Re: [Freeswitch-users] Remote-Party-ID issue and call pickup information

[SP]

before playing with mod_sofia, did you try the sip_cid_type variable?

http://wiki.freeswitch.org/wiki/Variable_sip_cid_type

On Sun, Nov 8, 2009 at 02:46, Yehavi Bourvine  wrote:
> Hello,
>
>   While trying to display the called party name  on SNOM phones I've found
> that the field sent to the phone needs to be changed slightly in order to
> make SNOM work: Insetad of sending P-Assterted-Identity SNOM expects
> Remote-Party-ID. I changed it in mod_sofia and now SNOM, Polycom and Cisco
> work ok. Just wanted to let the developers know...
>
>   And now a question: We have SNOM phones monitoring other extensions (BLF
> feature). When a call comes in, the monitoring phones get notification, but
> the name field (identity display) contains the calling extension number and
> not its display name. Can this be fixed?
>
>     Thanks! __Yehavi:
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Re: [Freeswitch-users] 3rd Party Dial Plan Tool

[SP]

You could also try consult...@freeswitch.org

On Wed, Oct 21, 2009 at 14:00, Brian West  wrote:
> none specifically exist... good ole trusty editor?
>
> /b
> PS: http://www.cudatel.com
>
>
> On Oct 21, 2009, at 1:37 PM, Jerry Richards wrote:
>
>>
>> Can anyone recommend a good 3rd party dialplan tool that will work
>> with
>> Freeswitch?
>>
>> Best Regards,
>> Jerry
>
>
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Re: [Freeswitch-users] Detecting a fax

[SP]

try using the hanup hook



On Mon, Oct 5, 2009 at 23:15, Mark Campbell-Smith
 wrote:
> Thanks for your help Mike and Tihomir.
>
> A little more playing around and I found that having  application="fax_detect"/> as well as  application="tone_detect" data="fax 1100 r +5000 transfer fax XML
> features" /> do not work together.
>
> Simply by removing fax_detect, the fax is detected beautifully.
>
> My problem now is trying to email the fax.  I followed the
> instructions on the wiki at http://wiki.freeswitch.org/wiki/Mod_fax,
> but the dialplan  is not executed
> after the rxfax command.  I know the script works because if I put the
> system command in another part of the dialplan and hard code the
> filename to attach, then the email is sent.
>
>    
>      
>        
>        
>         data="//usr//local//freeswitch//storage//${caller_id_number}-${uuid}.rxfax.tiff"/>
>         data="/usr/local/freeswitch/scripts/emailfax.sh
> /usr/local/freeswitch/storage/${caller_id_number}-${uuid}.rxfax.tiff"/>
>        
>      
>    
>
> ideas?
> Thanks!
>
> On Tue, Oct 6, 2009 at 1:32 AM, Tihomir Culjaga  wrote:
>> hi Mark,
>>
>> This is an inbound call leg and media channel (so far)  is open in reverse
>> direction only (application ringback). I'm afraid you have to answer the
>> call to be able to "hear" the fax tone.
>>
>> T.
>>
>>
>>
>> On Mon, Oct 5, 2009 at 2:32 PM, Michael Jerris  wrote:
>>>
>>> Fax tones are not played by the remote machine until after answer, the
>>> tone_detect application starts a media bug that listens for the tone,
>>> can you confirm the tone is happening at all.  Maybe the issue here is
>>> the timeout, try making that longer, or doing the tone_detect in
>>> execute_on_answer
>>>
>>> Mike
>>>
>>> On Oct 5, 2009, at 6:28 AM, Mark Campbell-Smith wrote:
>>>
>>> > Thanks for the response Mike,
>>> >
>>> > I read that page and this one (among others)
>>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
>>> > I'm still lost.  This is an extract of my dialplan
>>> >
>>> >    
>>> >      
>>> >        
>>> >        
>>> >        
>>> >        
>>> >        
>>> >        
>>> >        
>>> >        
>>> >
>>> > I would assume that on detecting a fax, the dialplan 'fax' is called
>>> > in context features.  This never happens.
>>> >
>>> > When is the fax tone detected?   Is it while the call is ringing or
>>> > can it be detected after the call is answered?  My goal is to be able
>>> > to have the same extension for a voice and fax call.  i assume that
>>> > the fax 'tones' are standardised and the ones on the wiki are correct?
>>> > Also, I guess this doesn't work with media bypass (which I don't
>>> > use).
>>> >
>>> > Thanks!
>>> >
>>> >
>>> > On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris 
>>> > wrote:
>>> >> check out
>>> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
>>> >>
>>> >> Note, you can't just have tone_detect as your last iten in the
>>> >> dialplan as the call will just get hung up.
>>> >>
>>> >> Mike
>>> >>
>>> >> On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
>>> >>
>>> >>> Hi
>>> >>>
>>> >>> I was hoping someone could help me to setup the fax detection / tone
>>> >>> detection application.
>>> >>>
>>> >>> I want to be able to transfer an incoming fax to a specific
>>> >>> extension.
>>> >>> In my default.xml file, I have the following (extracted):
>>> >>>
>>> >>>    
>>> >>>      
>>> >>>        
>>> >>>        
>>> >>>
>>> >>> I can't get the fax to be detected and transferred.  Is there any
>>> >>> way
>>> >>> this can be done?
>>> >>>
>>> >>> Thanks!
>>>
>>>
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>>
>>
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>
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Re: [Freeswitch-users] How to limit the number of incoming+outgoing calls via specific gateway?

[SP]

did you use the application limit on the inbound call?  You'll need to
in order to account for it.

On Mon, Oct 5, 2009 at 15:39, Dmitry Bely  wrote:
> On Tue, Oct 6, 2009 at 12:19 AM, SP  wrote:
>> Direction doesn't matter, it uses realm's and a few other vars.  Use
>> the same vars for both directions.
>
> Unfortunately it does.
>
>      
>
> generates limit_exceeded for the second outbound call, but if an
> incoming call is active FreeSWITCH still tries to use this gateway.
>
>> On Mon, Oct 5, 2009 at 14:57, Dmitry Bely  wrote:
>>> On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris  wrote:
>>>> http://wiki.freeswitch.org/wiki/Mod_limit
>>>
>>> Of course I've read that. But it only affects the number of outgoing
>>> calls (at least for gateways - chapter "Using mod_limit with an
>>> outbound gateway"). But I would like to limit the number of all calls
>>> (incoming+outgoing) via specific gateway. Any idea?
>>>
>>> - Dmitry Bely
>>>
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>>>
>>
>>
>>
>> --
>> Shannon
>>
>> ___
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>
>  Dmitry Bely
>
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Re: [Freeswitch-users] How to limit the number of incoming+outgoing calls via specific gateway?

[SP]

Direction doesn't matter, it uses realm's and a few other vars.  Use
the same vars for both directions.

On Mon, Oct 5, 2009 at 14:57, Dmitry Bely  wrote:
> On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris  wrote:
>> http://wiki.freeswitch.org/wiki/Mod_limit
>
> Of course I've read that. But it only affects the number of outgoing
> calls (at least for gateways - chapter "Using mod_limit with an
> outbound gateway"). But I would like to limit the number of all calls
> (incoming+outgoing) via specific gateway. Any idea?
>
> - Dmitry Bely
>
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Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey

[SP]

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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Re: [Freeswitch-users] Bind extention to a different Dialplan and cdr php?

[SP]

you can use different php includes

On Sunday, September 20, 2009, João Mesquita  wrote:
> Frank,
>
> That kind of logic needs to be performed at your application, if I am not 
> wrong. I will do some testing here, but I think that mod_xml_cURL sends 
> purpose and other information regarding the type of data it is requesting so 
> you can decide on your application exactly what to do.
>
> Regards,
>
> jmesquita
>
> On Sun, Sep 20, 2009 at 1:10 PM, Frank @ Impact 
> > 
> wrote:
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> Currently all incoming calls to my FS to
> all extensions are sent off by curl to a particular php
> script called dialplan.php.
>
> I would like to have certain extensions
> that are called to have their xml dialplan built by a
> curl to a different php script, say dialplan2.php.
>
>
>
> Is there a way to have certain extensions
> get their dialplan by calling a different php script other than the default?
>
>
>
> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael 
> Collins
> Sent: Tuesday, September 08, 2009
> 3:37 AM
> To:
> freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users]
> Bind extention to a different Dialplan andcdr php?
>
>
>
>
>
>
>
> On Sun, Sep 6, 2009 at 8:43 AM, Frank @ Impact 
> > 
> wrote:
>
>
>
>
>
>
> Is there a way to bind a particular extension to a
> different dialplan php and a different cdr php script than the default one?
>
>
>
>
>
>
>
>
>
> Could you re-phrase this question with a bit more
> detail? Thanks.
> -MC
>
>
>
>
>
>
>
>
>
>
>
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Re: [Freeswitch-users] Zaptel Analog FXO Outbound Dialing

[SP]

I have a Sangoma U100 usbfxo setup with wanpipe and zaptel.  Audio is immediate.

Obviously your milage may vary.  :)

On Sat, Sep 12, 2009 at 21:06, Michael S  Collins  wrote:
> Gang,
>
> Does anyone have a working analog setup? I've reproduced symptoms on
> two different systems, one with the Rhino 8 port card and one with the
> Nxtvox TDM400 clone. Symptoms are identical when dialing out on the
> FXO port. The outbound dialing established a connection with the
> called party but audio does not flow to the endpoint until approx 30
> seconds after the call is answered. Digilord is having the same issue.
> Anyway, before I open a bug report I wanted some feedback from those
> who are using FXO analog ports.
>
> Please let me know of your experiences. Thanks!
> -MC
>
> Sent from my iPhone
>
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Re: [Freeswitch-users] caller id on origination does not work

[SP]

You can try playing with this in your gateway profile
  

Not sure what it'll do to your registration, give it a try

On Tue, Aug 25, 2009 at 21:51, Shameem Shiek  wrote:

> I am setting the caller id like this in my ESL script:
>
> @con.sendRecv("api originate
> {origination_caller_id_number=15103245678}sofia/gateway/junctionnetworks/1...@number}
> &park()")
>
> And the caller id comes out as all zeroes.  The sip trace shows the "from"
> as shown in the sofia status command.  This is output of sofia gateway
> status and my gateway is under
> /usr/local/freeswitch/conf/sip_profiles/external/junctionnetworks.xml .
> Also, adding the gateway under /conf/directory/default did not work as the
> wiki suugests. What I need to do for the caller_id to come through?
>
>
> sofia status gateway junctionnetworks
>
> =
> Namejunctionnetworks
> Scheme  Digest
> Realm   jnctn.net
> Usernameusername
> Passwordyes
> From
> ;transport=udp>
> Contact 
> Exten   899xxx
> To  sip:usern...@sip.jnctn.net 
> Proxy   sip:sip.jnctn.net
> Context public
> Expires 3600
> Freq3600
> Ping0
> PingFreq0
> State   REGED
> Status  UP
> CallsIN 0
> CallsOUT1
>
>
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Re: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM

[SP]

Don't forget to press tab at the asterisk console!  :)

On Sun, Aug 23, 2009 at 18:23, Raymond Chandler
wrote:

> Could you load freeswitch with a couple hundred calls then run the test
> again.. and do the same to asterisk and see how the numbers stack up then?
> I'm just curious to see what happens at that point.
> -Ray
>
> On Aug 21, 2009, at 3:15 PM, Rogelio Perez wrote:
>
> Hi Everyone,
>
> I'm working on a PBX project for the 
> Sheevaplug
>  ARM
> based computer, with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU.
> So far I've found a big difference between Freeswitch and Asterisk
> performance times.
> This is a comparison of the time it takes them to perform different
> actions:
>
> startup Freeswitch: 3 min.
> startup Asterisk: 2 sec.
>
> call extension Freeswitch: 6 sec.
> call extension Asterisk: 0 sec.
>
> shutdown Freeswitch: 6.5 sec
>
> shutdown Asterisk: 0 sec.
>
>
> reload config Freeswitch: 1 sec.
> reload config Asterisk: 1 sec.
>
>
> Both were built from sources natively (no cross-compiling), and they use
> the default startup configurations.
> I have managed to lower the Freeswitch times by disabling most of the
> modules and recompiling, but it is still far away from Asterisk (i.e. FS
> startup time 2.5 min).
>
> 1. Is there any way to further improve Freeswitch performance for the ARM
> architecture?
> 2. Can this be related to the lack of a FPU (the Sheevalug emulates the
> floating point operations).
> 3. On the startup I see this error repeated many times: [ERR]
> switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be related?
>
> Thanks,
> Rogelio Perez
>
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Re: [Freeswitch-users] freepbx for freeswitch

[SP]

would it be pimped if it didn't?

On Wed, Aug 5, 2009 at 16:49, Diego Viola wrote:
> Yay for FreePBX =D
>
> Does Cudatel uses FreeSWITCH as the engine?
>
> On Wed, Aug 5, 2009 at 5:35 PM, Michael Collins wrote:
>> Of course, don't forget this:
>> http://www.cudatel.com
>>
>> For those who want a commercial solution built upon FreeSWITCH: You've got
>> it!
>> -MC
>>
>> On Wed, Aug 5, 2009 at 4:24 PM, Nik Middleton
>>  wrote:
>>>
>>> I'd heard rumours that this was going to happen and it's great news and
>>> good news for FS as well.  With a user friendly front end, FS is sure to
>>> fly.  I have no doubt that this will be the first of many.
>>>
>>> Regards,
>>>
>>> -Original Message-
>>> From: freeswitch-users-boun...@lists.freeswitch.org
>>> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
>>> Giovanni Maruzzelli
>>> Sent: 05 August 2009 22:00
>>> To: freeswitch-users@lists.freeswitch.org;
>>> freeswitch-...@lists.freeswitch.org
>>> Subject: [Freeswitch-users] freepbx for freeswitch
>>>
>>> Yay!
>>>
>>> http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-fut
>>> ure
>>>
>>> Darren Schreiber has made the announcement and is doinng a
>>> presentation of FreePBX V3 right now at www.cluecon.com.
>>>
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>>
>>
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Re: [Freeswitch-users] q: fresh installation: Couldnt open /usr/local/freeswitch/conf/freeswitch.xml

[SP]

2009-08-05 14:41:16.910763 [ERR] switch_xml.c:1282 Couldnt open
/usr/local/freeswitch/conf/freeswitch.xml *(No such file or directory)*

On Wed, Aug 5, 2009 at 13:46, tom wrote:
> hi just installed freeswitch via svn.
> - bootstrap
> - configure
> - make install
> - ./freeswitch
>
> gives me:
> acerdebian:/usr/local/freeswitch/bin# ./freeswitch
> Error: stacksize 4194303 is too large: run ulimit -s 240 or run
./freeswitch
> -waste.
> auto-adjusting stack size for optimal performance
> 2009-08-05 14:41:16.908170 [INFO] switch_event.c:565 Activate Eventing
> Engine.
> 2009-08-05 14:41:16.910182 [DEBUG] switch_event.c:553 Create event
dispatch
> thread 0
> 2009-08-05 14:41:16.910763 [ERR] switch_xml.c:1282 Couldnt open
> /usr/local/freeswitch/conf/freeswitch.xml (No such file or directory)
> Cannot Initialize [Cannot Open log directory or XML Root!]
>
>
> bump - help
>
> thx
>
>
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Re: [Freeswitch-users] Dialplan with lua - error missing closing angle bracket?

[SP]

check your whitespace

On Wed, Jul 15, 2009 at 13:33, Lars Zeb  wrote:

>  I copied the action element from http://wiki.freeswitch.org/wiki/Lua,
> “Sample Dialplan”.
>
>
>
> When I try to reloadxml, the cli tells me that there is a missing right
> angle bracket.
>
> +OK [[error near line 3130]: missing >]
>
>
>
> Do the docs need updating or have I totally blown it?
>
>
>
> Thanks, Lars
>
>
>
> 
>
>
>
>  
>
>
>
>  
>
>
>
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Re: [Freeswitch-users] Freeswitch architecture

[SP]

http://cluecon.com/node/3

On Wed, Jul 8, 2009 at 07:42, Maxim Tsvetov  wrote:

> Hi All
>
> Where can I get information about internal Freeswitch architecture:
>   1) how modules interoperates with each other (maybe using corba or com
> objects or something else)
>   2) how core interoperates with other modules
>   3) how javascript function is translated to internal commands.
>
> In addition if you cand send me some schemas of Freeswitch architecture
> that will be great.
>
> Regards,
> Maxim Tsvetov
>
>
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Re: [Freeswitch-users] Clustering Freeswitch

[SP]

openser/kamailio/ser/opensips you pick a name

DNS SRV

On Tue, Jul 7, 2009 at 16:29, Andres Gomez  wrote:

> Hello to all
>
> Exist any solution to clustering. Any load balancing appliance o heartbeat
> test?.
>
> Regards
>
> Andres Gomez.
>
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Re: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729?

[SP]

I'm now imagining bkw with his iphone in front of the mirror.

   - touchscreen - check
   - video - check
   - nice looking - you know I am


On Mon, Jun 29, 2009 at 20:58, Craig Askings wrote:

> Well Snom phones use linux. I wonder if the the Chinese factory that
> pumps them out does grey market versions. Unfortunately that doesn't
> meed the video, touch screen or even the nice looking requirements.
>
> Craig.
>
> p.s. Above thoughts are my own not my employers.
>
> 2009/6/30 Brian West :
> > Now who can we con into building an open platform thats nothing more
> > than a linux box shaped like a nice looking phone?  :P
> >
> > Video and Touch Screen are a requirement :)
> >
> > /b
> >
> > On Jun 29, 2009, at 8:41 PM, Jason White wrote:
> >
>
> --
> Craig Askings
>
> Network Engineer | Over the Wire Pty Ltd
> cr...@overthewire.com.au | www.overthewire.com.au
> Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365
>
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Re: [Freeswitch-users] Is there any license G729?

[SP]

FreeSWITCH runs on a Mac and can be configured as a UA... also supports WB
and UWB!

On Mon, Jun 29, 2009 at 20:13, seven  wrote:

> >
>
> Mike, but what UA are you using? I'm using X-lite and Zoiper on Mac/
> Linux/Windows, and I'd like to know some UAs supporting WB and UWB
> codecs.
>
> Thanks.
>
> > Well said! Like Steve has pointed out in the past: G729/G729A is a
> > race to the bottom. After using WB and UWB codecs all day every day
> > for the past 6 months I just can't live with G729 or even GSM for
> > that matter.  However, to each his own.
> > -MC
>
>
>
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Re: [Freeswitch-users] CDR loss possibility if FS freezes?

[SP]

clouds

On Thu, Jun 25, 2009 at 22:56, Brian West  wrote:

> Is solar an option? ;)
>
> /b
>
> On Jun 25, 2009, at 10:48 PM, Harmeet Singh wrote:
>
> > Just make sure the power is always there! ...I know that some parts
> > of the world this is not easy to achieve.
> >
> > Harmeet
>
>
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Re: [Freeswitch-users] mod_voicemail accounts on-the-fly

[SP]

Unless you use mod_xml_curl

On Sat, Jun 13, 2009 at 13:11, Adam Wilt  wrote:

> I'm sorry, I said dialplan, but I was talking about the directory.
> So the directory file must be edited every time a new mailbox is created?
>
>
>
>>
>> Message: 7
>> Date: Sat, 13 Jun 2009 12:58:57 -0400
>> From: Brian West 
>> Subject: Re: [Freeswitch-users] mod_voicemail accounts on-the-fly
>> To: freeswitch-users@lists.freeswitch.org
>> Message-ID: 
>> Content-Type: text/plain; charset="us-ascii"
>>
>>
>> Why would have to touch the dialplan?  Just one regex to catch it
>> all.. and send it to voicemail.  You still have to add users to the
>> directory for them to have a mailbox anyway.
>>
>> /b
>>
>> On Jun 13, 2009, at 12:18 PM, Adam Wilt wrote:
>>
>> > Hi, I have a question about mod_voicemail. I would like to use it
>> > independently of the dialplan and I was hoping to be able to add new
>> > voicemail accounts on-the-fly, without having to update the
>> > FreeSWITCH configuration files. But it seems to be forcing me to
>> > manually add each user to the dialplan.
>> >
>> > Does anybody know if mod_voicemail was made to work the way I'm
>> > trying to use it, and the best way to approach this?
>> >
>> > Thanks,
>> > Adam
>> >
>>
>> Brian West
>> br...@freeswitch.org
>>
>> -- Meet us at ClueCon!  http://www.cluecon.com
>>
>>
>>
>>
>> -- next part --
>> An HTML attachment was scrubbed...
>> URL:
>> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090613/bad009e9/attachment.html
>>
>> --
>>
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>>
>> End of Freeswitch-users Digest, Vol 36, Issue 129
>> *
>>
>
>
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Re: [Freeswitch-users] Cool names for my VoIP company

[SP]

Dasbus

On Thu, May 21, 2009 at 23:26, Diego Viola  wrote:
> Hey guys,
>
> I'm about to start my own ITSP with FreeSWITCH, and I'm looking some
> cool names for my VoIP company, if you know some please tell me :)
>
> Diego
>
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Re: [Freeswitch-users] As seen on DIGG

[SP]

really?? is that your story you're sticking to??


On Tue, May 19, 2009 at 16:37, Brian West  wrote:
> Well you can rest now... they were damaged in a washing machine accident!
>  :(
> /b
> On May 19, 2009, at 4:08 PM, Michael Collins wrote:
>
> i was kinda impressed with the dr. pepper pj's. :)
>
> On Tue, May 19, 2009 at 1:44 PM, Brian West  wrote:
>>
>> HAHAHAHAHAHAHAHAHAH
>> /b
>
> Brian West
> br...@freeswitch.org
> -- Meet us at ClueCon!  http://www.cluecon.com
>
>
>
>



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Re: [Freeswitch-users] As seen on DIGG

[SP]

I think we'd like to see the outfit!

On Tue, May 19, 2009 at 14:53, Brian West  wrote:
> FreeSWITCHers,
> Is it too much to ask for the community to help promote the project when
> possible?  Now I normally don't do this but I could put on a cheer leader
> outfit and grab some pomp pomps and get you guys and gals to click  well
> maybe if you click I won't put on the outfit!  :P  Not sure which is more
> beneficial to you!
> Thanks,
> Brian West
> On May 19, 2009, at 12:28 PM, Anthony Minessale wrote:
>
> please find it in your hearts to digg this story.
>
> http://digg.com/search?s=skype+switch
>
>
> Brian West
> br...@freeswitch.org
> -- Meet us at ClueCon!  http://www.cluecon.com
>
>
>
>
>
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Re: [Freeswitch-users] SNMP or Cacti to graph perrformance?

[SP]

Connect to the socket and subscribe to the heartbeat.  The heartbeat
event fires every 20 seconds and contains the session count.

On Tue, Apr 28, 2009 at 21:19, Ron McCarthy  wrote:
> will hitting it every 10 or 20 seconds hurt performance / eat a lot of CPU?
>
> Only reason id like to hit every second so I could get some sort of attempt
> on the call setups and call transactions.
>
> Has anyone even done anything like this before?
>
> Thanks
>
> On Tue, Apr 28, 2009 at 7:16 PM, Brian West  wrote:
>>
>> every second is way too high of a resolution.  10 seconds 20 seconds are
>> more sane!
>> /b
>> On Apr 28, 2009, at 9:11 PM, Ron McCarthy wrote:
>>
>> lso, if I have to make some custom plugin that montiors output the of show
>> channels /show calls via one of the API's, anyway we can get a realtime
>> stream so I don't have to query it every second to get correct data?
>>
>> Brian West
>> br...@freeswitch.org
>> -- Meet us at ClueCon!  http://www.cluecon.com
>>
>>
>>
>>
>>
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Re: [Freeswitch-users] I can not receive inbound calls thru FreeSWITCH

[SP]

most likely your hunting for 'winday' which is not found by any condition.

2009-04-15 22:57:42 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing
Cell Phone   FL->winday in context public

On Thu, Apr 16, 2009 at 00:06, winday  wrote:
>
> I'm a fresh user of FreeSWITCH. I just configed the basic things. Right now,
> all outbound calls are ok. I can call any number I want. But when I use my
> own mobile phone to call the phone number(I got from my sip provider) bind
> with my sip account, I can not receive the call.
>
> I configed that the inbound calls are all forwarded to extension 1001. But I
> got the following message:
>
> 2009-04-15 22:57:42 [NOTICE] switch_channel.c:592 switch_channel_set_name()
> New Channel sofia/external/352...@8.17.37.23
> [5a5e70c2-2a32-11de-865a-c350fb010f87]
> 2009-04-15 22:57:42 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing
> Cell Phone   FL->winday in context public
> 2009-04-15 22:57:42 [NOTICE] switch_core_state_machine.c:172
> switch_core_standard_on_execute() Hangup
> sofia/external/352...@8.17.37.23 [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-04-15 22:57:42 [NOTICE] switch_core_session.c:1019
> switch_core_session_thread() Session 1
> (sofia/external/3528713...@8.17.37.23) Ended
> 2009-04-15 22:57:42 [NOTICE] switch_core_session.c:1021
> switch_core_session_thread() Close Channel
> sofia/external/3528713...@8.17.37.23 [CS_DONE]
>
> I don't know why.
>
> The public.xml configration is:
>
>      
>      
>    
>    
>   
>
> The default.xml configration is:
>
>  
>
>    
>     
>    
>     
>      data="/usr/local/freeswitch/scripts/answermachine.js"/>
>    
>   
>
>
>
> --
> View this message in context: 
> http://www.nabble.com/I-can-not-receive-inbound-calls-thru-FreeSWITCH-tp23070965p23070965.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] Greedy codecs on the outbound side

[SP]

http://wiki.freeswitch.org/wiki/Codec_negotiation

http://lmgtfy.com/?q=freeswitch+late+negotiation


On Tue, Mar 31, 2009 at 17:04, Josh Forman  wrote:
> I know that on a sofia profile you can set it to be greedy or generous
> in codec negotiation on the inbound side.
>
> How does freeswitch negotate codecs on the outbound side and what ways
> are available to affect how this takes place?
> Is there a way to make the codec preferences of the inbound side the
> codec preference of the outbound side?
> If not is there a way to set outbound negotations as greedy vs generous?
> If we disable transcoding in freeswitch would that in effect make the
> codec negotated on the inbound side be our most prefered codec on the
> outbound side's codec negotation?
>
> Thank you for your help,
>
> Josh
>
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Re: [Freeswitch-users] Good time, people!

[SP]

Are you setting a Content Length header in the HTTP response??

2009/3/19 Леша... :
> Thats the thing!!
> Im using tcpdump to watch for packets - and i dont see any mistakes =\
> The xml i sent is allright, its like a piece from my static worked xml 
> dialplan.
>
> But I cant understand why does FS recognise it as a 130+ mb file :D
> Maybe i need to update s0mthing?)
>
> Brian West пишет:
>> Any reason you're feeding it a 130+ meg file over XML_CURL?
>>
>> /b
>>
>> On Mar 19, 2009, at 6:05 AM, Леша... wrote:
>>
>>> Hello!
>>>
>>> Has anybody faced such a problem with xml_curl?
>>> 2009-03-18 23:24:41 [INFO] mod_dialplan_xml.c:252 dialplan_hunt()
>>> Processing 1000->** in context default
>>> 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:114 file_callback()
>>> Oversized file detected [136089828 bytes]
>>> 2009-03-18 23:24:43 [ERR] mod_xml_curl.c:241 xml_url_fetch() Error
>>> encountered!
>>>
>>> Tcpdump log tells that transaction is allright (xml dialplan is fine
>>> and etc)
>>> But FS says its oversized =\ what could be wrong?
>>>
>>> My trunk is :
>>> FreeSWITCH Version 1.0.trunk (12573).
>>
>>
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>
>
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Shannon

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Re: [Freeswitch-users] FreeSWITCH forum community opened today!

[SP]

Another jab, in fun:
hey look, that first link looks like a forum?? wow :D

On Thu, Mar 12, 2009 at 13:55, Harry FSwitch  wrote:
> Greetings,
>
> Last week I submitted this post to the mailing list...
>
> http://www.nabble.com/Please-end-the-torment-td2235.html
>



-- 
Shannon

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Re: [Freeswitch-users] STUN error

[SP]

On Wed, Mar 11, 2009 at 11:32, Kristian Kielhofner
 wrote:
> On Mon, Mar 9, 2009 at 11:09 PM, Will Smith  wrote:
>> Thank you Brian, it works like a champ.
>>
>> Yes, west philadelfia born and raised?
>
> On a playground is where I spent most of my days...

Chilling out, maxing, relaxing all cool

>
> --
> Kristian Kielhofner
> http://blog.krisk.org
> http://www.submityoursip.com
> http://www.astlinux.org
> http://www.star2star.com
>
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Re: [Freeswitch-users] Adding an info digit to sip from

[SP]

On Wed, Feb 25, 2009 at 13:51, Josh Forman  wrote:
> The problem here is that what you are showing me produces:
>
> From: "1004" ;tag=1SparjgraS69m
> To: 
>
> when what I need to output would look like this:
>
> From: "1004" ;tag=1SparjgraS69m
> To: 
>
> with the "this=rocks" in the FROM field, not the TO field.
> I know that you can change parts of the from field by setting
> effective_caller_id_name and effective_caller_id_number, but I don't
> know how I would add that bit of data to the end of the SIP URI inside
> the < >
> Is there a variable that I could set or perhaps some method similar to
> overwriting the To header shown at 
> http://wiki.freeswitch.org/wiki/Sofia#Modifying_the_To
> :_header that can be used to accomplish this?
>
> Thanks
> Josh
>
> On Feb 25, 2009, at 2:35 PM, freeswitch-users-requ...@lists.freeswitch.org
>  wrote:
>
>>> Message: 3
>>> Date: Wed, 25 Feb 2009 09:29:26 -0600
>>> From: Brian West 
>>> Subject: Re: [Freeswitch-users] Adding an info digit to sip from
>>>       header
>>> To: freeswitch-users@lists.freeswitch.org
>>> Message-ID: <0e1e9f62-64c5-4b70-9c67-c7b5728db...@freeswitch.org>
>>> Content-Type: text/plain; charset="us-ascii"
>>>
>>> You can do something like this "sofia/blah/somenum...@someip:
>>> 5060;this=rocks"
>>>
>>> /b
>>>
>>> On Feb 25, 2009, at 9:22 AM, Josh Forman wrote:
>>>
 I'm trying to edit the sip headers to make the from field look like
 this:

 From: ;tag=gK0a00d6ea.

 I know that to read that data on an incoming sip message it is in $
 {sip_from_params}, but how can I add the ;isup-oli=27 part on an
 outgoing message?

 Thanks

 Josh
>>
>>
>>
>> --
>>
>> Message: 3
>> Date: Wed, 25 Feb 2009 10:53:39 -0600
>> From: Brian West 
>> Subject: Re: [Freeswitch-users] Adding an info digit to sip from
>>        header
>> To: freeswitch-users@lists.freeswitch.org
>> Message-ID: 
>> Content-Type: text/plain; charset=US-ASCII; format=flowed
>>
>> It will actually add it to both places
>>
>> INVITE sip:1...@conference.freeswitch.org;this=rocks SIP/2.0
>> Via: SIP/2.0/UDP 99.185.85.3;rport;branch=z9hG4bK0Kaa1322U42eK
>> Max-Forwards: 69
>> From: "1004" ;tag=1SparjgraS69m
>> To: 
>>
>> I verified it does indeed add it in both places.
>>
>> /b
>>
>>
>>
>> On Feb 25, 2009, at 10:34 AM, Josh Forman wrote:
>>
>>> Which variable would I need to set via the dialplan to do this
>>> though?  Your example looks like it would be the dialstring for the
>>> bridge application but if that works it would probably be added to
>>> the
>>> To header instead of the From, right?  I can't be sure since nothing
>>> I've tried has had any affect.
>>> Between looking at the wiki and random experimenting I haven't found
>>> anything that works thus far.
>>
>
>
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Re: [Freeswitch-users] Happy B-Day BKW!

[SP]

We could ban him from IRC for the day... that would be a gift :)

On Tue, Feb 17, 2009 at 19:48, Ken Rice  wrote:
> Ok Tomorrow is BKW's b-day...
>
> Lets as a community see if we cant pull together and get him something nice
> off his Amazon wish list...
>
> Visit his wish list here...
> http://www.amazon.com/gp/registry/wishlist/1BWDJUX5LYQE0
>
> If you want to chip in contact me on IRC (I'm SwK) or send something via the
> donate link at http://www.tollfreegateway.com/bailmeout.html be sure to put
> "For BKW" in the comment or drop me an off list email...
>
> If you want to send something via paypal direct to brian his paypal ID is
> br...@freeswitch.org
>
> Come on guys he works day and night on FS to help bring us arguably the best
> OpenSource Telephony platform out there!!
>
> Happy B'Day Brian!!
>
> Ken
>
>
>
>
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Re: [Freeswitch-users] Dynamic Dialplan

[SP]

Everything you can do in a static dialplan you can do via curl as
well.  Multiple extensions, search/conditions are allowed.  Don't sell
the curl short, it's very powerful and can get the ball rolling.

On Sun, Feb 8, 2009 at 21:21, Doug Blacksone  wrote:
> Hi,
>
> Right now, I am working on getting freeswitch configured for our call-center
> with more than 1000 agents.  There are several areas where we need the
> dialplan to be configurable based on some user detail in the database.
> Therefore, the dialplan needs to be some-what dynamic based-on inputs from
> the database.
>
> I would like to know from other implementation as to the most scalable way
> of doing high performance dynamic dialplan that is super scalable.
>
> There are three ways I can think of:
>
> 1. Static dialplan using customized freeswitch mod to access postgres for
> data
> pros: best performance
> cons: harder to program
>
> 2. Static dialplan using lua to access postgres for data
> pros: easy to program, maybe-performance is better than curl
> cons: need to search through all the extensions to find one dialplan,
> performance is slower than the first one.
>
> 3. curl-based dialplan using Java Servlet and HTTP
> pros: easy to program, freeswitch only gets one extension and no extension
> search
> cons: performance is slow than the other two
>
> Is this a correct analysis?
> If from a pure performance's perspective, how much performance can a
> customized mod gains in comparison to lua?
>
> For a production system that needs to be highly scalable, what do you
> recommend?
>
>
> Thank you very much for any input to our critical design decision.
>
> Doug

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