Re: [Freeswitch-users] Language settings for demo IVR

2009-12-15 Thread Dmitry Bely 
On Tue, Dec 15, 2009 at 12:15 AM, Michael Collins  wrote:
>
>
> On Mon, Dec 14, 2009 at 11:47 AM, Dmitry Bely  wrote:
>>
>> I'm playing with demo IVR from FreeSwitch distribution and have a
>> problem with language settings. I would like to use Russian as a
>> default language for voice messages so I set in vars.xml
>>
>>  
>>
>> and installed Russian sound files. It works almost correctly: all
>> phrases are played in Russian, but not explicitly specified .wav
>> files; say for
>>
>>    >        invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>>
>> I have
>>
>> 2009-12-14 22:17:57.506305 [ERR] mod_sndfile.c:194 Error Opening File
>>
>> [/opt/freeswitch/sounds/en/us/callie/ivr/ivr-that_was_an_invalid_entry.wav]
>> [System error : No such file or directory.]
>>
>> How to fix this and make it use the correct language?
>>
> What about this in vars.xml?
>
>  data="sound_prefix=$${base_dir}/sounds/en/us/callie"/>

Yes, that does the job. Thank you! But it looks a bit inconsistent.
Path to sound files is also set in $${base_dir}/conf/lang/ru/ru.xml.
Why duplicate the settings? And another problem is that you cannot
easily switch the language for your voice menu.

- Dmitry Bely

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[Freeswitch-users] Language settings for demo IVR

2009-12-14 Thread Dmitry Bely 
I'm playing with demo IVR from FreeSwitch distribution and have a
problem with language settings. I would like to use Russian as a
default language for voice messages so I set in vars.xml

 

and installed Russian sound files. It works almost correctly: all
phrases are played in Russian, but not explicitly specified .wav
files; say for

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Re: [Freeswitch-users] Status of ubuntu/debian packages.

2009-10-20 Thread Dmitry Bely 
William,

Where is mod_skypiax.so? Isn't freeswitch-skypiax package expected to
contain it?

- Dmitry Bely

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Re: [Freeswitch-users] Status of ubuntu/debian packages.

2009-10-19 Thread Dmitry Bely 
Can you build FreeSWITCH .deb package with tls transport support?
Anyway it already depends on openssl package.

Another problem:

2009-10-20 00:28:50.728557 [CRIT] switch_loadable_module.c:871 Error
Loading module /opt/freeswitch/mod/mod_cluechoo.so
**/opt/freeswitch/mod/mod_cluechoo.so: cannot open shared object file:
No such file or directory**
2009-10-20 00:28:50.728604 [CRIT] switch_loadable_module.c:871 Error
Loading module /opt/freeswitch/mod/mod_valet_parking.so
**/opt/freeswitch/mod/mod_valet_parking.so: cannot open shared object
file: No such file or directory**

Modules mod_cluechoo & mod_valet_parking are referenced in the config
but not present in the package.

- Dmitry Bely

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Re: [Freeswitch-users] Status of ubuntu/debian packages.

2009-10-19 Thread Dmitry Bely 
On Mon, Oct 19, 2009 at 10:45 PM, Patrick List
 wrote:
> On 10/19/2009 03:07 PM, Frank Carmickle wrote:
>> On Sun, Oct 18, Patrick List wrote:
>>> so there was no chance of your customized configs getting overwritten.
>>> Example:
>>> /opt/freeswitch/etc/freeswitch-$svn_version-samples/
>>>
>>> Maybe the maintainer of the debs can do something similar.
>>
>> We are planning on putting all of the different configs in 
>> /usr/share/doc/freeswitch/examples.  Now if I could find the time to get 
>> this done.
>
> That would obviously work. The advantage of sticking example configs in
> their own folder designating the release (svn rev) they belong to is
> that you can do a diff between two folders to see what has changed.
> Maybe that's useful to some. Just a thought though.

For most Debian packages dpkg just asks the user what he/she would
like to do with the newer config file from .deb: keep the old file,
replace it with the new file, diff old with new files etc. Not an
option here?

- Dmitry Bely

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Re: [Freeswitch-users] Status of ubuntu/debian packages.

2009-10-18 Thread Dmitry Bely 
On Sun, Oct 18, 2009 at 8:40 PM, Frank Carmickle  wrote:
> On Sun, Oct 18, Dmitry Bely wrote:
>> William,
>>
>> When I upgrade freeswitch-config package, all my customized config
>> files are silently overwritten with the default ones from .deb
>> package. This bug is quite annoying.
>
> Very sorry.  I can try to include some code that detects if your moving from 
> the old versions that don't have a config package to the new stuff that does.

I did have the previous version of freeswitch-config installed but
nevertheless the files were lost.

>  My way around this is not to install the freeswitch-config package at all.  
> I extract it
> some where else and diff my way through it.

Yes, but the bug should be fixed.

- Dmitry Bely

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Re: [Freeswitch-users] Status of ubuntu/debian packages.

2009-10-18 Thread Dmitry Bely 
William,

When I upgrade freeswitch-config package, all my customized config
files are silently overwritten with the default ones from .deb
package. This bug is quite annoying.

- Dmitry Bely

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Re: [Freeswitch-users] How to limit the number of incoming+outgoing calls via specific gateway?

2009-10-08 Thread Dmitry Bely 
On Tue, Oct 6, 2009 at 12:52 AM, SP  wrote:
> did you use the application limit on the inbound call?
> You'll need to in order to account for it.

Can you provide more detail? I have DID and gateway

  

  

  

  

(...)
  

  

(...)
  

  

that correspond to the same provider account. Then an incoming call is
active, outgoing one is not possible (provider will reject the it).
How to generate "limit_exceeded" then?

- Dmitry Bely

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Re: [Freeswitch-users] How to limit the number of incoming+outgoing calls via specific gateway?

2009-10-05 Thread Dmitry Bely 
On Tue, Oct 6, 2009 at 12:32 AM, M.Emran  wrote:
> Incoming Call-Limit http://www.howtonix.com/?p=89
>
> Outbound Call-Limit http://www.howtonix.com/?p=86

But what if I need to limit the total number of calls (in my case == 1)?

> On Tue, Oct 6, 2009 at 2:19 AM, SP  wrote:
>>
>> 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] How to limit the number of incoming+outgoing calls via specific gateway?

2009-10-05 Thread Dmitry Bely 
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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Re: [Freeswitch-users] How to limit the number of incoming+outgoing calls via specific gateway?

2009-10-05 Thread Dmitry Bely 
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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[Freeswitch-users] How to limit the number of incoming+outgoing calls via specific gateway?

2009-10-01 Thread Dmitry Bely 
My SIP provider allows only one call (incoming or outgoing) via one
SIP account. For FreeSWITCH I have configured it as public DID
extension and outgoing gateway. Now I would like to transfer to
another gw (or generate "limit exceded") when one tries to place an
outgoing call while incoming call is in progress. How tho do that?
Limiting the number of outgoing calls is easy (mod_limit), but how to
take into account incoming one?

- Dmitry Bely

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Re: [Freeswitch-users] FreeSWITCH Start up Script - Ubuntu Server

2009-09-28 Thread Dmitry Bely 
On Sun, Sep 27, 2009 at 5:42 PM, Aloysius Thevarajah Lloyd
 wrote:
> Hi All,
>
> I am trying to setup FreeSwitch on a Ubuntu Server.
>
> Where can I find the start up(boot time) script for FreeSwitch on a Ubuntu
> Server?

Why then not to use ubuntu .deb package? Init script is already there.

- Dmitry Bely

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Re: [Freeswitch-users] [Freeswitch-dev] Status of ubuntu/debian packages.

2009-09-25 Thread Dmitry Bely 
On Fri, Sep 25, 2009 at 9:52 AM, Jason White  wrote:
> [Just catching up on this thread.]
> William King  wrote:
>> I would be more than happy to share the code I use.
>>
>> Here is the git repo:
>>
>> http://git.home.quentustech.com:9191/cgit.cgi/freeswitch-ubuntu/
>
> When you would like your changes to the Debian build infrastructure in
> FreeSWITCH to be tested on Debian Sid, I'll gladly volunteer, excluding any
> modules that depend on proprietary software that I don't have and don't want,
> e.g., Skype.

If you don't use mod_skypiax then no Skype client software is
required. There is no dependency on the package level.

> I would also like to see these changes integrated into the FreeSWITCH
> repository to replace what is currently in the debian directory, once you have
> a version that is well tested.

It would be great if William's changes are committed to FreeSWITCH svn
repository. Anyway, debian folder in /trunk is outdated (has not been
updated since FreeSWITCH 1.0.3).

- Dmitry Bely

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Re: [Freeswitch-users] Status of ubuntu/debian packages.

2009-09-24 Thread Dmitry Bely 
On Fri, Sep 25, 2009 at 1:00 AM, Frank Carmickle  wrote:
> Hello
>
> On Fri, Sep 25, Dmitry Bely wrote:
>> On Wed, Sep 23, 2009 at 8:08 PM, William King  wrote:
>>
>> Another problem: all music packages in the repository (except 48Khz) are 
>> empty.
>
> If your speaking of the source package then they should be.  If the binary 
> package freeswitch-sounds-music-8000 is empty then we have problems.  It was 
> working the other day.  Please let me know.

Of course, I meant the binary packages:

http://ppa.launchpad.net/pbxbuntu-drivers/ppa/ubuntu/pool/main/f/freeswitch-sounds-music/

Note their sizes.

- Dmitry Bely

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Re: [Freeswitch-users] Status of ubuntu/debian packages.

2009-09-24 Thread Dmitry Bely 
On Wed, Sep 23, 2009 at 8:08 PM, William King  wrote:

Another problem: all music packages in the repository (except 48Khz) are empty.

- Dmitry Bely

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Re: [Freeswitch-users] Status of ubuntu/debian packages.

2009-09-24 Thread Dmitry Bely 
On Thu, Sep 24, 2009 at 2:14 AM, William King  wrote:
> Sure, post it here and I'll add it in the next build in a few hours.

See attached file.

Unfortunately mod_skypiax author did not placed config files
(skypiax.conf.xml, skypiax.X.conf) into conf/autoload_configs, so they
are not included into freeswitch-config and should be added manually.

BTW, why swig is a dependency for the source package? I recall Brian's
post where he insists that swig is never needed to build Freeswitch.

> -William King
>
> Dmitry Bely wrote:
>> On Thu, Sep 24, 2009 at 12:25 AM, Frank Carmickle  
>> wrote:
>>
>>> On Wed, Sep 23, Dmitry Bely wrote:
>>>
>>>> Can you enable mod_skypiax in your debian package?
>>>>
>>> We will be enabling as much as we can cleanly build on debian/ubuntu.  
>>> There will be a lot more to come.  We will be breaking the mods and end 
>>> points in to different packages so that you can install what you like.  If 
>>> you have something you would like to see in the package let us know.  Also 
>>> patches are welcome.
>>>
>>
>> Well, mod_skypiax just requires trivial one-line addition to
>> debian/rules and debian/freeswitch.install. It builds OK. If the patch
>> is required I can post it here.

- Dmitry Bely


freeswitch.diff
Description: Binary data
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Re: [Freeswitch-users] Status of ubuntu/debian packages.

2009-09-23 Thread Dmitry Bely 
On Thu, Sep 24, 2009 at 12:25 AM, Frank Carmickle  wrote:
> On Wed, Sep 23, Dmitry Bely wrote:
>> Can you enable mod_skypiax in your debian package?
>
> We will be enabling as much as we can cleanly build on debian/ubuntu.  There 
> will be a lot more to come.  We will be breaking the mods and end points in 
> to different packages so that you can install what you like.  If you have 
> something you would like to see in the package let us know.  Also patches are 
> welcome.

Well, mod_skypiax just requires trivial one-line addition to
debian/rules and debian/freeswitch.install. It builds OK. If the patch
is required I can post it here.

- Dmitry Bely

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Re: [Freeswitch-users] Status of ubuntu/debian packages.

2009-09-23 Thread Dmitry Bely 
On Wed, Sep 23, 2009 at 8:08 PM, William King  wrote:
(..)

> Any questions?

Another problem (on Ubuntu jaunty):

Setting up freeswitch (1.0.4+repack6-0ubuntu14925.1) ...
adduser: The --group, --ingroup, and --gid options are mutually exclusive.
addgroup: The user `freeswitch' does not exist.
chown: invalid user: `freeswitch:freeswitch'

- Dmitry Bely

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Re: [Freeswitch-users] Status of ubuntu/debian packages.

2009-09-23 Thread Dmitry Bely 
On Wed, Sep 23, 2009 at 8:08 PM, William King  wrote:
> Just to give everyone an update. There are working Ubuntu packages in a
> launchpad ppa. Debian users can add the ppa to their apt sources and
> build the package on your box. I'm currently using the packages on my
> home box and it is working great.
>
> Alright. I'm looking for people who want to use the packages. There are
> built packages for Ubuntu 8.04, 8.10, 9.04, and I'm working on 9.10 as
> well.  I'm building two apt repos. One will have nightly builds and the
> other will be for tagged releases, plus any major bug fixes.
>
> Thanks to Frank, we now have everything split out into separate files
> for everything. This is to try to reduce the amount of stuff you 'have'
> to download by default. We have the en-us-callie sounds packaged at 8k,
> 16k, 32k, and 48k, we also have packaged the russian-elena and music on
> hold at the same qualities. If there are other languages, or voices I'd
> be more than happy to package them. I just need the 48k. Also we have
> separated the packages out so you can specify which mods you want, such
> as mod_perl, mod_python, mod_lua are all separate so you can install
> them if you want.
>
> nightlies:
> https://launchpad.net/~pbxbuntu-drivers/+archive/ppa
>
>
> tagged releases:
> https://launchpad.net/~freeswitch-drivers/+archive/ppa
>
>
> The tagged released packages will start with the 1.0.5 tagged release
> which some say should be coming out soon.
>
> Any one who would like to help out feel free to sign up with a launchpad
> account, and request to join the driver team of either or both ppa's.
>
> Once we have some people using the packages, and testing them, I have
> already talked to some of the ubuntu official package maintainers about
> what would need to be done to add freeswitch into the ubuntu multiverse
> repo.
>
> Any questions?

Can you enable mod_skypiax in your debian package?

- Dmitry Bely

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Re: [Freeswitch-users] Affordable GSM gateway for one cellphone?

2009-09-22 Thread Dmitry Bely 
What about this one?

http://www.gempro.com.tw/gp-710.htm

On Tue, Sep 22, 2009 at 11:07 AM, Fred-145  wrote:
>
> Thanks for the link to PORTech's MV-370. For those interested, it can be had
> for a retail price of £150/€165 (before VAT).
>
> www.discountphonesystems.co.uk/acatalog/MV-370.html
> --
> View this message in context: 
> http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530601.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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- Dmitry Bely

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[Freeswitch-users] BLF docs/howto?

2009-09-21 Thread Dmitry Bely 
Have not found anything usable in the wiki/mail list archives.

I'm trying to setup BLF (busy lamp field) for Grandstream GXP-2000
phone. It offers BLF/eventlist BLF modes. Does Freeswitch  supports
both including the latter (RFC4662)? How to setup BLF on Freeswitch
side? Are there any examples?

- Dmitry Bely

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[Freeswitch-users] Skypiax false DTMF event

2009-09-09 Thread Dmitry Bely 
YPE  104  ][interface1][-1, 0, 0]
READING: |||USERSTATUS OFFLINE|||
2009-09-09 22:48:08.274202 [DEBUG] skypiax_protocol.c:104 rev
14707[(nil)|37 ][DEBUG_SKYPE  104  ][interface1][-1, 0, 0]
READING: |||CONNSTATUS CONNECTING|||
2009-09-09 22:48:31.374348 [DEBUG] skypiax_protocol.c:104 rev
14707[(nil)|37 ][DEBUG_SKYPE  104  ][interface1][-1, 0, 0]
READING: |||USER user1 ONLINESTATUS ONLINE|||
2009-09-09 22:48:31.374348 [DEBUG] skypiax_protocol.c:104 rev
14707[(nil)|37 ][DEBUG_SKYPE  104  ][interface1][-1, 0, 0]
READING: |||USERSTATUS ONLINE|||
2009-09-09 22:48:31.375500 [DEBUG] skypiax_protocol.c:104 rev
14707[(nil)|37 ][DEBUG_SKYPE  104  ][interface1][-1, 0, 0]
READING: |||CONNSTATUS ONLINE|||

Is this expected behavior? Of course, the main problem is probably in
hardware but does skypiax do its job right?

- Dmitry Bely

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Re: [Freeswitch-users] Proxy authorization

2009-09-04 Thread Dmitry Bely 
On Sat, Sep 5, 2009 at 1:08 AM, Brian West wrote:
> show me your XML for the gateway please.
> /b

It's fairly standard:



  

  












  


  

  


default_provider_register is set to true.

In the meantime I looked into the sources. If I understand them right,
proxy address is always used in REGISTER header:

sofia.c, line 1471
gateway->register_url =
switch_core_sprintf(gateway->pool, "sip:%s", proxy);

Probably it's incorrect as RFC 3261 says:

Request-URI: The Request-URI names the domain of the location
   service for which the registration is meant (for example,
   "sip:chicago.com").  The "userinfo" and "@" components of the
   SIP URI MUST NOT be present.

So the domain name (from-domain?) should be used there, not the proxy address.

> On Sep 4, 2009, at 3:43 PM, Dmitry Bely wrote:
>
> I'm started to suspect another thing.. Successful register (SIP phone)
> contains
>
> REGISTER sip:Domain SIP/2.0
>
> while unsuccessful one is
>
> REGISTER sip:1.2.3.4 SIP/2.0
>
> What parameter is responsible for Request-URI? Note that I need both
> IP address for proxy and symbolic name for SIP domain  (which is not
> mapped the resolvable DNS name).

- Dmitry Bely

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Re: [Freeswitch-users] Proxy authorization

2009-09-04 Thread Dmitry Bely 
I'm started to suspect another thing.. Successful register (SIP phone) contains

REGISTER sip:Domain SIP/2.0

while unsuccessful one is

REGISTER sip:1.2.3.4 SIP/2.0

What parameter is responsible for Request-URI? Note that I need both
IP address for proxy and symbolic name for SIP domain  (which is not
mapped the resolvable DNS name).

On Fri, Sep 4, 2009 at 11:37 PM, Brian West wrote:
> Try filling out contact-host too.  But if the far end gets pissed about your
> contact they are broken.
> /b
> On Sep 4, 2009, at 2:22 PM, Dmitry Bely wrote:
>
> Well, you are right. Looks like the problem is not with authorization
> but in the line
>
>  Contact: 
>
> that the gateway would like to see as
>
>  Contact: 
>
> I've found (almost undocumented) parameter extension-in-contact, but
> it still gives
>
>  Contact: 

- Dmitry Bely

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Re: [Freeswitch-users] Proxy authorization

2009-09-04 Thread Dmitry Bely 
On Fri, Sep 4, 2009 at 4:45 PM, Brian West wrote:
> I cannot change the way SIP Authentication works.  The first register is
> always sent without an authorization header then is challenged.  If you're
> getting an instant 403 then you have something wrong in your config and the
> remote system doesn't like it.  Please contact your provider and ask them to
> troubleshoot it with you.
> /b

Well, you are right. Looks like the problem is not with authorization
but in the line

  Contact: 

that the gateway would like to see as

  Contact: 

I've found (almost undocumented) parameter extension-in-contact, but
it still gives

  Contact: 

(1.2.3.4 is my IP address, 5.6.7.8 is gateway's one). Any idea how to
overcome this?

> On Sep 4, 2009, at 3:43 AM, Dmitry Bely wrote:
>
> Alas, I cannot change the way the provider's gateway works. It
> immediately responses with 403... BTW, it's Mera Damos
> (http://www.mera-systems.com ?). No workaround possible?

- Dmitry Bely

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

2009-09-04 Thread Dmitry Bely 
I had a similar problem when I needed to talk to a gateway using g729
while g711 was used by default. The following works for me:

vars.xml
(...)
  
  

sip_profiles/internal.xml
(...)


dialplan/default/01_example.com.xml
(...)
  
  

On Fri, Sep 4, 2009 at 9:54 AM, 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 questions:
>>     >        - does application "start_dtmf_generate" requires transcoding
>>     >        - if yes, can I set the variable disable-transcoding in
>>     my dialplan
>>     >
>>     >     regards,
>>     >     rod

- Dmitry Bely

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Re: [Freeswitch-users] Proxy authorization

2009-09-04 Thread Dmitry Bely 
On Fri, Sep 4, 2009 at 1:19 AM, Brian West wrote:
> There will not be an authorization header on the first register attempt...
> it only happens once we are 401/407'ed and the phone comes back and
> registers again.
> /b

Alas, I cannot change the way the provider's gateway works. It
immediately responses with 403... BTW, it's Mera Damos
(http://www.mera-systems.com ?). No workaround possible?

> On Sep 3, 2009, at 3:26 PM, Dmitry Bely wrote:
>
> Unfortunately even after that there is no "Authorization:" header in
> the REGISTER message:
>
>   REGISTER sip:1.2.3.4 SIP/2.0
>   Via: SIP/2.0/UDP 5.6.7.8:5080;rport;branch=z9hG4bKNBB3ygD85y3eF
>   Max-Forwards: 70
>   From: ;tag=Nrc6Z9yrNBS3H
>   To: 
>   Call-ID: a93d949a-98c1-11de-b6b8-8321249ad8d4
>   CSeq: 119885384 REGISTER
>   Contact: 
>   Expires: 600
>   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14707M
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>   Supported: timer, precondition, path, replaces
>   Content-Length: 0

- Dmitry Bely

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[Freeswitch-users] Proxy authorization

2009-09-03 Thread Dmitry Bely 
Hi,

My SIP provider's gateway requires authorization with correct realm
specified. So I configured gateway with "realm" parameter:

  












  

(default_provider_register = true)

Unfortunately even after that there is no "Authorization:" header in
the REGISTER message:

   REGISTER sip:1.2.3.4 SIP/2.0
   Via: SIP/2.0/UDP 5.6.7.8:5080;rport;branch=z9hG4bKNBB3ygD85y3eF
   Max-Forwards: 70
   From: ;tag=Nrc6Z9yrNBS3H
   To: 
   Call-ID: a93d949a-98c1-11de-b6b8-8321249ad8d4
   CSeq: 119885384 REGISTER
   Contact: 
   Expires: 600
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14707M
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Length: 0

and the provider responds with 403 User '111' is disabled (as
there is no correct authorization realm there). How to force
Authorization header? Am I missing something?

- Dmitry Bely

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