Pete Mueller wrote:
>
> Is there a way to detect DTMF during bridged conversation?
You can use bind_meta_app in your dial plan; see the wiki for details.
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On 17 Apr 2009, at 18:42, Meftah Tayeb wrote:
Hello,
please anyone here a implemented CURL for freeswitch using any
Language
and any
RDBNMS ?
i have a freeswitch contributed one implementation using PHP / MySQL
but
is not working.
any help is werlcome
thanks!
Yeah, we have done one for
Thanks for insight in this, Ill look into doing that.
In the directory though, we have other PBXs connected to us (basically we
provide some dialtone for a few people), and we stuck them in the directory
as it was the only way we dould have each user have their own context, but
we can't do a nice
On Fri, Apr 17, 2009 at 11:58 PM, Giovanni Maruzzelli
wrote:
> Then you install FS as service (freeswitch.exe -install servicename),
> start FS as a service (under "local system"), manually (again, from
> the "services" applet).
make sure the FS service is owned by "local system" and that "Access
> On Fri, Apr 17, 2009 at 4:02 PM, UV wrote:
>> Give a shout if you get Skypiax working as a service.
>> I'll be happy to contribute to its wiki about it once you get it working.
got Skypiax working as a service
I will document this better in the future, but following is the
general idea, from a
Quick question ya'll:
Is there a way to detect DTMF during bridged conversation? Or do I have to
use a conference for that?
Scenario: Caller A dials in and is bridged with caller B (in my case via lua
- session:execute("bridge", some_route)) If at any time during the
conversation caller
Just the usual goodies, here it is: ..
Original Message
Subject: Re: [Freeswitch-users] Issues detecting DTMF tones
From: Michael Collins
Date: Fri
Quick question: do you do anything interesting in the dialplan prior to
calling the script? For posterity's sake could you paste the extension info
here as well?
Thanks,
MC
On Fri, Apr 17, 2009 at 11:47 AM, wrote:
> below is the lua script that is running when the caller dials in. As
> mention
This isn't anything we can do anything about... its the gsm to PSTN
conversion of the digits... it'll take some time.. I have noticed this
on AT&T also while dialing digits on a remove IVR such as BofA.
/b
On Apr 17, 2009, at 1:47 PM, p...@privateconnect.com wrote:
I believe the isuse wit
below is the lua script that is running when the caller dials in. As mentioned before, swaping the streamFile and sleep make the problem go away. I am experiencing different issues all related to DTMF tones, again, I don't think FS is the problem. But I'm wondering how to find the culprit.Issues:
2009/4/15 Ron McCarthy
> Hi,
>
> We have several context setup with different users, all in the directory
> with IP based auth.
>
> Anyway we can turn on debug for just one context/user in the directory,
> when lots of people are making calls it's hard to see traffic to just for
> one user, and t
Meftah Tayeb wrote:
> Hello,
> please anyone here a implemented CURL for freeswitch using any Language
> and any
> RDBNMS ?
> i have a freeswitch contributed one implementation using PHP / MySQL but
> is not working.
>
"not working" is relatively loose. Do you have any more information than
t
Hello,
please anyone here a implemented CURL for freeswitch using any Language
and any
RDBNMS ?
i have a freeswitch contributed one implementation using PHP / MySQL but
is not working.
any help is werlcome
thanks!
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On Apr 17, 2009, at 12:07 PM, JuanMa wrote:
Hi,
I am using xml_curl, and what I'm trying to do is dynamically register
the users to the platform, instead of bulking all the users on the FS
boot time. So far I have successfully register the users and also make
calls through gateways or to stati
Hi,
I am using xml_curl, and what I'm trying to do is dynamically register
the users to the platform, instead of bulking all the users on the FS
boot time. So far I have successfully register the users and also make
calls through gateways or to static users (those users that are in the
phy
> If someone has a way to make true mirrors that support read/write
> this
> would be interesting.
Do it robustly, transparently and in real time and that's the problem of
distributed source code revision control mostly sorted as well.
Although I'm not sure I'd really want to use "the kernel any
Hi Brian,
The follower command worked per your suggestion
originate user/1...@192.168.1.102 &conference(3000-192.168.1.102)
Thanks a lot,
Prabhu
On Fri, Apr 17, 2009 at 6:43 AM, Brian West wrote:
> Are you calling a registered user? If so then please replace the @ with a
> % or user user/
If someone has a way to make true mirrors that support read/write this
would be interesting.
Mike
On Apr 17, 2009, at 4:06 AM, Will Boyce wrote:
> Special:Export will export a page to an XML format that can, in
> turn, be imported.
>
> There must be a way to automate that process (export ext
On Fri, Apr 17, 2009 at 4:02 PM, UV wrote:
> Give a shout if you get Skypiax working as a service.
> I'll be happy to contribute to its wiki about it once you get it working.
Yes, definitely!
gm
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On Fri, Apr 17, 2009 at 4:00 PM, UV wrote:
> You're suggesting running two instances of FS on same machine, have them
> communicate first over TCP then UDP via SIP over socket (Skype --> Skypiax
> --> FS1 --> FS2) on a production system? Hmmm... nah... :-)
>
That's exactly what the "farming clien
Give a shout if you get Skypiax working as a service.
I'll be happy to contribute to its wiki about it once you get it working.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Giovanni
Maruzzelli
Sen
You're suggesting running two instances of FS on same machine, have them
communicate first over TCP then UDP via SIP over socket (Skype --> Skypiax
--> FS1 --> FS2) on a production system? Hmmm... nah... :-)
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:fre
does it only work for registrations only or does it limit the user to calling
through that gateway only?
Anthony Minessale-2 wrote:
>
> It has not been used much but there was a way to add gateways to a user
> tag
> in the directory
> so if you register locally to FS to that user then it will
Are you calling a registered user? If so then please replace the @
with a % or user user/1...@192.168.1.102
/b
PS I do get called brain a lot!
On Apr 16, 2009, at 11:45 PM, Prabhuram Mohan wrote:
Thanks Brain/ Dave,
I ran the modified command as follows
originate sofia/default/1...@192.16
Hi Dave,
didn't fully get your explanation. Maybe I should explain how I do it right now.
I am using the http://www.codeplex.com/eventsocket library. I take my x-lite
phone with number 1000 and call into FS with number 301234. Under this number I
launch my app as async full.
It has not been used much but there was a way to add gateways to a user tag
in the directory
so if you register locally to FS to that user then it will register that
gateway and unreg it when he unregisters.
I don't have time to dig it up but the params are in there somewhere.
On Fri, Apr 17, 200
hi,
i was trying to do the stuff i had written above but seems a bit too tough
so on giving it a second though, the reason i need to do the above was each
client id is basically billed on my voipswitch so a way around it could be
was the client id and pass on voipswitch for all clients, i can cre
Anthony Minessale-2 wrote:
> A SIP proxy like openSIPS and a b2bua/media gateway like FreeSWITCH are
> meant to be used together.
> There is some overlap in SIP functionality but SIP is just one aspect of
> FreeSWITCH where SIP is the only
> thing OpenSIPS is for.
Thanks guys for the clarificati
mercutioviz wrote:
>
> definitely stop by IRC and talk real-time with others who've dealt with
> this
> kind of thing.
Thanks for the tip, but since this firewalls are a frequent issue with SIP,
I think it'd be very useful if someone who has good experience with this
wrote an article in the FS
Special:Export will export a page to an XML format that can, in turn, be
imported.
There must be a way to automate that process (export extire wiki to XML, rsync
and import and mirrors).
--
Regards,
Will Boyce
tel: 07933 515 987
url: http://willboyce.com
- "Jason White" wrote:
| Fr
Anthony implemented ${bridge_hangup_cause} in r13065 which has the value
of the last B leg bridge attempt. Works as expected!
The leg_delay_start problem still remains though. It would be great if
someone could do a test to see if leg_delay_start on a leg isn't
honored.
El jue, 16-04-2009 a las
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