Is there any example on how to use mod_fifo?
I am trying to implement a call centre queue as follows (much like
Asterisk queues) :
Inbound call- press 0 for operator - mod_fifo - 3 agents of whom any
one can get the call (doing round robin or whatever)
I checked out:
Sorry, just a little not sure to understand correctly -
there is no way to add a new SIP account without restarting
the SIP profile, and so any such changes will cause a call
drop?
Am I correct?
On Wednesday 02 July 2008 02:38, Brian West wrote:
Please take the time and make sure the wiki is
You can add users (people who register to you) but not gateways
without restarting the sip profile.
On Jul 2, 2008, at 3:14 AM, Anton wrote:
Sorry, just a little not sure to understand correctly -
there is no way to add a new SIP account without restarting
the SIP profile, and so any such
We have a FS box on Lylix/CentOS connected to les.net and making and
receiving calls, running commands from the console etc.
However, my tech resource is not available for the time being, and I need
some help setting up a basic Python hello world which will capture the
event of an incoming call,
You can disregard those 'solution folder' messages. The source
includes accommodations for building on linux, vs2005 and vs2008 in
one location.
In VS you can either build in release mode or debug mode by setting a
build variable.
On Wed, Jul 2, 2008 at 5:56 AM, Kin Quek [EMAIL PROTECTED] wrote:
if you add these 2 extensions: 7010 will be for agents who will hear music
till someone calls
and 7011 will be the customer who will hear hold music until an agent is
free.
extension name=test
condition field=destination_number expression=^7010$
action application=set
You don't need a extension created for the cisco... Just set it up to
forward the DID to the freeswitch boxes IP on its dial peer.. Then on
freeswitch you set up a profile w/ auth calls turned off then have a
separate context for that profile that does IP auth for the cisco something
like this
Оригинално писмо
От: Michael Jerris
Относно: Re: [Freeswitch-users] How to Configure SIP DID to IVR
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Сряда, 2008, Юли 2 19:24:22 EEST
^ seems like an invalid regex. is that literally what
you have there or you
Most likely its not actually matching the extension or it runs out of
actions to perform, can you post the full debug logs from the console?
Mike
On Jul 2, 2008, at 1:14 PM, Hristo Benev wrote:
Оригинално писмо
От: Michael Jerris
Относно: Re: [Freeswitch-users] How to
Here is the output:
---
2008-07-02 13:48:47 [NOTICE] switch_channel.c:533 switch_channel_set_name() New
Channel sofia/cisco/CallingNumber@CIscoIP
[c0d8586f-f6b9-4108-8676-c49e66f32e6d]
2008-07-02 13:48:47 [INFO] mod_dialplan_xml.c:222 dialplan_hunt()
Strange I changed regex to DID not ^DID and it worked?!
Оригинално писмо
От: Hristo Benev
Относно: Re: [Freeswitch-users] How to Configure SIP DID to IVR
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Сряда, 2008, Юли 2 21:00:59 EEST
Here is the output:
The ERR stun failed below is killing your call.
On Jul 2, 2008, at 3:08 PM, Hristo Benev wrote:
Strange I changed regex to DID not ^DID and it worked?!
Оригинално писмо
От: Hristo Benev
Относно: Re: [Freeswitch-users] How to Configure SIP DID to IVR
До:
Don't forget.
Brian West
sip:[EMAIL PROTECTED]
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Thanks Jeff.
The problem was Visual Studio 2005 + Windows SDK
Platform was not supported on Vista. I installed
Visual Studio 2008 and ignored the error messages and
FS built OK. Thanks for the help.
Can you please point me to how to test out the new
built FS? Is there any beginner's DOC
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