Hi,
I am in the processing of implementing a typical scenario for service call.
So, in my scenario, I would have several agents who may log in and log out
of the queue at different times of the day.
People can call in and will be routed to one of the available agents. If no
agent responses within
Brian West wrote:
> TOTALLY AWESOME!
>
> On Apr 22, 2008, at 1:07 PM, Michael Collins wrote:
>
>> Righteous!
yah bro!
There's a few examples on the Category:Javascript page
(http://wiki.freeswitch.org/wiki/Category:Javascript) that helped me a lot.
We have a temporary fix in tree for this that should keep it from
segfaulting in this situation, expect a more comprehensive fix in the
next few days.
Mike
On Apr 22, 2008, at 9:56 AM, Luis Jimenez wrote:
This is the debug:
freeswitch is running and phones 1000 & 1001 registered then i
TOTALLY AWESOME!
On Apr 22, 2008, at 1:07 PM, Michael Collins wrote:
> Righteous!
Brian West
sip:[EMAIL PROTECTED]
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Righteous!
> -Original Message-
> From: [EMAIL PROTECTED]
[mailto:freeswitch-
> [EMAIL PROTECTED] On Behalf Of Brian Snipes
> Sent: Tuesday, April 22, 2008 9:36 AM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Call for IVRs
>
> Didn't even know about it. I
Didn't even know about it. I'll do it, thanks.
Brian
On Tue, 2008-04-22 at 09:31 -0700, Michael Collins wrote:
> Brian,
>
> Did you already wikify the new sample IVR that bkw and I created? (It's
> extension 5000 in the default configs...)
>
> Just checking, thanks...
>
> -MC
>
> > -Ori
Brian,
Did you already wikify the new sample IVR that bkw and I created? (It's
extension 5000 in the default configs...)
Just checking, thanks...
-MC
> -Original Message-
> From: [EMAIL PROTECTED]
[mailto:freeswitch-
> [EMAIL PROTECTED] On Behalf Of Brian Snipes
> Sent: Tuesday, April
Can you be more specific with the steps to reproduce this.
We are trying to reproduce it with our snom and we cannot.
What version of the code are you using?
Can you give us a step by step of each configuration step you make on the
snom.
Also any differences between your FS config and the default
What about it?
Did you try my suggestion? I am not sure I could have explained it any
better?
Watch your console trace and maybe take a pcap of it. you can export
TPORT_LOG=1 in your shell to see the sip messages in the console.
The instant the phone you are calling sends us 180, we send you 1
What about OpenSer, i get the same results as Asterisk.
On Tue, Apr 22, 2008 at 10:11 AM, Anthony Minessale <
[EMAIL PROTECTED]> wrote:
> The ringing is not passed across until the other phone (the one you are
> calling) sends a 180 Ringing.
>
> As soon as it sends it, we pass that indication to
The ringing is not passed across until the other phone (the one you are
calling) sends a 180 Ringing.
As soon as it sends it, we pass that indication to the calling phone (your
phone).
Asterisk just assumes you should hear ringing and sends it instantly on it's
own before anyone knows that the ca
This is the debug:
freeswitch is running and phones 1000 & 1001 registered then i change
credentials of 1001 to 1002.
i shut down FS with ...
then i start FS again
if i reboot phone 1002, this send an UNREGISTER MESSAGE to FS and then i see
the following in the console:
[EMAIL PROTECTED]> Segment
Ok, my network topology is:
1 server HP ML-110 FS installed.
2 Snom 360
1 Switch Linksys SRW224P
using default dialplan installed by make samples
this is the debug of de FS console when you dial from 1000 to 1001:
[EMAIL PROTECTED]> 2008-04-22 08:33:43 [NOTICE] switch_channel.c:531
switch_chann
I think it will always work. But I am not 100% sure.
On Tue, Apr 22, 2008 at 8:44 AM, Jonas Gauffin <[EMAIL PROTECTED]>
wrote:
> Ahh. doh. Never thought of that. You are of course correct.
>
> Do deflect work both with unanswered and answered calls?
> Or do I need to use a check whether the call
Ahh. doh. Never thought of that. You are of course correct.
Do deflect work both with unanswered and answered calls?
Or do I need to use a check whether the call have been answered when
it hits the dialplan?
On Tue, Apr 22, 2008 at 3:18 PM, Anthony Minessale
<[EMAIL PROTECTED]> wrote:
> Can you s
Can you still do a 302 once you answer?
I thought that only works before you answer. it's a final response to an
invite like a 488 etc.
You can try the deflect app instead which is similar in the end result
redirect but it sends a SIP refer (like a blind transfer on a sip phone)
so it works on an
I would like to ask anyone that has working IVRs to please post them to
the list or on the wiki ( http://wiki.freeswitch.org ). If you post to
the list I will add them to the wiki for you. The few items on the wiki
now were created no less than 11 months ago. There have been a lot of
questions r
I do a transfer to an extension that is redirected to another one.
=> ivr app that transfers the call
1201 => uses the redirect app in the dialplan to 1203
1203 => extension that should be ringing
this works:
caller -> 1201 (redirects to 1203) -> 1203 is ringing
caller -> (doing transfe
Didn't you say you wanted to do 302?
transfer app is a FS internal transfer.
302 is initiated with session.execute("redirect",
"sip:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
");
If you do actually mean to transfer within FS, (meaning send the current
call back to the dialplan)
Be sure to exit your sc
can you provide the console output for this failure?
On Apr 22, 2008, at 2:23 AM, Jonas Gauffin wrote:
> The transfer is made with:
> session.setVariable("ringback", "%(1000,4000,425)");
> session.execute("transfer", user.extension + " XML");
Brian West
sip:[EMAIL PROTECTED]
___
Hi all,
I am new to this freeswitch, i have started my career recently on voip.
i am able to install freeswitch sever in my PC and i am able to connect two
xlite clients via this freeswitch server. Now my problem is i want to configure
a freeswitch client. please can any one help me out.
R
Ok, ive done some more testing.
The redirect works fine in this case (with the dialplan that i sent in
my last post):
Caller -> 1201 (which is redirected to 1203) -> 1203 is ringing
The same dialplan fails in this scenario:
Caller -> ivr menu (javascript) -> transfer to 1201 (same redirect is
mad
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