On 10:12 Mon 19 Jul , Nasir Iqbal wrote:
> Try 3 second wait between Answer and ReceiveFAX
I'am added but this don't help.
; extensions.conf part with fax
exten => fax,1,Goto(543,1)
exten => 543,1,Answer()
exten => 543,n,Set(FAXFILE=/var/spool/asterisk/fax/${CALLERID(num)}.tif)
exten => 543,
On 19 July 2010 00:35, Anthony Messina wrote:
> On Wednesday, July 14, 2010 01:44:54 pm bruce bruce wrote:
>> Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones,
>> how can one receive distinctive ring tones for INTERNAL calls ONLY?
>
> Using Aastra 4801 CT phones...
>
> [ext
Excellent!
I finally got it working, it was ODBC drivers issue actually. Installed the
proper compatible version and its working.
There are still few errors which i see on asterisk console:
[Jul 19 13:58:51] WARNING[30213]: res_config_odbc.c:1032 require_odbc:
Realtime table book...@meetme req
Try 3 second wait between Answer and ReceiveFAX
On Mon, Jul 19, 2010 at 9:27 AM, Stefan Schmidt wrote:
> Alexander Aksarin schrieb:
> > Hello, All. I have a problem with receiving fax through T.30. I'm
> > calling 543 number from fax machine, then start sending fax and fax
> > machine send docu
Alexander Aksarin schrieb:
> Hello, All. I have a problem with receiving fax through T.30. I'm
> calling 543 number from fax machine, then start sending fax and fax
> machine send document without problem. But asterisk don't receive fax.
> I can't find good documentation for app_fax and I'am googl
Hello, All. I have a problem with receiving fax through T.30. I'm
calling 543 number from fax machine, then start sending fax and fax
machine send document without problem. But asterisk don't receive fax.
I can't find good documentation for app_fax and I'am googled this
errors.
Please help me.
so
On Wednesday, July 14, 2010 01:44:54 pm bruce bruce wrote:
> Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones,
> how can one receive distinctive ring tones for INTERNAL calls ONLY?
Using Aastra 4801 CT phones...
[external-context]
; Calls entering from outside the system
e
I'm looking for a good Linux Softphone that
a> has "Consultation Transfer" built in, I know you can do this by
dialling what ever is in features.conf but this is not ideal.
b> has the ability to handle more than 2 lines eg calls at a time.
c> Works with Asterisk.
d> Has a feature where someone can
bruce bruce wrote:
> Hi Everyone,
>
> If I receive a call on a ZAP line and pickup the call and drag and
> drop it (by mouse) into a Parking Lot through FOP, it just hangs up.
> Is this feature supported by FOP?
>
I don't believe so, how would Asterisk know what phone to ring on timeout?
Doug
-
Marta Silva wrote:
>
>
> I have configured the (ttyIAX) modems with the specific
> valuesI was hoping I could choose which one to use for each of the
> sip clients (defined on the GXW box), and this way, select the
> outbound numberif not, how can I do this?
>
My iaxmodems are configure
Has anyone used FreeSide to do billing with Asterisk?
How easily did you find it integrated?
With external systems?
With your credit card processor?
How easy was it to add additional fields or 'service' types?
--
_
> On Sat, Jul 17, 2010 at 6:52 PM, David Shauger
> wrote:
>> Can anyone provide the settings in Audacity to create a proper wav file
>> without having to do additional conversion in the cli? Has to be a way
>> to do this with less steps.
On Sun, 18 Jul 2010, David Backeberg wrote:
> If your
On Sun, Jul 18, 2010 at 09:56:30AM -0700, Vieri wrote:
> > As I said above, once you have purchased your SIP channel
> > you can make
> > free calls to your PBX using the special number but it's
> > only INBOUND
> > AFAIK.
[lots snipped]
With Skype's just released SkypeKit it should be possible t
I am not aware with any logging option, but If you want to monitor
registration status. Asterisk Realitme can help you.
For example if you are using Realtime SIP configuration then you can find
registration info at "regserver" and "regseconds" fields
On Sun, Jul 18, 2010 at 7:28 PM, Bram Bosboom
--- On Sun, 7/18/10, Alejandro Imass wrote:
> > Hi,
> >
> > I'm trying to integrate Skype and Asterisk but I'm
> only interested in these 2 things:
> >
> > 1) allow any Asterisk SIP extension to call any Skype
> "user". I do not need to call landlines via Skype.
> >
>
> I think this is _explic
On Sat, Jul 17, 2010 at 6:52 PM, David Shauger wrote:
> Can anyone provide the settings in Audacity to create a proper wav file
> without having to do additional conversion in the cli? Has to be a way to do
> this with less steps.
If your goal is to 'minimize steps', you should do a batch on the
On Sat, Jul 17, 2010 at 3:31 PM, Tzafrir Cohen wrote:
> Would the ability to temporarily change logging settings from the CLI do
> the trick?
>
For debugging purposes, I would agree with this statement. Many times
when trying to capture logs for a bug, you need to first modify your
logger.conf, r
On Saturday 17 July 2010 17:23:19 Steve Edwards wrote:
> > On Fri, Jul 16, 2010 at 10:59:43AM -0700, Steve Edwards wrote:
> >> "Request For Comments on a Feature Suggestion" -- just wondering if
> >> others would find this useful.
> >>
> >> It occurs to me that being able to just enter:
> >>
> >>
Can asterisk log the registration date/time in a database? Is there a
standard option to do this?
I know it being logged in the asterisks 'full' (debug) log and we are
probably able to script something with the API interface but there might
be somewhat easier if there is a option to make aste
On Sun, Jul 18, 2010 at 7:48 AM, Vieri wrote:
> Hi,
>
> I'm trying to integrate Skype and Asterisk but I'm only interested in these 2
> things:
>
> 1) allow any Asterisk SIP extension to call any Skype "user". I do not need
> to call landlines via Skype.
>
I think this is _explicitly_ not suppo
On Fri, 2010-07-16 at 17:34 +0100, Paddy Grice wrote:
> Seems BLF only work on called extensions - is there a way to show busy
> for the calling extension?
You don't say what version of asterisk you are running this on, or have
any config snippets, so difficult to say what might be wrong.
Check
Hi,
I'm trying to integrate Skype and Asterisk but I'm only interested in these 2
things:
1) allow any Asterisk SIP extension to call any Skype "user". I do not need to
call landlines via Skype.
2) allow Internet Skype "users" to call my Asterisk PBX Skype "user" and route
the call to a speci
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