On 01/28/2011 Kevin P. Fleming wrote:
Loading or not loading a MOH provider is not going to change Asterisk's
behavior regarding hold/unhold of endpoints; if you want Asterisk to
pass
through hold/unhold indications over SIP, unfortunately it can't do
that yet... although most of the
On 30 Jan 2011, at 09:21, Pezhman Lali wrote:
Faxter is an opensource email to fax gateway,
please check it, let me know if any bug.
Only bug i can see is the attitude of the developer...
As for the bugs, having the config variables liberally scattered throughout the
script makes it's use
Hi,
Here is my confing:
[out]
Exten = _X.,1,Noop()
Exten = _X.,2,Dial(SIP/${EXTEN}@peer,60,gcU(do_dtmf_cc-take-call,s,1))
Exten = _X.,3,Playback(tt-monkeys)
Exten = _X.,4,Playback(tt-monkeys)
Exten = _X.,5,Playback(tt-monkeys)
Exten = h,1,Noop(ABCDEFGHIJKLMNOPQRSTUVWXYZ)
On 01/31/2011 02:06 AM, Urs Buob wrote:
On 01/28/2011 Kevin P. Fleming wrote:
Loading or not loading a MOH provider is not going to change Asterisk's
behavior regarding hold/unhold of endpoints; if you want Asterisk to
pass
through hold/unhold indications over SIP, unfortunately it
Hi,
Here is my confing:
[out]
Exten = _X.,1,Noop()
Exten = _X.,2,Dial(SIP/${EXTEN}@peer,60,gcU(do_dtmf_cc-take-call,s,1))
Exten = _X.,3,Playback(tt-monkeys)
Exten = _X.,4,Playback(tt-monkeys)
Exten = _X.,5,Playback(tt-monkeys)
Exten = h,1,Noop(ABCDEFGHIJKLMNOPQRSTUVWXYZ)
[do_dtmf_cc-take-call]
Sorry for resurrecting an old thread...
Tilghman Lesher writes:
Out of curiosity, what platform are you running on? On most platforms
that are able to run Asterisk, with the possible exception of Solaris,
increasing the maximum file descriptor for use with select(2) is
possible.
I am not
HI
I have used this xml file for registering a sipp client with
asterisk i dont know whether it is correct or not.. could you please rectify
and inform me where i did wrong
?xml version=1.0 encoding=ISO-8859-1 ?
!DOCTYPE scenario SYSTEM sipp.dtd
scenario name=registration
send
Hello All,
I have asterisk installed in our call center and i want to know how to do in
order to save all the calls (inbound and outbound) if there is any tool
Thanks in advance
Kind Regards.
--
_
-- Bandwidth and
Thanks you @ Godson Gera , @Sherwood McGowan , @ CF
Thank you for mentioning.
I have tried all the options (excluding AMI) but in vain.
Let me show you what happens
When the call starts core show channels shows me
Channel Location State Application(Data)
On 01/31/2011 12:51 PM, salaheddine elharit wrote:
I have asterisk installed in our call center and i want to know how to
do in order to save all the calls (inbound and outbound) if there is any
tool
Yes, there is.
Tom
PS: Sorry, I couldn't resist!
--
On 01/31/2011 12:51 PM, salaheddine elharit wrote:
I have asterisk installed in our call center and i want to know how to
do in order to save all the calls (inbound and outbound) if there is any
tool
OK, now to be somewhat more helpful, this is a common scenario. You
should search for
On Monday 31 January 2011 07:26:25 Benny Amorsen wrote:
Sorry for resurrecting an old thread...
Tilghman Lesher writes:
Out of curiosity, what platform are you running on? On most platforms
that are able to run Asterisk, with the possible exception of Solaris,
increasing the maximum file
From: Kevin P. Fleming kpflem...@digium.com
Sent: Thursday, January 27, 2011 3:08 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] res_fax
On 01/27/2011 09:21 AM, Bryant Zimmerman wrote:
Kevin
That is grate. I dove into the code
Hi all,
I've got an agi script that calls the directory function, which seems to
work to a point. However, once the caller has selected an entry, I need my
agi script to find out which extension was selected. I've RTFM'd and don't
see that the extension is returned. Nor is a variable set, as
Benny Amorsen benny+use...@amorsen.dk wrote:
Sorry for resurrecting an old thread...
Tilghman Lesher writes:
Out of curiosity, what platform are you running on? On most platforms
that are able to run Asterisk, with the possible exception of Solaris,
increasing the maximum file
On Mon, 31 Jan 2011, Mike Diehl wrote:
I've got an agi script that calls the directory function, which seems to
work to a point. However, once the caller has selected an entry, I need
my agi script to find out which extension was selected. I've RTFM'd and
don't see that the extension is
On 01/31/2011 02:08 PM, Bryant Zimmerman wrote:
*From*: Kevin P. Fleming kpflem...@digium.com
*Sent*: Thursday, January 27, 2011 3:08 PM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] res_fax
On
Hello!
Im new to Asterisk configuration and I have few questions regarding its
configuration.
I have 4 PSTN lines connected to TDM410. I can make exact 60 minutes of free
calls from each of 4 pstn lines... Can I configure Asterisk to call thru
pstn line that has free minutes? For example
On Monday 31 January 2011 15:16:13 cov...@ccs.covici.com wrote:
Benny Amorsen benny+use...@amorsen.dk wrote:
Sorry for resurrecting an old thread...
Tilghman Lesher writes:
Out of curiosity, what platform are you running on? On most
platforms that are able to run Asterisk, with the
On Mon, 31 Jan 2011, Piotr Górski wrote:
I have 4 PSTN lines connected to TDM410. I can make exact 60 minutes of
free calls from each of 4 pstn lines... Can I configure Asterisk to call
thru pstn line that has free minutes? For example
Outgoing calls are going through PSTN 1 for 60 minutes.
From: Kevin P. Fleming kpflem...@digium.com
Sent: Monday, January 31, 2011 5:13 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] res_fax
On 01/31/2011 02:08 PM, Bryant Zimmerman wrote:
Steve Edwards asterisk@sedwards.com wrote:
On Mon, 31 Jan 2011, Mike Diehl wrote:
I've got an agi script that calls the directory function, which seems to
work to a point. However, once the caller has selected an entry, I need
my agi script to find out which extension was selected.
On 01/31/2011 05:06 PM, Bryant Zimmerman wrote:
I just replaced the res_fax.c file with the one from 304599. Would I
just keep doing that as I step forward on versions of 1.8.x?
If this is the case how would I get any other critical changes to
res_fax.c that may occur after rev 304599?
How
Hey guys,
I'm sorry this isn't * related but if there *is* an 'answer' to this
question, I suspect someone on this list will know it. :)
I'm trying to work out what technology to use;
Situation:
Mobile Linux computer connected via 3G/GPRS to internet.
The computer is likely to encounter
Channel Location State Application(Data)
SIP/NTT00- 99449046902115@vicid Down AppDial((Outgoing Line))
Local/99449046902115 99449046902115@defau Up
Dial(SIP/NTT00/449046902115||o
Local/99449046902115 8302@default:2 Up Playback(conf)
Hi all,
when i was trying to register a sipp client by using
register_client.xml file with .csv file in asterisk server i have
encountered an error that
10642150.240891127.0.0.1127.0.0.1SIPStatus: 481 Call
leg/transaction does not exist
I dont kknow how does
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