Am Montag, den 09.07.2007, 17:21 +0200 schrieb Matthias Huber:
When i send more than one messages shortly after the other, my log
(/var/spool/asterisk/sms ) looks like this
and only two of four messages arrive.
What am i doing wrong ?
I am using an AVM B1 PCI with chan-capi and 1.4.4.
Is there something simple like gastman that provides functionality
to establishing conferencing?
--
Eric Smith
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On Fri, Jul 13, 2007 at 10:54:29PM -0500, RENZZO SOTOMAYOR wrote:
Hi! I am new here. Well I'm doing a call center using asterisk and I'm
looking for an open source screen pop software to pop the caller's
information, its call history and others things. i was looking around and
find the
Al,
If you don't mind I am actually writing to you about a little different
matter. I am new to the asterisk biz and am not too far from where you are
located in PA. I am interested in starting a VOIP business like your own and
was wondering how you are finding the market for new customers? I
Hi list,
I am searching for a possibility to do a certain call transfer method
which is called path replacement in QSIG. But I want to do that in
DSS1 (EuroISDN). If my asterisk does a call transfer I want the machine
to signalize on dchan that the call path has to be replaced to a direct
connect
Hi!
I am searching for a possibility to do a certain call transfer method
which is called path replacement in QSIG. But I want to do that in
DSS1 (EuroISDN).
They keyword to search for is explicit call transfer (ECT). At least
chan_capi-com (http://www.melware.org/ChanCapi) comes with
Philipp von Klitzing schrieb:
Hi!
I am searching for a possibility to do a certain call transfer method
which is called path replacement in QSIG. But I want to do that in
DSS1 (EuroISDN).
They keyword to search for is explicit call transfer (ECT). At least
chan_capi-com
Hi!
I am searching for a possibility to do a certain call transfer method
which is called path replacement in QSIG. But I want to do that in
DSS1 (EuroISDN).
They keyword to search for is explicit call transfer (ECT). At least
chan_capi-com (http://www.melware.org/ChanCapi) comes
how can I fix this just started ..
Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18
(Ring Begin)...
== Starting Zap/1-1 at bell,s,1 failed so falling back to exten 's'
== Starting Zap/1-1 at bell,s,1 still failed so falling back to
context 'default'
Jul 14 14:32:35
What do you mean establishing conferencing?
There is a page for meetme conferences in the gui...
-bk
- Original Message -
From: Eric Smith [EMAIL PROTECTED]
To: Asterisk Users Mailing List asterisk-users@lists.digium.com
Sent: Saturday, July 14, 2007 2:58:08 AM (GMT-0800) America/Tijuana
Dear Mojo;
Looking to the below example again, there are two
lines for s-NOANSWER and s-BUSY, one line with priorty
1 and other with priority 2, and both lines are
calling the Voicemail application, so the question is:
when it will jump to priority 2 for s-NOASNWER and
s-NOBUSY?
Last thing, like
Hi List;
[incoming]
include = parkedcalls
exten=103,1,Dial(SIP/Bob,,tT)
exten=104,1,Dial(SIP/Charlie,,tT)
When we use tT and when we use t alone or T alone, I
know this for call parking, but I do not know what the
tT does?
Regards
Bilal
On 16:28, Thu 05 Jul 07, Philipp Kempgen wrote:
Since the list was switched over to API-Digital almost
every message I get is older than a week. Coincidence?
Is anyone else having trouble?
Regards,
Philipp
I got this message today July 14
Yes, I have the same.
--
Michiel van Baak
Check out this gui for meetme.
http://sourceforge.net/projects/web-meetme/
On 7/14/07, Eric Smith [EMAIL PROTECTED] wrote:
Is there something simple like gastman that provides functionality
to establishing conferencing?
--
Eric Smith
___
So last night thanks to the help of the fine folks on this list I got
the newest Zap 1.4 and Asterisk 1.4 compiled and installed and
seemingly working except that after a few calls and a few keystrokes
I get a kernel panic. The first time I had just typed reload after
cleaning up some
Well, this is now the third active thread on this subject, but I guess
you won't see this message for a while. Has anyone dissected the
headers of a delayed message yet? We should be able to tell for sure
where the holdup is. All of the messages are coming through on time
for me, so it won't do
I lost one channel on an FXO module on a Sangoma A200 card due to a
lightening zap in the area (well - it died the same night as a major
thunder storm came through)Is there a recommended/standard
surge protector for phone lines I should be using? My server has 2
POTS lines.
I like ADM as it has a URL popup feature (open a URL with a DID or
CallerID in URL). The problem is that for each call, I tend to get 4
or 5 popups... But as the other author said, there are many
programs to choose from...
Todd
On Jul 13, 2007, at 11:54 PM, RENZZO SOTOMAYOR wrote:
Dave Donovan wrote:
On 7/10/07, *Stephen Bosch* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Jason Aarons (US) wrote:
Since many CLECs (Competitve Local Exchange Carriers in NA) offer
fractional PRI, combined with Internet/Data, I haven't seen any demand
for ISDN
Hi,
I work with
gnudialer
vicidal
Best Regards
On 7/14/07, Todd H [EMAIL PROTECTED] wrote:
I like ADM as it has a URL popup feature (open a URL with a DID or
CallerID in URL). The problem is that for each call, I tend to get 4
or 5 popups... But as the other author said, there are
Never mind the 1.2.18 messed and did not recognize the s extension any
more so I just upgrade to 1.2.21.1 and fixed the problem,.weird.
otis
OCOSA ListAcct wrote:
how can I fix this just started ..
Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18
(Ring
Has anyone come up with to timestamp a Recording? I am using a pretty
simple dialplan to record a audio file for a hotline. I'd like to
store the date and time it was recorded somewhere, Ast DB or MySQL DB.
Then when the audio file is played back to a caller, the system will
say something like.
Yes, just get a MSTAPI compliant program that integrates with asterisk.
On 7/13/07, Al lists [EMAIL PROTECTED] wrote:
I was wondering if any of you guys are aware of ability to call customers by
click on customer's phone number in ACT?
___
How about connecting it to UPS (uninterpretable power supply)
Standard of the shelf $40.00-$80.00 will do the trick.
--
#Joseph
On Sat, 2007-07-14 at 22:17 -0400, Todd H wrote:
I lost one channel on an FXO module on a Sangoma A200 card due to a
lightening zap in the area (well - it died
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