Yeah I know about A-D but can we send more than those !? I've read about
h245-alphanum thing but that is definitely not in asterisk, so what other
options are there is I've to send more than just A-D ?
On Tue, Feb 28, 2012 at 12:42 PM, Matt Darnell wrote:
> On Mon, Feb 27, 2012 at 8:23 PM, Sammy
On Mon, Feb 27, 2012 at 8:23 PM, Sammy Govind wrote:
> Hi list,
>
> What possibilities are there in asterisk to send an alphanumeric DTMF
> from/to asterisk !?
>
> Regards,
> Sammy
Do you mean A-D? You send those just like 0-9*#
-Matt
--
__
Hi list,
What possibilities are there in asterisk to send an *alphanumeric
DTMF*from/to asterisk !?
Regards,
Sammy
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live in
HI,
*Asterisk 1.8 *allows to read SIP response codes in the dialplan with *
${HASH(SIP_CAUSE,)}* - This is working fine for me too.
This is the dialplan line just after the DIAL()
same => n,NoOp(SIP return code : ${HASH(SIP_CAUSE,${CDR(dstchannel)}):4:3})
Ref: http://www.voip-info.org/wiki/view
Thanks Virendra,
That was a good start but not quite what I am looking for. I want to know
the details of listening to a specific event rather than listening to all.
My questions specifically is regarding the Manager command "originate". I
just used it to dial an extension first and then dialling
- Original Message -
From: Gaurav P
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Sunday, February 26, 2012 7:00 PM
Subject: [asterisk-users] pstn bridge to asterisk - phones connected to pstn
stop ringing when asterisk answers
Hi All,
I am using an Obi110 to bridg
Hello,
I'm running Asterisk C.3.7.2 and DAHDI 2.4.1.2
No changes in hardware since last April and asterisk version since last summer.
Since last Friday one of the call centers I support has been complaining of
lagged audio, dropped calls, and "two calls at once". Now I run 5 call centers
off a
On 02/24/2012 01:29 AM, Takehiro Matsushima wrote:
Hi, Sam
Yes, I’m understanding that the backend of AstDB is bdb or
sqlite(since asterisk10).
So, I suggested to place files of them on shared disk (like DRBD).
This is the solution that is documented in the user's manual for the
Digium R-Seri
Hello,
I am using a mix of Call files and AMI telnet from a perl app to place
calls. I sometimes get this in the CLI:
-- Attempting call on sip/551234@for 1@:1
(Retry 1)
[Feb 27 13:47:07] == Using SIP RTP CoS mark 5
[Feb 27 13:47:07] -- Got SIP response 503 "No Circuit Available"
Thank you Jonathan,
I already do the steps you mention, my configuration is:
in res_odbc.conf
enabled => yes
dsn => asterisk-connector
pre-connect => yes
in odbc.ini
[asterisk-connector]
Description = PostgreSQL connection to 'asterisk' database
Driver = Postgr
Maybe his google-fu is phisher-fu!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Monday, February 27, 2012 12:53 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] CDR An
On Mon, 2012-02-27 at 18:28 +, Noah Engelberth wrote:
> I’ve been tasked with finding and implementing a CDR/Queue analyzer to
> provide information to management about the call center’s performance.
> My Google-fu seems to be returning a lot of things that are more or
> less abandoned projects
Downgrade actually worked fine in this instance - from 1.8.9.2 to 1.8.9.1
and I concluded it wasn't Asterisk that was the issue. Thanks Patrick.
It would be great to keep this feature stable. As it helps in case of
regressions, etc...
On Mon, Feb 27, 2012 at 10:09 AM, Kevin P. Fleming wrote:
>
I've been tasked with finding and implementing a CDR/Queue analyzer to provide
information to management about the call center's performance. My Google-fu
seems to be returning a lot of things that are more or less abandoned projects.
Does anyone have any recommended solutions for a CentOS 6 /
The AstLinux team is happy to announce the release of version 1.0.2. This
release features several security updates. All current users are encouraged to
upgrade as soon as possible. Please see the documentation at
http://doc.astlinux.org for upgrade details or the official release pages.
Upd
Hi,
We heavily use meetme/SLA functionality in Asterisk, and continuously run into
issues with dahdi timing. The two errors we get are:
ERROR[6518] res_timing_dahdi.c: Failed to configure DAHDI timing fd for 0
sample timer ticks
WARNING[22024] app_meetme.c: Unable to write frame to channel
On 02/27/2012 09:05 AM, Jason Parker wrote:
On 02/26/2012 06:22 PM, Patrick Lists wrote:
On 25-02-12 19:47, Jason Parker wrote:
yum and rpm do not support downgrades.
Incorrect. There is yum downgrade. See man yum.
yum downgrade is extremely broken. It fails, often, potentially leaving a
On 02/26/2012 06:22 PM, Patrick Lists wrote:
> On 25-02-12 19:47, Jason Parker wrote:
>> yum and rpm do not support downgrades.
>
> Incorrect. There is yum downgrade. See man yum.
>
yum downgrade is extremely broken. It fails, often, potentially leaving a
system in an unrecoverable state. That
Hi Arstan¡¡¡
Great Job¡¡¡
All perfect with this ODBC version.
this is the first reference to this problem solved i found.
Best Regards and Thanks
El 27/02/2012 2:58, Arstan escribió:
I did install CEL logging for Debian 6. Here's my notes, I hope it helps
How to cel logging in asterisk 1.
Hi.
I am new to asterisk.
I have an ivr application with asterisk and voiceglue. I make a call from
asterisk (say to A) and when callee press a button voiceglue transfer the
callee to another number (say to B). When I look cdr records the billsec
between A and B always 0 and billsec with A shows th
On 27/02/12 00:00, Gaurav P wrote:
Hi All,
I am using an Obi110 to bridge my PSTN line to Asterisk. Inbound and
outbound calls work fine, but I noticed that phones connected directly
to the PSTN line stop ringing as soon as Asterisk answers and rings one
of my extensions. I'd like the regular ph
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