On Thu, Mar 15, 2012 at 12:09 PM, Richard Mudgett rmudg...@digium.com wrote:
I currently have an Asterisk 1.6.2.18 server running a patched (see
below) libpri 1.4.10.2 connected to a Toshiba Strata CTX670. All
external calls come in via the Strata and then are routed to the
Asterisk server
Hi,
I'm using SendText to send a text message when the user picks up a line in a
SLA setup (even though I'm not sure the problem is related to SLA). I'm on
Asterisk 10.2.1 (same in 1.8.9)
[from-office]
..
same = n,SendText(hi)
same = n,SLAStation(line1234)
..
Here is a simplified version of
Am 16.03.2012 04:14, schrieb Ast Coder:
I would be more interested in a system where quality routes are tested
with different providers because rate really doesn't matter if a call
can't be placed or if a destination is a fake one. We have seen many
fake destinations with top tier providers but
If we had reports of every call, we could downgrade status of routes that
had frequency calls not completed that were outside the norm.
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus
Sent:
On 03/16/2012 09:43 AM, Matt Hamilton wrote:
Hi,
I'm using SendText to send a text message when the user picks up a line
in a SLA setup (even though I'm not sure the problem is related to SLA).
I'm on Asterisk 10.2.1 (same in 1.8.9)
[from-office]
..
same = n,SendText(hi)
same =
snip
pri intense debug:
TEI: 0 State 7(Multi-frame established)
V(A)=31, V(S)=31, V(R)=42
K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
T200_id=0, N200=3, T203_id=8192
[ 00 01 54 3e 08 02 01 b3 62 1c 66 9f aa 06 80 01 00 82 01 00 a1
31
02 02 01 3a 02 01 0c 30 28 0a
- Original Message -
Greetings-
First off, my apologies for the slightly OT nature of this post. It
does involve Asterisk to a degree, but errs a bit on the side of
Audiocodes inquiry. I accept all responsibility for my actions and the
consequences. :)
The scenario is this: I
I had many of the same problems with sip station. If you just need sip
termination, Check out flow route. The service just seems to work properly
for me, and they respond to tickets. You can open up new cases through
their site.
On Mar 15, 2012 11:48 AM, Jake Wicke j...@nxtphase.net wrote:
Jake,
We are DIDForSale support asterisk. We do IP based authentication and do
not require registration. You can test our DIDs without paying anything. I
am sending you the rates just to make it easy to compare apple to apple, no
run around for pricing ;) . Let me know if I can have an opportunity
hi folks,
i was wondering if some one has a2billing script,which can be used to
install a2billing easly ?
thanks in advance
--
*
HARAZ Tahar
*
*Engineering Student at the National Institute for Posts and
Telecommunications (INPT)
*
*
Phone: +212 6 78030050
E-mail: harazta...@gmail.com
On 03/15/2012 02:38 PM, Warren Selby wrote:
Second, this is kind of outside the box thinking, so it may not work at
all, but try setting the NAT on that peer to no, and then tcpdump the
incoming registration attempts and see if you can see the internal
private IP address of the packet. If
We have had success with this:
https://sites.google.com/site/a2billing2asterisk/
On Fri, Mar 16, 2012 at 1:28 PM, Tahar .H harazta...@gmail.com wrote:
hi folks,
i was wondering if some one has a2billing script,which can be used to
install a2billing easly ?
thanks in advance
--
*
Hi,
I've tried upgrading one of my servers with yum update to the latest
dahdi/asterisk, and found out that my 4th gen TE410P is failing the dahdi init
with
Running dahdi_cfg: DAHDI startup failed: Input/output error
Rolling back to 2.5 restores the normal operation, and reading the dahdi
List,
I am getting this warning message in cli. How to fix this problem.
*translate.c:163 framein: no samples for ulawtolin
**
*Best Regards,
Mahesh Katta
http://www.buzzworks.com/
--
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