thank you evryone for you help and support
i make this command lspci -d d161:* and i get :
09:02.0 Communication controller: Digium, Inc. Wildcard TE210P dual-span
T1/E1/J1 card 3.3V (rev 02)
that is mean i have one card diguim with 2 dual span that is right ??
please advice
2011/9/26 JT
Monday, September 26, 2011, 11:33:50 PM, Kristijan wrote:
Gergo, why do you want to use mISDN? Use Dahdi. Or do you want to
use a very exotic isdn card which is only supported by mISDN? tell
us more.
Well, I want to use one channel driver for all my installations. Now I
have to reinstall a
dahdi show status?
2011/9/27 salaheddine elharit salah.elharit...@gmail.com
thank you evryone for you help and support
i make this command lspci -d d161:* and i get :
09:02.0 Communication controller: Digium, Inc. Wildcard TE210P dual-span
T1/E1/J1 card 3.3V (rev 02)
that is mean i have
there is no dahdi i have asterisk 1.4 there is a zaptel instaed dahdi
when i put dahdi show status i get command not found
regards
2011/9/27 Anton Kvashenkin anton.juga...@gmail.com
dahdi show status?
2011/9/27 salaheddine elharit salah.elharit...@gmail.com
thank you evryone for you help
Dears;
Yes, I tried all what mentioned, and did not work ... So I am contacting the
service provider.
Thanks a lot.
Regards
Bilal
---
Hi
try to add this
exten = _90Z,1,Set(CALLERPRES()=allowed)
exten = _90Z,n,Set(CALLERID(num)=5631040)
if you get
Dears;
I am facing now a problem in the recording the calls that coming via the queue,
the problem that I am not able to make the filename contains the agent (for
example its extension) who received the call.
Actually by looking to the below settings, it is clear that the agent name (it
the
Right -- I've been wrestling with this problem for over a year now.
Tinkering with modules.conf load order, noload= / preload= etc I even
modified the internal priority of the timers in the C code of the timing
modules so that after a reload the pthread timer would still be at a higher
prio than
I am starting up 4 calls over the AMI.
It appears as though the first 3 start up and go out right away.
The 4th call is delayed like 15 seconds.
Any thoughts on why this fourth call might be getting delayed...
Everything is working its just delayed.
Jerry
--
Hi Nick,
Understand your reasoning - though as Matt points out sql db isn't in the
core so compiling it there would preclude seemless upgrades. Also, I
personally would be concerned putting the calls right into the call-file
thread might create an issue if you hung on a db query or insert.
Hello David,
At first I assumed asterisk used call files out of the box for
normal-initiated/instantiated calls however,
this is incorrect. I think call files was the easy approach for client
just to place a file with call details
in some location. I am trying to do the same with a db record. My
On Tue, Sep 27, 2011 at 09:01:32AM +, salaheddine elharit wrote:
thank you evryone for you help and support
i make this command lspci -d d161:* and i get :
09:02.0 Communication controller: Digium, Inc. Wildcard TE210P dual-span
T1/E1/J1 card 3.3V (rev 02)
that is mean i have one
On Tue, 2011-09-27 at 03:47 -0700, bilal ghayyad wrote:
Dears;
I am facing now a problem in the recording the calls that coming via the
queue, the problem that I am not able to make the filename contains the agent
(for example its extension) who received the call.
Actually by looking
On Sun, Sep 25, 2011 at 10:59:29AM -0700, bilal ghayyad wrote:
Yes I tried Set(CALLERID(num)=5631040) as shown in the below dialing,
and no success. Also I tried Set(CALLERID(num)=1040) and I tried
Set(CALLERID(num)=065631040) as the city code is 06 and when we call
any mobile, it is appearing
ok thanks a lot for your support
2011/9/27 Shaun Ruffell sruff...@digium.com
On Tue, Sep 27, 2011 at 09:01:32AM +, salaheddine elharit wrote:
thank you evryone for you help and support
i make this command lspci -d d161:* and i get :
09:02.0 Communication controller: Digium, Inc.
Danny's suggestion of using System instead of AGI was correct, it was wrong of
me to use AGI here as there is no communication. The same script does
communicate with asterisk when executed with different options, so I had just
kept the same line using different options at different places.
On Tue, 27 Sep 2011, Mehmet Avcioglu wrote:
Today I changed the AGI calls to System at places where there is no
response back to asterisk. Kept everything the same, instead of
AGI(script.php,var1,var2) made it System(/path/script.php var1 var2)
and it worked without a problem. This way I have
The Asterisk Development Team is pleased to announce the second beta
release of
Asterisk 10.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
With the release of the Asterisk 10 branch, the preceding '1.' has been
removed
from the
Hi,
I have a grandstream HT 503 ATA which seems to support TLS/SRTP. I
generated the keys and certificates for both the server and the device
as mentioned at
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
I then copied the grandstream.crt and grandstream.key over to the ATA.
Hi,
It seems there is random behavior that causes screening mode to be
activated when a user calls and the line answered and then forwarded using
a dial command such as:
EXEC Dial SIP/13365551212@8x8|60SIP/13365541212@8x8
|60SIP/13365531212@8x8
Hello Everyone,
I was wondering if there is an OCF resource agent already created for
Asterisk, allowing for it to be able to include nodes in
a pacemaker cluster? This is for a corosync/pacemaker cluster.
Cheers,
Nick.
--
_
make sure the option '|60' is only included after the devices, IE. at the
end of the dial.
Dial(SIP/13365551212@8x8
mailto:SIP/13365551212@8x8SIP/13365541212@8x8SIP/13365531212@8x8|60|dgF(c
allFlo-in^3^1)M(record^39ff6274-c0f0-453d-aa05-402a7bd6d567
Hi all and thanks for reading.
I am having a very strange issue. When dialing out with a certain
carrier, asterisk 1.6.20 will play music on hold instead of a ring
tone, although this behaviour is NOT what I want.
Example dialplan execution:
-- Executing [0021266xxx@test:13]
Has anyone written a C wrapper to ease development with the AMI? I found a
couple of c++ ones, but not C.
Thanks!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Are you looking for just a parser? A parser + state machine? Or a complete
service that entails those components plus some sort of high-level API that
exposes them to outside callers?
--
This message was painstakingly thumbed out on my mobile, so apologies for
brevity, errors, and general
Thanks All. Here is my config:
*On my Firewall NAT:*
/I allowed the following ports: 4569,5004-5082, 1-2/
*
On Asterisk Box:*
Here is the extensions.conf:
/[general]
static=yes
autofallthrough=yes
[avaya-internal]
exten = s,1,Answer()
exten = s,2,background(pls-entr-num-uwish2-call)
If you can post any relevant code sections and CLI output for this then
it'll be lot better to determine whats causing this. I never got any problem
initiating as many call as u can say from AMI !
On Tue, Sep 27, 2011 at 5:36 PM, Jerry Geis ge...@pagestation.com wrote:
I am starting up 4 calls
Correct me if I'm wrong or don't know anything other than AMI Originate
Event or a call file to kick start a call from asterisk ! So making a new or
modifying asterisk call-file cron job/poller seems like a nice idea but why
put on extra load on Asterisk. (See pbx_spool.c if still want to modify).
:P I'd this very similar situation/ project Carl - and guess what. The
filename is created before the call actually hits QUEUE application so these
Queue variables are not populated by then so filename won't contain the
Agent Number.
UNLESS you move the file after queue to a new filename
Very strange indeed! post the dialplan lines as well. Seems like a very
normal Dial command execution. Also complete SIP packets for this particular
behaviour can show some insider. Which version of Asterisk you are using?
On Wed, Sep 28, 2011 at 6:44 AM, Alejandro Recarey
I see a couple of conflicting extensions as well as something I assume
copy-paste malfunction. Please paste the CLI logs of the call.
On Wed, Sep 28, 2011 at 8:26 AM, Malvin Rito
mr...@mail.altcladding.com.phwrote:
Thanks All. Here is my config:
*On my Firewall NAT:*
*I allowed the
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