You have to create yourself the odbc commands using func_odbc.conf file,
like:
[GET_HUNTLIST_TYPE]
dsn=asterisk1,asterisk2
synopsis=Get the Hunt List type
readsql=SELECT hu_type,hu_ringtime from hu_huntlists where hu_id='${ARG1}'
then you cna use it in your dialplan (I use AEL):
Dne 30.4.2012 11:09, Bharat Lalcheta napsal(a):
Hiii all,
I am using asterisk 1.8.9.2 and compile all modules related to calendar.
neon version is 0.29.6. OS is ubuntu 11.10.
I configured ical for zimbra, caldav for google mail and ews for
exchange 2010 calendar.
ical and caldav setup
On Fri, May 04, 2012 at 09:24:56AM -0700, bilal ghayyad wrote:
What is happening with me that when I used fedora core 16, I compiled
and installed dahdi 2.6 and then compiled and installed asterisk 1.4
and it did not create chan_dahdi. I tried to select it by running make
menuselect and I
On Fri, May 04, 2012 at 08:34:49PM +0200, Jonas Kellens wrote:
Hello,
what does it mean when you read in the backtrace file :
Reading symbols from /lib64/libgcc_s.so.1...(no debugging symbols
found)...done.
No debugging symbols are avaialble for libgcc_s . Libgcc is an external
library,
On Fri, May 04, 2012 at 11:53:41PM +0200, Jonas Kellens wrote:
I have selected don't optimize in the menuselect for better
information in the trace and now you tell me that it's still useless
?
(As mentioned in a separate post, this is not related to the missing
debug symbols from the
On 30-04-12 11:09, Bharat Lalcheta wrote:
Hiii all,
I am using asterisk 1.8.9.2 and compile all modules related to calendar.
neon version is 0.29.6. OS is ubuntu 11.10.
I configured ical for zimbra, caldav for google mail and ews for
exchange 2010 calendar.
ical and caldav setup working
Hello,
I am having a problem with SendDTMF - it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.
I use software phone to test it... when I dialed 501, I cant hear anything
for about 10
Am 06.05.2012 13:46, schrieb Shahid H:
Hello,
I am having a problem with SendDTMF - it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.
Log the actual DTMF to your console, set in
Am 06.05.2012 13:46, schrieb Shahid H:
Hello,
I am having a problem with SendDTMF - it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call.
PS: You are only allowing the GSM codec for your
Try using Dial(SIP/+44797XX@voipms,30,D(ww0788XX)t)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shahid H
Sent: Saturday, May 05, 2012 11:20 PM
To: asterisk-users@lists.digium.com
Subject:
Thanks for the suggestion Markus. Here what I did:
In the logger.config I have added 'dtmf':
console = notice,warning,error,dtmf
and then in sip.conf:
allow=ulaw
allow=alaw
; allow=gsm
dtmfmode=inband
I've added a test to call my mobile:
exten =
Now you have a totally different issue. 8-)
While the call is up do a sip show channels in the CLI. This will show you
the ACTUAL codec for the call. Likely the call was still using GSM. Did you
remember to put a disallow=all before the allow= lines?
I recommend dtmfmode=rfc2833 with
Hii All,
Thanks for helps. Problem solved. Its due to exchange side authentication
problem. In exchange IIS server ,for EWS site only window authentication
enabled which is not supported by neon library. Just enabled basic
authentication for EWS site on IIS and its working fine now.
Thanks for
When I changed back to dtmfmode=rfc2833 and I cant hear the DTMF
sound.. completely silent.
Indeed I have put disallow=all before the allow=ulaw allow=alaw
sip show channels in the CLI show during a call:
78.129.xxx.xx +447715d909406db14d2 0x4 (ulaw) No
Tx: ACK
Here is another debug log:
== Using SIP RTP CoS mark 5
-- Executing [123@test2:1] Dial(SIP/test2-0008,
SIP/+44776@voipms,,D(1ww2ww3ww4)) in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/+44776XX@voipms
-- SIP/voipms-0009 is making progress
I have a Digium TDM400P card that appears to have died. The first noted
symptoms were that dahdi would fail to reload on boot. On closer
inspection, the card looks totally dead; no lights on at all. I have
tried moving it to a different PCI slot, and removing the other PCI card
(a 3com 10/100 NIC)
On 05/06/2012 11:42 AM, Greg Woods wrote:
Second, since the parts of this card are very expensive, I am wondering
if these symptoms likely mean that the main board of the card is dead,
but the FXS and FXO modules might still be good. In that case, I could
just get a new main card and move the
On Sun, May 6, 2012 at 12:42 PM, Greg Woods g...@gregandeva.net wrote:
I have a Digium TDM400P card that appears to have died. The first noted
symptoms were that dahdi would fail to reload on boot. On closer
inspection, the card looks totally dead; no lights on at all. I have
tried moving it
Hey guys,
I have managed to get to work Thanks for the help..
I just registered a new account at sipgate.co.uk and test it on asterisk...
and DTMF worked well :)
It seem voip.ms dont work well when sending DTMF to UK.
Do anyone know UK/Europe voip provider to allow you change any callerID
On Sun, 2012-05-06 at 12:46 -0400, Andrew Latham wrote:
Sounds like you did a kernel update and did not rebuild DAHDI.
I haven't done a kernel update on this particular machine in quite some
time, since long before the card started failing. It is still running
Fedora 14, so there aren't even
Very occasionally in my logs I see things like this. In this case 7
lines starting with the first line and each ending with one of the
group of 7. Took about 10 seconds for the 7 tries.
[2012-05-03 16:58:27] NOTICE[31850] chan_sip.c: Sending fake auth
rejection for device unknown
I'm about to receive approval to design and deploy an Asterisk-based
phone system for my company. I will immediately have to start writing
specifications. I'm working on the hardware design and the architecture
right now. I'd like a second, third, fourth, 1,000th opinion.
800 SIP phones. All
On 12-05-06 02:00 PM, Nunya Biznatch wrote:
I'm about to receive approval to design and deploy an Asterisk-based
phone system for my company. I will immediately have to start writing
specifications. I'm working on the hardware design and the architecture
right now. I'd like a second, third,
For 100% High Availibility and Hot Failover, I would recommend one of those
Red-fone Fonebridges.
Also getting 800 Phones all register on single server is crazy, add a SIP
proxy to distribute load evenly between 2 Ast boxes.
For Wireless you might consider using DECT phones from Snom instead of
I have deployed multiple installations with 1500 phones per server using
standard HP DL-380's.
It's not that crazy. They can handle it pretty easily.
Antonio.
Op 06-05-12 23:19, Mitul Limbani schreef:
For 100% High Availibility and Hot Failover, I would recommend one of those
Red-fone
Thanks for the info. It got me digging deeper. I definitely don't want
to screw this one up, but I've got to pinch pennies to get this done, so
don't want to buy anything that would just be nice to have. ...but if I
have to get it, that's what I'll do.
Have any of you seen this?
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