Hi lists,
Does anyone know if the following error message (from a debug screen) was
a
deliberate change from the behavior in asterisk V1.4.18 or just an
overlooked
parsing error in progressing to V1.6.0? Since, in this case, the string (Hi
there)
is quoted, it doesn't seem as though the
Hi,
I'm trying to use AddQueueMember() to add a member to a queue and trying to
make this logged member in the queue between reloads and restarts of
asterisk.
I configure en queues.conf:
[general]
Persistentmembers=yes
And Extensions.conf:
exten=
Check the number of calls waiting in the queue, then play the message
if
more than 0
example code (written in the TBird IDE)
Exten = 100,1,Answer()
Exten = 100,n,Set(NumWaiting=${QUEUE_WAITING_COUNT(${QUEUENAME})})
Exten = 100,n,GotoIf($[${NumWaiting} = 0]?JoinQueue)
Exten =
Hi All;
So what should I do now to remove that bug? Any
advise?
Regards
Bilal
---
Hence the reason I kept CCing your off-list emails
back to the list.
Guys like Tzafrir are aces.
(Just a reminder that if this is indeed the case then
this is a bug
inflicted by me and fixed
How do they get away with that?
On Mon, Apr 28, 2008 at 7:23 PM, Jonathan C. Bailey
[EMAIL PROTECTED] wrote:
Try the RPM from Trixbox. If you need something to open the file on Windows,
7zip works fine..
Asterisk didn't benefit much from having HT enabled on a P4
with HT capability.
There are several things that make a difference when optimizing for a
specific processor in order to take advantage of its features.
Gcc version used to build asterisk (and the system in general) and compile
Hello List!
We have problems setting the right caller id on outgoing calls. The
Asterisk Pbx is located
in Bucarest(Romania), our Telco provider is rcs-rds.ro. We have the
local telefon number
40787 00-99, associated to our PRI E1 Line. Where 00-99 are the DID
numbers available.
The telco is
On Mon, Apr 28, 2008 at 8:34 PM, Vieri [EMAIL PROTECTED] wrote:
How can I get a list of the callers within a specific
queue at any given moment?
I need to get the caller IDs of all active calls in a
queue then send them out via a udp socket to a
listening application on the network (the
--- Atis Lezdins [EMAIL PROTECTED] wrote:
On Mon, Apr 28, 2008 at 8:34 PM, Vieri
[EMAIL PROTECTED] wrote:
How can I get a list of the callers within a
specific
queue at any given moment?
I need to get the caller IDs of all active calls
in a
queue then send them out via a udp
On Tue, Apr 29, 2008 at 1:22 PM, Vieri [EMAIL PROTECTED] wrote:
--- Atis Lezdins [EMAIL PROTECTED] wrote:
On Mon, Apr 28, 2008 at 8:34 PM, Vieri
[EMAIL PROTECTED] wrote:
How can I get a list of the callers within a
specific
queue at any given moment?
I need to get
This may not be the right place to ask, but I have an annoying issue with
a Sipura/SPA1000-2.0.10(e) ATA device connected to an Asterisk box. (The
system is remote to me, so I've only been able to observe this by dialling
into a VoIP phone on-site, then run commands on the box remotely!)
Hi,
On Apr/23/2008, Steve Totaro wrote:
On Tue, Apr 22, 2008 at 7:10 AM, Carles Pina i Estany [EMAIL PROTECTED]
wrote:
Hello,
We have an Asterisk server with a TE410P Quad-Span togglable E1/T1/J1
card, 3 SPANs configured and OK and one SPAN unconfigured.
In our tests it
Greetings,
upgrading from 1.4.17 to 1.4.19 some asterisk gateway of ours (used
for gatewaying ISDN-PRI and SIP), I noticed an annoying thing: when
the PSTN party answers, for a few seconds (4/5 sec typical) some SIP
client could not hear anything (the ringing was heard well!), then the
audio
Also, you say no network issues but what is the rating of your
switches PPS (often overlooked for speed such as 100mb or 1000mb)?
100 Mbps, enough for 50 - 60 calls
Thanks,
I asked for PPS (packets per second) not Mbps they are very different.
Thanks,
Steve Totaro
Polycom is affiliated with the project in some way.. They also have an official
Polycom moderated vendor forum.
-Jon
- Original Message -
From: Andreas van dem Helge [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
Hi all,
i have found the two possible solution for doing RAS with Asterisk:
1) PPPD (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+PPPD)
I have downloaded the tgz proposed in the Wiki: app_pppd-060822.tgz
But it do not compiler under Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q.
First
Jerry Geis wrote:
I have xinet tftp running on centos 5.1
It seems to be running on the local network eht0 fine. My box has 2 nics.
however when I connect to eth1 for tftp I get:
in.tftpd[5084]: tftpd: read(ack): Connection refused
How can I get tftp working on BOTH eth0 and eth1 for my
Hello,
I'm selling Digium TE420B (Four Port E1 PRI PCI-E x1 with Echo
Cancellation), a brandly new one. The price is $1200. Urgent!
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To UNSUBSCRIBE or
An amazing change from the old days when you could only get firmware
from a Polycom authorized distributer.
Jonathan C. Bailey wrote:
Polycom is affiliated with the project in some way.. They also have an
official Polycom moderated vendor forum.
-Jon
- Original Message -
From:
Lol poorly moderated.post a question and then listen to crickets
waiting for an answer.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Tuesday, 29 April 2008 8:54 AM
To: Asterisk Users Mailing
Robert,
You can access CDR information within the dialplan using the CDR
variable. I'm doing something very similar with a DISA feature for our
employees. We use ODBC to validate them against an existing MSSQL server
(check their employee ID pin number) then when all is well, I write
some
On Tue, Apr 29, 2008 at 02:36:36PM +0200, Tobias Wolf wrote:
Hi all,
i have found the two possible solution for doing RAS with Asterisk:
1) PPPD (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+PPPD)
I have downloaded the tgz proposed in the Wiki: app_pppd-060822.tgz
But it do
Hello:
I have asterisk configured to use sip with two providers.
Checking with the command
sip show registry
I found that sometimes is not registered.
?Is it there anyway to configure asterisk to restablish the connection with the
providers automatically?
Thanks in advance for any answer.
Lee, John (Sydney) wrote:
Check the number of calls waiting in the queue, then play the message
if
more than 0
example code (written in the TBird IDE)
Exten = 100,1,Answer()
Exten = 100,n,Set(NumWaiting=${QUEUE_WAITING_COUNT(${QUEUENAME})})
Exten = 100,n,GotoIf($[${NumWaiting} =
This is my understanding of hyper threading, which I believe to be accurate.
Basically, as some have mentioned previously, the OS 'sees' your single
physical core processor as 2 logical processors, in generally, logical
processors are treated exactly as if they were real processors, and in the
Hello all
As always I'm trying the mailing list as a last resort as I'm out of
options. I am seemingly unable to dial international numbers over our BT
ISDN30 line.
I've checked with BT and the number format they're expecting is:
00CCnumber
(where CC is the country code).
But this
Hi All,
I'm trying to debug DTMF issues I have with certain endpoint
conferencing systems (external, 3rd party).
On our A*k server I log DTMF, and I see that coming through in the log.
What I'd like to see is what is sent onto our VoIP carrier over SIP.
I can do a tcpdump of the
You might need to set the dialplan to international or so in the config
files.
Zoa
Stuart Ford wrote:
Hello all
As always I'm trying the mailing list as a last resort as I'm out of
options. I am seemingly unable to dial international numbers over our BT
ISDN30 line.
I've checked with
Many people think ZapRAS is for modem dialin. None of the RAS stuff
support modems, as far as I know. The RAS stuff in Asterisk is for
networking via ISDN, rather than modem.
Tzafrir Cohen wrote:
On Tue, Apr 29, 2008 at 02:36:36PM +0200, Tobias Wolf wrote:
Hi all,
i have found the two
See below
In article [EMAIL PROTECTED], Stuart Ford [EMAIL PROTECTED] wrote:
Hello all
As always I'm trying the mailing list as a last resort as I'm out of
options. I am seemingly unable to dial international numbers over our BT
ISDN30 line.
I've checked with BT and the number
Hello! As the subject states, I'm experiencing an odd issue with one of my
client's systems. We're running a Sangoma A400D card with 8 FXO channels. The
POTS line on channel 5, whenever called, generates the following message on the
asterisk console:
Apr 29 10:35:08 WARNING[16927]:
2008/4/29 Tony Mountifield [EMAIL PROTECTED]:
[snip]
What values do you have in zapata.conf for pridialplan, internationalprefix,
nationalprefix and localprefix?
Try each of the following two sets of parameters:
(A)
pridialplan=dynamic
internationalprefix=00
nationalprefix=0
On Tue, Apr 29, 2008 at 05:00:26PM +0100, Steve Davies wrote:
2008/4/29 Tony Mountifield [EMAIL PROTECTED]:
[snip]
What values do you have in zapata.conf for pridialplan,
internationalprefix,
nationalprefix and localprefix?
Try each of the following two sets of parameters:
(A)
Eric Wieling schrieb:
Many people think ZapRAS is for modem dialin. None of the RAS stuff
support modems, as far as I know. The RAS stuff in Asterisk is for
networking via ISDN, rather than modem.
This is exactly what we want ZapRAS to use for ;)
Maybe anyone can enlighten me in
Alexander Lopez schrieb:
If you want to do RAS (modem or data) I would suggest going with a
Portmaster PM3(Livingston, then purchased by Lucent, They are reliable
and pretty cheap. You can use Asterisk to route the calls into it if you
use a two port card.
You can find them at
Tzafrir Cohen schrieb:
On Tue, Apr 29, 2008 at 02:36:36PM +0200, Tobias Wolf wrote:
Hi all,
i have found the two possible solution for doing RAS with Asterisk:
1) PPPD (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+PPPD)
I have downloaded the tgz proposed in the Wiki:
The thing is that's still the case. If they really wanted change
they'd post the newest version of the firmware at www.polycom.com
which they dont (well technically yes but you need to be a member of
their reseller crap)
The byproduct of the corporate bureaucracy,. Isn't it great?
On Tue, Apr
Thanks guys.
I understand the ODBC and channel variables using Asterisk. However,
I am really looking for an ability for a webserver to subscribe to
channel status in asterisk and be informed when a call comes in and
show the callerID in real-time.
From the AMI, there is Newcallerid Event which
--- Atis Lezdins [EMAIL PROTECTED] wrote:
So, I suppose if
MySQL dies in
middle of operation, SELECT should fail and Asterisk
should just
continue with what it has in memory. Btw, You should
be able to also
use static or dynamic queue members (not realtime)
in combination with
realtime
Hi,
im trying to config ooh323 in asterisk. i compiled the one inside the
asterisk-addons im trying to connect my softphone(Xmeeting).
here's my ooh323.conf
[marq]
type=friend
context=avaya
ip=dynamic
port=1720
e164=888
username=marq
secret=marq
--
Regards,
Mark Quitoriano
On Tue, Apr 29, 2008 at 1:02 AM, Patrick
[EMAIL PROTECTED] wrote:
On Mon, 2008-04-28 at 14:49 -1000, Matt Darnell wrote:
Anyone seen anything on the IP670 the Color Expansion?
Great timing. Yesterday I was looking at the IP650 and wondered when the
successor to the IP650 would arrive.
Is Sugar CRM the best Free CRM to be integrated with Asterisk ?
Is Asterisk VoiceRD Integration the best integration patch to be used
with Sugar CRM ? Is any other ?
Regards,
Fernando
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Is Sugar CRM the best Free CRM to be integrated with Asterisk ?
Is Asterisk VoiceRD Integration the best integration patch to be used
with Sugar CRM ? Is any other ?
Regards,
Fernando
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On 18:16, Tue 29 Apr 08, Fernando Berretta wrote:
Is Sugar CRM the best Free CRM to be integrated with Asterisk ?
Is Asterisk VoiceRD Integration the best integration patch to be used
with Sugar CRM ? Is any other ?
Have a look at Covide: http://sourceforge.net/projects/covide
/shameless_plug
vicidial ... vicidialNOW
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Hi,
Asterisk 1.4
Working (jitter buffers created as expected):
ZAP - SIP
SIP - ZAP
Not working (no jitter buffers created):
SIP - chan_local (with /nj) - ZAP
SIP - chan_local (with /j) - ZAP
SIP - chan_local (with no flags) - ZAP
I have this in zapata.conf:
jbenable=yes
jbforce=no
jbimpl=fixed
I have a PRI connected to a traditional PBX using NI-2 and a typical
config (further below). When I call from a SIP/IAX phone to an extension
on the PBX, only the number makes it through. If I plug that same port on
the PBX to a carrier the PBX presents both name and number.
Hints or pokes to
Thanks Olle and Jared for your reply. This clears a lot of my doubts.
Thanks once again.
Regards,
Aadil
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Johansson
Olle E
Sent: Wednesday, April 23, 2008 8:01 PM
To: Asterisk Users Mailing List -
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