2008/1/10, IT-Connect [EMAIL PROTECTED]:
stoffell schrieb:
Has anyone been able to get ISDN-BRI support to work reliably on
Asterisk 1.4? If so, I'd love to know how you did it (hardware,
distro, kernel, modules, versions, config files).
Maybe your best bet is using bristuff, the
Hi,
When sending a PDF file to an ISP email2fax service, I something get errors
in return (1 miss out of 3).
Once, I striped the document I intended to send from any logo or complex
formatting to have it successfully sent.
I couldn't spend time trying to pine point which element in my document
Hi guys,
I need to interconnect an asterisk server (1.4.17) with an alcatel pbx,
because i want that an asterisk extension can call an alcatel extension
and viceversa. I need to do this throug a NT port (of the digium card)
for technical reasons (linked with alcatel hardware).
Card
Kev,
Change the HOST from sip.nsw.iinet.net.au to 203.215.3.1
Also, see:
http://forums.whirlpool.net.au/index.cfm?a=wikitag=iiNetPhone_asterisk
Sincerely,
Trevor Hammonds
-Original Message-
From: Kev S
Sent: Thursday, January 10, 2008 9:28 PM
S N I
Dear friends of cyber-telecom,
How are you doing these days? Cyber-Telecom.net website has now got a new
face-lift and as well as that we have added many new products.
As existing customer, we are offering you a one time 20% off discount for
all our product ranges in return for the
Everyone into VoIP and asterisk is welcome to what may be the largest
weekly gathering of Asterisk users worldwide:
http://VoipUsersConference.org
During the conf, we monitor IRC channel #voip-users-conference on Freenode.net
Community site (blogs, forum, video, etc) http://food4wine.ning.com
Hi,
Is full rfc4662 (SIP Resource Lists notification) support planned for 1.4 or
1.6 ?
see http://bugs.digium.com/view.php?id=10354
Now 7.1 Snom phones also support it.
Regards
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Just tried, Same issue unfortunately :(
Sorry if this message came through multiple times, Email client being silly
Regards
Kev
On 1/11/08, Paul Hales [EMAIL PROTECTED] wrote:
Are they expecting numbers in a 61 format?
PaulH
On Fri, 2008-01-11 at 16:27 +1100,
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
What is the Dial command you use?
Can you post the relevant part of your diaplan?
exten = _X.,1,Dial(Zap/g0/[EMAIL PROTECTED],,r)
In addition: are you sure that there are channels set for group=0 ?
Maybe try a channel directly: Zap/1 or Zap/2 instead
Hello,
if you give more information about your configuration, it will be
easier to help you.
Are both clients on the same network or are they separated by a NAT?
What is the configuration of these SIP clients in sip.conf? Do you
experience these delays with other soft phones, with non-SIP
hi, all,
can anybody tell me how to be a part of asterisk developer team.
I am so much intersted.
thnks in advance.
Bhrugu Mehta
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To UNSUBSCRIBE or
Hi,
I have a question about the definition of agents.
The agents.conf file looks like this:
[general]
persistentagents=yes
[agents]
maxlogintries=5
ackcall=no
wrapuptime=500
musiconhold = default
group = 1
agent = 1311,1311,Tom
agent = 1531,1531,Tim
and here is the queues.conf:
[general]
Hello everybody,
is it possible to execute some dialplan code when an extension changes its
state?
Say that SIP/204 becomes busy. I know that using the hint priority some phones
can show its state. Am I able to call custom code in the same way?
Thanks in advance,
--
Dr. Andrea Spadaccini
Friends,
In April, I'm organizing a conference here in Sweden about the new IP
Communications platform - SIP, Instant messaging, audio, video, text.
Open Standards, Open Source. Asterisk is part of a new platform that
we're building. The new version, 1.4, integrates with Jabber and
Doug wrote:
At 14:54 1/10/2008, John Millican wrote:
Hello all,
I know this has been discussed before but I am not finding the thread on
voip-info or site:lists.digium.com through google.
I have a site with * ver. 1.4.15 (started out as 1.4.2 or so) running on
openSuSE 10.2, Dual core AMD
Hi,
I just had my production box deadlocked - no calls could go trough,
CLI didn't load. Last lines in log were:
[Jan 11 09:15:43] VERBOSE[7265] logger.c: -- Executing
[EMAIL PROTECTED]:40] GotoIf(SIP/204.11.200.152-c0070ed0, 1?41:57)
in new stack
[Jan 11 09:15:43] VERBOSE[7265] logger.c:
Hi everyone,
I'm new to this list, so I don't know if this issue has been posted before.
My problem is with the Authenticate command.
From alle internal SIP phones the authentication is working just fine,
but when a call is comming from my IAX2 trunk,
where the calls should come from, the
Stefan Guenther wrote:
Hi,
I have a question about the definition of agents.
The agents.conf file looks like this:
[general]
persistentagents=yes
[agents]
maxlogintries=5
ackcall=no
wrapuptime=500
musiconhold = default
group = 1
agent = 1311,1311,Tom
agent = 1531,1531,Tim
Hi All
I am getting some delay while taking with software phone. I am using Xlite
software phone in both side. Please help me to reduce this delay.
Regards
Niru
Be a better friend, newshound, and
OK. I will continue this thread. I have learned a lot through a lot of
tcpdumps. So I am top posting so new understanding does not get hidden.
Senario:
Asterisk publicly addressed behind a firewall. Two different firewalls
available: Linksys WRT54G running sveasoft and Centos using
C F wrote:
You can use app_page.
If you call a local channel that uses app_chanisavail first then you
should be able to call as many as you need to.
How do you make chanisavail work for SIP devices? On a system we tried
to get paging to work it always returns status 0 (Unknown) for all SIP
On Fri, Jan 11, 2008 at 03:53:11PM +0100, Jaap Winius wrote:
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
What is the Dial command you use?
Can you post the relevant part of your diaplan?
exten = _X.,1,Dial(Zap/g0/[EMAIL PROTECTED],,r)
In addition: are you sure that there are channels
-- Forwarded message --
From: Brian Hutchinson [EMAIL PROTECTED]
Date: Fri, 11 Jan 2008 12:22:09 -0500
Subject: Trying to build MFC/R2
Hi Moy,
I downloaded your astunical-1.4.16 and have been studying it.
I have a machine running the latest download of asterisk-1.4-17 and was
Hans Witvliet schrieb:
On Thu, 2008-01-10 at 08:35 +0100, IT-Connect wrote:
stoffell schrieb:
Has anyone been able to get ISDN-BRI support to work reliably on
Asterisk 1.4? If so, I'd love to know how you did it (hardware,
distro, kernel, modules, versions, config files).
On Thu, 2008-01-10 at 08:35 +0100, IT-Connect wrote:
stoffell schrieb:
Has anyone been able to get ISDN-BRI support to work reliably on
Asterisk 1.4? If so, I'd love to know how you did it (hardware,
distro, kernel, modules, versions, config files).
I've run Asterisk
Hello asterisk users!
Does asterisk (or any of its plugin) support MRCP to interface with 3rd
party professional TTS engine?
I've found very little information about this, for example:
http://www.zen-turkey.com/blog/default.aspx?id=12t=Asterisk-MRCP-Integr
ation
Thanks in advance!
Regards,
Hi,
ATAL: Error inserting ztdummy
(/lib/modules/2.6.22-14-386/misc/ztdummy.ko): Unknown symbol in
module,
or unknown parameter (see dmesg)
Are you sure that the source of your kernel is the same as the running
kernel?
I.E. Have a look at the source it is using while compiling
I set my queues up using the SIP device and added the following into my
extensions_internal.conf file
;Pause/unpause Queue
exten = 424,1,PauseQueueMember(|SIP/${CALLERID(num)})
exten = 424,2,Playback(unavailable)
exten = 424,3,Hangup
exten = 425,1,UnPauseQueueMember(|SIP/${CALLERID(num)})
exten =
[EMAIL PROTECTED]
;Pause/unpause Queue
exten = 424,1,PauseQueueMember(|SIP/${CALLERID(num)})
exten = 424,2,Playback(unavailable)
exten = 424,3,Hangup
exten = 425,1,UnPauseQueueMember(|SIP/${CALLERID(num)})
exten = 425,2,Playback(available)
exten = 425,3,Hangup
I do something similar but
Quoting Tzafrir Cohen [EMAIL PROTECTED]:
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/1000-081f68b0,
Zap/g1/[EMAIL PROTECTED]||r) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/[EMAIL PROTECTED]
Again, you're calling an incorrect number. You dial
Hi All,
We have some polycoms hardphones. The exten 6610 is being recorded
with MONITOR cmd () as below:
exten =6610,10,SetVar(CALLFILENAME=/usr/local/apache2/htdocs/x/${UNIQUEID})
exten = 6610,11,Monitor(gsm,${CALLFILENAME},m)
exten = 6610,12,Dial(SIP/6601,30,tT)
On Fri, Jan 11, 2008 at 05:21:27PM +0100, Stefan Guenther wrote:
BTW: Do I need a soundcard to use the local channel in a queue?
Asterisk forwards calls to agents like this:
-- Executing [EMAIL PROTECTED]:1] Answer(Local/[EMAIL PROTECTED],2, ) in
new stack
No. A sound card is not needed.
Robert Moskowitz wrote:
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9 times? (Ok, I
I have followed all the stuff in that guide, and changed the ip address but
still having the same issue :(
Regards
Kev
On 1/11/08, Trevor G. Hammonds [EMAIL PROTECTED] wrote:
Kev,
Change the HOST from sip.nsw.iinet.net.au to 203.215.3.1
Also, see:
Also, Just letting everyone know i fixed it.
I needed to add a custom registration string in sip.conf
Thanks !
On 1/11/08, Trevor G. Hammonds [EMAIL PROTECTED] wrote:
Kev,
Change the HOST from sip.nsw.iinet.net.au to 203.215.3.1
Also, see:
A good place to start is where any developer would start, at the site
for the project they want to work on. http://asterisk.org/developers You
can also get there by clicking on the giant link that says code for
asterisk on the front page.
Anthony
Bhrugu Mehta wrote:
hi, all,
can anybody
hi, all
I am using asterisk 1.2.12.1 and zaptel 1.2.7 and libpri 1.2.1 version.
I have created Ivrs(very big) .It works fine in sip phone , but when i call
through zaptel digit sens problem occured. Asterisk doesn't sens any digit
pressed.Our pstn is CORAL pbx.
any suggesion..
thnks,
Bhrugu
Hi Moy,
Bottom line up front (yes I'm military):How do I now configure some of my
spans as normal Euro ISDN and a few with R2? Do I use a combination of
/etc/asterisk/unicall.conf and /etc/zaptel.conf or just unicall.conf for
everything?
See below for the rest of the details and thanks in
On Thursday 10 January 2008 16:52:11 Norman Franke wrote:
How does one get asterisk to timeout realtime request via res_odbc to
unixODBC? I've set timeouts as appropriate for freetds (which
unixODBC is using.) However, it doesn't seem to work. It takes over 3
minutes to timeout a connection
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