Thanks a lot..
ALl the information i need and much more:)
very useful:)
Oguzhan Kayhan escribió:
I mean..making a single trunk between a pstn or telco with 2 or more
PRI's..
I mean instead of using 32 channels to use 64 or more..
I am trying to increase the capacity between my PSTN and
Hi,
In 1.6.1, it seems Attended Transfer do not behave exactly behave like Blind
Transfer when transferer hangs up before callee answers :
- in Blind Transfer, caller (ie transferee) is hearing Ringing tone when
callee's phone is ringing
- in Attended Transfer, caller (ie transferee) is hearing
Hi
I would like the option to set the codec used on a call by call basis.
I have a tdm410 2fxs + 1fxo.
when I make calls to my vsp, they go through as ulaw, I am guessing
because I have allowed if for the vsp (g729, alaw and ulaw).
I would prefer to use g729 from the fxs to the vsp but I would
Hello,
Is there a way to remove a global variable from dialplan ?
Regards
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Hi,
I've just discovered IMPORT function existence.
It can be use to import values from channel's Variable section but
unfortunately, il can't be use to access to values from Info section
(I'm referring here to sections Info and Variables dumped by DumpChan
application).
Is there a way to work
Hi,
I've read this :
http://www.trixbox.org/forums/vendor-forums-certified/aastra-endpoints/9143i-re-map-transfer-key
Has anyone tried this ?
I would be very happy to get few more details on how to do this.
Regards
___
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Hello,
I have asterisk 1.6.1.1 box behind NAT. On the same local network I've
SIP proxy server too. The problem appears with RTP.My provider's RTP IP
addresses are public. When asterisk sends SIP invite to SIP proxy, it
defines local RTP IP, but not externIP. Maybe somebody knows how to solve
You could try this one:
http://www.voip-info.org/wiki/view/Asterisk+func+Devstate
If I can add a warning, be wary of having both ACD (Queue) and non-ACD
traffic on the same operator - you risk having awful performance.
Just my two eurocents,
l.
2009/6/12 Lee, John (Sydney) john@compuware.com
Hi Guys,
Any suggestions on any open source soft phones that has IAX and
SIP support.
I would also like to some programming over it and interface it with
address book or LDAP in order to make the call making easier for the
users.
Thanks
Manoj
On 15 Jun 2009, at 12:05, Manoj Panicker - FOES wrote:
Any suggestions on any open source soft phones that has IAX
and SIP support.
I would also like to some programming over it and interface it with
address book or LDAP in order to make the call making easier for the
users.
Hello guys,
Is there a decent click-to-dial CTI which works well with Asterisk?
We have vanilla asterisk implementation and I have tried a few (ADA,
Outcall etc) but they have poor documentation and don't work very well.
We are looking for an application which can allow us to dial a number
from
twinkle.
2009/6/15 Manoj Panicker - FOES manoj.panic...@emirates.com
Hi Guys,
Any suggestions on any open source soft phones that has IAX and
SIP support.
I would also like to some programming over it and interface it with address
book or LDAP in order to make the call making
I'm currently using Ekiga. I don't think I'd reccomend it though; it
lacks a lot of basic features.
--
Christopher Stamper
Email: christopherstam...@gmail.com
Web: http://tinyurl.com/2ooncg
gTalk: http://tinyurl.com/6e359r
Skype: cdstamper
___
--
On Mon, Jun 15, 2009 at 12:51:25PM +0100, Geraint Lee wrote:
twinkle.
Twingle is a good SIP phone. But does not support IAX.
At the moment the only one I can think of is yate-gtk :-)
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406
Hi,
I try Noojee Click and Outcall, and for my context they work fine. Some
times ago I tried SanpANumber, but it was bought by Digium and substitute
with ADA.
Bye
Marco
2009/6/15 Stefanov, Milen milen.stefa...@compuware.com
Hello guys,
Is there a decent click-to-dial CTI which works
Hello All,
We have a requirement of hosted multi-tenant PBX where we can map DID for
different clients. Each client should have saperate interface of Reporting,
Call Recordings, Voice Mail and other features. Please suggest some solution
or let us know if have it to sell ?
Regards,
Kashif Naeem
Excuse me?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 15 June 2009 15:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Open Source Soft Phone
On 15 Jun 2009, at 13:36, Manoj Panicker - FOES wrote:
Excuse me?
You sent this message twice. Send it once, and wait for a reply.
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Quoting Jaap Winius jwin...@umrk.to:
Previously, I had the PrivacyManager working for me exactly as would
be expected, but after upgrading the OS to Debian lenny and Asterisk
to v1.4.21.2 that's no longer the case. Anonymous callers are still
confronted with the PrivacyManager, but now no
I posted a simple PERL agi that uses hints to do a similar thing to Devstate
last week. Here it is:
#!/usr/bin/perl
use strict;
use warnings;
# define variables
# show hints will get hint information from the dialplan
my $cmda = '/usr/sbin/asterisk -rx show hints ';
my
Remove the Set in extensions.conf and reload the dialplan. If you don't
have that capability, just do a set to null.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Monday, June 15, 2009 4:07 AM
To: Asterisk
Olivier wrote:
Hi,
In 1.6.1, it seems Attended Transfer do not behave exactly behave like
Blind Transfer when transferer hangs up before callee answers :
- in Blind Transfer, caller (ie transferee) is hearing Ringing tone
when callee's phone is ringing
- in Attended Transfer, caller (ie
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
I try to set up a gateway gtalk to sip.
I test asterisk 1.6.1 and 1.4.21 from debian repository and the result
is identical : no sound during the call.
my jabber.conf :
[general]
debug=yes
autoprune=no
autoregister=no
[allo-gw]
type=client
I use snapanumber for dialing from Outlook works great.
Don't know what Digium did to it when they made it Outcallbut you're
not the only one who has said they had a problem with it.
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
mailto:d...@cognation.net +1-212-203-4357
Hello All,
We have a requirement of multi-tenant Predictive Dialer which we can sell to
multiple call centers. Each call center will have saperate interface for
setting up campaigns and Reporting. Please suggest some solution or let us
know if have it to sell ?
Regards,
Kashif Naeem
Business
On Monday, June 15, 2009, Steve Howes wrote:
On 15 Jun 2009, at 13:36, Manoj Panicker - FOES wrote:
Excuse me?
You sent this message twice. Send it once, and wait for a reply.
Only received once here. My mail server is configured to remove
duplicated messages - but a different timestamp
On 15 Jun 2009, at 15:54, Geoff Lane wrote:
On Monday, June 15, 2009, Steve Howes wrote:
On 15 Jun 2009, at 13:36, Manoj Panicker - FOES wrote:
Excuse me?
You sent this message twice. Send it once, and wait for a reply.
Only received once here. My mail server is configured to remove
Hi,
Just wondering what the popular open source call statistics / metrics
packages are for Asterisk? Preferably an all-in-one package that
supports queues and calls from the CDR information generated by
Asterisk.
Whats everyone using? Favorites?
Thanks,
Marc
On Mon, Jun 15, 2009 at 10:54 AM, Geoff Lane ge...@gjctech.co.uk wrote:
Only received once here.
Only once here also, using gmail.
IOW, it looks to me like the list server had a hiccough and
Christopher wrongly accused the OP.
Steve did the 'accusing', not me... ;-)
--
Christopher
Thanks for your information!
Now I tried to send a Sip Messages, instead of a Sip Info.
Between two softphones the exchange of sip messages works fine. But the
message relay over the asterisk doesn't work:
Status 415 Unsupported Media Type
Does someone know, how to activate the exchange
Hi,
I've been editing my dialplan to launch custom instructions anytime a SIP
REFER-based transfer occurs.
The only hook I could find is catching an hangup event which is tied to a
Zombie channel
(ie a channel named like SIP/1234-vhvebjvnvZOMBIE).
Is this a feature or a bug ?
In other words, do
On Monday 15 June 2009 04:06:31 am Olivier wrote:
Is there a way to remove a global variable from dialplan ?
Set(GLOBAL(foo)=)
--
Tilghman
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To
On Monday 15 June 2009 04:03:48 am Olivier wrote:
I've just discovered IMPORT function existence.
It can be use to import values from channel's Variable section but
unfortunately, il can't be use to access to values from Info section
(I'm referring here to sections Info and Variables dumped by
The Asterisk Development Team is pleased to announce the third beta of
Asterisk 1.6.2.0. Asterisk-1.6.2.0-beta3 is available for immediate download at
http://downloads.digium.com/pub/asterisk/
This is an incremental release of the 1.6.2.0 branch as the previous beta was
released just over a month
Hello Asterisk-users,
I am new to Asterisk. I got SIP Calls to work between two computers using a
soft phone and asterisk in the middle. Since then, I have been trying to get
my soft phone to make a PSTN call with terrible failure for about two days
now.
On Windows using asteriskwin32:
I have a
Kashif
This changes things. We can do, but this is hardly a simple off-the-
shelf.
I will call you midday Tuesday if I might
Steve
On Jun 15, 2009, at 3:46 PM, Kashif Naeem wrote:
Hello All,
We have a requirement of multi-tenant Predictive Dialer which we can
sell to multiple call
On 2009-06-14 at 00:19, Jeff LaCoursiere (j...@jeff.net) wrote:
[hft0]
type=friend
username=hft0
secret=mysecret
context=outtrunk-office
host=192.168.200.99
Change the above to host=dynamic
I just did this and did a 'reload'.
reg.1.server.1.address=jtsd05
Can the phone
Well, lets just take the OP out and shoot him!
GEESH! Can we all just move on, or MUST we waste more and more time and
messages sent to reportedly 10,000 people on this unimportant issue.
The original responder could have simply answered the guy's question or
even better said nothing, instead
On Mon, 15 Jun 2009, Jim Gottlieb wrote:
On 2009-06-14 at 00:19, Jeff LaCoursiere (j...@jeff.net) wrote:
[hft0]
type=friend
username=hft0
secret=mysecret
context=outtrunk-office
host=192.168.200.99
Change the above to host=dynamic
I just did this and did a 'reload'.
Pardon my ignorance, but can you register the external sip name to your
internal ip (192.168.x.x)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Monday, June 15, 2009 2:13 PM
To:
On 2009-06-15 at 19:12, Jeff LaCoursiere (j...@jeff.net) wrote:
chan_sip.c: Registration from 'sip:6193644...@jtsd05.nccom.com' failed
for '192.168.200.99' - Username/auth name mismatch
I am a bit confused as to the names and addresses involved here. Which
name/address is the server,
Jim Gottlieb wrote:
I'm evaluating using Polycom phones for our call center and I've set
up my first phone (a SoundPoint 560) to give it a try.
The phone is working and can successfully place and receive calls.
But every minute, there's an error in the log file:
chan_sip.c:
On 2009-06-15 at 17:04, Dave Fullerton
(dfullertaster...@shorelinecontainer.com) wrote:
Try changing reg.1.address to hft0. My hunch is asterisk is looking at
the from of 6193644...@jtsd05 and going huh? I don't know a
6193644...@jtsd05.
That makes sense and it fixed it. Thanks!
Hi
I have come across a problem, with my tdp410 and soekris board
(basically pc on a chip amd geode cpu).
I am using the box as a firewall/asterisk box. The problem occurs when I
drop ppp and I get dead loop dectiotn going, I seem to lose interrupts
and get lots of messages in syslog from
Alex Samad wrote:
Hi
I have come across a problem, with my tdp410 and soekris board
(basically pc on a chip amd geode cpu).
I am using the box as a firewall/asterisk box. The problem occurs when I
drop ppp and I get dead loop dectiotn going, I seem to lose interrupts
and get lots of
I have a production PBX (1.6.0.9) 150+ phones with MS Exchange 2007 and OCS
working very well out of the box. We're using SIP/TCP support in 1.6.x; Believe
it or not the most challenging part is to get MWI signaling back from Exchange.
Let me know if I can help.
Jim
j...@sigma-networks.com;
All right Steve Thanks. I thought it never went. My apologies.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Howes
Sent: 15 June 2009 16:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Mon, Jun 15, 2009 at 08:19:33PM -0500, Lyle Giese wrote:
Alex Samad wrote:
Hi
[snip]
as you can see with the interrupts the wctdm24xxp0 is above eth0 (local
lan) and eth3 (my adsl)
eth1 is wireless and not heavily used
So any one had this problems, any other possible
Hello List!
I have 2 asterisk servers, The Admin(.20), and the Call Center(.21).
The Admin server contains the 1XXX extension and the Call Center hosts
the 2XXX extensions. I would like for our Admin folks to be able to
call the Call Center folks (and vice versa).
The call will go over the
Dan Pilcheck wrote:
The call will go over the server fine, but when the Call Center server
answer, the CLI returns:
NOTICE[4296]: chan_iax2.c:7398 socket_read: Rejected connect attempt
from 10.0.10.20, request '2...@2xxx' does not exist
What context are the phones in the extension range
On Monday 15 June 2009 20:00:11 Alex Samad wrote:
I have come across a problem, with my tdp410 and soekris board
(basically pc on a chip amd geode cpu).
How to engage digium to providing a fix for this ?
http://www.digium.com/en/supportcenter/
--
Tilghman
Need help pls..Anyone?
On Mon, Jun 15, 2009 at 10:58 PM, Shiva Kumar shi...@gmail.com wrote:
Hello Asterisk-users,
I am new to Asterisk. I got SIP Calls to work between two computers using a
soft phone and asterisk in the middle. Since then, I have been trying to get
my soft phone to make a
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