On Thu, 28 Aug 2008, Michael Graves wrote:
> I've had essentially no problems with my snom m3s. Someone from snom
> has been in touch to confirm that they are now putting more effort into
> the firmware for this phone. There are a few new features that I'd like
> to see that are already in their p
Hello Philipp,
Yes, I have autofill set in queues.conf. I suspect that this behaviour is
because the Polycom phones I use have 2 lines. Has anyone used this function
with polycom phones before? Also, my agents are Dynamic, perhaps this works
better with Static agents?
Here's my queues.conf (with
Yup I just copy and paste to it but it shown not a known channel.
On Thu, Aug 28, 2008 at 6:47 PM, Steven Howes <[EMAIL PROTECTED]> wrote:
> Did you tab complete it to make sure it was right?
>
> On 28 Aug 2008, at 11:39, Rilawich Ango wrote:
>
>> I got the message below after I issue the sof
On 08/28/08 08:18, Jay R. Ashworth wrote:
>On Wed, Aug 27, 2008 at 06:38:48PM -0700, Trevor Peirce wrote:
>> I'm pretty confident the Linksys device does not support this
>> functionality. Asterisk can't really do much with it anyway as it can't
>> answer the call waiting call as long as the orig
Hi ,
I have check zapte.conf in and after make some correction that problem
solve.
But now I am facing other problem. We are using here Postgres Database
and the data from CLI it can't insert in Postgres Database. I have also
here mention below cdr_pgsql.conf, modules.conf and cdr.conf
cdr
Call files can do something like this - you can choose the number to
call and where to connect the call to (within the dialplan)
PaulH
Julien Claassen wrote:
> Hello all!
>Is there a way to (mis)use asterisk itself as a softphone? Can I make a
> call
> from within the CLI? Can asterisk f
When I set out to develop a basic service provider Perl AGI framework
for Asterisk three or four years ago, I wanted to design something
that would make developing additional Perl AGI apps under this
framework scalable and easy to do. One of the features I wanted to
have in this framework w
I've had essentially no problems with my snom m3s. Someone from snom
has been in touch to confirm that they are now putting more effort into
the firmware for this phone. There are a few new features that I'd like
to see that are already in their plans.
Someone did report to me that they had diffic
> > Better still - is it possible to SSH (or some sort of connection
method)
> > from a remote PC to the Asterisk server and make a call using CLI?
>
> Sure, you can use the CLI 'console dial' command.
>
Do you mean that I will be able to hear the call from my PC if I do
'console dial' on the rem
On Thursday 28 August 2008 19:05:23 Lee, John (Sydney) wrote:
> >> Hello all!
> >> Is there a way to (mis)use asterisk itself as a softphone? Can
> >> I make a call
> >> from within the CLI? Can asterisk from itself produce a ringtone? I
> >> Or can bind a system-command to incoming calls?
> >> Any
Let me know if you find out - We played around with this for a while but
could never get it to work. We ended up sending multiple messages with blank
lines to get the spacing we wanted.
Thanks,
Matt G
: http://www.voipphreak.ca
: http://www.ratemydialplan.com
-Original Message-
From: [E
>> Hello all!
>> Is there a way to (mis)use asterisk itself as a softphone? Can
>> I make a call
>> from within the CLI? Can asterisk from itself produce a ringtone? I
>> Or can bind a system-command to incoming calls?
>> Any help is sincerely appreciated!
>You can install a browser softphone on t
Is there a way to include a linefeed in the message sent by JabberSend?
--
Eric Chamberlain
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.ast
How can you configure Asterisk to forward the calls you don't want to answer
back on the 2nd PRI line?
Does this traffic increase the load on the asterisk server, or is it
completely dealt with by the 2 port card?
Thanks,
Dan
On Thu, Aug 28, 2008 at 3:46 AM, Paul Hales <[EMAIL PROTECTED]> wrote:
Oh, by the way, the agent who will be doing the assisted transfer will be
using eyebeam.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: August 28, 2008 5:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Transfers o
Hi,
I have the same question as:
http://lists.digium.com/pipermail/asterisk-dev/2003-November/002320.html
..which like all important things was never answered.
How do I do an assisted transfer on AgentLogin()? I don't use zaptel, it's
just pure SIP/VoIP.
Help please.
Thanks,
Mark.
Every one PSTN line connected to the FXS port of sipura..
Though these 4 lines comes in one cable if that has to do with anything!
> Date: Thu, 28 Aug 2008 14:10:53 -0400
> From: [EMAIL PROTECTED]
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] sip conversations overlappin
You can install a browser softphone on the same server and make calls
from any browser. Ted
- Original Message -
From: "Julien Claassen"
To: "asterisk users mailinglist"
Subject: [asterisk-users] Console softphone
Date: Thu, 28 Aug 2008 11:16:59 +0200 (CEST)
Hello all!
Is t
Hi,
The settings provided did not work...the state is still OFF HOOK...
ANy other ideas?
--- On Tue, 8/26/08, Guillermo Salas M. <[EMAIL PROTECTED]> wrote:
From: Guillermo Salas M. <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] X100P Card in OFFHOOK state
To: "Asterisk Users Mailing List -
Ruchir wrote:
> Have you set dtmf mode rfc2833 or avt in your phone?
>
No, I have not changed anything in the phone. The sip.cfg setting is the
default:
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
_
> They want an after hours application that checks inbound caller ID
> numbers and matches them to a list, say 5 to 10 numbers of special VIP
> customers, if there is a match on the list, then forward the call
> straight to a cell phone, instead of ringing local extension and then
> to voicemail.
>
Hi again!
I searched through the net some more about this problem. It seems to be
something in mISDN. But noone got a satisfactory answer. People got this
problem or similar ones since 2005.
If there's anything I can do to help solve this, please tell me so. One
thing I found out is, that
Hi
I think that you can use quicktime..
bye
Date: Thu, 28 Aug 2008 23:40:37 +0500From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: Re: [asterisk-users] GSM recordings
Use winamp media player with their gsm extension.Shariq
On Thu, Aug 28, 2008 at 10:40 PM, Gustavo A Gonzalez <[EMAIL PROTECTED]> wrot
Use winamp media player with their gsm extension.
Shariq
On Thu, Aug 28, 2008 at 10:40 PM, Gustavo A Gonzalez <[EMAIL PROTECTED]
> wrote:
> Hi folks!
>
>
>
> I want to play gsm agents recordings from a web interface, to do that,
> someone knows some media player that launches when I click on th
RoLaNd RoLaNd wrote:
>
> Hi all,
>
> i'm facing this weird prob...my topology is as such:
>
>
>
> -
> -
>
> when am on a call, sometimes when some1 else tries
Hi!
Maybe I'm dense: You need a player that can be launced from within your
browser when you click on a remote or local file?
If it can be a simple system-application, you could try sox (in every linux
distro). I think it plays gsm just fine. If you need some kind of applet, that
opens in
Hi everyone!
If I try to originate a call from the CLI using my ISDN-card, I get errors
from the mISDN driver. Here's what I did:
CLI> originate mISDN1/029213399096 application Jack system:playback_1
system:capture_1
The output I get:
P[ 0] --> * NEW CHANNEL dad:Extern1 oad:(null)
P[ 1] re
Hi folks!
I want to play gsm agents recordings from a web interface, to do that,
someone knows some media player that launches when I click on the file that
I want to hear?
Thanks!
Gustavo A. González
Dto. de Infraestructura
Despegar.com, Inc.
[EMAIL PROTECTED]
___
Hi All;
If I need to see on my Polycom LCD the caller id of the other caller extension
(for example, if 801 called the polycom of 802 then how can I let the LCD of
polycom of the extension 802 to display the 801 as caller)? My polycom model is
330.
Also, I have IAX trunk between two Asterisk b
Hi all,
i'm facing this weird prob...my topology is as such:
-
-
when am on a call, sometimes when some1 else tries to call out.. i hear the
ac
Hi,
I was wondering if there's any sense in increasing audiobuffer above the
minimal '2' in meetme, if every channel is already dejittered before
(Local/.../nj - as described at:
http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/)
Will it help in anythin
I have a client testing one of these and he is happy with it so far.
I don't know if there are any known problems yet with this phone, but would
be interested in knowing about your review.
Tom
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Fred Posner
On Thu, 28 Aug 2008, Jaap Winius wrote:
Hi list,
Are there any reliable wireless SIP phones available on the market
yet?
My Linksys WIP330 should arrive today. I've always wanted to test how
well it would work in hotspots... will let you know.
Fred Posner
[EMAIL PROTECTED]
Tel: +1 (2
Anyone else having timeouts to Voicepulse?
Fred Posner
[EMAIL PROTECTED]
Tel: +1 (212) 937-7844 x501
Fax: +1 (954) 252-4187
www.teamforrest.com
smime.p7s
Description: S/MIME cryptographic signature
___
-- Bandwidth and Colocation Provided by h
On Thu, 28 Aug 2008, Jaap Winius wrote:
Hi list,
Are there any reliable wireless SIP phones available on the market
yet?
My Linksys WIP330 should arrive today. I've always wanted to test how
well it would work in hotspots... will let you know.
Fred Posner
[EMAIL PROTECTED]
Tel: +1 (2
On Thu, 28 Aug 2008, Jaap Winius wrote:
Hi list,
Are there any reliable wireless SIP phones available on the market yet?
Six months ago I went for the Siemens Gigaset 675IP. Although there
was a bug in the MWI support, unit #1 seemed fine for the first few
weeks, so I bought #2 and #3. Then th
Jaap Winius wrote:
> Hi list,
>
> Are there any reliable wireless SIP phones available on the market yet?
>
>
> Since the firmware seems to be the same, there's no way I'm going to
> upgrade to the 685IP. I was thinking of trying out the Snom M3, but
> according to voip-info.org, that model
We use
http://www.areski.net/asterisk-stat-v2/about.php
http://www.micpc.com/qloganalyzer/
on Asterisk 1.2, don't know how well they work with later versions
regards,
Drew
Mark Hamilton wrote:
> Doesn't Queuemetrics run on a license basis?
> Anything else that's probably open source and free?
On Thu, Aug 28, 2008 at 08:24:58AM -0500, David A. Bandel wrote:
> On Wed, Aug 27, 2008 at 11:01 PM, lizhong zhu <[EMAIL PROTECTED]> wrote:
> > hello, all of users:
> > i have a problem with loading chan_dahdi.so. when i start asterisk, it
> > always reports the can not open channel 1 in ...
> > he
El jue, 28-08-2008 a las 01:32 -0700, mahboob zaman escribió:
> hi.
>
> i have two IP phones that are in H323 protocol. How can i test that
> these two phones are working? For IP phone (SIP) i used asterisk
> server. can i use asterisk server to test the ip phone with H323
> protocol.
>
I've wr
El jue, 28-08-2008 a las 10:32 -0400, BerkHolz, Steven escribió:
> asterisk linkedin group
>
>
>
> I have created an asterisk linkedin group for anyone interested.
>
>
>
> http://www.linkedin.com/e/gis/45252/66270A773F53
>
Thank you, I've joined it.
There is a group for spanish users fo
asterisk linkedin group
I have created an asterisk linkedin group for anyone interested.
http://www.linkedin.com/e/gis/45252/66270A773F53
Thank You,
Steven BerkHolz
- MCSA - MCSE -
Manager of Information Systems
HIROTEC AMERICA
Please visit us on the web at www.h
Thank u very much, Russel.
I will definitely contact with digium, & then updates you.
Shariq
On Thu, Aug 28, 2008 at 5:27 PM, Russell Bryant <[EMAIL PROTECTED]> wrote:
> Shariq Khan wrote:
> > I m facing problem with TDM2400P pstn card. When someone dials, the
> > voice quality is crappyIns
On Thursday 28 August 2008 08:59:29 Geraint Lee wrote:
> On 2008/8/28, Cory Andrews wrote:
>
> > Just a heads up, Hitachi is effectively ceasing production of their
> > IP5000 and IP3000 WiFi SIP Phonesproduct availability is next to nil
> > on these. They also have no plans apparently to conti
Hi,
I'm looking for the SEP.cnf.xml (and XMLDefault.cnf.xml) file
for a Cisco 7911 with SIP firmware 8.3.5. If anyone on the list has one
I sure would appreciate it if you could send me a copy. If you prefer to
email it privately please use my "from" email address without the "-list".
The reas
> I you have such a problems with siemens you should consider
> 8 voip port
> linksys gateway with dect bases, their gateway is rock
> solid and cheap.
I'd recommend against buying new analog POTS gear myself. We've got a mix of
SNOM 300 corded VOIP phones and generic DECT bases attached to Lin
I've used several hitachi dmp330's they work great, roam between wireless
access points with no loss of audio or connection for that matter.
it will be a great shame if hitachi stop producing them, they are the most
reliable wireless sip phones i've come accross... stay well away from
pirelli phon
On 26/08/2008 Nhadie wrote:
> if i use an IAX trunk, how do i dial a SIP user?
don't think of them as SIP or IAX users, they're just extensions.
If one box has extensions 1xx and the other 2xx then your dial plan on
the 1xx box needs to send the calls dialled to 2xx extensions to other
box and
Just a heads up, Hitachi is effectively ceasing production of their IP5000 and
IP3000 WiFi SIP Phonesproduct availability is next to nil on these. They
also have no plans apparently to continue producing WiFi phones.
Cory J. Andrews
Director New Market Initiatives
VoIP Supply, LLC.
454 Son
On Aug 28, 2008, at 9:06 AM, Jaap Winius wrote:
> Hi list,
>
> Are there any reliable wireless SIP phones available on the market
> yet?
We typically prefer DECT in which case the SNOM M3 is a strong
contender, but recently our customers have told us good things about
Polycom's new wifi h
I you have such a problems with siemens you should consider 8 voip port
linksys gateway with dect bases, their gateway is rock solid and cheap.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Ph
We had some problems with siemens 675ip with audio, but with the correct
setup they disappeared, we are using one base and 2 phones.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Ariz
On Thursday 28 August 2008 08:06:37 Jaap Winius wrote:
> Are there any reliable wireless SIP phones available on the market yet?
I've gotten a Hitachi WIP3000, which works great. Supports b & g, all the
wireless encryption standards, scans networks, everything a laptop softphone
would do, but in
On Wed, Aug 27, 2008 at 11:01 PM, lizhong zhu <[EMAIL PROTECTED]> wrote:
> hello, all of users:
> i have a problem with loading chan_dahdi.so. when i start asterisk, it
> always reports the can not open channel 1 in ...
> here is my setting: in etc/system/dahdi.conf:
> # Global data
> fxsks=1
> fxs
No problem with Snom M3 phones here :) Just make sure you run the latest
firmware. There are a couple of annoying bugs but these will be fixed in the
next FW release which should be the start of September.
Regards,
--
--[ UxBoD ]--
// PGP Key: "curl -s http://www.splatnix.net/uxbod.asc | gpg
Hi list,
Are there any reliable wireless SIP phones available on the market yet?
Six months ago I went for the Siemens Gigaset 675IP. Although there
was a bug in the MWI support, unit #1 seemed fine for the first few
weeks, so I bought #2 and #3. Then the problems started. Of the three
unit
Hello Russell!
Thanks a lot for your answer. A few more questions:
1. If I want to dial over my ISDN card it would be:
CLI> originate misdn/029213399096 ...
Right?
2. The input and output ports do they mark the ports to connect to like:
CLI> originate sip/[EMAIL PROTECTED] system:capture_1 sy
randulo wrote:
> So this will be an option in selectmenu? (or menuselect or whatever
> it's called, it's been a long time since I've built asterisk)
Yes, it is treated just like other Asterisk modules. It will be built
by default if you have the proper dependencies installed. You can
optionall
Shariq Khan wrote:
> I m facing problem with TDM2400P pstn card. When someone dials, the
> voice quality is crappyInstead of hearing.
Please contact Digium technical support for assistance with this
problem. They are the experts when it comes to debugging these types of
issues. This assis
Julien Claassen wrote:
>The question: Can I (mis)use my asterisk CLI interface to make and recieve
> calls coming in/going out via the ISDN-card, while using my soundcard I/Os
> under JACK as a phone?
Yes, you can. You actually have two options for doing this. One is
using app_jack and th
On Wed, Aug 27, 2008 at 06:38:48PM -0700, Trevor Peirce wrote:
> I'm pretty confident the Linksys device does not support this
> functionality. Asterisk can't really do much with it anyway as it can't
> answer the call waiting call as long as the original call is still engaged.
To the OP: is you
Tobias Ahlander schrieb:
> I have a sample queue with two dynamic agents. When the first caller calls
> in to the system, the first agents phone starts to ring. Then another caller
> calls in to the queue, but the other phone doesn't start to ring until the
> first agents pick up his queued call.
http://www.voip-info.org/wiki/view/Asterisk+H323+channels
Google is your friend.
PC
---
Paul Catchpole CCNA
Cisco Enterprise Network Consultant
Bluecat Certified Engineer
www.paulcatchpole.co.uk
0121 285
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of m
Hi,
Thanks for reply. can u give me information in detail? How can i compile and
can i add chan_h323 ?
Thanks
mahboob
On 8/28/08, map <[EMAIL PROTECTED]> wrote:
>
> Yes you can.
> Obviously you have to compile, configure and add chan_h323 to Asterisk.
>
> Map
>
> On Thu, Aug 28, 2008 at 10:32
Have you set dtmf mode rfc2833 or avt in your phone?
On Thu, Aug 28, 2008 at 4:29 PM, Chris Mason (Lists) <[EMAIL PROTECTED]>wrote:
> I have a client with 30 extensions, all Polycom 501 phones, an Asterisk
> 1.2.30.2 installation, and trunking over SIP to TelIAX. Everything works
> fine except wh
Hello List,
I have a sample queue with two dynamic agents. When the first caller calls
in to the system, the first agents phone starts to ring. Then another caller
calls in to the queue, but the other phone doesn't start to ring until the
first agents pick up his queued call.
I want the second ca
Hello,
I've installed Asterisk and Asterisk GUI 2.0. The GUI says "No Analog
Card found" and /etc/asterisk/ztscan.conf is empty.
I see the following message from asterisk,
-- Executing [EMAIL PROTECTED]:1]
System("Local/[EMAIL PROTECTED],2", "uptime >
/var/lib/asterisk/static-http/config/sysinf
I have a client with 30 extensions, all Polycom 501 phones, an Asterisk
1.2.30.2 installation, and trunking over SIP to TelIAX. Everything works
fine except where they need to use DTMF to navigate IVRs such as
Dell.com. The tones are not recognized at all.
My sip.conf lists for each extension:
Did you tab complete it to make sure it was right?
On 28 Aug 2008, at 11:39, Rilawich Ango wrote:
> I got the message below after I issue the soft hangup.
> sip01*CLI> soft hangup Local/[EMAIL PROTECTED],2
> Local/[EMAIL PROTECTED],2 is not a known channel
>
> Any other way to kill the call witho
I got the message below after I issue the soft hangup.
sip01*CLI> soft hangup Local/[EMAIL PROTECTED],2
Local/[EMAIL PROTECTED],2 is not a known channel
Any other way to kill the call without affecting other queues and calls?
On Thu, Aug 28, 2008 at 4:09 PM, Steven Howes <[EMAIL PROTECTED]> wrote
hi,
i think i'm getting somewhere (i hope) with this combo.
i have tried registering to the Virtual IP and i'm getting unauthorized.
i set sip debug to try and see the difference and found out i am missing
this:
Authorization: Digest
username="200200",realm="sip.mydomain.com",nonce="4cbc7dba"
Most likely at this point what I should be using for a hardware platform
that will work in a small area that I can put in place and leave it put and
running for years on end like you can with other pbx equipment.
The pbx installer I worked with says that 16 channel sip cards for the
system cost a
Hello all!
Is there a way to (mis)use asterisk itself as a softphone? Can I make a call
from within the CLI? Can asterisk from itself produce a ringtone? Or can I
bind a system-command to incoming calls?
Any help is sincerely appreciated!
Kindest regards
Julien
Music
Yes you can.
Obviously you have to compile, configure and add chan_h323 to Asterisk.
Map
On Thu, Aug 28, 2008 at 10:32 AM, mahboob zaman <[EMAIL PROTECTED]>wrote:
> hi.
>
> i have two IP phones that are in H323 protocol. How can i test that
> these two phones are working? For IP phone (SIP) i us
Hei!
I need to accomplish something and I don't know how. Asterisk version is
1.2.13. I need to make a routing decision based on how long the call has been
in ringing state. Lets say I have a few extensions and I want to ring each of
them for 5 seconds (I can't use queue for technical reasons)
Try CLI> soft hangup Local.
On 28 Aug 2008, at 09:01, Rilawich Ango wrote:
> Hi ,
>
> Actually, there are 3 queues in the server. Only one queue (2700)
> has problem. I want to reset or remove the caller only in 2700
> without affecting other queues or calls. Does it work for my case?
>
>
hi.
i have two IP phones that are in H323 protocol. How can i test that
these two phones are working? For IP phone (SIP) i used asterisk
server. can i use asterisk server to test the ip phone with H323
protocol.
--
Mahboob Zaman
System Engr
Systems & Services Limited
Cell: +8801712280308
__
Chris Maciejewski wrote:
> Hi,
>
> You can find some info about differences between 1.4 and 1.6 here:
>
> http://svn.digium.com/view/asterisk/branches/1.6.0/UPGRADE.txt?view=markup
>
> Kind regards,
> Chris
>
>
Although reading the 1.4 UPGRADE.txt isn't a bad thing either, since all
the thing
Hi ,
Actually, there are 3 queues in the server. Only one queue (2700)
has problem. I want to reset or remove the caller only in 2700
without affecting other queues or calls. Does it work for my case?
On Thu, Aug 28, 2008 at 11:49 AM, Andy Kuo <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Try CLI>
On 28 Aug 2008, at 08:22, Andreas M. wrote:
> http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi,
You can find some info about differences between 1.4 and 1.6 here:
http://svn.digium.com/view/asterisk/branches/1.6.0/UPGRADE.txt?view=markup
Kind regards,
Chris
2008/8/28 --[ UxBoD ]-- <[EMAIL PROTECTED]>:
> Hi,
>
> I would like to give 1.6 a try and was wondering about the configuration
Hello,
is it possible, to execute a script (cmd system or agi), after successfull sip
registration from an
extension?!
To go in more detail , only the first register is important, re-register
messages should be ignored.
Currently i use the "action url" in snom phones, that is executed after
su
>
> Hi All,
>
> I'm using A2billing application in order to make callback calls through my
> asterisk server...Everything looks fine except the voice quality...There is
> a lot of cuts in the call with different codecs(G711, and G729)...Please
> note that when making a call from any extensions to
Hi,
I would like to give 1.6 a try and was wondering about the configuration files.
Can I just copy them across to a new install or are they completely different
? Is there a document which shows what I would need to change ?
Best Regards,
--
--[ UxBoD ]--
// PGP Key: "curl -s http://www.spl
84 matches
Mail list logo
