Sigma Networks wrote:
I would like to get in contact with users/consultants who are or have
worked with the Cisco phones and Asterisk to trade information.
Cisco has reluctantly made SIP available on their phones and most of the
information on voip-info and other wiki's appears to be
On Sun, Mar 2, 2008 at 4:26 AM, C F [EMAIL PROTECTED] wrote:
1. Way to slow to boot
2. Lack of features, can't reconfigure the buttons to show something
decent, like BLF, and the buttons you could configure are limited even
though they are soft buttons. Compare
On Sun, Mar 2, 2008 at 4:26 AM, C F [EMAIL PROTECTED] wrote:
1. Way to slow to boot
2. Lack of features, can't reconfigure the buttons to show something
decent, like BLF, and the buttons you could configure are limited even
though they are soft buttons. Compare
hey Folks,
Just curious if anyone has suggestions on how one can get a near
FREE(I hope) DID number.
I am experimenting with asterisk, for home use.
thanks,
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asterisk-users
On Sun, 2 Mar 2008, Mike wrote:
hey Folks,
Just curious if anyone has suggestions on how one can get a near
FREE(I hope) DID number.
I am experimenting with asterisk, for home use.
Telling people what country you're in will really help here.
If you're in the UK, I'll give you a free
Sean Dennis ha scritto:
Sigma Networks wrote:
...
My current questions are:
1. How to remotely reboot 7970s. I have both web access and SSH
access to the phones. The instructions I have for SSH are to use
(1) user/pass (or whatever is in the confg) and then (2)
try sipgate.co.uk
M
On Sun, Mar 2, 2008 at 8:21 PM, Mike [EMAIL PROTECTED] wrote:
hey Folks,
Just curious if anyone has suggestions on how one can get a near
FREE(I hope) DID number.
I am experimenting with asterisk, for home use.
thanks,
On Fri, Feb 29, 2008 at 02:12:18PM +0100, Patrick wrote:
On Fri, 2008-02-29 at 11:09 +, Nuno Fernandes wrote:
Hi,
We've just bought a new cisco 7965g and web are trying to connect it to
asterisk. I've bought smartnet and downloaded
[snip]
How can i install the sip firmware?
Hi.
Is it possible to override the standard DB function in
Asterisk?
My dialplan contains a lot of calls to Set(DB(...))
and ${DB(...)} which of course use astdb to
store/read data. I would like to stop using astdb and
switch to Clustered MySQL (I don't suppose clustered
astdb exists?).
So
On Wed, Feb 20, 2008 at 08:39:07PM +0200, ik wrote:
Hello,
I have the following settings for manager on two Asterisk 1.2.24 (that
have installed over a year ago):
[user]
secret = password
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
write = call,command
On one server,
Any Asterisk people going to Cebit ?
Let's meet! If you go and would like to go for a drink and meet some
others from the voip business, please add your name to the list below
Joachim Vanheuverzwijn (zoachien AT securax.org) - Attractel.com -
wednesday / thursday.
Tan Aksoy - Telappliant -
On Sunday 02 March 2008 05:33:49 Vieri wrote:
Is it possible to override the standard DB function in
Asterisk?
No.
--
Tilghman
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To UNSUBSCRIBE or
Thanx alot for reply
I mean i have to use the fxo to connect to the pstn line and i do not know if
there is any asterisk functions ,Application, options that could help to know
what is the real call duration [ how to deal with pstn line signaling how to
detect the pstn ringing tone or
continuing discussions of 79xx issues. i've seen referenced and am
experiencing difficulty getting a 7970 to work behind NAT to a public
asterisk server. i am successful with 7960s.
1. SIP load is 70.8-3-3SR2S
2. config works fine if 7970 is connecting to an asterisk server a
On Sat, 1 Mar 2008 22:26:18 -0500, C F wrote:
On Sat, Mar 1, 2008 at 9:31 AM, Michael Graves [EMAIL PROTECTED] wrote:
When in doubt there is only one sure answerPolycom. Without a doubt the
best functionality, performance and reliabilityeven in the lower cost
models. Although the
On Sat, 01 Mar 2008 18:34:36 -0600, Timothy C Litwiller wrote:
Everyone seems to think a used proprietary system would be better - I
looked on ebay for over a week and never found something that indicates
that you can conference and have voicemail. In this day and age how can
you get along
On Sun, Mar 02, 2008 at 08:30:17AM -0500, aymen warfalli wrote:
Thanx alot for reply
I mean i have to use the fxo to connect to the pstn line and i do not
know if there is any asterisk functions ,Application, options that
could help to know what is the real call duration [ how to
The documentation of how to use the 79xx series' phones and features
with Asterisk is really hard to find and put together. The higher end
phones like 7970 are more like converged PC+phone, a thin client to
telephony and network apps. But it's really hard to target it as a
development and
Have the some symptoms mentioned in http://bugs.digium.com/view.php?id=12099
After upgrade to Zaptel 1.4.9.2 I can't dial out at all, with 1.4.8 there
were just random dial-out problems.
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/210-081e9968, ) in new stack
-- Executing [EMAIL PROTECTED]:2]
I'll admit in my case, the main reason the boot time of the Polycom
drives me nuts is that I don't see the phones unless I'm installing them
for the first time or supporting them when something isn't quite right.
It's for this reason that I very much appreciate a phone that either (a)
boots
On Sun, Mar 2, 2008 at 3:21 AM, Mike [EMAIL PROTECTED] wrote:
Just curious if anyone has suggestions on how one can get a near
FREE(I hope) DID number.
Hey Mike - give IPKall a try:
http://www.ipkall.com/
They'll give you a free Washington state DID along with free SIP to
your asterisk
On Sun, Mar 2, 2008 at 10:21 AM, Mike [EMAIL PROTECTED] wrote:
hey Folks,
Just curious if anyone has suggestions on how one can get a near
FREE(I hope) DID number.
If you are in the USA, see http://www.IPKall.com it's free, works great.
Another idea, if they still do this is
The 601 is powered by PoE with 2 sidecars, so Polycom wants us to put
an actual Power Supply on the phone - thinking the voltage is dropping
and causing the reboot. I don't buy that, but we are putting one on
next Monday. We'll see.
That's almost certainly your problem. When you
Kevin P. Fleming wrote:
...
The messages in bug 12099 are *not* errors, they are annoyances only.
The latest SVN branch 1.4 code of Asterisk will no longer generate them,
Using today's svn 3915:
..
Answer(Zap/2-1, ) in new stack
--
On Sun, Mar 02, 2008 at 11:36:03AM -0500, sean darcy wrote:
Kevin P. Fleming wrote:
...
The messages in bug 12099 are *not* errors, they are annoyances only.
The latest SVN branch 1.4 code of Asterisk will no longer generate them,
Using today's svn 3915:
Tzafrir Cohen wrote:
On Sun, Mar 02, 2008 at 11:36:03AM -0500, sean darcy wrote:
Kevin P. Fleming wrote:
...
The messages in bug 12099 are *not* errors, they are annoyances only.
The latest SVN branch 1.4 code of Asterisk will no longer generate them,
Using today's
sean darcy wrote:
In any event, as least for me the TDM400P seems to have problems with
zaptel svn - not just an annoyance.
As I've mentioned previously, the changes to fix this for good (assuming
they work properly) are in
http://svn.digium.com/svn/zaptel/team/kpfleming/battery_alarms,
Kevin P. Fleming wrote:
sean darcy wrote:
In any event, as least for me the TDM400P seems to have problems with
zaptel svn - not just an annoyance.
As I've mentioned previously, the changes to fix this for good (assuming
they work properly) are in
Kevin P. Fleming wrote:
sean darcy wrote:
In any event, as least for me the TDM400P seems to have problems with
zaptel svn - not just an annoyance.
As I've mentioned previously, the changes to fix this for good (assuming
they work properly) are in
Tilghman Lesher [EMAIL PROTECTED] writes:
On Sunday 02 March 2008 05:33:49 Vieri wrote:
Is it possible to override the standard DB function in
Asterisk?
No.
Is it permitted to modify the astdb outside Asterisk, while Asterisk
is running? It is a SQLite file, right?
/Benny
On Sun, Mar 02, 2008 at 11:17:20PM +0100, Benny Amorsen wrote:
Tilghman Lesher [EMAIL PROTECTED] writes:
On Sunday 02 March 2008 05:33:49 Vieri wrote:
Is it possible to override the standard DB function in
Asterisk?
No.
Is it permitted to modify the astdb outside Asterisk, while
Thanks for the update.
I have Sangoma A104D and wanted to use ss7 signalling. I came accross
chan_ss7 but found sifira is not in active development. But is this
chan_ss7 stable and can be used in production server implementation.
We are going to have 2 to 3 boxes with ss7 signalling using
sean darcy wrote:
Kevin P. Fleming wrote:
sean darcy wrote:
In any event, as least for me the TDM400P seems to have problems with
zaptel svn - not just an annoyance.
As I've mentioned previously, the changes to fix this for good (assuming
they work properly) are in
Hi All;
If someone used speex and has experience with its
settings, then who can help to explain the following:
1) When it is recommended to use VBR (vbr = true)?
2) If there relation between setting the vbr = true
and the abr value (for example to be 0 or 1 or 10) and
the relation between this
First hit for speex asterisk settings in Google answers all of your
questions.
http://www.voip-info.org/wiki/view/Asterisk+config+codecs.conf
On Sun, Mar 2, 2008 at 5:54 PM, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi All;
If someone used speex and has experience with its
settings, then who
Hi Guys,
I'm new in VoIP, I heard from a friend that asterisk is good in VoIP service
especially on SIP. I'm planning to replace our old PBX system (legacy of
Panasonic) to VoIP so that even out of the country we can still communicate
cheaper than regular phone. But I have a few questions
My incoming calls work just fine with 1.4.9.2 and TDM400P but I'm completely
unable to dial out the line.
even this doesn't work:
exten=*903,1,Answer()
exten=*903,n,Dial(Zap/3)
exten=*903,n,Hangup()
I use this to pick up the line shared with analog phone. Zap/3 and Zap/4 are
connected to PSTN.
By default cdr is stored in a delimited fille, but can easily be put into a
mysql table.
Securing it is a good question. Asterisk is secure on its own, and I would
secure the rest like ssh by changing the default ports and such.
I don't know about bsd.
You will fall in love with asterisk
Martin wrote:
.
In any event, as least for me the TDM400P seems to have problems with
zaptel svn - not just an annoyance.
As I've mentioned previously, the changes to fix this for good (assuming
they work properly) are in
I just finished downloading and install the asterisknow. It's pretty cool
GUI... How do I see my CDR on asterisknow?
Thank you...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard Neese
Sent: Monday, March 03, 2008 10:39 AM
To: Asterisk on BSD
On Mon, 3 Mar 2008 10:14:02 +0800, NOC Ph [EMAIL PROTECTED] wrote:
This questions might annoyed experts. Please bear with me...
The journey of a thousand miles begins with a single step. Lao
Tzu.
Free PDF of Asterisk: The Future of Telephony, Second Edition
A simple inrease of DEFAULT_BATT_DEBOUNCE to 32 (originally 4) work for
me... Probably a nasty hack but allows me to dial
(I also need DEFAULT_RING_DEBOUNCE increased to 256 to minimize problems
with caller ID, which is right above)
Martin
My incoming calls work just fine with 1.4.9.2 and
On Tue, 26 Feb 2008 01:19:23 +0200, Atis Lezdins [EMAIL PROTECTED]
wrote:
To help you on your way of minimizing modules, here's some basic setup
that generally works
Thanks much for sharing your modules.conf.
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Hi marek,
gr8. I am working on chan_ss7 now..
Regards,
Joel
- Original Message -
From: marek cervenka [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, March 03, 2008 3:55 AM
Subject: Re: [asterisk-users]
On Mon, Mar 03, 2008 at 10:14:02AM +0800, NOC Ph wrote:
Hi Guys,
I'm new in VoIP, I heard from a friend that asterisk is good in VoIP service
especially on SIP. I'm planning to replace our old PBX system (legacy of
Panasonic) to VoIP so that even out of the country we can still
Hi Tilghman,
Thanks for taking interest in my problem.
I just want to send a http post request to my website without changing the
dial plan. So I have added slightly modified http post code and some other
code to channel.c got from curl/curl.h.
After adding the code I compiled the asterisk code
It's not only Asterisk you have to secure. You install Asterisk on a
system. It is that system you have to secure. You'll probably expose
some sort of management interface to the world (a web interface, ssh,
whatever).
[NOCPH] I have to open the SIP port and web. Another question, the SIP port
is
On Mon, Mar 3, 2008 at 12:40 AM, NOC Ph [EMAIL PROTECTED] wrote:
[NOCPH] I have to open the SIP port and web. Another question, the SIP port
is 5060 UDP, how about the conference? Does it use the same port also?
That's a good start, but you'll also need to open the RTP ports as
well - these
On Mon, Mar 03, 2008 at 02:40:03PM +0800, NOC Ph wrote:
It's not only Asterisk you have to secure. You install Asterisk on a
system. It is that system you have to secure. You'll probably expose
some sort of management interface to the world (a web interface, ssh,
whatever).
[NOCPH] I
I have asterisknow 1.0.1 install on my box. Seems working now... but how can
I see or monitor the calls and CDR is important?
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Monday, March 03, 2008 2:06 PM
To:
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