So the phone is telling you that no line is available.
Either increase number of calls per registration, or add more
registrations, but you need an open line to make a conference call or
perform a transfer.
Yes the phone also gave you a beep when additional calls came in and
allowed the
General question.
If you install a Digium card in an Asterisk system, and install zaptel drivers,
do this give any benefit of echo cancellation? Our PSTN gateway is a separate
Audiocodes box, so the zaptel card wouldn't actually be connected to anything.
I'm wondering though doing this would
We are looking to deploy asterisk at one of our locations
that will have about 50 phones. I have been buying different phones to test
there quality and feature set.
So far we have a
Grandstream 2000
Grandstream HandyTone 488
Cisco 7912
Polycom SoundPoint IP
And we are looking
Hello,
I'm using asterisk for a while, fully ip, and i'm
now moving to ISDN, so i got a TE110P.
The first problem is the "Unexpected Channel
selection 3" message. I'm unable to find, on the web, the meaning of such
message.
Can somebody explain me?
My second problem is when a call come
If pricing is an issue, I've had very good experience with GXP-2000. Otherwise, I really like the SPA-941/2.- DanielOn Jun 28, 2006, at 3:05 PM, Forrest Beck wrote: We are looking to deploy asterisk at one of our locations that will have about 50 phones. I have been buying different phones to
Forrest This topic is rehashed
almost weekly, and if you rummage through the recent archives of the listserv
here http://lists.digium.com/pipermail/asterisk-users/
Youll find lengthy discourse on the
subject.
Its a somewhat subjective topic.
Cory Andrews
Executive Vice President
Hi,
Douglas Garstang wrote:
If you install a Digium card in an Asterisk system, and install
zaptel drivers, do this give any benefit of echo cancellation? Our
PSTN gateway is a separate Audiocodes box, so the zaptel card
wouldn't actually be connected to anything. I'm wondering though
doing
Polycom phones support STUN - that should solve the issue too.
Cullin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dr. Michael J.
Chudobiak
Sent: Wednesday, June 28, 2006 1:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Because: usg(2)[EMAIL PROTECTED] causes the app to exit with a non-zero status,
Because: [EMAIL PROTECTED]|g(2) causes asterisk to hard crash.
And trying to use g2 in either case doesn't work either.
Cullin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
The other problem is that if you add the gain to the original message,
it seems to me the volume on the phone will be too loud as compared to
the volume of the emailed message. Just a thought.
On Wed, 2006-06-28 at 15:41 -0400, Cullin J. Wible wrote:
Because: usg(2)[EMAIL PROTECTED] causes the
Yes - and it seems to prevent presence hints from working until the
phone is rebooted..
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Jun 26, 2006, at 9:28 AM, Douglas Garstang wrote:
Is
Not all of them... This has been a sticking point for the SoundPoint IP 501's.AlexOn 6/28/06, Cullin J. Wible
[EMAIL PROTECTED] wrote:Polycom phones support STUN - that should solve the issue too.
Cullin-Original Message-From: [EMAIL PROTECTED][mailto:
[EMAIL PROTECTED]] On Behalf Of Dr.
Hi Tj,
yes, you can run two TDM400s (or more) on the same cpu, and the channels are
1 to 4, and 5 to 8. (also, you can set one or more groups for outgoing
calls).
Interrupts are the main issue. As far as possible avoids that the cards
share interruptions.
cheers
Fabio
-Mensaje
Yes, I have more than one call per line enabled on the phone itself. I
have a value of 3 entered there, and that should be sufficient I would
think. So, the message I'm getting is coming from Asterisk. How do I
see what the console is saying?
Jerry Jones wrote:
Do you have more than one
We have the Polycom IP501's here and for £140+VAT
there great value and have excellent call quality.
Regards,
Dean.
- Original Message -
From:
Daniel Salama
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, June 28, 2006 8:18
PM
Just my opinion and I am no expert. I took the time
to crawl first. I installed basic Linux then I installed basic
Asterisk. After spending some time crawling with this setup I set up some
extensions and made some calls and watched the messages on the CLI while
Asterisk did it's thing.
Sure, but if one needs that many, much better off to use the Sangoma
A200 No MB problems and up to 24 channels.
John Novack
Fabio wrote:
Hi Tj,
yes, you can run two TDM400s (or more) on the same cpu, and the channels are
1 to 4, and 5 to 8. (also, you can set one or more groups for
The same thing happened to me... I had to get a linux expert to take care of it for me.
I believe the files were either libpamor libss. They were telling asterisk to shutdown. I believe they deleted the files it that fixed it.
Tighten down your firewall.
bp
On 6/28/06, Doug Lytle [EMAIL
Here is a firefox plugin that connects to
asterisk via IAX protocol.
http://moziax.mozdev.org/
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Il Neofita
Sent: Tuesday, June 27, 2006 8:55
AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject:
Hello,
I am trying to install Asterisk-Addon and got the following problem:
-
cc -shared -Xlinker -x -o app_saycountpl.so app_saycountpl.o
cc -fPIC -I../asterisk -D_GNU_SOURCE -DMYSQL_LOGUNIQUEID
-I/usr/local/mysql/include
Olle,
Will the metermaid patch help this issue?:
http://bugs.digium.com/view.php?id=7435
I believe that the fix is in res_features.c , but I do not want to
pursue it if it is already there.
Also, thanks for your hard work on that patch.
Our receptionist will really like the PARKINGEXTEN
Hi again,
Could anyone tell me from where I can get non-commercial G729 codec and its
installation procedure for Asterisk?
Thanks
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update
It's in french but it's free:
http://www.linphone.org/
On 6/27/06, Il Neofita [EMAIL PROTECTED] wrote:
Hi,
someone know a good webphone, possibily a free one
Thx
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing
We're not seeing that behaviour...
-Original Message-
From: Anthony Rodgers [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 28, 2006 1:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] '500 Internal Server' Error on
SIP NOTIFY
Yes
There is one there, with instructions:
http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/Cheers,
Christophe
On 6/28/06, Wasif [EMAIL PROTECTED] wrote:
Hi again,Could anyone tell me from where I can get non-commercial G729 codec and itsinstallation procedure for Asterisk?
On Wed, Jun 28, 2006 at 03:55:12PM -0500, Erick Perez wrote:
It's in french but it's free:
http://www.linphone.org/
http://www.linphone.org/?lang=us
But it's not a web phone by any means. Writing a soft phone in HTML and
javascript is practically impossible. So you'll see some Java applets,
Sorry
if this has been posted before but Im having an issue where I get the following
on my CLI.
ast_read:
Dropping incompatible voice frame on Local/XX of format ulaw since our
native form has changed to slin
A
call comes in on our main to toll free number on an ATT T1 line
On Wed, Jun 28, 2006 at 04:39:29PM -0400, William Piper wrote:
The same thing happened to me... I had to get a linux expert to take care of
it for me.
I believe the files were either libpam or libss. They were telling
asterisk to shutdown. I believe they deleted the files it that fixed it.
I have checked with SNOM, they have came to the same conclusion I
did, and that is Asterisk is not matching the external number to the
dial plan, and thus not making the external call.
I have taken a different approach, and now looking at the Asterisk-
PBX to manage the call diversion. A
There's a non-commercial version that doesn't break the patent holders rights?At least go to the end of the page on this link and read the Legal Stuff - Important, please read paragraph.
RobOn 28/06/06, Christophe Ngo Van Duc [EMAIL PROTECTED] wrote:
There is one there, with instructions:
On 6/28/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Wed, Jun 28, 2006 at 04:39:29PM -0400, William Piper wrote: The same thing happened to me... I had to get a linux expert to take care of
it for me. I believe the files were either libpam or libss. They were telling asterisk to shutdown. I
On 28 Jun 2006, at 22:17, Tzafrir Cohen wrote:
On Wed, Jun 28, 2006 at 03:55:12PM -0500, Erick Perez wrote:
It's in french but it's free:
http://www.linphone.org/
http://www.linphone.org/?lang=us
But it's not a web phone by any means. Writing a soft phone in
HTML and
javascript is
Hello -I currently have 10 DID's coming into one Asterisk server, I seem to be having some difficulty routing based on the DID dialed and am hoping someone on the list can assist me.Here's the relevant info:
Ingress SIP trunk:IP: 123.45.45.3456DID's
Douglas Garstang a écrit :
General question.
If you install a Digium card in an Asterisk system, and install zaptel drivers,
do this give any benefit of echo cancellation? Our PSTN gateway is a separate
Audiocodes box, so the zaptel card wouldn't actually be connected to anything.
I'm
Douglas Garstang wrote:
General question.
If you install a Digium card in an Asterisk system, and install zaptel drivers,
do this give any benefit of echo cancellation? Our PSTN gateway is a separate
Audiocodes box, so the zaptel card wouldn't actually be connected to anything.
I'm wondering
I too am using 2.2.2, but I'm loading my config files via HTTP. I was having some difficulty when I was using TFTP. Things were not as reliable for me, so I switched to HTTP. I've been stable since.
On 6/28/06, Herchi Silviu [EMAIL PROTECTED] wrote:
Hi Tom,
Thank you for your interest in my
If price is an issue, then Grandstream is the go.
If quality is the issue, then Snom or Cisco.
(poet laureate)
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Forrest
Beck
Sent: Thursday, 29 June 2006 5:06 AM
To: 'Asterisk Users Mailing List - Non-Commercial
Hi
From: Noah Miller [EMAIL PROTECTED]
Sorry for the long delay in responding. I didn't see you message until
now due to the postfix problems on the mailing list.
No problem. I've decided to dump the rPath PoundKey linux distro because it
was still using Asterisk 1.2.5 and it was pointless
Christopher Aloi wrote:
Hello -
I currently have 10 DID's coming into one Asterisk server, I seem to be
having some difficulty routing based on the DID dialed and am hoping
someone
on the list can assist me.
snip
Unless I'm misunderstanding you, how about trying this:
1. In your sip.conf:
-Original Message-
From: Jean-Michel Hiver [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 28, 2006 4:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Echo Cancellation
Douglas Garstang a écrit :
General question.
If you
How can I use the column setvar in my dialplan?
I am not sure if it is for that what I need:
Many phones have the same jump in place, but need a few variables
different, like tariff, silent, need_password,
I have for tariff = 4 variations, for silent=2, for need_password=2
... If I solve it
in sip.conf:
setvar=key=value
And then in your dialplan, you'll have a (global?) variable called ${KEY}
Doug
-Original Message-
From: Ronald Wiplinger [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 28, 2006 8:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
a) No it won't unless you connect it to a TDM circuit
b) I have an audiocodes too (mediant 2000 4E1), and I've
found the echo
cancellation to be superb. I'm surprised to see you're having issues!
Me too, given we're on fiber gigabit ethernet, with only a few test calls in
progress!
Same version, same problem...
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Doug Lytle
|Sent: Wednesday, June 28, 2006 11:42 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] asterisk shutdown
|
|Anton
Yes, I use tar balls 1.2.9.1.
And installation and setting of ooh-323 succeeded.
Best Regards,
Tetsuya
On Tue, 27 Jun 2006 23:39:35 -0700
Martin Joseph [EMAIL PROTECTED] wrote:
On Jun 27, 2006, at 8:24 PM, Richard Scobie wrote:
Tetsuya Yamamoto wrote:
I can't makel asterisk addon,
Im my case, the box is closed down so I dont think its an
intruder issue... Im puzzled...
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William
PiperSent: Wednesday, June 28, 2006 4:41 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
Hello - Thanks for the suggestions, I actually learned since my post that the problem is related to the formation of the initial SIP invite sent from our Sonus gateway through our NexTone SBC. Using your idea regarding the variable I think i've found a work-around.
When the Sonus sends the inital
maxlogins =3D 3;
if ((s =3D ast_variable_retrieve(cfg, general, maxlogins))) {
if (sscanf(s, %d, x) =3D=3D 1) {
maxlogins =3D x;
} else {
ast_log(LOG_WARNING, Invalid max failed login attempts\n);
}
}
And yes, that should be documented
asterisk -r
set verbose 3
On Jun 28, 2006, at 3:23 PM, Mike Staver wrote:
Yes, I have more than one call per line enabled on the phone
itself. I have a value of 3 entered there, and that should be
sufficient I would think. So, the message I'm getting is coming
from Asterisk. How do I
Has anyone considered the idea of splitting the sip registration
information in a realtime database from the actual configuration of the
peers?
I mean, instead of having a table full of the configuration information
(i.e. name, regexten, secret, etc) and registration information (i.e.
ipaddr,
Hi Group,
Just want to asked if some of you have experience 1
way audio. Currently I am using two asterisk box. One
handles the prepaid platform, and the other one is for
media gateway connection. I am using asterisk version
1.2.4 and a 4E1 digium card with echo cancellation. My
interconnection
Sounds like the registration interval in the phones is less than the required registration interval of the server. I had this occur when using a SIP phone with an ITSP.
Michael
On Wed, 28 Jun 2006 12:04:40 -0400, Von L. wrote:
Hello,
Here is a breakdown of the issue I am
I would say NAT somewhere misconfigured.
On 6/28/06, leonimar cape [EMAIL PROTECTED] wrote:
Hi Group,
Just want to asked if some of you have experience 1
way audio. Currently I am using two asterisk box. One
handles the prepaid platform, and the other one is for
media gateway connection. I am
Hi,
Does anyone know what asterisk do during the register invite
process using ARA (realtime) with Nat enabled?
Say there is an UA1 send an register request to asterisk. Asterisk
will parse the register request header (in which source file?). It
will get the necessary information update
Could you show me the what we need to change in asterisk to support
multi-asterisk with a common database operation?
On 6/23/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
- unplug [EMAIL PROTECTED] wrote:
BTW, do you mean this function will be included in next release?
When
will be the
Paul Hayes wrote:
Neil Cherry wrote:
[snip]
How did you get access to the web config? What user and is it
the default password/access code?
type it's IP address into a web browser. Username: admin, password: psw
is the default.
Thanks, I was missing the psw.
--
Linux Home Automation
JP Carballo wrote:
Ronald Wiplinger wrote:
I got a request for one customers to set-up 100 accounts.
I use usually the Caller-ID as the card number.
Is there a way to make it for 100 accounts easier?
To generate 100 cards is not a problem, but if it would work with one
account number would
This is known issue, we fixed it by putting an answer() in the dial plan before it gets forwarded,thefix transcode_via_sln=no (detailed in the bug tracker)didn't work for me. YMMV.
http://bugs.digium.com/view.php?id=4101
On 6/28/06, Kevin Savoy [EMAIL PROTECTED] wrote:
Sorry if this has been
Hi,
We have a page on the wiki just for phone reviews, but I think it needs
a bit of format change. Instead of individual reviews for each phone, I
think each person should review all phones they have worked with and
list the phones they have had access to and rank them in relation to
each
On Wed, 2006-06-28 at 19:10 -0700, Mike Fedyk wrote:
Hi,
We have a page on the wiki just for phone reviews, but I think it needs
a bit of format change. Instead of individual reviews for each phone, I
think each person should review all phones they have worked with and
list the phones
If anyone's interested, I've just put together a sip realtime patch,
figured anyone that uses realtime may want to have a look at it. The
patch basically takes the stuff asterisk updates (fullcontact, ipaddr,
port, regseconds, and username) out of the sippeers table and puts it in
it's own table.
Anybody know of a VoIP (preferably IAX) carrier here in the Atlanta
area? Using ones in Chicago and Denver is hit and miss these days.
Traceroutes seem to change often and the latency is inconsistent. I'm
wondering if someone more local would improve things.
--
Paul Dugas, Computer Engineer
On 6/28/06, Jan Berggren [EMAIL PROTECTED] wrote:
I am using Trixbox 1.0(Asterisk 1.2.7.1)at a customer site. They whishes to
change the default Voicemail behavior.
Standard behavior
No answer/Busy - send to Voicemail
Requested behavior
No answer/Busy - message that if you press 9 you will
It is totally nat, first try to port map the ports through your firewall, on the network page set the rtp and sip ports plus the nat ip to use. I had the exact same problem and this was the only solution.
Or add the following to your config for the phone:
nat.mediaPortStart="5004"
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