On Tue, 2006-06-06 at 11:37 +0600, [EMAIL PROTECTED] wrote:
try asterisk -rx 'show channels'
that is what I did try, yes I ommited the quotes in the email guess it
wasnt understood that it returns only the header and not any information
on what channels are in use nor any information on how
Hi Patrick,
Let me see if I am following you here. When a caller calls in, obviously
you want them to be in the first queue level based on your dial plan.
Now, how do you want the caller to reach the next queue? Is the only way
a caller going to go to the next queue via a transfer from the
On 12:33, Mon 05 Jun 06, Douglas Garstang wrote:
I guess this is wy beyond my knowledge of subversion. I just started
playing with the directory structure I might use, and first thought was
something like this:
[EMAIL PROTECTED] ~/cfg $ ls -l
total 16
drwxr-xr-x 2 dougg users 4096
At 10:58 PM 6/5/2006, you wrote:
exten = _310.,1,NoOp
It can get tricky if several pattern cover the same range. But I odn't
believe that this is the case.
There's likely lots of ways to do what he wants, I thought he asked
for a solution to a particular problem, matching a 3 digit number
Hi,
It is impossible for
CPU to do so heavy task. Is there anybodyuse4E1 card work well on one PC
server? For the PCI limited and the software dsp, the voice quanlity will be not
acceptable. Could anyone give me some evidence?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On
I also want to know this.
This would be very useful in Call center for remote attendent. The E1
gateway will do this very well, but where is the BRI voip gateway?
Hawk
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of James Harper
Sent: Tuesday, June
Hi,all.
There have any STUN spport for asterisk?
thanks,,,-- Jeffery `∧ ∧︵ ミ^r^ミ灬)~iaxtel Num: 1-700-576-1311fwdnet Num: 728150http://www.diaip.com
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
On Tue, 2006-06-06 at 15:17 +0800, Chen Fan wrote:
Hi,all.
There have any STUN spport for asterisk?
thanks,,,
where asterisk queries a stun server or where asterisk acts like a stun
server?
Becuase stun is totally self contained it would be silly (in my opinion
anyway) to have a stun
James and Armin,
Turn on asterisk debugging too. Capi seems to be working okay, maybe
asterisk isn't picking up the call for some reason. Maybe:
asterisk -r
set verbose 9
set debug 9
capi debug
then make an incoming call and copy the output into an email and send it
to the list (unless it
Hi,
I am in a problem. Can anybody help me out.
I am trying to establish connection using hyperterminal through IAXsoft
modem using asterisk PBX. I have done the following settings in the
configuraion files of asterisk.
1) iax.conf file:
[iaxmodem]
type=friend
Try to see this...
http://www.patapsco.co.uk/applications/isdn_conversion_sharing_and_simulation/share_pris/share_pri.htm
On 6/6/06, Hao Xu [EMAIL PROTECTED] wrote:
I also want to know this. This would be very useful in Call center for remote attendent. The E1 gateway will do this very well,
Wes Baehr ha scritto:
(Sometimes) When I’m monitoring calls, I hear a very bad jitter –
usually only on one of the bridged channels. So at first I thought it
was just the one end of the conversation actually causing the jitter –
but it’s not. So I called in from another device to spy at the
I thought I had this fix but I just upgraded from 1.2.7 to 1.2.9 and it
seems to be happenning again
Using Sipura SPA-3000 with *. I want DTMF tones to go thru when I call out
on PSTN - I.E. if I call my bank or external VM and need to put out DTMF.
I found with 1.2.7 if I turned off all
hi,
We need STUN client support for asterisk...
becasue the service provider only offer STUN interface,, so i can not connect asterisk to their server
i have found that there someone is develop res_stun.c ..but still not release...
regards
On 6/6/06, trixter aka Bret McDanel [EMAIL
On Tue, 2006-06-06 at 16:12 +0800, Chen Fan wrote:
hi,
We need STUN client support for asterisk...
becasue the service provider only offer STUN interface,, so i can not
connect asterisk to their server
all stun does is resolve your external IP by sending data to a foreign
server
HI,
There is a parameter NAT can be set in the configuration file. Is
it the way that we can use to support NAT by setting nat=yes in the
file instead using other NAT resolving tools like stun?
On 6/6/06, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
On Tue, 2006-06-06 at 16:12 +0800,
Just a quick question: Is there a driver for a normal modem to be used as an
FXS line (to connect a normal analogue phone to your PC)?
Thanks,
Pieter
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE
Which chip set?
Cheers,
Madhawa
pieter Claassen wrote:
Just a quick question: Is there a driver for a normal modem to be used as an
FXS line (to connect a normal analogue phone to your PC)?
Thanks,
Pieter
___
--Bandwidth and Colocation provided by
On Tue, Jun 06, 2006 at 11:09:28AM +0200, pieter Claassen wrote:
Just a quick question: Is there a driver for a normal modem to be used as an
FXS line (to connect a normal analogue phone to your PC)?
A normal modem may serve as an FXO line with the proper hardware, as it
is basically a phone.
Hi,
I use an E1-Board to hand the calls over to internal SIP-Clients. My
Question is which Dialstatus is set when the SIP-client is unreachable.
I tried with NOANSWER but does not seem to be suitable.
Does anyone of you have a solution?
In voip-info.org wiki there is a Dialstatus CHANUNAVAIL but
-d6c5, 0 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(SIP/200-d6c5,
recordingcheck|20060606-110927|1149588567.614) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060606-110927|1149588567.614: Outbound recording
I want that incoming callers to hear a welcome message while the phones
ring. I know I can use Dial with the m(class) option to make the same
with musiconhold, but the problem is that musiconhold does not start
from the beginning of my mp3 file. If I use Playback or Background, the
phones do
PRI is primary rate ISDN, consisting of (normally) 30 channels or
(rarely - US only I think) 24 channels.
BRI is basic rate ISDN, and consists of 2 channels.
They are not the same thing. The redfone is a PRI to TDMoE converter,
I'm after something that does the same thing for BRI.
Thanks
James
Hi list!
Are there any changes in the behaviour of the Dial command between
1.2.7.1 and 1.2.8.?
I am forwarding calls to my legacy PBX using :
exten = s,1,Dial(Zap/g1/8210,90,r)
Ever since I upgraded to 1.2.9 it seems as if the Legacy PBX is no longer
receiving the extension I am calling on
Hi!I use the following configuration to register my asterisk server to my SIP provider:register = 12345:[EMAIL PROTECTED]/12345sip.conf
:[sipout-test]type=peerusername=12345fromuser=12345fromdomain=provider.comsecret=passwdinsecure=veryhost=sip.provider.com
I believe if you use the new native music on hold feature it always
plays the music on hold starting from the beginning.
On Tue, 2006-06-06 at 11:15, Tommaso Calosi wrote:
I want that incoming callers to hear a welcome message while the phones
ring. I know I can use Dial with the m(class)
I am working on an AGI script to do just this. The idea is to use the
XML database search at localcallingguide.com and decide if the call is
local or LD based on parameters that you provide to the script. Will
keep you posted on the progress of this. I'm hoping to have it done
soon. The
Tzafir:This was the result of the test you gave me. Obviously the message was not sent.On my environment I have a mail server and the [EMAIL PROTECTED] server. I just want to take one email out of the box, no matter if it is to internet or inside my network. Thanks for your help and please keep
Today is Swedens national day - since a few years a holiday too.
We don't have a tradition on how to celebrate.
Sweden has not been to war for a very long time, so there's no real
spirit
for the country here - it's been aroundfor such a long time, so
what? :-)
Guess we have to learn from
Hi,
We are trying to port over a PABX to our network. Both PRI's seem to be
live however, whenever someone dials out from the PABX Asterisk happens to
report :
-- Extension '' in context 'samsungincoming' from '736327438' does not
exist. Rejecting call on channel 0/31, span 2
If crc4 is
I will look out for that software. I´ll let you know. Thank you.YrvingAlex Robar [EMAIL PROTECTED] escribió: Yrving,OK, so you've definetly got a sendmail configuration problem. If you're not big on config files, try using Webmin. It's a web-based interface for thousands of administrative tasks
Samsung PABX?
Its TEPRI probably configured in overlap mode so you need to configure
asterisk span that is connected to PABX to overlap mode as well.
When user selects the outside line in overlap mode PABX connects to
asterisk and then sends the digits to it as the user presses the key's.
If
Thanks mate. All going well.
Regards,
Sahil Gupta
VoiceValley
On Tue, 6 Jun 2006, Boris Bakchiev wrote:
Samsung PABX?
Its TEPRI probably configured in overlap mode so you need to configure
asterisk span that is connected to PABX to overlap mode as well.
When user selects the outside line
Does any of these asterisk users know, one Mr Jeffry from Kochi, India,
Please let me know
Thanks
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Please any one knows how to configure the weather on
asterisk or if there a weather channel I can subscribe to it
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail
[EMAIL PROTECTED]/Trixbox includes built-in weather AGI scripts. You should have some good luck getting weather on your Asterisk box if you look into those scripts.AlexOn 6/6/06,
Khaled Chehab [EMAIL PROTECTED] wrote:
Please any one knows how to configure the weather on
asterisk or
http://nerdvittles.com/index.php?p=134On 6/6/06, Khaled Chehab
[EMAIL PROTECTED] wrote:
Please any one knows how to configure the weather on
asterisk or if there a weather channel I can subscribe to it
*
No employee or agent is
CPE-Asterisk(NT Mode)-ip-Asterisk(CPE)-NT?
maybe it will fit for you? if yes, i think you can work with the following
budget:
via epia board ~85$
mini itx case (small size!) ~85$
ram ~20$
DiskOnChip (or HDD) ~20 - ~50
HFC BRI ~50$
so globally ~300-350/side
you can also go for patton something of
- Doug Crompton [EMAIL PROTECTED] wrote:
I am getting the following error at the end of 'make install' 1.2.9
I have not tried to find it but I suspect there is just a misplaced
punctuation. It runs fine.
That part of the Makefile has not been touched in quite a while, so there
should
- trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
It acts almost like a race condition that 'wins' when the channel
count
is low, but looses almost always when it gets to a moderate level.
Why
I was thinking it was a threading issue.
I believe this has been a known problem for a
On Tuesday 06 June 2006 02:56, Hao Xu wrote:
It is impossible for CPU to do so heavy task. Is there anybody use 4E1
card work well on one PC server? For the PCI limited and the software dsp,
the voice quanlity will be not acceptable. Could anyone give me some
evidence?
YATE has multiple
- Remco Barendse [EMAIL PROTECTED] wrote:
Did I goof up or did something change?
No, there should not be any behavioral changes between 1.2.7.1 and 1.2.8 except
for bug fixes. The only change that I see in the ChangeLog for 1.2.8 that could
be relevant is the one regarding
On Tue, 6 Jun 2006, Esteban Guana-Jarrin wrote:
James and Armin,
Turn on asterisk debugging too. Capi seems to be working okay, maybe
asterisk isn't picking up the call for some reason. Maybe:
asterisk -r
set verbose 9
set debug 9
capi debug
then make an incoming call
Wether the SIP client is not registered or does not exists at all you
will get CHANUNAVAIL.
Regards
On 6/6/06, Christophorus Laube [EMAIL PROTECTED] wrote:
Hi,
I use an E1-Board to hand the calls over to internal SIP-Clients. My
Question is which Dialstatus is set when the SIP-client is
On Tue, 6 Jun 2006, Kevin P. Fleming wrote:
- Remco Barendse [EMAIL PROTECTED] wrote:
Did I goof up or did something change?
No, there should not be any behavioral changes between 1.2.7.1 and 1.2.8 except
for bug fixes. The only change that I see in the ChangeLog for 1.2.8 that could
I tried with CHANUNAVAIL but I was not successful. I want to try to call a
SIP client. If it is not answering and cannot be found I want wo call
someone else.
How can I do that? NOANSWER and CHANUNAVAIL do not work out.
Wether the SIP client is not registered or does not exists at all you
will
- Steve Underwood [EMAIL PROTECTED] wrote:
Asterisk should really import a recent version of Speex. The last time
I
checked it had an ancient version. Quality has improved, and
computation
has significantly reduced.
As far as I know, Asterisk has never included the Speex library in
this is what I have, and it works on Asterisk-1.2.1
[macro-sipextens]
exten = s,1,Macro(validate_extension)
exten = s,2,Dial(SIP/${sipprefix}${num},${calltimeout}|${calloptions})
exten = s,3,Macro(catch_dial_response,${DIALSTATUS})
so, After Dial, I catch the dial response, and heres the catch
On 15:46, Tue 06 Jun 06, Remco Barendse wrote:
On Tue, 6 Jun 2006, Kevin P. Fleming wrote:
- Remco Barendse [EMAIL PROTECTED] wrote:
Did I goof up or did something change?
No, there should not be any behavioral changes between 1.2.7.1 and 1.2.8
except for bug fixes. The only change
I am using [EMAIL PROTECTED] v 2.6
I want to active or deactivate voicemail from command line
Like database put AMPUSER/VOICEMAIL/111 ENABLE this command is applicable at
version 2.8 but it don't work at 2.6
Any one can help me ??
Regards
*
Check out this example dialplan: http://pastebin.ca/19456
That should give you everything you need.
bp
On 6/6/06, Moises Silva [EMAIL PROTECTED] wrote:
this is what I have, and it works on Asterisk-1.2.1[macro-sipextens]exten = s,1,Macro(validate_extension)
exten =
OK Great. I will wait for your success!
Are you using Perl?
Doug
On Tue, 6 Jun 2006, Jason Bachman wrote:
I am working on an AGI script to do just this. The idea is to use the
XML database search at localcallingguide.com and decide if the call is
local or LD based on parameters that you
Hi,
maybe this helps ?
http://www.snom.com/wiki/index.php/DHCP
http://www.snom.com/wiki/index.php/Massdeployment_Firmware_Release_5
For further questions in that regard feel free to contact us at
[EMAIL PROTECTED] !
Regards.
On Friday 02 June 2006 05:49, Remco Barendse wrote:
Hi list!
On Tue, 6 Jun 2006, Michiel van Baak wrote:
On 15:46, Tue 06 Jun 06, Remco Barendse wrote:
On Tue, 6 Jun 2006, Kevin P. Fleming wrote:
- Remco Barendse [EMAIL PROTECTED] wrote:
Did I goof up or did something change?
No, there should not be any behavioral changes between 1.2.7.1 and
Hi Erick. Just for record I could compile the application but could not
use. It was crashing the asterisk.
Then I downloaded the 0.5 release and used the 64_Makefile to compile.
Now its working on test environment but until now everything is ok.
Does anyone know a good syslog server to use for grandstream phones? I want
to set this up to see what is happening with the grandstreams. Easy and
Free preferably.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing
hmmm... I am a huge fan of syslog-ng, but the stock syslog on your *
system should work well...
Matthew Warren wrote:
Does anyone know a good syslog server to use for grandstream phones? I want
to set this up to see what is happening with the grandstreams. Easy and
Free preferably.
On Tue, 6 Jun 2006, Michiel van Baak wrote:
On 15:46, Tue 06 Jun 06, Remco Barendse wrote:
On Tue, 6 Jun 2006, Kevin P. Fleming wrote:
- Remco Barendse [EMAIL PROTECTED] wrote:
Did I goof up or did something change?
No, there should not be any behavioral changes between 1.2.7.1 and
Hello,
I've installed a QuadBri card from Junghanns, and have some problems for
compiling software.
Compiling Zaptel and LibPRI are OK. But when i want to compile ztgsm
files, i have the error:
link /usr/src/linux-2.6 to your kernel sources first !
I work with kernel 2.6.8-2-686, and i have
Yes it does, I just set our system up that way.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Gareth
Blades
Sent: Tuesday, June 06, 2006 6:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Playback welcome
On 11:02, Tue 06 Jun 06, Matthew Warren wrote:
Does anyone know a good syslog server to use for grandstream phones? I want
to set this up to see what is happening with the grandstreams. Easy and
Free preferably.
Hi,
Use the syslog on your asterisk box :)
--
Michiel van Baak
[EMAIL
I'm having a problem with a new installation of asterisk 1.2.5 with a
digium dual port T1 (span 1 connected to an outside line, and span 2
connected to a CAC access bank I channel bank with 24 fxs ports). When I
start Asterisk (either from safe_asterisk or asterisk -vvvc) it will
immediately
www.kiwisyslog.com works perfect but windows only hth
-Original Message-
From: Matthew Warren [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 06, 2006 9:03 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] syslog server
Does anyone know a good syslog server to use for
2006/6/6, Gareth Blades [EMAIL PROTECTED]:
I believe if you use the new native music on hold feature it alwaysplays the music on hold starting from the beginning.Where can I find this new native music on hold feature ?
In Asterisk 1.2.x ?Regards
___
Is anyone using Vonage on an FXO port in Asterisk? How well does it work?
Specifically, any echo/delay problems?
Second part, I am assuming it is possible to separate fxo ports for least
cost routing correct? In other words, I would like the routing to be such
that any local or 1-800# dials
On Tue, Jun 06, 2006 at 05:27:58PM +0200, Olivier Saulnier wrote:
Hello,
I've installed a QuadBri card from Junghanns, and have some problems for
compiling software.
Compiling Zaptel and LibPRI are OK. But when i want to compile ztgsm
files, i have the error:
link /usr/src/linux-2.6 to
Ok I tried the 1.2.7 install again and I get the same result. I do not
remember that from when I installed it the first time. Obviously my shell
does not like something in the context that checks for old modules. At
first I thought it was because there were old modules there but deleting
them
On Tue, Jun 06, 2006 at 10:10:17AM -0600, Nathan Bell wrote:
I'm having a problem with a new installation of asterisk 1.2.5 with a
digium dual port T1 (span 1 connected to an outside line, and span 2
connected to a CAC access bank I channel bank with 24 fxs ports). When I
start Asterisk
2006/6/6, Woodoo People .pGa! [EMAIL PROTECTED]:
CPE-Asterisk(NT Mode)-ip-Asterisk(CPE)-NT?Have you already tried such setup ?What are the benefits of using Asterisk instead a dedicated CPE ?regards
___
--Bandwidth and Colocation provided by Easynews.com
I used a scenario like this before but I always ran into intermittent echo
issues that were just not worth the hassle for me so I switched to a sole IP
origination and termination service.
Just my personal experience!
HTH
Curt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Ok trying this again... is there anyone using the SPA-3000 with *
I am not sure if this is a specific problem to it or not. This is
something I really need to fix!!!
When dialing out using * interfaced to an SPA-3000, fxs,fxo, I cannot
access (reliably) DTMF menus at the called party, after
On Tue, 6 Jun 2006, Kevin P. Fleming wrote:
- Remco Barendse [EMAIL PROTECTED] wrote:
Did I goof up or did something change?
No, there should not be any behavioral changes between 1.2.7.1 and 1.2.8 except
for bug fixes. The only change that I see in the ChangeLog for 1.2.8 that could
Hi, I tried to find a reference in terms of size but got back a bunch
of tech documents and couldn't get the idea of wav49 format.
wav49 format is supposed to be half the size of a normal wav right?
so, how much disk space takes to save one minute of audio in wav49?
I trying to do some capacity
The Asterisk Development Team today re-released Asterisk 1.2.9.1 and
Asterisk 1.0.11.1 to address a security vulnerability in the IAX2
channel driver (chan_iax2). The vulnerability affects all users with
IAX2 clients that might be compromised or used by a malicious user, and
can lead to denial of
Dear list (and more specifically Bret),
I am getting one-way (inbound only) audio when trying to place a SIP call
via voip.trxtel.com (i.e. [EMAIL PROTECTED]). The Cli spits out
== Forcing Marker bit, because SSRC has changed 5 times after atempting a
native bridge. I realize this is most
I am
running on 1.2.7.1
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of
OlivierSent: Tuesday, June 06, 2006 12:17 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Playback welcome message while
The errors were just these:
Jun 6 10:03:06 ERROR[14553] pbx_dundi.c: Unable to load config dundi.conf
Jun 6 10:03:06 ERROR[14553] chan_iax2.c: Unable to load config iax.conf
There were other warnings and notices for the other conf files, and that
was it.
However, I just noticed a brand
I am using it with my [EMAIL PROTECTED] setup. I did not face any issues with
echo, but
once in a while, the trunk does NOT get disconnected even after the call has
been completed. So I had to manually plug the phone cable out from FXO and
plug it back again. But I think that's something to do
try setting dtmf playback length to .5 in the admin section of the
Sipura and try again.
On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote:
Ok trying this again... is there anyone using the SPA-3000 with *
I am not sure if this is a specific problem to it or not. This is
something I really
Thanks, it works for me too.
Tim Sharp wrote:
Yes it does, I just set our system up that way.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Gareth
Blades
Sent: Tuesday, June 06, 2006 6:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I'd suggest calling whoever you buy your phones from. The distributor I work with requires that you are Polycom certified to be able to purchase phones from them, but once you are certified with Polycom you can actually download the firmware from their extranet.
On 6/5/06, Douglas Garstang
- Brent Torrenga [EMAIL PROTECTED] wrote:
== Forcing Marker bit, because SSRC has changed 5 times after
atempting a
native bridge. I realize this is most certainly a NAT issue, the *
server is
behind one. Sip.conf has externip=, and localnet=.
google is your friend :-) We've already
Hi All,
I'm having a really weird can reinvite issue. I've been banging my head
around on this for days now..
I have two asterisk servers. One at 172.20.0.11 One at 172.20.2.5
172.20.0.11 is a hosted box and serves multiple offices
172.20.2.5 is a box on site at a customer's office.
A phone
It would be helpful if responders would tell us what FXO hardware they
are using and which vonage ATA device it connects to.
Padmanaban Balasubramaniam wrote:
I am using it with my [EMAIL PROTECTED] setup. I did not face any issues with
echo, but
once in a while, the trunk does NOT get
Tried that makes no difference. Did it for you? What DMF method(s) are you
using. Looking at a goggle search yields lots of talk on this but no real
solution. Apparently there is an rfc2833 issue and * is working on it???
Also it appears the codec used might be an issue. This is a serious
problem
Also to expand on this... when listening to opposing phone in a connected
call over PSTN you hear a click followed by a very short burst of DTMF
audible energy. Same in both directions.
I can't be the only one having this problem!
Doug
On Tue, 6 Jun 2006, Tom Vile wrote:
try setting dtmf
Hello,
I have an asterisk server running with 23 g.729 licenses. I have
also purchased a sound file from thevoice.digium.com. I need to
covert this file (uLaw, PCM I think) to g.711, g.729 g.723 for use
with an IVR system. Is there a way I can convert the files using the
g.729
Using AVT in my sipura with above settings and it work fine going out
the PSTN. There was an issue a while back with an older version of
Asterisk with one of my providers but it has been fine since the
upgrade. I also use ulaw for calls.
On 6/6/06, Doug Crompton [EMAIL PROTECTED] wrote:
Tried
Well, this kinda sux.
We have three Asterisk servers. Phones register to a single,
primary server.
When a phone on one wants to reach a phone on another, we use
DUNDi to discover the destination pbx and IAX to transfer the
call to the other Asterisk box. This seems to be a fairly
common
William Piper wrote:
For Problem #2:
I'm not sure what you are asking. Perhaps post your dialplan for this
problem we will take a look.
bp
On 6/4/06, *M.Hockings* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Problem 2) Incoming sip calls from my voip provider get rejected
Have you tried turning off icmp redirect on your router?On 6/6/06, Brett N [EMAIL PROTECTED] wrote:
Hi All,I'm having a really weird can reinvite issue. I've been banging my head
around on this for days now..I have two asterisk servers. One at 172.20.0.11 One at 172.20.2.5172.20.0.11
is a hosted
We use Motorola v551's as extensions on our Asterisk system with a
homebrew find me/follow me dialplan. It works great except where coverage is
poor then of course the inbound call hits voicemail. This has nothing to do
with Asterisk and everything to do with our cellular provider, but since you
in Fact we saw similar problems with all sipura products. We think its a default value thats not quite right for the north american market, these units are built and tested in asia mostly.one simple test to check it out is call this number
www.nextwavetitaniumplus.com Toll-Free Account
We released a critical update for idefisk. (Version 1.37 now ships with
a patched iaxclient library).
Everybody is urged to update asap. (
http://www.asteriskguru.com/idefisk/free/ )
A big thanks to coresecurity and Steve Kann for the early warning.
Zoa.
The Asterisk Development Team
In my experience, this can be pretty cumbersome. I could be wrong but I
think the reason I stopped doing it was that the phone would restart
when you applied ANY changes, and you'd have to wait like 90 seconds or
more to be able to re-access the phone via http.
Moj
Avi Miller wrote:
Stephen
Tzafrir Cohen a écrit :
wget
http://rapid.dotsrc.org/rapid/pool/main/z/zaptel/zaptel-modules-2.6.8-2-686_1.2.5-4+2.6.8-16sarge1_i386.deb
It use as dependance the zaptel deb package, but in the website, only
release for i386 is available, is it good??
When i depackage it, i have the
I have a small Cepstral howto on my blog..
http://www.voipphreak.ca/archives/269-Even-More-Asterisk-Weather-Now-Cepstral.html
On 06/06/06, David K Parker [EMAIL PROTECTED] wrote:
http://nerdvittles.com/index.php?p=134
On 6/6/06, Khaled Chehab [EMAIL PROTECTED] wrote:
Please any one
AVT??? I have ulaw allowed (only) - When you call your cell via
pstn/spa-3000/* and listen on both while pressing dtmf do you hear good
clean tones of enough duration to allow detection, in both directions?
Do you access DTMF required services over pstn, like banking, vm, etc
from local * system?
The only thing I have found that tends to point to an * problem is
http://bugs.digium.com/view.php?id=6667
It is a long read and I have no ideas what the disposition is. It was a
discussion back in late March. This seems to apply to all or many SIP
connected devices and around implementation of
I have a handful of Linksys PAP2-NA's all talking nicely to Asterisk using
standard telephones. I've been running them for the better part of this
year. No complaints whatsoever. We chose the PAP2-NA's mainly due to cost
and especially the ease of provisioning.
In an effort to inexpensively
1 - 100 of 187 matches
Mail list logo
