Since many on this list use cisco ip phones I thought they may find this
information worthwhile to know
http://www.SecurityTracker.com/alerts/2005/May/1014043.html
--
Trixter http://www.0xdecafbad.com
UK +44 870 340 4605 Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
Actually G.729A is a reduced complexity version, and G.729B is a version
with silence suppression. The data rate while sending voice is exactly
the same, although the quality of G.729B should be a little higher.
However the average rate for B can be lower if the silence suppression
is used.
Ivan Meic (Vox Mundi) wrote:
Actually G.729A is a reduced complexity version, and G.729B is a version
with silence suppression. The data rate while sending voice is exactly
the same, although the quality of G.729B should be a little higher.
However the average rate for B can be lower if the
On Tue, May 24, 2005 at 08:18:24AM -0400, Christopher Kenna wrote:
Sorry about last posting, typo...
I just added 2 Digium X100P cards. When my * box boots, it found them
and configured them. When I enter genzaptelconf, it comes back with
the following error:
genzaptelconf generates
Welcome to the Asterisk users community!
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Asterisk.org is a fast moving project. New code is added every
day.
The next community meeting is
Hi,
spandsp-0.0.2pre18 works fine for txfax with HT488(version-1.0.1.2),
but rxfax doesn't work. After some FAX sounds, it hangup!
Could someone tell me how to debug?
The following is the * CLI log
to 192.168.0.161:43222
-- Executing NoOp(SIP/4881-bde9, ) in new stack
-- Executing
On Wed, 25 May 2005, Shane Burrell wrote:
Anyone with any comments on DSS buttons and general phone features?
The BLF (Busy Light Field) part of the DSS buttons are not active in the
latest firmware.
The microphone part of the speaker phone needs some work, possibly just
software (too low
C F wrote:
are these phones behind nat?
Yes, but correctly registered. The same fones dont have any problems
when registered to a SER Server.
Can constantly reloading the configuration cause problems?
cheers,
Arnd
___
Asterisk-Users mailing
Hi,
I was having this problem with Gradstream BT101's with Asterisk @ Home
version 0.7.
The problem was that there was a sip channel still open (as far as asterisk
and the phone were concerned) however this sip channel was not actually in
use. The existence of this sip channel meant that whilst
El mar, 24-05-2005 a las 11:10 -0500, Nathan Pralle escribi:
Don't know if this had been done before, but I went and HTMLized for my
own use (and anyone else's) the Asterisk Sounds listing(s). Here are
the links to the finished pages:
Thank you very much. I'm looking for this for a
Hello Everybody,
I was going thru the C code of Asterisk. Does anybody know how does one go
about modifying the C code of Asterisk? Please do reply
Regards,
Bharat M. Sarvan
EZZI BPO Pvt Ltd.,
PUNE.
___
Asterisk-Users
Hi!
I'm running the latest asterisk (cvs of yesterday) and I'm seeing some
SEGFAULTS.
The box is running fine for a while but then bombs out.
The box has a TE110P PRI card and some sip phones, no other hardware.
Any ideas?
Thanks!
Remco
___
After I did rxfax(...|debug), I got the followings;
..(snip)...
DIS: 80 00 ce f4 80 80 81 80 80 80 18
HDLC underflow in state 9
Changed from phase 4 to 3
T4 timeout in state 9
Changed from phase 3 to 4
DIS:
Prefer 256 octet blocks
Can receive fax
Supported data signalling rates:
Hi,
I have the following config:
[7960] --skinny-- [Cisco CCM] --SIP_trunk-- [asterisk] --SIP--
[X-lite]
Is there a chance to avoid the RTP stream from passing through the Cisco
CCM ? I would like to have all RTP handled by the *.
This is just a testbed, for a larger project. What I want to
Fellow Canadians:
Please excuse this brief bit of self-promotion but in my defence it does
pertain to this thread.
Being in Canada it makes it very difficult to find companies that will
ship COD from the US. If I was to order I would only order COD from
now on from VoipSupply.
We are in
Paul Dracevich wrote:
I have set it up, but I get an error, to do with the keys, if I can get
past that part I will have no problems setting up the mappings, dial
rules etc.
Do u have any ideas, on this error?
Have you generated the keys?
Are they in the right directory?
bye
Ronald
The clock on your computer still thinks it's April... Please fix!
As for editing C code.. any decent text editor will do the job.
On Mon, 2005-04-25 at 13:26 +0530, Bharat M. Sarvan wrote:
Hello Everybody,
I was going thru the C code of Asterisk. Does
anybody know
Hi all,
I have problem with my Asterisk.
I'm using the softphone Xten-Lite.I've
removed the SIP client information in sip.conf. The softphone can't register to
Asterisk, but it can make outgoing calls.
I've tried to add back the SIP client information
into the sip.conf, but make a wrong
I've got a Polycom IP 600 that doesn't want to DHCP. It DHCPDISCOVERs,
there's a DHCPOFFER, it DHCPREQUESTs and a DCHPACK is returned, but 3
seconds later it repeats the process and never boots. The phone works fine
with a static IP, and the DCHP setup works ok for an IP 500. I updated to
On Tue, May 24, 2005 at 10:58:17PM -0700, trixter http://www.0xdecafbad.com
arranged a set of bits into the following:
Since many on this list use cisco ip phones I thought they may find this
information worthwhile to know
Just a further warning, I was informed last night of a few other known
Hi all,
I'm
trying to setup Asterisk trunk to Siemens HiPath 4000 V2.01
What
would be the best way to do so? I am a bit confused because as far as I've
understand this PBX doesnt support H323, but I saw somewhere someone who
created a cornet trunk and it worked using H323.
So
if
Hi
I am currently developing a IVR application using
PHP/AGI. I am using the PHPAGI class at
http://phpagi.sourceforge.net/ to handle the
commuication with my *.
The application basically asks a caller to enter in
some information which is then processed and a answer
is read back out to them. I
Hi all,
I have Asterisk working great for a while now, and until now, only
needed to forward calls which originate from one country (Holland, +31).
We have a Wildcard TE410P card and configured it in zapata.conf as:
switchtype: euroisdn
pridialplan: unknown
...etc
This was all fine, but now we
On Monday 25 April 2005 08:56, Bharat M. Sarvan wrote:
Hello Everybody,
I was going thru the C code of Asterisk. Does
anybody know how does one go about modifying the C code of Asterisk? Please
do reply.
How many times must you ask this?
If you do not know how to
hello all,
i been doing a voip project in campus. i incharge at
the asterisk side. i'm using Digium TE110P. the 31
channel are already configured and then i get green
light.
to make asterisk work with TE110P i must have the zap
module. but i don't know how i can make zap module. i
have go
Hello
Did you tried a deadagi in place of agi
A++
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part
de Jon Farmer
Envoyé : mercredi 25 mai 2005 11:40
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] PHP/AGI Problem
The problem is as
hello there,
i also using the same board and configure for the E1
before this i also got the same problem.
what the problem are:
1. cabling
2. and the zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31 # set this to 1-15,17-31 for E1
dchan=16 # set this to 16 for E1
i get this
pridialplan: - called party
prilocaldialplan: - calling party
try:
prilocaldialplan= international
-b
- Original Message -
From: Leon de Rooij [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, May 25, 2005 11:41 AM
Subject: [Asterisk-Users] Possible to send
--- Thierry Wehr [EMAIL PROTECTED] wrote:
Hello
Did you tried a deadagi in place of agi
A++
I am calling the PHP app via deadagi. I believe what
might be happening is that the application is going
into a internal loop waiting for DTMF to know what to
do next. I am going to investigate
hello
like if 6000 is the main exchange number. any one dial
to 6000 will be asked for pressing his desired
extension then he can press his desired extension then
his number is diled
exten=6000,1,Background(enterdesiredexten)
exten=6000,2,Wait(2)
On 25/05/05, Kamran Ahmad [EMAIL PROTECTED] wrote:
hello
like if 6000 is the main exchange number. any one dial
to 6000 will be asked for pressing his desired
extension then he can press his desired extension then
his number is diled
exten=6000,1,Background(enterdesiredexten)
Bob Goddard wrote:
On Monday 25 April 2005 08:56, Bharat M. Sarvan wrote:
Hello Everybody,
I was going thru the C code of Asterisk. Does
anybody know how does one go about modifying the C code of Asterisk? Please
do reply.
How many times must you ask this?
Fantastic, it works ! Thanks you very much :)
Leon
On Wed, 2005-05-25 at 12:04 +0200, barney wrote:
pridialplan: - called party
prilocaldialplan: - calling party
try:
prilocaldialplan= international
-b
- Original Message -
From: Leon de Rooij [EMAIL PROTECTED]
To:
Kamran Ahmad wrote:
hello
like if 6000 is the main exchange number. any one dial
to 6000 will be asked for pressing his desired
extension then he can press his desired extension then
his number is diled
exten=6000,1,Background(enterdesiredexten)
exten=6000,2,Wait(2)
IMHO!
I just see a skype channel as something good for asterisk.
Skype has broad coverage.
I can't imagine that skype wouldn't be interested in selling corporate accounts
skype trunk lines.
Imagine having unlimited or X amount of continious calls coming in on SkypeIN
and out on SkypeOUT from
Hello, All!
We was upgrade our Asterisk from version
0.7.2 to 1.0.7.
And havebig problem.
When asterisk starts:
---
*CLI h.323 show codecsAllowed
Codecs:
Table: G.729A{sw} 1 G.729{sw}
2 G.723.1{sw} 3
G.711-uLaw-64k{sw} 4Set:
0: 0:
Also for some odd reason when I ring an extension attached to my
sipura 2100 ATA it takes it about 12 seconds to start ringing after I
dial it (sits there with dead air on the calling phone).
After you dial, push '#' to actually start the call. Or update the dial
maps in the sipura. But # is
Look for a CCIE or better. Contact me off list if you can't find one.
On 5/24/05, Adam Collard [EMAIL PROTECTED] wrote:
Is anyone here familiar with configuring Cisco routers? I have a Cisco
3620 with 3x WIC-1DSU-T1, 1x 2FE-2W, and 1x 1E-2W. I have 2 T1 lines
being brought in by ACD.NET, a
What have other admins done to retrieve detailed call information about
the queue system? Anyone develop their own that they don't mind
sharing?
You can try this perl script it was useful for me. After parsing I do
reports based on generated queue_statistic.csv in Excel...
cut
--- [EMAIL PROTECTED] wrote:
I'm trying to setup Asterisk trunk to Siemens HiPath
4000 V2.01
i suppose you mean version 2.0 ;-)
What would be the best way to do so? I am a bit
confused because as far
as I've understand this PBX doesn't support H323,
but I saw somewhere
someone who
Just got a net4501 board, installed cf card/Monowall. Does anyone have a
monowall firewall with Asterisk behind it, any problems, can external SIP
phones work?
What firewall rules are you using?
Chris Mason
Int: (305) 704-7249 Fax: (815)301-9759
___
Ivan Meic (Vox Mundi) wrote:
Actually G.729A is a reduced complexity version, and G.729B is a version
with silence suppression. The data rate while sending voice is exactly
the same, although the quality of G.729B should be a little higher.
However the average rate for B can be lower if the
On Wednesday 25 May 2005 11:27, Ronald Wiplinger wrote:
Bob Goddard wrote:
On Monday 25 April 2005 08:56, Bharat M. Sarvan wrote:
Hello Everybody,
I was going thru the C code of Asterisk. Does
anybody know how does one go about modifying the C code of Asterisk?
Matthew Boehm wrote:
We have no need for vlans in our office
Then why not disable vlan altogether? Press the settings button while
the phone is booting up, go to Network Settings and disable vlan. (in
the Network Settings menu you might need to press **# to unlock the
phone. I have 7912's
Hi Zen,
See http://www.soft-switch.org/foip.html
Steve
Zen Kato wrote:
After I did rxfax(...|debug), I got the followings;
..(snip)...
DIS: 80 00 ce f4 80 80 81 80 80 80 18
HDLC underflow in state 9
Changed from phase 4 to 3
T4 timeout in state 9
Changed from phase 3
On Wednesday 25 May 2005 10:15, Asterisk User wrote:
Hi all,
I have problem with my Asterisk.
I'm using the softphone Xten-Lite. I've removed the SIP client information
in sip.conf. The softphone can't register to Asterisk, but it can make
outgoing calls.
I've tried to add back the SIP
-Original Message-
From: Jon Farmer [mailto:[EMAIL PROTECTED]
Sent: 25 May 2005 11:14
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] PHP/AGI Problem
--- Thierry Wehr [EMAIL PROTECTED] wrote:
Hello
Did you
Hey all,
I have read on voip-info.org that to configure MoH asterisk requires the
use of mpg123. I have installed mpg123 and restarted asterisk. But,
when i put a call on hold i get this error:
May 25 14:13:03 WARNING[1872]: res_musiconhold.c:865
local_ast_moh_start: No class: default
You may also want to check the following link
http://www.voip-info.org/wiki-MSN%20PHP. This is work in progress, but I
think it may help you. it is based on IM messaging protocol to/from MSN
Messenger. I don't believe there is a redirect to hard phones, but I think
that could be part of command
Steve, what would help a bunch of people trying to implement your
spandsp is some kind of help document that at least attempts to
describe some of the debug statements shown below. When the average
person reads hdlc underflow or T4 timeout in state 9, we don't
have a clue what those statements
Can you post your conf file for the musiconhold??? Sounds like you haven't
defined a default class / context - I could be wrong
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of yusuf
Sent: Wednesday, May 25, 2005 8:32 AM
To:
Being in Canada it makes it very difficult to find companies that will
ship COD from the US. If I was to order I would only order COD from
now on from VoipSupply.
We are in Canada; we are authorised Digium, Grandstream, Sipura, and
Snom resellers; and we ship COD (within Canada).
I have
A quick hack would be to do something like
exten = h, 1, system(kill the php)
of course this would knock out all the calls in progress using php.
Perhaps in the hangup extension you could send a message to your php
script that would end it's blocking. I don't suppose SendDTMF would
send to the
Anton Krall said:
I think I once read something about creating a peer on sip.conf that
should be Guest in order to allow any server to connect without a
password to yours and go to the specified context.. Am I right?
Sorry Anton, I have no idea. Let me know if you do figure out how to have a
Hi there
I have been using Asterisk Meetme with Ztdummy for timing.
It seems to work fine and I havent had any major problems. I am now
moving into a production environment and am wondering if it is better to use a
Zaptel card? Are there any problems with Ztdummy? I will probably have
--- Alex Barnes [EMAIL PROTECTED] wrote:
Although this isnt a substitute for a correctly
terminating script,
I would have thought that the PHP 'maximum script
execution time'
variable would kick-in
and kill the script eventually.
Well I have already tried that I have the first line
of
Hi there
I am using Meetme. Now, I know it is possible to mute a user
in a conference, but is it possible to stop a user receiving audio at a
specific time (basically when they speak) and to do this through the Manager
API.
Looking at the Asterisk wiki it seems there might be some
--- yusuf [EMAIL PROTECTED] wrote:
Hey all,
I have read on voip-info.org that to configure MoH
asterisk requires the
use of mpg123. I have installed mpg123 and
restarted asterisk. But,
when i put a call on hold i get this error:
May 25 14:13:03 WARNING[1872]: res_musiconhold.c:865
On May 25, 2005 09:46 am, Rich Adamson wrote:
Steve, what would help a bunch of people trying to implement your
spandsp is some kind of help document that at least attempts to
describe some of the debug statements shown below. When the average
person reads hdlc underflow or T4 timeout in state
no man iax2 trunking not working i don't know why its really odd
iax2 trunk debug command shows
IAX2 Trunk Debug Requested
Beginning trunk processing
Ending trunk processing with 0 peers and 0 calls processed
wat's that means how can i enable trunking on one ser iax2 show
channels command shows:
Andrew Kohlsmith wrote:
On May 25, 2005 09:46 am, Rich Adamson wrote:
Steve, what would help a bunch of people trying to implement your
spandsp is some kind of help document that at least attempts to
describe some of the debug statements shown below. When the average
person reads hdlc
Rich Adamson wrote:
Steve, what would help a bunch of people trying to implement your
spandsp is some kind of help document that at least attempts to
describe some of the debug statements shown below. When the average
person reads hdlc underflow or T4 timeout in state 9, we don't
have a clue
i know there is example in extension.conf
but that is not working in my case
i am unable to get the extension pressed by user after
listening menu
like how to get when 2000 pressed.
because it is not dialing 2000
exten = 6000,1,Background(k-enterexten)
exten = 6000,2,Wait(2)
Im doing something similar using centericq for testing.. Works well but
sometimes the message arrives to the user too late..
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Alex Vishnev
|Sent: Miércoles, 25 de Mayo de 2005 07:40 a.m.
|To: 'Asterisk
No luck so far but I still think you need to define [guest] or something on
sip.conf
For example, Steven Sokol has its own extension configured into every copy
of iaxphone... How does iaxphone connect to his asterisk with the need of a
user? Or does I have a user hardcoded into the app and
Hi All,
Now I configured a linux box as a router. And I
installed Asterisk on it.
My problem is whenever the WAN is offline all the
sip extensions will logon failed. My sip extensions
are connected to Asterisk through LAN. Why the LAN
side sip phones cannot logon when WAN is offline.
Steve, what would help a bunch of people trying to implement your
spandsp is some kind of help document that at least attempts to
describe some of the debug statements shown below. When the average
person reads hdlc underflow or T4 timeout in state 9, we don't
have a clue what those
For example, Steven Sokol has its own extension configured into every copy
of iaxphone... How does iaxphone connect to his asterisk with the need of
a user? Or does I have a user hardcoded into the app and multiple people
can use that username?
I would think he has a default context in
Use your favorite text editor. That is how you modify the code. Personally,
I like pico.
-Matthew
Bharat M. Sarvan wrote:
Hello Everybody,
I was going thru the C code of Asterisk.
Does anybody know how does one go about modifying the C code of
Asterisk? Please do
Check if the SIP.conf is configured to bind into the ip address of LAN or
not? You have to define the internal address there, not the WAN ip address.
RG,
Gentian
- Original Message -
From: lanfei chen [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Rich Adamson wrote:
Steve, what would help a bunch of people trying to implement your
spandsp is some kind of help document that at least attempts to
describe some of the debug statements shown below. When the average
person reads hdlc underflow or T4 timeout in state 9, we don't
have a clue
I am calling the PHP app via deadagi. I believe what
Well, that is your problem. Don't use deadagi. DeadAGI is for use if you
want to continue processing after the call hangs up. That is why your
scripts are continuing to run. Use regular AGI.
-Matthew
-Original Message-
From: Jon Farmer [mailto:[EMAIL PROTECTED]
Sent: 25 May 2005 14:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] PHP/AGI Problem
I have also tested detecting the channel_status and
that doesn't seem to work either.
I am also very interested in CRM integration. Anything I can do to help?
One thing I don't understand is how is the browser being launched on the
person's PC. Or is it not launched automatically?
Anyone know of a simple app running on the desktop to do this? I looked
into IPSwithcBoard and it
Try changing wait(2) to background(silence/9)
See if that helps, as you can't accept digits during a wait.
Also, check out the the waitexten command.
Hope this helps,
Jon.
On Wednesday 25 May 2005 05:14 am, Kamran Ahmad wrote:
hello
like if 6000 is the main exchange number. any one dial
The 7912's aren't SIP compatible are they? I just scrolled thru the network
settings menu and there is nothing in there about disabling VLAN. The Admin
VLAN is set to blank but it still searches.
-Matthew
Mark wrote:
Matthew Boehm wrote:
We have no need for vlans in our office
Then why not
Hi all,
I just want to inform you that we have opened up a Swedish, no-charge,
independent user forum at http://www.voip-forum.se/
We do discuss things such as Asterisk, Skype, hardware and various
configurations.
(Of course, we welcome all people from Scandinavia.)
Welcome!
Regards,
Trunking works for me. I'm not sure what the problem is but can have you try
different things till we find it.
Notransfer=yes doesn't work for me. Calls still transfer.
Try putting trunk=yes in EVERY user.
Also I don't use type=friend. Try setting up a seperate user and peer
context.
Well, that is your problem. Don't use deadagi.
DeadAGI is for use if you
want to continue processing after the call hangs
up. That is why your
scripts are continuing to run. Use regular AGI.
I get the same behaviour if I use deadagi or just agi
Regards
Jon
As one of the members said. The idea behind this is not to have to install
any app on the computer but base everything on web apps. Im still trying to
find out how to mix everything together but its a mixture of Asterisk
Manager usage and some PHP coding for example.
You can probably just open a
Has anyone had problems with a small electrical type of hum on the 841's
handset. It is there on all of the three phones I bought, and also do the
sound like the microphone is cheap and kind of a high pitched talking into a
can. I can live with these as long as I know that this is what the phones
Wow I found a fellow pico user... I'm constantly receiving ridicule for my
use of pico... I cannot stand vi... If I don't have pico sometimes the ol'
emacs...
But vi is garbage.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent:
Sounds good to me!!
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Nabeel Jafferali
|Sent: Miércoles, 25 de Mayo de 2005 09:05 a.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Guest
|
| For
-Original Message-
From: Steve Underwood [mailto:[EMAIL PROTECTED]
Sent: 25 May 2005 15:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] rxfax(spandsp-0.0.2pre18) and HT488
What is the page supposed to say? Something like:
IF YOU TRY
Why are you using DeadAGI? Use AGI or EAGI instead, unless you actually
want to run on a dead-channel.
-Original Message-
From: Jon Farmer [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 25, 2005 8:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
Modifying the C files of asterisk is very different from modifying the
config files. If you are not currently a programmer, this will be a very
long and slow process. First, a good understanding of the C language
programming statements and language syntax is needed. Next, you will need
to
Hello,
We use astGUIclient, it does have server side apps that have to be installed
on your Asterisk server, but it does have callerID popups that allow you to
search a customizable web page when a call comes in. We are also releasing a
new version of the astGUIclient app next week that is
David,
this is my config via DHCP:
67: Startupserverwebserver.mydomain.com
66: Startup filesnomstartup.cfg
File snomstartup.cfg
setting_server:
http://webserver.mydomain.com/snom/conf/snomcfg.php?MAC={mac}
subscribe_config: on
File snomcfg.php
?php
$filename =
| -Original Message-
| From: [EMAIL PROTECTED]
| [mailto:[EMAIL PROTECTED] On Behalf Of
| Michiel van Baak
| Sent: Martes, 24 de Mayo de 2005 04:45 p.m.
| To: asterisk-users@lists.digium.com
| Subject: Re: [Asterisk-Users] CallerID
|
| On 23:44, Tue 24 May 05, Anton Krall
Joel Duffield wrote:
Has anyone had problems with a small electrical type of hum on the 841's
handset. It is there on all of the three phones I bought,
Yes, though I would not call it hum, somewhat more like a small amount
of data hash.
Other than that, the sound quality of the phone through
Jon Farmer wrote:
Well, that is your problem. Don't use deadagi.
DeadAGI is for use if you
want to continue processing after the call hangs
up. That is why your
scripts are continuing to run. Use regular AGI.
I get the same behaviour if I use deadagi or just agi
Regards
Well then you
The method we use for web popups on incoming calls in the astGUIclient
client app that we are working on for release next week is to use
AJAX(Javascript + XMLHTTPRequest) It works in Firefox and IE5+ and doesn't
require any META refreshes. We've been using this internally for the last
month and it
You have been replied to - we do not use digital certs, we do not
reply when you have some sort of Spam blocker. This time I am
responding even though that is not policy.
It seems it is their policy not to answer.
FYI info I tried to get an account with them a week ago. I did not get
any
Anton Krall a écrit :
No luck so far but I still think you need to define [guest] or something on
sip.conf
sip.conf
[101]
type=user
username=Guest
insecure=very
host=dynamic
permit=0.0.0.0/0.0.0.0
context=fromsipguest
callerid=SIP Guest 101
nat=yes
canreinvite=no
allow=all
Extensions.conf
I feel I need to stand behind voipsupply.com here as well.We have
ordered several ATAs as we are starting up our VoIP business, and just
recently became a reseller with them.. so far we've had nothing but
success with them and love them.
___
You're lucky you didn't let the smoke out of your card - some HDSL units
in the USA have some serious voltage/current on the pair that goes into the
telco side to power the unit.
Glad to hear it turned out okay.
Tim
- Original Message -
From: Remco Barende [EMAIL PROTECTED]
To:
Steve, what would help a bunch of people trying to implement your
spandsp is some kind of help document that at least attempts to
describe some of the debug statements shown below. When the average
person reads hdlc underflow or T4 timeout in state 9, we don't
have a clue what those
Yes I do. Works fine. It's important to let Monowall create the forwarding
rules for you after you create the NAT entries. If you create it manually,
it is hit-and-miss. My config is:
NAT:
WAN UDP FROM: 4569 NAT IP: ASTERISK IP LOCAL PORT 4569 (IAX)
WAN UDP FROM: 5060 NAT IP: ASTERISK
On May 25, 2005 10:02 am, Steve Underwood wrote:
So let me get his clear. If I don't document things I am in the wrong,
whereas if I do document them I am in the wrong. Is that it?
No no; I have to admit that I took the foip page as a background on faxing and
that leads into why faxing over
could you post the script, the output of the script in the asterisk
console and which asterisk version are you working with?
On 5/25/05, Jon Farmer [EMAIL PROTECTED] wrote:
--- Alex Barnes [EMAIL PROTECTED] wrote:
Although this isnt a substitute for a correctly
terminating script,
I
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