On Tue, 19 Oct 2004 10:34:02 +1000, Adam Goryachev
[EMAIL PROTECTED] wrote:
Just a heads-up that asterisk is getting a mention in spam now... oh,
and make sure you NEVER EVER buy anything from this company.
[SNIP]
NEWS: VocalScape Inc. Announces DELETED for Asterisk IP PBX Users.
As
Hi, List
I have used Windows Messenger for video call via Asterisk Server.
But Windows Messenger function can't match our requirement.
We are looking more SIP Video Phone can use under Asterisk.
Any suggestion for video Phone(Software or Hardware)?
Also I still have an question about video/audio
Hi,
On Monday 11 October 2004 19:12, Dave Cotton wrote:
On Mon, 2004-10-11 at 11:51 -0500, Mike Meyer wrote:
Someone pointed me here
http://www.snom.com/downloads/share
http://www.snom.com/download/share
!
That where the SNOM support team sent me. Seems that they may be
Hi,
no, it is
http://www.snom.com/download/share
!!!
Sven
On Monday 11 October 2004 17:18, Alex Barnes wrote:
Someone pointed me here
http://www.snom.com/downloads/share (had to guess at URL as the Snom
site appears to be down or uber slow but if that's not it its damn close
:-P )
Hi,
do a SIP trace or PCAP trace of the scenario via the webinterface and you will
see exactly, what is going on...
Regards,
Sven
On Thursday 14 October 2004 21:53, Magnus Jungsbluth wrote:
Hi,
I'm running the 1.0 release of Asterisk an it is working nicely with our
snom 105 phones. Hold
This tends to be a religious issue. I guess I am an older admin. :)
I come from the school of thought that it is a good idea to
reboot a server that is not meant to be used interactivly (console or
terminal) on a schedule. Most software does not require it. In my
experience, the systems
Fabian Garcia wrote:
I understand asterisk invokes sendmail in order to send email
notifications of messages left. Is there another application less
complicated than Sendmail, I already got mail servers else where
and they are the ones I want to use.
All major MTAs emulate the sendmail
Benjamin on Asterisk Mailing Lists wrote:
When I connect to the Sipura to dial out on the PSTN line connected to
the Sipura's FXO port, it gives me the dialtone of the PSTN line and
then I can hear the DTMF for the number I dialled beforehand.
It does work but the customer perceives this delayed
Magnus:
I would like to decide using the callerid which music on hold is tobe
played: That is: play free music to calls from the outside
but play copyrighted music if I put an internal call on hold
(i.e. a co-worker).
Is this possible ?
Yes, and it's easier than intercepting the hold
Kristian Kielhofner wrote:
1) There is a lot of code in the dump from /admin/advanced.
Note that if you're interested in only changing a few parameters, you
need not post everything.
2) The password is all *'s (not good to PUT it back like that).
See previous point.
-- PhoneBoy
On Tue, 19 Oct 2004, David H Hickman wrote:
This tends to be a religious issue. I guess I am an older admin. :)
I come from the school of thought that it is a good idea to
reboot a server that is not meant to be used interactivly (console or
terminal) on a schedule. Most software does
Hello List,,
I have a customer that has a broken voicemailof a nortel option 11 ,,
can we offer something to replace with Asterisk? anyone there that all
ready implement something as this , please contact me because I'll
need service to setup one.
right now they have 8 digital ports for the
Hi,
I have Digium TDM400P. I have succesfully installed and got the demo.
However I have a problem with DIAL command. I have 2 FXS port (Zap/3 and
Zap/4).
Both of them are connected a inner telephone line.(100-Zap/3 and 101-Zap/4)
When i call 100 with phone 102, 101 redirects 102 to 103.
So i use
Hi !
I have been working on making my asterisk server work with Vonage
services. I have been able to recieve calls on my asterisk machine but i
couldnt call through that account to other people. Means if i call a zap
channel and then dial 1 314 652 ... then i get an error like
Executing
Adam
On UK keyboards ,I have to type a £ to get a # into Firefly. The proper
# key does nothing. If you are updating the code, perhaps you might look
at this?
Many thanks
Peter
-Original Message-
From: Adam Hart [mailto:[EMAIL PROTECTED]
Sent: 16 October 2004 07:46
To: Asterisk Users
Hello *-users
I'm trying to implement a simple follow me solution. The case is that I
would like to be able to pickup the incoming call on a line (whatever)
hang it up and repick it on another line (mobile)
Currently i'm using the following to accomplish this:
exten = 31xxx,1,Wait(1)
exten
Title: Message
Hiall,
I am hoping
that someone out there is using the Snom phones "configuration via HTTP server"
functionality.
I have
downloaded and read the FAQ many times but I am having trouble getting the
settings to take effect. Probably as I haven't formatted things
correctly. For
On Mon, 2004-10-18 at 21:11, Fabian Garcia wrote:
I understand asterisk invokes sendmail in order to send email
notifications of messages left. Is there another application less
complicated than Sendmail, I already got mail servers else where and
they are the ones I want to use.
How about simply doing a blind transfer to your cellphone (or other
phone...)? You could setup a special extension, say extension *1, to
dial your cellphone so you don't have to dial the whole number every time.
Pascal C. Kocher wrote:
Hello *-users
I'm trying to implement a simple follow me
Good day all
How do I record a call on a vpb channel?
Thanks
Altus
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To UNSUBSCRIBE or update options visit:
I usually use Qmail www.qmail.org, in my humble opinion it is more straight forward to
configure than sendmail.
On Mon, 2004-10-18 at 21:11, Fabian Garcia wrote:
I understand asterisk invokes sendmail in order to send email
notifications of messages left. Is there another application
Try googling for 'linux mail server how to'
All you really need is a simple setup (sometimes called a 'smarthost' if
I recall correctly) that forwards mail to another smtp server.
Take a look at EXIM and Postfix. IMHO they are both much easier to setup
then sendmail.
Fabian Garcia wrote:
Hi,
Hello Brian
How about simply doing a blind transfer to your cellphone (or other
phone...)? You could setup a special extension, say extension *1, to
dial your cellphone so you don't have to dial the whole
number every time.
Thank you for the reply, the log extension is really a DDI, not
Hi,
I have a Grandstream Budge Tone 100 and i wanted to use the supervised call
transfert feature but i don't find any tips for that. So there is my
question : Is this feature is implemented on GS BT100 and if it is not, it
is possible to implement it directly on Asterisk. Juts for your
Hi,
I haven't worked with Vonage myself but I usually get
this error back from my termination provider when the
number I have sent them is incorrect.
It might be worth checking you have used the correct
prefix (011 or 00) and area code etc.
Regards,
Aaron
Hi !
I have been working on
Hi all,
was just wondering if there were any
special things I had to do to set the outgoing caller ID on a UK BRI (EUROISDN)
line. I've got a line in my extensions.conf which says:
exten = _9.,1,SetCallerID(3317**)
This is then followed by the dial command.
So I dial 9 followed by my mobile
On October 18, 2004 09:11 pm, Fabian Garcia wrote:
I understand asterisk invokes sendmail in order to send email notifications
of messages left. Is there another application less complicated than
Sendmail, I already got mail servers else where and they are the ones I
want to use.
Nullmailer;
Benjamin:
So I dial 9 followed by my mobile number and the call comes
through fine but the display says No Caller ID.
Assuming that your mobile is displaying caller IDs for other numbers and
your ISDN lines are with BT.
There are two ways in which the number can be withheld:
1 -
On October 19, 2004 05:48 am, james wrote:
There are several replacements, but sendmail isn't any harder to config.
You usually only need to change 3 lines in the sendmail config.
I suppose the reasons people are so anti-sendmail are several:
1. Security. Sendmail has a track record of being
On Tue, 19 Oct 2004 [EMAIL PROTECTED] wrote:
I'm at work now and don't have my access to my asterisk box (which isn't
much use as I can't post debug data or other lines from the config files).
Just wondered if anyone had done this and where I was going wrong (I have
tried different number
Hello all,
I'm still having problems with the IVR options.
When I press on my mobile phone one of the digits related in the IVR
options, press 1 for .,press 2 for.., press 3 for..
After I press the one, the second or the tirth key on my mobile phone, I
can't hear nothing more, I can't
I also have a Samsung i700 phone, and their newer i600 (Mobile Windows
2003), both through Verizon. With both phones you can purchase
unlimited Internet access for $79 per month. Both phones are able to
browse the Internet, send and receive email, and use the roll-up
keyboard. As such, I'm
Hi Jacky
Try using Eye Beam from X-Ten for vidio with Asterisk.
www.Xten.com
Doug Reid
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jacky
Sent: Tuesday, October 19, 2004 8:20 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP video support problem
Hi
We use the Grandstream range, the work very well with Asterisk
although the run at 10BASET so best to keep them on a separate
network. They have all the functionality and work very well, not
the best looking phone but you get what you pay for!
Doug Reid
-Original Message-
From:
There is currently no such feature on the BT100 although someone did post
two weeks or so ago that firmware 1.0.5.12 would have it. As yet, there is
no hint of this new firmware. Alternately I think there is a patch around
somewhere to do it within Asterisk, play detective and see if you can
I am experiencing a problem with the RECORD FILE functionality in AGI when
I am doing a Record_file.
After approx 20 mins + the Record_file ceases to accept escape digits and
therefore records for ever or until my timeout I set. It acts like a dead
application, just recording without the
Hi Nick,
yep, my mobile displays caller id for
other numbers - and it even works perfectly displaying caller id information
set by a cheap ISDN pbx on the *same* ISDN line as the Asterisk box. Curious.
Even without setting a callerid on the outgoing calls I get No Caller
ID on my mobile (or
Yeah, thats what I figured, BUT, if you transfer an incoming call to
another internal user, music on hold switches to INTERNAL, and if the
2nd agent does a another transfer, the incoming call gets INTERNAL
music. I search for a way to define somewhere in extensions.conf a
extension that is
Laugh. I use a bare-bones install of QMail on my main asterisk server. It of
course emulates sendmail and the like.
But on my remote Asterisk server, I use ssmtp, it came as a prerequisite to
Asterisk. When I emerged asterisk, ssmtp came with it. Works great.
Configured it to use my main Asterisk
For the SIP client. I just can't imagine using a SIP client over a
connection that has 250+ ms response times. If I make it go online to the
802.11 networks, I can use the SIP Client with ULaw and get high quality SIP
calls at any hotspot. I wouldn't do this for every hot spot, but it'd be a
nice
Does anyone know if the sip firmware on the 79xx phones would support *
pushing the called name back to the calling phone?
Maybe using the SIP Info method?
--
respectfully, Joseph ===
-= ** =
___
I didn't read this whole discussion but I've used the G729A codec
using X-Ten Pro on my laptop while connected to the 1xRTT network to
my Verizon phone via Bluetooth and it worked rather well. (as long as
I had a full signal on my cell phone). It's not something you can
call stable, or probably
Hi All,
Does anyone here use the Speex codec on their * server? I see that
Voicepulse Connect supports Speex and I'd like to try using it in
wideband mode. I'm wondering if it might be a suitable alternative to
GSM at comparable data rates. Any idea how I setup the codec for
wideband mode?
Did someone have experience with:
-
Chan_modem
-
Chan_capi
-
Chan_misdn
What is the best???
_
Erwan
Desvergnes - ANDIUM -
82/86 rue Château Gaillard
69100 Villeurbanne
Tel. 04 3743 44 45
/ Fax 04 37 43 44 44
E-mail:
I have my brother-n-law in Australia who just purchased a SPA-3k.
He is wanting to connect to my * server.
For the * entry I have the following:
sip.conf:
[2203] ; Dustintype=friendhost=dynamiccontext=defaultsecret=supersecretpasscodemaxexpirey=1800defaultexpirey=1600callerid="Dustin-Debbie"
Title: Message
Can you enable "sip debug ip 'HisIPAddressinAU'"
And copy out the REGISTER message and responses.
Might help narrow down what the problem is.
-Original Message-From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: 19
October 2004 13:51To:
[EMAIL
I've just used chan_capi it's very easy to use with Fritz!Cards and
therefore I like it ;-)
Did someone have experience with:
- Chan_modem
- Chan_capi
- Chan_misdn
What is the best???
Ruben Fagundo wrote:
I have an asterisk server running on Redhat 8.0 with a Digium TDM400P
w/4 FXO modules (TDM04P)
There are 2 lines going into the Digium card. One line is a Vonage
digital line, and the other line is a Comcast voice line. I have a SIP
Grandstream 100 phone connected to the
On Tue, 2004-10-19 at 11:54 +0200, Altus Syman wrote:
Good day all
How do I record a call on a vpb channel?
Part of the point behind the way asterisk is built is that at the
application point of view, the channel is mostly irrelavent. Therefore
you record by using record.
--
Steven Critchfield
You also are having a problem realizing that we have now seen your
message SEVERAL times and shoved into other threads that are irrelavent
to recording or AGI. You are not helping yourself by doing this.
--
Steven Critchfield [EMAIL PROTECTED]
___
On Tue, 19 Oct 2004 12:50:41 +, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
UDP pointing to my NAT'd * server.
He is also NAT on his side
Are you saying this is NAT on both ends? aka double NAT?
If so, use tunneling. Double NAT is a bitch.
rgds
benjk
--
Sunrise Telephone Systems,
I've come up with a temporary solution to locate the softkeys where I
want them...
set up the following key:
KEY blank IS OR Blank
GOTO offhook
ENDKEY
then you can do
SHOWKEYS park
SHOWKEYS blank
SHOWKEYS xfer
Hi,
I'm sure a lot people can help you here, maybe I'm the first. See below
inline:
On Tuesday 19 October 2004 11:20, Alex Barnes wrote:
Hi all,
I am hoping that someone out there is using the Snom phones
configuration via HTTP server functionality.
I have downloaded and read the FAQ many
Have you got any problem with sound on the 2nde chanel ???
-Message d'origine-
De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Envoyé : mardi 19 octobre 2004 14:59
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] mISDN, CAPI, ISDN ???
I've just
Definitely choose chan_capi.
Chan_modem is almost deprecated, bad quality and very few features.
Chan_misdn seems to be a very good project but it is still young.
Zaphfc in theory it's wonderful (zap echo cancellation, timing etc.) but
you have to use older * versions, (till new kapejod's
Has anyone tweaked the txgain values on an T100P
card(hooked up to PRI)with success?
People complain about loudness.
Thanks,
Don Dawson.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
I am using a Sonicwall 3060. The SonicWall has 6 hardware interfaces. My
asterisk box is on one interface configured as a DMZ. It still goes through
NAT, but is exposed as a public ip of x.x.x.56, and private IP 192.168.3.2.
The public ip of the firewall is x.x.x.50.
I am using the connect
Hi Doug,
Are you using eyeBeam with Asterisk? I posted in another message to this
group this text:
---
Hi everyone,
Is anyone using Xten eyeBeam Video softphone with Asterisk? It supports
few types of H.263 codecs for video. I have tried to use it with
If you have a solidly re-produceable bug, suggest that you go to
http://bugs.digium.com/login_page.php
Sign up, and post the bug.
Regards,
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original Message-
From: [EMAIL
Oh ok, so there are other threads with Recording...Where are they?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Tuesday, 19 October 2004 11:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
On Tue, 2004-10-19 at 14:58 +0200, [EMAIL PROTECTED] wrote:
I've just used chan_capi it's very easy to use with Fritz!Cards and
therefore I like it ;-)
Worked straight out of the box on an AVM C2, hope it does the same with
2 Fritz!Cards in the same machine.
--
Dave Cotton [EMAIL PROTECTED]
You are mixing oranges and apples here i guess.
G729is a MediaTransmission Protocol Codec the other is a Compressed
AudioFile format.
There are no .g729 audio files as far as I
know.
Seshu Kanuri
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Victor
CartesSent: Monday,
Hi Guys,
i need to do some kind of CDR for all clients inside my network, but they do
not register/use the same
sip-server, some of them use iptel, others fwd and various other services.
Can i somehow put asterisk in the (control-)path between my clients and the
other services
(iptable-redirect
SER most definitely does CDR archiving via MySql database. It's a
hellaciously fast and stable proxy - sounds like it'd be a good choice
for the core of your network with all the different components.
On Oct 19, 2004, at 10:01 AM, Andreas Anderson wrote:
Hi Guys,
i need to do some kind of CDR
On Tue, 19 Oct 2004 09:44:12 -0400, Gene Willingham
[EMAIL PROTECTED] wrote:
What I think is happening is: If I receive an inbound call on IAX during an
IAX registration, the call does not get setup. I appear to be unavailable
to the other server. When a call fails I noticed using tcpdump
The SIP client in Windows Messenger 5.0 seems to work fine with Asterisk
though.
Peter
-Original Message-
From: Robert Rozman [mailto:[EMAIL PROTECTED]
Sent: 11 October 2004 22:08
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Dave Cotton wrote:
On Tue, 2004-10-19 at 14:58 +0200, [EMAIL PROTECTED] wrote:
I've just used chan_capi it's very easy to use with Fritz!Cards and
therefore I like it ;-)
Worked straight out of the box on an AVM C2, hope it does the same with
2 Fritz!Cards in the same machine.
Sadly
I would like to find a way to list active extensions
with either the manager api or an agi script.
Using ExtensionState in the manager api I can't seem to get the syntax
right.
I tried the show channels with the exec command and it did not seem to
work.
And I tried Channel Status with the agi
I don't know for the C2 but for the USB one it doesn't.
AVM says it's normal.
-Message d'origine-
De : Dave Cotton [mailto:[EMAIL PROTECTED]
Envoyé : mardi 19 octobre 2004 15:53
À : Asterisk List
Objet : Re: [Asterisk-Users] mISDN, CAPI, ISDN ???
On Tue, 2004-10-19 at 14:58 +0200,
Looks like you're dialing on a zap channel, no?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 19, 2004 6:15 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Working Asterisk With Vonage
Hi !
I have been working on making my
Seem It doesn't work for the USB one. And for the pci one, the current
drivers it's not then same.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Massimo De
Nadal
Envoyé : mardi 19 octobre 2004 16:24
À : Asterisk Users Mailing List - Non-Commercial
only avm active cards permit multiple installation straight forward (b1,
c2 and c4)
with fritz! pci you can do the hack mentioned above, for fritz! usb you
can't install more then one.
This limitiation is due to avm drivers design, they choose to allow
multiple installation only on hi-end
hi all
we're setting up a rather large end-user VoIP system, and due to
pressure from norwegian telephony authorities, we consider choosing
something instead of G.711A, possibly G.729.
does anyone know if it is possible to test-drive G.729 without paying
Digium for it?
roy
yes, It's not the same,
but applying the same hack to newer drivers it's not so difficult,
almost for pci fritz!
Erwan Desvergnes wrote:
Seem It doesn't work for the USB one. And for the pci one, the current
drivers it's not then same.
-Message d'origine-
De : [EMAIL PROTECTED]
I have been using this testing various fax machines. I have one source
that always works, Jfax. When I send my self a fax, it always gets
converted and sent.
Today I was passing the basement sysconsole and saw a bunch of debug
stuff slide by for a fax that seems to give trouble. I know l'm
I personally think for a codec that's almost 1/3 the size of ULaw, the
quality is great. I consider ULaw above telephone quality, and g729 to be at
telephone quality.
But just 5 minutes ago I moved a user over to g729a. Changed the
SIP000.cnf file for the Cisco phone, but forgot to change the
On October 19, 2004 10:47 am, Roy Sigurd Karlsbakk wrote:
does anyone know if it is possible to test-drive G.729 without paying
Digium for it?
You're too cheap to blow $20 (I assume you need 2 licenses) on a test?
Seriously it's not that expensive and if it doesn't work it doesn't work -- I
Anyone here know about NFAS trunkgroups?
I have a TE405P card with spans 1, 3 and 4 connected to T1s. I have
happily had NI2 PRI running on them with each trunk having its own
D-channel. Using v1-0 from CVS.
/etc/zaptel.conf has the following:
span=1,1,0,esf,b8zs
span=2,0,0,esf,b8zs
Deon Rodden wrote:
I personally think for a codec that's almost 1/3 the size of ULaw, the
quality is great. I consider ULaw above telephone quality, and g729 to be at
telephone quality.
uLaw *is* telephone quality. Its what the PSTN uses. G.729 is much
inferior, but at 1/8th the size (not
/SNIP/
Have you looked into that open-source implementation of G729? There
was something on the WIKI about 3 different implementations of it. One
being where you paid license per channel fees, one that was
free/open source, and another I can't remember. Check the WIKI.
/SNIP/
There is no such
I wrote:
Anyone here know about NFAS trunkgroups?
Just a little more info (please see original message for main details):
I have changed /etc/asterisk/zapata.conf as follows:
[trunkgroups]
trunkgroup = 1,24,72
spanmap = 1,1,1
spanmap = 3,1,3
[channels]
switchtype = national
signalling
G'Day All;
Greetings and best wishes. I need some help as follows:
My Grandstream 100 is at a remote location on broadband and connects to
my * server else where.
From a POST line I dial the 3 to the * server and selects the ext # of
the remote GS100 IP phone.
The GS100 rings. When answered I
States right here:
http://www.voip-info.org/tiki-index.php?page=Convert%20WAV%20audio%20files%20for%20use%20in%20Asterisk
Asterisk can play anything it has a format and codec for. Including wav,
gsm, g729, g726, wav49 all of which can be used for Playback and Background.
So, how can you make
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ferguson, Michael
Sent: 19 October 2004 16:18
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Almost there--Remote connection
[snip]
The * server is behind a firewall and I have opened ports
Record them from a phone that speaks g729 right to raw .g729 files.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent: Tuesday, October 19, 2004 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial
The 1-10100 was given to me by a prior post so I really do not know.
I will change the forewall to allow 1-2 and see if it works.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl Dyson
Sent: Tuesday, October 19, 2004 11:22 AM
To: Asterisk
My firewall script has something to the effect of:
# Allow Existing traffic through
-A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT
# Incoming VOIP Ports
-A INPUT -m state --state NEW -m tcp -p tcp --dport 5036:5045 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 5036:5045
Hi,
We try to send Fax through IP Network but without success. The
other party use NetCentrex SoftSwitch and our communication protocol
between us is H.323 (OpenH323). The error that the other party receive
is: bearer capability not imoplemented.
Is it possible to send Fax using
On October 19, 2004 11:01 am, Tony Mountifield wrote:
Anyone here know about NFAS trunkgroups?
Yes, I worked with them in the dialup world on AS5248s and MaxTNTs.
I have a TE405P card with spans 1, 3 and 4 connected to T1s. I have
happily had NI2 PRI running on them with each trunk having its
yep, my mobile displays caller id for other numbers - and it even works
perfectly displaying caller id information set by a cheap ISDN pbx on the
*same* ISDN line as the Asterisk box. Curious. Even without setting a
callerid on the outgoing calls I get No Caller ID on my mobile (or other
phones -
On Tue, 2004-10-19 at 23:14 +1000, Simon Smith wrote:
Oh ok, so there are other threads with Recording...Where are they?
You put this message into a thread about AGI Get Data', You put this in
a thread about video door phones. You put it in a thread in -dev with
subject line of Unusual problem.
Thanks. I think that's Iptables. No?
I have a hardware firewall.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Deon
Rodden
Sent: Tuesday, October 19, 2004 11:35 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE:
We just put in for a demo of one of these today. Do you know if it does
any sort of QOS or traffic shapping. The specs don't seem to mention it.
Scott Wolf
Ben Merrills wrote:
Just thought I
would let the
list know, as we got our pre release versions today of the new
Hi ,
I tried this a lot, but with no sucess , even in a local network , there
is always some loss and you receive only chunks of the original file .
Pedro.
Miroslav Nachev wrote:
Hi,
We try to send Fax through IP Network but without success. The
other party use NetCentrex SoftSwitch and our
Did someone have succeed to compile
chan_misdn???
Ive got an error when in try to compile
chan_misdn.c:68: error:
`__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared
here (not in a function)
thanks
_
In article [EMAIL PROTECTED],
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On October 19, 2004 11:01 am, Tony Mountifield wrote:
Anyone here know about NFAS trunkgroups?
Yes, I worked with them in the dialup world on AS5248s and MaxTNTs.
OK, I was too vague! What I really want is someone who
Hi Marconi,
I couldn't access URL.
I want to try your patch.
Marconi Rivello wrote:
Where to get it:
http://www.carcara.lncc.br/marconi/mr_busydetect.patch
tatsuya
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Asterisk-Users mailing list
[EMAIL PROTECTED]
After long email communication with May Lin, Wellgate's International
Sales Dept./Project Manager, I was asked to supply them with a list of
email addresses of people with the same FXO/FXS sip version hardware's
bugs related to registration and anything else. They're trying to find
out the
Well,
assuming that some of these CODECS do error correction and drop any
information that hasn't come through instead of doing error detection and
request to re-transmit the lost information, is somewhat expected. Are there
any Fax over IP protocols?
Yiannis.
-Original
There is no way to convert existing files to g729? The only reason we need
the licenses is to access voicemail since they are in GSM. All our phones
have g729 built in. But if you try and access VM, you get that No coversion
for GSM to g729 error. But if all the voicemail sounds where in g729,
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