On Thu, Sep 11, 2008 at 8:10 PM, C. Chad Wallace
[EMAIL PROTECTED] wrote:
At 8:29 AM on 11 Sep 2008, John Millican wrote:
Not directly on-topic for this list, but I'd not heard of OpenSIPS
before, so I had a look at the website. It looks to be a fork of
OpenSER. Does that mean OpenSER
On Tue, Sep 9, 2008 at 3:34 PM, Darren Sessions [EMAIL PROTECTED] wrote:
I would suggest using OpenSIPS with Asterisk and bypass IAX all together for
this particular application.
An OpenSIPS solution will take care of your traveler's NAT issues (and could
handle the registrations) while you
Hi all,
The usual suspects will be gathering today at 12 EDT. Join us on the
VUC if you have the time:
Details: http://VoipUsersConference.org
PSTN 1(724) 444-7444 and enter 22622# 1#
SIP [EMAIL PROTECTED] DTMF 22622# 1#
IRC: #voip-users-conference on Freenode.net
RSS:
Thanks! You're the best!
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
=== AND MY PERSONAL PAGES AT: ===
Russell!
This time it's really a problem:
when I use application Jack I get input and output. When I use
functionJACK_HOOK with the same options, just copied from the Jack call, I
only get one way. the o-option doesn't work. I connect it to my microphone,
sstem:capture_1. So nothing
Some addition...
Something I find even stranger is that jack_lsp shows, that the asterisk
input AND output ports do exist and ARE CORRECTLY connected. So I should get
audio from my microphone and still I don't.
Hope that helps...
Kindest regards
Julien
Music was my
xx-montague-gardens*CLI show uptime
System uptime: 38 years, 37 weeks, 4 days, 10 hours, 47 minutes, 11
seconds
Amazing. Especially considering:
[EMAIL PROTECTED]:/var/log uptime
09:58:14 up 18:42, load average: 0.21, 0.09, 0.02
Steve
___
--
On 09:59, Fri 12 Sep 08, Stephen Davies wrote:
xx-montague-gardens*CLI show uptime
System uptime: 38 years, 37 weeks, 4 days, 10 hours, 47 minutes, 11
seconds
Amazing. Especially considering:
[EMAIL PROTECTED]:/var/log uptime
09:58:14 up 18:42, load average: 0.21, 0.09, 0.02
On 12 Sep 2008, at 09:20, Michiel van Baak wrote:
On 09:59, Fri 12 Sep 08, Stephen Davies wrote:
xx-montague-gardens*CLI show uptime
System uptime: 38 years, 37 weeks, 4 days, 10 hours, 47 minutes, 11
seconds
Amazing. Especially considering:
[EMAIL PROTECTED]:/var/log uptime
Joseph L. Casale wrote:
Now that we have voicemail working, people have asked to be able to
dial in externally and be able to access their voicemail. My dial plan is
You can either setup a context for just checking voice mail or you can
use the following option under the voice mail
On Thu, Sep 11, 2008 at 9:15 PM, Gordon Henderson
[EMAIL PROTECTED] wrote:
On Thu, 11 Sep 2008, Russell Bryant wrote:
[EMAIL PROTECTED] wrote:
Is the idea to switch to another video source or stay with the callers
camera? An option for both would be nice. I could see a help desk
placing
On 12 Sep 2008, at 10:13, Tim Panton wrote:
I'd guess the battery on your motherboard has died so it is going back
to 1970 at
boottime.
Watchout, because this can also mean that your BIOS is about to
loose all settings too which can cause it to forget how to talk to the
harddrive :-(
Hmm
On Fri, Sep 12, 2008 at 11:13 AM, Tim Panton [EMAIL PROTECTED] wrote:
I'd guess the battery on your motherboard has died so it is going back
to 1970 at
boottime.
Why do hide the truth, Tim? It's much more likely the motherboard
traveled back 38 years in time, is it not?
r
2008/9/11 Stefan Schmidt [EMAIL PROTECTED]:
Steve Davies schrieb:
Thanks for that excellent information - Now does anybody know the XML
to provision that field? Normally you take the text on the screen
Call Pickup Code and replace space with underscore
Call_Pickup_Code ua=na *8#
On Fri, Sep 12, 2008 at 10:13:11AM +0100, Tim Panton wrote:
On 12 Sep 2008, at 09:20, Michiel van Baak wrote:
On 09:59, Fri 12 Sep 08, Stephen Davies wrote:
xx-montague-gardens*CLI show uptime
System uptime: 38 years, 37 weeks, 4 days, 10 hours, 47 minutes, 11
seconds
Amazing.
Hi,
The Dial command has the g option, voip-info.org says:
If the g option is specified, and the called party hangs up before
the calling party, then Dial continues execution at priority n+1.
and this works well. But I need to continue the execution if the
caller hangs up first too.
What do I
On Fri, Sep 12, 2008 at 2:35 PM, Gergo Csibra [EMAIL PROTECTED] wrote:
Hi,
The Dial command has the g option, voip-info.org says:
If the g option is specified, and the called party hangs up before
the calling party, then Dial continues execution at priority n+1.
and this works well. But I
Thanks Shaun___
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Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi List;
First of all, how can I know that the telephone line is not fixed in the fxo
port?
Then, if the Dial function used to place a call via the zaptel (via the fxo
port), and no telephone line was fixed in the fxo, can I have any returned
error to know that the telephone line is not fixed
On Fri, Sep 12, 2008 at 7:35 AM, Gergo Csibra [EMAIL PROTECTED] wrote:
Hi,
The Dial command has the g option, voip-info.org says:
If the g option is specified, and the called party hangs up before
the calling party, then Dial continues execution at priority n+1.
and this works well. But I
Sean Bright wrote:
Thomas Kenyon wrote:
In trying to upgrade my test machine from 1.6.0beta9 to 1.6.0rc6 when I
try to make menuseletc I get the following error.
This is using gcc 4.1, libgtk 2.0, on an intel Core2Duo machine running
an up to date Debian etch.
Asterisk builds okay (not
2008/9/12 randulo [EMAIL PROTECTED]
On Fri, Sep 12, 2008 at 11:13 AM, Tim Panton [EMAIL PROTECTED] wrote:
I'd guess the battery on your motherboard has died so it is going back
to 1970 at
boottime.
Why do hide the truth, Tim? It's much more likely the motherboard
traveled back 38 years
The best way I can think of is:
wget http://ftp.digium.com/pub/asterisk/asterisk-1.4.21.2.tar.gz
tar -zxvf asterisk-1.4.21.2.tar.gz
cd asterisk-1.4.21.2
./configure
make menuselect (You don't have to select anything)
make
make install
make samples
Pascal Bruno wrote:
I am about to
Stephen Davies wrote:
Why don't you guys believe that my Asterisk has just been up for 38 years?
Because Mark was born in 1977 and he's 31.
http://en.wikipedia.org/wiki/Mark_Spencer
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Maybe that robot in his office doubles as a time machine.
___
Chris Hoff
Telecommunications Administrator
SEI LLC
Voice +1 701 298 8865 Ext 2189
Mobile +1 701 361 5976
Fax +1 701 298 8860
Email [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
Nominated for dumbest thread ever
On Fri, Sep 12, 2008 at 9:34 AM, Christopher Hoff
[EMAIL PROTECTED] wrote:
Maybe that robot in his office doubles as a time machine.
___
Chris Hoff
Telecommunications Administrator
SEI LLC
Voice +1 701 298 8865 Ext 2189
On Fri, Sep 12, 2008 at 3:24 PM, Doug Lytle [EMAIL PROTECTED] wrote:
Stephen Davies wrote:
Why don't you guys believe that my Asterisk has just been up for 38 years?
Because Mark was born in 1977 and he's 31.
Which proves the time travel explanation!
On Thu, Sep 11, 2008 at 08:11:09PM -0500, Russell Bryant wrote:
The Jack application acts as an endpoint for a call.
A bit of nomenclature: is Jack the name of an Asterisk application? Or
are you referring to JACK, the Jack Audio Connection Kit, whose name is
all-caps, directly? And if not,
Julien Claassen wrote:
Something I find even stranger is that jack_lsp shows, that the asterisk
input AND output ports do exist and ARE CORRECTLY connected. So I should get
audio from my microphone and still I don't.
Hope that helps...
Can you share the dialplan that you're using?
Tzafrir Cohen wrote:
I usually configure the entire span of 24 channels (23 B + 1 D) and
only the turned up channels go into service. This is good for a
couple of reasons.
Also note that Zaptel will anyway reserve all the 24 (for T1) or 31 (for
E1) Zaptel channels for the span. So
Jay R. Ashworth wrote:
A bit of nomenclature: is Jack the name of an Asterisk application? Or
are you referring to JACK, the Jack Audio Connection Kit, whose name is
all-caps, directly? And if not, of course, is Jack something that
connects JACK to Asterisk?
Sorry for the confusion.
There
Ok very good, how about for the asterisk addonds and sounds? Can you
provide me the commands to get, build and install for the 1.4.21 version?
Thanks a lot guys.
On Fri, Sep 12, 2008 at 6:07 AM, MFH [EMAIL PROTECTED] wrote:
The best way I can think of is:
wget
On Fri, Sep 12, 2008 at 09:04:57AM -0500, Russell Bryant wrote:
Jay R. Ashworth wrote:
A bit of nomenclature: is Jack the name of an Asterisk application? Or
are you referring to JACK, the Jack Audio Connection Kit, whose name is
all-caps, directly? And if not, of course, is Jack
Dear All,
I have the following scenario...When a customer dial 111 number a beep
message will iplay in order to record and playback his voice...Else he'll be
routed to another call flow as you can see in the context below:
[a2billing]
exten = _X.,1,Gotoif($[${EXTEN} = 111] ?
Hi all
I'm just having a problem now and I don't have any idea how to do this.
It is pretty simple. When a customer calls, to speed up the navigation
in the dialplan, I want something like
Welcome. Please enter your 10 digit customer number or press * to register
So, I want to read up to 10
Asterisk 1.6 installed with last zaptel...
On cli, when typing zap show channels, I get No such command 'zap show
channels' (type 'help zap show' for other possible commands)
Help doesn't help, of course...
I have a zaptel conf on the /etc/asterisk...
Any Idea?
Olivier
On Fri, Sep 12, 2008 at 09:53:48AM -0400, Bill Michaelson wrote:
Tzafrir Cohen wrote:
I usually configure the entire span of 24 channels (23 B + 1 D) and
only the turned up channels go into service. This is good for a
couple of reasons.
Also note that Zaptel will anyway reserve
On Fri, Sep 12, 2008 at 11:07 AM, hh174 [EMAIL PROTECTED] wrote:
Asterisk 1.6 installed with last zaptel...
On cli, when typing zap show channels, I get No such command 'zap show
channels' (type 'help zap show' for other possible commands)
Help doesn't help, of course...
I have a zaptel
On Fri, Sep 12, 2008 at 11:09 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Fri, Sep 12, 2008 at 11:07 AM, hh174 [EMAIL PROTECTED] wrote:
Asterisk 1.6 installed with last zaptel...
On cli, when typing zap show channels, I get No such command 'zap show
channels' (type 'help zap show' for other
2008/9/12 Doug Lytle [EMAIL PROTECTED]
Stephen Davies wrote:
Why don't you guys believe that my Asterisk has just been up for 38
years?
Because Mark was born in 1977 and he's 31.
Oh dear. Maybe this will help: ;-) :-)
Steve
___
--
Hello Russell!
Certainly, here's the shortened dialplan:
exten = NUM,1,System(ast_picker ring.wav)
exten = NUM,2,Answer()
exten = NUM,3,GotoIf($[${SYSTEMSTATUS} = SUCCESS]?4:7)
exten = \
NUM,4,Set(JACK_HOOK(manipulate,i(sstem:playback_1)o(system:capture_1)=on)
exten = NUM,5,System(ast_connect)
I have a 7921 wireless phone working with Asterisk, and I want to tighten
the wide open port range of my IPTABLES now.
I tried allowing only SCCP port (2000) in/out and found that my audio was
gone. A quick look at my iptables message shows source port 15886 and dest
port 25968 used:
FORWARD
Stephen Davies wrote:
Because Mark was born in 1977 and he's 31.
Oh dear. Maybe this will help: ;-) :-)
I knew you were joking, maybe I should have added a :=P
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety,
Hello!
I'll classify the subject. :-) I have a nasty firewall, I don't have to much
power over. It's javascript based in configuration and I can't use any
graphical browser. The only other person at my home, doesn't know too much
about computers.
So I know, from experience, that SIP is
On Fri, Sep 12, 2008 at 11:19 AM, OCG Technical Support [EMAIL PROTECTED]
wrote:
I have a 7921 wireless phone working with Asterisk, and I want to tighten
the wide open port range of my IPTABLES now.
I tried allowing only SCCP port (2000) in/out and found that my audio was
gone. A quick
Thomas Kenyon wrote:
Sean Bright wrote:
Thomas Kenyon wrote:
In trying to upgrade my test machine from 1.6.0beta9 to 1.6.0rc6 when I
try to make menuseletc I get the following error.
This is using gcc 4.1, libgtk 2.0, on an intel Core2Duo machine running
an up to date Debian etch.
SCCP (aka Skinny), H323, MGCP, and SIP all use the RTP protocol for
audio. For all signalling protocols (except maybe H323) use rtp.conf
for the RTP ports.
OCG Technical Support wrote:
I have a 7921 wireless phone working with Asterisk, and I want to
tighten the wide open port range of
Jay R. Ashworth wrote:
On Mon, Sep 08, 2008 at 11:28:13AM -0500, Matthew Fredrickson wrote:
For DMS100's version of TBCT, called RLT, one leg *must* be inbound and
the other *must* be outbound. No other combination is going to work.
This is explicitly mentioned in the protocol in RLT.
Ok.
Hi,
Am Freitag, den 12.09.2008, 11:03 -0400 schrieb Ruddy Gbaguidi:
Hi all
I'm just having a problem now and I don't have any idea how to do this.
It is pretty simple. When a customer calls, to speed up the navigation
in the dialplan, I want something like
Welcome. Please enter your 10
In article [EMAIL PROTECTED],
Ruddy Gbaguidi [EMAIL PROTECTED] wrote:
Hi all
I'm just having a problem now and I don't have any idea how to do this.
It is pretty simple. When a customer calls, to speed up the navigation
in the dialplan, I want something like
Welcome. Please enter your 10
I'm setting up a 7921 and now want to add a second line to the phone. In my
SCCP.conf file I have:
autologin = 235,299
However, on reloading SCCP the phone fails to login to the second line with
this error:
[Sep 12 12:46:49] WARNING[12224]: sccp_actions.c:185 sccp_handle_register:
On Fri, Sep 12, 2008 at 10:56:40AM -0500, Matthew Fredrickson wrote:
Will I actually need to do PRI debug on that span to tell?
Or will seeing hangup messages while I'm still talking be the solution?
Seeing hangup messages on the console while the audio path remains
indicates success
Dear, I'm looking for IP phones (directly connected to the RJ-45 port
from my LAN) that support any level of encryption for use with an
Asterisk 1.4 SIP server we have.
What branch and type can I use
What is the encryption mechanism I can have with this equipments ???
Greetings
Jay R. Ashworth wrote:
On Fri, Sep 12, 2008 at 10:56:40AM -0500, Matthew Fredrickson wrote:
Will I actually need to do PRI debug on that span to tell?
Or will seeing hangup messages while I'm still talking be the solution?
Seeing hangup messages on the console while the audio path remains
I've added lines like this:
speeddial = 123,test
speeddial = 260,Bob
in the [device] section for my 7921, but the speed dials do NOT appear on
the menu (click right from the main screen). Am I missing something obvious
here?
Thanks
MD
Thanks for the hint. Sorry about that.
If I use your soution, I cannot make any difference between a user
pressing * and a user that reach the timeout because he didn't enter any
digit.
In both cases, I will have an empty string
Karsten Wemheuer wrote:
Hi,
Am Freitag, den 12.09.2008, 11:03
Hi thanks for the hint.
That will works I think.
But now, if I'm in an AGI script and I want to stay in there and don't
want to jump from an extension to other in the dialplan,
how can I do it ??
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Ruddy Gbaguidi [EMAIL PROTECTED] wrote:
On Fri, Sep 12, 2008 at 12:12:56PM -0500, Matthew Fredrickson wrote:
Can *you* confirm, off hand, that 1.2 would do TBCT at *all*? Someone on
IRC thinks it wouldn't.
It will only attempt it for DMS100 switchtype. You must have 1.4 libpri
for any other switchtype.
Will libpri 1.4 work
On 12:51, Fri 12 Sep 08, OCG Technical Support wrote:
I'm setting up a 7921 and now want to add a second line to the phone. In my
SCCP.conf file I have:
autologin = 235,299
However, on reloading SCCP the phone fails to login to the second line with
this error:
[Sep 12
Hi Michel,
Am Freitag, den 12.09.2008, 17:41 +0300 schrieb michel freiha:
Dear All,
I have the following scenario...When a customer dial 111 number a beep
message will iplay in order to record and playback his voice...Else
he'll be routed to another call flow as you can see in the context
On 13:15, Fri 12 Sep 08, OCG Technical Support wrote:
I've added lines like this:
speeddial = 123,test
speeddial = 260,Bob
in the [device] section for my 7921, but the speed dials do NOT appear on
the menu (click right from the main screen). Am I missing something obvious
I am having problems with echo cancel using dahdi and latest (as of
Saturday) version of asterisk 1.4.
The problem only occurs between zap and sip or iax. The far end gets
an echo. I can even get it by calling my own analog phone hooked up
to an ata! Zap to Zap is just fine.
Here is my
Does your box run on the Mr. Fusion power supply?
Doug Lytle wrote:
Stephen Davies wrote:
Because Mark was born in 1977 and he's 31.
Oh dear. Maybe this will help: ;-) :-)
I knew you were joking, maybe I should have added a :=P
Doug
Hi Ruddy,
Am Freitag, den 12.09.2008, 13:22 -0400 schrieb Ruddy Gbaguidi:
Thanks for the hint. Sorry about that.
If I use your soution, I cannot make any difference between a user
pressing * and a user that reach the timeout because he didn't enter any
digit.
In both cases, I will have an
Chan_sccp again...
From what I read chan_sccp is the successor to chan_skinny.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel van
Baak
Sent: September 12, 2008 2:08 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Setup speed dials
I have a call between two people. I know their channel identifier. I want
to trasfer a call away from one person and pass it to another person.
To start, let's talk about a blind transfer. My system places both outgoing
calls to people and bridges them together (cheaper, works via AGI).
On 15:37, Fri 12 Sep 08, OCG Technical Support wrote:
Chan_sccp again...
From what I read chan_sccp is the successor to chan_skinny.
No, it's a fork that never contribute back anything to asterisk.
The last year there have been activity in chan_skinny again, and I can
say it works ok for my
Is there a way to force asterisk to take care only of sip signaling without
forcing it to take care of rtp traffic?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register
http://www.taylortelephone.com/asterisk/
There are install scripts for Centos 5 Asterisk 1.4. They should work just fine
on FC9. If you have a problem just email me.
Jonn
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pascal Bruno
Sent: Friday, September 12, 2008
Asterisk 1.6rc4 will only use dahdi. I just went though this on my test system.
Jonn
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Friday, September 12, 2008 10:12 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Pascal Bruno [EMAIL PROTECTED] writes:
Ok very good, how about for the asterisk addonds and sounds? Can you
provide me the commands to get, build and install for the 1.4.21 version?
Thanks a lot guys.
If you can't figure that out on your own, you really should stick with
the
Should have been 1.6.0rc6.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonn R Taylor
Sent: Friday, September 12, 2008 4:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk 16 and zapata
Asterisk
You wont need things like PHP, MySQL, etc but you do need some of the other
things otherwise you'll get errors. And while I run these as automated
batches, I suggest you take my commands and do them one line at a time.
Keep an eye out for errors.
yum -y install kernel kernel-devel ntp
yum -y
Thanks Jonn!!!
On Fri, Sep 12, 2008 at 2:02 PM, Jonn R Taylor [EMAIL PROTECTED]wrote:
http://www.taylortelephone.com/asterisk/
There are install scripts for Centos 5 Asterisk 1.4. They should work just
fine on FC9. If you have a problem just email me.
Jonn
It's like the same except you wget a different package and I don't think
you have a menuselect option and you do it before you compile asterisk.
For addons I think there might be some configuration if you are
planning to use the database stuff which I don't use. The sounds come
with the
Hi,
is anyone Siemens OpenStage 20 SIP phone connected to asterisk 1.4 ?
Since V1 R4.11.0 the phone shows Number unavailable each time an
outgoing call gets connected. To users this looks like an error
message. It is a bit confusing.
This problem did not occur when V1 R3 was used, but this had
Il Neofita wrote:
Is there a way to force asterisk to take care only of sip signaling
without forcing it to take care of rtp traffic?
Yes. The canonical way is to enable canreinvite=yes on both SIP peers
(incoming and outgoing legs), which will cause Asterisk to send a new
INVITE within
In article [EMAIL PROTECTED],
Ruddy Gbaguidi [EMAIL PROTECTED] wrote:
Hi thanks for the hint.
That will works I think.
But now, if I'm in an AGI script and I want to stay in there and don't
want to jump from an extension to other in the dialplan,
how can I do it ??
Ah, you didn't say
Hello
I updated the Ports collection to compile the latest Asterisk, but
after running make config, make just returns without doing
anything:
=
# pkg_version -v | grep asterisk
asterisk-1.4.20.1_1needs updating (port has
1.4.21.2_3)
^C
# cd
http://www.voip-info.org/wiki-IAX
http://www.voip-info.org/wiki-IAX+versus+SIP
http://www.voip-info.org/wiki/view/Asterisk+IAX+clients
Terve,
Stefan
--
Last words of a stormchaser:
Where is that rotation on the radar?!
___
-- Bandwidth and
The short answer is SIP.
Stefan Gofferje wrote:
http://www.voip-info.org/wiki-IAX
http://www.voip-info.org/wiki-IAX+versus+SIP
http://www.voip-info.org/wiki/view/Asterisk+IAX+clients
Terve,
Stefan
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1)
I would have said the short answer is IAX
:)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: September 12, 2008 7:31 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Which internet phone protocol best to choose
The short
Hello!
IAX I can basically understand, although I wasn't aware in the slightest,
that other standard softphones supported it.
But why SIP? Correct me if I'm wrong. there's a standard SIP-port. Then you
send out the request to talk, then server and client negotiate a port for the
audio
On Friday 12 September 2008 18:31:23 Alex Balashov wrote:
The short answer is SIP.
Stefan Gofferje wrote:
http://www.voip-info.org/wiki-IAX
http://www.voip-info.org/wiki-IAX+versus+SIP
http://www.voip-info.org/wiki/view/Asterisk+IAX+clients
The longer and more accurate answer is that it
Alex Balashov schrieb:
The short answer is SIP.
Maybe not behind a firewall which you don't have control over. IAX is a
single-port-protocol and as such much less problematic with firewalls
and NAT.
Read the second link in my previous mail.
Terve,
Stefan
--
Last words of a stormchaser:
Where
Julien Claassen schrieb:
IAX I can basically understand, although I wasn't aware in the slightest,
that other standard softphones supported it.
They don't. Well - it depends, what you see as standard. There are very
good multi-platform combined SIP/IAX clients like Zoiper. But Zoiper is
not
Hello Stefan!
Sorry for the miss-understanding. I didn't refer to your mail about IAX, but
about the one sayng SIP. I read your links and it seems I'll delve into it.
I'll try to quote next time. I hate doing this, it always looks a bit
unorganised, while writing... :-(
Kindest regards
The OP asked, if I recall, about the protocol which is likely to be
supported rather universally by softphones and a wide variety of clients.
That is not a feature of IAX.
Tilghman Lesher wrote:
On Friday 12 September 2008 18:31:23 Alex Balashov wrote:
The short answer is SIP.
Stefan
On Fri, Sep 12, 2008 at 7:43 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Friday 12 September 2008 18:31:23 Alex Balashov wrote:
The short answer is SIP.
Stefan Gofferje wrote:
http://www.voip-info.org/wiki-IAX
http://www.voip-info.org/wiki-IAX+versus+SIP
Really? I thought both IAX and SIP are, at 3 characters apiece, equally
short.
However, if you get into IAX2, then yes... SIP is definitely a shorter
answer.
N.
Alex Balashov wrote:
The short answer is SIP.
Stefan Gofferje wrote:
http://www.voip-info.org/wiki-IAX
But user just needs to enter * instead of *#
We are doing this because 80% of the callers already have an account,
so, instead of playing :
If you have an account, press 1, if not press 2
we prefer to play
Enter you account now or press * if you don't have any
Karsten Wemheuer wrote:
Hi
Hello I've been trying to add a string to CIDNAME for incoming calls from
PSTN to tag calls so I know how to answer more appropriately. I have tried
numerous combinations to no avail and hope someone can point me in the right
direction. My context from extensions.conf is listed below.
Thanks for your help.
This can be add to Read command as feature
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Ruddy Gbaguidi [EMAIL PROTECTED] wrote:
Hi thanks for the hint.
That will works I think.
But now, if I'm in an AGI script and I want to stay in there and don't
want to
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