I am trying to locate the manual for that level software. If it's not here
at home it is at my office and I will look everything up in the morning.
- Original Message -
From: Scott Wolfe [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
On 20:47, Sat 23 Apr 05, Stefan Gofferje wrote:
Besides chan_capi does not understand Busy() and Congestion(), that
probably is a matter of how fast the faxmodem picks up the call and to
what value your timeout is set.
However, I think, I have another strategy regarding security... I have
Has anybody success with speed dialing?
If so, I am sure you can help me to get into this club.
tgj wrote:
Hi Ronald,
It seems like you need to put in default as your context. However I think
your problem was that you put the number in CallerID column and The CallerID
in the Name column. I was
Can anybody help me to figure out how much memory per minute is consumed
for voicemail applications? And how many concurrent calls can be
handled at a time in Asterisk? so that, I can choose the specification
for Server to setup Asterisk for a large number of users.
Hi !
What is the easiest esyest way for implementation
of ztdummy on a Debian (testing) system?
Thore
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Hi All,
Debian Sarge (most recent update yesterday). Running a custom-built
2.6.8 kernel (Debian kernel doesn't have the Traverse transparent mode
patch - the 2.4 patch seemed to apply to the 2.6 sources OK).
:01:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN
Title: Nachricht
-
having the kernel sources or kernel headers installed
-
uncommenting ztdummy in zaptel'sMakefile
- make
/ make install :)
assuming you have an uhci chip on your main board and kernel
2.4x
With
kernel 2.6 make a make linux26 and things are more easy
regards
Manuel
Hi,
This is how I got ztdummy on debian sarge:
$ apt-get install kernel-headers-2.6.8-2-386 dpatch kernel-package zaptel
zaptel-source
$ cd /usr/src
$ ln -s kernel-headers-2.6.8-2-386/ linux
$ cd linux
$ make-kpkg modules_image
$ dpkg -i
Hi,
Does * support QSIG?Some experience with
it?
Which card are adequated?
Regards.
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Dear All :
How can I enable the announcement Feature of Meet-me rooms ?
So that when I enter the conference room , the system ask me
about my name ,, then announce all the existing people in the room about my entrance
..
Also when I go out of the conference an
announce should be played
Hi !
I this working with kernel 2.4?
Thore
- Original Message -
From: Samuel T. Cossette [EMAIL PROTECTED]
To: Thore [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Sent: Sunday, April 24, 2005 1:45 PM
Subject: Re:
Hi,
I am currently having a problem where I am making outbound calls via
IAX, these calls are then being routed by my provider through a SIP
connection to a service providing PSTN access.
The problem I have is the the DTMF is being sent inband over the SIP
connection, and I am only receiving
Of course!
The trick are the kernel headers but they must of course fit onto the
installed kernel.
The problem is: For the zaptel stuff you need more then the downloaded
stuff. You need the kernel sources or headers.
Manny
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
Hi folks.
I have a problem with a pythonscript designed for joining one in-wav
and one out-wav
after recording a call.
Yes, I have the wav-files after a successfull recording...
Python stops at line 5:basename = sys.argv[2]
like:
[EMAIL PROTECTED] bin]# python
Is anyone running an Asterisk server and connecting over VSAT? I'd love to
talk to you about your exteriences, or any experiences with VSAT with or
without Asterisk.
Chris Mason
Int: (646)722-0001 Fax: (815)301-9759
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Hi,
I know it's a redundant question but this time it's different, I'm
looking for a mature enough management and administration GUI to which I
can further contribute and work with the developers to deliver a well
featured package to meet with what people really demand, it's a shame
that people
Sorry, I forgot to show the script, here it is:
#!/usr/bin/python
import tempfile,os,sys,re,time
monitordir = sys.argv[1]
basename = sys.argv[2]
def runcmd(cmd):
print cmd
os.system(cmd)
#mix to one wav
inwav = os.path.join(monitordir, basename+-in.wav)
outwav =
Hello everyone:
I have just reinstalled asterisk and astcc. Asterisk
is working fine, but not astcc. When I try to make iax
call through voipjet, Astcc is working fine till the
pont where it tries to make the call, it gives a
congestion message.
Here is the message i get when i attempt to make
In article [EMAIL PROTECTED],
Mohamed Farid [EMAIL PROTECTED] wrote:
Dear All :
How can I enable the announcement Feature of Meet-me rooms ?
So that when I enter the conference room , the system ask me about my
name ,, then announce all the existing people in the room about my
entrance ..
When do you use Registerport 5060 and when 1720 ??
bye
Ronald
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Anyone have any ideas here?
We are using 8 channels of EM Wink with a T100P for outgoing LD and
incoming tollfree numbers and are apparently connected to a Nortel
DMS-250 at the CO. We are receiving ANI DNIS just fine and can
dial-out domestically with DTMF but have two issues that are still
I run m0n0wall (http://m0n0.ch/wall) on a Soekris 4501 embedded PC
(http://www.soekris.com). Very tweakable. Under $200.
Michael
On Fri, 22 Apr 2005 10:42:20 -0700, Max Clark wrote:
Hi all,
I am looking for good (sub $200 dollars) routers to support VoIP
installations. What is available at
Thanks Henry,
-Scott
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, April 23, 2005 11:05 PM
Subject: Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
I am trying to locate
This link points to a page about a switch...not a router.
Michael
On Fri, 22 Apr 2005 18:14:05 -0400, Iassen Hristov wrote:
Maybe this fits the bill.
http://www.gigafast.com/products/product_detail/EE2400-SS.htm
It retails for less than $100
Message: 9
Date: Fri, 22 Apr 2005 10:42:20 -0700
Hello everyone,
I'm been toying with the idea of allowing my users to use meetme but
have had some service quality issues (which I know are being
addressed) but am concerned about making work for myself for something
I can outsource...
Junction Networks (http://www.junctionnetworks.com) seems
The 5060 is usually SIP Proxy listen port.
And the 1720 is usually h323 gatekeeper's listen port.
On 4/24/05, Ronald Wiplinger [EMAIL PROTECTED] wrote:
When do you use Registerport 5060 and when 1720 ??
bye
Ronald
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Asterisk-Users mailing
Hi,
I need to use cidsignalling=dtmf where the callerid comes after the first
ring.
Looking in source code of chan_zap.c I understand that cidsignalling=dtmf
only works when cidstart=polarity.
Is this right? or also works with cidstart=ring?
Thanks
Alejandro
The idea:
1. IPv6 is experimental, I would like to set it up as an extra box
2. MeetMe could kill my bandwidth. I would like to co-locate it.
How can I combine different boxes / at different location to one system?
How can I reach each other?
Does anybody have experience in doing that?
bye
Ronald
[EMAIL PROTECTED] wrote:
On Fri, 22 Apr 2005, Peter Bowyer wrote:
On 22/04/05, Ian Hailey [EMAIL PROTECTED] wrote:
Hello everyone,
I am trying to receive DTMF commands on asterisk from PSTN calls
terminated at my asterisk box. I have tried to terminate the PSTN calls
with both SIP and IAX
Ian Hailey wrote:
[EMAIL PROTECTED] wrote:
On Fri, 22 Apr 2005, Peter Bowyer wrote:
On 22/04/05, Ian Hailey [EMAIL PROTECTED] wrote:
Hello everyone,
I am trying to receive DTMF commands on asterisk from PSTN calls
terminated at my asterisk box. I have tried to terminate the PSTN
calls
with
Please contact me Urgent...
Atentamente,
Franz Schuverer Arrue
GLOBAL GROUP, INC.
www.telefoniaglobal.net
[EMAIL PROTECTED]
Tel. (504) 221-4062 (Honduras
Tel. (507) 322-2259 (Panamá)
Tel. (866) 978-0976 (U.S.A.)
(SKYPE) franz1969
(MSN MESSENGER) [EMAIL PROTECTED]
(YAHOO MESSENGER) [EMAIL
Hi all,
What is the best client's protocol for my softphones in Windows pcs? and
what is the best way for connecting clients, I meant should I use one
protocol for all the clients or some mix (SIP/IAX/oh323) of protocols?
what
is better?
Thanks in advance
Kumara
Hi,
I'd kindly ask if anyone can provide working configuration examples for
Asterisk-Fritz-mISDN combo.
Thanks in advance,
regards,
Rob.
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Hi all,
What is the best client's protocol for my softphones in Windows pcs? and
what is the best way for connecting clients, I meant should I use one
protocol for all the clients or some mix (SIP/IAX/oh323) of protocols?
what
is better?
Thanks in advance
Kumara
Sorry, I
For my two cents .. IAX/IAX2 is the only way to go .. It stops most if not
all Firewall issues as well as double NAT ...
BRW
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kumara
Jayaweera
Sent: Sunday, April 24, 2005 10:54 AM
To:
The digitmap is in your telephone. Used to terminate dialing and send
the dialed string to *.
On Apr 23, 2005, at 11:56 PM, Jaime Blanco wrote:
Jerry,
when you say digitmap, you mean in my extensions.conf file?
Thanks.
Jaime
From: Jerry [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List -
I am building a simple form validation that needs to
do a simple validation on the _length_ of a phone
number. As we all know, different countries have
different phone number lengths.
For example, Australia phone numbers can be either 6
or 7 digits, while USA phone numbers are always 10
digits.
Jerry wrote:
The digitmap is in your telephone. Used to terminate dialing and send
the dialed string to *.
Grandstream BT phones don't have a digitmap feature.
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
___
What is the best client's protocol for my softphones in Windows pcs? and
what is the best way for connecting clients, I meant should I use one
protocol for all the clients or some mix (SIP/IAX/oh323) of protocols?
what
is better?
I would say IAX. If you only use 1 protocol for
Hello,
how can I prevent Asterisk from trying to create a native bridge between
an incoming call from a SIP provider and an extension attached to a
SIP ATA?
My Asterisk is behind a firewall, and the native bridge invariably fails.
Thanks in advance for any suggestion!
(I DID search the list
add
canreinvite=no
to the sip user definition blocks for the SIP provider and for the SIP ATA.
Regards,
Marc
Wolf N. Paul wrote:
Hello,
how can I prevent Asterisk from trying to create a native bridge between
an incoming call from a SIP provider and an extension attached to a
SIP ATA?
My Asterisk
I always answer myself because my mind would be tired
at the time of asking the question. This time, i made
a mistake in the trunk configuration in astcc web
interface. peer/trunk was [EMAIL PROTECTED] and
should've been [EMAIL PROTECTED]
--- chawki hammoud [EMAIL PROTECTED] wrote:
Hello
Hi
Is there any help for me to register my quantium A800 (SIP) with my Asterisk
.
Please help me what should me my Sip.conf
now present i did
[1234567]
type=friend
context=sip
username=
secret=
nat=yes
host=dynamic
canreinvite=no
defaultip=XXX.XXX.XXX.XXX
disallow=all
allow=g729
allow=gsm
On Fri, 22 Apr 2005, Chris Coulthurst wrote:
Is there a specific SIP or IAX phone that truly shines above the rest
where it comes to 'happy' compatibility with Asterisk? I guess I'm
talking about feature sets, like early-dial, off hook call announcing,
conferencing, echo suppression, etc
Hey guys, I am aware that Asterisk may be a bit overkill for what I need but I
haven't found any other software. Here's what I need to do:
I have 1 computer with 2 modems in it. Each modem (regular 56k )is plugged into
a different phone line ( line A and line B ).
Whenever a call comes on line
The short answer is Yes. However, you would need X100P cards and not regular
modem cards. These cards can be found on eBay for about $7 each.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, April 24, 2005 12:34 PM
I have a T1 (PRIs) getting installed tomorrow and plan to plug it into
a Sangoma A101. My question is are there any specifics I need to tell
the CLEC's engineer regarding the configuration for Asterisk to see
it?
This is obviously new to me so any help is most appreciated.
Regards,
Jess
I finally figured it out ... working with BT100 you need to make a little
voodoo ritual first :-) ... so follow the steps --exactly-- if you have
trouble
This is my working configuration behind Linksys WRT54G router:
- Upgrade firmware 1.0.5.23
- Reset BT100 to factory defaults
- SIP Server:
Customer has integrated access arrangement with 16 channels of data/8
for voice that is split via customer cisco equipment. No local dialing,
LD and incoming 800 service only via the t1. Qwest provides both the
local loop and LD/800 service but it is provided via re-seller PNG.
We have
On 24 Apr 2005, at 18:53, Kumara Jayaweera wrote:
Hi all,
What is the best client's protocol for my softphones in Windows
pcs? and
what is the best way for connecting clients, I meant should I use
one
protocol for all the clients or some mix (SIP/IAX/oh323) of
protocols?
what
is
Why don't you use Vonage (what ever that might be :) to forward to a
free account at a sip or iax phone provider somewhere in the world, make
your European asterisk register with that account and dial out locally?
:)
Of course this and much furtehr similar works! :)
-Ursprüngliche
Excellent news!
Now, remember that I am in Europe and out of my Linksys VoIP router I have a
phone line coming out which I believe is North American standard.
I am really not knowledgeable enough about the differences in both Networks so
do I need a special card for Europe ?
Otherwise, I'm set to
Don't forget you may like to support digium by buying an official
tdm400P
I know more expensive then a $7 clone but will work better on lines
different to the 600ohm US pstn
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL
bill black wrote:
Anyone have any ideas here?
We are using 8 channels of EM Wink with a T100P for outgoing LD and
incoming tollfree numbers and are apparently connected to a Nortel
DMS-250 at the CO. We are receiving ANI DNIS just fine and can
dial-out domestically with DTMF but have two
Joseph wrote:
We have the same problem with 7960, just randomly it will stop *hearing*
the dtmf tones and you have to hangup and call back.
This problem was fixed in CVS long ago, and current stable releases have
the fix as well. When you are running a copy of Asterisk that is 4/5
months old,
According to the Mitel manuals that version of SX-200D can only use a
regular 24 channel T1. It can not use a PRI interface. You are going to
have to configure * to use a standard T1 not a PRI D4/AMI is the correct
signaling.
- Original Message -
From: Scott Wolfe [EMAIL PROTECTED]
I'm sitting here with my dunce cap on. My weak excuse is that I haven't
ever played with g729 before.
I have a Sipura 841. I have the phone config set to use g729. Its
appropriate sip.conf entry, and the IAX stanza for my ITSP all set to
disallow=all, allow=g729.
But as soon as I dial, I
http://www.qwest.com/largebusiness/products/voice/callingcards/lb_dial_guide.html
based on that info, i'd say you are about to have a very crappy day. G
sorry to reply to my own post, forgot to suggest trying to send calls
over another network.
http://www.thedigest.com/faq/picodes.html
Greg Boehnlein wrote:
I've got Polycom and Cisco phones and quite frankly, I prefer the Polycom
Soundpoint IP-500 and 600 to my Cisco's now. All things being equal
between the phones, the following are why I prefer the Polycoms:
1. Better speakerphone than the Cisco 7960s. Despite the fact that
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian
Capouch
Sent: Sunday, April 24, 2005 3:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] g729 passthrough?
I'm sitting here with my dunce cap on. My weak
jltaylor wrote:
;;;
Brian,
Add to the [general] section in sip.conf the following:
disallow=all
allow=g729
allow=ulaw
allow=alaw
For some reason Asterisk will not pass audio through itself without trying
to transcode unless you have this in your config.
Don't ask me why it will
TE101P card T1 EM trunk to telco
on a SIP-PSTN call, after dial
SIP phone hears two seconds busy tone (1) then ring tone
how do we get rid of busy tone?
(1) two second busy
(480+620/500 0/500 480+620/500 0/500)
---
extensions.conf:
;
; dial-out to
If Feature Group B signaling is working properly (and you have Feature Group
B trunks), then
to reach your Asterisk box you would dial from the PSTN (1)+950+WXXX {W is 1
or 0 based on the number assigned to you}.
If you are dialing out {terminating where you look like the carrier} on
FGB then it
I upgraded my FC3 to kernel 2.6.11. I installed bristuff 0.2.0-RC8 and I
cannot call out using zaphfc. I can receive calls, but can't get out. Here
is what I got:
-- Starting simple switch on 'Zap/4-1'
-- Executing Dial(Zap/4-1, Zap/1/**348|60|rTt) in new stack
[ 00 e7 0e 12 08 01
On 20 Apr 2005, at 17:12, Moody wrote:
Signate offers an interesting product they call 'webcall', which
basically contacts a client at a number they provide then connects
that person to a sales staff. Some potential for abuse but a nice idea
for support etc.
I know that it is possible to do
On Sun, Apr 24, 2005 at 01:15:06PM +0200, Thore wrote:
Hi !
What is the easiest esyest way for implementation of ztdummy on a Debian
(testing) system?
Thore
On testing/unstable you basically:
apt-get install module-assistant
m-a a-i zaptel
to build a zaptel package for your kernel.
On Sun, Apr 24, 2005 at 06:45:15AM +0200, Remco Barende wrote:
When using bristuff I do get an error too if I don't load zaptel first but
not with the tdm driver.
I know that in my modprobe.conf it is specified that ztcfg should be run
after loading the module but why doesn't it?
Hi
We have a fairly simple Asterisk setup: E1 card, 8 FXO lines connected
to old PBX, and some SIP phones, used by a callcenter with queues.
Almost all calls are incoming (through E1 line), answered by some
callcenter operator (using SIP phones, call assigned by queue app),
and in some cases, are
[EMAIL PROTECTED] wrote:
On 20 Apr 2005, at 17:12, Moody wrote:
Signate offers an interesting product they call 'webcall', which
basically contacts a client at a number they provide then connects
that person to a sales staff. Some potential for abuse but a nice idea
for support etc.
I know that it
I found if put AGI on zaptel channel, when execute stream file there is no
voice and execute set callerid got no effect.
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Bookmark this page .. It has saved me more than once in dealing with pages
with different languages ..
http://babelfish.altavista.com/babelfish/
BRW
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Wiplinger
Sent: Sunday, April 24, 2005
I need some serious help!! I have been in the process of building an
Asterisk system to replace a Cisco Call Manager. I have most everything
setup, but only got to test the PRI today. To make a long story short,
my Call Manager is half broken and I need to go live with * a lot sooner
than I
Hello-
I made some adjustments to the Ast-Tapi to do a similar thing on my
site. It was a very easy modification. Here is a sample running on our
demo server. I would appreciate it if people don't just try it though
-- since the calls are routed to my sales staff who I pay per call...
heh.
Mark Johnson wrote:
I need some serious help!! I have been in the process of building an
Here are my interrupts:
cat /proc/interrupts
CPU0
0: 960018 XT-PIC timer
1: 4 XT-PIC keyboard
2: 0 XT-PIC cascade
3:9565339 XT-PIC
Normally, plain old PBX DID trunks are em_w (dtmf).
Strange, the only other problem might be the timing of the wink.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Ackley
Sent: Sunday, April 24, 2005 8:27 PM
To: Asterisk Users Mailing List -
List,
I have been using asterisk for a couple of weeks now, to support some
Cisco 7960 and 7920 phones, and have been enjoying the learning
experience. I have gotten the phone firmware upgraded, Broadvoice
connectivity, basic dial plan, and voice mail working. However I am
sure that there is
Michael Welter wrote:
Try 'vmstat 1'--are you getting 40% system utilization every n
seconds? If so, unload the wcfxo and wcfxs modules and test again.
I tested and I do in fact get from 40-50% system util every 5 seconds or
so. After removing the wctdm module, the system util drops to 0 and
i am trying to get G723 passthrough
get the same error.
how to configure passthrough for g723/g729 ?
On 4/24/05, Brian Capouch [EMAIL PROTECTED] wrote:
jltaylor wrote:
;;;
Brian,
Add to the [general] section in sip.conf the following:
disallow=all
Anybody having problems with cvs head?
I gave a problem with queues and agents... I have defined joinempty=no on
queues.conf and eventhough there are no agents logged in, the call are
getting queued.
Also, I have the following statements:
exten = s,9,GotoIfTime(00:00-11:59|*|*|*?10:12)
exten =
Ditto!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Simon
Sent: Sunday, April 24, 2005 7:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk best practices
List,
I have been
I would like to start a discussion about real and big voip providers.
Lets say for example voicepulse or vonage or any other.
What software do they use?
Is anybody using asterisk?
What kind of connections they have to the internet?
What kind of equpment is used by them?
I hope that this group
I figured out the problem with gotoiftime.. Still have the problem with the
queues though :(
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Domingo, 24 de Abril de 2005 09:45 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial
Mark Johnson wrote:
Michael Welter wrote:
Try 'vmstat 1'--are you getting 40% system utilization every n
seconds? If so, unload the wcfxo and wcfxs modules and test again.
I tested and I do in fact get from 40-50% system util every 5 seconds or
so. After removing the wctdm module, the system
Hi Everybody can someone tell me why I can hear audio? My call is to my
proxie which is directing it to my Asterisk box. The Voice mail is
playing but I think its playing to my proxie.
the phone is on 198.31.185.246:63257
Here is from the sip debug. Thanks
Sip read:
INVITE sip:[EMAIL
Some things i've wanted to look into is sphinx2 with asterisk. They
have EAGI demos but surely they don't work for me! How about voicemail
linked to your online activity (aim) http://ruk.ca/article/1832 .
Dream it and write it.
On 4/24/05, Brian Watters [EMAIL PROTECTED] wrote:
Ditto!
Asking someone to spend eleven times more money on a hardware purchase
($220 vs $20) as a gesture of good will is expecting a bit much, I'd say.
Certainly I can understand that Digium doesn't stand to make much money
selling X100Ps at $10 each, and I can certainly understand them choosing
to
Have you check on what price a nec ip pabx is going for lately?
Whilst I appreciate that digium should be selling their cards for less -
if there was no digium there would be no asterisk - therefore price of
clones x00p's is irrelevant.
Dean
-Original Message-
From: [EMAIL PROTECTED]
What year is this? 2005 right? Doesn't everyone on the planet know that you
get what you pay for these days? If you want to experiment with Asterisk
there is nothing wrong with using clone X100P cards at $6.95 a pop. If you
are putting in a production machine that is mission critical to the
HI,all!
I have setted up an Asterisk server on my linux.My sip.conf and extensions.conf configurations like these:
sip.conf
[general]port=5060disallow=allallow=alawallow=ulawallow=gsmnat=yes
dtmfmode=rfc2833canreinvite=no
Hi all,
I am still unable to initiate a call transfer with the keypresses
defined in features.conf in a couple month old version of asterisk from
CVS HEAD.
Before I go ripping things apart, I was really wondering if this is by
design, or should it work on all my devices? I have an iaxy, phones
Hello,
that is even possible without MODEM hardware. It should work with a
simple call forwarder/diverter. It connects to both line ends and works
more or less like a analogue 2-port pbx with a fixed programmable
forwarding number. Offered e.g. in Germany from AUERSWALD (A-BOX)
Franz wrote:
Please contact me Urgent...
Hi Frantz,
I can do custom programming. Here is some information about my company:
http://ykoz.net/intl/
Let me know what you're after and I'll send you a preliminary quote.
Cheers,
Jean-Michel.
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