I think I heard that satalite first hops are usually in the 400ms
range. Are you prepared for half second or more delays?
Well. For all practilal reasons. NO.
But, apparently the C band is suitable for VOIP.
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to travel...
yup, normally latencies over a satellite link is
about 200/500 msec ... so you will notice that
on your calls, since max delay that human ear
won't notice is about 150 msec (of course, this
is an average...)
just my 2 cents
matteo.
Have you got tried above in real life
Joe Hughes wrote:
A friend of mine uses a 1024/512kbps satellite link here in the UK,
the latency is roughly 600ms.
Do you know what gateway hardware is he using?
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Dan wrote:
Hi,
It would be great if the IAX protocol will be able to tranfer fax
data (even converted in another format) between Asterisk boxes, using
low bandwidth codecs like GSM. I know that this is possible only with
the G.711 now (passing faxes using the audio stream), but maybe
Dan wrote:
Hi,
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 12, 2003 3:42 PM
Subject: RE: [Asterisk-Users] FAX, IAX and *Maybe I'm
dreaming...:-)
Dan wrote:
Hi,
It would be great if the IAX protocol
Stephen Wingfield wrote:
Dear ALL:
Following Mark's request: if all those who want to meet in Paris on
20th could please email me asap. Please confirm if there is a time
you would not be able to make it. I will then judge numbers and make
a posting/reply to you all on Sunday with time/venue
Patrick Cantwell wrote:
All,
If you currently own a Sipura SPA2000, avoid going to the sipura
website and upgrading the firmware. I upgraded my SPA2k a couple of
days ago from 1.0.9 (what it came with) to 1.0.18 off the site, and I
am having issues with my SPA rebooting itself every
You can!!! :)
Use one of those FXS to FXO converters found at eBay, connect X100P to
the FXO port and you all setup.
Ta
SJ
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Well, I just placed an order with BT. As soon I get the gear I will let
you know.
Ta
SJ
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Why would you need one of those? Those are designed for plugging a
voip line into a POTS line or such so that you can dial a number, hit
the converter, and pass your call out the VoIP line. All you need to
do is plug the ata186 into your X100P. No extra hardwarerequired. -Pat
Steven Critchfield wrote:
On Tue, 2003-12-09 at 12:28, Senad Jordanovic wrote:
Why would you need one of those? Those are designed for plugging a
voip line into a POTS line or such so that you can dial a number,
hit the converter, and pass your call out the VoIP line. All you
need to do
Michael Rowley wrote:
Hey,
Here is a quesion for you. I am still battling with the phone system
for my new buisiness.
6 incoming lines, 1 fax, DSL. 8 phones max, will provably start with
5
to save money.
I was thinking of using Asterisk, but having difficulty finding
appropriate
Miguel Cavazos wrote:
On Fri, 2003-12-05 at 07:13, Steven Critchfield wrote:
During Phreaknic, Mark was showing off a Xbox running asterisk with 4
S100U interfaces connected to the game ports on the front. It was
interesting. In the end, I don't think it is cost effective as a real
PC since
Nick Bachmann wrote:
Hello
I have couple of Grandstream phone and some of them after a day or
two just stops receiving calls, you can still make a call from that
phone but you cannot receive calls until you restart the phone. Is
it a wrong configuration of phone or Asterisk ? Thanks for any
Michael Bielicki wrote:
You mean on US or EU dialup, but I doubt you will get any success on
far
east dialup or african dialup with that. Here you would either need
speex or g723.1.
Mark Spencer wrote:
You can definitely do that with GSM and G.729 when running IAX /
IAX2.
Mark
Title: Message
just
need to buy g729 licences from www.digium.com, install it and off you go.
:)
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd
WallaceSent: Tuesday, December 02, 2003 6:37 PMTo:
[EMAIL PROTECTED]Subject:
Mark Spencer wrote:
I'm coming to Paris Dec 19. I was wondering if there was any
interest in having an Asterisk get together in Paris sometime near
there. Any one out there interested? Anyone in Paris who could help
organize something like that? :)
Mark
Anton Yurchenko wrote:
Hello,
is there a way to disable call waiting in sip? I`m using grandstream
101 and even when the phone is in use I hear ringing in the headset.
It is pretty annoying , is there some way to disable this? I cant find
anything like it in the grandstream docs.
Thanks
Andrew Gillham wrote:
Senad Jordanovic wrote:
Hi,
Just received recently released Grandstream handytone 286 ATA for
testing.
I can call ATA from any other extensions and conversations seems to
be of quite good quality. However placing calls from ATA is not
possible at all to any
Billy Huddleston wrote:
change dtmf to info on both * and in the handytone.
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, November 25, 2003 8:01 PM
Subject: [Asterisk-Users] Handytone 286 - calling out
Hi,
Just received
Steven Critchfield wrote:
On Wed, 2003-11-26 at 04:27, Asterisk wrote:
Hello Asterisk-ers,
Thanks to WipeOut! you've kinda answered something I wondered about.
I've been looking for a post like yours for the last 3 hours (I
didn't want to get told off for not looking first)!
I don't have
Hi,
Just received recently released Grandstream handytone 286 ATA for
testing.
I can call ATA from any other extensions and conversations seems to be
of quite good quality. However placing calls from ATA is not possible at
all to any extensions.
After dialing there no indications of any kind
Does * supports time zone setting per EACH user/device?
Ta
SJ
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www.loligo.com/asterisk/
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tddy
Sent: Thursday, November 20, 2003
5:01 AM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Help
configuring CISCO 7960
Yes, I went to cisco site but I
could
I am curious as well if UK caller ID will be supported.
Anyone else out there with the same requirement?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Boardman
Sent: Thursday, November 20, 2003 11:06 AM
To: [EMAIL PROTECTED]
Subject: Re:
Good idea :)
Also, Oftel is planning 055. code specifically to be used for VOIP...
Ta
SJ
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Does this implementation of H323 for * terminates and originates calls
successfully from Cisco 5300?
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If * is configured to write call records data to a remote server, and
remote server becomes unavailable for some reason!
Will * continue to try to write missing records or will it give up or
ELSE?
Ta
SJ
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Funny, I am doing the same at the moment... :)
We are allowing * to dump call records onto a remote database server.
Once there we can do all sort of things with it.
My only concern is if this remote server goes down! What happens to the
call records which were not written to remote server? The
AGI has set autohangap
command. That should do it.
Ta
SJ
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak
Sent: Thursday, November 13, 2003
12:20 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Limit
timeout of outgoing
You need to add # before dialing the rest of numbers.
Ta
SJ
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Sure, I will be happy to test it for you...
Please let me know more details.
Ta
SJ
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1.
install h323 support on your *. All docs are in
/usr/src/asterisk/channels/h323. You MUST follow the instructions to the
letter.
2.
configure Multitech MV110 FXO box to send/receive calls from *
3.
configure * (in h323.conf) to send/receive calls to Multitech MV110 FXO
box
Ta
SJ
Look, at the codecs
compatibility between the phones and canreinvite=X
in your sip.conf
Ta
Senad
I made few successful calls (in/out).
However, the application did crash few times during conversation, and
now while trying to start it the application shows this error message:
Run Time error '341':
Invalid control array index
I am using XP-PRO, Service pack 1.
Did you reload after you made the change?
dave
Yes, many times.
BTW, The servermail variable works fine.
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AFAIK these only work with voicemail2.. check your extensions.conf and
make sure you are using voicemail2 and not just voicemail..
Yap, that did it. :)
Ta
Senad
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Finaly, someone has started the IAX soft phone ball :)
Thanks, Dan...
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Does any one know what below means?
-
DEBUG[6151]: File chan_sip.c, Line 4904 (handle_request): Check for res
for 2298
DEBUG[6151]: File chan_sip.c, Line 973 (find_user): Call from user
'2298' is 1 out of 0
Yes, we have!
Nice device, works fine on public IP but behind
NAT it has problems.
PCphoneline are sorting out NAT problems as far as I know.
Ta
Senad
Found the answer.
It was not codec, but instead missing [ in local context.
Ta
Senad
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Look at RTP (/etc/asterisk/rtp.conf) packets, and its firewall configuration.
Title: Leterhead
Sip.conf
[general]
register=FWDNUMBER:[EMAIL PROTECTED]/EXTENSION
[fwd]
type=friend
username=FWDNUMBER
secret=FWDPASSWORD
host=fwd.pulver.com
context=YOURINBOUNDCONTEXT
extensions.conf
[inboundcontext]
exten = EXTENSION,1,Dial(SIP/SOMEOTHEREXT)
If you are load balancing with a director that spreads the load
across
multiple servers, the first problem would be sharing the SIP
registration information between the two or more servers..
This is so that if UA1 is registered on Server1 and UA2 is registered
on
Server2.. Then when a call is
Hi,
Try this:
http://www.loligo.com/asterisk/
Ta
Senad
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I prefer 7960.
Works out of the box with no problems, it
is sexy and after all it is Cisco gear.
Ta
Senad
Hi,
I would love to participate in your test.
We have several * machines.
Please let me know further details.
Ta
Senad
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hi,
You need to upgrade SIP image for the phone using TFTP server.
All image(s) need to be uploaded to TFTP server, from where 7960 will
download
SIP image and conf files.
here is more info:
http://www.cisco.com/warp/public/788/voip/handset_to_sip.html#intro
Ta
Senad
-Original
Hi,
Apart from cisco site, does anyone knows where the SIP images
can be downloaded from?
Ta
Senad
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Hi,
Has anyone managed to get www.pcphoneline.com USB devices (FXS and Phone)
devices working with *.
It registers and works with an ease with FWD, but with *, it does not even
try to register.
I tried all sort of configuration settings and the PC I am on is using a
public IP.
It does not work
try to read this:
http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html
senad
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I am not a coder hence this question:
If a web interface (similar to vonage account management) gets produced
using PHP/MYSQL to administer
*, does that require licence from Digium if the code is not open source.
Thanks...
Senad
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great, thanks for that.
What if interface triggers CLI commands?
Senad
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* has application called Record.
type show appliaction record for more info at CLI prompt.
Senad
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yes, that is correct.
I just realised that myself.
Thanks
Senad
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Is it not possible to bundle all of the TCP/UDP traffic in
a CIPE tunnel (or similar).
has anyone tried it? what were the results?
senad
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look
into /usr/src/asterisk/channels/h323
and
FOLLOW instructions EXACTLY as in README.
Senad
Hi,
I have enabled ztdummy in order to have * compile it.
I can modprobe ztdummy with no problems.
The sole reason, i need ztdummy is to heve musiconhold and meetme working.
However when I start *, it says this and does not start.
Hi,
I have enabled ztdummy in order to have * compile it.
I can modprobe ztdummy with no problems.
The sole reason, i need ztdummy is to heve musiconhold and meetme working.
However when I start *, it says this and does not start.
Hi,
I have enabled ztdummy in order to have * compile it.
I can modprobe ztdummy with no problems.
The sole reason, i need ztdummy is to heve musiconhold and meetme working.
However when I start *, it says this and does not start.
yes, I do
but was it not supposed to be specified?
Senad
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hmm.. sorry for double post of the same issue.
I had mail returned, hence I thought it did not get through.
Senad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Senad
Jordanovic
Sent: 25 September 2003 21:35
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users
here is one phone company listing some of the codes...
i hope this will help you...
www.onetel.co.uk
senad
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We had calls dropping every few mins but after we have upgraded ATA's
firmware to 2.16
the problem was solved.
Senad
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(IT WORKS HERE! fix yours) etc...
- wasim
On Thu, 18 Sep 2003, Senad Jordanovic wrote:
well, i have same problem...
it sounds like nufone is not allowing calling of #800.
anyone from nufone care to comment?
Senad
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hmm.. this is weird...
I can call all other numbers except #800...
and I can not see anything wrong in my conf files.
here they are, if you find something, please shout...
iax.conf
---
[general]
port=5036
bandwidth=high
Deny=all
allow=iLBC
register= bicomus:[EMAIL PROTECTED]
John, Tx
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I did not read the comment made by Alistair hence why I am replying to it
now.
And thanks, to PJ Welsh for bringing it up. Your points are true, valid and
I am sure most people will agree with you. (Even the old timers where
newbies at some stage)!
Full VOIP understanding takes time. There are
I found that site very useful as well, but is very slow.
The webmaster of that site...!!!
I can provide FREE hosting for that site and it should be much faster. (
Another two cents from me)
Please do get in touch if interested.
(Web hosting is something I do not need spoon feeding for CERTAIN.
have you more info on this free phone offer? please send it to me off the
lest?
senad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Koehler
Sent: 15 September 2003 23:08
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Grandstream Source?
You
Hi,
What are real life bandwith stats for * supported codecs?
Is it true one can run 6-32 conversations over DSL, as stated in this list?
Senad
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Hi,
Anyone had ATA 186 calls stopping for some strange reason on EVERY call made
between them.
My setup is like this:
ATA1 * ATA2
ATA1 is behind NAT, ATA2 is on a Public IP.
My SIP.conf is:
-
[2201]
type=friend
username=2201
secret=
host=dynamic
mailbox=2201
cheers,
and Thanks
Senad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tilghman
Lesher
Sent: 13 September 2003 23:41
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] h323 v oh323
On Saturday 13 September 2003 17:31, Senad Jordanovic wrote:
Any
www.telappliant.co.uk
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Angel Gabriel
Sent: 13 September 2003 13:02
To: * Users
Subject: [Asterisk-Users] UK Suppliers
Can anyone please direct me to UK based suppliers of equipment. Website
URL's would be
asterisk and let it sit there hanging
for a few minutes. Stop Ctrl-C in the terminal session running tcpdump
and send me the file foo that was created by tcpdump. You might want
to gzip it if it's large and send it to me off list.
-Original Message-
From: Senad Jordanovic [mailto:[EMAIL
I am not 100% sure but I think a minimum of full duplex sound card is
required.
Anyone else knows exactly?
Senad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Angel Gabriel
Sent: 13 September 2003 13:22
To: * Users
Subject: [Asterisk-Users] How to test
do you want/need a diguim card or a third party?
senad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Angel Gabriel
Sent: 12 September 2003 10:10
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] UK based guy, wants card for machine.
I guess the subject
Any knows a link where to download:
Open H323 v1.11.7
and
PWLib v1.4.11
Thanks
Senad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jeremy
McNamara
Sent: 12 September 2003 20:04
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] h323 v oh323
I have been reading archive post in regards to h323 support, and I am not
clear on this:
1.
Is h323 enabled and ready to use if one compiles asterisk, zaptel and libpri
(as shown on asterisk.org) ?
2.
If it is, is it h323 or oh323?
3.
If it is not, does one just need to follow instructions in
tarballs from openh323.org
Follow Jeremy's instructions in the /asterisk/channels/h323/ directory
EXACTLY!
good luck
Regards,
Sean Langley, P.Eng
Firmware Engineer
General Dynamics Canada
(403)730-1482
[EMAIL PROTECTED]
-Original Message-
From: Senad Jordanovic [mailto:[EMAIL
I am really desperate to have any help on this problem below
as it prevents us from making any further progress.
Is there anyone out there who can help?
Thanks
Senad
-
Hi,
Allowing registration to iconnect by using
of sip.conf. Also, are you using the latest CVS release of *?
-Original Message-
From: Senad Jordanovic [mailto:[EMAIL PROTECTED]
Sent: Thursday, September 11, 2003 1:52 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Segmentation fault due to SIP registration
NUMBER 2
I am really desperate
Hi,
Allowing registration to iconnect by using
register = XXX in sip.conf file does not start *,
but instead produces segmentation fault and * hangs.
Commenting out register = allows * to start with no
problems.
Anyone know why is this happening?
here is program I would recommend:
www.helpandmanual.com
really easy to use, full of the features.
Senad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Leif Madsen
Sent: 06 September 2003 17:02
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] OT:
Hi,
Just got hold of Ericsson webswitch 100 G4 (4 FXO ports).
IT uses H323 as codec. The plan is to use it for incoming/outgoing calls on
two PSTN lines.
I have ATA 186 which is using SIP to use asterisk services.
I can not figure out:
1. where in asterisk do I edit conf files so it uses
hi, what does tr means at the end of line?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew
Gillham
Sent: 05 September 2003 06:29
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk and Cisco 7960
Andrew Joakimsen wrote:
exten =
PROTECTED] Behalf Of Steven
Critchfield
Sent: 05 September 2003 20:27
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk and Cisco 7960
On Fri, 2003-09-05 at 14:00, Senad Jordanovic wrote:
hi, what does tr means at the end of line?
There is documentation, it is even within quick access
have
you looked at digiums site? there are few simple sample
there.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Adelino
BaenaSent: 09 August 2003 21:47To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] H323 and
SIP
Dear
:[EMAIL PROTECTED] On Behalf Of
Senad Jordanovic
Sent: Friday, August 08, 2003 1:14 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] X-Lite - No sound + chan_sip issue
Hi,
X-Lite logs into * with no problems. I dial 1000 and *
plays greeting, but
i can not hear it.
Tried many
I agree as well... phpbb is much better solution.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of James Taylor
Sent: 09 August 2003 15:08
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] list proposal
I'd still vote for phpbb.
Then we could have
the allow=all in sip.conf to allow=alaw and see if
that works.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Senad Jordanovic
Sent: Friday, August 08, 2003 1:14 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] X-Lite - No sound
+ chan_sip issue
Change the allow=all in sip.conf to allow=alaw and see if that works.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Senad Jordanovic
Sent: Friday, August 08, 2003 1:14 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] X-Lite
Hi,
X-Lite logs into * with no problems. I dial 1000 and * plays greeting, but
i can not hear it.
Tried many times with the same result.
After quite few tries * complains about:
-
WARNING[81926]: File chan_sip.c, Line 388 (retrans_pkt): Maximum
James,
Thanks for that!
Do you have a list of those US AIX providers?
Also, is there anyone in UK providing
termination/origination on AIX?
Senad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of James Sharp
Sent: 04 August 2003 18:59
To: [EMAIL PROTECTED]
Hi,
Has anyone used http://www.pcphoneline.com/ products with asterisk?
Senad J
attachment: winmail.dat
, 28 Jul 2003, Senad Jordanovic wrote:
MessageHi,
I know of a person who would probably write the docs for FREE.
He is British with a superb understanding of English language.
Anyone in digium interested, please let me know!
Senad Jordanovic
-Original Message-
From: Damian Flynn
Title: Message
Hi,
I know
of a person who would probably write the docs for FREE.
He is
British with a superb understanding of English language.
Anyone
in digium interested, please let me know!
Senad
Jordanovic
-Original Message-From: Damian Flynn
[mailto:[EMAIL PROTECTED
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