It looks like you have you * on public IP and your phones on private, most
likely behind NAT if so in your sip.conf under each [grandstreamX] you most
likely need: nat=yes
- Original Message -
From: Chandra [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 03, 2004 1:44
www.voip-info.org had the most visitors ever today (Jan 2).
Slowed down a little, but otherwise no problems.
The Tiki Wiki software is rather resource greedy, its on a dedicated server, but it
does LOTS of
database queries for each page it displays.
Jim
James H. Thompson
[EMAIL PROTECTED]
Hi
Ive been trying to get this to work for ages now, basicaly im trying
to do if ${woteva} = (nothing), or its none existenant then do
label 1, else label 2. for my last called function, so it will play a
different message if theres no last call in the system or it was
anonymous. ive tried
Hi!
[xlite1]
type=user
Make this [xlite1user]
Adjust your extension.conf accordingly.
[xlite1]
type=peer
Make this [xlite1peer]
Adjust your extension.conf accordingly.
The alternative is to merge both entries and use type=friend instead.
my grandstream is also not registering to *.
You
SW: Thanks a million for the statement that I only need these two files and
they can be just about empty !
David Carter: many thanks for those files which I will study
Rich Adamson: That is so re-assuring! That may sound odd but its realy
helpful to have the problems I am facing acknowledged and
In case anybody is trying to work out the currency I used - it's
actually British Pounds, but the £ sign isn't being handled by the
mailing list. I've noticed that the mailing list is also having
problems removing the HTML or Microsoft OLE email components, and is
constantly filling the list
On Sat, 2004-01-03 at 06:31, John Coll wrote:
SW: Thanks a million for the statement that I only need these two files and
they can be just about empty !
David Carter: many thanks for those files which I will study
Rich Adamson: That is so re-assuring! That may sound odd but its realy
Alastair,
You were correct. The program was generating the same call IDs for all the
INVITEs. In fact it is a small routine of just 2 lines. I checked all my
routines, except this one. I never expected a bug in such a small routine.
It taught me a good lesson.
Sorry, i was not able to post
Hi Olle!
I put something into trouble ticket (I guess you get this as email).
BTW 2.03f is available at http://snom.com/download/share.
Christian
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] Im Auftrag von Olle E. Johansson
Gesendet:
Alastair,
You were correct. The program was generating the same call IDs for all the
INVITEs. In fact it is a small routine of just 2 lines. I checked all my
routines, except this one. I never expected a bug in such a small routine.
It taught me a good lesson.
Sorry, i was not able to post
Steven - thanks for that. OK I will try and ask interesting and directed
questions :)
I appreciate the support from several people. Rich Adamson encouraged me to
hang in there so I've been back at the shell prompt and edited configuration
files down to the bare essentials and still get the same.
Hi John,
Try adding username=5702 and username=5703 to each of the configs in
sip.conf. I recall I had this problem with the Grandstreams.
-Original Message-
From: John Coll [mailto:[EMAIL PROTECTED]
Sent: Saturday, January 03, 2004 11:56 AM
To: [EMAIL PROTECTED]
Subject: RE:
On Sat, 2004-01-03 at 17:55, John Coll wrote:
;
; liza:/etc/asterisk/sip.conf
;
[general]
port = 5060
bindaddr = 0.0.0.0
externip = 10.0.1.198
[5702]
type=friend
host=dynamic
context=johnhome
reinvite=no
canreinvite=no
qualify=300
callerid=John workroom #1 5702
mailbox=5702
John,
Obviousely, this would not work. Look at my example before;
[5702] ==
type=friend
username=5702 ==
context=internal
dtmfmode=info
username and context should match.
Better get it working in a simple LAN first, why NAT, why voicemail ..
Go to basics :),
SW
From: John Coll
Hi!
You started out with a much too complex setup. Start small, test, and
then add things step by step - don't configure everything at once!
Asterisk and 2 Grandstrem phones on a LAN. The * box (liza) is RH9 and
happens to have a firewall connected to the outside but * and the SIP
phones are
Dave
You were right!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dave Cotton
Sent: 03 January 2004 17:19
To: Asterisk List
Subject: RE: [Asterisk-Users] Newbie - getting two local
phonestocommunicate would be a good start :)
On Sat, 2004-01-03 at
On Sat, 2004-01-03 at 18:59, John Coll wrote:
Dave
You were right!
In the words of that welsh comedian I know because I was there.
As others have said there's a steep learning curve for *, but as one
who's climbed just some of it, I can say it's worth it.
--
Dave Cotton [EMAIL
Dave
You were right
disallow=all
allow=ulaw
allow=alaw
gave me two-way voice! Whew! Thanks a million. I wonder if I really should
have found that for myself ... I've added it to the voip-info.org wiki
OK lets see if the next step is a bit easier :)
thanks again all
john
-Original
And why you have two different entries for the same object?
Posting two times the same questions with other data will not help to
resolve the issue more quickly...
- Original Message -
From: Glenn Dalgliesh [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Saturday,
Add this to ur sip.conf ..that would help u.
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
-B
And sip.conf contains this
[general]
port = 5060
bindaddr = 0.0.0.0
externip = 10.0.1.198
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0
Hi. I've noticed a problem with the expression parsing in Asterisk. If the variable
is not defined, I will get a parse error. Yeah, there are ways around it, but I would
think that it should return false if 0, null, or undefined. I would change it, but I
have no idea about bison and I only
Mr. West,
Sorry to burst your bubble, but that is not me. My
name is Barbara Simpson. Either you are lying or
someone is trying to remove any credibility from my
original post. I now stand by my original post with
more conviction than ever.
There were a lot of insightful replies. However,
What happens when you change the configuration of the GS phone to
send DTMF via SIP INFO?
Send via SIP, RTP or INLINE AUDIO.
Make sure you change your dtmfmode= in your sip.conf to match the mode
set on the phone..
Yes.. that solved it. I added dtmfmode=info to sip.conf and set SIP
Ah. I suppose this isn't you, either.
http://www.worldogl.com/view_clan_info.php?clanid=5363
On Saturday 03 January 2004 14:12, Me wrote:
Mr. West,
Sorry to burst your bubble, but that is not me. My
name is Barbara Simpson. Either you are lying or
someone is trying to remove any
I have set up my * box to provide free calling. You can access it by
dialing 1-700-945-2475. Once you hear the prompt dial 1 the area code
and number. I would also like to test some direct incoming IAX
connections from some other * boxes to see if I can terminate PSTN calls
that way. If you would
That was the beginning of the all female quake clan
girlz of destruction http://www.girlzgotgame.net/.
Annie and I joined a 2v2 ladder. Yes, that's me,
however, that was nearly 3 years ago. Why bring this
up? What does this have to do with our discussion?
Are you just trying to prove your skills
On Saturday 03 January 2004 18:37, Me wrote:
That was the beginning of the all female quake clan
girlz of destruction http://www.girlzgotgame.net/.
Annie and I joined a 2v2 ladder. Yes, that's me,
however, that was nearly 3 years ago. Why bring this
up? What does this have to do with our
We just got 1 Nehemiah in the office. Performance-wise it's pretty much
a P3-class machine, IIRC the FPU is running at full clock speed compared
to the 800MHz version. We do have problem booting a 686 optimized kernel
on it. Can't install White Box Enterprise Linux (community distro based
on
On Sat, 2004-01-03 at 14:12, Me wrote:
Mr. West,
Sorry to burst your bubble, but that is not me. My
name is Barbara Simpson. Either you are lying or
someone is trying to remove any credibility from my
original post. I now stand by my original post with
more conviction than ever.
You
The £ came through here OK...
---
These optional licenses (which can also be purchase separately, and are
approx £10/$15) are to upgrade the number of users on the Cisco Call
Manager Platform.
---
Dan (in UK)
On Sat, 2004-01-03 at 13:34, Adthrawn wrote:
In case anybody is trying to work out
As far as your original post goes, Asterisk doesn't regularly segment fault.
There are many stable installations. We have a bunch of happy users. This is
remarkable since Asterisk is still a beta product.
There is plenty of useful information on the sites you panned if you are
smart enough to
Hi All,
Can we stop this thread pl. This lady has no
intentions to learn asterisk.
She is just a troll and wasting our time. With her
corporate attitude, what
she expects is support that available with paid
commercial products. Her
company has enough money to buy commercial products,
let she go
I now stand by my original post with
more conviction than ever.
There were a lot of insightful replies. However, none
of them were able to address the real problems of the
asterisk community and come up with solutions. If you
can't see your own faults, you are in for a bumpy
ride.
How
Paul Mahler wrote:
As far as your original post goes, Asterisk doesn't regularly segment fault.
There are many stable installations. We have a bunch of happy users. This is
remarkable since Asterisk is still a beta product.
888.480.4638, my toll-free number, is routed to wherever I choose to
Hello Everyone,
I just got my Dev Kit TDM today... :D
I installed the X100P ok (wcfxo); however, when I tried to 'modprobe wcfxs' for
the TDM400P(TDM10B), I got this error message:
/lib/modules/2.4.19/misc/wcfxs.o: init_module: No such device
Hint: insmod errors can be caused by incorrect
On Thu, 01 Jan 2004 17:50:32 +0100
Philipp von Klitzing [EMAIL PROTECTED] wrote:
Hi!
We needed the client browser to be open all the time for dynamic data to
load without the page refreshing. After looking at all of our options we
decided on programming it ourselves using flash
Tilghman,
Thanks for bringing this to my attention. I agree with the comment from
'siggi' - it seems that this should be configurable on a per handset basis,
not per voicemail user as currently implemented in the most recent patch
that has been hung on this bug. It's a useful hack that may help
Thilo,
I wasn't too sure about the packet based prioritization so I stuck with
the physical port based model. That is, I made port 1 high priority all
the time, then plugged my * sever into that port. Actually, that
segment has all the ip phones and nothing else.
The problem I had initally with
Please forgive me if this is a silly question. I've been following this
thread in the hope that I could put my * server and snom 200 into
full-time service very soon. I need to find out how to have the lines
configured so that it does not return a busy reply when only one call
instances is
On Saturday 03 January 2004 22:51, Darren Nickerson wrote:
On Friday, January 02, 2004. Tilghman Lesher wrote:
On Thursday 01 January 2004 12:57, Darren Nickerson wrote:
That worked a treat - thanks! Comedian Mail is now able to
download to the handset and there's a lot more
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