On Sun, Jun 06, 2004 at 04:25:32PM -0400, Tim Sailer wrote:
On Tue, Jun 01, 2004 at 07:49:29PM -0500, Yelson Vivas wrote:
Hi everybody
i have a problem trying to connect an incomming phone call from pstn to my
(soft phone) iaxcomm, the phone rings but when i try to answer the call,
On Wed, Jun 02, 2004 at 08:14:44AM +0100, gARetH baBB wrote:
On Wed, 2 Jun 2004, Adam Hart wrote:
Can I recommend you label files with version numbering - this must be
about the third ? fourth ? firefly-thirdparty you've released.
.. but have firefly-thirdparty.exe be a symbolic link to
Hi,
I have two softphones connected to an Asterisk stable. I have two
extensions, say 1000 and 2000. When 1000 calls 2000, the call cannot be
completed; the softphone (either Diax97a , SJphone, Firefly 1.8) on
extension 2000 will ring, but as soon as the call is picked up, extension
2000 will
On Wed, Jun 02, 2004 at 11:25:26AM +0200, Tor Houghton wrote:
Hi,
I have two softphones connected to an Asterisk stable. I have two
extensions, say 1000 and 2000. When 1000 calls 2000, the call cannot be
completed; the softphone (either Diax97a , SJphone, Firefly 1.8) on
extension 2000
On Wed, Jun 02, 2004 at 05:37:48PM -0300, [EMAIL PROTECTED] wrote:
I donwloaded two IAX Clients (firefly and IAX phone) and they did register
with *. It would make authenticated calls, but wouldn't actually register
with the
server.
[snip]
qualify=1000
i have found that firefly, diax
Are there any softphone clients that can use IAX/IAX2 for MacOS X?
Regards,
Tor
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
On Thu, May 20, 2004 at 07:01:09PM +0300, Dan wrote:
:-)
Dan
P.S. You can really decode DTMF tones with your ear/brain?..:-)
not far off, but i mostly use the feedback to double check that i didn't
dial a number wrong.
tor
___
Asterisk-Users
On Wed, May 19, 2004 at 09:16:02AM +0300, Dan wrote:
A new version with some cool features (not available on any other soft
phone) will be available at the end of the week.
Send me a mail if you need further assistance.
Looks promising -- one request (I'm sure there will be more); how
On Thu, May 20, 2004 at 12:38:34AM +0300, Dan wrote:
Hi Tor,
What do you mean by DTMF feedback?
When you hit a key, make DIAX play back the corresponding DTMF tone to you.
You can enable the key beep in DIAX, but what's the reason to get a DTMF
type of feedback?
The beep is not enough?
Hi,
I upgraded my local Asterisk (the last version was quite old), and since
then, whenever anyone tries to call me via SIP/IAX thru my external
Asterisk, they get 403 Forbidden as soon as I pick up.
I have no trouble picking up when someone calls via PSTN.
Basically, my phone (Firefly
This has been mentioned before on this list, but in order for md5.c to
compile successfully (OpenBSD 3.3), the following must change in md5.c:
#if defined( __FreeBSD__ ) || defined( __OpenBSD__ )
# include sys/endian.h
Change this to be:
#if defined( __FreeBSD__ ) ||
On Wed, Apr 14, 2004 at 02:52:16PM -0400, James Moran wrote:
Anyone have any suggestions on free sip phone software for windows??
Only have one IP phone and want to have one other computer hooked up to
my Asterisk box for testing.
Have you tried Firefly?
Hi,
I just upgraded to the recent CVS, and IAX1 no longer seems to be available.
Is there a way to reenable it?
Tor
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
On Tue, Apr 13, 2004 at 02:56:48PM -0400, Jeremy McNamara wrote:
Use IAX2, it is a better IAX protocol.
Jeremy McNamara
P.S. If you really must have it, dig thru the channels/Makefile, but
there is zero reason to use it any longer.
Well, I use IAX1 between the clients on the
On Tue, Apr 13, 2004 at 04:58:19PM -0400, James Golovich wrote:
# If you really want IAX1 uncomment the following, but it is
# unmaintained
#
#CHANNEL_LIBS+=chan_iax.so
Thanks all, I'll move to IAX2 after I've tested the notransfer option.
Tor
On Sat, Mar 13, 2004 at 11:20:32AM +1100, Duane wrote:
Tor Houghton wrote:
PHONES1=IAX/[EMAIL PROTECTED]
Did you try IAX2/[EMAIL PROTECTED] ?
Erm, no.
Haha, I cannot believe I spent days trying to fix that.
It works!
My internal asterisk took the call! Yay!
Thanks!
Tor
On Sat, Mar 13, 2004 at 11:20:32AM +1100, Duane wrote:
Tor Houghton wrote:
PHONES1=IAX/[EMAIL PROTECTED]
Did you try IAX2/[EMAIL PROTECTED] ?
Actually, I think I found the culprit. It seems (ho hum), that the IAX
softphone re-registered (reinvited?) with the external IAX server, so
Hi,
I'm having a bit of a problem. I have two Asterisk servers, one serving SIP
clients on the outside of a NAT, the other on the inside. The internal one
also serves PSTN and IAX clients.
When I call someone (who is on SIP) from any phone registered with the
internal Asterisk, I get through to
Hi,
I've applied the OpenBSD patches as noted on
http://www.voip-info.org/tiki-index.php?page=Asterisk%20OpenBSD%20patch
but there are a few files that still need changing with the current CVS.
I've collected them all here (including the ones from the wiki):
On Thu, Mar 11, 2004 at 11:42:05AM -0600, Tilghman Lesher wrote:
On Thursday 11 March 2004 07:45, Tor Houghton wrote:
Of course, I hope these make it into the tree so that OpenBSD users
don't have to manually patch + search in future.. :-
Anything you hope makes it into the tree should
On Thu, Mar 04, 2004 at 02:06:52PM -, Jon Shamash wrote:
[snip]
it should be
exten = 66,1,Dial(SIP/66)
Incidentally, is there a difference between = and =, or are both allowed?
Tor
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi,
I've got a question or two about SIP calling channels. As I understand,
there is no facility for Asterisk to make outbound calls as if it were a SIP
proxy.
As I understand it, it is not possible to add an extention that simply
states if no match so far, try SIP/url (from what
Hi,
So it's like this. I've had siproxd working for me on an external host to
which I've established a tunnel (my SIP client is behind a NAT gateway).
Of course, I've got to have mailbox functionality at the very least, so a
friend of mine told me about Asterisk, which I grabbed from the CVS and
23 matches
Mail list logo
