Displaying 7 results from an estimated 7 matches for "libretel".
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libertel
2005 Jan 06
0
re: asterisk and libretel
hi list,
is anyone succesfully using asterisk with libretel port-of-call
(www.libretel.com)? If so, i would be grateful for configs..i set up
libretel to forward to mynumber@myserver.com:5070 (asterisk is running
on 5070 and SER on 5060) and when i call the number i see SIP messages
with ngrep but the asterisk CLI doesn't seem to catch them. I assume i...
2004 Jan 25
2
Incoming SIP matching
Incoming FWD calls from other FWD users, iaxtel, or via ipkall, need to
have dtmfmode=rfc2833. However, incoming FWD calls from the dialup
access numbers (such as libretel) need to have dtmfmode=inband. To
solve this problem, I created a second FWD account and configured
sip.conf as follows, in order to match the incoming number to the proper
dtmfmode:
[fwd-rfc]
type=friend
secret=*****
host=dynamic
dtmfmode=rfc2833
username=76153
callerid="CLARK THOMAS B&q...
2005 May 07
0
Getting DTMF to work with SIP?
Folks, from googling, I see that the dtmfmode
parameter is not valid in the [general] context.
My problem is that my overseas DID through Libretel
seems to want to come into the [general] context!
And, having done that, I get my welcome message, but
then the DID does not accept the DTMF when I try to
dial an extension! It plays the welcome message,
waits, and then times out (and hangs up nicely, yes).
I've actually tried setting the dtm...
2005 May 12
0
Asterisk, SIP and NAT: Help needed!
I've been googling and talking with Libretel about my
setup and the fact that incoming calls to my asterisk
box through the Libretel number reach my box (I hear
the greeting being played) but then don't accept DTMF.
Here is a rough diagram of my setup:
Asterisk |
server | NAT <------------ Libretel
| ro...
2003 Nov 07
0
sipdtmfmode problem
Greetings. I'm having a bit of a problem using the sipdtmfmode app. I have two
incoming paths to * from pstn via FWD that use differing dtmfmode. IPKall
wants rfc2833, libretel wants inband. If I set dtmfmode= in the fwd peer
config in sip.conf each works seperately, and I'm trying to use gotoif and
sipdtmfmode to switch based on the CID calling. Output seems to indicate
sipdtmfmode runs and does what it's supposed to, but it doesn't actually work.
I've...
2005 May 15
2
Voip Provider in Brazil
Hi all,
Is there a VOIP provider that can deliver local Rio de Janeiro numbers?
I am looking for a normal Rio number for my Asterisk box.
Thanks
Johannes
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2004 Jan 26
0
Digium FXO Card
...gt; Sent: Sunday, January 25, 2004 3:01 AM
> Subject: [Asterisk-Users] Incoming SIP matching
>
>
> > Incoming FWD calls from other FWD users, iaxtel, or via ipkall, need to
> > have dtmfmode=rfc2833. However, incoming FWD calls from the dialup
> > access numbers (such as libretel) need to have dtmfmode=inband. To
> > solve this problem, I created a second FWD account and configured
> > sip.conf as follows, in order to match the incoming number to the proper
> > dtmfmode:
> >
> > [fwd-rfc]
> > type=friend
> > secret=*****
> >...