Sales
2004-Jul-24 11:03 UTC
[Asterisk-Users] Please help I fear I have missed something very important! but what?
Sorry about this, I have been struggling with the basics of my asterisk
config.
I set up two sip peers and two phones. And I set up lots of dial masks
for outgoing calls, all my outgoing calls were working great, however
incoming calls were a different matter altogether, I cannot get incoming
calls to work. So I have gone back to a very basic FWD config, with one
phone which as far as I am aware should work, but doesn't. I cannot find
info on how to fix this.
Below is my sip.conf
[general]
port = 5060
bindaddr = xxx.xxx.xxx.xxx
context = sip
register => 2xxxx:xxxx@fwd.pulver.com/1001
[fwd]
type=friend
secret=xxxxxx
username=xxxxxx
host=fwd.pulver.com
;
;
[1001]
type=friend
username=xxxxxx
host=dynamic
secret=xxxxxxx
callerid=Home <1001>
dtmfmode=RFC2833
mailbox=1001
context=sip
and here is my extensions.conf:
[general]
static=yes
writeprotect=no
;
[globals]
HOME=SIP/1001
;
[sip]
exten => 1001,2,Dial(SIP/1001,20,t)
include => fwdnet
;
[fwdnet]
exten => _8.,1,Dial,SIP/${EXTEN:1}@fwd,t
Now as I said I can call out no probs by dialing 8 then the FWD number,
but incoming calls don't work, and as far as I can see that should ring
ext 1001 for 20 secs.
Could someone please help a complete Linux/Asterisk Newb, as apart from
this I have learnt a hell of a lot. But it's the last thing I need to
solve.
The linux box for this testing has a unfirewalled public IP address, so
there is no problems with NAT
Please please can someone help. If I have missed something important
then I aplogise, as I have been scouring the wiki and the archives to no
avail
Regards
Stuart Buchanan
------------------------------------------------------------------------
------
This email is only for the intended recipient. If you have received this
email
in error please notify the sender and delete the message immediately.
This email has been checked for viruses to ensure that any attachments
are free
from viruses. You should, however, carry out your own virus check before
opening any attachment. We accept no liability for loss or damage caused
by
software viruses.
------------------------------------------------------------------------
------
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20040724/532544d6/attachment.htm
Elman Efendiyev
2004-Jul-24 11:52 UTC
[Asterisk-Users] Please help I fear I have missed something very important! but what?
Looks like you missed 's' extension for incoming calls
You need something like this in extensions.conf
exten => s,1,Answer
exten => s,2,Dial(SIP/1001,20,t)
See sample of extensions.conf in asterisk distribution (make samples if
you didn't install samples)
--
Sincerely,
Elman Efendiyev
elman@earlinvest.com
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Sales
Sent: Saturday, July 24, 2004 9:03 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Please help I fear I have missed something
very important! but what?
Sorry about this, I have been struggling with the basics of my asterisk
config.
I set up two sip peers and two phones. And I set up lots of dial masks
for outgoing calls, all my outgoing calls were working great, however
incoming calls were a different matter altogether, I cannot get incoming
calls to work. So I have gone back to a very basic FWD config, with one
phone which as far as I am aware should work, but doesn't. I cannot find
info on how to fix this.
Below is my sip.conf
[general]
port = 5060
bindaddr = xxx.xxx.xxx.xxx
context = sip
register => 2xxxx:xxxx@fwd.pulver.com/1001
[fwd]
type=friend
secret=xxxxxx
username=xxxxxx
host=fwd.pulver.com
;
;
[1001]
type=friend
username=xxxxxx
host=dynamic
secret=xxxxxxx
callerid=Home <1001>
dtmfmode=RFC2833
mailbox=1001
context=sip
and here is my extensions.conf:
[general]
static=yes
writeprotect=no
;
[globals]
HOME=SIP/1001
;
[sip]
exten => 1001,2,Dial(SIP/1001,20,t)
include => fwdnet
;
[fwdnet]
exten => _8.,1,Dial,SIP/${EXTEN:1}@fwd,t
Now as I said I can call out no probs by dialing 8 then the FWD number,
but incoming calls don't work, and as far as I can see that should ring
ext 1001 for 20 secs.
Could someone please help a complete Linux/Asterisk Newb, as apart from
this I have learnt a hell of a lot. But it's the last thing I need to
solve.
The linux box for this testing has a unfirewalled public IP address, so
there is no problems with NAT
Please please can someone help. If I have missed something important
then I aplogise, as I have been scouring the wiki and the archives to no
avail
Regards
Stuart Buchanan
------------------------------------------------------------------------
------
This email is only for the intended recipient. If you have received this
email
in error please notify the sender and delete the message immediately.
This email has been checked for viruses to ensure that any attachments
are free
from viruses. You should, however, carry out your own virus check before
opening any attachment. We accept no liability for loss or damage caused
by
software viruses.
------------------------------------------------------------------------
------