Eric C. wrote:
Hello all,
Just starting to setup asterisk v 1.4.11 and need to run three distinct phone
systems for three different companies.
So far, I have inbound lines going to the appropriate dial plan within the
extensions.conf file. I'm using
exten = _X.,1,NoOp(FROM NUMBER
Hello all,
Just starting to setup asterisk v 1.4.11 and need to run three distinct phone
systems for three different companies.
So far, I have inbound lines going to the appropriate dial plan within the
extensions.conf file. I'm using
exten = _X.,1,NoOp(FROM NUMBER:
0 my.public.ip.address:10078
0.0.0.0:* udp0 0
0.0.0.0:10079 0.0.0.0:*
[3]
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
my.wan.ip.address:5060;branch=z9hG4bK-3c9c7b77;received=my.wan.ip.address;rport=5060
From: Eric C
Eric C. Snowdeal III wrote:
after registering the phones correctly and receiving a 200 o.k.
message i can connect to other registered softphones and pstn
endpoints [ via an voicepulse account ], but after making the initial
connection, i can't hear any sound and i get disconnected after
Holger Schurig wrote:
i'm new to asterisk and am having trouble placing outbound calls. i
Bug Grandstream so that they finally fix their buggy software.
The GS phone sends occassional SIP packets to port 0, not to port 5060, as
tcpdump or (better) ethereal will show you.
There's a page on
i'm new to asterisk and am having trouble placing outbound calls. i
know this topic has been discussed ad nauseum in the past [1] , but i
can't seem to find a workaround and i'm wondering if my newbie-ness is
getting the best of me.
after registering the phones correctly and receiving a 200
