Displaying 5 results from an estimated 5 matches for "gkorproxy".
2003 May 24
4
Free World Dialup behind NAT
...:5060
Left Click on the "SIP Proxy" button
and for
"Out Bound Proxy:" use: 192.246.69.247:5082
This is how you should configure an ATA:
# Set UseTftp to 0
# Set UID0 to your FWD account's FWD number (e.g. 16000).
# Set PWD0 to your FWD account's password.
# Set GkOrProxy to fwd.pulver.com
# Set UseSIP to 1
# Set SipPort to 5060
# Set SIPRegInterval to 3500
# Set SipRegOn to 1
# Set outbound proxy to 192.246.69.247:5082
# Press the Apply button at the bottom of the web page, then
# restart the ATA 186 by powering it off and then on again.
I'm pretty shur...
2003 Mar 06
1
NAT working outbound with Asterisk and ATA-186 phones
...]
type=friend
username=2410
secret=somepasswordhere
host=dynamic
context=intern
canreinvite=no
nat=1
On your Cisco ATA-186:
Set your IP address information as usual (use DHCP, or static,
whatever your site requires)
UID0: [your UID]
PWD0: [this UID's password]
UseSIP: 1
SIPRegInterval: 240
GkOrProxy: [ip address of your Asterisk server]
Gateway: [ip address of your Asterisk server]
ConnectMode: 0x00460400
OutBoundProxy: [ip address of your Asterisk server]
The ConnectMode flags are used in v2.14 and v2.15 to "re-register"
phones with the correct data. See
http://www.cisco.com/un...
2003 Dec 22
1
Fw: Questions and finding
I installed * to primarily test its voicemail feature. I installed it on a server WITHOUT any telco board (i.e., digium). Installation looks ok, however I am having problems.
MY SETUP:
2xATAs are configured to use * as GkorProxy
Asterisk is registered to my SER SIP/RTP Proxy
1.) First test
- ATA1 calls to ATA2. When voicemail starts playing, it just cuts-off after 5-10seconds (consistently).
- Solution: I have to reconfigure ATA to use OutboundProxy to be Asterisk IP.
- Am I doing the right thing?
2.) Second test:
- A...
2004 Jun 23
0
connecting to Iconnect here using asterisk
...searching the mailing lists and several sites but did not find an
answer.
My current configuration:
* Asterisk installed on the Gateway (Bound to internal network and to
Internet) (Not behind NAT).
* Several Cisco ATA186 adapters with SIP firmware (Behind NAT with asterisk
set as their "GkOrProxy").
Current State:
* I manage to place calls to my other internal phones.
* Asterisk does not register at Iconnecthere.
* I am not able to place calls through Iconnecthere.
Thanks,
Configuration files: (commented lines represent options I tried).
n= my Ic...
2003 Sep 11
2
SIP client<->NAT<->Asterisk<->NAT<->SIP client. only works with canreinvite=no.
Hi!
I have this configuration:
SIP client A <-> NAT box A (real external IP) <-> Asterisk server (real
IP) <-> (real external IP) NAT box B <-> SIP client B
The echo test form any of the clients to the asterisk server is working
just fine, even without canreinvite=no.
When I try to call from SIP client A to B, wihtout the canreinvite=no in
the sip.conf, the call