Displaying 3 results from an estimated 3 matches for "server1_abc".
2010 May 11
1
asterisk-users Digest, Vol 70, Issue 23
...ct: Re: [asterisk-users] Dialing a SIP Peer without using
register strin
To: asterisk-users at lists.digium.com
Message-ID: <hsarab$ok7$1 at dough.gmane.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Remove username and secret and use IP authentication on both side
[server1_abc]
type=peer
host=192.168.0.20
context=default
dtmfmode=rfc2833
canreinvite=yes - canreinvite with nat=yes is not working
insecure=invite,port
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
nat=yes
qualify=yes
[server2_abc]
type=peer
host=192.168.0.21
context=default
dtmfmode=rfc2833
canr...
2010 May 11
1
asterisk-users Digest, Vol 70, Issue 24
...P Peer without using
> register strin
> To: asterisk-users at lists.digium.com
> Message-ID: <hsarab$ok7$1 at dough.gmane.org>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Remove username and secret and use IP authentication on both side
>
> [server1_abc]
> type=peer
> host=192.168.0.20
> context=default
> dtmfmode=rfc2833
> canreinvite=yes - canreinvite with nat=yes is not working
> insecure=invite,port
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> allow=gsm
> nat=yes
> qualify=yes
>
>
>
&g...
2010 May 10
1
Dialing a SIP Peer without using register strin
Hi,
I am new to this list and this is first time i m posting here. please help
me out
currently I am working on dialing a sip peer on an asterisk server from 2nd
asterisk server. scenario is like this.
on my system i am using this peer in sip.conf.
[abc]
type=peer
username=abc
secret=mysecret
host=192.168.0.20
context=default
dtmfmode=rfc2833
;restrictcid=no
canreinvite=yes