Displaying 3 results from an estimated 3 matches for "itsp1".
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itsp
2006 Jun 22
1
SIP Channel hangup problem with re-INVITE enabled - ugrent
Hi List
I have UAs registered with Asterisk and make outbound calls via ITSP1,
everything is fine without re-INVITE. When people call 178, the actual
number 112233445566 at ITSP1 network will be called.
When UA or called telephone (112233445566) hang up, the call and associated
channels are cleared.
Sip.conf
[general]
canreinvite=no
nat=no
[ITSP1]
type=p...
2011 Jun 21
1
: Re: ITSP failover for PRI
...way with the following lines in
extensions.conf failover works.
If one ITSP is unavailable the call flow cascades to the second ITSP and
connects with audio.
[outgoing]
exten => _1NXXNXXXXXX,1,NoOp(${CALLERID(all)="" <>}) exten =>
_1NXXNXXXXXX,2,Dial(SIP/${EXTEN}@ITSP1)
exten => _1NXXNXXXXXX,3,Dial(SIP/${EXTEN}@ITSP2)
If we attempt calls from the PBX over the PRI connected to the Astlinux
Gateway the calls connects, but there is no audio.
This is what we see:
ITSP1:
Accepting call from 'XXXXXX' to 'XXXXXX&...
2007 Feb 08
0
SIP Re-Invite behind a NAT
...egister => me: xxxx@my1.itsp.com:5060
register => me: xxxx@my2.itsp.com:5060
; This section is because i'm behind nat
;
externip=999.999.999.999 ;Outside address
localnet= 192.168.0.148/255.255.255.0 ;Inside Network
;
;============================================================
;
[voip-ITSP1]
context=incoming-sip
type=friend
host= my1.itsp.com
username=me
secret=xxxxxxx
nat=no
canreinvite=yes
insecure=port,invite ; do NOT remove this
qualify=yes ; do NOT remove this
dtmfmode=rfc2833 ; should match what is set on your account
disallow=all
allow=ulaw ; set in/out codec here
;
[voip-ITSP2...