Displaying 4 results from an estimated 4 matches for "sip_trunk".
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2005 May 25
2
RTP path with Cisco CCM
Hi,
I have the following config:
[7960] <--skinny--> [Cisco CCM] <--SIP_trunk--> [asterisk] <--SIP-->
[X-lite]
Is there a chance to avoid the RTP stream from passing through the Cisco
CCM ? I would like to have all RTP handled by the *.
This is just a testbed, for a larger project. What I want to achieve, is
actually this:
[Cisco Phone] <--skinny--> [Cisco...
2007 Oct 23
2
register => to let Asterisk register to another softswitch via SIP
...help.
This is if I need to let Asterisk register with
another softswitch (so I used register =>), what if I
need asterisk to send call for the softswitch without
register to it (directly)? If I removed the register
=> then how it will distiguish the IP address in the
"host" at the [sip_trunk] is the IP address of the
softswitch that need to register with it and not the
IP address of the original caller sip endpoint?
Your help is highly appreciated.
Regards
Bilal
The same way you do it with IAX2, pretty much.
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
On Fri, 19 Oct 200...
2015 Feb 18
1
SIP trunk no audio
...NEW -m tcp -p tcp --dport 8000:60000 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 8000:60000 -j ACCEPT
I am using asterisk 11.16
box A is
[boxab_sip]
type=friend
username=boxa_sip
secret=***
disallow=all
allow=ulaw
allow=alaw
allow=gsm
dtmfmode=rfc2833
host=DNS Name here
context=sip_trunk
insecure=port,invite
box B is
[boxab_sip]
type=friend
username=boxab_sip
secret=***
disallow=all
allow=ulaw
allow=alaw
allow=gsm
dtmfmode=rfc2833
host=DNS Name here
context=sip_turnk
insecure=port,invite
Is there something I am missing?
The one piece I have not done before is SIP trunk - to - SIP...
2007 Jul 12
0
No subject
...let Asterisk register with
another softswitch
(so I
> used register =>), what if I need asterisk to send
call for the
> softswitch without register to it (directly)? If I
removed the
register
> => then how it will distiguish the IP address in the
"host" at
the
> [sip_trunk] is the IP address of the softswitch that
need to
register
> with it and not the IP address of the original
caller sip
endpoint?
Unless I am missing something here, I suppose the
answer is that
Asterisk can distinguish the IP endpoints because they
are ...
distinct.
Here is the ess...