Hello again,I love to learn. That’s why I’m trying myself. Thank you
againsathees
--
View this message in context:
http://opensips-open-sip-server.1449251.n2.nabble.com/incoming-DID-will-not-pass-to-asterisk-tp7595694p7595772.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.c
The OpenSIPS project owners offer some excellent consulting solutions
that will save you a lot of time.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Thank youi Will look into
--
View this message in context:
http://opensips-open-sip-server.1449251.n2.nabble.com/incoming-DID-will-not-pass-to-asterisk-tp7595694p7595767.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.___
Users
Hello Mahan,
My suggestion would be to do a lot more research:
i)
http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration
ii)
http://www.voip-sip.org/wp-content/uploads/2011/08/Building-Telephony-Systems-with-OpenSIPS-1.6.pdf
Take a few weeks and get to know opensips and build
Hello Terrance.Yes it’s the basics scripts.I’m trying to setup OpenSIPS as a
proxy to asterisk via dispatcher modules. First part I will able to register
sippers on asterisk via dispatcher modules. I’m trying work my way up. This
is my second part of the test; send incoming calls to asterisk IVR.
On Tue, Mar 10, 2015 at 6:06 AM, mahan77 wrote:
> Hello Terrance,
>
> Thank you for your time to replay back.
> It was basic dispatcher Config file and posted in the first place.
>
> This is my sip trace.
>
> interface: eth0 (192.168.1.0/255.255.255.0)
> filter: (ip or ip6) and ( port 5060 )
>
>
Hello Terrance,
Thank you for your time to replay back.
It was basic dispatcher Config file and posted in the first place.
This is my sip trace.
interface: eth0 (192.168.1.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )
U 192.168.1.64:5060 -> 192.168.1.150:5060
.
Gb.."
?.
U 87
Hello Sathees,
It is the appropriate place. I myself would need some more information such
as SIP trace, and maybe
more of the config?
Terrance
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Is it wrong question to ask in this mailing list?
--
View this message in context:
http://opensips-open-sip-server.1449251.n2.nabble.com/incoming-DID-will-not-pass-to-asterisk-tp7595694p7595723.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
Hi,
I need some help please.
I’m trying to pass all sip packets to asterisk via dispatcher module.
The problem I’m having if UA signed in the incoming DID will not pass to
asterisk. The error message was “SIP/2.0 401 Unauthorized”
How can I solve this problem?
preshiead
Sathees
This is my scr
10 matches
Mail list logo