[OpenSIPS-Users] forwarding calls from Asterisk to OpenSIPs
Julian Kay
juliank at vazycomm.com
Sun May 10 23:03:21 CEST 2015
Thanks again!
all my extensions have insecure=invite,port. I think Asterisk is forwarding
the call as: from anonymous.
right now when the user makes a selection I use a GOTO in Asterisk to send
it to an Asterisk CONTEXT and from that CONTEXT I use:
Dial(SIP/opensips/1005) opensips is the peer name in sip.conf
As per the sip trace
Opensips gets an INVITE
FROM "Anonymous"<sip:anonymous at vazycomm.com>
TO <sip:1005 at vazycomm.com:5060>
Thanks, I do appreciate the input very much
Juls
From: Stefano Pisani [mailto:stefano.m.pisani at gmail.com] On Behalf Of
Stefano Pisani
Sent: Sunday, May 10, 2015 2:37 PM
To: Julian Kay
Subject: Re: [OpenSIPS-Users] forwarding calls from Asterisk to OpenSIPs
Use insecure = invite,port in extensions configuration.
s
Il 10/05/2015 18:39, Julian Kay ha scritto:
Thank for the input!!
When I do, it fails and a SIP trace shows error: Failed to authenticate on
INVITE to "Anonymous" <mailto:sip:anonymus at vazycomm.com>
<sip:anonymus at vazycomm.com>
I'm using basically using the "residential" script with some hooks for
Asterisk. Asterisk is doing some IVR, once a department / individual is
selected I want asterisk to transfer the call to that extension.
Again, thanks for any help!!!
Juls
From: users-bounces at lists.opensips.org
[mailto:users-bounces at lists.opensips.org] On Behalf Of Stefano Pisani
Sent: Thursday, May 07, 2015 4:24 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] forwarding calls from Asterisk to OpenSIPs
Just define a peer in sip.conf
then Dial(SIP/peer_name/OpensipsAccount) on extensions.conf. Very simple.
regards,
s
Il 07/05/2015 08:32, Julian Kay ha scritto:
Hello Everyone;
I want to be able to forward calls from Asterisk to a phone registered with
OpenSIPs, can anyone give me some information on how this can be done? Or a
link to the info I would need to do this.
Thanks for any help
Juls
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