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:[email protected]>

TO <sip:[email protected]:5060>

 

Thanks, I do appreciate the input very much

Juls

 

 

 

 

From: Stefano Pisani [mailto:[email protected]] 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:[email protected]>
<sip:[email protected]> 

 

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: [email protected]
[mailto:[email protected]] 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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