Hi Nabeel,
OpenSIPS does not assume anything by default. If you want to have any
new calls to user A rejected (if A already in a call, with other users
or any service), you need to script this.
You should use dialog profiles to count the ongoing calls for the user A
and when you get a new call for A and the profile counter is <> 0, just
do the reject.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 08.08.2016 01:44, Nabeel wrote:
Hi,
The busy reply should be generated in exactly the same way as when the
callee is busy on the phone with another SIP user (from the OpenSIPS
subscriber table). The only difference is that the callee is on the
phone with asterisk voicemail instead of a SIP user from the
subscriber table. I'm not sure who exactly should generate the reply
but I think it depends on how OpenSIPS is interacting with the
Voicemail server during a call to voicemail. Perhaps it's not treating
the voicemail service as another 'user' so the caller didn't receive
the busy signal.
Nabeel
On 7 Aug 2016 6:57 p.m., "Bogdan-Andrei Iancu" <[email protected]
<mailto:[email protected]>> wrote:
Hi Nabeel,
Who should generate the 486 Busy reply ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com <http://www.opensips-solutions.com>
On 06.08.2016 02:56, Nabeel wrote:
Hi,
OpenSIPS does not receive a '486 Busy' signal when a callee is
using the Asterisk voicemail service. If a user is currently
listening to voicemail or leaving a voice message, an attempt to
call that user should result in a '486 Busy' signal. From that
response, I will be able to play a busy tone on the caller's
side. Please let me know how to fix this.
Nabeel
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