I do not have to script anything as the end-user (already in a call) is
generating the 486 reply, not OpenSIPS itself. Make a sip capture and
check which party is generating the 486.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 10.08.2016
On 10 August 2016 at 19:38, Bogdan-Andrei Iancu wrote:
> 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.
>
In the case
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
Is there any way to make OpenSIPS handle a call with voicemail in exactly
the same way as a call with another SIP user?
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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
Hi Nabeel,
Who should generate the 486 Busy reply ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
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
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