Hi List
Just in case someone else runs into the same problem migrating from
chan_sip to res_pjsip.
In chan sip you did define the voicemail variables in the peer section.
I did configure most of that stuff into the endpoint of pjsip,
including:
mailboxes=
voicemail_extension=
Well, after
Hi Joshua
> The chan_pjsip module doesn't prevent that. You'd need to provide the
> full SUBSCRIBE now that it is actually finding the endpoint and coming
> in.
Ok, let's see if we can solve the mystery..
pjsip.conf
[endpt-home](!)
type=endpoint
disallow=all
allow=g722
allow=alaw
allow=gsm
On Sat, Dec 2, 2017, at 07:53 AM, Benoit Panizzon wrote:
> Hi List
>
> Just in case someone else runs into the same problem migrating from
> chan_sip to res_pjsip.
>
> In chan sip you did define the voicemail variables in the peer section.
>
> I did configure most of that stuff into the
I am having a really bad day trying to get incoming calls to work
on Asterisk 13 with PJSIP. We just migrated from Asterisk 1.8 where
everything was working but there seems that something got lost in
translation. No matter what I try I always get a 401 Unauthorized
message when receiving
On Sat, Dec 2, 2017, at 06:33 PM, Carlos Chavez wrote:
> I am having a really bad day trying to get incoming calls to work
> on Asterisk 13 with PJSIP. We just migrated from Asterisk 1.8 where
> everything was working but there seems that something got lost in
> translation. No matter
