With chan_sip there is the variable SIP_MAX_FORWARDS to set
Max-Forwards. This counter is persistant after a redirect. I can't find
the equivalent for PJSIP, so I went the way of header manipulation. Only
to find out that any headers added to the outbound leg are lost after a
redirect (with
On Thu, Aug 02, 2018 at 05:29:23PM +0200, Daniel Tryba wrote:
> With chan_sip there is the variable SIP_MAX_FORWARDS to set
> Max-Forwards. This counter is persistant after a redirect. I can't find
> the equivalent for PJSIP, so I went the way of header manipulation. Only
> to find out that any
On Thu, Aug 02, 2018 at 02:40:48PM +1000, Patrick Wakano wrote:
> In my opinion, Asterisk should at fail the Dial and proceed with whatever
> was configured in the dialplan I tried some other 4XX SIP codes, but
> the only one I found not behaving properly is the 400 one
I think you are
Hi All
With the below config, I just keep gettings this in the Asterisk 13.22.0
CLI:
WARNING[15872][C-0051]: channel.c:6343 ast_request: No channel type
registered for 'Agent'
whenever a caller gets sent to that agent queue with logged in agents
waiting for calls on Asterisk 13.
3997 and
On Thu, Aug 2, 2018, at 5:54 AM, Stefan Viljoen wrote:
> Hi All
>
> With the below config, I just keep gettings this in the Asterisk 13.22.0
> CLI:
>
> WARNING[15872][C-0051]: channel.c:6343 ast_request: No channel type
> registered for 'Agent'
chan_agent doesn't exist anymore[1] in
Hi there; I'm trying to dial into a Zoom conference, send some digits,
wait, send a name, and be "in the room", as it were.
I thought this would work:
same =>
n,Dial(PJSIP/02036950088@voipfone-205,12,r(callWaiting)D(WWW12345W#WW::))
But it didn't, so I tried all of these:
same =>
Hi Guys
Found the solution for this...!
https://wiki.asterisk.org/wiki/display/AST/New+in+12#Newin12-channels_chan_a
gent
and
https://reviewboard.asterisk.org/r/2657/diff/1/
and
https://blogs.asterisk.org/2016/02/10/converting-from-chan_agent-to-app_agen
t_pool/
clarifies the situation.
