Hello,
I have implemented a simple sip client (using pjsip 1.16) running on OpenWRT having FXS interface to an analog phone.
The sip client is registering to an Asterisk server.
The client is working fine.
When I dial out from the analog phone the termination of the dialing currently is done by 0.5 seconds time out.
This works but seems not ideal.
At the moment my pjsip client has no information about the dialplan (kept in Asterisk server).
Can I implement more intelligent dialing (from my FXS sound device) so Asterisk gets each dialed number and responds with number complete or something?
Best Regards
Dimitar Penev
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20150813/c8854a64/attachment.html>
