Hi everyone,
I am running asterisk with release 12.1.0.rc3 and PJSIP.
I have a peer which sends OPTIONS method for session keep-alive, and the
asterisk is not responding to it. That of course disconnects the call after
a few minutes.
Is there a settings in the PJSIP.conf to respond to in dialog
Is it possible to get .spec or .src.rpms for packages on packages.digium.com?
I specifically need to rebuild kmod-dahdi-linux-fwload-vpmadt032 for the
kernels available in CentOS 5.10. I see there’s source at
downloads.digium.com/pub/telephony/firmware/releases, but .spec files are not
Hi Jeff,
On 04/08/2014 12:13 PM, Jeff Brower wrote:
Darrel- The G729 essential patents were *granted* in 1996, but applied
for prior to June 8 1995. That means their lifespan is either 20 years
from their application date, or 17 years from their grant date,
whichever is greater
Hi,
I have trouble establishing a call between between two SIP phones. One sip
phone is, with asterisk server, at home behind a firewall. The second sip
phone is an iPhone with 3G wireless connection.
When I call from the SIP device at home the SIP account on the Internet
(iphone + 3G) I can
