Archives : February-2016
everyone. We have an Asterisk server running Debian Squeeze, with Asterisk v1.8.13.1 (basically, the Debian Stable version for Squeeze, but with some minor source code changes specific to our site). Were trying to upgrade to 11.13.1 (The Debian Sta..
one of my client have hundreds of siemens openstage phones i want implement provisioning (1) for Asterisk and donate the code to some OSS provisioning projectcan you recommend some live provisioning project?thanks(1) http://wiki.unify.com/images/c/c7/OpenStage_Provisioning_Interface_Developer%27s..
all, Can someone recommend what hardware to use for a 1000 analogue line capacity asterisk PA..
all, I am currently using asterisk 11, and I am trying to figure out how to set the uri parameter telephone-context. I need to set it for outbound calls for a specific carrier when making emergency calls and dont seem able to find the option to set it.Rega..
Is it possible to use serveral protocols for a single transport section in pjsip.con?In sip.conf you could use transport=udp,ws,wss so you cound use webrtc along with your phones but if I try: [transport-udp] type=transport protocol=udp,ws,wss bind=0.0…
This question was asked by Chirag on March 4 2015 earlier, but I am following exactly the same procedure here and I cannot even get my clients to register on Asterisk.Heres my PJSIP.conf:[transport-tcp]type=transport protocol=tcp bind=0.0.0.0:5061…[endpoint_internal](!)type=endpo..
Getting the some errors making dahdi 2.11.0.Seems same as listed here http://forums.asterisk.org/viewtopic.php?f=1&t=96455In that link they say to use 2.10.2 but thats from December. Is there a fix yet for this?Travis Ryan Director of Information Technolog..
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I use TLS transport for all my endpoints on my production system (Asterisk 11) .I need to debug some NAT traversal issues, and would like to use the ‘sngrep’ tool which shows SIP messages from a packet capture.Per the developer of ‘sngrepâ€..