We use a obfuscation software to encrypt/mangle both SIP/RTP which sits before asterisk. What happens is sometimes we dont get any voice. after some rtp set debug we found out that when received ip of the rtp stream is routers public ip, everything wo..
I hope this doesnt already exist, but I couldnt find anything to help.I am installing a brand new Asterisk server, and want to use the Polycom IP331 phones.Does anyone have any steps on how to configure these?I have softphones working just fine, ..
Ive been lurking on the dev discussion on creating nat=auto. It all leads me to think theres no reason to use nat=no. We have about 60 internal sip extensions connected to an multihomed asterisk box where the external ip is not nated. Each of the inter..
Im not sure whether this is possible but if it is, Im sure someone on here might know … Is it possible to use Monitor() to record a conversation, but make it start a new pair of wav files at intervals (eg every 15 minutes) if the calls go on ..
Is there any reason not to run Asterisk on an Intel Atom ..
It doesnt work at all with the dahdi timers The reason it works it works till the first reload is because you are preloading it before dahdi so it starts and uses the pthread timer later when you reload it starts using the dahdi timer and there it g..
I noticed the CLI shows that the music on hold actually stops for some reason?
Heres the output of my CLI:
Connected to Asterisk 220.127.116.11 currently running on localhost (pid = 6363)
Verbosity is at ..
How can I get a SIP trace to troubleshoot a one way of communications? I need to see what is happenning in the packets to know the reason of the problem.
All, Along with my asterisks server, all incoming calls to my D-linkDPH-80 ip phones are are working fine while calling from soft phones with good voice clarity. But not able to make outgoing calls from the same D-link DPH-80 ip phones to either s..
Is there a way to disable all SIP registration and block any requests? The reason Im asking is this particular Asterisk server will just be originating calls. Ive noticed sip attacks where the attacker attempts to register a user 100x per second caus..