my webrtc calls ends after ~60seconds with "res_rtp_asterisk.c: DTLS failure occurred on RTP instance '0xb6c02a94' due to reason 'sslv3 alert unexpected message', terminating". any ideas where can be problem? or howto debug this problem?
asterisk13.4.0-rc1 + sipml5 latest (chrome,firefox)
Someone on this list uses the address @sedwards.com
I doubt this is their actual email address as there is no MX record for sedwards.com and I can't find registration for their domain either.
Part of my mail servers reject these emails because they cannot be replied to, or are likely to be spam.
Every so often I get a mail from the list management to say that I've been unsubscribed because of excessive bounces and it takes a single click to re-register.
It's a bit of a niggle for me. What do you think I should do? Change my servers so that I don't…
The results of a security experiment were published this week, in which an Asterisk PBX was set out in the wild to see who would attack it and how:
What I find particularly interesting is that people/bots are scraping support websites looking for valid IP's of PBX's, and valid credentials!
A good reminder to everyone on this list to not publish the IP of their PBX's, or even account names (in postings) as they will be quickly targeted....
Ia had a server overload today because someone did a call forward to their own extension. To do a call forward I write a key called CFWD with the extensión number and number to dial . The main script tests if the key/value exists and dials the number stored in the database. What is an easy way to prevent dumb people from creating a loop? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 --
I'm new with Asterisk and I have to create a dial plan that will invoke a binary code. That is, asterisk will execute a program in the same machine. How to do it?
Let me explain what I have to do:
In the project that I am currently working, there is smartphones, SIP servers and doors/gates to be unlocked remotely. When the user executes an application on his/her phone, it will presents a button to unlock a remote gate or door. By pressing such button, the application will send a SIP INVITE to the SIP server (Asterisk). In this moment, a…
I read the pages that Steve sent to the list. It sounds nice, but I didn't found any documentation about available SIP-Header on my phone (ST2022, not ST2030!).
Is there a possibility to ask the phone which header it understand? Or to get this list in other ways?
Thanks a lot Luca Bertoncello (email@example.com)
Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls).
Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP...
Well, now I have some time to spend with "fooling"... My phone will receive calls from 3 numbers. All that was done in my dialplan. Now, it would be nice, if I can signaling on the phone which number will be called, so that,…
I currently run an Asterisk server on a NetBSD system. It mostly works but sometimes I have weird issues. As far as I can tell they are usually NAT issues.
I have a Cisco SPA-2102 with two phone numbers installed. I have NAT Mapping and NAT Keepalive enabled. No STUN server. Both are using 5060. This is behind an ADSL through a WRT54GL with no special port handling.
The server is 11.15.1. My sip.conf includes this:
[general] context=unauthenticated allowguest=yes udpbindaddr=0.0.0.0 nat=force_rport,comedia srvlookup=yes qualify=yes
All of this works fine. It also works fine with the few clients that I have connected. However, certain changes cause…