List, I need your help with information going out on my SDP. Is it possible to update the Media Address on a per-call basis or a per-channel basis?Reason:My Asterisk is in a private network and needs to connect to UA on its internal network and a..
I am working with a customer and their SIP provider is IPitimi.The customer needs to sometimes provide various caller id number for the calls going to IPitimi.They are processing calls for multiple businesses who want their caller id to show up.W..
all,Checking on the asterisk source code Ive seen that SIP will always use the IP address in the c= field of SDP to send media. Is that correct?Is there a case where asterisk would send media to the received source IPaddress instead of the one he ..
HelloI use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom Germany. We have sometimes problems with incoming and outgoing calls. I hope I can explain it understandable.For example, Asterisk sends a REGISTER to 188.8.131.52 (tel.t-online.de..
I want to have Kamailio in front of one or more Asterisk boxes.I dont think it is necessary for Kamailio and Asterisk to register with one another. Id like for PJSIP to recognise Kamailio by its IP address.I have two boxes, both have public IP address..
I am currently trying to send faxes via T.38 using PJSIP (newest version 2.3) with Asterisk 13.0.2. After having configured PJSIP, I have seen several things the cause of which I would like to know.1) Ports and IP addresses which PJSIP bind toI h..
Asterisk Project Security Advisory – AST-2014-012 ProductAsterisk SummaryMixed IP address families in access control lists may permit unwanted traffic.Nature of AdvisoryUnauthorized Access SusceptibilityRemote unauthenticated sessions Severity ModerateExplo..
Yesterday a customer was attacked from the following IP addresses so
add them to your blacklist:
iptables -A INPUT -s 184.108.40.206 -j DROP
iptables -A INPUT -s 220.127.116.11 -..
If you are using Grandstream GXP 21XXX and 14XX phones and you are doing any kind of remote firmware updates or config updates DO NOT use the 18.104.22.168 BETA version.We have found a bug in it that causes HTTP updates to not work if you are using a dom..
To put it simply, is the process where Asterisk tries to redirect the RTP media stream to go directly from the caller to the callee. Be careful that some devices do not support this (especially if one of them is behind a NAT). The default setting..