Archives : June-2015
Home » 2015 (paged 39)
!So, new day, new problem…I tried right now to call from my cellphone a peer in my Asterisk. The cellphone has correct credentials, but its NOT registered on my Asterisk, now.I just tried to call a peer in my network, from a peer not yet register..
!Im working hard to securing my Asterisk… Now I deleted all possibility to access the node as anonymous and every call through the proxy will be checked (just known peers are allowed to use it). Furthermore, I restricted the registration of my h..
The Dial() application lets you specify two or more destinations, separated by & characters.When you execute an application call of this sort in your dialplan, Asterisk dials all of the destinations in parallel.If theyre SIP clients, each will rece..
HiIs there any way to set the presence state of a peer to in-use in asterisk1.8?The idea is to integrate DND buttons on phones to B..
!Today I tried to change the NAT-configuration on my Firewall to useanother port for SIP. I configured it so:/sbin/iptables -t nat -A PREROUTING -p udp -m udp –dport 10000:10100-j DNAT –to-destination 192.168.20.120/sbin/iptables -t nat -A PREROUT..
The Asterisk Development Team has announced the releases of:DAHDI-Linux-v2.10.2-rc1DAHDI-Tools-v2.10.2-rc1dahdi-linux-complete-2.10.2-rc1+2.10.2-rc1This release is available for immediate download at:http://downloads.asterisk.org/pub/telephony/dahdi-li..
Asterisk-user.Im starting in a soft-phone project with lots of requirements and some of then caused me some doubts about Asterisk. Could someone tell me if Asterisk can help me with some requirements? See below:1 – My SIP server (Asterisk) will h..
!Very strange… I ran the Asterisk CLI for other tasks, and suddenly I got this message:== Using SIP RTP CoS mark 5– Executing [000972592603325@default:1] Verbose(SIP/192.168.20.120-0000002a, 2,PROXY Call from 0123456 to 000972592603325) in new stac..
Sorry if off topic, but is anyone here on this list using it?I am currently searching for a good router for my home network wich supports SIP. Ma..
I have configured a certified asterisk 13 server with chan_mobile and res_pjsip. I have a Cisco 7940 hardphone and I use ekiga as softphone client.Now the problem is, using the hardphone Im able to call the softphone and hear everything properly. ..