Archives : June-2015
I have this in my sip.conf:exten => *98,1,Verbose(0,CALLERID number is ${CALLERID(num)})same => n,VoicemailMain(${CALLERID(num)}@LocalSets,s)same => n,HangupHowever, my extensions are set up so that they always show the external number, not the extension:[foobar2](client-phone)secret=xxxxxxxxxxxxxxxxxxxxxxxxxx..
Dear asterisk-users,I have listened that a diaplan on Asterisk can extract information from proprietary SIP messages header fields. That is, if Asterisk receives a SIP message with a modified HEADER (containing proprietary fields) , is it possible..
This is really three problems, as follows:(1)Accessing the Google Contacts API to retrieve someones details based on their phone number.(2)Passing the incoming callers number to an AGI script.(3)Displaying the details retrieved from Google on your screen.Presum..
again!About my previous E-Mail…I though about it and I think, that maybe Im just very stupid… Since I called an INTERNAL number, Asterisk tried to call it.I tried right now to call an EXTERNAL number (using my context[myproxy]) and the behavior..
!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..