!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..
Author : Luca Bertoncello
Home » Articles posted by Luca Bertoncello (paged 5)
!Another day, another problem… Im checking with Nagios my Asterisk at home, and since yesterday I noticed that, after my IP changes (Deutsche Telekom drop the DSL-line every 24 hours, so that I have a new IP every day), the peer of an VoIP-provi..
again!I always try to get my mobile phone work with my Asterisk. I tried to install Asterisk on my PC (with public IP), but it has problems, too… I think, my UMTS-Provider doesnt want to connect to dynamic IP or my DSL-Provider does not want it, t..
again!I decided, just for fun, to install Asterisk on a server of mine (available in Internet) and to log on my mobile phone on this server.This Server communicate with my Asterisk at home and if I call a phone at home from my mobile phone (logged..
!Since the internal calls work as expected and I can register my Asterisk on an external provider, Id like to add a new feature and allow my mobile phone to connect to my Asterisk and manage calls.Well, first of all, my Asterisk is NOT direct on Inter..
!Im trying to configure my Asterisk to accept SIP-TLS connections, too.I followed this HowTo: http://remiphilippe.fr/sips-on-asterisk-sip-security-with-tls/But as soon I try to connect to my Asterisk using SIP-TLS I get on Asterisk-CLI:== Problem sett..
again!I just noticed, that my Asterisk (running on an OpenWRT-Switch) writesthe logs using GMT… On the Switch the time is right configured and a date says me thecurrent LOCAL time.I didnt found in logger.conf or other file an option to set the timezo..
again!Im thinking about using my mobile phone to receive (and send) callswhen Im not at home (for example in holiday). I can make my Asterisk reachable from Internet, of course, or I canuse a VPN, thats not the problem…My question is: can I log..
!I configured Asterisk to forward the incoming call for a number toboth phones. I wrote that:exten => _00493512222222,n,Dial(SIP/00493512222222&SIP/00493511111111,,R)of course it works… Now the problem is, that when a phone get the call, on the ot..
!I read the pages that Steve sent to the list. It sounds nice, but Ididnt 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? Orto get this list in ot..