Archives : June-2015
we think that there is a mistake with the asterisk-11.18.0.patch. The file look likediff –git a/.version b/.version index c5df2aa..150754a 100644— a/.version+++ b/.version@@ -1 +1 @@-11.18.0-rc1\ No newline at end of file+11.18.0 \ No newline at ..
everyone,i have question about fax detection on dahdi channels. does dahdi channels detect fax and pass it? if yes, does it detects both types of fax (g711pass through and T.38)? finally, how can i enable it on dahdi_channels? i set faxdetect=both..
Ive been having problems segfaulting when receiving faxes. So I tried FFA. I used the register utility and my license is in /var/lib/asterisk :ls /var/lib/asterisk/licenses/ -l total 4-rw-r–r–. 1 root root 325 Jun5 22:41 FFA-DNCXXXXXXX.licbut:Execut..
dialplan[FaxIncoming]exten=s,1,NoOp(Incoming fax on 46-va) same=n,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d)}_${STRFTIME(${EPOCH},,%H%M)})same=n,Answer()same=n,ReceiveFAX(${FAXFILE}.tif,df)same=n,Hangup()exten=>h,1,NoOp(FAXSTAT..
!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..
Sorry for a bit of a newbie post but we all had to start somewhere right ..Im wondering if someone can briefly explain the difference between blind and attended transfers and why they would generate two very different CD..
despite some searching I havent found an answer to this question:Is there a way I can see in the log, or find out in the dialplan, what codec has been negotiated for a SIP channel? If possible, Id like to do this in both Asterisk 11 and in an old ..