asterisk users - Jan 2019

Thursday January 31 2019
TimeRepliesSubject
2:30PM 0 features.conf disconnect and local channels
10:40AM 0 Dailplan with playtones
10:36AM 2 Dailplan with playtones
10:26AM 0 Dailplan with playtones
9:59AM 2 Dailplan with playtones
8:38AM 0 tel URI
8:22AM 2 tel URI
 
Wednesday January 30 2019
TimeRepliesSubject
8:03AM 0 [Asterisk-video] asterisk playing video call to a local display
 
Monday January 28 2019
TimeRepliesSubject
2:29PM 0 INVITE from DID: No matching endpoint found but completes the call anyway
 
Saturday January 26 2019
TimeRepliesSubject
5:55PM 3 INVITE from DID: No matching endpoint found but completes the call anyway
 
Friday January 25 2019
TimeRepliesSubject
6:11AM 0 trying to upgrade asterisk and Debian -- not working
 
Thursday January 24 2019
TimeRepliesSubject
2:46PM 1 trying to upgrade asterisk and Debian -- not working (John Covici)
7:49AM 0 trying to upgrade asterisk and Debian -- not working (John Covici)
6:17AM 2 trying to upgrade asterisk and Debian -- not working (John Covici)
 
Wednesday January 23 2019
TimeRepliesSubject
3:42PM 0 trying to upgrade asterisk and Debian -- not working
 
Tuesday January 22 2019
TimeRepliesSubject
3:44PM 0 [asterisk-app-dev] Who uses the ari/sounds resource?
3:30PM 0 [asterisk-app-dev] [asterisk-dev] Who uses the ari/sounds resource?
2:56AM 0 ChanSpy "Audiohook has stale audio in its factories" problem
12:32AM 0 opus
 
Monday January 21 2019
TimeRepliesSubject
3:42PM 0 [asterisk-app-dev] Who uses the ari/sounds resource?
 
Friday January 18 2019
TimeRepliesSubject
5:06PM 1 Enhanced Messaging and softphones
4:28PM 0 Enhanced Messaging and softphones
3:58PM 2 Enhanced Messaging and softphones
3:30PM 0 Enhanced Messaging and softphones
3:21PM 2 Enhanced Messaging and softphones
 
Thursday January 17 2019
TimeRepliesSubject
3:48PM 0 Early media using ARI
3:40PM 2 Early media using ARI
2:42PM 0 Is it possible to find real domain names instead of IP in SIP URI ?
9:38AM 2 [OT] Are anonymous international calls allowed ?
6:44AM 0 ConfBridge: Identifying troublemakers
 
Wednesday January 16 2019
TimeRepliesSubject
5:42PM 0 Simple one-word offline free speech recognition in Asterisk (or as an AGI)?
4:18PM 0 CDR/CEL Radius features
 
Tuesday January 15 2019
TimeRepliesSubject
6:26PM 0 Cannot originate to extension unless /etc/hosts is edited constantly? [Tony Mountfield]
5:16PM 1 (NAT) direct media to host on local net when registering from external address
5:06PM 0 Cross-compiling Asterisk 16
5:01PM 0 (NAT) direct media to host on local net when registering from external address
4:46PM 0 Various extensions ring once and go to voicemail
4:29PM 1 Cannot originate to extension unless /etc/hosts is edited constantly?
4:17PM 2 (NAT) direct media to host on local net when registering from external address
4:00PM 0 (NAT) direct media to host on local net when registering from external address
3:46PM 2 (NAT) direct media to host on local net when registering from external address
3:30PM 0 what service does asterisk need to avoid netsock error ?
3:08PM 0 Various extensions ring once and go to voicemail - Thomas Peters
2:56PM 0 How to build and use your custom asterisk .deb package ?
2:53PM 3 Various extensions ring once and go to voicemail - Thomas Peters
2:41PM 1 MWI Delayed on Polycom VVX phones
2:29PM 0 MWI Delayed on Polycom VVX phones
2:23PM 2 MWI Delayed on Polycom VVX phones
1:29PM 0 Adding Subscribe Handlers in PJSIP
6:04AM 0 Various extensions ring once and go to voicemail - Thomas Peters
 
Monday January 14 2019
TimeRepliesSubject
10:30PM 0 Various extensions ring once and go to voicemail
10:08PM 2 Various extensions ring once and go to voicemail
10:02PM 0 Various extensions ring once and go to voicemail
9:34PM 2 Various extensions ring once and go to voicemail
8:28PM 0 Various extensions ring once and go to voicemail
6:42PM 2 Various extensions ring once and go to voicemail
5:47PM 0 Is the R2 list still up?
5:41PM 0 Problem receiving calls with Telmex in Mexico...
5:18PM 0 Out of queue - no pickup after 0ms
6:37AM 0 Outbound caller ID ignored
 
Sunday January 13 2019
TimeRepliesSubject
3:45PM 2 Outbound caller ID ignored
2:57AM 1 Overhead pager announcement in "background" channel
 
Saturday January 12 2019
TimeRepliesSubject
1:21PM 0 [asterisk-app-dev] ARI-client Node.js objects
 
Friday January 11 2019
TimeRepliesSubject
5:53PM 1 chan_sip bind port
4:14PM 0 [asterisk-app-dev] Multiple ChannelDestroyed events for the same channel
3:47PM 2 [asterisk-app-dev] Multiple ChannelDestroyed events for the same channel
2:50PM 0 Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joel)
10:08AM 0 Detecting a fax
10:04AM 0 Detecting a fax
9:23AM 4 Detecting a fax
9:19AM 0 Detecting a fax
9:12AM 2 Detecting a fax
8:59AM 0 Can SIP domain help to set multiple SIP trunks between two boxes ?
7:34AM 0 Switched from Asterisk 1.8 to 13 - CDR
6:13AM 0 Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Tony)
3:15AM 0 Compiling error
 
Thursday January 10 2019
TimeRepliesSubject
6:51PM 1 Problem with AudioCodes MP-114 ATA
3:18PM 0 Hint and state
2:13PM 2 Hint and state
 
Wednesday January 9 2019
TimeRepliesSubject
10:07AM 0 Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joshua C. Colp)
6:00AM 2 Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joshua C. Colp)
 
Tuesday January 8 2019
TimeRepliesSubject
6:49AM 0 Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joshua C. Colp)
 
Monday January 7 2019
TimeRepliesSubject
6:42PM 0 Timeout for AGI/HAGI connections
5:25PM 0 [asterisk-app-dev] ARI Node JS Bridge.addChannel
5:23PM 3 [asterisk-app-dev] ARI Node JS Bridge.addChannel
4:23PM 1 Configure SIP reply timeout (timerb in sip.conf)
6:47AM 1 Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero
 
Friday January 4 2019
TimeRepliesSubject
9:20PM 0 CyberMegaPhone WebRTC Video Conference demo
5:23PM 2 CyberMegaPhone WebRTC Video Conference demo
 
Wednesday January 2 2019
TimeRepliesSubject
12:28AM 1 Custom langagues