asterisk users - May 2013

Friday May 31 2013
TimeRepliesSubject
9:57PM 3 Help me understand these log messages
7:21PM 2 TE410P PCI card in 1U rackount server
11:53AM 6 WebRTC softphone for Asterisk - any suggestion?
 
Thursday May 30 2013
TimeRepliesSubject
7:29PM 0 Re: asterisk-users Digest, Vol 106, Issue 41
3:06PM 6 how to launch a URl when dialing a number
12:54PM 2 Queue Periodic Announce not working...
9:46AM 3 Executing a dynamic sequence of applications
 
Wednesday May 29 2013
TimeRepliesSubject
8:18PM 0 [asterisk-announce] DAHDI-Linux and DAHDI-Tools 2.7.0-rc1 Now Available
4:53PM 0 IM through Asterisk
 
Tuesday May 28 2013
TimeRepliesSubject
6:11PM 0 Initial cut off audio
4:45PM 12 Asterisk - Soundcard - Recording?
6:21AM 4 DTMF recognized after call establishment
 
Monday May 27 2013
TimeRepliesSubject
5:17PM 0 ChanIsAvail function is breaking the round robin strategy
5:14PM 7 RED on DAHDI channel
6:56AM 12 Not able to build the chan_sip.c module
6:19AM 13 G.729 codec in pass-thru mode
 
Saturday May 25 2013
TimeRepliesSubject
3:14PM 0 Asterisk 1.8 wrong Def. Username
 
Friday May 24 2013
TimeRepliesSubject
8:29PM 6 Asterisk 11 dtmf not recognised
5:46PM 7 asterisk-gui-2.1.0-rc1
1:34PM 0 Pri-Debug-Log / Is Early Media supported by provider?
10:32AM 1 Registration timed out - for created users
 
Thursday May 23 2013
TimeRepliesSubject
3:08PM 3 Asterisk on Solaris
1:49PM 1 GotoIf function
1:41PM 0 Seeking for TTS engine supporting Hebrew
9:57AM 6 Integration with skype
9:49AM 1 Jabber
12:04AM 0 Diversion vs. P-Asserted-Id vs. Remote-Party-Id vs. P-Charge-Info vs. From Fields
 
Wednesday May 22 2013
TimeRepliesSubject
2:39PM 3 Error 488 Not Acceptable Here
1:56PM 0 Changes to the community service maintenance notifications
12:11AM 0 Automatic Speech Recognition and Text To Speech using iSpeech
 
Tuesday May 21 2013
TimeRepliesSubject
5:11PM 3 Failed to authenticate device "Ext 110"
3:19PM 8 Asterisk Log rotate not working
1:21PM 0 Planned maintenance for community services on May 23, 2013
 
Monday May 20 2013
TimeRepliesSubject
9:05PM 8 Stress testing Asterisk
1:02PM 2 Passcode
1:01PM 1 Secure Calling
1:00PM 1 Loopback question
12:58PM 1 Question
 
Saturday May 18 2013
TimeRepliesSubject
6:10PM 0 Asterisk 1.8 vici and the fax, SMS, gtalk, Jaber channels
6:01PM 7 Performance Asterisk large installation on Vmware/Xen
4:34PM 1 Asterisk 1.8-cert and AGC
 
Friday May 17 2013
TimeRepliesSubject
9:03PM 0 Asterisk 11.4.0 Now Available
9:03PM 0 Asterisk 1.8.22.0 Now Available
8:35PM 9 Auto dialer scripts and software
9:47AM 0 Temporarily features (transfer) off during Read
 
Thursday May 16 2013
TimeRepliesSubject
11:45PM 2 wanpipe and digium, oslec and hardware echo canceller
6:37PM 0 Planned maintenance for community services on May 16, 2013
6:09PM 0 Re: asterisk-users Digest, Vol 106, Issue 23
5:14PM 0 Asterisk High-availability/failover solutions
1:41PM 4 11.4: motif can only handle one channel at a time?
1:26PM 0 AstriCon 2013 (our 10th AstriCon) needs YOU!
10:13AM 1 Call Transfer question
 
Wednesday May 15 2013
TimeRepliesSubject
10:12PM 1 SetCallerPres questions
7:10PM 2 Polycom and forwarding.
6:18PM 9 Cut offs on outgoing SIP calls
3:10PM 1 How to allow AMI access to Originate yet deny Application: System
2:17PM 0 Re: 3. mfcr2 channel state IDLE 0x00 and call trace log file not ended ?? (Leonardo Rivanera)
 
Tuesday May 14 2013
TimeRepliesSubject
7:42PM 10 Using PHPMyAdmin to remotely access Asterisk MySQL Database
7:21PM 0 mfcr2 channel state IDLE 0x00 and call trace log file not ended ??
4:16PM 11 dial and bridge
3:30PM 2 Monitoring SIP trunk status on call by call basis
2:55PM 0 Call Diversion Override
12:16PM 1 Asterisk 11.3 and Kamailio 4.0 Realtime Integration Tutorial
 
Monday May 13 2013
TimeRepliesSubject
9:27PM 3 Upgrade from 1.0.x to AsteriskNOW 3.0
7:44PM 1 amiDebugger - might make your life easier if you program through the AMI
9:29AM 5 Sangoma Wanpipe Driver
 
Sunday May 12 2013
TimeRepliesSubject
2:04AM 4 Integrate Astreisk with SIP interface
1:40AM 7 time zone setting in asterisk
 
Saturday May 11 2013
TimeRepliesSubject
9:26PM 0 HD Voice -- connecting Asterisk into HD Voice compatible mobile phone
2:52PM 1 Which channels are required for FAX, GTALK and Jaber
11:23AM 1 AMI Originate issue
8:15AM 8 dahdi driver not getting install
2:53AM 0 11.4: no incoming gv/xmpp
2:42AM 2 Tier 1 Service Providers (AT&T, Level 3)
 
Friday May 10 2013
TimeRepliesSubject
10:27PM 5 ISP trunk session ID?
8:30PM 0 Job Posting
6:45PM 1 Asterisk 12 and OPUS Codec
12:21AM 3 qualify=yes: OPTIONS: How to Change?: `From: "asterisk"`
 
Thursday May 9 2013
TimeRepliesSubject
8:30PM 0 DID providers
7:36PM 0 Planned maintenance for community services on May 11, 2013
7:23PM 17 monitoring Asterisk 1.8
7:13PM 1 chanstats console errors
3:01PM 1 Elastix vs vicidial
8:46AM 4 question about CDR
2:01AM 0 No early media on 302 redirect via two servers
 
Wednesday May 8 2013
TimeRepliesSubject
8:48PM 0 Transfer cmd via AsyncAGI
12:55PM 0 Confbridge Dynamic video_mode
 
Tuesday May 7 2013
TimeRepliesSubject
9:54PM 1 Obtaining high voice quality in dahdi channel
4:23PM 4 Asterisk and hylafax: how to debug ...
10:52AM 4 passing '302 moved temporarily' back to the SIP provider
5:25AM 3 Get Channel Variables in AMI Event NewExten
1:58AM 0 (no subject)
1:53AM 1 НА: asterisk-users Digest, Vol 105, Issue 40
1:31AM 1 chan_alsa and confbridge
 
Monday May 6 2013
TimeRepliesSubject
10:54PM 4 What is bootstrap.sh for ? Possible bug in 11.3.0 ?
6:48PM 1 Installing on an OpenVZ instance
10:37AM 0 MRCPSynth() change voice
9:56AM 0 OT - Question on Aastra 6735i - Was: Sip phone displaying caller name while on call
9:24AM 1 OT - Differences between Aastra 6730i and 6750i series
1:34AM 4 Joining an astablished call
 
Sunday May 5 2013
TimeRepliesSubject
3:07PM 4 Testing 911 call
8:33AM 4 Connecting Multiple Asterisk instances getting "Unable to create channel of type 'SIP'"
5:33AM 2 GotoIf DIALSTATUS - not working
4:26AM 0 Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
4:02AM 0 Re: BLF and asterisk Queue
2:01AM 0 11.4.-rc1: new segfault in iksemel ??
12:43AM 2 My new Polycom 450's can't xfer to 4-digit extension
 
Saturday May 4 2013
TimeRepliesSubject
3:07PM 1 AMI help needed
8:32AM 4 Cisco 9971 help
 
Friday May 3 2013
TimeRepliesSubject
6:53PM 0 Digium D70 visual voicemail - won't play
5:14PM 2 changing ringtones to a group of phones
7:15AM 0 VoIP Incoming Issue
 
Thursday May 2 2013
TimeRepliesSubject
9:53PM 8 Playing a sound file during a call
9:34PM 1 Building Asterisk 11.4.0-rc1 with PJSIP 2.1
11:19AM 5 debug strategy for one-way audio calls
3:04AM 0 Queues with different technologies for members, like Khomp Driver
 
Wednesday May 1 2013
TimeRepliesSubject
6:11PM 1 Call "stuck" in queue
9:47AM 1 SMS Scenario
6:54AM 0 Re: asterisk-users Digest, Vol 105, Issue 39
4:33AM 2 Re: multiple provider for incoming