asterisk users - Aug 2011

Wednesday August 31 2011
TimeRepliesSubject
9:12PM 0 phone + video
7:49PM 7 Faxes suddenly failing
6:36PM 0 Asterisk 1.8.6.0 Now Available
3:43PM 15 cli command show codecs
 
Tuesday August 30 2011
TimeRepliesSubject
10:19PM 0 Transfer to VoiceMail Asterisk 1.6
9:48PM 0 subscriptions from ekiga to asterisk
6:53PM 4 MOH making calls appear hung up
4:32PM 5 dahdi with isdn nt_mode, phone no signal still.
2:44PM 2 Avaya to Asterisk Voice mail
2:16PM 2 same sip peer as user and provider
1:44PM 10 T.38 passthru on 1.8.5
9:16AM 1 FREE webinar video about Auto-Dialer Business Model (Telemarketing)
 
Monday August 29 2011
TimeRepliesSubject
8:15PM 0 http manager getconfig crashes on reading large files/categories (50+ lines)
6:56PM 6 Dragging the dialup customers along, possible?
3:49PM 0 Scheduled Maintenance for Asterisk Project community services
10:04AM 1 Asterisk is delaying DTMF (SIP INFO) relay in MeetMe
8:30AM 0 Presence for channels other than SIP.
7:25AM 0 Create ou update values ins ASTB when Asterisk is stopped
7:15AM 2 Dialing multiple endpoints and CallerID presentation
 
Sunday August 28 2011
TimeRepliesSubject
4:26AM 0 chan_dahdi.conf waitfordialtone
 
Saturday August 27 2011
TimeRepliesSubject
3:31PM 11 USB or Ethernet based FXO device ?
12:33PM 1 how to play wav files to all members in konference
 
Friday August 26 2011
TimeRepliesSubject
6:02PM 16 Good, inexpensive wireless VOIP handsets ?
5:52PM 6 Looking for ideas for nice **Asterisk** home phone system
3:53PM 2 SIP Trace to troubleshoot one way of communications
2:29PM 2 app_sms testers required
2:18PM 0 Queue Group not forwaring calls to agents
10:05AM 10 Wanted a modified SIP message body
9:41AM 4 DMTF via rfc2833 and SIP-INFO simultaneously
6:21AM 1 Asterisk spontaneous reboot
12:51AM 6 red alarm on tdm400 fxo (fxs signalled)
 
Thursday August 25 2011
TimeRepliesSubject
7:58PM 5 "Core Show" being assumed before commands
6:36PM 3 Thunderbird extension using AMI to dial
2:20PM 0 Possibly odd sip.conf security requirements. Possible?
1:26PM 10 Bind SIP over TCP port in asterisk 1.4.42.
10:33AM 1 security: SIP header spoofing CHANNEL(recvip)?
6:27AM 3 SIP client on a mobile?
4:37AM 0 What sort of information does LIDB provide?
 
Wednesday August 24 2011
TimeRepliesSubject
7:12PM 0 Softhungup missing from Asterisk 1.6.20-1 - *without any notice*
5:21PM 10 System Command not executing php
5:18PM 3 1.8.5 Voicemail duration incorrect
2:17PM 1 [OT] Yealink T26/28/38 and Open-VPN
2:01PM 6 Asterisk Integration with Android device
1:16PM 9 How to know how many user is connected
12:16PM 0 Fwd: Re: Welcome to the "asterisk-users" mailing list
9:15AM 0 Time-limited calls -- revisited
4:40AM 1 Invite with replaces handling issue
 
Tuesday August 23 2011
TimeRepliesSubject
10:28PM 3 Help with pri call giving error.
 
Monday August 22 2011
TimeRepliesSubject
8:11PM 46 Looking for ideas for nice home phone system
5:47PM 0 netsock error? some sip clients crashing!
5:17PM 1 res_odbc with informix
5:38AM 1 allow anonymous call
 
Sunday August 21 2011
TimeRepliesSubject
9:29PM 5 espeak module for asterisk
9:26PM 0 Flite module for asterisk
 
Saturday August 20 2011
TimeRepliesSubject
4:46PM 7 Outgoing calls fail in chan_gtalk
4:23PM 5 How is a ping test delay "ms" different from status in Asterisk "sip show peers"?
12:00PM 2 Sytem Commands not executing
5:09AM 1 Gtalk channel problem
 
Friday August 19 2011
TimeRepliesSubject
7:37PM 5 Outbound Dial
1:14PM 10 Possible Bug? .call files executing multiple times
11:08AM 0 Can't use SmartVoip/JustVoip after update
3:11AM 0 ChanSpy on 1.6.2.20
1:12AM 5 Playback while dialing out
 
Thursday August 18 2011
TimeRepliesSubject
3:29PM 0 cisco 7945 sip lines on 2 different asterisk servers
12:42PM 2 Asterisk 1.8 SIP_CAUSE performance regression
11:42AM 1 issue with the detection of the call status after sending it using Orginate (DAHDI/1/...., app, ...
10:31AM 1 1.8.5 CLI colors are gone?
10:20AM 2 How to get presence using AMI
 
Wednesday August 17 2011
TimeRepliesSubject
10:01PM 15 Polycoms rebooting themselves
6:35PM 3 Spy just a range of extensions
1:08PM 1 Asterisk 1.8.5.0
 
Tuesday August 16 2011
TimeRepliesSubject
3:16PM 5 Asterisk scaling
2:31PM 3 PRI Problem
1:37PM 1 Strange Asterisk notices
9:38AM 1 Celebrating 10 years SER SIP Router in Vienna: 8th September 2011
9:20AM 0 AGI dialplan
8:34AM 2 Speed Dials Management....
12:40AM 3 Asterisk -> Office 365 Unified Messaging... anyone done it?
 
Monday August 15 2011
TimeRepliesSubject
11:29PM 0 1.4.38 passing a Regular expression containing a pipe character to a macro ?
10:54PM 0 SIP trunk trouble. Please help.
9:48PM 4 AMI Commands - not working as Expected, Maybe???
5:46PM 5 Linksys/Cisco 504G randomly restarts
5:20PM 5 issue with some numbers
2:11PM 0 app_swift for Asterisk 10
1:09AM 3 Queue Breakout Input being Ignored
 
Sunday August 14 2011
TimeRepliesSubject
10:17PM 0 dynamically alter list of offered codecs (for faxing)
5:26PM 0 Adhearsion 1.2.0 Released at Lone Star Ruby Conference V
4:26PM 5 DAHDI-linux-complete on CENTOS kernel 2.6.32
7:36AM 1 1.6.2.20 ${DIALSTATUS} disagrees with CDR(answered)
7:08AM 1 callgroup more than 63
12:12AM 2 asterisk-users Digest, Vol 85, Issue 23
 
Saturday August 13 2011
TimeRepliesSubject
11:34PM 5 Echo problem in the analoge lines
11:28PM 6 DID to display the calling number
 
Friday August 12 2011
TimeRepliesSubject
8:59PM 1 One way audio when using originate...
8:23PM 1 Message prints even if verbose level is Zero
3:59PM 0 Conference calls through web-interface with moderation using Asterisk?
3:23PM 8 .call files in /var/spool/asterisk/outgoing
1:57PM 5 Queuing outgoing calls
12:37PM 0 Interrupting a call in progress?
11:06AM 11 Queue agent login notification
 
Thursday August 11 2011
TimeRepliesSubject
10:40PM 0 Asterisk 1.8.5 - Ubuntu Pkg from diguim Repo - OPENH323 error
9:31PM 0 experiences sharing
8:55PM 1 TLS Error on 1.6 and 1.8
7:04PM 3 Where to proceed next
9:23AM 0 Background music during a call
8:07AM 13 Trouble with *8 Pickup
7:03AM 6 Problem setting for incoming termination
6:27AM 6 Any Method for capturing ISUP packets in DAHDI/ASTERISK
5:50AM 1 Asterisk reporting
1:12AM 1 Asterisk+internal phones+recorded messages
 
Wednesday August 10 2011
TimeRepliesSubject
12:30PM 2 Asterisk 1.8 Install Problem
11:57AM 0 Unable to enable echo cancellation on channel 1 (No such device)
9:59AM 1 BT killed Ribbit
7:23AM 3 ulimit
 
Tuesday August 9 2011
TimeRepliesSubject
8:38PM 0 A lA orden los libros!!!
7:38PM 19 FAX Issues
12:16PM 6 block all numbers begin by 00 and 1
10:36AM 0 Look c8m9kl
 
Monday August 8 2011
TimeRepliesSubject
6:57PM 1 1.8 issues with Local Bridging
4:37PM 4 SRV question
2:52PM 3 CEL and MySQL
1:41PM 1 Version 1.8 strange expression error
9:46AM 2 DAHDI Callerid and transfer problem
8:43AM 0 Timer B in sip.conf cannot be changed
6:30AM 0 DAHDI-Linux 2.5.0 and DAHDI-Tools 2.5.0 Released
5:18AM 0 Reconfiguring smoother module in 1.8
2:32AM 4 Polycom and auto answer
 
Sunday August 7 2011
TimeRepliesSubject
6:49PM 4 fail to correctly build 1.8.5 ??
2:49AM 2 Dahdi does not build against Kernel 3.0.0
 
Saturday August 6 2011
TimeRepliesSubject
11:11AM 0 openh323 or ooh323
6:35AM 14 Firewall Issue
 
Friday August 5 2011
TimeRepliesSubject
8:32PM 0 Asterisk 1.6.2.20 Now Available
8:31PM 1 Receptionist Extension cannot be Pickup()'ed
8:10PM 0 asterisk-users Digest, Vol 85, Issue 10
6:20PM 4 No more CDR record for simple Hangup?
3:41PM 7 Assistance sending mass sms to cellphones
2:43PM 3 error: Autodestruct on dialog
2:18PM 0 Fax Detection with chan_ss7
12:53PM 3 Ring delay problem
12:26PM 0 Audio when a call is on hold.
12:11PM 0 Asterisk 1.8.5 eventfilter not working in manager.conf
11:17AM 1 ASterisk is Going stop whenever restart the server
11:16AM 0 Call Park announcing to Caller rather than callee.
11:02AM 0 (no subject)
10:55AM 1 Send Refer with replaces from asterisk
10:53AM 2 display name
8:32AM 3 Custom Dialplan
 
Thursday August 4 2011
TimeRepliesSubject
7:39PM 16 Answering machine answers after pickup a phone.
4:59PM 3 REGISTER forwarding problem
3:20PM 11 pickupgroup
2:36PM 1 burned module X400M
8:21AM 3 Customizing sip response codes for PBX Sip trunk
7:28AM 1 wctdm24xx IRQ missing
2:02AM 0 UK BT ISDN30 settings?
 
Wednesday August 3 2011
TimeRepliesSubject
2:49PM 2 Need a volunteer for a Patch
2:12PM 17 Increasing volume ?
1:20PM 2 Debugging Sip
12:45PM 0 trustrpid in sip.conf
11:36AM 1 Changing sip response codes
9:55AM 3 Know the number of concurrent dial ?
9:32AM 0 Barging in PBX
9:31AM 0 *8 pickup not releasing channel
8:06AM 1 dundi
7:28AM 2 Asterisk reload, to execute file
6:01AM 1 Segmentation Fault
5:41AM 0 CHANSPY
2:15AM 2 snom and srtp
1:38AM 0 Outgoing call issue in D-link DPH-80 ip phones
1:22AM 6 TE410P hardware problems
 
Tuesday August 2 2011
TimeRepliesSubject
7:51PM 4 Codec negotiation issue (no audio format found to offer)
5:06PM 3 Problem with (asterisk1.8-iksemel1.4-GoogleVoice)
4:42PM 6 use ILBC installed from asterisk yum repositories
2:23PM 0 SMS within asterisk users
9:58AM 4 MixMonitor and attended transfers
8:43AM 0 How to stop Dial from waiting for extra digits if number is incomplete.
8:29AM 0 asterisk speech to text and text to speech?
 
Monday August 1 2011
TimeRepliesSubject
8:24PM 3 Problems with AMI connections (Asterisk 1.8.3.2)
5:02PM 5 T38 Fax