asterisk users - Oct 2011

Monday October 31 2011
11:03PM 0 (no subject)
9:59PM 11 custom automated meeting
8:47PM 2 Calls from PSTN on SPA3102
8:31PM 4 Nat Phone in Asterisk 10
2:53PM 0 asterisk and iax2 errors
2:05PM 2 Starting asterisk turns bash console text white in rxvt
12:53PM 0 Endpoint Capabilities?
12:33PM 3 sip issue
11:09AM 13 Problem with Atxfer for the calling party
10:14AM 3 Temporarily disabling voicemail recordings (but not greetings)
Sunday October 30 2011
10:23PM 1 Meetme does not return back to the dialplan
1:33PM 1
Saturday October 29 2011
6:14PM 4 Issues with Xorcom astribanks when running Ubuntu 11.10 (oneiric) as a quemu/libvirt guest
8:43AM 3 googleapps calendar
5:13AM 9 Queue announcements and MOH blanking out on calls from PSTN over IAX2
Friday October 28 2011
8:13PM 0 File permissions mysteriously changed
7:10PM 1 Network testing for VoIP
Thursday October 27 2011
11:02PM 12 Asterisk Executing outbound dial number twice
3:53PM 6 Recording a meetme conference
3:30PM 8 Sangoma Card with 16E1 SS7 signaling
2:48PM 0 DAHDI spans up without physical connections
2:36PM 3 Check which client access Asterisk using AMI
12:15PM 0 OPTIONS support for SDP
10:04AM 7 Unknown warning
8:58AM 0 Still having trouble to configure gxw4108 with asterisk 1.8 need enlightenment
12:53AM 1 Asterisk 1.8 RealTime problem with ipaddr field
12:16AM 5 Tips & best practices for asterisk troubleshooting & parsing logs
Wednesday October 26 2011
10:11AM 0 OPTIONS to determine codec capability before an INVITE
Tuesday October 25 2011
6:13PM 0 OPTIONS to query endpoint capability
4:17PM 0 Asterisk 1.8 and CDR Mysql: Unqiue ID and if any commands required in extensions.conf
1:28PM 6 Concurrent call monitoring
12:25PM 11 bug in queuemanager?
11:30AM 3 Asterisk does not accepts SIP registration
Monday October 24 2011
2:32PM 0 [asterisk-dev] SIP, NAT, security concerns, oh my!
11:46AM 0 device state of SIP device is stucked into NOT_INUSE, and cannto be reverted to unavailable
9:46AM 4 Storing a variable at a context and using it in another context
9:35AM 1 Voicemail: playing a message to give option if need to transfer for operator
9:19AM 2 ${CALLERID(num)} after doing transfer from extension to extension
Sunday October 23 2011
2:36PM 1 Questions on IAX client
9:46AM 2 Peer and User Clarification
9:22AM 2 how to know RTP por of a SIP client in
Saturday October 22 2011
8:38PM 3 Strange behavior over Zap chennels
12:29AM 0 DAHDI-Linux and DAHDI-Tools Released
Friday October 21 2011
3:45PM 4 Skype Messaging with Asterisk 10?
1:40PM 0 No Voice path during NCS call with Asterisk 10.0.0
12:46PM 0 Video Softphone
11:54AM 1 Asterisk dialplan macro output
7:46AM 2 how to know RTP por of a SIP client in the dialplan
Thursday October 20 2011
10:16PM 1 Astricon: GPG Key signing event
9:41PM 2 10.0 CallerID question
7:58PM 0 problems getting to run
7:45PM 3 Cutting noise and voice
4:01PM 6 question about queues.conf
2:28PM 3 Unable to build sip pvt data
1:28PM 3 RTP ports used by Asterisk in dialplan
11:09AM 0 Video call Setup in Asterisk 1.4.17
10:53AM 1 elegant way to change codec whn failing over to another line
10:35AM 0 Call popups with Thunderbird (and potentially other PIMs)
9:47AM 1 Sipp and asterisk 10
9:05AM 0 ASR & ACD Analysis & Monitoring from Master.csv
8:12AM 3 Monitor does not work well (little cuts in the audio file)
Wednesday October 19 2011
9:35PM 7 Can we use MySQL native connector for ARA?
8:22PM 1 Asterisk replying 491
7:45PM 2 G729 and Dahdi: Inbound forcing ulaw!
7:25PM 1 Problem E1 PRI
3:37PM 2 Problem with video phone call, error in sdp media handling?
3:33PM 11 Asterisk call transfers not working
1:47PM 1 DTMF fun
12:41PM 3 strange delay behaviour in SIP call with same codec
11:58AM 1 Asterisk sponteanous reboot : core dump file
10:34AM 0 Detecting Special Information Tone in Asterisk
9:50AM 10 Running as non-root
5:13AM 2 Outgoing call failure
1:22AM 8 How to use menuselect.makeopts?
Tuesday October 18 2011
11:44PM 3 GoogleTalk Calls
11:15PM 0 DID and how caller id will appear
9:21PM 3 nvfaxdetect in 10.0
3:46PM 0 How does asterisk parse configuration files
2:29PM 0 Asterisk IAX Trunking Stops - Too much delay in IAX2 calltoken timestamp from address
1:35PM 7 Problems during calls
12:27PM 1 make progdocs
12:26PM 4 voicemail
7:30AM 1 Chanspy() not working with group in asterisk 1.4.42
7:08AM 2 How does Asterisk parse extensions.conf file?
5:58AM 0 Chanspy() on group
Monday October 17 2011
11:48PM 2 Conference solution to handle 10, 000 participants - possible at all?
7:22PM 2 Asterisk Centos RPM packages question
6:36PM 0 Asterisk Now Available (Security Release)
5:44PM 0 AST-2011-012: Remote crash vulnerability in SIP channel driver
5:30PM 1 SIP Device and ZAP device
4:32PM 1 forwarding early media
2:33PM 2 Request hangup on local channel
1:29PM 0 Integration asterisk with pabx
1:28PM 0 chanspy() with group
Sunday October 16 2011
9:49PM 0 Voicemail in asterisk 1.8.7, stop working
7:05PM 0 PRI E1 call termination issue
6:55PM 0 **OT** Fwd: oFono 1.0 has been released
5:51AM 6 Any help with these error messages???
Saturday October 15 2011
9:12AM 3 Emulate and script emulation of users calling in/receiving calls, transferring calls &etc
Friday October 14 2011
8:00PM 0 Correction: Scheduled Maintenance for Asterisk Project community services
3:29PM 21 Problem with outbound dialing from remote phone
10:21AM 7 one way voice with IVR
9:17AM 1 Get the total amount of lines/channels for a SIP-trunk?
Thursday October 13 2011
9:04PM 12 Free ticket for Astricon
7:03PM 0 Cisco 3925 Integrated Services Router
4:21PM 3 Phones flapping with * and Sonicwall.
9:09AM 0 many sip dialog/ opened channels.
12:24AM 4 Asterisk 10 'database show' CLI command
Wednesday October 12 2011
9:27PM 16 Binding asterisk to two static IPs
8:25PM 0 Scheduled Maintenance for Asterisk Project community services
5:49PM 1 Asterisk HoneyPot
5:24PM 1 AGI not Installed?
8:13AM 3 failed to extend from 512 to 676
3:42AM 6 FXS ports on TDM410P card...
Tuesday October 11 2011
8:15PM 5 permit -- deny not working
5:11PM 2 Question on meetme and t option
4:58PM 13 Reporting for Asterisk Call Center
4:13PM 0 Call deflection with Libpri/Dahdi on BRI/PRI lines
4:06PM 28 CallerID inconsistently presented through ISDN/cellular networks
1:53PM 3 Asterisk to asterisk IAX trunk
11:50AM 0 Failure to write to tcp/tls socket
11:41AM 0 Asterisk 1.8.7 and VoiceMailMain
11:16AM 2 BT line: unavailable vs withheld numbers?
10:55AM 0 Is it recommended to let Asterisk run with "backtrace options"
9:24AM 2 Queuing strategy
6:50AM 3 t.38 interop with metaswitch
5:58AM 4 Asterisk Incorrect information in Queue events-AMI
Monday October 10 2011
11:32PM 2 Queue calls to agent end prematurely with diastatus cancel
9:35PM 10 Maybe slightly OT but..
3:40PM 2 Dialout from MeetMe to another conference (Asterisk 1.4)
12:08AM 1 Which mISDN required for chan_misdn in 1.8 & 10?
Sunday October 9 2011
1:40PM 4 Grandstream GXP2000 - copy configuration from handset
Friday October 7 2011
5:17PM 0 DIDs in Singapore
3:03PM 2 Add SIP diversion header in originate from AMI?
1:01PM 5 Asterisk 1.8.7 and client outside network
12:46PM 11 Asterisk 1.8.7 and ReceiveFAX
11:20AM 0 Problem With Playing Busy Tone
Thursday October 6 2011
9:25PM 3 Cisco AS5400XM
8:37PM 15 Digium FFA + Gafachi T38 outgoing issues
4:25PM 4 Which SIP phone LCD expansion module and >100 asterisk-compatible BLF ?
2:57PM 13 dahdi show status command not avilable in CLI
7:12AM 7 A manager event whenever an hint value changes
12:28AM 1 Questions on Dahdi
Wednesday October 5 2011
8:06PM 0 Asterisk 1.8 Manager Perl Script Problem [SOLVED]
7:36PM 4 making announcements
7:04PM 1 Different revisions of Digium cards
6:42PM 3 call pickup
4:17PM 9 meetme
8:41AM 0 Passive wait in dialplan
8:35AM 6 Passive wait in dialplan?
6:36AM 4 Reinvite dialplan application [Was: OT - SIP - Toggle to autoanswer after ringing]
6:30AM 3 Reduce the wav file size
Tuesday October 4 2011
6:40PM 9 Asterisk (Trixbox) - VirtualBox - Linux Host
6:25PM 1 Database Lookup Advice
5:48PM 5 music on hold
3:15PM 0 OT - SIP - Toggle to autoanswer after ringing
2:21PM 23 Beep file with Record
11:54AM 0 Lag with Call Transfer (Patching)
8:33AM 2 rtp.conf and Asterisk as a sip agent/client
Monday October 3 2011
11:38PM 3 Delay before ringing from PSTN`s call
10:01PM 0 Asterisk 1.8 Manager Perl Script Problem
5:51PM 2 Keeping Voice Call Active During Data Connectivity Loss
Sunday October 2 2011
4:05PM 2 Asterisk in the Cloud with Diamonds
3:20PM 7 Sipgate trunk doesn't bridge with other trunk, but works with local extensions
11:53AM 1 asterisk_1.8.7.0-1digium1 100% CPU
Saturday October 1 2011
10:30PM 3 Make asterisk cluster appear and operate as a single server?
10:02PM 1 Converting dahdi_monitor unit to dbm0