asterisk users - Jul 2012

Tuesday July 31 2012
3:41PM 0 As Kevin Fleming says "So long, and thanks for all the fish!", we say thank you - and look to the future
1:59PM 4 So long, and thanks for all the fish!
12:22PM 3 Digium IP Phone D40 quality, very bad
8:44AM 1 Static noise on bridged calls to PSTN, although the trunk line is clean on its own
8:43AM 0 AGI not generating sip 180/183 status
Monday July 30 2012
7:42PM 0 Asterisk 10.7.0 Now Available
7:42PM 1 Asterisk Now Available
7:06PM 0 111 Useful and/or Funny New Prompts For Asterisk, Courtesy of Allison Smith
11:02AM 1 problem with 8 port card
10:08AM 2 libpri error
9:36AM 4 Multi-Tenant PBX with Asterisk
9:11AM 1 (no subject)
8:43AM 0 (no subject)
Sunday July 29 2012
7:34PM 0 just did sched_add waitid Warnings
Saturday July 28 2012
10:43PM 3 best PRI gateway?
3:38PM 1 How to send a SIP MESSAGE outside a call
2:18PM 2 Asterisk + Google Voice
9:43AM 3 No audio playing back voicemail from odbc
9:19AM 2 Multiple DID for SIP Trunk
6:58AM 2 MixMonitor creating file on non-bridged calls with option b
4:58AM 3 Asterisk on Dynamic IP to a SIP extension
Friday July 27 2012
4:55PM 1 CAS T1 - No Ringback
2:07PM 0 MWI not working - Asterisk
11:31AM 0 (no subject)
3:33AM 1 still got ReceiveFax() problem, how to properly setup asterisk fax?
Thursday July 26 2012
11:21PM 1 asterisk crash
10:03PM 2 Call ID of the second call leg
8:28PM 2 Polycom Presence with Asterisk
5:34PM 1 Asterisk Realtime issue after registering with x-lite
1:53PM 0 Realtime Queue and Queue_members
1:02PM 3 callback on busy
10:37AM 1 callback - disa
10:08AM 2 What TTS to use?
5:22AM 1 Confbridge examples for Asterisk 10?
1:09AM 1 Dahdi+Redfone+Channel Bank+E&M
Wednesday July 25 2012
7:35PM 1 SIP/GSM-gateway recommendation?
6:24PM 4 Video conferencing?
1:42PM 0 How to play DTMF digits without blocking
1:10PM 1 res_odbc crashing asterisk after freetds dsn reconnects
9:17AM 2 Asterisk
Tuesday July 24 2012
9:06PM 1 passing arguments to macros from originate command
8:46PM 1 echo canceler query
4:45PM 5 DAHDI problems
2:19PM 2 Video call using Asterisk
9:22AM 0 Play announcement during Dial just once
8:37AM 2 Finding the position of a character in a string
Monday July 23 2012
4:23PM 1 Asterisk 1.8.12 and Fax?
3:40PM 0 Digium Phones: Heads Up
11:30AM 2 T.38 Gateway
7:55AM 2 file and on SayNumber() app
4:22AM 8 PBX, IVR and Conferencing Platforms From the Same Installation of Asterisk
Saturday July 21 2012
10:35PM 1 - SIP retransmission problem
5:46PM 2 Less good call quality using Asterisk
12:24PM 0 Asterisk and IPTV
Friday July 20 2012
5:53PM 4 Voicemail Emails
2:48PM 1 T.38 (PRI) Fax Debugging
12:22PM 0 SIP Inband DTMF problem
7:15AM 2 freepbx asterisk
Thursday July 19 2012
11:30PM 1 Route incoming calls
8:49PM 1 Channel is rsrvd and does not turn off
2:34PM 1 Agent receives call while making calls
Wednesday July 18 2012
9:35PM 0 Can not get my Eicon Diva running with Asterisk...
4:08PM 4 Remote party ID - sort of working...
3:06PM 1 Asterisk 1.8.13 / res_fax / res_fax_digium
11:44AM 5 How to work around asterisk ss7
11:30AM 3 Using Asterisk 10.6 as a T38 Fax gateway
10:53AM 1 Telecom HU cannot callforward to external number
6:27AM 4 asterisk 1.8 on Solaris/sparc
Tuesday July 17 2012
6:10PM 0 Unplanned service outage within next hour for Asterisk community services
1:29PM 0 How I could Insert a record into the table with REALTIME
10:24AM 0 Asterisk Capacity
Monday July 16 2012
6:47PM 2 any working calling card solution "open source"
4:44PM 0 Asterisk 10.6.1 Now Available
4:44PM 0 Asterisk Now Available
11:48AM 1 Changing auto mixmonitor output file name
11:35AM 1 Getting error while dialing outside (PRI Connection card TE110P ASTERISK1.8 AND FREEPBX 2.10
1:23AM 1 QoS : tos and cos settings
Sunday July 15 2012
4:40PM 0 Mediatrix 4400plus ISDN setup
2:17AM 0 asterisk-users list testing -
1:46AM 0 asterisk-users list testing -
Friday July 13 2012
9:42AM 8 How to set SIP to auto answer in the dial plan .
9:00AM 2 Recommended VOIP Monitoring Tools
8:56AM 1 trouble with asterisk behind router
4:17AM 7 How to Auto Answer a sip phone
Thursday July 12 2012
8:36PM 2 Issue with a ticket system subscribed to asterisk-users
5:56PM 0 chan_sip sending from wrong source, address when multiple interfaces are used
5:38PM 1 chan_sip sending from wrong source address when multiple interfaces are used
12:55PM 1 Asterisk with OpenBTS and mobile phone
12:53PM 1 weird dect beheaviour multiple handsets
12:07PM 0 chan_ss7 quick patch to enable RBT
4:18AM 0 Audiocodes 310HD - on Asterisk Server
2:08AM 0 wrong RTT QoS information always reported
Wednesday July 11 2012
2:08PM 1 Inconsistency in CDR between NO ANSWER and BUSY calls
7:35AM 3 click to call
Tuesday July 10 2012
9:09PM 0 Planned service outage for community services on July 12, 2012
8:15PM 0 Asterisk 10.6.0 Now Available
8:15PM 0 Asterisk Now Available
4:20PM 2 Flowroute: howto set outbound callerid (ast 1.4)?
3:42PM 1 10.6.0-rc2: tmp full of core.PBX
3:07PM 1 connections to manager
2:04PM 3 channel not available and jump to next group channels
Monday July 9 2012
3:04PM 0 Queue timeoutpriority=app doesn't working as explained in conf.sample
1:24PM 2 seems like call is picked and returned to me
9:26AM 2 SRTP Encryption Per Device
Saturday July 7 2012
6:48PM 2 Rookie / sip and extensions
9:46AM 1 Trixbox or FreePBX or Elastix or PBX In a Flash
Friday July 6 2012
9:18PM 1 sip.conf and binaddr issue
9:18PM 1 Trixbox or FreePBX?
9:16PM 1 Can I install Asterisk normally and then installing the GUI
9:01PM 1 Maximum concurrent calls using call files
3:53PM 2 DAHDI DTMF problem?
3:15PM 1 Asterisk trying to call a queue with no members
1:34PM 1 Can I install Asterisk normally and then installing the GUI "asterisk now"
10:56AM 2 call file and NFS server
Thursday July 5 2012
10:25PM 2 FreePBX: How to hangup if the caller did not press # after the voicemail message
10:20PM 7 FreePBX: using context other than the default context and the generation for the configuration
9:44PM 2 sip and extensions
9:04PM 0 AST-2012-011: Remote crash vulnerability in voice mail application
9:04PM 0 AST-2012-010: Possible resource leak on uncompleted re-invite transactions
5:47PM 1 touch command not behaving for future calls in asterisk 1.4.41
4:00PM 0 Asterisk 1.8.11-cert4,, 10.5.2, 10.5.2-digiumphones Now Available (Security Release)
2:15PM 1 OT - Multi Function Printer with one touch scanning/emailing
1:21PM 4 OT - Integration with building intercom systems
9:34AM 1 sip set debug on always showing error
7:19AM 1 Elastix
6:52AM 1 Regrading Speech Recognition.
2:20AM 2 basic sip quesiton
Wednesday July 4 2012
5:50PM 0 Asterisk 1.8.13 PlayTones App
11:32AM 1 Timer1 RFC and SIP.CONF
9:32AM 1 Queue Member login from IAX trunk
Tuesday July 3 2012
10:35PM 0 QUEUEMEMBER_STATUS incorrect?
9:08PM 1 Outbound Asterisk calls default directmedia specifications
8:21PM 0 Free PBX: hangup even if did not dial # in the voicemail
3:56PM 1 IAX trunking stopped working
12:30PM 3 How to play different different hold music.
11:43AM 3 AMR - Segmentation Fault
7:38AM 3 Centos 6 mISDN
Monday July 2 2012
9:15PM 1 Call recording and hosted PBX platform using Asterisk
5:05PM 0 (no subject)
2:55PM 0 Allworx 9212
Sunday July 1 2012
8:46AM 1 T.30 Fax session error: Received bad response to DCS or training
7:34AM 5 port 5060 is blocked by ISP