asterisk users - Nov 2011

Wednesday November 30 2011
10:39PM 1 Installing asterisk on a server vs appliance(e.g digium mypbx)
9:20PM 2 Walkie talkie to sip phone interface
8:47PM 3 how to find out one way latency
7:55PM 1 Sound files with MixMonitor not playable with Media Player
4:08PM 0 vall directly extensions from E1-PRI line
3:26PM 2 Issue with Polycom SPIP650 and its sidecar
1:38PM 1 Question on PAP2 linksys showing off-hook
12:25PM 1 s/n ratio detection etc...
6:51AM 1 Best VoIP conferencing phone ?
1:05AM 0 SLA and polycom
Tuesday November 29 2011
11:03AM 3 When dialing the number, I need to see it in the Cisco LCD Phone
10:47AM 2 SIM to E1 gateway, and SMS gateway
10:38AM 2 OT: Does IEEE 801.2q include VLAN trunking?
9:50AM 3 app_mysql and asterisk 1.4
Monday November 28 2011
8:30PM 1 Recommendations
3:29PM 2 Call Parking Realtime
1:32PM 0 AsteriskGUI - Which version for 1.8 ?
8:47AM 1 Queue-Tip/Adhearsion installation tip
1:22AM 1 centos asterisk 1.8 rpms: chan_gtalk and res_jabber missing?
Sunday November 27 2011
10:38PM 0 Displaying entered digits in the LCD of the IP Phone when is requested to enter it
10:04PM 0 SMS problems.
5:47PM 6 Does Asterisk alter the Headers of INVITE Message
6:33AM 0 Automatic IVR generator (alpha)
Saturday November 26 2011
11:55AM 3 A new hack?
Friday November 25 2011
10:11PM 1 android won't play wav49: how to change format
6:00PM 0 Install Adhearsion on Debian [SOLVED]
5:19PM 1 Install Adhearsion on Debian
3:32PM 1 hwo to stok variable wiith menu
1:18PM 1 Sporadic yellow alarms in dahdi_tool output
7:57AM 3 Call Parking
5:18AM 1 Rgarding asterisk 10 stable release
Thursday November 24 2011
10:36AM 2 PSTN Frequency parameters
Wednesday November 23 2011
12:03PM 1 MWI for non-subscribed Realtime peers?
11:37AM 2 Is it possible call land into extensions.ael configuration file not in extensions.conf
7:16AM 3 safe_asterisk ?
6:40AM 0 how to call a ring group via the dial plan language in asterisk?
2:46AM 2 Sending more than one call the agent while he is already in a call !! ringinuse=no/yes
Tuesday November 22 2011
11:22PM 1 Receive sms via bluetooth with chan_mobile and an Android phone?
11:00PM 1 Follow me unreachable message default
10:34PM 4 How to program a 100ms delay between the ringing of queued calls w/ ringall
3:00PM 2 sip show peers
2:40PM 1 Asterisk refuses INVITE (401) and I don't know why
7:14AM 2 Resell VoIP Servcies
6:57AM 0 no sound with ICES ?
Monday November 21 2011
10:17PM 1 AEX410P drops DTMF digits
5:50PM 3 difference between playback and background?
5:43PM 1 CDR uniqueid - across multiple servers?
5:30PM 0 no more trunk with asterisk
4:56PM 1 queue ring delay
12:13PM 2 CDR mysql with asterisk 1.4
10:35AM 1 video calls not working
10:20AM 1 vigor 2920 problems
10:12AM 0 Dependencies for BETTER_BACKTRACES on Centos 5.6
6:42AM 7 How to use password file with Authenticate Application
6:38AM 2 Continue AGI after Dial() following caller hang up?
Sunday November 20 2011
11:32PM 0 hello
8:49PM 1 check if devices reachable in queue
7:25PM 4 Deleting AstDB family at start
2:43AM 1 SIP registration issues
Saturday November 19 2011
8:39PM 0 Queue: The call keep going to agent until the agent drop the call
Friday November 18 2011
8:02PM 2 Setting outbound PRI Callerid with Asterisk 10.0.0-beta2
7:23PM 4 Using asterisk with DSP chips
7:10PM 1 I want to participate in development
2:33PM 1 Polycom Phantom Ringing
10:29AM 1 Call files and spool directiory shared amongst several asterisk servers
10:16AM 1 Asterisk Developer
Thursday November 17 2011
6:13PM 0 2 same sip extension number on 2 asterisk - call not passing on certain condition
4:10PM 1 Question about Read() application
3:44PM 0 DTMF dropping in Read Command
2:20PM 0 10-rc2: how to debug dropped calls?
1:17PM 1 How to unregister a sip trunk
1:00PM 0 ast_debug messages not showing up
11:30AM 2 Fax not detected by Asterisk
11:00AM 3 AMI: anything to glue originate to events?
6:34AM 0 Use Polycom FX with Asterisk
1:46AM 0 Grandstream HT503 colgado
Wednesday November 16 2011
6:49PM 5 Polycom Attended Transfer
6:06PM 2 polycom soundpint ip650 question
1:46PM 4 Limit monthly calls by context
12:01PM 9 Skype For Asterisk (SFA)
10:18AM 1 Server-to-server BLF
9:23AM 3 Does Asterisk Support SIP Video Call ?
5:06AM 1 Asterisk Send out SIP Invites to external network- howto
Tuesday November 15 2011
3:58PM 3 Standard UIDs, especially for asterisk?
3:12PM 2 More than one route to a destination
2:47PM 2 Forcing a CODEC
1:28PM 4 Multiple SIP endpoint registrations
10:56AM 2 Goto Queue, does not work, it should play message or any thing
6:01AM 2 Calling an independent gateway from asterisk
1:45AM 1 asterisk bin file may change when running
Monday November 14 2011
11:51PM 2 trouble with sip connection and registration
9:51PM 2 How do extensions "stay registered"
6:47PM 1 Becoming a CLEC
3:40PM 0 Dial Application / hangup with option h or H / featuremap / more than 1 valid key
2:08PM 2 TE122
1:34PM 1 Monitor() - splitting long calls into several sound files
9:47AM 0 Asterisk 1.6 AEL Macro vs GoSub
5:54AM 2 unavailable state not reported to Cisco SPA50X phone
3:25AM 1 Call to Asterisk registered sofphone from an independent unregistered Endpoint
Sunday November 13 2011
10:19PM 2 Logging Specific Verbose Level To Seperate File
8:55PM 0 shared_lastcall for 1.4.42
10:05AM 0 Music on Hold does not " Kick-in" until second try , on outgoing calls.
Saturday November 12 2011
1:39PM 1 Warning from Exchange Calendar
Friday November 11 2011
10:50PM 2 10.0.0-rc1: dahdi doesn't see card
10:19PM 1 10.0.0-rc1: won't start: "empty buf size"
8:36PM 0 Text to speech modules (espeak, flite)
5:26PM 3 crashing : Can't send 10 type frames with SIP write
12:08AM 1 What the variable that return the IP Phone username to use it for AddQueueMember
Thursday November 10 2011
8:44PM 1 FXS - Power Alarms
6:10PM 1 Frequent Asterisk Restarts
5:07PM 0 DTMF issue with and SIP Trunks [WORKING]
4:38PM 2 Asterisk 10.0.0-rc1 Now Available
3:38PM 5 DAHDI has broken by pbx :-(
3:28AM 1 IAX2 availability testing
12:03AM 1 DTMF issue with and SIP Trunks
Wednesday November 9 2011
11:01PM 0 ayrv2by jg4yjbf3r
5:09PM 1 ConfBridge 1.6.20 user count
10:22AM 1 Permanent sip and agi debug on?
Tuesday November 8 2011
10:47PM 2 Licensing question.
5:09PM 0 Asterisk lost registration bug with NAT keep-alive
2:51PM 1 No call progress sounds
9:45AM 0 Many SIP-480 responses
6:44AM 1 Realtime Queue - changing strategy to linear needs Asterisk restart
Monday November 7 2011
2:28PM 4 4 sec delay in voice menu (asterisk)
1:38PM 1 Missing core functions.
8:46AM 2 Dahdi complete - dahdi_dummy not compiled
Saturday November 5 2011
9:23AM 1 Where do I find error message descriptions?
Friday November 4 2011
12:21PM 0 augeas lens asterisk
12:15PM 0 What does this error mean?
9:34AM 2 9. any live queue monitor recommendation? (Jean Chassoul)
7:57AM 1 problem when exiting from "record file" function without pressing the escape digit
12:10AM 0 Queue status question.
Thursday November 3 2011
11:59PM 0 any live queue monitor recommendation?
11:20PM 15 DID from Direct from Telco
6:36PM 1 2 pbxes
5:04PM 1 Asterisk as SoftSwitch - Hardware
10:10AM 2 duration limits in Dial() not being enforced at correct time
Wednesday November 2 2011
5:41PM 1 Change indications in Dialplan
2:56PM 1 Unable to build sip pvt data - Switching equipment congestion
2:06PM 1 libpri : Q.931 Called Party Number interpreted as empty
1:08PM 1 Option 'd' of application Dial not working in 1.8.8-rc2
4:47AM 2 FFA - Asterisk
Tuesday November 1 2011
5:08PM 10 State of Asterisk+Virtualization+Timing