asterisk users - Apr 2006

Sunday April 30 2006
TimeRepliesSubject
11:58PM 0 how to make messages button on ip500 work
10:42PM 4 PRI Issue: D-Channel woes
6:36PM 2 WARNING[12785]: acl.c:244 ast_get_ip_or_srv: Unable to lookup '????'
2:19PM 35 FreePBX in production?
12:46PM 1 integrated voip originator, to digitize audio once and only once?
10:57AM 1 Asterisk 1.2.7.1 & Fritz!PCI or AVM A1
10:05AM 6 Asterisk is stripping my area code
10:01AM 1 Change in audio file while listening to it
9:22AM 7 Legacy PBX integration
9:18AM 4 newbie-too much latency
8:41AM 0 Intermittent problem dialling out on a SIP channel
6:21AM 0 Fwd: can modify CHAN_SIP.c to generate a new exten=> ext, 2, dial(tech/peer) ?
5:56AM 0 some sip clients unreachable on sip-reload
5:35AM 1 Error : ast_readaudio_callback: Failed to write frame
4:16AM 6 How to monitor DTMF tones in a call?
12:56AM 0 PRI got event: HDLC Bad FCS (8) on PrimaryD-channel of span
 
Saturday April 29 2006
TimeRepliesSubject
10:33PM 3 problame with outbound calls on pri
8:55PM 0 NuFone - How to switch to another provider?
8:53PM 13 Compare to Skype
8:11PM 0 [OT]Cisco 2621XM with (2) T1/PRI inetrfaces for sale
7:49PM 5 Codec G729 no longer works.
6:17PM 11 (Semi-OT) QoS Question FTP Living with Asterisk
6:11PM 2 RE: Install/Upgrade
6:02PM 1 Large Asterisk with Regexten, Regcontext, DUNDi, , , , , , , , , but not load balance...
4:50PM 2 Unable to Make Asterisk-addons
11:40AM 0 Audio Muting at seemingly random times
9:58AM 0 Locate Me Function with freePBX
9:04AM 2 How many asterisk process's are "normal"?
8:13AM 1 Telephone support charging system with Asterisk?
7:35AM 0 canreinvite, bandwidth, dial option
5:19AM 1 asterisk to use an outbound proxy
4:13AM 2 NOTIFY Problem
4:12AM 2 Help with Mediatrix 1204
1:00AM 0 Is there a way to monitor the DTMF tones on a channel?
 
Friday April 28 2006
TimeRepliesSubject
9:37PM 3 stupid trick of the day (fried polycom)
9:06PM 1 IAX + GSM codec is good quality
8:02PM 1 [SPAM] [asterisk-dev] Disable 407 proxy authentication for outbound domains
7:10PM 4 Random 1-way audio on IAX2 Connections
7:06PM 5 Call Queue Transfer
6:51PM 1 Remote UNIX connection disconnected over and over
5:39PM 2 two box share one real time configuration database.
4:53PM 0 Rhino T1 and 4-port FXO cards
4:34PM 0 How to use the cmd SMS
4:11PM 2 Asterisk DNID/RDNIS with Dial iax2
3:29PM 1 Bristuff 1.2.7.1?
3:07PM 1 Cell phones and DTMF
2:36PM 0 Asterisk and Panasonic KX-T336
2:27PM 0 Digium TE210P and faxing, is it possible?
1:26PM 0 DNSMasq - Why the stuff hits the fan when the net connection is down
12:19PM 7 Odd internal vs. External dialplan issue
11:48AM 0 Configuration OpenPri for logger
11:07AM 6 Dual Timing Sources
10:33AM 3 Official TE411P echo settings??
10:12AM 0 What is i2 ? 911 Candian Style
9:57AM 0 New astGUIclient VICIDIAL Release 1.1.11
9:20AM 0 OT: Phishing with phones
8:34AM 1 Basic Linux Advice
8:12AM 3 Problems if GXP-2000 phones and Asterisk are not on the same network
7:44AM 1 RESOLVED - TE405P vs. SoundCard problem (in reality - TE405P No Voice Problem)
7:12AM 1 Warning: No path to translate with SJPhone
6:50AM 2 Integrics release Enswitch 2.0
6:14AM 3 How to transfer outgoing calls
5:29AM 5 Dial 'R' option gone?
4:27AM 0 IVR answers and questions instead of MOH in a queue, how?
3:39AM 3 caching of sip account
3:21AM 0 video support on iax trunk
3:18AM 0 Which h323 channel for asterisk and gnugk ?
2:02AM 0 how to dial the real time iax user
1:43AM 2 Asterisk dialing
1:41AM 0 Help on multiple dialed sip-channels.
1:35AM 2 mISDN: No DID/extension information returns busy to caller
 
Thursday April 27 2006
TimeRepliesSubject
8:45PM 0 HINTING, how it works... Please explain
6:00PM 0 How can conference room can call out?
5:48PM 0 Info system
4:36PM 0 Call Pickup with CID info
3:35PM 1 Polycom 501 unregistered itself?
1:21PM 0 Looking for input on which way to go with smallbusiness setup
1:20PM 0 Please Help i have a error with unicall and AT&T
1:08PM 7 Looking for input on which way to go with small business setup
12:09PM 0 What happened to my subscription?
11:50AM 1 PrivacyManager & FastAGI: Rewrite or use?
11:22AM 0 createlink option in agents.conf can't be disabled?
11:10AM 9 Recomended Commercial PBX Bundles/Software
11:03AM 2 Snom 320 HOLD and TRANSFER not detected
10:43AM 1 Analog GSM Gateways
10:26AM 1 Excessive Asterisk delay to answer on ZAP inboundcall
9:55AM 0 Receive fax (libtiff problem?)
9:22AM 16 Polycom NTP issue
9:18AM 17 PRIs from two different telco
8:46AM 0 access to caller/pickupgroup in extension.conf
8:03AM 1 Slip/Frame Error between Mitel SX-200 and Asterisk
7:38AM 2 Very stupid question regarding Polycom Soundstation 4000
7:12AM 0 Guest Account - SIP and IAX
7:07AM 0 chan_sip.c patched with t.38
6:58AM 5 PRI configuration
6:28AM 4 Seize phone line
6:03AM 2 asterisk spandsp and txfax
5:46AM 4 TE405P vs. SoundCard problem
5:46AM 2 Transfer - context/priority
5:34AM 0 Need help configuring Asterisk with Alepo
5:19AM 0 URGENTS: seek people for video tests with asterisk/ser/rtpproxy + eyebeam
5:10AM 10 Interesting Dial-Plan Question
4:17AM 0 zt_pri-error
3:27AM 0 Autodial feature doesn't return $DIALSTATUS values
2:17AM 1 Asterisk to Dial a number , after getting a mail notification ,
2:07AM 1 Asterisk Voice Problems
1:55AM 0 GXP-2000: disable provisioning
1:37AM 3 Asterisk Hangs the whole system
12:32AM 6 SATA hard disk compatibility
12:29AM 0 SV: treating an incoming call as a local extension
12:03AM 7 Extreme delay before * processes call files
 
Wednesday April 26 2006
TimeRepliesSubject
11:35PM 0 replacing step-by-step giving echo
9:25PM 1 Accessing PARKEDAT variable in AGI
9:21PM 2 treating an incoming call as a local extension
9:19PM 2 getting asterisk to reliably answer a voip line
9:09PM 0 func_odbc and 1.2.7.1
8:56PM 8 Unable to accept incoming PSTN calls
7:30PM 10 Asterisk as a phone survey system
6:50PM 1 Paging on Aastra analog phones.
5:55PM 0 no audio for ring group.
5:35PM 2 Problem with a TDM-400P
4:28PM 1 Problems with Eicon Diva V-4BRI - 2nd Port
2:17PM 0 Hook Flash via SIP INFO command?
2:06PM 0 cell mobile network (GSM) to Asterisk
2:01PM 2 Status of Queue
12:44PM 0 clipcomm versus sipura/linksys
12:43PM 0 Help! * Won't Start after SVN Trunk Update - SuSE 10
12:43PM 0 A@H and channel announcement
12:41PM 1 Phone Emergency - Need IAX Help
11:43AM 1 Question about the zaptel-1.2.5-patch
10:27AM 1 cannot transfer to call waiting call on ip500
10:24AM 0 Callback help
10:00AM 1 Explain to me VoIP termination service.
9:52AM 0 How to configure Asterisk to handle multiple Gizmo accounts?
9:31AM 1 asterisk no longer compiles on gcc 2.95
9:25AM 0 (Wanted OEM VOIP and SKYPE PRODUCTS)
9:16AM 0 Re: Asterisk-Users Digest, Vol 21, Issue 132
8:17AM 2 Early media after a dial command
8:05AM 0 kernel - module problem
7:33AM 2 Stuck in Queues
7:25AM 17 Camp on?
6:53AM 0 Re: [Serusers] Sip t38 gateway tests
6:47AM 3 No Caller-ID With Cisco PAP2T-NA
6:27AM 0 RE: SOLVED: No audio when dialing in via PRI with Q.SIG
6:19AM 0 Configuring QoS Params in UIP-200
6:11AM 1 IAX calls dropping after minutes
5:52AM 1 ODBC Storage for voicemail messages in database
5:41AM 14 I am looking for a webphone on MY SITE
5:36AM 9 Sphinx2
5:28AM 2 2 analogue ports on an Alcatel PBX patched to 2 FXO ports on my *
5:27AM 9 astcc: need partial pin code
5:21AM 1 Registering to H.323 Cisco gatekeeper
5:02AM 3 7960G SIP Issue
3:04AM 0 do extensions must be numbers in Asterisk@Home?
2:51AM 1 Sip Phones with BLA Support
2:14AM 0 Avoiding deadlock... Problem
1:59AM 7 Excessive Asterisk delay to answer on ZAP inbound call
1:40AM 0 Stange behaviour on 4port BRI Asterisk 1.0.10-BRIstuffed-0.2.0-RC8q
1:17AM 1 Intergrate Asterisk IP PBX with Legacy PBX, continuing existing funtionality of legacy pbx
12:50AM 1 # and call speed
12:24AM 0 CISCO 7960G - SIP Configuration
12:01AM 0 SV: Need some help on queues with agents(SIP members)with multiple phones.
 
Tuesday April 25 2006
TimeRepliesSubject
11:40PM 7 Need some help on queues with agents(SIP members) with multiple phones.
9:49PM 4 test numbers in different countries!
9:23PM 0 Trying to set up automatic announcement upon
8:25PM 0 Trying to set up automatic announcement upon transfer for IVR in AAH 2.8
6:46PM 0 Here I am facing problem of Voice Breakage
6:22PM 16 USB conference phone
5:06PM 1 Agents <--> Extensions
4:18PM 3 56K Dialup and VOIP over same PRIs
3:09PM 0 Pressing ## end the call and return to menu
2:32PM 2 Touch tone recognition issues
2:21PM 1 queues that do not play music
2:09PM 1 TE410 and 411
1:37PM 3 One Way Audio....in the middle of a call
1:22PM 1 Splitting Zap channels into trunks?
1:05PM 2 FastAGI Connection Failure and Hangup
12:41PM 2 Help on chan_misdn and MSN's
12:30PM 0 Re: Asterisk-Users Digest, Vol 21, Issue 132
11:46AM 4 Sip t38 gateway tests
11:38AM 8 Auto Logout from queue
10:45AM 6 MFCR2 in Brazil, someone?
10:00AM 2 TDM400P: flash on analog phones doesn't work
9:07AM 1 Updated: No audio when dialing in via PRI withQ.SIG
8:49AM 0 Question on connecting to another system
8:40AM 0 Updated: No audio when dialing in via PRI with Q.SIG
8:02AM 10 About Softphone IAX free for Pocket PC
7:19AM 9 Really Old Rotary Phone
6:34AM 1 Lastest stable build
5:51AM 0 Voicemail being cut-off
5:38AM 1 res_perl voor asterisk 1.2.4
4:21AM 0 SQL update failing/long fullcontact
4:16AM 1 Festival , Cannot hear the words after ","
4:07AM 0 No sound in one calling direction, men using PRI with E1 and Q.SIG
3:47AM 1 Another undefined pri_restart failure
3:16AM 5 Background asynchronous AGI
3:05AM 2 PRI got event: HDLC Bad FCS (8) on Primary D-channel of span
1:33AM 22 billing realtime
1:02AM 1 CHANUNAVAIL, busy and congestion
 
Monday April 24 2006
TimeRepliesSubject
11:27PM 0 Development news :: New AEL and configuration system
9:54PM 0 A@H 2.6 : problem connecting call from PSTN
9:23PM 2 Polycom Delay
8:52PM 0 Gsm Gateway , again !
8:45PM 1 Dialing Ring Groups from the Digital Receptionist-
7:47PM 3 (no subject)
7:31PM 2 Asterisk2Billing
5:20PM 0 codec variable for incoming calls
4:32PM 1 [Issue] Does the *-pbx cmd page honour the absolute timeout value?
4:18PM 1 Change name User-Agent
3:02PM 0 chan_gsm_bt Impression
2:58PM 1 Pinouts for T1/E1 crossover cable WAS "RE: what cable to connect a legacy PBX to a TE410P ?"
2:57PM 6 Two asterisk process in one hardware.
2:30PM 1 Re: Shielding of T1/E1 cables WAS RE: PinoutsforT1/E1 crossover
2:20PM 12 Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)
2:15PM 0 this is just a post test
2:04PM 1 E1 testing
1:05PM 0 SUSE 9.3, modprobe, and zaptel
1:01PM 3 Channel Restart and Dropped calls
12:56PM 5 Some questions re. T1 cards & QoS
12:23PM 0 HINTS with Polycom stops working after aster isk reload
11:35AM 3 HINTS with Polycom stops working after asterisk reload
11:34AM 0 Time out if channel does not ring
11:17AM 0 eyeBeam 1.5
10:50AM 1 Zap channels not disconnecting after PSTN line hangs up (getting empty voicemails)
10:50AM 0 Asterisk to Linphone sound playback delay, and then choppy
10:30AM 4 Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover
9:57AM 2 CallerID/variable setting.
9:55AM 3 Question about Asterisk realtime
9:50AM 1 Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover
9:47AM 0 Pinouts for T1/E1 crossover cable WAS "RE: whatcable to connect a legacy PBX to a TE410P ?"
9:39AM 1 Re: Shielding of T1/E1 cables WAS RE: Pinoutsfor T1/E1 crossover
9:07AM 5 SMP kernel on Pent 4?
9:05AM 0 getting listed in Directory Assistance, the phone book
8:54AM 5 SIP HEADER FROM: without CALLERID(name)
8:51AM 0 fxotune Problem
8:41AM 3 Faster Sound Files
8:32AM 0 Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossovercable WAS "RE: what cable to connect a legacy PBX to aTE410P ?"
8:23AM 0 fax and URA
7:40AM 3 Help!!!!! DTMF detection is not working on Zap lines
7:29AM 0 Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover cable WAS "RE: what cable to connect a legacy PBX to a TE410P ?"
7:01AM 0 strange problem with Telasip DID, please help
6:49AM 1 Queue reload
6:47AM 0 Digium cards for sale
6:45AM 12 Shielding of T1/E1 cables WAS RE: Pinouts for T1/E1 crossover cable WAS "RE: what cable to connect a legacy PBX to a TE410P ?"
4:57AM 5 User Defined VoiceMail announcement?
4:41AM 2 Quintum D3000
4:16AM 3 outbound calls to sip urls
4:15AM 2 1.2.4/7 and chan_modem
4:06AM 15 Hi...Please help me
3:46AM 1 compiling zaptel-1.2.5
3:38AM 2 Dreadful results from zttest with TE210P and Dell 2850?
2:16AM 3 MeetMe Call Out to invite
1:14AM 1 X100P Polarity Reverse Detection
12:52AM 2 annoying noise on analog phones on tdm400p
 
Sunday April 23 2006
TimeRepliesSubject
11:54PM 0 sending special infoa fter login
11:41PM 0 1/3 packets are reported dropped by tethereal
7:48PM 0 Clearpath?
6:36PM 0 SPA3000 in Singapore
6:32PM 0 Re: Asterisk-Users Digest, Vol 21, Issue 132
5:10PM 1 Zap - Cahnnel bank - one way audio
2:56PM 1 Asterisk hangs up on incoming PSTN line to analog extension
12:58PM 0 New backport of T.38 fax passthrough functionality to asterisk-1.2.7.1
12:01PM 4 call queue problems
10:49AM 0 asterisk at home, broadvoice and iptables
10:40AM 6 E1 connexion
9:20AM 0 RE: Asterisk-Users Digest, Vol 21, Issue 130
9:12AM 0 RE: Asterisk-Users Digest, Vol 21, Issue 130
8:38AM 0 Asterisk and SER hangup issue
6:08AM 0 Routing through ENUM
3:58AM 2 Accessing functions from AGI
3:43AM 1 SIPredirect
3:19AM 1 FritzCard, mISDN & "Anlagenanschluss"
3:17AM 5 Codec G729 / x86_64 bits.
2:54AM 1 Setting up a t38 fax gateway
 
Saturday April 22 2006
TimeRepliesSubject
6:39PM 7 Need help with getting EXTEN from pstn hunt group
3:18PM 0 Pinouts for T1/E1 crossover cable WAS "RE: whatcable to connect a legacy PBX to a TE410P ?"
3:14PM 1 Pinouts for T1/E1 crossover cable WAS "RE: whatcable to connect a legacy PBX to a TE410P ?"
1:05PM 0 Asterisk on FreeBSD + Passive ISDN BRI
11:21AM 1 Pinouts for T1/E1 crossover cable WAS "RE: whatcable to connect a legacy PBX to a TE410P ?"
9:34AM 0 Pinouts for T1/E1 crossover cable WAS "RE: whatcable to connect a legacy PBX to a TE410P ?"
9:26AM 1 PANASONIC KX-TS208W - Speakerphone IncompatibleWith Asterisk 1.2.3
9:13AM 3 PANASONIC KX-TS208W - Speakerphone Incompatible With Asterisk 1.2.3
9:07AM 0 Pinouts for T1/E1 crossover cable WAS "RE: whatcable to connect a legacy PBX to a TE410P ?"
8:59AM 13 Pinouts for T1/E1 crossover cable WAS "RE: what cable to connect a legacy PBX to a TE410P ?"
7:31AM 0 What about NCS and Asterisk?
7:03AM 1 anybody get experience with dell powerconnect 3424 and QOS for asterisk traffic?
6:19AM 5 Sipura SP3000 question
3:31AM 6 RE: SPA 3000 - UK Replacement
3:15AM 4 what cable to connect a legacy PBX to a TE410P ?
2:12AM 10 Connecting to a cluster of SIP servers
 
Friday April 21 2006
TimeRepliesSubject
7:26PM 2 1.2.7.1 on FC5 won't make install
3:35PM 1 Error installing oh323
2:04PM 0 Very high size-32 usage
2:02PM 1 server choice
1:29PM 2 confused about iax and voip providers termination
1:25PM 4 extension match sip address
11:47AM 1 Grandstream Budge Tone 101 keeps deregistering
11:13AM 0 Easier install of QueueMetrics on Asterisk@Home
10:51AM 2 wellgate FXO unit
10:37AM 1 MWI in multi-PBX setup
9:51AM 0 SIP domain in Asterisk
9:45AM 1 roundrobin strategy in queues not working as described?
9:38AM 0 HANGUPCAUSE on SIP channels
9:36AM 2 Definitive list of sounds
8:53AM 1 Parallel Dial: Busy detection - stop when any is busy?
8:39AM 0 MoH issue
8:07AM 17 Power over Ethernet (PoE) switch recommendations
7:41AM 5 Separating Asterisk SIP extensions from dialing each other.
7:04AM 1 Flash Panel / Queue Slots
5:07AM 0 Airspan / Arelnet GW and Asterisk
4:54AM 0 record_in / record_out configuration parameters
4:08AM 0 How to select Ceptral's Voice in Asterisk's Swiftapplication??
3:59AM 2 How to select Ceptral's Voice in Asterisk's Swift application??
3:59AM 2 Asterisk on Red Hat AS 4?
3:28AM 1 AAH or Fedora an Asterisk by sources
3:18AM 6 Unicall MFRC2 Problems with BrT.
2:57AM 1 Real-time Database Front-end
2:10AM 45 some EICON Diva 4BRI questions
1:19AM 0 problem with sphinx2
12:33AM 0 USB VoIP phone with G729 support
12:04AM 2 Modem connection
 
Thursday April 20 2006
TimeRepliesSubject
11:10PM 3 Get sysdate + 5 minutes
9:22PM 0 Colour coding the dialplan -- NoOp and ANSI codes?
7:46PM 1 MeetAsterisk in Europe - register today!
3:02PM 1 Background() and Read()
2:38PM 0 Agents and Realtime
2:35PM 2 queues and the '*' key
2:24PM 1 Problem with TE110P configuration
1:43PM 4 channels change names
1:25PM 2 Asterisk FAx-to-Email
1:24PM 4 Asterisk Won't start after SVN Trunk Update
12:11PM 0 Asterisk (RFC 3389)
12:01PM 0 Suggestion Request: Coloc Provider in Chicago, IL area
11:43AM 3 enablling Te110p with PRI
11:39AM 10 Announcement System for a Charity
11:04AM 28 TDM2400P
8:46AM 2 MeetMe: lots of buffer overruns/underruns when connecting over IAX
8:44AM 1 zaptel and zapata configuration
8:13AM 0 does anyone know anything about chan_btp or btpd?
8:09AM 2 Cubix Softphone + Asterisk 1.2.6
7:24AM 0 Re: Asterisk-Users Digest, Vol 21, Issue 113
5:03AM 0 Internet connection
4:39AM 3 CDRs and billing
4:32AM 3 Playback(something,noanswer) on Zap?
4:12AM 0 Fwd: why DUNDi ${IPADDR} has been transfered to 127.0.0.1?
4:11AM 0 Best Fax send through Asterisk plan?
3:45AM 5 still some moh troubles
3:36AM 3 asterisk + mobicents
3:13AM 1 Asterisk & MGCP reinvite
3:00AM 2 SPA-3000 Bug? Dropped calls while registering.
2:39AM 0 why DUNDi ${IPADDR} can not transfer to 127.0.0.1?
2:19AM 2 avm b1with chan capi and siemens hipath
1:52AM 0 asterisk and siemens hipath 3500
12:36AM 0 Happy story
12:25AM 1 How to stop Asterisk picking up my incoming calls?
12:05AM 0 Dial two extensions at the SAME time and connect them when possible
 
Wednesday April 19 2006
TimeRepliesSubject
11:47PM 4 Jingle support - can we test the feature ?
10:36PM 1 asterisk 1.2.7.1 crashing my newly built system
9:21PM 0 sip.conf codecs: ulaw, alaw and g729
9:13PM 3 ANNOUNCE: Asterisk Jobs and Consulting Site
8:21PM 3 dundi trouble
7:17PM 1 lost audio after zaptel
6:46PM 6 Upgrade from 1.2.4 to 1.2.7.1
2:44PM 3 Codec problem from SIP to H323
2:40PM 1 Error installing asterisk
2:16PM 1 Voice mail issuse when pressing 0
1:39PM 7 Fwd: sip.conf and jump from register to the extension
12:58PM 4 Ring a grop of extension, then playback a file, then transfer to external number
11:35AM 5 Asterisk 1.2.7.1 DTMF anomaly
11:15AM 2 Asterisk IVR / Scalability
11:12AM 3 Meetme codec translation and callerID library.
10:33AM 3 clearing "stuck" channels without a restart
10:26AM 1 Where to buy Eicon DIVA cards
9:55AM 0 How to route all incoming calls on an analog trunk to a specific ring group
9:23AM 5 What's the best way to combine multiple VOIP lines into a single number?
9:19AM 0 Asterisk 1.2.6 and 9133i
9:02AM 1 Delayed voice for 10 secs
8:37AM 4 SLIN format
8:34AM 1 Callerid matching in extensions.conf
7:58AM 1 Sip channel variables
7:48AM 0 Music on Hold bug? User disconnect Sip user agent
7:08AM 2 Call Center with No TDM components
6:59AM 5 Kernel panic - suggestions?
6:30AM 0 Using Asterisk modules in external application
6:11AM 0 FW: NuFone Update: DIDs (Correction)
6:10AM 3 PRI caller ID
5:42AM 0 oh323: asterisk crashes on a dial
5:40AM 0 Problem with Voice quality, please help
5:25AM 2 Unable to allocate socket: Too may open files
5:02AM 0 Re: new_callback_call and conf disconnect
4:01AM 4 Asterisk 1.2.7.1 and IAX modem / channel
3:57AM 8 Asterisk and 7960s
3:38AM 13 Music on Hold bug? User disconnect Sip user agent and called party stills MOH
3:20AM 0 error when executing sphinx!!!
2:50AM 0 Calls stuck in queue...
2:38AM 0 polycom unable to start recoding
1:28AM 0 LCDC and lcd.conf, p_, c_
1:08AM 0 Receiving Faxes...
12:53AM 1 Sending SIP NOTIFY / How to get remote SIP port?
12:02AM 1 need stand-alone FXO ports
 
Tuesday April 18 2006
TimeRepliesSubject
7:03PM 14 Asterisk service crashes
5:06PM 0 Voicemail exits
3:57PM 5 Remember the incoming context?
3:24PM 9 Outgoing voice distortion with Unicall
2:00PM 3 Outbound calls are failing
2:00PM 0 Problem Using Asterisk Call Files with Zap PRI
1:51PM 0 Polycom IP 501 buddy list: Got SIP response 500 "Internal Server Error"
1:49PM 1 Asterisk & GNUDialer issue
1:47PM 2 polycom blind transfer button
1:30PM 4 UK Asterisk sound files
1:03PM 0 re: Sixtel Services
12:57PM 0 RE: Asterisk-Users Digest, Vol 20, Issue 31
12:48PM 24 FW: NuFone Update: DIDs
11:58AM 0 re: Sixtel Services
11:57AM 0 Realtime goto problem
11:56AM 0 re: Sixtel Services
11:29AM 4 PRI blocking on incoming calls
11:18AM 0 Asterisk Performance 350 Concurrent
11:18AM 0 Aastra 9133i Phones Asterisk 1.2.6 and MWI
9:48AM 11 T1 to cross connect remote PBX and asterisk
9:47AM 7 ISDN in Japan?
9:44AM 1 Double Ring - TelIAX/Cisco 79[46]0
9:40AM 0 Asterisk crash with Digium
9:35AM 0 Voicemail Issue - Failed to lock path
9:27AM 0 Asterisk Performance 350 Concurrent ChannelsWorking Nicely
9:26AM 3 eyeBeam + ASterisk 1.2.7.1 + Instant Message
9:23AM 0 Gizmo Call In
9:11AM 0 BSR 1000 and Asterisk
8:50AM 0 Help Getting Local Exchange Dialtone on PRI
8:34AM 0 Asterisk Performance 350 Concurrent Channels Working Nicely
8:24AM 4 correct version of asterisk for oh323
7:54AM 0 IVR and voicemail issues ?
7:50AM 2 bad voice quality
7:28AM 3 IVR: playing multiple streams simultaneously?
6:52AM 0 Re: Cisco 7940/7960 SIP 8.2 Freely
6:49AM 0 Noise on IAX or SIP trunk between 2 Asterisk
6:31AM 1 voicemail kicking in after user has already disconnected
6:04AM 1 Change email/pager VM alerts body text dynamically?
5:19AM 0 Asterisk/FreePBX/Alcatel2400
5:14AM 1 Granstream GXP2000 Distinctive tones
3:39AM 2 Cisco 7970 SIP - few questions
2:34AM 7 Grandstream Budgetone and Mac mini?
2:09AM 5 HardPhone PlanetVIP-150T - Starts music on Hold and i can't get the call again
 
Monday April 17 2006
TimeRepliesSubject
3:15PM 4 Looking for a good VoIP Provider in the UK-
3:09PM 0 Asterisk settings for roaming users
1:53PM 7 Asterisk hyperthreading compiling.
12:00PM 9 Orative
11:35AM 1 astcc and inwards billing
11:28AM 0 H.323 question, take so long time to call
10:39AM 0 Sip Notify cisco-check-cfg - Does it still work with 8.2?
10:17AM 0 Asterisk Like Phone Switch ?
10:12AM 4 Billing Server Open Source
9:47AM 0 Setting CDR dnid and Billing
9:39AM 0 IAX phone hardware recommendation
8:44AM 4 voicemail use external smtp server for sendingmail
7:52AM 13 Don't see my post
7:17AM 7 Cannot dial out with Polycom 501 after upgrade
7:00AM 2 Agents, Queues, and Voicemail
2:57AM 4 Probs with asterisk
2:09AM 15 multiple asterisk process ?
12:45AM 2 cdr_pgsql failing to load in head
 
Sunday April 16 2006
TimeRepliesSubject
11:29PM 1 Snom 190, Asterisk and Intercom
11:25PM 1 Cisco 7940/7960 SIP 8.2 Freely Downloadable
4:18PM 0 Flash Key and R Italian Key
3:35PM 0 External voicemail and MWI on internal phone
9:03AM 5 Problems with several SIP Providers (one way echo)
7:54AM 1 Can Astcc allow dialing phone number more than once
6:46AM 2 How do I limit the lenght of a call
6:16AM 4 [Fwd: Re: voicemail email-from]
6:04AM 0 What is Multi-layered-Access control
1:09AM 1 Faxing and PCI (was Re: Digium cards, sodisappointing !)
 
Saturday April 15 2006
TimeRepliesSubject
9:24PM 5 voicemail email-from
6:12PM 12 Phones that work well through NAT
12:17PM 6 FreePBX in Production systems?
11:58AM 1 CDR query
11:47AM 3 asterisk voicemail question
1:52AM 6 Cisco 7960 International
 
Friday April 14 2006
TimeRepliesSubject
3:19PM 0 Ztmonitor shows RX is always on. FIXED.
3:09PM 0 7941/61 IP Phone SIP phone load - for CCM v5.0
2:14PM 7 My consulting story
1:59PM 0 Bluetooth (chan_btp): dialing external phone number through BTP/Zap when bluetooth device not present?
12:56PM 1 asterisk or ser
11:19AM 50 attended transfer issue
11:14AM 2 How to get 1.2.7 asterisk
10:46AM 1 Re: Asterisk-Users Digest, Vol 21, Issue 81
9:59AM 5 asterisk 1.2.7.1 and app_rxfax
8:54AM 3 Polycom 501 resource full problems ...
8:23AM 4 Unicall and Fax
7:49AM 1 Packet Testing
7:41AM 1 A weekend of upgrade is coming for me - any hints?
7:30AM 0 Work available - India
6:41AM 2 change/toggle flash operator panel components
6:39AM 3 asterisk with bluetooth headset howto
5:54AM 5 tdm2400p and asterisk 1.2.1
3:29AM 6 Asterisk hardware for new office suggestion
3:17AM 0 How to cross compile asterisk for Axis ETRAX 100LX foxboard embedded device on Debian
2:43AM 0 Cisco 7970 SIP
12:47AM 3 HELP! Bad sound quality
 
Thursday April 13 2006
TimeRepliesSubject
9:09PM 1 Asterisk no sound from sound card
7:37PM 0 troubles with a Gateway audiocode (Mp104 fxs)
7:33PM 0 Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. ???? Xlite
5:26PM 5 Will VoIP ITSP's be Next?
5:01PM 85 Digium cards, so disappointing !
4:07PM 1 What is Multi-layerAccess control
3:36PM 2 Anyone played with app_amd?
2:33PM 0 connecting Digium E1 pri card to panasonic TD-500
2:21PM 0 SIP/ShoreTel REFER support
1:54PM 0 spa-942 support Page() / Intercom correctly?
1:10PM 1 call center running Asterisk-soundquality-critical!
1:02PM 0 DTMF sensitivity
12:21PM 0 CANADA 911 Update
11:48AM 1 call center running Asterisk -soundquality-critical!
11:29AM 2 Static on ZAP channels
11:16AM 1 Early Media Enable?
11:11AM 4 Asterisk 1.2.7.1 Released
10:58AM 1 Ztmonitor shows RX is always on.
10:42AM 0 Help Cas Circuit
10:33AM 1 Set language in Asterisk auto-dial out
10:07AM 3 Asterisk 1.2.7 Page()
10:06AM 0 SIP register question
10:04AM 1 Sipura 2100
10:04AM 1 placing call with agi
9:53AM 2 DTMF Not working for only one number
9:33AM 5 Segfault on Inbound call?
9:25AM 0 Display "Confideltial" or "unknown" on calledid display
9:22AM 1 Display "Confideltial" or "unknown" on called iddisplay
9:01AM 5 Display "Confideltial" or "unknown" on called id display
8:55AM 3 NAT/STUN Server
7:36AM 0 fxotune error
7:31AM 3 app_meetme.so
7:30AM 0 (no subject)
6:59AM 1 sip nat bug
6:36AM 0 Codec GSM Makefile Patch for IA64
6:28AM 1 Question on parkinglot
6:20AM 1 AgentCalled event
6:02AM 7 OT: MWI on Treo 600/650
5:10AM 2 How to terminate ringing call before it is answered?
4:54AM 2 IP logging
4:51AM 1 ast_sched_runq ran 281 scheduled tasks all at once
4:47AM 0 Hangupcause to handle Called party disconnect ? PSTN----E1----OldPBX---E1--Asterisk
3:01AM 0 Any way to prevent this from happening
2:28AM 1 voicemail use external smtp server for sending mail
1:48AM 6 Background music in call
 
Wednesday April 12 2006
TimeRepliesSubject
11:57PM 0 Re: Double sip logins
10:30PM 2 Announcement: New Texas User Group formed
10:09PM 1 Problem with Voice Quality
8:46PM 2 freepbx dialing prefix
8:35PM 2 How to terminate ringing call before it is answered
6:59PM 0 RE: Asterisk-Users Digest, Vol 21, Issue 70
6:57PM 0 Multiple phones in same call
6:05PM 1 ASterisk Back2back
4:19PM 0 Asterisk 1.2.7 Released!
12:51PM 0 Playback sound file while on-line
12:50PM 8 Company List
12:32PM 1 Call Forward and AGI
12:30PM 1 SIP MWI
11:43AM 43 DUNDi with SIP
11:09AM 4 Cisco 7960 won't dial (sccp)
11:07AM 3 Callback Agents and Dial 'g' option
10:43AM 0 call center running Asterisk-sound quality-critical!
10:33AM 0 Config with TE210P, Asterisk and Legacy PBX and FreePBX?
10:24AM 2 Recording queue transfers
10:14AM 4 call center running Asterisk -sound quality-critical!
10:13AM 7 playback soundfile stored in mysql database
9:56AM 2 Polycom VLANs
9:38AM 2 * 1.2.4 & 1.2.6: "Ringing" anamoly
8:44AM 3 Setting Codecs on the Fly
8:42AM 4 CAPI Installation Eicon Diva Server
7:53AM 4 call center running Asterisk - sound quality-critical!
7:28AM 1 Macro-hangupcall - has a Wait(5) - Ast@Home --- why?
7:23AM 0 Trunking Protocols
6:56AM 4 URL in Queue App / Determining the DID/Queue at Agent's Phone
6:26AM 1 DID'S Romania - Bucharest
5:49AM 11 SIP conections, with RTP not going trough Asterisk
5:10AM 0 Oh323 inband DTMF
4:22AM 0 g.726 codec not working in one direction
4:09AM 1 free video (soft) phone available?
3:48AM 2 help -- voicemail
3:40AM 1 Failed to recieve Fax: Asterisk - IAXModem - Hylafax
3:26AM 1 SIP call hangup from asterisk CLI
3:15AM 2 billing with PostgreSQL
2:16AM 1 iax2 show netstats
1:21AM 2 Where is the difference sip.conf - Real-time ?
 
Tuesday April 11 2006
TimeRepliesSubject
11:10PM 0 SPA-3000 call pickup behind a PABX
10:55PM 0 Polycom SIP 1.6.5 reloading
9:24PM 5 TE410P upgrade to TE411P => (solution to) no more fax carrier detection !
8:56PM 0 How to config firewall for RTP/RTCP?
8:33PM 0 AstriCon Update: Europe Early Bird Ends Saturday
7:30PM 17 Performance: Xeon or Opteron?
6:08PM 2 call center running Asterisk - sound quality- critical!
5:41PM 5 Cisco 7960 6.3 unlock/reset?
5:19PM 3 Re: update - 512 Simultaneous Callswith DigitalRecording
4:51PM 4 E1 Disconnection Asterisk behind an old PBX
2:31PM 2 res_config_mysql.so: undefined symbol: __stack_chk_fail
2:22PM 0 chan_btp: dialing external phone number when bluetooth not present?
2:07PM 0 core dump...
1:38PM 0 TNT Max Config
1:29PM 0 RE: Fatpipe Support - Authorization to open Box - fwrps2001101288
1:14PM 0 HELP NEEDED: "odbc show" crashes Asterisk... and I have no idea of what is going on!!!
1:00PM 0 STUN Server info
12:21PM 2 nic aliases not working
11:51AM 0 Cisco 7970 SIP Config
11:02AM 0 log messages...
10:40AM 1 ExternalIVR
10:17AM 2 Snap for Asterisk
10:12AM 0 XO Callerid NAME
8:28AM 0 PRI outbound call error detection
8:26AM 1 Virtual terminal running CLI
8:15AM 0 Logoff time of an agent.
8:06AM 1 Agent with multiple phones in multiple queues
7:54AM 1 AW: Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use??
7:33AM 3 Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use??
7:06AM 1 Night Mode and indicators
7:05AM 7 Why is the internet connection important to LAN and PSTN calls?
6:52AM 6 Re: Received VNAK: resending outstanding frames?
6:14AM 3 Question on clicking
5:15AM 2 Automatic 3 Way Call
4:56AM 8 OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)
4:42AM 2 G726-40 required - Help!
4:18AM 7 Trial Version of Asterisk Interface Available
4:14AM 3 the best billing tool for Asterisk
2:39AM 1 Major issue: More incompatible frame messages
1:44AM 0 Differences 1.0 vs. 1.2
1:26AM 1 Native music on hold on 1.0
12:51AM 0 Echo in some queues but not others
12:35AM 1 Database server
 
Monday April 10 2006
TimeRepliesSubject
10:48PM 7 Bandwidth Management
10:05PM 10 SPA-941/942 Bulk provisioning
8:49PM 7 asterisk credit card processing
8:33PM 3 Vertical
8:30PM 16 Asterisk BRI in the USA
7:52PM 2 Choppy Sound when using linux router or asterisk
7:46PM 3 One digit too short dialed, stay for ever there in the dialplan!
3:56PM 0 ANI and DNIS Seperation on a PRI(TelephonyNumbering Plan (E.164/E.163) (1)'*4105556654*8005550215*' ])
3:44PM 6 [ISSUE] Honouring Silent Caller ID Numbers
2:54PM 23 te110p and interrupts
2:12PM 2 HTML / PHP
2:00PM 0 Is this possible? (queue setup)
1:50PM 0 Problem with Asterisk and Grandstream HT286
1:30PM 2 Wanted any /all used out of service Digium boards Mark
1:04PM 1 Group funcations not functioning
12:32PM 4 Polycom TOS
12:28PM 2 Asterisk and Cisco Callmanager
11:19AM 2 Problem - Voicemail resets phone
11:12AM 0 RE: still no solution for me, if one
10:56AM 4 E1 PRI problem with TE205P
10:51AM 12 Texas User Group
10:03AM 1 ANI and DNIS Seperation on a PRI (TelephonyNumbering Plan (E.164/E.163) (1) '*4105556654*8005550215*' ])
9:54AM 1 RE: still no solution for me, if one provider
9:40AM 0 Audio problems
9:31AM 7 still no solution for me, if one provider fails.
9:20AM 1 ANI and DNIS Seperation on a PRI (Telephony Numbering Plan (E.164/E.163) (1) '*4105556654*8005550215*' ])
9:03AM 10 App Page() in 1.2.5
8:37AM 2 faxing setup
8:29AM 0 Asterisk/InterTel Axxess via MGCP? Anyone?
8:25AM 1 "chan_iax2.c: Ooh, voice format changed to ..."
8:07AM 1 SIP channel unavailable/busy/really not there
8:05AM 0 NORTH CAROLINA: Any interest in starting NC User Group?
7:54AM 2 RTP Timestamp errors
7:27AM 2 Outbound calls through Broadvoice
7:14AM 1 How to set AbsoluteTimeout for DirectoryApp() ? Is this the safest way?
7:04AM 6 call center running Asterisk - sound quality - critical!
6:45AM 1 [asterisk-dev] RTP mixer in Asterisk
6:44AM 1 Directory App() is running for a while, like blocked/freeze? in the same name...
6:29AM 0 RTP mixer in Asterisk
5:56AM 14 callerid name inboune from PRI
5:14AM 9 Asterisk stops responding when internet is down
4:13AM 2 AMP / Maintenance-Button missing
4:12AM 1 Asterisk to CCM4 SIP Trunk one-way audio problem.
3:48AM 5 GXP-2000 phones stop registering
3:43AM 0 WG: G729a error
2:59AM 4 Call me for testing my system
2:10AM 0 Asterisk evaluating CLIP, then getting out of the way
1:50AM 1 Re: update - 512 Simultaneous Calls with DigitalRecording
1:21AM 1 G729a error
12:56AM 1 Asterisk is not reconnecting
12:40AM 0 How to avoid "Avoiding initial deadlock...."
 
Sunday April 9 2006
TimeRepliesSubject
11:55PM 4 queue_log timestamp?
11:21PM 0 Realtime oracle compiling problem
11:13PM 0 (no subject)
11:10PM 3 Voipstunt, voipbuster, .... not working properly?
9:09PM 2 Asterisk Dial Command Timeout not Accurate (not even close)
8:06PM 1 PRI Group Calling
7:06PM 6 Instant Message?
6:20PM 0 for review
5:26PM 0 Provisioning Server...
12:09PM 0 How to avoid "Avoiding deadlock..."
9:28AM 0 txfax tiff file format
7:21AM 4 GXP-2000 and Voicemail
2:06AM 2 how to communicate two PCs on LAN with Asterisk
 
Saturday April 8 2006
TimeRepliesSubject
11:52PM 2 oh323.conf problem
10:20PM 2 MACRO_RESULT=ABORT
9:02PM 9 How to set busy
7:24PM 11 Force codec
3:08PM 1 ANI on a PRI
2:45PM 0 Re: [asterisk-dev] bug or bad chan_sip.c
2:38PM 0 Re: [asterisk-dev] bug or bad chan_sip.c
1:51PM 1 unable to enable stutter dialtone
11:18AM 5 quadBRI PCI ISDN on Suse Linux 10
10:21AM 0 Call parking query
5:01AM 2 question about DISA
4:39AM 0 FW: CallerID
2:57AM 5 HELP !!!!!
1:22AM 0 Quintum ASM400 FXO configuration
1:03AM 5 AAstra 9133i register double account.. ??
 
Friday April 7 2006
TimeRepliesSubject
11:13PM 0 May be OT , but comparing
9:46PM 3 Problems with registering iaxy
8:26PM 2 Announcing Astmanproxy 1.20
8:04PM 7 [OT] Centrex Question
4:53PM 0 Canada Nomadic 911 - From the Yes it will Screw Your Biz Dept
1:07PM 0 simple wav ringtones?
12:57PM 3 can we lend a hand?
12:35PM 0 Call tracking through chan_agent using the Manager API
11:50AM 0 Audiconferencing System fon Asterisk
11:21AM 3 DIALSTATUS for Multiple Dialled Numbers
10:03AM 2 Telephony newbie need advice for integration Nortel MICS 4.1 with Asterisk via T1/E1 interface
9:45AM 1 Bell Canada Requests $987.14 Rate increase 9 11 /VOIP Providers
9:37AM 2 Attended Transfer howto
8:55AM 2 wellgate registration 3802
8:28AM 9 Beeps and noises during calls
8:17AM 1 Cisco 7912 Phones & XML
8:06AM 1 Fedora 'service asterisk start' problems
7:51AM 0 editing the asterisk -addons makefile
7:44AM 1 transfer call after advise
7:42AM 2 Uplink Skype2Sip
7:33AM 0 Re: IVR: Cant hear my message
7:22AM 1 regexp in gotoif
7:02AM 1 OT: local calling guide
6:54AM 0 Inbound PRI calls drop after 5 seconds using
6:32AM 1 Inbound PRI calls drop after 5 seconds using Sangoma A101
6:14AM 1 suggestions on an IP T1 to TDM T1 gateway solution
3:25AM 2 407 proxy authentication
2:35AM 0 match callerid against outgoing calls
2:29AM 0 MINNESOTA: TwinCities Asterisk Users Group - Saturday 04/08/2006
1:35AM 6 gotoif
12:27AM 0 Dial Plan Problem with extensions ringing multiple phones connected on different * servers
12:18AM 0 CRTC Stops Bell Canada 911 rate increase for 45 days only
12:17AM 0 Line in use
12:12AM 0 0-ECRS
 
Thursday April 6 2006
TimeRepliesSubject
11:48PM 1 Bell Canada Requests $987.14 Rate increase 911 / VOIP Providers
11:41PM 3 Look What 911 Will Cost in Canada
8:59PM 0 Pickup Group in VPB
8:42PM 1 Integrics ITSP 1.6 released
7:56PM 4 asterisk box as a voip gateway
7:32PM 0 Problem with playing old gsm files
6:57PM 3 # IP601's with POE per Catalyst 3560G-48PS
6:50PM 0 SIP to another PBX w/ forwarding set
6:50PM 0 Telasip
3:52PM 0 Re: Asterisk-Users Digest, Vol 21, Issue 38
3:49PM 1 Suggested MeetMe feature: 'i' without review.
3:38PM 0 What Media Gateway (connected via SS7) do you use
3:36PM 3 OT: HOWTO: Create a 90mbit bonded link 600 m etre s away with Cat 3 or telco wire [long]
3:35PM 0 OT: HOWTO: Create a 90mbit bonded link 600 m etres away with Cat 3 or telco wire [long]
3:22PM 1 OT: HOWTO: Create a 90mbit bonded link 600 metres away with Cat 3 or telco wire [long]
3:17PM 1 digium card for xseries 346
2:57PM 4 queue/agent and macros?
2:47PM 6 Steps to make trunked iax2
2:44PM 5 OT: HOWTO: Create a 90mbit bonded link 600 metre s away with Cat 3 or telco wire [long]
2:42PM 1 Originate
2:42PM 0 Directory Issue
1:32PM 3 Asterisk in FreeBSD
12:27PM 0 Re: MWAnalyze question
11:59AM 6 Call/Contact Center.
11:39AM 0 ringing indication in handset when 2 extensions answer simultaneously?
10:57AM 0 TDM400P and Junghans quadBRI
8:47AM 1 Networld Interop, Vegas 2006
8:29AM 1 increasing volume level to console/dsp
8:19AM 0 New user needs a hand starting
8:06AM 1 Planet VIP-320 DECT gateway with Asterisk?
7:59AM 3 Cisco 7960 - hints
7:42AM 5 chan_sccp and hinting
7:39AM 1 pause / unpausequeuemember
7:29AM 0 audiocodes with asterisk:- newbie
7:00AM 0 Open channels
6:40AM 2 Call Parking and multiple contexts
6:35AM 0 (no subject)
6:18AM 2 Using Call Progress
5:36AM 2 Voicemaster
5:32AM 2 TDM2400P problems
5:03AM 0 FXS module failed
4:49AM 0 Asterisk dialing over asterisk to PSTN
4:19AM 1 FXO/FXS and E1 in same system
4:18AM 1 qozap errors on junghanns QuadBRI
3:51AM 0 not get ring tone with chan-capi and avm b1
3:15AM 0 Call transfer to cell phone [UPDATE]
3:02AM 0 AW: Dial out on Zap
2:49AM 0 Dial out on Zap
2:28AM 0 Call transfer to cell phone
1:52AM 1 Incoming call redirected to mobile
1:51AM 0 Fwd: Hangup Supervision
1:49AM 1 IVR : Can't hear my message
1:34AM 2 Hinting a conference room
12:56AM 0 chan_modem_i4l delay again..
 
Wednesday April 5 2006
TimeRepliesSubject
11:57PM 0 Chan-sccp - Asterisk dies
10:59PM 1 Fwd: [dmuars] Eh up - March 144 results altered
8:11PM 0 E-911 Canada Info - Hot Off the Press
6:55PM 0 WOW! Sphinx is awesome... but....(asterisk+sphinx+menus)
6:35PM 3 Setting ptime attribute in SDP invite
4:30PM 7 Fedora Core 4 - problem with kernel 2.6.16-1.2069_FC4
4:25PM 1 WebMeetme Problem Please help!!!
2:52PM 5 What causes deadlock?
2:48PM 0 What does this error mean "app.c: Huh....? no dial for indications?"
2:47PM 21 How to restrict simultaneous phone registrations
1:37PM 7 Sending Access codes to a 5EE switch.
1:20PM 2 IAX2 Origination Problem
12:24PM 10 transforming g729 files to wav files
11:31AM 0 The Asterisk bug tracker :: please think twice before opening a report!
11:21AM 0 Asterisk RealTime queue - periodic-announce
11:16AM 1 zaphfc NT Mode. Extension not recognized...
11:12AM 2 Asterisk on BSD?
10:41AM 0 SIP client looses register and then i need to restart my pc to get registered on Asterisk 1.2.5
10:40AM 0 one-waysilence during calls
10:26AM 4 legacy Alcatel 4200/4400 and Asterisk (QSIG/PRI) and callerid
10:23AM 0 Asterisk support to Tornado M5 IP Phones
10:14AM 0 Favorite softphone with command line interface
10:07AM 0 Re: Asterisk start/stop
9:46AM 0 SHOWCHANINFO Not Working
9:32AM 0 TE110P errors
9:15AM 4 fax server functionality on Asterisk
9:07AM 3 SIP Asterisk Polycom Reinvite
9:02AM 3 can't start chan_capi with asterisk group
8:56AM 0 Patch 5779 on 1.0.9?
8:27AM 2 long delay between "Ring Begin" and "SIP/XXX is ringing"
8:17AM 4 chan_modem_i4l delay
6:37AM 5 queue issue
5:38AM 0 oh323 - cant load module
5:23AM 9 WOW! Sphinx is awesome... but.... (asterisk+sphinx+menus)
5:16AM 0 Querying number of people in a call queue from dialplan
3:29AM 1 Got SIP response 302 "Moved temporarely"
3:18AM 7 Dial Plan Logic Problem
2:31AM 0 Error Header field Via
2:18AM 4 Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
1:56AM 0 Hangup Supervision Issue on Digium TDM11B
12:23AM 3 VPB cannot call out
12:09AM 0 extensions.conf - switch => statement?
 
Tuesday April 4 2006
TimeRepliesSubject
11:52PM 4 Asterisk svn starting problem
11:45PM 0 Asterisk-addons compiling problem
11:38PM 0 some problems with asterisk and E1
10:04PM 2 Milliwatt Test Number List
9:54PM 5 asterisk-ooh323, asterisk 1.2.6 and netmeeting
9:54PM 2 VoiceMail realtime not working in asterisk-1.2.6
7:17PM 0 sip hang channels
6:27PM 1 voipstunt: "Forbidden - wrong password ..."
6:23PM 2 speech rec what works
5:55PM 1 Too many open files
5:44PM 1 Need 25-50 Linksys boxes
5:15PM 1 Set(CDR(anything_but_userfield_or_accountcode)=bla) broken?
4:56PM 2 Distinctive Ring on SPA941
4:37PM 1 not transmit audio on sipura 941
3:07PM 0 AST eating CPU 100%->Resource temporarily unavailable
3:00PM 0 Applying patch.
2:58PM 0 Anyone have a definitive list of Managereventsper category?
2:42PM 0 Opensource solutions to SPIT
1:40PM 2 WebMeetme defines.php?
1:14PM 0 Jitter in SIP connection
12:36PM 8 queueue recording and what to do next
12:34PM 1 Anyone have a definitive list of Manager eventsper category?
12:17PM 1 Ideal Setup for T1/PRI and TE110P - second try
11:59AM 0 Anyone have a definitive list of Manager events per category?
11:35AM 2 Can't get Pickup app working
11:12AM 2 voicemail context issue
10:46AM 1 Ideal setup for PRI/T1 and TE110P
9:25AM 2 Can't recieve Fax: No carrier detected - Ast erisk + iaxmodem + Hylafaxv --- sorry.wrong log.
9:18AM 5 Any Aheeva Users?
8:57AM 0 X100P small test gives cracking sound at the voip side
8:19AM 1 Can't recieve Fax: No carrier detected - Asterisk + iaxmodem + Hylafaxv --- sorry.wrong log.
8:12AM 0 Can't recieve Fax: No carrier detected - Asterisk + iaxmodem + Hylafax
7:44AM 4 Phones are all auto answering
7:39AM 3 Possible PRI fault?
7:21AM 1 Realtime Database Lookup
6:45AM 0 Jitter in SIP calls?
6:39AM 7 IAX connection refused between 2 asterisks 1.2.5
5:47AM 6 Hangupcause is not enough on PRI
3:39AM 0 New User's Query , which card TE411P or TE410P
2:31AM 4 Auto Attendant Question
2:11AM 4 Fax over 2 bridged TE110P channels
1:38AM 3 E1 te110p problem
1:35AM 0 Dial(L(x...)) distinct to SET TIMEOUT(absolute)
12:52AM 17 Loading module chan_zap.so failed! PLZ help me!
12:32AM 0 QSIG and multi-PBX receptionnist
 
Monday April 3 2006
TimeRepliesSubject
11:16PM 0 RE: Re: Re: Compatible Asterisk Connectivity Cards :Sangoma
9:10PM 3 MeetMe/Asterisk Timer
7:36PM 3 GoDaddy royally screws over aussievoip.com.au and soft-swtich.org
7:30PM 3 Monitor or mixmonitor
6:52PM 0 random beeps during calls
5:22PM 4 Need More Simultaneous Voice Channel Capacity on Asterisk
5:21PM 4 New Skype<>SIP gateway
5:21PM 2 Blocked channels, according to our telco... leading to CONGESTION status
3:46PM 3 Need to Install Fax to Email feature
2:06PM 0 warnings during parsing of misdn.conf
1:41PM 1 Meetme admin
1:26PM 1 web meetme
1:14PM 1 Asterisk compiling problems...
12:34PM 0 Inter-Asterisk SIP and CalleriID
12:04PM 0 411 Directory: First, Last or Both?
11:50AM 1 Hardware question about Redfone's foneBridge
11:31AM 0 Lockups after Asterisk upgrade
11:21AM 2 Interrupting a call
11:18AM 0 Critical Transaction failed: Client non-INVITE - SJPHONE connected to Asterisk
10:43AM 5 Stupid newbie question
9:58AM 0 Maximum duration of Voicemail messages
9:26AM 0 Is it a stun problem: 63 to 1800 msec
8:41AM 3 Unable to connect to remote asterisk (does / var/run/asterisk.ctl exist?)
8:35AM 1 Anybody success using Asterisk 1.2.6 and Spa nDSP 0.0.3 pre 6?
8:28AM 0 Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
8:25AM 17 Pickup() h323
7:44AM 0 Anybody success using Asterisk 1.2.6 and SpanDSP 0.0.3 pre 6?
7:40AM 6 Beginner: PBX for my house
7:31AM 14 Hinting
7:30AM 5 SIP Responsecodes
7:25AM 4 Random music not so 'random'
6:54AM 6 Coice recognition IVR?
6:43AM 2 Callback auto dialing
5:42AM 14 Frustrated with echo...
5:11AM 0 Concurrent calls to voipstunt and other providers
4:36AM 0 AMILogin and case sensitive
4:04AM 5 call transfer to external phone number
3:50AM 0 Bad Pick up line
3:48AM 0 Annonuce Me Feature
2:38AM 1 update asterisk in a production system
2:35AM 4 bristuff for * 1.2.6/zaptel 1.2.5
2:10AM 6 R2 protocol error
1:56AM 5 Diva Server BRI echo options
1:42AM 0 No CDR in Macro after Dial
1:13AM 17 update - 512 Simultaneous Calls with Digital Recording
12:02AM 0 Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.
 
Sunday April 2 2006
TimeRepliesSubject
11:48PM 0 Comparison of Business Edition VS Open Source
10:31PM 8 ASTCC: How to reset "in-use" flag automatically ?
8:45PM 1 ZapBarge but ability to talk to the agent
8:41PM 27 Compatible Asterisk Connectivity Cards : Sangoma
7:48PM 5 Who is on a call?
5:16PM 1 Information about LOCAL/ Channel
2:32PM 0 Voicemail() - Reading exit or return results
1:19PM 2 Connecting Asterisk to "traditional" phone central
12:54PM 0 Subversion mirrors of Asterisk, Zaptel and libpri rebuilt
12:46PM 1 Asterisk "answering machine" replacement, "WaitForRing()", application return values
12:28PM 0 can automon work with MixMonitor
10:38AM 0 no audio between sip channels * 1.2.6
9:30AM 6 Asterisk 2.0 Where to download
8:04AM 2 DID registration status
4:58AM 6 polycom overlap dialing?
2:40AM 6 Cisco 7960 nat problems.
12:05AM 1 morcdr v0.1 released
 
Saturday April 1 2006
TimeRepliesSubject
9:48PM 0 G729 Passthrough question
6:46PM 2 Problem: ringtones stop unexpectedly
6:09PM 9 H323 on way voice
4:54PM 1 vmail access problem
4:41PM 1 channel.c:787 channel_find_locked: Avoided initial deadlock for '0x8446b50', 10 retries!
4:17PM 4 Install problem with res_snmp.so from current trunk (bug?)
3:32PM 4 chan-capi: Sending digits on a bri (isdn) d-channel
12:09PM 1 Incorrect CDR results
11:54AM 2 TO have ringing tone instead MOH
11:22AM 1 Problem: ringtones stop unexpectedly when multiple channels are dialed
11:12AM 2 AGI hangup problem
10:07AM 0 Free Software/Open Source Telephony-Summit 2006
5:58AM 2 Asterisk box with unreliable ping/latency
5:08AM 14 Newbie question - sip.conf incoming contexts
4:52AM 0 INX (Internationalnumber.com)
1:40AM 1 How to use Sendtxt?
12:04AM 1 voicemail to email sending problems
12:01AM 3 ooh323 and g729 - any issue?