asterisk users - Apr 2005

Saturday April 30 2005
11:08PM 1 Broadvoice limits???
7:01PM 2 Asterisk@Home bug
4:59PM 8 Programing a call forward feature to cel phones
4:47PM 1 How to bridge 2 calls
4:10PM 1 Kernel 2.4 or 2.6
3:57PM 3 SIP over IAX2
2:58PM 8 Problem with Sangoma/Adtran 600 installation
2:53PM 1 Problem with PSTN
2:27PM 1 Asterisk on Radio Tonight
2:14PM 2 Call-park timeouts..
1:52PM 1 Hotel CDR Software
1:18PM 1 Send DTMF *AFTER* channels are bridged
12:08PM 22 A good SIP receptionist phone
12:04PM 0 X-lite and * behind Firewalls
11:56AM 1 Zaptel and Boostringer
11:32AM 0 ANNOUNCEMENT: Asterisk-java 0.1 released
10:34AM 1 Amp extensions script
10:24AM 1 help with compiling addons for cdr
10:10AM 2 transfer pstn call to voip line, thus freeing up pstn line
10:09AM 1 Intel 536EP
10:07AM 0 Chan_modem_*
8:41AM 0 7910 and Skinny
8:39AM 1 IPSwitchBoard version 0.111 released
7:49AM 1 Polycom IP500 Forward problem codec issue
6:25AM 3 Dynamic phone groups.
5:54AM 0 Avaya 4610SW IP phone?
12:19AM 0 sipp example
Friday April 29 2005
11:22PM 1 CID Number problem
10:01PM 0 Call routing
9:51PM 0 CallerID on cell phone
9:29PM 1 Can't get incoming calls with IAX trunks (FWD &Teliax)
9:03PM 2 Can't get incoming calls with IAX trunks (FWD & Teliax)
8:16PM 2 Asterisk and sendmail
5:57PM 0 Polycom IP500 Ringer Volume
4:26PM 3 Need info : lspci
2:57PM 1 Any workaround for long DISA timeout before it actually dials ?
2:45PM 3 Caller-ID Block
2:21PM 4 Paging and intercom
2:14PM 1 UTSTARCOM Wifi handset?
1:29PM 1 txfax and Ghostscript 8.51
1:27PM 0 More TDM questions....
1:10PM 3 Bouncing DTMF?
1:03PM 0 Asterisk@Home 1.0 released
1:01PM 0 Curious behaviour for pound (#) key with SIP X-lite SoftPhones
12:19PM 0 Detecting DeadLocks
12:10PM 0 ISPCON: SIP CPE experts wanted for panel
11:32AM 1 GR-303 zaptel and zapata configurations
11:19AM 0 SIP/IAX softphone with g729/723
11:17AM 0 Adtran 600
10:07AM 1 User events - a dumb question
10:03AM 4 IAX2 one way audio
9:50AM 1 Sip endpoints that support re-invite??
9:12AM 1 the beginning of voice menu is cutted
8:33AM 3 Channel bank of E1s? (one E1 input --> 2 x E1 output)
8:30AM 1 Asterisk Manager interface, setting global vars
8:22AM 1 chan_zap graceful failure
8:16AM 1 Queue Monitor Filename Problem
8:05AM 3 quadbri bristuff ztcfg fail
7:42AM 2 Recording in a call center
7:11AM 1 Asterisk on VMWare ESX/blade servers
6:23AM 0 EuroISDN bearer capability pass thru from (fax) a/b adapter on OctoBRI to TE410P
6:00AM 1 T1 Technology and VoIP Gateway Primer
5:43AM 3 Realtime feature
5:03AM 0 Cost field in Call Detail Records (cdr)
4:42AM 0 DNID empty on incoming calls
4:05AM 2 asterisk-oh323
3:59AM 0 IPSwitchBoard Version 0.110 Released
3:08AM 3 bri error
2:58AM 0 Barge In With Queues
2:19AM 0 how to configure ser and asterisk together to share the load
2:18AM 0 how to share asterisk load with ser server
2:17AM 0 how to share asterisk load with ser
2:16AM 0 (no subject)
12:57AM 0 Some * scripts: Pull asterisk config from LDAP and authenticate() against voicemail passwords
12:38AM 1 first few seconds of call is lost
12:17AM 7 Pattern Matching
Thursday April 28 2005
11:32PM 0 Voicemail Broadcasts
9:43PM 2 vmail.cgi: -rwsr-sr-x as root *still* won't read the files
9:11PM 0 How to Restrict Number of Lines
9:08PM 0 [Fwd: Re: [Fwd: Voicemails stopping]]
9:07PM 0 Re: Asterisk-Users Digest, Vol 9, Issue 255
9:04PM 1 How to prevent number of agents
7:05PM 1 Help to configure asterisk to dial to an PSTNline
7:04PM 0 How do I add an IP to an Exten
6:53PM 3 missing first digit when dial extension / dtmf problem ???
6:52PM 3 voip connection problems
5:16PM 1 Assigning DID and Extension with similar value
5:07PM 1 Traffic Testing
4:58PM 11 Problems with TDM400P card
4:56PM 2 Re: Re: T1/DS1/ISDN PRI
4:44PM 1 chan_capi crashes asterisk
3:57PM 0 E1 legacy multi PBX integration?
3:53PM 1 Gabled voice problem on Asterisk for two remote users
3:21PM 2 Sipura SPA-841 and firewall
3:08PM 0 Asterisk not paying attention to NAT Setting
2:55PM 5 Asterisk Hardware Recommendation
2:38PM 1 music on hold on R key not working.
2:24PM 1 Asterisk SIP sound issue
2:14PM 0 Prompts and MoH not working - AAH .09
2:00PM 0 Spandsp compile error
1:48PM 1 Asterisk Home .9 with TDM11B
1:42PM 0 Hints: What are they? How do they work?
1:38PM 1 Help to configure asterisk to dial to an PSTN line
1:22PM 0 can asterisk send AT commands to a modem?
1:17PM 0 PRI ISDN NFAS configuration needed
1:07PM 3 Install Asterisk on CCM MCS-7835 Server
12:40PM 0 Re: T1/DS1/ISDN PRI
12:33PM 3 Music on Hold can' t hear it!
12:30PM 1 Asterisk CVS and bristuff-0.2.0-RC8a-CVS: no callerid
12:17PM 0 help / advice needed on a project
11:49AM 2 HTTP mirror, Digium's FTP server
11:49AM 0 No audio playback
11:11AM 13 Polycom IP500 - Phone TIme
10:52AM 1 RE: Number of production asterisk systems (Christopher Jacob)
10:49AM 4 Web interface Suggestions
10:38AM 0 BIND VoIP anyone?
10:17AM 2 VoicpulseConnect problems?
9:48AM 1 SIP calling Error from MP108 please help - confs included
9:43AM 1 Delete voicemail
9:09AM 3 Number of production asterisk systems
9:01AM 4 start asterisk
8:54AM 2 Console Warning Message
8:50AM 2 IAX attempt -> Segmentation fault
8:20AM 1 Experienced Asterisk Consultant in Chicago, IL
8:07AM 2 Prefix to CALLING Number ?
8:06AM 0 proper 2-card ISDN modem.conf configuration?
7:04AM 2 asterisk-h.323
6:59AM 0 MGCP and CISCO 7960?
6:42AM 0 Agents CallBackLogin and HangUp to calling party on pick-up
6:39AM 0 Advice on Adtran 600 setup
6:28AM 1 800 number provider suggestions
6:22AM 0 Asterisk Agents
5:59AM 12 Newer Dell Servers + TDM card
3:32AM 0 RSS feed Asterisk-Users
2:51AM 0 problem with skinny
2:38AM 1 H323 FAX
2:35AM 0 sip and analog
2:34AM 0 (no subject)
2:25AM 0 Problem with X101P(Red Alarm)
1:48AM 2 Monitoring B chans and G.729 High Water Marks
1:38AM 0 Incoming calls and CAPI
1:07AM 1 Eicon DIVA PCI ISDN cards (notserver) workwithasterisk!
12:48AM 1 Eicon DIVA PCI ISDN cards (not server) workwithasterisk!
12:40AM 0 call recording problem
Wednesday April 27 2005
11:37PM 0 Linux SoftPhone with Sound Daemon Support
11:22PM 2 Asterisk@home questions
8:32PM 0 Questions about ongoing calls
8:28PM 1 TDM400 doesn't know the hangup signal in china
6:42PM 1 Automatic Follow-Me Forwarding Based on Cell GPS
6:04PM 1 Transcoding times
5:33PM 8 Linksys/Cisco buys Sipura
5:32PM 0 Any other MoH source except *
5:30PM 0 Asterisk on a media stream vs. direct RTP communication between endpoints
5:27PM 1 Dialing out...
5:19PM 2 ATA 186 MGCP Firmware
4:48PM 5 IAX aproprietary protocol
4:04PM 4 * and Sipgate (UK)
4:03PM 0 ser rtpproxy asterisk problems....
2:59PM 1 SIP -> capi problem (no sound)
2:54PM 1 UK (english) sound files (Paul R
2:40PM 2 CDR Billing Question.
2:39PM 0 Anyting special needed for fax on a ATA186?
2:26PM 0 Call Type = Data
1:49PM 3 UK (english) sound files (Paul R)
1:48PM 0 Asterisk on Solaris 10 x86
1:27PM 2 Don't know what to do if second ROSE component is of type 0x6
12:49PM 0 wip 5000 in 12 hour time mode - anyone?
12:06PM 4 Panasonic KX-TD1232 Signaling
11:57AM 3 [Fwd: Voicemails stopping]
11:37AM 0 Public IP for SIP and NAT
11:35AM 0 Remote Phones - No Audio In Either
11:02AM 18 RJ45 to RJ11?
11:00AM 1 j'ai un probleme de connexion
10:24AM 2 Receiving Incoming Calls not working properly on TDM400P with 4 FXO modules
10:22AM 0 Transcoding Capacity
10:18AM 1 oh323 Zone
10:06AM 2 tonezone in tunisia
10:04AM 0 [Fwd: Supervised transfer problem.]
9:53AM 0 Re: Using Asterisk to dial a number and thenwait to dial the extension
9:42AM 1 Confused on G723 and G729
9:35AM 1 QuadBRI card on Suse 9.2 Unable to load qozap.ko
8:41AM 0 Determinating SIP Phone status
8:06AM 1 Connection Timeout problem with SIP phones from Gnet
7:40AM 6 Redirect two channels to each other?
7:26AM 2 Determinating Phone status
7:18AM 1 Grandstream BT101 Firmware
7:09AM 0 No audio playback after upgrade from 1.0.1
6:49AM 5 Cisco SIP Firmware Price Increase
6:26AM 0 SetGroup on dialed calls?
5:26AM 1 Dialing out from remote.
5:02AM 1 All lines are busy
4:21AM 2 noload breaksa IAX
4:14AM 2 agent monitor filename
4:12AM 1 Dial Tone
3:54AM 1 Asterisk doesn't disconnect when I hang up SIP (SIP -> PSTN call)
2:53AM 2 Supervised transfer problem.
2:30AM 1 Cisco 7.4 SIP firmware
2:08AM 1 b0rked hfc config
1:46AM 0 do I configure ISDN in zapata.conf?
1:42AM 1 No music on hold when transferring call
1:20AM 0 (no subject)
12:37AM 0 call a ldap result via my x-lite
12:23AM 2 RTP vs cRTP vs IAX
Tuesday April 26 2005
11:58PM 4 call a peer over the asterisk manager with a php script
10:25PM 2 Zaptel FXO crashing.
7:25PM 0 removing monitor IN/OUT wav file
7:22PM 0 Recommend IP Phones???
6:17PM 0 Auto Fax transfer problem?
6:03PM 2 SIP, Asterisk and NAT
5:56PM 1 The TCP in Asterisk
5:55PM 3 No Audio sent using playback cmd
5:45PM 2 US$100 bounty for two features in voicemail
5:16PM 7 Polycom Images
5:05PM 2 Warm standby boxes - keeping config syncronised?
3:52PM 1 Queue Management and Command Execution
3:23PM 0 Controlling extentions through Management Interface (fwd)
3:20PM 1 Variable names in dial plans
3:20PM 0 Controlling extentions through Management Interface
2:25PM 0 amaflags=Documentation
1:58PM 1 Is There Media Accelerator For Better AsteriskCalls
1:23PM 0 AAH 0.9 - SIP DTMF negotiation problem
1:12PM 1 Turn off Music on Hold
12:06PM 0 Stable Asterisk Version ??? / SIP problem ???
11:44AM 3 Remote Phones - No Audio In Either Direction
11:40AM 0 CID signalling for DTMF
11:29AM 6 Extensions / Contexts
11:19AM 1 the CLI dial command
11:16AM 0 Establish a ppp connection with a Pipeline 50
11:04AM 0 Voicemails stopping
11:02AM 2 Checking for a sound file
10:50AM 1 rxfax over IAX ulaw
10:34AM 1 Strange queue agent issue - Agent busy problem
10:31AM 2 Is There Media Accelerator For Better Asterisk Calls
10:12AM 1 Using Asterisk to dial a number and then wait to dial the extension
10:08AM 1 Cisco 7290 calling problems :-( - Sorry if this comes through twice
9:50AM 1 pri_dchannel: PRI got event: HDLC Abort
9:47AM 0 Re: Asterisk-Users Digest, Vol 9, Issue 224
9:45AM 1 Polycom Config - SIP 1.4.1
8:52AM 1 Cisco 7290 calling problems :-(
8:36AM 0 Polycom SIP 1.5.0 Firmware
8:05AM 0 asterisk xlite nat problem
8:02AM 0 japanese voice files
7:52AM 1 Fail over solutions
7:33AM 0 call an ldap search result
7:31AM 1 CLI dial command
7:20AM 0 X100P + spandsp locks machine with zaptel & asterisk 1.0.7
7:07AM 1 Cisco to buy Sipura
7:07AM 1 I wanted to understand
7:04AM 2 Zap/PRI: received AOC-E charging
6:58AM 3 YAC and IPs
6:56AM 2 SIP behind IPTables/NAT
6:48AM 1 Incoming Not Answering
6:43AM 1 return a value from dial macro
5:56AM 1 Dial CLI Command
5:40AM 4 Digium for ETSI ISDN
5:14AM 0 Cisco Systems to Acquire Sipura Technology
4:57AM 2 Shanghai or Bangalore DIDs
4:50AM 4 pridialplan/TON question
4:36AM 8 Good FXO for UK use.
4:28AM 1 How to set jitter buffer for SIP
4:21AM 0 ForkCDR question
4:09AM 0 i need Asterisk free Billing systems
4:08AM 1 Asterisk and Cisco Call Manager
3:57AM 0 bri cli error
3:45AM 2 Group/Broadcast Voicemail
3:08AM 0 Unexpected control subclass 17
2:53AM 0 ACD in Asterisk
2:02AM 1 how to use dialparties.agi
1:51AM 4 IP Softphone Recommendations
1:31AM 1 SIP/NetMeeting
1:29AM 0 help to configure sip server asterisk
12:55AM 0 Error on the Mysql, realtime database HELP soclose so far; .
12:38AM 0 Problem with long delay. VPN ?
12:29AM 5 VOIP Gateways & Asterisk
12:22AM 1 NO ringback tone for VOIP call to another SIP server
Monday April 25 2005
11:39PM 1 Distinctive ring on BT100
9:24PM 1 SV: Re: IPswitch: How to use speed dialing?
9:05PM 0 Only want softphone account from Vonage:
8:50PM 2 Digium Quad Span Cards
8:14PM 2 Error on the Mysql, realtime database HELP so close so far; .
6:35PM 1 Citel Handset Gateways
6:29PM 2 Playback dosen't play Playtone(Congestion) does ?
5:44PM 0 Asterisk ADSI
5:23PM 0 stanaphone now terminating fax
4:55PM 0 T1 E&M wink issues - bad int'l dial-outandoccasional dropped calls
4:45PM 2 Siemens SX66 wi-fi handset released
4:35PM 2 Polycom ip500 (Not-Registered)
4:20PM 0 using goto to do selective dialing
3:54PM 0 Transfers tend to fail after upgrade to 1.0.7
2:51PM 0 Dialing to a remote extension
2:30PM 0 Cannot make outgoing calls on Mediatrix 1204 from Asterisk
1:54PM 0 Asterisk replacing CCM using Catalyst 6608
1:32PM 7 Polycom IP4000 Conference Phone
1:04PM 5 voip problems
12:54PM 2 Has anyone used Libretel DIDs with Asterisk?
12:05PM 4 Dial Plan - How to prepend a digit
11:56AM 4 Phone Recommendation.
11:20AM 0 Does ztmonitor record the audio channel?
10:08AM 16 Broadvoice Down?
9:44AM 5 Grandstream ATA 286 problems
9:08AM 7 Alternatives to SpanDSP??
8:16AM 0 QoS Help and survey
8:07AM 4 astrecipes v2.0
7:08AM 0 Repost: Dialing problem - Cisco 7290 to anything
6:38AM 1 Basic telephony hardware questions
6:31AM 0 asttapi and identapop pro
6:08AM 2 Call Recording via monitor
6:04AM 0 need resources to include iax softphone functionality in vb6 app
4:39AM 2 What small PC can take 8 FXS + 8 FXO cards
4:39AM 0 chan_capi: no dialstatus, no causes, no branches
3:24AM 5 UK (english) sound files
3:23AM 0 each 64K channel's ABCD bits for E100P Digium Cards.
1:55AM 0 asteriks without h/w
1:34AM 0 [ANNOUNCEMENT] Amatix InstantPBX
1:19AM 2 signaling during a call
12:55AM 0 Zap event On hook(1) handling problem
12:34AM 1 No busy tone when dialing out over ISDN with Polycom 500 IP
Sunday April 24 2005
9:46PM 3 Trouble with call parking/transfer
8:24PM 1 Why can't I hear audio?
8:07PM 0 What software and types of connections are used by VOIP providers
7:45PM 1 Problems with gotoiftime and cvs head
7:00PM 2 Asterisk best practices
6:10PM 3 Static and echo on PRI
5:00PM 1 Transfers fails, even after upgrade to 1.0.7
2:44PM 3 T1 E&M false busy after dial
1:56PM 2 g729 passthrough?
1:00PM 0 T1 E&M wink issues - bad int'l dial-out andoccasional dropped calls
12:55PM 0 Need info on necessary config of new T1/PRIs
12:34PM 5 Can Asterisk do the following for me ?
11:38AM 4 How to prevent native bridging between SIP channels
11:15AM 1 sm bounty validate length of e164/e212 number for all countries
10:57AM 1 What is the best client's protocol for my softphones
10:53AM 2 Re: Asterisk-Users Digest, Vol 9, Issue 215
10:48AM 0 Fritz+chan_misdn - any working example ?
9:30AM 0 How can several Asterisk boxes working together?
9:28AM 0 cidsignailling mode question
9:17AM 0 Feedback on Junction Networks conferences?
8:44AM 1 T1 E&M wink issues - bad int'l dial-out and occasional dropped calls
8:18AM 1 Registerport 5060 or 1720?
7:51AM 1 Astcc Working but Can't Make The Call
7:07AM 0 Asterisk management GUI
7:05AM 0 VSAT and Asterisk
7:05AM 1 Asterisk2mp3
6:03AM 0 inband DTMF with IAX
5:50AM 0 QSIG.
5:40AM 2 Meetme Announcement
4:26AM 0 Netjet/Linux/Asterisk issue
4:15AM 5 ztdummy and Debian
3:14AM 0 help:Memory Consumption
Saturday April 23 2005
6:01PM 1 SIP registration behind Linksys WRT54G
12:07PM 3 Provisioning Lines
11:36AM 1 How to replace VM busy.gsm and unavail.gsm messages with custom files
10:55AM 2 ztcfg doesn't do anything from /etc/rc.d/rc.local
9:31AM 2 [Fwd: FW: IAX help]
8:05AM 1 OctoBRI and 2.6kernel
6:17AM 1 PA168 ip phone setup iax2 to LiveVoip
4:29AM 0 Dial While on IVR
3:32AM 0 usb phone(AU-100) and usb phone adapter(TJ560B)
3:29AM 0 chan_sip.c:7174 handle_request : Failed to authenticate user
3:25AM 0 ast_expr.y:243 to_integer:Overflow
3:17AM 1 Failed to authenticate
12:11AM 7 Hotel billing in IPSwitchBoard
Friday April 22 2005
11:16PM 0 Most affordable 8-port NT-capable ISDN card
11:14PM 0 Connecting Elmeg CS100 ISDN system phones to Asterisk
11:10PM 6 Best of the best of IP Phones
8:49PM 4 Cisco 7960 won't register as SIP device
7:38PM 4 if outgoing call fails with provider 1 then auto try provider 2
5:59PM 0 wIPPhone with Asterisk
5:20PM 2 console /distinctive ring
5:19PM 0 Libunicall Compile Error
5:09PM 5 IAX help
2:53PM 0 TDM-fxo card and zttest - logic probem?
2:44PM 0 Asterisk + Cisco 2620
2:32PM 0 Grandstream : low bandwidth codec (ilbc doesn't work, any other ? )
2:25PM 2 Recommendations for Spanish Voice Talent
2:19PM 2 Questions about a 7960 and images
12:28PM 0 Upgrade Cisco 7940/7960 firmware
11:45AM 1 Re: routing in extensions.conf
11:38AM 0 Digium Hardware Problem
11:30AM 0 IAX channel
11:12AM 4 Quadbri & bristuff: can * respond only to 1 MSN and leave 1 number to other ISDN phones ?
10:42AM 7 QOS Routers
9:52AM 2 voice pulse connect - no dtmf
9:20AM 0 ASTCC Database Creation
9:01AM 4 TE11OP -> Mitel 200Sx??
9:00AM 0 changing port on IAX2 protocol
8:40AM 3 chan capi: Long incomingmsn line in capi.conf?
8:32AM 1 No sound with voicemail and musiconhold?!?
7:55AM 1 Error loading zaptel on RHEL4
7:37AM 6 can't make my PRI dial out
6:51AM 0 Dynamic queue member behaviour
6:36AM 1 Alcaterl IP-touch phones
6:34AM 1 No such context/extension
6:20AM 0 Asterisk acting as PBX + SIP Proxy ... possible?
5:58AM 3 Mysql using Sip and voicemail
5:33AM 2 IAX2 Error
5:20AM 1 callto: URL (URI) tag for dialing
5:12AM 1 Asterisk transcoding
4:56AM 2 Asterisk Restart after crash
3:28AM 3 Dell PowerEdge SC1425 w/ TE405P?
3:14AM 1 DTFM tones almost completly muted.
2:22AM 2 X100P delayed ring on incoming calls?
1:08AM 0 Spandsp 0.0.2Pre15 with bristuff-0.2.0-RC8 Problem - Hangup
12:52AM 0 dialling problem with astcc
12:41AM 0 How to attended/supervisor transfer
12:26AM 1 Echo cancelling with Adit 600
12:16AM 0 Asterisk increased memory
Thursday April 21 2005
11:51PM 0 LiveVoip status report
10:41PM 0 Provider offering IAX and T.38 origination+termination?
10:09PM 1 Digium Card Issues
7:48PM 1 TE110p - universal voltage?
6:54PM 0 100 & AAH .9
6:39PM 4 Bug?
6:39PM 8 Email to Fax
6:22PM 2 asterisk@home 0.9 zap problems
6:18PM 0 Music Onhold Problem
6:15PM 0 Looking for an IAX(2) or SIP DID provider for LA, Orlando and Chicago areas.
6:07PM 0 GS ATA 286 goes deaf.
5:06PM 4 Demo phones with advertisement announcements
4:18PM 2 Playing mp3's while recording voicemail
3:41PM 0 does ast_app_getdata() reset timeout?
3:19PM 1 Recording Queue agents
2:54PM 1 forwarding Sip call to IAX and vice-versa
2:48PM 2 Provisioning lines 5 and 6 via TFTP
2:39PM 2 using * for Internet call waiting
2:16PM 0 CVS-HEAD: Sip not paying attention to context
1:55PM 1 Queue member persistent stats
1:08PM 1 Multiple Line config help
12:35PM 3 Broadvoice gateways!
12:23PM 5 One touch voicemail on Cisco 7940/60
12:15PM 0 leastrecent queue option
12:04PM 1 Problems with app_dbodbc.c
12:01PM 1 adding a thrid asterisk server
11:08AM 0 AgentCallbackLogin AckCall Problems (Current CVS-HEAD)
10:17AM 1 Libunicall Make Error
9:59AM 1 ZAP - outgoing call using different D-Channel each time ?
9:54AM 0 Re: Basic Setup Question
9:06AM 1 Error in starting asterisk
9:03AM 1 Max concurrent faxes using SpanDSP?
8:41AM 1 security
8:00AM 1 dialup internet via Asterisk
7:51AM 0 * not send SIP Notify for IAX2 channel
7:45AM 1 503 Error
7:36AM 1 PBX replacement
7:31AM 1 Queues configuration
7:18AM 0 hint priority and realtime in asterisk cvs-head
7:11AM 1 Odp: Re: capi problem with dialout
6:58AM 1 Bristuff and Belgium
6:39AM 2 i like my colors, thanks..
5:08AM 1 Deny certain extension
3:37AM 0 Problems with soundcards
3:09AM 4 capi problem with dialout
2:42AM 3 Dial W option usage
2:30AM 0 Attended Transfer
2:29AM 0 Problem using ztdummy kernelmodul with Kernel 2.6.8
2:11AM 0 Asterisk Cisco Connection
1:33AM 0 RE: Large Asterisk Setup (~500 Concurrent Calls + Scalability)
1:26AM 0 Asterisk Cisco Conection
Wednesday April 20 2005
11:18PM 2 Fax Problems
11:05PM 0 Tonelist questions
9:46PM 1 Zap channels busy. Have to soft hangup.
8:26PM 6 Recommended Linux Dist. for Asterisk
8:01PM 4 asterisk home wiring question
7:47PM 0 Queuing with busy detect
7:36PM 0 Volume of call waiting beeps
7:32PM 1 mpg123 won't compile, arch x86_64
7:15PM 1 Large Asterisk Setup (~500 Concurrent Calls +Scalability)
7:14PM 1 What do I need to get started?
7:01PM 0 Ringing problems was TDM400P Revision question.
5:51PM 0 error in asterisk and LOTS OF log files generated
4:55PM 3 TE110P
3:30PM 3 chan_unicall.c compile error
3:27PM 0 What do Digium use for tracking support tickets?
3:07PM 0 spa 3000 pstn with amp
3:03PM 1 TE110P card installation errors
2:03PM 1 Anyone have a GXP-2000 working with Asterisk yet?
2:01PM 0 Help with [codec_g729.c:196 g729tolin_framein:Invalid data]
1:46PM 0 Lucent EMRS PRI Card
1:43PM 1 Adit 3104 - user experiences?
12:58PM 3 GotoIf in Stable 1.0.4
12:58PM 0 One-Way NO audio (and sometimes both ways)
11:42AM 0 choose audio codec with chan_sccp driver and 7920 wireless?
11:18AM 11 BYOD provider other than broadvoice
11:14AM 1 Annoying SIP registration problem behind ?Linksys?
10:58AM 0 Recommendations for IAX/SIP ATA
10:46AM 1 FXS --> FXO Converter
10:40AM 1 Dialplan not showing up.
10:29AM 1 Zap Extensions unavailable after a call
10:08AM 0 Route SIP calls to provider
9:58AM 0 Active calls not responding to entries
9:51AM 3 Line Noise UPDATE - If you've got line noise, read this
9:07AM 0 ADSI phones in the UK
8:58AM 1 RE: Re: a simple question
8:56AM 0 ???
8:50AM 2 Monitor via Manager question
8:40AM 0 Which free calling card app most suitedforcommercial use?
8:25AM 3 Transfer of incoming call from external to internal number
8:19AM 2 Cisco 7960 SIP registration???
8:12AM 4 webcall
7:56AM 0 RxFax not hanging up...
7:46AM 1 CVS Head and SetLanguage
7:32AM 1 Can I do something with Caller-ID?
7:29AM 2 Wait in Dial String
6:56AM 4 G723.1 and G729 on Athlon 64
6:47AM 1 General voip mailing list
6:46AM 1 Which free calling card app most suited forcommercial use?
6:46AM 0 FXO lines on TDM04B not responding
6:01AM 0 Help with [codec_g729.c:196 g729tolin_framein: Invalid data]
5:23AM 0 IPSwitchBoard connects to CDR
5:21AM 2 A question about queues
5:00AM 0 Asterisk + Adit 600 questions
4:26AM 3 Issues of reliability, hardware, platforms
3:52AM 0 Cisco 2800 with Asterisk
3:29AM 1 Snom 360s and Asterisk
3:25AM 4 Asterisk and VAD
3:21AM 1 TE410P PCI-slot
3:11AM 1 NAT issues
2:36AM 0 Cisco ATA Help
2:18AM 0 "friendly networks" via **
2:17AM 3 Setting SIP username for CallerID
1:52AM 2 OH323 incoming audio stutter
12:09AM 2 IAX realtime HELP
Tuesday April 19 2005
11:32PM 0 Text Messages
11:20PM 1 NAT and only been able to have 1 SIP phone behind
11:06PM 2 RealTime ignoring switch => Realtime/context@realtime_ext
10:44PM 1 help needed for sound device setup
10:06PM 1 Sample AGI Scripts in C needed.
9:46PM 2 CVS-HEAD and CheckGroup/SetGroup
9:27PM 0 VoiceMail Config Questions
9:14PM 2 SIP Phone Compatability
8:19PM 0 Libunicall
8:06PM 1 FW: Cisco 7920 - chan_sccp - asterisk@home .9
7:59PM 0 Cisco 7920 - chan_sccp - asterisk@home .9
7:18PM 1 Cisco 7960/7960G
7:05PM 0 Asterisk@Home v.0.9 and Digum
7:01PM 1 NuFone problems to non-na numbers
6:29PM 0 Looking for a softswitch
5:11PM 0 Attended transfer on sipura ATA/Phone?
4:50PM 0 TDM400P and SCSI/SATA = * noise problems???
4:33PM 0 Industrial Cordless handsets analog or voip based ??
3:56PM 11 US$200 bounty for * paging feature
3:03PM 0 Help needed on Utstarcom F1000 Wifi Handset.
2:29PM 0 Which free calling card app most suited for commercial use?
2:15PM 0 Any work around for ISPs that block port 5060
1:30PM 0 New AstManProxy Manager Proxy v0.98
1:23PM 1 Fax and spandsp
1:21PM 0 BRI channels not answering
1:17PM 0 Show accountcode in both directions?
1:16PM 1 Asterisk Netgear FSM7326P and Cisco 7960 on VLAN
1:11PM 0 Server failing to Boot with
1:09PM 0 Unix softphone
12:58PM 0 Latest CVS breaks voicemail app
12:44PM 0 Dutch callerid: sending on FXS?
11:59AM 1 ATA - PBX
11:21AM 0 mysql from dialplan
11:20AM 1 ASterisk OH323.CONF Gateway & Gatekeeper
11:01AM 4 Asterisk Business Case - Who is using it!?
10:37AM 2 OutBOund Dial problem
10:36AM 5 Conference solution for 100+ users
10:14AM 1 PRI - T1 feasibility
9:34AM 1 Extensions unavailable after to sucessfull call (Registration lose)
9:31AM 0 Testing the TDM01A
9:25AM 1 Re: Any work around for ISPs that block port ....
9:12AM 2 Want to use Asterisk instead of existingMeridianNorstar system ... need some help
8:54AM 0 Asterisk and Request Tracker, RT.?
8:54AM 3 Using voicemail independently from Asterisk PBX
8:17AM 1 Cisco 7960 directory.xlm
8:16AM 1 Testing my TDM01A
7:36AM 1 VoIP PSTN numbers in Australia?
7:29AM 0 Sipura PSA-841 -suitable headset
7:18AM 1 Any work around for ISPs that block port 5060 and69
7:12AM 7 Firefly w/*?
7:11AM 1 IPv6 possible?
7:10AM 1 Soft Video phone for Windows XP
6:48AM 5 Any work around for ISPs that block port 5060 and 69
6:31AM 1 Re: [Serusers] Ser + Asterisk
6:28AM 1 802.1p , precedence and TOS
6:18AM 5 Voicemail email text:
6:04AM 0 Snom NOTIFY on IAX2 channel
5:53AM 0 answered time
5:40AM 3 IPTables
5:10AM 0 SIP users, OH323 to provider, g729 - high level of echo
4:59AM 0 Looking for some real basic doccos...
4:58AM 0 AT-320 phones with IAX2
4:31AM 8 VPN/Asterisk combo
3:20AM 3 Newbie - VoIP route SIP calls to provider
3:09AM 2 Asterisk and T.38.
2:43AM 1 TE405p PRI ISDN [E1] RED Recovering ?
2:08AM 2 Installed ztdummy, Asterisk doesnt work anymore
2:08AM 6 Asterisk with Softswitch
1:51AM 2 DID ~ Extension
1:31AM 1 Sipura SPA-841 distinctive ring
12:53AM 0 Astrisk + Cisco 5350
12:49AM 0 codec negotiation with CISCO 7960 and Firefly softphone
12:33AM 7 Billing
12:33AM 0 Codec/Phone negociation(s)
Monday April 18 2005
11:57PM 1 CLI Numbers
11:18PM 1 Asterisk timer on Digium's TDM cards?
11:03PM 0 Queues-Agents Problem
11:01PM 1 DTMF in outbound calls
10:23PM 1 Vici Dialer
10:20PM 0 Remapping Woes in features.conf
10:18PM 6 Asterisk & POE
10:11PM 1 HELP: How to detect a hangup tone?
9:25PM 1 G729 Key Registration Problem
9:09PM 1 Want to use Asterisk instead of existing Meridian Norstar system ... need some help
8:03PM 0 SIP calls being lost "frame from cahnnel" error
7:36PM 0 Extension busy issue on TDM01A
7:33PM 1 Blind Transfers - any ideas?
7:30PM 1 DTMF intermittently stops working
7:30PM 4 Citrix
6:33PM 2 Problems with incoming calls on a E1 ISDN PRI
5:37PM 1 wcte11xp digium card
5:03PM 0 Why *3* entries?
5:02PM 1 RealTime Vs. AGI and PHP or MySQL calls within extensions.conf
4:56PM 0 [Announcement] Updated Web-MeetMe
4:09PM 0 zombie channels & missed transfer
4:05PM 1 asterisk on MIPS
3:42PM 2 system wide speed dialing
3:36PM 1 snom 220 hints lost after reload
3:32PM 1 Problems with Cisco ATA 186/MGCP
3:24PM 1 Looking for ATAs
2:40PM 1 Junghans QuadBRI and fax detection
2:10PM 1 callback broken?
1:53PM 2 Fedore CORE 2
12:30PM 0 ackcall with AddQueueMember
12:00PM 1 Random SIP Phone Problem
11:57AM 1 Hold on outbound calls and the SNOM 190
11:14AM 1 [Fwd: Re: Only one PRI out of four working on TE405p?]
11:11AM 0 maximum value for LEN(x)
10:12AM 0 Voicemail not working...
10:04AM 2 Only one PRI out of four working on TE405p?
9:47AM 0 Unable to specify channel 1: No such device
9:33AM 0 Cisco 7970 startup problem
9:14AM 0 Strange tones when placing a PSTN call.
9:11AM 0 Lots of RTP checksum errors
9:01AM 1 Asterix Manager Proxy in Java/EJB?
8:51AM 3 Can I use Asterisk for a modified Hoot and Holler?
8:44AM 8 Calling Card
8:34AM 2 Snom subscribe/notify problem
8:24AM 4 Motherboard failure with 2 Digium TE405P car ds
7:34AM 2 Motherboard failure with 2 Digium TE405P cards
7:28AM 4 Help compiling zaptel in Debian
7:22AM 0 Indicating when other party has answered
7:08AM 0 Follow-me script - user changeable options
6:56AM 3 99% CPU - CVS 03.28.05
6:22AM 0 Fw: Analogue phone transfering
6:16AM 3 queue - transfer calls
5:13AM 1 Changing Codecs when dialing out...
5:01AM 3 Cisco External Directory
4:35AM 1 Still having broadvoice issues
2:31AM 0 Error on install of AMP
2:26AM 1 Distributed organizations - large scale public sector rollout
1:16AM 1 Got SIP response 302 "Moved Temporarily" back....
1:09AM 1 analog gsm router
Sunday April 17 2005
10:58PM 2 Dynamic Dialplan - Turn VM on/off?
10:35PM 1 dynamic callrouting and billing?
10:29PM 1 hangs pc
8:31PM 1 Digium G.729 vs. IPP G.729
6:22PM 3 Can anyone send me sample config files for asterisk and X-Lite?
6:11PM 3 Register two account at Broadvoice with one asterisk box
4:48PM 1 extension dialing resistivity
2:37PM 3 Unbelievable...
2:03PM 0 E & M signalling with WCTE11XP - not all calls go through
12:53PM 3 spandsp and cvs head
12:34PM 1 High Availability - Again
12:26PM 2 ISDN BRI vs. VOIP DID's, is it worth it?
12:22PM 0 RE: Asterisk-Users Digest, Vol 9, Issue 152
10:08AM 1 IPP g729 & x86_64
9:49AM 0 Bandwidth Reduction using Compressed RTP
8:29AM 0 app_dtmftotext.c
6:44AM 1 res_perl compile problem
6:31AM 0 cisco mgcp and CARD.XML
3:58AM 0 Zaptel fxo & late distinctive ring
3:34AM 1 IPSwitchBoard Version 0.91 Released
2:55AM 2 Illegal instruction (core dumped)
1:41AM 1 Line name same as user name
12:59AM 0 Point-to-Point Asterisk Link to Reduce Bandwidth
12:09AM 1 OT VoIP related jobs in Eu
Saturday April 16 2005
10:13PM 2 Park a call then hunt for a *willing* person
8:59PM 4 Hitachi WIP-5000/IP-5000 firmware
7:19PM 0 Can't Native Bridge Any More
7:04PM 3 recommandation for four (4) port FXS ATA
4:44PM 0 Receptionist Module
4:20PM 2 SIP/iax devices in Russia
4:18PM 3 problem connecting multiple boxes via IAX2
2:45PM 2 Slightly [OT] Asterisk Backends
1:50PM 1 first few seconds of outgoing calls cut off
1:11PM 0 can't use 2 port gws simultaneosuly
12:18PM 0 Sipura SPA-2000 correct settings for Fax in The Netherlands/Europe
11:42AM 0 Sipura SPA-1001 Setup/Review
10:17AM 1 BT100 wrong NAT detection
9:21AM 0 Problem with wipphone
9:18AM 1 zap device detects hangup when phone switches from answer machine announcement to recording
9:08AM 1 2 Questions
8:58AM 0 Asterisk and openbrick
8:53AM 0 Mitel 5055 dead after wrong flash, any tips appreciated
8:33AM 1 Cisco/Asterisk codec negotiation problems
7:58AM 1 Is this normal - Long time to make call - What is your average with your Hardware?
7:48AM 3 VOIP to PTSN provider
7:23AM 0 Lots of RTP checksum error
7:10AM 1 OT: Sourcing Equipment at the HK Electronics Fair
7:06AM 0 [Fwd: Re: Debugging zaphfc + PBX integration]
6:27AM 0 OT: Interview With Kevin Fleming
5:50AM 1 IPSwitchBoard now has Zap Support
5:03AM 0 Codec Linux Bandwidth Reading
5:01AM 2 IPswitch: How to use speed dialing?
3:10AM 3 Installing Asterisk@Home on VMware Workstation 4.5.2- build 8848
3:00AM 1 Asterisk@Home & ISDN BRI
2:36AM 0 FXO GW Dial in/out syntax
2:10AM 4 Asterisk and network problems
1:43AM 2 Extensions busy queue
12:39AM 0 acd+transfer+asterisk-1.0.7
Friday April 15 2005
11:15PM 1 Asterisk connect to Asterisk
8:48PM 0 Help needed to configure Asterisk and SJPhone
8:00PM 1 Dialplan help needed
7:05PM 0 AMP/Asterisk
6:04PM 3 a simple question .
4:32PM 0 IAXTEL Passord
3:39PM 0 How do I connect my Asterisk PBX to a serviceProvider
3:30PM 3 What is the good client softphone for windows?
3:27PM 0 How do I connect PC clients to my Asterisk PBX
3:24PM 1 How do I connect my Asterisk PBX to a service Provider
3:21PM 1 How do I make Extention in my Asterisk PBX
2:24PM 5 IAX softphone
2:24PM 4 Can't Modprobe ztcfg
2:22PM 0 FW: USB Controller ztdummy
1:35PM 1 fax detect/transfer problem solved
1:20PM 2 Anyone already pionered outing calling with user selcted background noise?
12:51PM 1 USB Controller ztdummy
11:41AM 1 Keypad disabled on AriaVoice SIP phone -- Fixed
11:30AM 0 How do I connect two Asterisk in different domains
11:08AM 5 OT: USB handsets / softphones
10:50AM 8 OT: google groups Asterisk-test and now Asterisk-Users marked as spam on Gmail
10:25AM 2 H323 Large Scale
10:09AM 4 large analog to asterisk
9:54AM 0 How to avoid CTL file request for Cisco 7970
9:48AM 0 Outgoing PRI Call Early Media Detection
9:43AM 0 about volume in Playback() files
9:12AM 2 Bridging 2 Zap channels
8:56AM 0 Maybe not worded right, Answering a call
8:25AM 0 Ring requested on unconfigured channel 0/31 span 1?
7:51AM 2 sipXphone
7:28AM 1 ilbc codec in Asterisk
6:57AM 0 Polycom IP500 phones do not update time fromtime server
6:27AM 0 Asterisk working on FC3+X100P+France Telecom line
6:25AM 0 Question on Asterisk CDR / "In-Network Calling" / MySQL CDR
6:22AM 2 Debugging zaphfc + PBX integration
5:50AM 0 SIP Message Waiting Notification
5:31AM 0 Excessive re-registration of Broadvoice account in Asterisk@Home 0.8
4:41AM 0 howto forward UAC codec capabilities to the PSTN gw
4:38AM 0 Problems with a SMS-capable Phone on a ZAP Channel / Question about native bridging on digium cards
4:26AM 0 g729 not work with DTMF and AGI
4:18AM 2 Slack 10 install - THANK YOU - & Cisco Reseller Help
4:05AM 1 SIP stack pluggable?
4:04AM 0 VIC2BRI and J4BRI
4:00AM 0 SIP through firewall is intermittent
3:55AM 0 IAX2 to IAX2 - one way audio
3:49AM 1 Asterisk live chat problem
3:31AM 1 Analogue phone transfering
3:14AM 2 Empty voicemail attachments?
2:58AM 0 LiveVoip incoming, no ringback still
2:38AM 0 qos test
2:19AM 0 Urgent .... Asterisk <-> Cisco CCM SIP TRUNK
2:15AM 3 *8 nor *8# works for me!
2:14AM 0 UDP Sip Data: GS Grandstream - remote office
2:08AM 0 E1 PRI: Unable to set channel to linear mode?
1:49AM 0 OH323 and outgoing calls problem.
1:29AM 4 Asterisk PBX with X100P in India
1:12AM 0 Soekris net4801 usb isdn avm fritz
12:48AM 0 OctoBRI - unable to specify channel 1
Thursday April 14 2005
11:38PM 2 Grandstream BT Volume
9:43PM 3 distribute outbound calls
8:00PM 0 dropping inbound calls from certain regions
7:18PM 1 DID reseller structures
6:46PM 1 How do I make a call thru *PBX
6:33PM 0 Mark Spencer and John "Maddog" Hall visiting Toronto - come and join us
6:31PM 6 cisco 7960 SIP setup
5:11PM 1 DTMF does not work with g729 and AGI
4:57PM 0 Cant respond to prompts from SPA1001
4:14PM 2 Fax questions
3:44PM 2 ISDN BRI and signalling
3:38PM 1 Dial Macro Arguments
3:34PM 3 codec introducing huge latency
3:25PM 1 I dont want to hear the FXS port ring - TDM400?
2:57PM 1 asterisk + OH323 + NAT + gnomemeeting
2:38PM 2 Problem with Livevoip incoming context
2:38PM 0 How to reduce asterisk CPU-LOAD?
2:34PM 0 Bizarre - VM just stopped for one user
2:27PM 0 Routing on called number via SIP
2:12PM 1 MFCR2 compile requirements
2:00PM 0 Custom/Vanity DIDs
1:37PM 0 Call Parking timming out to the wrong extension
1:09PM 0 Voicemail delivery to pbx or mobile/panasonic dbs
12:52PM 0 Fritz Card going Crazy to make it compile
12:13PM 11 Overheard conversation. Comments please !
12:11PM 2 Invalid extension handling
11:42AM 5 Line Presence:
11:39AM 2 making an action based on the status of multiple extensions
10:10AM 0 Re: Polycom IP500 phones do not update time from
10:05AM 4 Voicemail name (greet.wav) is not played
10:00AM 7 TDM400P Revision question.
9:52AM 1 Call Files to Terminate a call to the dialplan not directly to a channel
9:34AM 1 Wall Mount PC Case
9:33AM 0 asterisk hosting
9:22AM 5 Polycom IP500 phones do not update time from time server
9:06AM 4 Siemens optiPoint 420 phone and Asterisk
8:57AM 0 Matching on '@' in extensions
8:55AM 1 G.729A codec amd64/intel x86-64 optimisation?
8:17AM 5 Toshiba CTX100 integration with PABX for two site
8:14AM 1 Polycom IP500 phones and Presence feature
8:01AM 2 Voicemail Email
7:33AM 3 Who is a QUALITY IAX Termination Provider for 800 DID's?
7:29AM 1 Zap won't dial out?
7:28AM 1 Segregating a test version of asterisk - libpri/zaptel locations
7:26AM 0 cisco 79xx and SIP call statistics
7:24AM 1 Steal a call from a SIP extension
7:10AM 3 Ring two extensions at the same time
7:06AM 1 MoH stopped working with cisco 7912/7960
6:45AM 3 delay problem in asterisk
6:30AM 0 IPSwitchBoard Version 0.86 Released
6:02AM 1 SIP Incoming Problem
5:31AM 1 BOUNTY: app_hangup from exten => h
5:28AM 1 BOUNTY - ztdummy & modules
5:07AM 1 sip phones make connection but no-sound is heared
5:04AM 1 Asterisk@home first experience
4:41AM 1 Re: Asterisk-Users Sip Reload or Realtime
4:36AM 1 Dialing rules
4:20AM 0 no voice tone
3:59AM 2 ISDN BRI + echo cancelling + Fax
3:08AM 0 Dropped calls from Junghans octo-bri card
3:02AM 1 lost DTMF digits
2:50AM 0 <register> syntax and limitation therewith
2:10AM 2 voicetronix bri
2:03AM 2 IAX blind transfers
1:58AM 1 Is there a SIP protocol stack inside asterisk?
1:51AM 1 need urgent help
1:39AM 3 Hylafax and Asterisk
1:00AM 1 pbx to asterisk
12:35AM 1 Cisco 7960 command-line dialer
Wednesday April 13 2005
11:55PM 2 RTP problem
11:35PM 1 trying the xc-ast queue_log analyzer
11:31PM 1 Strange intermittent NAT problem with BT100s
10:56PM 5 RTP not being sent by asterisk
10:01PM 0 Re: Running asterisk without special hardwar e
9:58PM 1 oh-323 compilation error !
9:14PM 1 Channel 0 on Zap ???
9:07PM 2 Changing IRQ's on TDM
8:53PM 1 cannot dial two phones using zap
7:10PM 2 New Zealand Telco (TelstraClear) query
7:09PM 1 bashing my head against broadvoice
7:02PM 2 Cannot dial two phones at the same time
6:45PM 1 show translation
6:28PM 1 SIP Deadlock problem.
6:28PM 1 ZyXEL Router Terrible Voice Quality
6:22PM 0 H.323 in CVS Head
6:13PM 5 Telephone line installation.
5:34PM 3 does meetme need ztdummy
4:58PM 2 trying to figure out a few error messages in *
4:13PM 0 connecting Asterisk as SIP gateway to a Verso BHT1000
3:27PM 0 about sip and skinny
3:04PM 2 Polycom Vendor Recommendation
2:40PM 0 Help, Outbound Problems
2:10PM 1 DISA() and predefined ACCOUNTCODE variable
1:55PM 0 SIP Clients over Wan losing connection
1:51PM 1 Advice sought on how to automatically and sa fely reboot * box
1:27PM 9 Asterisk@Home 0.9 released
1:11PM 0 Advice sought on how to automatically and safely reboot * box
12:53PM 0 Choppy music on hold
12:45PM 3 Why does this Macro Loop?
12:42PM 2 TDM card periodic buzz
12:30PM 1 PSTN VOIP integration not allowed in INDIA
12:29PM 2 Loop Detection
12:25PM 0 AW: SIP registration fails
12:08PM 1 Grandstream Won't hangup like Polycom 600 will
11:51AM 4 VOIP Regulations in INDIA
11:39AM 1 asterisk from cvs head crashes on via samuel 2, kernel 2.6.11-gentoo-r4
11:23AM 2 SPA-3000 and quiet voicemail
11:06AM 0 need to ask you about your dlink and NAT/voip
10:56AM 0 Clipcomm CG-410 with asterisk?
10:39AM 0 Asterisk crashing? (gdb trace included)
10:18AM 2 IAX introducing huge latency
10:08AM 0 Asterisk on debian sarge doesn't start with CAPImodule errors
10:07AM 0 polycom dial...rings 4 ever, but redial connects
9:59AM 1 sip reload or realtime
9:11AM 1 SIP ACD system for station to station calls
9:02AM 3 IAXy Provision
8:57AM 2 Pretty Voicemail Docs
8:50AM 0 CVS-HEAD Zaptel with 1.0.x CVS Asterisk
8:20AM 2 ZAP channel hangs up with no apparent reason
8:19AM 2 Newbie Question on how to handle main office number
8:12AM 3 Unable to register license for G729 codec
8:02AM 0 FRAME_CONTROL (5) dropping calls on PRI
7:48AM 0 IPSwitchBoard is now Event Driven
7:16AM 1 Transferring a call
7:14AM 3 Zaptel and Fritz Card
7:06AM 1 SNOM 220 with >7 "lines"
7:02AM 0 PCI 1xE1, 2xE1 cards from Russia for MFC/R2 signaling for Asterisk IP-PBX
6:53AM 1 ni1 (ppp) and national isdn on te110p
6:44AM 1 Asterisk on debian sarge doesn't start with CAPI module errors
6:21AM 3 Who is willing to help an Asterisk newby?
6:05AM 2 OH323 and Asterisk CVS-HEAD-03/21/05-15:32:10
6:03AM 4 3-Way Calling in Asterisk
5:51AM 1 x-ten lite error
5:31AM 0 Question about Routing Order in .conf files
5:27AM 3 Cisco 7940G SIP Conversion
5:11AM 0 Asterisk / Quintum CRSP codec problems
4:01AM 0 codec quality
3:16AM 4 ISDN Fritz and TDM400
2:55AM 0 Turtle Firewall - Sip user
2:08AM 2 iaxcomm
12:13AM 0 Sipura SPA-841 and Asterisk 1.0.7 with chan_misdn
Tuesday April 12 2005
11:52PM 1 Codecs and * pass through...
11:40PM 1 New PRI install with new te110p
11:16PM 1 attension mark spencer
10:22PM 0 FW: binding Asterisk to virtual IP
10:19PM 0 chan_sip.c:7215 handle_request
10:09PM 1 Interesting Cisco 7960 issue, the phone picks up without me!
9:49PM 2 Problem reading digits from OH323 caller
9:48PM 1 weird call transfer problem
9:42PM 1 Article on IAX in Network World
9:35PM 2 invalid extension (need help)
9:26PM 0 invalid extension(need help)
9:14PM 0 Looking for comments on robustness of SpanDS P / app-rxfax / mime-construct <--solved
9:03PM 0 Hangup after Transfer problem.
8:46PM 0 Testing call back to back mfcr2 using chan_unicall with TE400P
8:36PM 1 Running asterisk without special hardware
8:29PM 1 Compile/modprobe issue
8:10PM 0 semantics terminology
7:39PM 2 Question about Macros
7:09PM 1 Blank voicemails being sent to users
6:52PM 0 Australian anologue callerid
6:48PM 4 New SNOM 190 Firmware
6:25PM 2 Outbound calling stops working after configuring trunk lines
6:11PM 0 OH323: Sending CallerID to H323 voip provider...
4:06PM 1 Realtime Friends
3:48PM 0 Looking for comments on robustness of SpanDS P / app-rxfax / mime-construct <--More information, and I figured out wh y
3:30PM 3 binding Asterisk to virtual IP
3:17PM 1 voice mail playback
2:50PM 1 Re: [Asterisk-Dev] Iax Trunking LD Service
2:47PM 0 Jitter problems IAX to Livevoip
2:37PM 0 TDM02B on 2 a/b ports of a PBX not working.. . help
1:57PM 3 IAX2 - Between two ASterisk Servers
1:56PM 0 OH323 and outgoing calls.
1:33PM 0 503 Service Unavailable
1:23PM 0 RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars
12:57PM 2 Looking for comments on robustness of SpanDSP / app-rxfax / mime-construct
12:54PM 1 Re: How do I reduce echo on asterisk
12:05PM 3 overwriting config file problem
12:04PM 3 RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars
11:59AM 0 Asterisk@Home - Newer Mobo - Memory
11:55AM 0 Anybody who is using a live stream as MOH
11:54AM 0 Music on hold not working between SIP clients
11:34AM 1 g729 versus g711
11:29AM 0 Accountcode in SIP.CONF not set
11:21AM 1 Attempting native bridge of
11:11AM 0 Iaxy, Transfer, & #
10:56AM 0 Dumb question ?
10:48AM 3 Cisco 7960s and skinny
10:36AM 0 Re: Dumb question ?
10:30AM 0 Meetme and billing
10:16AM 4 Local Echo
9:51AM 0 LCDial and default provider
9:48AM 1 How many licenses of G729 do I need?
9:10AM 0 Noises on ZAP Channels
8:59AM 1 How do I reduce echo on the Caller side
8:52AM 1 Multiple TDM cards on the same box
8:47AM 0 Re: Asterisk-Users Digest, Vol 9, Issue 104
8:36AM 1 QoS TOS numbers and Cisco IOS
8:26AM 0 Dialing Out (My mistake, here is the entire message)
8:01AM 0 RE: Ebay listing selling Asterisk @ Home (Blah Blah)
7:55AM 5 Acceptable voice time delay
7:37AM 0 Power Consumption of a Digium Wildcard TE410P
7:18AM 5 multiple line usage on Polycom IP300
7:09AM 0 Internet Conection Broken and asterisk can not route any calls
7:05AM 0 Meetme disconnecting clients that use VAD
7:02AM 0 NENA CAMA Trunks for 911 and *
7:00AM 1 Low cost box for hosting Asterisk and atleastoneTDM400p - THIN CLIENT MAYBE?
6:34AM 2 How to get list of codecs
6:05AM 0 multiple asterisk boxes with "show channels"
5:14AM 2 TE410P and X101P problem
4:40AM 6 Version 0.80 of IPS released
4:23AM 1 Problem with fxo
3:15AM 3 TE110P - NT-Mode ?
2:57AM 0 H.323 Question
2:36AM 1 Asterisk Addons compile errors
1:56AM 1 Supervisor monitor / barge in - automatically on call setup?
1:45AM 0 Voicemail quota
1:36AM 0 Asterisk on HP DL380 G4 - problems
12:42AM 0 Problem with * transfer
12:19AM 1 Has anyone got Asterisk working behind a NAT connection to users within a NAT
Monday April 11 2005
10:18PM 1 Losing CallerName info if no CID sent
9:45PM 2 Zyxel P2000W Finally (Almost) Working
8:28PM 3 Trunk Seize - Line 1 - CO1: Does it exist in an Asterisk environment?
8:04PM 0 H.323 General Questions
8:03PM 4 Asterisk management portal
7:48PM 0 ChanSpy -- Slow/garbled audio and console errors
7:41PM 1 Remote phone often appears to be disconnected
7:25PM 0 Asterisk Realtime - can't see sip friend
7:24PM 0 E911?
7:22PM 0 Asterisk did not play music when pressing hold button on SJPhone
6:23PM 1 Problem Detecting Answer on a PRI Outcall (sometimes)
6:06PM 1 Monitor with Asterisk@Home
5:35PM 0 OT: Thunderbird threading
5:11PM 1 Dialing Out
5:11PM 0 Advice in provider for business use
4:38PM 1 Best FXO Voip Gateway for Asterisk
4:32PM 0 ASTCC - IVR prompts
4:23PM 0 Asterisk-Users] RE:Asterisk Voice mail with CCM
3:56PM 1 Dialogic cards compatibility
3:43PM 1 Pre-install questions
3:36PM 2 Linksys PAP2 Dual Incoming Calls
3:25PM 0 Q931 Setup message
3:14PM 0 Maximum amount of users on one asterisk server?
3:03PM 2 broadvoice config problem.
2:52PM 0 Re: Using manager interface to play aanouncmentsinaMeetMe
2:41PM 1 Check for stutter dialtone on ZAP FXO channel
2:35PM 4 RE: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars
1:58PM 1 Sip transfer and redirect in a Company setting
1:40PM 0 RE: Ebay listing selling Asterisk @ Home and AM P for over 1000 dollars
1:01PM 1 RE: Ebay listing selling Asterisk @ Home and AMP for over 1000 dollars
12:55PM 1 Low cost box for hosting Asterisk and at leastoneTDM400p
12:39PM 3 BROADVOICE - Incomming calls are dropped after 1-2 min
12:34PM 1 "Refresh" asterisk internal database?
11:55AM 2 Play Sound File Without Answer Channel
11:54AM 0 Re: Asterisk-Users Digest, Vol 9, Issue 96
11:41AM 3 Low cost box for hosting Asterisk and at leastone TDM400p
11:24AM 1 Suggestions about where to start from
11:19AM 0 debugging broken/distorted sound with SIP
11:11AM 1 Connection to SIP Gateway
11:02AM 0 Asterisk T.38 framing
10:47AM 1 Bounty: Request for PRI Debug
10:34AM 2 499 Error on X-lite / asterisk setup
10:29AM 1 Line Noise HELP!
10:09AM 1 IPswitch Monitor Extension
9:56AM 0 DID via SIP/IAX
9:53AM 0 Roadmap for Asterisk??
9:47AM 1 Play Sound File Without Answer Channel ??
9:46AM 1 TDM400p reliability????
9:38AM 2 Low cost box for hosting Asterisk and at least one TDM400p
9:31AM 0 Need to Reduce Latency
9:30AM 0 Rebooting Asterisk box shows Asterisk failing to shutdown
8:34AM 1 Is it possible to PickupChan / Dial Pickup / Steal a call that has not been answered?
8:27AM 3 Getting CVS HEAD
8:25AM 2 timed Loop
8:22AM 1 Why 's' doesn't take over unknown extension in context ?
8:16AM 2 Manipulate Asterisk Database from manager?
8:13AM 0 Intercom with Aastra 480e?
7:54AM 4 (no subject)
7:52AM 2 Problem with X101P
7:49AM 1 wcfxo problem
7:34AM 1 Sangoma A101 + Rhino channelbank
7:15AM 1 Interface bonding + asterisk
6:31AM 1 Shared call appearances
5:50AM 0 IPS version 0.79 released
5:17AM 2 SIP Attended/Supervised transfer & features.conf
5:04AM 0 TDM02B on 2 a/b ports of a PBX not working... help
3:45AM 0 Username containing an "@"
3:31AM 0 Direct Broadband connection of ip phone to LiveVoip?
2:28AM 0 call forwarding and parking
2:15AM 3 CDR and TDS
2:12AM 2 Aculab
2:06AM 1 UK CallerID patch with 1.0.7 / 1-0 CVS
1:30AM 3 Can I exit from asterisk console without stopping asterisk?
1:26AM 1 Supply ringing noise to IAX callers
1:19AM 0 voicetronix dtmf
1:18AM 0 Snom 'virtual' extension monitoring?
12:53AM 2 IAX calls between asterisk boxes works 1 way only
12:38AM 0 Conferance DialPlan
12:23AM 1 TDM400P power supply
12:21AM 3 Setgroup & Checkgroup
Sunday April 10 2005
11:09PM 2 VAD/DTX implementation through zaptel cards
10:40PM 1 SIP outbound call audio quality change
9:55PM 1 From OH323 to SIP or OH323 without gatekeeper
9:05PM 1 Cannot open chan_zap:
8:32PM 1 Callback application
8:13PM 0 problem with astrisk on MFC R2
8:08PM 1 problem with unicall
8:01PM 0 Broadvoice problem: Bad request!
7:28PM 0 restructuring my dialplan
6:36PM 3 VM answer call after 20 sec...
5:31PM 3 no ring on inbound SIP calls
5:28PM 3 SIP outgoing problem
5:05PM 1 PTSN POTS Differences
4:50PM 1 unexpected crash ......
3:12PM 2 snom360 & hint priority
2:34PM 2 Problems trying to compile H323 from CVS-STABLE
2:24PM 0 Need /etc/zaptel.conf for TE110P
2:21PM 1 International callback strategies
2:01PM 1 Fax detect/transfer problem?
12:50PM 0 binding virtual IP address
11:47AM 2 sipura 3000 - "Call Leg/Transaction Does Not Exist" - only happens sometimes
11:25AM 8 Sipura SPA-841 Phone Review
11:24AM 0 Yet another version of IPS Freeware
11:01AM 5 Multiple Servers and 1 Central Voicemail
10:28AM 0 problem with unicall and asterisk
9:52AM 0 SMS suddenly not sending out
9:17AM 2 S100I - competitive price?
8:19AM 0 How To conferance
7:49AM 1 UK PSTN Calling From OH323 Problem
7:08AM 0 decimal arithmetic operations in Asterisk
7:06AM 0 Why do calls go silent after 10 minutes
6:43AM 3 search the mailing list
5:30AM 1 ignorepat changing the sound of dialtone
5:29AM 2 Asterisk::LCR - Least Cost Routing for Asterisk
5:22AM 2 Asterisk becomes after one month unstabled
5:15AM 2 Snom only one way audio
5:11AM 0 CVS compile issue on res_odbc.o
4:33AM 0 problem
3:51AM 0 IPSwitchBoard Version 0.77 Released
2:57AM 2 Cann't get CallerID on Zap channel, Please Help!!
1:50AM 0 Fax, which one do I need?
1:47AM 3 Can you comment on this Qos script? How does one shape RTP?
1:44AM 0 Any free SIP softphone with IM capacity for Windows?
1:42AM 1 How to upgrade safe?
12:54AM 0 question about oh323
Saturday April 9 2005
11:19PM 1 Using zap channels for fax
8:13PM 1 Multiple Servers and One Central Voicemail
7:18PM 0 Confusion re; 407 Proxy Authentication Required
6:56PM 2 Terrible crackling on analogue line and X100Pcard
6:39PM 0 Stanaphone - eureka
4:20PM 0 Using manager interface to play aanouncments in aMeetMe
1:07PM 1 Syntax error near unexpected token 'fi'
12:46PM 1 OT: ManxPower 2005 European Tour
12:05PM 1 AgentLogin to MeetMe conference?
12:04PM 3 CallerID name lookup AGI script
11:57AM 2 FWD no longer doing IAX?
11:04AM 6 DS3000P - 20 E1 capacity on single card
11:01AM 2 Asterisk Dual Servers
10:57AM 1 SPA and NAT traversal
10:24AM 1 Asterisk as protocol conventer beetwen SIP and H.323
8:50AM 2 Dialing With Backgound Music
8:28AM 1 How to change language using manager interface?
6:56AM 4 Terrible crackling on analogue line and X100P card
5:58AM 0 Shorewall settings?
5:51AM 0 dyndns alias clients: needs to register in iax.conf as well?
5:43AM 1 Call rejected by XXX: No authority found
3:47AM 0 HOW TO SET THE TIME TO DIAL AFTER astcc-accountnum and astcc-phonenum
2:59AM 2 Hardware dimesioning issues
1:40AM 1 fax pass through on te410p
12:48AM 1 sip phone extensions at a remote site
12:40AM 0 Astcc Patch
12:06AM 2 g726 > gsm not working with sipura
Friday April 8 2005
11:17PM 0 zap to sip caller id "forwarding"
10:11PM 0 SIP Softphone for testing with Asterisk
8:27PM 1 Using manager interface to play aanouncmentsin aMeetMe
7:49PM 0 Frame slip on Wildcard TDM400P?
7:42PM 0 Phone card implementation issues in IAX
7:40PM 3 "s" extension doesn't work with ata
5:30PM 3 How many FXS/FXO ports do I need?
5:19PM 1 Running a Marco from the dial command
5:01PM 6 Asterisk Memory Requirements
4:45PM 2 Asterisk and RT (Request Tracker) setup?
4:20PM 0 debugging voice quality issues
3:33PM 4 UK ISDN with Asterisk
2:16PM 2 Convertnig from Norstar to * to save money
1:20PM 3 Warning, flexible rate not heavily tested!
12:22PM 2 codec translation hints
12:14PM 2 RE:: Connecting asterisk to existing PBX - newbie
11:49AM 1 Connecting Asterisk to a SIP Gateway
11:43AM 0 Allison sounds in native format
11:20AM 1 Dell suggestions for Quad T1 system
11:07AM 1 SIP peer doesn't report busy properly
10:57AM 0 call forwardin and parking
10:23AM 0 Asterisk - SER - sharing single mySQL db and tables
9:48AM 2 Cannot access voicemail
9:45AM 2 Asterisk based CallAccounting software- 1strelease
9:38AM 3 Long wait for ring
9:38AM 1 asterisk missing dtmf what would cause that
9:35AM 2 - low bandwidth codecs
9:17AM 0 Echo on analog line related to FXO card?
9:16AM 0 Asterisk 1.0.5-BRIstuffed-0.2.0-RC7e
9:12AM 1 PCI-PRI cards - what to buy????
8:57AM 2 SNOM 190: Unknown SIP command 'PUBLISH'
8:35AM 4 Channel bank replacement
8:23AM 1 Several INVITE messages sent by Asterisk
8:12AM 0 Test settings
8:01AM 3 PRI card and TDM400P in same box
7:20AM 0 Asterisk@Home .8 SPA-2000
7:13AM 2 AMP 1.10.007 problem on cdr_mysql_table.sql
7:06AM 5 Any opinions on quality/service of Teliax?
7:04AM 2 inquire about connected channel (show channels)
7:03AM 2 snom and "hint" priority
6:49AM 2 X100P doesn't check for dialtone
5:44AM 0 Can a SIP Phone talk directly with anoyher SIP phone (ext a to Ext b)
5:11AM 0 linejack and iax2 !
5:06AM 2 Call from publicIP to PrivateIP
4:59AM 1 Difference Between NAT=yes and QUALIFY=yes and STUN...
4:32AM 0 NVFaxEmail
3:10AM 1 oh323 DTMF bug
3:08AM 2 Asterisk and CAS
2:55AM 0 compiling oh323 Undefined symbol in res_features & Others
2:17AM 1 Fw: Registration Problem with Firefly Softphone
1:24AM 0 G723 call through GW
1:12AM 2 Delayed dial under Asterisk ?
12:57AM 11 Asterisk based Call Accounting software - 1st release
12:44AM 1 Undefined symbol in res_features & Others
12:21AM 1 external access to voicemail?
Thursday April 7 2005
11:44PM 2 iax / realtime problems
11:39PM 2 "404 User Not Found" when calling between two X-Lites
11:03PM 1 Re: Livevoip IAX DTMF troubles
9:32PM 2 Off Topic - Employment Opportunity - PERL, Melbourne, AU.
9:26PM 1 Looking for feedback on IAX2 Phones from Netweb
8:42PM 11 Asterisk Google Group?
8:36PM 1 Canreinvite issue
8:31PM 1 Asterisk quit abnormally
8:08PM 1 how to pass G723.1
7:11PM 0 RE: Asterisk-Users Digest, Vol 9, Issue 67
7:10PM 0 RE: Asterisk-Users Digest, Vol 9, Issue 67
6:59PM 2 failover outbound dialplan
6:45PM 2 stand alone Voice Mail
6:43PM 0 Melbourne Asterisk Consultants
6:38PM 5 Answering without ringing from PRI
6:11PM 1 Asterisk Max TNT
6:10PM 1 Livevoip responds to DTMF via IAX issue
4:50PM 0 Can somebody with HEAD please test MOH on agent calls? M3976
4:24PM 6 Getting a good deal on a PRI
4:19PM 2 Using manager interface to play aanouncmentsin a MeetMe
3:55PM 2 Local Number Ports
3:54PM 0 Low volume in recorded messages
3:37PM 0 Out of Office AutoReply: fedora 3
3:22PM 1 zaptel.conf digium and quadBri together (e1 and isdn together)
3:20PM 2 How to turn off automatic pick up for Incoming calls A@H v0.6
3:17PM 3 Using manager interface to play aanouncments in a MeetMe
3:04PM 3 x100p disconnect on "D" tone
3:02PM 0 Voice controlled calling? Pt 2
2:30PM 0 Conferancing with different interface
2:09PM 1 FW: Out of Office AutoReply: Voice controlled calling?
1:38PM 1 "Mic-To-Speaker-loop" on ZAP lines???
1:34PM 7 Voice controlled calling?
1:02PM 1 TE405P vs TE410P
12:57PM 2 SIP UA behind NAT and REINVITE ???
12:19PM 4 Database lookups?
12:18PM 0 Remembering State
11:29AM 2 Zap (analog line) and volume
11:28AM 1 Asterisk Vs. Cisco, et. al.?
11:01AM 0 IPSwitchBoard Version 0.76 released
10:54AM 5 oh323 compilation
10:52AM 0 How Can I make 2 instances of the nufone H323 channel run?
10:15AM 2 Lag in meetme
9:41AM 2 Can asterisk get code for cmd Authenticate from Database
9:02AM 1 AES vs AEC
8:39AM 0 CDR mysql userfield column truncated at 239 characters
8:26AM 0 open source Asterisk Application of the year ?
8:18AM 0 patch to add distinctive ringing to queues
8:15AM 3 PRI Advice...
8:07AM 7 unlimited iax termination
7:45AM 0 Voicemail localization
7:43AM 2 Fax to email problem
7:38AM 1 SetCdrUserField
7:37AM 0 X100p, IRQ and Noises
7:35AM 2 IVR - newbie question
7:20AM 5 My Sangoma Experience - Review
7:09AM 2 open source Asterisk Application of the year?
7:08AM 1 SpanDSP HELP
7:06AM 3 [OT]: Wiki Etiquette
6:39AM 1 mangle + to 00
6:29AM 0 Cisco ATA 186: Only incoming - no outgoing call
6:01AM 2 Micronet 128K TA Card
5:55AM 0 call behind NAT
5:53AM 1 IAX2 trunk frames documentation.
5:08AM 1 GSM Hardware Setup
4:37AM 1 voip phone reviews
4:15AM 0 Voice mail through Call Manager
4:13AM 1 "404 User Not Found" when calling between two SIP UA's
4:12AM 0 slightly-ot: Where to buy sip phones in Massachusetts
4:01AM 5 T.38 fax with SIP devices
3:41AM 1 keeping dynamic queue members over restart?
2:54AM 0 Help: Problem with X101P
2:50AM 1 [again] Sangoma PRI vs TE410?
2:44AM 0 Integration of Alcatel 4400 with Asterisk
2:19AM 3 Linux & Asterisk
1:54AM 0 How Can i phone traditional PSTN Phone using TDM11B
1:42AM 0 Fax - Capi
1:19AM 0 Manually adjusting volume in an IAX channel
1:19AM 1 about mpg123
1:15AM 3 capi segfault when incoming call is answered
12:43AM 2 Access Voicemail From Outside
12:43AM 5 Call Interception
12:30AM 1 Acccess Voice Mail From Outside Line
12:28AM 2 Measure the Signal of Zap
12:21AM 3 Digium TDM400 Failover on Power Loss
12:13AM 0 Cant Hear Any Sounds
12:03AM 2 Latest CVS chokes Sipura SPA-841
Wednesday April 6 2005
11:57PM 0 X100P call keeps ringing and ringing
10:21PM 3 Curry and Asterisk
9:49PM 1 Configuring the Sipura for static IP and registering with Asterisk.
9:39PM 3 Help using wav files for IVR
9:37PM 2 Beeps during Sip to Sip phone calls
9:18PM 0 Any gsm -> g7231 codec translator?
9:01PM 1 rout call from ser to asterisk
8:40PM 2 MWI for SER and Asterisk - ast_data vs "realtime"
8:18PM 0 Problems with new Asterisk@Home install and Broadvoice, no incoming calls
7:59PM 3 Account Codes with SIP
7:42PM 1 Direct Channel Answering
7:04PM 3 Receiving calls from and to H323 devices
6:12PM 4 Asterisk on Slack 10.0
5:30PM 0 Asterisk and broadsoft
4:12PM 1 DIDs in 510, 408, 916 415 area code
4:06PM 2 Realtime UPDATE
4:03PM 0 Cisco 7940 SIP and No compatible codecs!
3:36PM 1 Context overlap?
3:30PM 1 How to avoid that certain calls come into the voicemail (e.g. wakeup calls)?
2:07PM 1 AMP & Handset Provisioning
1:33PM 1 "Choppy" sounds after transferring to ISDN client or after a time
1:27PM 3 Cisco 7940 Outgoing Audio
1:23PM 12 Liveviop problem
1:17PM 2 SIP messages truncated to 256 characters
12:47PM 5 Asterisk .call files
12:44PM 0 Asterisk, ACD, Queues and Call Transfer Issue
12:26PM 0 Ingate Firewall and Asterisk Integration
12:23PM 0 Problems using Asterisk 1.0.3 with Vocal 1.4
12:07PM 4 SRV Bounty
11:55AM 2 Connecting asterisk to existing PBX - newbie question
11:49AM 4 SIP - SIP Problems
11:40AM 2 Web interface for realtime Mysql friends/peer
11:12AM 0 NOTICE: chan_sip:7654 handle_request: Registeration failed HELP !!
10:41AM 0 I want to call another pc with TDM11B Card
10:34AM 6 Keypad disabled on AriaVoice SIP phone
10:28AM 2 Script Perl Autodialer
10:02AM 0 Fritz Card ISDN in UK - Unable to
10:00AM 2 Any success with BRI in the US?
9:43AM 0 Cant Hear Any Sound
9:24AM 0 Call gets cut off after 5 minute
9:05AM 2 ser <-> asterisk configs anyone?
8:47AM 0 V-9970 Paging Setup
8:33AM 1 dial out and "all circuits are busy"
8:23AM 0 looking for some draft (sip - iax2 mapping)
8:00AM 6 Asterisk and phone system
7:45AM 1 Latest Bristuff crashes on modprobe -r qozap ?
6:55AM 1 Multiple BroadVoice Accounts Problem with Incoming calls
6:41AM 0 SMS with VOIP phone WIP 5000 from hitachi
6:37AM 3 Cisco 7960 forgets VLAN setting
6:33AM 2 IPTABLES Firewall
6:29AM 1 Syntax checker for Asterisk config files
6:21AM 6 how can i connect a cost display on asterisk
6:00AM 0 Fritz Card ISDN in UK - Unable to dial.
5:35AM 2 Snom 190 + NAT
5:34AM 3 Zaptel Compile on a virtual dedicated host.
5:01AM 0 IPSwitchBoard - Now in Spanish
4:38AM 1 Problem compiling 2nd AVM Fritz
3:40AM 1 Got S-frame while link down
3:30AM 1 asterisk is giving error- unable to write audio data
3:26AM 1 wcte11xp works only after cold reboot
3:08AM 0 Asterisk & Windows Messenger 5: Which is the correct/preferred DTMFmode setting?
2:58AM 3 Fritz Card ISDN in UK - Unable to dial. 0x3301/0x3302 errors
2:38AM 3 fedora 3
2:28AM 0 (no subject)
2:05AM 0 FXO-FXS parameters
1:59AM 0 X-Lite codec related...
1:31AM 1 How can I add entry for a UA into asterisk when asterisk is running?
12:45AM 1 IAX2 and NATs that increment ports
12:25AM 2 How can I make base calls with X-Lite via Asterisk?
12:10AM 0 Fritz Card ISDN in UK - Unable to dial. 0x3301/0x3302 errors.
12:09AM 4 Voicemail and SJphone
Tuesday April 5 2005
11:53PM 2 SCCP
10:33PM 1 query about cdr configuration
10:29PM 0 Asterisk @ Home 0.8 Question
10:05PM 1 Help with simple callback application from newbie
9:05PM 6 About Audio Latency from PSTN to SIP
7:54PM 3 Grandstream HandyTone-488, * -> FXO problems
7:25PM 0 Re: Asterisk-Users Digest, Vol 9, Issue 44
7:21PM 0 .conf to realtime conversion script?
7:19PM 1 Stopping Retransmission Found: 102 Error with Polycom IP300
6:37PM 0 RE: Digium ISDN card
5:50PM 0 Asterisk and Nortel Meridian DTMF
5:17PM 1 Automatic Start
4:40PM 3 Petition for IAX firmware
4:03PM 1 Cell phone friendly MOH
2:52PM 1 Atcom AT-320 multiple lines?
2:44PM 4 TE405P and Dell Poweredge 6450 Incompatible?
2:24PM 1 new user TDM400P and T1 card problems
1:59PM 2 Multiple CDR Locations
1:09PM 5 Dialogic D/300SC-1E1 and D/600SC-2E1 with *
1:04PM 0 contracting
12:39PM 2 Should PRI running over t100p be able to survive short yellow alarms?
12:30PM 0 Polycom IP500 does not show elapsed call time on LCD.
12:24PM 3 AGI call problem
12:19PM 0 chan_iax2 stops listening to packets
11:28AM 2 Queue works, but the caller hears silence instead of ring tone
11:24AM 1 PortaSIP/PortaBilling incompatibility (provider:
11:22AM 0 Accessing Conferencing Bridges
10:46AM 2 Sound quality with Xten Xlite softphones...
10:37AM 1 AAH 0.6 to 0.8 Upgrade
10:24AM 5 multiple PBXs on one server.
10:24AM 9 asterisk sounds
10:19AM 0 voicemail access
10:06AM 0 Concurrent calls: best provider?
10:02AM 0 Question about sending Caller-ID Name over PRI NI2 bug 3554
10:00AM 0 Command Reference
9:52AM 0 Digtial Receptionist Recorded Greeting LocationProblem
9:44AM 3 using asterisk as a gateway for residential IP telephony clients
9:42AM 2 realtime management for sip with mysql
9:42AM 3 Cisco 7940/60 failed to take SIP image from tftp server
9:24AM 0 Digtial Receptionist Recorded Greeting Location Problem
9:13AM 2 sip <-> oh323 / real-time / g729 - one way audio
9:04AM 6 TE110P/Hipath3750 - Yellow Alarm
8:55AM 1 Agents
8:18AM 0 Awaiting Ack
8:09AM 5 How do I retrieve voice mail in Asterisk
8:01AM 1 VOIP 911 Mandatory in Canada
7:33AM 1 Best Performance
7:23AM 1 mysql and confs
7:07AM 0 SIP -> PSTN: mISDN DTMF tones generation
7:06AM 0 re: Problem: Compiling error for SpanDSP
6:53AM 2 Asterisk and C# (Dotnet)
6:53AM 1 OT: CRTC mandates 911/E911 for VoIP in Canada
6:50AM 1 prevent callerid spoofing between asterisks
6:32AM 1 VoIP network configuration using Asterisk and SIP
6:11AM 0 Tracing dial plan branching
5:40AM 2 D Channel Becoming "CORRUPTED"?
5:36AM 1 Asterisk and HylaFAX integration
5:17AM 0 Problems registering Asterisk with Vocal
5:05AM 0 Checking for the in-use conference
4:53AM 2 fault tolerant asterisk system
4:43AM 2 MOH and OptiPoint400 std SIP
4:31AM 1 how to make asterisk only for SIP and direct RTP
3:52AM 4 busy line status on CISCO 7940/7960
3:35AM 8 WRT54GP2A-AT
3:21AM 0 i3micro VTA111 and Asterisk
3:03AM 0 Realtime queues?
3:00AM 1 SIP / PTT over Cellular
2:37AM 0 No Voice to POTS.
2:33AM 1 "Multiplexing" (or what ever the term is) FXO ports into a "Trunk"
2:08AM 1 zaptel not starting issues
1:58AM 0 asterisk FWD intelligent routing
1:57AM 0 Asterisk Realtime - extensions configurationhelp / solved
1:56AM 0 sniffing bridged video call on zap channels
1:54AM 0 E100p zapata errors
1:37AM 1 Incomming Call issues
12:51AM 1 SIP Phone binary
12:41AM 2 Outgoing calls on PRI
12:01AM 0
Monday April 4 2005
11:51PM 2 livevoip callerid
10:46PM 2 Outgoing faxes with chan_capi?
10:34PM 1 help regading outbound calls
10:31PM 2 Asterisk Discussion Forums provided by Digium
10:04PM 5 asterisk on UML
8:45PM 2 Weird Errors with Realtime and MySQL
7:51PM 0 Compatability with AudioCodes MP-108
7:26PM 3 Voicemailbox detection:
6:18PM 3 Transient SIP Registration Issues
6:05PM 3 Detecting Downed SIP Phone
5:17PM 0 Digium Hires Kevin Flemming
4:49PM 4 AAH 0.6 - Change Network Gateway
4:45PM 4 Set system time over the phone
4:36PM 0 SIP/SDP packaged in Multipart/Mixed mime type
4:33PM 1 ASTCC - not saving configuration
4:12PM 0 zapata.conf parameter order - feature or bug?
3:40PM 5 Channel bank question
3:32PM 1 SIP and firewall
3:20PM 1 chan_sccp compile error
2:58PM 3 TE405P takes ~5mins to load.
2:43PM 1 Authentication with DB Support
2:40PM 0 DTMF Caller ID in Brazil
2:37PM 2 Operators guide
2:23PM 0 L2 QoS switch
1:49PM 0 CVS-HEAD compile? Was: Checkgroup and transfers
1:42PM 0 IP Address of caller variable?
1:25PM 0 Asterisk queue crash
1:19PM 0 SIP microphone not working after ZAP and sounddriver installation
1:05PM 1 SIPTone II and PoE
1:04PM 2 call redirection from outside line?
12:49PM 3 Livevoip DTMF via IAX almost
12:25PM 0 distinctive ringing in a queue?
11:34AM 2 ZAP problem (No channel type registered for 'Zap')
11:34AM 3 Asterisk on WRT54GS
11:19AM 2 compilation of asterisk
11:03AM 0 vmail.cgi - can't forward messages
10:59AM 5 monmp3thread: Request to schedule in the past?!?!
10:46AM 1 mISDN + chan_misdn and DTMF
10:31AM 0 How to control codecs when originating a call?
10:23AM 0 SIP phones to Asterisk using MAC addressinsteadofIP address
10:21AM 0 IPSwitchBoard speaks other languages
9:59AM 2 Ring Twice
9:57AM 1 Can't see ANI2 (aka info digits) from PRI t1
9:51AM 2 Can I set queue not to hangup?
9:51AM 5 Asterisk and clarent
9:50AM 3 rookie getting started question
9:42AM 2 FGD Support
9:00AM 1 Asterisk Realtime - extensions configurationhelp
8:44AM 0 Asterisk and Ingate registration
8:19AM 1 Problem registering 'SJPhone'?
8:18AM 0 Does the agent queue app support Aftercall and AUX agent status?
8:06AM 1 Just a test
7:56AM 2 newbie - want to use asterisk as an internal PBX
7:46AM 1 configuring md5 authentication
7:33AM 1 SIP phones to Asterisk using MAC address insteadof IP address
7:21AM 1 Asterisk Realtime - extensions configuration help
7:15AM 0 Distributed services such as voicemail using Asterisk
7:08AM 0 SIP phones to Asterisk using MAC address instead of IP address
6:58AM 1 Browser based configuration of Asterisk
6:47AM 1 IAXy audio troubles (only on INCOMING calls)
6:43AM 1 X-Lite to Zap, no Voice on other phone!
6:37AM 0 PRI: received SETUP message for call that is not a new call, wicked!
5:54AM 0 Re: ASTCC question: Trunk LOCAL
5:43AM 2 Best way for nated sip peers thru a database
5:06AM 2 Sending faxes and call accounting
4:27AM 1 Zap - What is going on?
3:51AM 0 How do you do Line Hunting in Asterisk?
3:23AM 1 Supervised transfer problems
2:56AM 4 Realtime & voicemail
2:54AM 1 Difficulty in configuring Asterisk to ensure that Call Transfers (SIP phone) are properly recorded and billable
2:41AM 0 Planet VIP 450
2:22AM 2 Asterisk+Sipgate - just one step away..
12:49AM 1 Zaptel group members - dial out on a availible port via trial and error?
12:38AM 1 error while compiling asterisk-1.0.7
12:33AM 1 How to send email from the dial plan?
Sunday April 3 2005
11:55PM 0 Wellgate 3701
11:49PM 1 Previous sip reload not yet done
10:50PM 4 how to configure groups using a sip phone
10:48PM 4 V92 modem with asterisk
10:31PM 2 Music On Hold and ATA-186 w/Silence Supression
9:52PM 6 Asterisk@Home Question
9:19PM 1 AGI Dial Plan
8:41PM 0 creating conference call
8:12PM 1 Asterisk <-> Altigen
8:03PM 0 Re: Asterisk-Users Digest, Vol 9, Issue 21
7:51PM 0 Joshua Chessman
6:24PM 2 Asterisk Realtime Capabilities
5:37PM 2 AS5300+SIP+ASTERISK or AS5300+MGCP
5:10PM 3 problems with call-forward from ccme to * on sip trunk
4:50PM 0 VG248 and Asterisk
2:04PM 0 Asterisk on Suse minimal installation based on Suse Rescue - what to add to be bootable on HD partition ?
1:21PM 0 Looking for res_config_pgsql
12:56PM 3 Detecting when a called mobile is not reachable?
12:33PM 3 Authenticating username
10:20AM 2 SET & CHECK group
10:00AM 1 SIP dialing in two extensions
9:32AM 0 Asterisk with Jasomi Peerpoing
7:14AM 0 Re: Asterisk Discussion Forum
4:26AM 0 IPSwitchboard Version 0.73 Released
3:33AM 5 Router with QoS recommendations
3:06AM 0 Re: Asterisk Discussion Forum
3:03AM 1 Where to post my impovements to ASTCC?
Saturday April 2 2005
11:50PM 2 New to asterisk.
11:47PM 4 How does asterisk know the did called on?
11:46PM 0 configuration problem
10:41PM 1 Fwd: NDN: Re: Delaying answer of incoming calls
10:00PM 0 Replace Adtran 608 With Asterisk
9:37PM 2 Sip registration Problems With Zyxel P2000W
9:11PM 1 Macro Extension with Realtime and Mysql DB
9:09PM 1 How to reset IAXy?
9:00PM 0 TE410P and Fax Server
8:59PM 0 Problem with asterisk -> ohphone
8:42PM 3 Asterisk Discussion Form
8:07PM 1 D-Link router/Voip gateway locked to Lingo?
7:16PM 1 Packetization
6:38PM 0 Long Distance Acces Code
5:30PM 2 Re: Are there online forums instead of, this email
5:18PM 6 Buying some Polycom IP300s
5:03PM 4 xlite regestration fails but calls to thru
4:02PM 0 Sipura - GSM or iLBC?
3:36PM 1 OH323 core dump
12:48PM 3 Zaptel Anti-MMX Optimizations
12:46PM 1 Book Review: VoIP Telephony with Asterisk
11:44AM 3 how to tell what ${DIALSTATUS} is being set
11:20AM 3 Asterisk Auto-Startup on Ubuntu/Debian
11:20AM 2 problem detecting answer on pri card
11:07AM 0 Outbound calls with xlite and Xpro PocketPC
10:38AM 2 Passing varibles *out* of macros
10:26AM 1 SjPhone&H323
9:33AM 1 Dynamic Zap/{channel} allocation for out going possible?
9:24AM 7 Starting with Asterisk-SIP
8:37AM 7 Asterisk Voice mail with CCM
8:14AM 1 Two accounts at one provider and a 302 redirect problem
5:27AM 4 astcc problems
5:14AM 1 {extensions.conf} Dialing plans with queues....
4:37AM 1 Registration to multiple GKs
4:19AM 1 H.323 call '.....' cleared, reason 8 (Transport failure)
4:01AM 3 :: Strange way of receiving calls ::
3:42AM 1 Delaying answer of incoming calls
3:35AM 0 (no subject)
3:24AM 0 Version 0.72 of IPSwitchBoard Released
3:23AM 1 VAD In Asterisk with Zaptel
3:01AM 1 Little question
2:10AM 3 Shorewall firewall rules
Friday April 1 2005
11:30PM 1 at-320 phone configuration difficulty
9:41PM 7 Queues
7:07PM 0 Display agent logged into a queue on the phone
6:35PM 1 OT(?) Your subject line
5:31PM 1 Random outbound:
4:46PM 0 Strange DTA problem
4:02PM 1 Preserving CallerID when forwarding to cellphone
3:45PM 3 Call bridging
3:10PM 1 Datafire 2977
2:42PM 4 Squeaking / chirping on ZAP Digium TDM400P
2:32PM 3 What's the use of a multi line phone?
1:25PM 1 Codec not negotiating
1:01PM 0 Make voicemail use Maildir...
12:13PM 3 Issues with ringing on FXS ports
10:47AM 0 Faxing through Broadvoice - HT286
10:07AM 1 "Unable to create channel of type Zap" Message
9:52AM 0 Must be configuring something wrong spanDSP rxfax
9:38AM 1 ADSI Input from 480 keypad?
9:27AM 1 Sending DTMF back in a dialed/answered channel before bridging a call
9:24AM 1 MF Trunk Signaling
8:54AM 2 Does asterisk@home support Dual-Processor installations?
8:48AM 1 Specify Codec In Outbount Calls?
8:42AM 1 Voicemail Email Bouncing
8:31AM 3 Snom and Multiple calls
8:25AM 1 blind transfer question
8:25AM 0 Zyxel Prestige 2002 (ATA)
8:23AM 0 new release of chan_misdn !
7:36AM 1 Q.931 to SIGTRAN interface
7:29AM 2 Looping messages
7:12AM 1 Dial'ing multiple SIP devices impossible when forward activated
6:45AM 2 Maybe an echo cancellation problem?
6:12AM 0 Optimizing speex (was Re: Erratic CPU load )
5:59AM 1 LDAP and Asterisk
5:52AM 0 [OT] Announcing
3:57AM 2 queue.conf config
3:40AM 3 Eicon Diva Server BRI Setup
3:28AM 0 [Fwd: Problem with dial out via chan_capi]
3:02AM 1 H.323 call '...' cleared, reason 15 (Call ended due to security checks)
2:26AM 3 Problem with dial out via chan_capi
2:07AM 4 using unixODBC
1:59AM 0 Problems getting FXO channel working - Unable to create channel of type 'Zap' (cause 0)
1:52AM 5 really small box
1:07AM 0 Playback starts before call answer
12:53AM 1 register => with realtime
12:40AM 21 *** Asterisk 2.0 Stable release out now
12:32AM 1 Parking no
12:01AM 1 User Regerstation, allowing non-registered users on *