asterisk users - Nov 2005

Wednesday November 30 2005
11:03PM 2 1.2.0 PRI dropping calls occasionally...
10:34PM 1 two sip phone communication using asterisk server
7:51PM 1 Call transfer with voicemail password
6:49PM 1 format
6:38PM 5 Asterisk cluster and astdb
5:32PM 1 sip to sip, not comunication
4:41PM 1 polycom backlight?
3:22PM 1 FW: CDR issues
3:17PM 1 Queues and Servicelevel
2:48PM 0 CDR issues
2:15PM 3 How to exit from Asterisk console.
1:46PM 1 Disposition failed in Asterisk-1.2.0-stable
12:38PM 1 problem with zaptel 1.2.0 and pulse dialing
12:38PM 1 Queue calls...
11:25AM 3 Snom 320s and the hint priority
10:45AM 5 hierarchical VoIP system
10:34AM 1 pbx or asterisk?
10:26AM 2 Sipura SPA-3000 & SPA-2002 - Unable to dial *99
10:26AM 0 asterisk starting problem. Warning 2224 (
9:25AM 2 Disable IAX2 native bridging / Monitor() app
9:08AM 0 Debian Sarge + Asterisk 1.2 + chan_mISDN not starting
8:46AM 0 Got SIP response 400 "Invalid Subscription-State"
8:43AM 3 IAX Service providers in Australia for unlimited inbound
7:57AM 0 Transfer call error
7:44AM 0 Astfax problem
7:29AM 1 Astfax with current CVS
7:07AM 2 MeetMe with the V (video) option
4:15AM 1 Tone busy in zaptel
3:38AM 1 Compiling Asterisk 1.2 from Source on Debian Sarge - Problems
3:33AM 1 Help transfer call
3:11AM 0 BRIStuff and PRI
2:25AM 1 IP GSM Gateway is giving uncomplete SIP signalization to PRI interface - can I somehow avoid that in Asterisk ?
1:31AM 1 TE210P & Linux SMP
12:16AM 0 Page() application examples.
Tuesday November 29 2005
11:49PM 0 werror compiling libpri
9:20PM 0 ASTCC not completed
8:40PM 2 Active SIP Peer?
5:53PM 2 zapata directory not found in svn .
5:48PM 1 Problem with IAX2 jitterbuffer and DTMF reception with 1.2.0
4:55PM 8 Static on inside end of conversation
4:41PM 3 Pasting phrases together....
4:02PM 0 How to disable SIP Options methods on asterisk
3:59PM 0 Comas versus pipe command in AgetCallBackLogin
3:51PM 0 All agent calls going to powered down agent extension?
2:30PM 2 Cisco CP-7940G drops time from display
2:01PM 0 Digital Cellsocket
1:49PM 0 RE: Asterisk-Users Digest, Vol 16, Issue 232
1:41PM 1 route call based on codec? (g723 gets message, g729 goes to conf connection)
1:25PM 6 Voicemail and sendmail
1:17PM 0 Question on Monitoring and Transferring...
11:57AM 1 Queuelog
10:54AM 0 cause 17 - User busy ?
10:36AM 1 Voicepulse not accepting new customers. (FCC E911)
10:25AM 0 ResetCDR with CDR
10:12AM 0 Monitoring Zaptel Errors
9:15AM 3 Caller ID Block (*67)
8:48AM 0 moh on optipoint400
7:56AM 1 qozap.o error
7:25AM 0 FW: Fax sending problems
4:58AM 1 Problems with auto dialout
3:59AM 2 TDM400 revisions problem: Rev J not working!!
3:59AM 1 Load spikes with 1.0.10
3:41AM 5 asterisk@home isdn
3:38AM 0 setting variables in a .call file - how?
2:38AM 0 Hangup after 18 sec on PRI channel
2:24AM 0 SNOM Phones MWI, Hold & Retrieve buttons notworking with Asterisk v1.2
1:38AM 0 Problem with Ext calling
Monday November 28 2005
11:14PM 1 VegaStream
9:27PM 1 Digitmap problems
7:17PM 2 delayed pickup in ZAP interface and issue with hang up-s (fwd)
7:00PM 1 SIP Trunk in incoming ? it's possible ?
6:52PM 0 Avaya 4620SW - SIP response 400
5:59PM 1 Channels not showing up in Asterisk
5:59PM 1 Is a BUG ? Hints and incominglimit
5:51PM 0 tranfered calls audible but low volume
5:44PM 2 SIP rapid INVITE re-transmission: bug, or config problem?
4:37PM 0 Philippines asterisk mailing list / yahoo groups! (PINOY AKO! PINOY TAYO!)
4:24PM 3 Newbie question on 1.2 extension configs
4:01PM 7 US e911 reminder
3:30PM 2 Accepting Inbound SIP Connections
3:19PM 1 small office setup
2:53PM 1 Comedian Voicemail? PROBLEMS?
2:38PM 0 Interface Cards that support QSIG
2:32PM 8 SNOM and 1.0.9
2:24PM 1 misdn, busy detection
2:10PM 0 Avaya 4620SW Invalid Subscription-State - Issue
1:26PM 1 DTMF errors
1:24PM 2 cdr_manager.conf
1:12PM 3 Wrong usage of in the extension?
1:06PM 1 Help connecting Avaya S8700 and Asterisk through H.323 trunk
12:41PM 0 AGI + CDR
12:08PM 1 Problem with pulses dialing on asterisk 1.2
12:00PM 0 Asterisk project converts to Subversion version control system
11:49AM 1 Problem with Internet connection
11:48AM 1 SNOM Phones MWI, Hold & Retrieve buttons not working with Asterisk v1.2
11:42AM 1 PROGRESS with cause code 31 received
11:25AM 1 Emailed voicemail messages not being deleted
10:15AM 0 Call progress from sip gsm gateway to pri interface - doesn't get through
9:55AM 0 how to stop ringing while talking
9:47AM 1 IAX jitterbuffer and trunking settings between 1.0.9 and 1.2
9:27AM 0 How does DTMF get sent over a PRI in Asterisk
9:19AM 0 Trunk SIP howto ?
8:46AM 0 Can 'spandsp' ack as an intermediary between a fax machine and a TDM400P?
8:39AM 1 Download Ringtones for 7960's?
8:22AM 1 AGI script always returning 0
8:20AM 0 Realtime Extensions Problem
7:30AM 0 New mailing list: AstCallCenters
7:15AM 0 Problem forwarding zap to sip
6:35AM 0 troubles with voicemail
5:53AM 2 Upgrade Cisco 7910 with Asterisk ?
5:31AM 0 A rather big setup.
5:00AM 2 Legacy PBX integration problem
3:28AM 3 Problem with ADIT 600 and FXO configuration
12:42AM 11 SIP tapi
12:16AM 1 CDR Accounting PRoblem
Sunday November 27 2005
11:25PM 0 PBX Manager beta1 release
9:05PM 0 "CPE does not support Call Waiting Caller*ID"?
6:50PM 2 Does it mean I was blocked by STUN?
6:30PM 0 Intel G729 Codec Install error on asterisk@Home2.0
6:28PM 2 New Asterisk user - Dumb Questions
5:59PM 0 chan_bluetooth background scanner
5:36PM 0 Script to update externip for A@H/AMP [was Re: SIP Extension behind NAT, Asterisk on NAT (DMZ)]
4:40PM 8 Zaptel errors on Debian
3:13PM 1 Asterisk 1.2 and Athlon64 platforms
12:56PM 1 Asterisk cdr mysql
12:40PM 0 Failover with 1.2 Dial applications
11:34AM 0 chan_bluetooth with Plantronics Heaset (some good stuff)
11:11AM 0 ANNOUNCEMENT: Asterisk-Java 0.2 released
9:50AM 1 A question about transfering calls
9:02AM 3 Reboot stops TD400P cards from outgoing calls until first incoming call arrives
8:17AM 2 Voicepulse Open Access status?
6:43AM 1 IAx/g729 client for MAC
6:17AM 0 zaptel 1.2.0 and correct settings in zapata.conf for Germany
5:26AM 0 trunk not registering -newbie
5:02AM 0 calling to mgcp device
4:57AM 7 Dialplan help
4:56AM 1 Intel G729 Codec Install error on asterisk@Home 2.0
12:21AM 2 a2billing / php agi debugging
Saturday November 26 2005
10:20PM 0 1.2 page command
10:06PM 0 Possible Bug in Asterisk 1.2.0 with Queues and MOH
7:53PM 4 OH323 channel in asterisk 1.2 ...Ouch ... error while writing audio data!
6:14PM 1 Trouble with Channels
5:46PM 8 Would DECT cordless phones work with Asterisk and VOIP?
4:09PM 0 Voice recognition ...
3:32PM 0 cid_rewrite and update
3:25PM 0 Plantronics Bluetooth Headset help..
2:00PM 1 Anyone using Parlay VoXip SIP Gateway with Asterisk ?
1:43PM 3 IAXmodem fax polling
1:25PM 3 Context mix-up
12:25PM 0 Re: SIP Extension behind NAT, Asterisk on NAT (DMZ)
11:49AM 0 H standard extension
11:26AM 1 WARNING[19541]: res_features.c:844 builtin_atxfer: Did not read data.
10:48AM 4 Small office with all employee's offsite
10:26AM 1 chan_bluetooth max simultaneous channels
8:56AM 2 cdr enhancement with 'rate' column
8:11AM 4 problems with callback.agi script
7:05AM 1 callback.agi script
7:02AM 1 Asterisk dial plan
6:19AM 1 Camping-on-busy
6:10AM 1 Asterisk and Cisco Phone 7910
2:10AM 0 optiPoint 410/420 SIP firmware for Asterisk
12:52AM 1 SIP Forward
Friday November 25 2005
11:10PM 3 configure intel modems.....
7:44PM 2 Problem about outgoing calls with verizon.
7:20PM 0 Music On Hold Crashing?
6:36PM 0 RE: [Asterisk-biz] GSM Gateway for £60
5:59PM 0 Polycomm 500 not saving web changes any more
5:55PM 1 Transfering to a bridged call
4:35PM 3 global numbering plans
4:13PM 1 can I have T1 and E1 on the same TE406 card?
2:53PM 2 Distinctive Ring Detection not working
1:57PM 2 misdn, 2x HFC cards
1:45PM 2 Polycom IP50X Park Softkey
1:25PM 1 Asterisk 1.2 stability problem.
1:21PM 5 Sangoma problems!?
1:06PM 2 Asterisk callback system
12:48PM 0 Asterisk and Siemens HiPath 3750 issues
12:36PM 2 Narrowing RTP port range
12:31PM 3 Truncated CDR records
11:41AM 0 Re: think people dont help that easily
11:16AM 1 Bristuff: qozap.o error
11:08AM 1 CallerID not passing through to Polycom 500 (SOLVED, sort of)
11:01AM 0 A2Billing questions are off topic for this list
10:48AM 0 smsq sending 7 at a time ?
10:24AM 1 Distinctive ring?
10:17AM 2 "Local Directory" feature on Polycom Soundpo int 501s
9:55AM 1 "Local Directory" feature on Polycom Soundpoint 501s
9:47AM 1 Dialplan pattern match discrepancy
9:44AM 2 Help with 2billing please.
9:23AM 1 Problem with SIP register
9:02AM 0 Manager log
8:58AM 0 is it possible to force faxdetect / disable echo cancellation for a given extension?
8:39AM 1 speex & ilbc
8:21AM 3 How to initiate a call from a web page?
8:19AM 0 Second TDM22B board install issue
8:07AM 0 SIP response 484 "Address Incomplete" incorrectly handled
7:46AM 0 authentication question
7:40AM 3 Philippines Asterisk users, anyone?
7:03AM 0 busy channels
6:59AM 4 Siemens OptiPoint 4xx
6:22AM 3 Looking for Info on Asterisk scripting
5:50AM 1 authentication fails to provider after upgrading to 1.2.0
5:33AM 2 Command line
3:22AM 3 sound problem, please help!
2:36AM 1 TE411P
2:29AM 1 Really lightweight itemised billing
2:08AM 0 help need for the configuration
1:51AM 0 has someone zaphfc with xenomai working?
1:18AM 0 Re: Bad quality...
Thursday November 24 2005
10:53PM 20 NewBie to Ast Server, help need for the configuration
10:50PM 1 Asterisk and Japanese Caller ID
9:02PM 0 Recommended PCI latency time?
7:31PM 0 Asterisk + SER problem,ua cann't hangup
5:45PM 4 Pros and Cons of T1/E1 cards
5:10PM 1 harry's project
5:04PM 1 Preventing long-distance call forwarding
4:29PM 2 chan_misdn crashes : init_stack: success but entitylist not empty
3:33PM 1 Bad quality
2:41PM 1 Newbie requesting help!
2:00PM 1 (AMUG) Asterisk Montreal User Group today's meeting
1:48PM 5 Linksys SPA-841 Disconnects from Asterisk
11:58AM 0 SIP softphone with subscription/hint support?
11:35AM 1 Send fax using PRI connection to TE405P
10:17AM 0 Re: Asterisk not picking up calls.
10:05AM 1 Re: Queue Callback - SOLVED
9:48AM 0 Voicemail notifications alwats sent as asterisk@hostname
9:28AM 0 H323 to H323 calls problem
9:18AM 2 CallerID not passing through to Polycom 500
7:55AM 1 jittering with Iax2 and Meetme on Asterisk 1.2.0
7:45AM 0 Don't Outgoing call with Zap
6:20AM 2 Lag in speech
6:13AM 0 GUI and Asterisk Realtime
6:06AM 1 HELP! on disconnecting stale calls.
5:14AM 0 Compatibilidade com PABX Intelbras
4:35AM 0 Sip dosenot fall to default 's' , STRANGE?
4:01AM 1 Fax sending problems
3:28AM 3 [Asterrik-Users] Bristuff for Asterisk 1.2 error
3:26AM 4 PRI problems again - What should I do ?
3:01AM 0 1.2.0 using 1G of RAM
2:21AM 0 hint problem
1:58AM 0 pstn-destination beeing cut in logs and cdrs
1:08AM 1 Calling Asterisk PABX in "anonymous" mode...
12:21AM 7 Looking for Windows based Asterisk
12:08AM 0 Voicemail email format, please help!
12:01AM 1 zaptel 1.2.0 on (Tettnang)
Wednesday November 23 2005
11:59PM 0 MTP Requirements for getting * talking to CCM for Voicemail
11:06PM 2 Loss of Registration for SIP Trunks
10:05PM 1 Asterisk as Softswitch
8:29PM 0 Codec negotiation (not the same old stuff)
7:45PM 1 Looking for Windows based Asterisk Management Client
6:41PM 2 Satellite, SIP Invite 488 Codec Rejection, SIP Timing Issue??
6:20PM 1 ASTCC - card in use
6:06PM 0 1.2.0 voicemail: unable to create lock file?
5:05PM 1 QSig and MD110
3:36PM 1 Asterisk DNS SRV lookups
3:11PM 1 Invite with Replaces
2:33PM 6 Asterisk + WiFi Phones
2:10PM 2 Modem Connections to PPP Server
2:07PM 1 Not receiving fax
1:45PM 5 [Asterisk-Dev] hello
1:28PM 1 PhoneCALL version 2.7-RC1 Released!
12:55PM 0 Cisco FXO hangup detection
12:41PM 0 Warning LSP Low
12:23PM 2 manager interface behavior
12:23PM 0 How to connect a Cisco Router with FXO module to Asterisk
11:58AM 1 astman make error
11:48AM 0 Call transfer with phones that cannot handle more than one line
11:11AM 0 Using transfer button in SJPhone
8:16AM 2 Querry about the modem
7:32AM 0 agent transfer problem
7:17AM 4 Asterisk SIP architecture question
6:45AM 0 How to make Broadvox work with Asterisk 1.2.0
6:20AM 0 [patch] sqlite3 support for asterisk 1.2.0
6:12AM 0 Re: [Users] open letter (2)
5:43AM 0 7960 audio quality when calling remote asterisk box
5:42AM 0 Calling lines
5:41AM 1 ISDN cards using CAPI interface
5:16AM 1 presence and Asterisk crash
4:11AM 0 queue problem
4:08AM 4 Aastra 1.3 firmware
3:25AM 0 Re: [Users] open letter (2)
2:46AM 3 Asterisk and DrayTek Vigor2600VGi
2:42AM 0 how to configure analog phone
2:34AM 5 open letter (2)
2:19AM 0 RE: [Serusers] Re: open letter
2:13AM 1 Asterisk server behind NAT, and SIP clinet behind another NAT.
Tuesday November 22 2005
11:13PM 7 Help need to reset Adit 600 for Asterisk install
9:32PM 1 (New SIP phone, and more)
9:24PM 1 NVFaxDetect and NVBackgroundDetect on Asterisk 1.2
9:17PM 0 Is there a way to see what agents are in a group from CLI
9:16PM 1 Strategy=ringall does not ring all agents.
9:14PM 0 Sipura SPA-841 Disconnects from Asterisk
8:41PM 6 ver1.2 installation problem
8:38PM 2 Clearwire and Asterisk
8:09PM 0 Idle time between agent ringing too long
5:34PM 1 Dial ZAP with group (g2) erroneously says call answered when it is still ringing
4:49PM 1 Bristuff for Asterisk 1.2
4:09PM 0 SetGroup & GROUP_COUNT advise appreciated
3:23PM 3 Does Voipjet uses IAX2 trunking
3:15PM 2 Asterisk 1.2 Aastra/Sayson 480i DTMF Problem
2:56PM 4 Call parking on Polycom IP501
2:47PM 3 high CPU usage when using -c
2:42PM 3 TDM400 FXO port 1 only problem.
2:33PM 0 Possible SIP/NAT Problem with 1.2
2:23PM 1 sip URL peering
2:18PM 0 AstLinux VMware images now available (use with the free player)
1:23PM 1 Asterisk 1.2 + Debian Sarge
1:20PM 3 Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!...
1:06PM 2 Test numbers for ENUM (,, etc.)
12:18PM 2 Re: [Asterisk-biz]
12:09PM 2 Not able to call with phonzo
11:00AM 0 Re: [Users] open letter
10:15AM 0 DUNDi
10:09AM 1 Unable to register Zyxel WIFI Phone as SIP Clientto Asterisk
10:01AM 5 1.6.3 Polycom Firmware?
9:45AM 4 Virtual Modems Revisited
9:28AM 1 installing Asterisk from source
9:12AM 4 I need suggestions for on equipment
9:02AM 2 Re: [Asterisk-biz] VoIPJet Support Contact -We have US unrestricted termination for .095
8:47AM 2 Best Communications Line for VoIP
8:40AM 3 Which is Better!
8:25AM 2 ATA verse Wildcard TDM400P
8:22AM 2 Master Telephone
8:20AM 4 Server Side AgentCallbackLogin
8:14AM 4 Slightly OT - Anyone know of an external ringer compatible with Cisco phones
8:14AM 0 Re: [Serusers] open letter
8:12AM 0 REPOST:How do you get a sound to play to caller on answer?
8:10AM 0 Unwated outgoing Zap channel briding
7:57AM 2 Digital Assitant Help
7:38AM 5 Bad Lines - What can the phone company do?
7:32AM 0 SIP debugging tools - Suggestions experience?
7:23AM 7 open letter
7:12AM 0 setting caller ID with Voicepulse
7:04AM 2 Asterisk 1.0.10
6:16AM 1 Call waiting issue
5:42AM 0 sip routing
3:17AM 0 PAP2 and double ringback tone
3:00AM 1 HELP - ! No D-channels available!
2:35AM 1 channel_find_locked
2:33AM 0 outbound sip proxy
12:36AM 0 standard extension with forwarding
Monday November 21 2005
11:19PM 1 Using Long Distance Operators
10:38PM 2 problem with registration of SIP phone
10:28PM 4 SIP Extension behind NAT, Asterisk on a public domain
9:32PM 1 priority jumping
9:18PM 2 equivalent to SetvarIf ?
8:46PM 1 ETel
8:11PM 0 odd transfer behavior
6:09PM 1 Setting up FXO in router
6:02PM 1 Rejected connect attempt
5:27PM 0 inquiry into Asterisk scripting scenario (VXML, AGI, IVR, etc)
5:22PM 2 chan_capi_cm-0.6.1: ISDN1: too much voice to send for NCCI=0x10101
5:08PM 0 PAP2-NA 3.1.3
3:58PM 1 Problems with fax failing when bridged across TDM400Pvers E
3:50PM 2 Detect alternate line in Broadvoice inbound context
3:44PM 1 Zyxel P2000Wv2 cannot do agent login, SJPhone work just fine?
3:06PM 2 recall button using tdm400 Australia
2:49PM 1 Asterisk 1.2.0 AddOn's compile error with MySQL 5.0.15
1:45PM 0 hint for MGCP (devicestate): bug 5515
1:40PM 0 H.323 and video
1:34PM 2 v1-2 install mkdep loop
1:22PM 0 Motherboard Selection Assistance
12:59PM 1 legacy pbx
12:54PM 0 allpage.agi
12:49PM 0 asterisk 1.2 unable to request echo training on channel
12:31PM 1 Help on x101p disconnect when called party answers
12:23PM 0 Include files in AEL
12:10PM 0 MOH: Most Efficient Method
11:46AM 0 mISDN + Fedora + asterisk 1.2
11:43AM 0 AGIphp Installation
11:36AM 1 can't receive calls with CAPI ISDN
11:00AM 1 Select multiple columns from MYSQL cmd...
10:43AM 1 Asterisk and embedded system
10:35AM 1 How to deal with echo in MeetMe?
10:10AM 1 How do you disable realtime?
10:08AM 3 Linksys SPA941
9:45AM 4 Anyone parked in your Asterisk?
8:17AM 1 Asterisk versions after the 1.2 release
8:12AM 2 AMP installation
7:58AM 1 Please Help with Zaptel
7:48AM 4 addmailbox script
7:30AM 0 split line authorization problem (ATL IP400 phone)
7:17AM 1 MySQL - Realtime install procedure?
7:00AM 1 zyxel p2000w
6:56AM 0 New firmware for Aastra/Sayson IP phones
6:56AM 1 h323 question
6:51AM 1 Asterisk crash: "using deprecated BYE/Also transfer method"
6:18AM 0 v1.2 and features.conf
6:17AM 0 HT486 and RFC2833
6:13AM 2 AstLinux 0.2.9 Released
5:57AM 0 Problem with Broadvoice
5:48AM 0 User identification
5:47AM 1 Problem with SIP channels
5:34AM 2 Can not build zaptel with kernel-2.6.12
3:45AM 0 How do you get a sound to play to caller on answer?
3:38AM 0 how to configure the LCS with Asterisk--->>Anyone, please?????
3:02AM 0 Problem with multiplier
2:40AM 3 Asterisk to Fax Server
2:34AM 4 E1 Gateway
2:27AM 0 zaptel compilation help!
1:54AM 1 Death at 2am
12:56AM 0 RTP question
12:24AM 6 Realtime Problems
12:06AM 0 CallProgress breaks DTMF - RFC2833
Sunday November 20 2005
11:55PM 0 General Bandwidth Voice Gateway
7:52PM 1 aastra 480i config files
6:47PM 0 Monitor() creating choppy audio files
4:17PM 0 Re: Call Leg/Transaction problem
1:51PM 1 Database update after hangup
12:42PM 2 Asterisk MySQL CDR - MySQL starting too late
12:04PM 1 DNID on IAX2 trunks?
10:02AM 1 Weird 1.2 stable problem
9:16AM 4 International Dialing Code
7:36AM 1 stopped sounds
5:01AM 0 CallProgress breaks DTMF
4:06AM 2 Dutch callerid and x100p
2:50AM 0 Illegal instruction on starting asterisk (was Newbie question)
2:07AM 1 SIP response 481, SIP client
1:02AM 0 AMP, huge number of Zs
Saturday November 19 2005
10:24PM 6 Can Asterisk Set CallerID on Broadvoice?
10:14PM 0 Abdul Lateef Khan wants to talk to you using Google Talk
8:20PM 2 TDM400p card problem
6:44PM 0 mfcr2 and 1.2
5:57PM 1 Free 411 Service
4:55PM 1 AMP partially not working, Apache dying on segfaults?
4:41PM 2 ztcfg segfault
3:16PM 3 asterisk.conf question
2:42PM 2 cmd dial timeout don't work in asterisk
2:13PM 1 VoIP connection US --> EU with ADSL a problem ?
2:01PM 1 Allowing Called user to accept call before transfer
12:50PM 1 simple setup
12:38PM 0 call parking and realtime_ext
11:44AM 1 Clipcomm CG-410 and caller-id from PSTN
11:42AM 0 i3micro VTA-111 to Ast 1.2
10:35AM 1 Asterisk 1.2 compile error - Any suggestions would be appreciated
10:08AM 0 AstBill Live CD with Asterisk 1.2 Released
10:02AM 1 Wildcard FXO takes too long to answer incoming calls
9:50AM 1 Audio in MeetMe Conferences Garbled After Upgrade to 1.2
8:44AM 1 MOH during M() macro execution
8:28AM 2 customized softphones
8:09AM 2 Dial() and j option: What is correct?
7:05AM 3 chan_bluetooth and Ericcson T68 problem
6:37AM 7 OT: Where to buy a T1 crossover cable for * and channel bank
4:25AM 1 ztdummy problem on SUSE 9.3
3:38AM 1 cmd dial timeout don't work in asterisk 1.2 ?
2:51AM 3 return Credit Time
12:41AM 1 meetme + sendtext
Friday November 18 2005
11:07PM 3 [Fwd: call status with FXO]
10:17PM 1 Retrieve multiple variables from database using MYSQL cmd
7:58PM 1 What's the streamplayer util and how to use it?
6:20PM 1 IAX Webphone to Dial a Support Extension Only
6:09PM 0 Page command in 1.2
6:04PM 0 RE: RE: Asterisk-Users Digest, Vol 16, Issue 151
5:51PM 0 SOLVED: Polycom MW beep
5:41PM 0 How do I test my Asterisk IAXmodem Hylafax HylaClient ?
4:43PM 0 secondary host= in iax.conf
4:39PM 3 GotoIf always goes to true?
4:12PM 1 Zaptel Error
3:46PM 1 Asterisk Compilation Error
3:37PM 5 Forward Voicemail to remote server?
3:35PM 0 Channels that won't die
3:27PM 2 Modifications to Voicemail
3:07PM 2 RE: Asterisk-Users Digest, Vol 16, Issue 151
2:27PM 0 No Caller Name Displayed SIP->SIP
2:20PM 2 IAX and Firewall
1:31PM 0 Asterisk app_ices problem
12:56PM 1 WARNING[2757]: Failed to write frame
12:51PM 1 Getting invalid extension during agent login.
12:45PM 0 AMP 1.10.010 released
12:09PM 0 Voicemail ODBC storage and realtime
12:04PM 2 mISDN and chan_isdn for 1.2
11:46AM 0 which g729 codec to use ?
11:29AM 1 Asterisk feature codes???
11:00AM 1 auto assigning SIP port
10:44AM 1 A2billing warnings with new Asterisk 1.2
10:41AM 0 AEL and n+101 apps
10:38AM 0 Cisco IP phone NAT config
10:33AM 5 VOIPJET - are they down
10:00AM 1 Context restrictions for long distance access, examples not clear?
9:59AM 1 R2 variations by country
9:52AM 5 Provisioning server
9:45AM 1 call transfer and pick chan_h323
9:27AM 2 FAX difference IAXModem / Hylafax and spandsp app_rxfax
9:22AM 0 Specirfic IP to specific context sip.conf
9:11AM 6 Asterisk 1.2 error: "Ouch ... error while writing audio data: : Broken pipe"
8:39AM 1 Cisco phones port range
7:54AM 3 OT: Softphone with Bluetooth support for *
7:51AM 0 Asterisk 1.2 and music-on-hold question
7:40AM 1 phone intergration
7:32AM 3 Asterisk 1.2 - Windows Messenger ?
7:24AM 1 Remove older version of Asterisk
7:02AM 1 Examples of LIMIT_CONNECT_FILE and other LIMIT_XX Options
6:42AM 1 'ztmonitor' stopped working after using 'fxotune'
6:32AM 4 Sipura SPA-841 Second Line Help
6:19AM 0 Re: Asterisk en france
5:46AM 2 Problems with Read() in outgoing calls
5:32AM 1 In France asterisk never detect hang up. Why ?
5:18AM 2 wcfxo loads correclty after issuing twice the command "ztcfg -vvvv" !!
4:31AM 4 Contact field in SIP HF between asterisk + ser
2:46AM 1 Streaming mp3's when dialing a particular extension.
2:17AM 2 Problem switching from external ISDN-2 to PBX ISDN-2
2:14AM 1 Re: Re: SIP - Loop detected (Matt Riddell) (Matt Riddell)
2:12AM 5 Newbie question. (Long)
2:10AM 1 gpx-2000 early dial support
1:28AM 0 re: problem with asterisk and SIP on same box with 1.2
1:09AM 10 create my own soft Phone
12:25AM 0 Subject: Eicon Diva Server query
Thursday November 17 2005
11:34PM 0 SPA 3000 and MWI
11:14PM 0 Asterisk voicemail responses - feature?
10:22PM 1 1.2 under OS X?
9:31PM 7 Eicon Diva Server query
9:24PM 1 SIP INVITE IP address variable?
8:06PM 1 how to originate a call and capture it's DIALSTATUS
7:32PM 2 Help with shell script for externnotify
5:56PM 2 Hung Zap channels
5:28PM 0 Missing smp kernel package in Asterisk 1.2installation...
5:17PM 1 What's the best way to stream and/or convert MP3 and WAV files?
5:13PM 3 multi tenant with queues
5:08PM 2 Missing smp kernel package in Asterisk 1.2 installation...
5:00PM 0 Asterisk 1.2.0 and memory usage
4:11PM 2 SER & Asterisk combination to get around NAT
3:53PM 1 realtime callerid
3:39PM 1 no longer loading all config files?!?!?!?!?!!!!!!...
3:09PM 1 1.2 won't compile: res_config_odbc.c
3:07PM 0 Overlapping sounds in asterisk and asterisk-sounds
2:52PM 2 call levels
2:43PM 0 Cisco SIP translation-rule Question
2:23PM 2 VoIP Gateway Providers
2:13PM 2 HFC ISDN card and mISDN driver
1:57PM 2 CVS v1-2-0 make problems?
1:09PM 1 Asterisk 1.2 Change in: agi_channel
12:46PM 2 Sound Choppy
12:19PM 0 sip.conf settings for / broadvox?
12:19PM 0 64bit libs in /usr/lib
12:01PM 4 Bristuff / Junghanns / Customer Service
11:49AM 2 Poor sounds on Adtran 750
11:19AM 0 (AMUG) Asterisk Montreal User Group: Dedicated Mailinglist + Next Meeting
10:02AM 0 How to specify multiple agent groups with a queue?
9:35AM 2 Asterisk + FXO + FAX SIP
9:07AM 6 Wireless SIP Phones with Asterisk
9:03AM 0 Sipura doesn't get caller id and hangup with Siemens Combiset
8:47AM 0 Suggestions for tunning SJphone with Asteris k?
8:37AM 5 Mission-Critical Deployments
8:29AM 0 Bridgind and decoding.
8:17AM 0 D Channels reseting every 30 seconds
7:48AM 0 CallerID Length
7:22AM 1 PSGW 2.2 Skype gateway?
7:14AM 0 Suggestions for tunning SJphone with Asterisk?
6:42AM 1 Dazed and Confused
6:20AM 1 Hangup detection - TDM400P
6:13AM 0 IAX softphone's sporadic performance - Keep Alive Issue?
5:50AM 8 GSM Gateway / Terminal for sale
5:46AM 5 chan_bluetooth
5:15AM 4 /spool/outgoing delays
5:05AM 3 1.2 chan_modem not installing?
4:59AM 2 Register redirect
4:53AM 0 Voicemail email format
4:47AM 1 SIP Channel and jitter buffer
4:44AM 6 suggestions for hard phones?
4:26AM 1 chan_capi fails when Asterisk doesn't start under root user
3:23AM 1 Hardware HDLC in Zaptel - Bug ID 5313
2:21AM 5 stop asterisk when Idle
2:20AM 1 RE: Re: SIP - Loop detected (Matt Riddell)
2:10AM 1 AGI Dial command return status
1:24AM 5 New asterisk management tool
1:20AM 4 is there any free pocket pc softphone??
12:19AM 2 SIP - Loop detected
Wednesday November 16 2005
11:37PM 0 upgrade from 1.0.9
10:55PM 0 Hangup problem with other EPBX
10:49PM 14 Asterisk 1.2 Released!
8:04PM 5 ip phone
8:02PM 1 Asterisk drops call when calling other VOIP
7:07PM 4 IAX offline Voicemail
7:04PM 3 hold problem w/ GXP-2000 1.01.12
4:56PM 1 Weird behavior on incoming calls
4:54PM 2 app_icd anyone? on 1.2?
4:25PM 3 receive fax with asterisk
4:08PM 0 1.2rc2 build problems
4:07PM 1 List of Motherboards or Servers that are testedok with Asterisk and Digium boards
3:58PM 1 TDM04b on FreeBSD
2:04PM 0 asterisk and cisco ubr900 configs using h.323.
1:24PM 1 Contention Detection with Zap ??
1:17PM 1 Aastra 9133i registration errors
12:38PM 0 (no subject)
12:33PM 0 AT&T Merlin Communications System 6102 Cartridge Music on Hold and Paging
11:51AM 1 SER authenitification failure on ASTERISK
11:38AM 2 Outgoing sound very low
11:21AM 0 Read() application behavior change: bug or feature?
9:55AM 0 high availibilty (heartbeats) - a good way to
9:38AM 1 zapata.conf for T1 PRI
9:00AM 0 Cisco Security Advisory: Fixed SNMP Communities and Open UDP Port in Cisco 7920 Wireless IP Phone
8:26AM 0 SpanDSP and broken faxes (cut short pages)
7:48AM 1 List of Motherboards or Servers that are tested ok with Asterisk and Digium boards
7:45AM 0 bluetooth headset with softphone or directasterisk
6:55AM 0 NOTICE: ast_unregister_indication_country
6:37AM 0 Asterisk and Inter-tel
5:48AM 0 misdn for BRI
5:42AM 0 is the 'Zaptel Under the Hood' down?
5:41AM 0 Asterisk T.38 question
5:39AM 1 Is the '' down?
5:30AM 1 calling to asterisk and listening to music (GSM)
5:28AM 0 A-Z carrier Registration
3:55AM 0 Heads up - AVM C2/C4 on AMD 64 bit processors
2:16AM 0 Queue Autologoff over trunks
1:37AM 0 A simple network environment: a configuration issue or a bug in Asterisk?
1:26AM 1 Problem with octo bri
1:18AM 3 Recording voice messages in mp3 format
1:06AM 4 Asterisk @ Home password recovery
1:04AM 2 Compile problems, 1.2 rc2 and SUSE 9.3
12:56AM 0 Asterisk pop-up
12:56AM 0 Price info in SIP packet?
Tuesday November 15 2005
11:57PM 2 Dialing out with FXO
10:28PM 3 Agent not ready
10:28PM 1 Changing 5060 port
10:14PM 0 asterisk can't load chan_misdn (FC4)
8:15PM 0 Has anyone bought anything from Asteriskmall? yourexpirence?
8:13PM 1 PRI HDLC abort on dchan
8:11PM 0 Cisco Call Manager and H323 trunk correction(MTP)
7:00PM 1 Directory Command - Odd Hangup When Pressing "*"
6:58PM 2 Queue Monitoring..
6:40PM 0 SDT Message Signal
6:36PM 0 Mention VoiceMail2 in UPGRADE.txt?
5:33PM 1 Cisco Call Manager and H323 trunk correction (MTP)
5:06PM 1 not work DTMF
4:50PM 5 reply to today's posting
4:34PM 3 A2billing questions
3:19PM 0 res_musiconhold.c: Music on Hold class 'default' already exists
3:05PM 7 g729 status in New Zealand
2:58PM 1 Anyone got zaphfc running 2 cards with NT and TE simultaneously?
2:19PM 0 1.2rc2: Problem with channel bank, Ring/Off- hook in strange state 6
2:00PM 3 bluetooth headset with softphone or direct asterisk
1:46PM 0 "Call/Transaction Does Not Exist" back from
1:40PM 0 mtp-2
1:20PM 0 1.2rc2: Problem with channel bank, Ring/Off-hook in strange state 6
1:01PM 1 Automon / wW options ?
1:01PM 2 Max number of Digium cards a server can support?
12:59PM 1 Infinitum bloquenado SIP ???? / Is Infinitum
11:44AM 0 FXO module picks up but incoming callers only hear the ringing tone.
11:24AM 0 Has anyone bought anything from Asteriskmall? your expirence?
11:17AM 1 Using variables for context names
11:09AM 0 Problem with Zap/1 picking up on OUTBOUND calls from analog extensions
11:08AM 5 Incoming call trunk fwd not work
11:08AM 1 Can't start * with
11:04AM 5 g.729 pass thru mode
11:00AM 2 Problem with call drops
10:51AM 0 FW: Asterisk 1.0/1.2 on cobalt Raq2-4]
10:36AM 0 backup routing for IAX outbound
9:40AM 0 Asterisk peer authentification
9:19AM 12 Editing Asterisk config files with WORD Pad
9:15AM 0 Illegal redirection
9:13AM 1 Infinitum bloquenado SIP ???? / Is Infinitum blocling SIP ????
9:09AM 2 unexpected debug output from console
8:51AM 0 Play a message at the begining of a call
8:44AM 0 Problem: Can't make outgoing call
8:38AM 2 help and guidance needed from gurus
8:34AM 1 FW: Asterisk 1.0/1.2 on cobalt Raq2-4
8:26AM 1 Restore Asterisk log files after deleting...
7:33AM 4 Message waiting notification
7:26AM 1 speech to text for only digits
7:22AM 1 Possible bug in agent monitoring
7:19AM 4 canreinvite=yes
7:18AM 2 Cisco 7960 Multiple Line Appearance
6:39AM 0 Queue Callback
6:34AM 4 Multiple emails
6:15AM 8 Multiple Outbound SIP Trunks
5:38AM 1 Asterisk and Agents
4:41AM 1 Monthly tips for the community?
4:39AM 0 app_conference compiling for asterisk
4:22AM 0 Configuring Asterisk Queues using real time(MySQL)
4:16AM 0 usb cellphone
4:15AM 1 errors with chan_zap.c when installing asterisk-1.2.0-rc2
3:58AM 2 Cisco 7905 sccp Hold and Message buttons
3:49AM 1 Polycom Softkeys & Voicemail Button
3:30AM 1 A2Billing problems. still.
3:02AM 3 remove asterisk?
2:42AM 2 voicemial maxmsg
2:35AM 0 ERROR utils.c:509 tvfix:
2:18AM 3 SIP => H.323 Terminator
1:42AM 1 E1 PRI slips on TE410P
1:08AM 0 Asterisk + Voicetronix Card
1:08AM 0 H323 config question
Monday November 14 2005
9:26PM 6 Asterisk hobby box
7:42PM 0 TDM400 FXS Pulse Dialing Craziness
6:07PM 1 Can someone explain the 's' extension
5:16PM 0 Looking for T38 Solutions
4:37PM 17 "open" asterisk?
3:56PM 4 Using RxFAX and TxFAX together
3:47PM 1 Hardware: Dell/Acer
3:42PM 2 voicemail locking
3:33PM 0 VoIP provider for South America Termination
3:13PM 1 Asterisk 1.0/1.2 on cobalt Raq2-4
2:50PM 2 Problem with Cisco local conference and hangup
2:39PM 2 Mixmonitor
2:05PM 2 Dedicated echo canceller hardware
11:42AM 0 Grandstream - No dialtone in handset after firmware update..
10:31AM 4 Media gateway recommendations?
10:24AM 2 Connecting analog lines to Asterisk for IP telephony device use
9:54AM 1 How do I know if I have CRC-CCITT (README.Linux26)
9:22AM 1 PRI to SIP
9:17AM 0 TDM400 cards and modem/fax devices
9:14AM 4 Fritz card usb v2.1 - Capi installation problem
8:39AM 1 Comments in AEL files?
8:38AM 0 problem to connect h323 temination
8:28AM 1 Problem with 827-4v and asterisk as a pstn GW
8:07AM 0 connect to gateway h323
7:58AM 6 asterisk sample size adjustment
7:53AM 2 Maximum Number of SIP Phones Supported By Asterisk
7:28AM 1 SIP signaling and canreinvite=yes
7:25AM 0 OT: Aastra PT 390 Question.
7:07AM 0 Configure Asterisk to call from softPhone(SIP Channel) to Analog phone(Modem Channel)
6:21AM 3 IAXy echo?
6:06AM 0 Brooktrout MPAC 1200 card with Asterisk
5:57AM 1 How to check how many G729 codec licenseinstalled
4:21AM 2 newbie question regarding asterisk
3:54AM 1 MYSQL issue in UPDATE..
3:00AM 7 Snom clients deregistering
2:50AM 0 Promblem dialin from an internal E1
2:32AM 11 ISDN card required
12:06AM 0 re: a2billing /areski help
Sunday November 13 2005
11:10PM 1 Asterisk realtime extensions context inclusion
10:35PM 0 showing iax incoming calls
10:21PM 0 PSTN Trunk
10:20PM 1 g.729 codec
10:10PM 5 Anybody tried it from India ?.
10:08PM 1 TDM Echo issue
10:06PM 1 Regarding TDM400P
8:17PM 0 load module failed, returning -1
8:16PM 2 Asterisk Installation exits with following error
8:01PM 1 Sipura SPA-2002 Double Ring
7:19PM 1 Zaptel cards on SuSE?
6:19PM 0 Notices at beginning of call
6:11PM 1 iax-qos-openbsd...
5:26PM 0 MONTREAL USER GROUP MEETING, Tuesday 15th, 5pm
5:14PM 0 Advice on Asterisk-based home voicemail+fax+datasystem
5:00PM 2 How to get Referred-By header
4:59PM 2 Advice on Asterisk-based home voicemail+fax+data system
4:18PM 2 How to do "asynchrononous" Dial?
3:43PM 0 DeStar 0.1 released!
3:26PM 1 Asterisk overlap dialing (PRI)
2:30PM 3 X100P troubles?
12:38PM 0 Any experince with Voip Reach
11:31AM 0 Playing Music at the Back Ground while the conversation is on and recording the same in real time
11:12AM 1 fwd - Iax
11:08AM 1 Cron still running after uninstalling asterisk
10:34AM 3 spandsp-0.0.2pre21c broken?
6:37AM 1 "check for res for"
6:36AM 3 Upgrading 1.0.9 to 1.2 beta
6:07AM 3 Format of music for native MoH?
4:31AM 8 How to check how many G729 codec license installed
Saturday November 12 2005
9:44PM 0 codec error connecting to cisco gateway
8:57PM 1 A2billing with Mysql-5.0.15
5:47PM 1 WARNING[3035]: Invalid priority/label ' ' at line 17
5:46PM 0 Example of Pass-Thru Codec
5:08PM 1 NEC NEAX 2400 Integration with Asterisk
4:43PM 0 Warning CONFIG_ZAPATA_DEBUG on 2.6.14
2:32PM 1 Help with this
12:43PM 1 vigortalk and transfers
9:29AM 1 Unable to play dialtone
9:20AM 0 problems compiling spandsp-0.0.2pre21c under 1.2rc2
8:45AM 1 REaltime does not unregister sip peers "on the fly"
8:20AM 3 Does IAX2 Trunk Work between IAX and SIP
8:14AM 1 asterisk@home KDE or GNOME?
7:46AM 1 PRI testing using TE205 and loopback cable?
5:45AM 1 Capi problem
3:25AM 1 How to let caller continue after Dial cmd
3:22AM 1 callcentrum - call any, ring one
12:35AM 1 debian sarge & zaptel 1.2 & TDM400P
Friday November 11 2005
10:34PM 1 Snom 360 Opinions
10:11PM 1 Problems after upgrade...
9:58PM 0 Asterisk 1.2.0-rc2 Released!
9:10PM 0 How to add an Asterisk predictive dialer to a Strata VI phone system
8:52PM 6 Nextone <-> Asterisk <-> DID provider
8:49PM 1 Quantumvoice vs Broadvoice - Multiline
6:41PM 2 Fail over?
4:54PM 0 Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA
4:50PM 0 AAstra 9116 Weirdness.
4:02PM 2 7940 paperweight
3:19PM 0 Problem with CallerIDNum
3:13PM 2 [Announce] Web-MeetMe v1.4.0
3:02PM 1 GoToIf Regular Expression
2:54PM 1 Asterisk behind a NAT
1:54PM 3 2 SIP phones on Y data connector on 1 ethernet
1:31PM 0 IAX2 phones
12:57PM 0 MINNESOTA: TwinCities Asterisk Users Group - Saturday 11/12/2005
12:35PM 2 Non-numerical caller id in Budgetone 101 Ip Phon
11:38AM 1 Setting up IP PBX
11:00AM 3 IAX2 multiple audio frames per UDP packet?
10:23AM 1 asterisk high load high availability servers
9:16AM 0 sip.ld for a SoundStation IP 4000
9:09AM 3 Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 1
9:07AM 2 New Asterisk WEB Interface ( astwebmgr )
9:05AM 0 Comand Read issue (Asterisk rel. 1.0.9)
8:53AM 1 SV: Call p2p
8:48AM 0 missing name part in to field of SIP header
8:18AM 2 GPS data from cell phones
8:04AM 0 Digium TDM400 on freebsd
7:58AM 1 MOH/Media Server
7:55AM 0 command returns a result code of -1 (indicating failure)
7:03AM 0 DISA multiple calls with single dialup
6:02AM 0 Re: Asterisk-Users Digest, Vol 16, Issue 85
4:39AM 2 CAPI problem under gentoo with AVM C2 - asterisk claims CAPI not installed
4:30AM 2 IAX2 calls being droppped
3:48AM 2 sip ignores context definition?
3:11AM 1 Result branching in AEL
1:00AM 2 Voicemail file as MP3
12:37AM 1 A2Billing Postpay
12:25AM 3 TDM400P + FXO module = PSTN woes
12:11AM 2 Softphone with Lotus Notes support?
Thursday November 10 2005
11:24PM 1 one outgoing call == one call per minute
8:44PM 1 iaxy and comfort noise generation / sound quality
8:08PM 1 Asterisk: BUS Error in SPARC/Linux (debian)
7:31PM 0 SIP Registration from Verizon DSL
7:28PM 2 Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 18
7:05PM 0 Disable Comfort Noise on Grandstream Phones?
5:59PM 1 How do I apply the asterisk patches?
4:58PM 0 FW: Re: MAX TNT SIP / Asterisk
4:52PM 0 Trouble completing a call.
4:47PM 2 Siemens optiPoint 420 phone and Asterisk SOS
4:42PM 1 A2billing problem.The system disconnects me immediatelly after asking me the PIN
4:07PM 1 NvFaxDetect , rxfax, Quantumvoice SIP : Dropping incompatible voice frame
4:05PM 1 Errors With Hint
4:04PM 0 features while on hold
3:50PM 1 Needed - Pager notification script
3:38PM 0 RE: 4 HFC cards
3:29PM 0 autoattendant timeout with include statement
3:26PM 2 Digium TDM Revision I Card
3:23PM 1 Digium TDM mothercard version I
3:17PM 0 agents in database
3:02PM 1 muiconhold.conf without restarting asterisk
2:56PM 0 RE: (BAD!!!) Sound quality of the NEW GRANDSTREAM BT 101 and 102 MODEL
2:29PM 1 txfax and rxfax problem with spandsp 0.0.2pre21c and 1.2rc1
2:27PM 0 New astGUIclient/VICIDIAL version released 1.1.8
1:49PM 1 SoundStation IP 4000 App and Cfg files.
1:30PM 2 asterisk 1.0.10?
1:24PM 1 Disa dialplan
1:20PM 1 PAP2-NA and SRV
12:31PM 6 Planet Network - VIP-153
12:28PM 1 Possible problem with Zaptel/Asterisk with 1.2rc1
12:19PM 3 Linksys PAP2: supported codecs
12:02PM 0 MeetMe bugs in 1.2.0rc1?
11:56AM 0 3COM 655005001 SIP phone on asterisk
11:25AM 3 Little OT.. SER Question
11:16AM 0 ast_merge_contexts_and_delete: Requested contexts didn't get merged???
11:12AM 0 Call Transfer Problem with IAX2
11:11AM 0 loaded despite being told not too!
11:03AM 0 Cannot find where error message is comming from...
11:02AM 1 Need help can't figure out what wrong with zapata.conf
10:52AM 1 Ex-girlfriend mode on invalid/no CID?
10:47AM 0 sched.c: Attempted to delete nonexistent schedule entry
10:38AM 13 voicemail to two emails?
10:36AM 1 Asterisk 1.2-rc1 and sip show inuse
10:36AM 1 How do I factory reset a Grandstream BT-102
10:33AM 2 (Some problems sending this menssage) Sound quality of the new BT 101 and 102 models
10:24AM 1 Bug in 1.2rc1
10:02AM 1 looking for keypad free sip phones
9:49AM 1 TDM400 Card
9:35AM 0 TE110P Zaptel config questions
9:34AM 0 Nortel BCM 3.6 and Asterisk 1.0.9 via H.323
9:21AM 1 Clarification on module
9:17AM 0 Sound quality of the new BT 101 and 102 models
8:53AM 0 NAT'd SIP extension, no audio
8:29AM 0 Cell phone as digital trunk line
8:21AM 3 Simple Dial for If Busy Send to Voicemail
8:00AM 0 Phones no longer register - except one?
7:56AM 0 terminal emulation application that uses SIP
7:38AM 0 H323 still no rtp traffic
7:36AM 1 SIP Redirect/Transfer
7:14AM 2 SIP and VPN
6:24AM 2 New revision of my MFC/R2 software available
6:18AM 1 Call p2p
6:04AM 2 Queues with one Agent set to DND
5:50AM 2 ITS Telecom Hardware
4:39AM 3 IM / presence asterisk-1.2-RC1
4:18AM 1 H263 algoritm in 1.2.0.rc1
3:48AM 0 sorry for posting many times
3:14AM 0 Asterisk 1.0.9 + TE210 --- Long
1:37AM 1 SIP NAT register
12:25AM 1 Test environment (Windows Softphone)
12:25AM 0 chan_iax2: ast_sched_runq
12:02AM 2 Can't create iax channel
Wednesday November 9 2005
11:05PM 6 Asterisk Crashing (high load issues)
9:56PM 1 Intel Desktop MotherBoards *NOT* Unsuitable for Digium Boards
8:58PM 1 Changes from 1.2beta2 to 1.2RC-1
7:56PM 2 Zaptel Outbound Caller ID on E1 in UK
7:46PM 4 DTMF detection in TE406P ??
7:12PM 0 Asterisk 1.2.0-RC1 Crashing with g729=?ISO-8859-1?Q? codec ?= =?ISO-8859-1?Q? and ?= =?ISO-8859-1?Q? ATA_?= 186
7:03PM 2 DTMF method AVT
6:27PM 1 MusicOnHold does not play
6:20PM 1 Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 186
4:23PM 3 Error compiling app_rxfax on 1.2-rc1
3:57PM 3 Wits end with echo
3:56PM 0 system command vs mailfax and quotes?
3:18PM 2 what is the role of trunk=yes
3:10PM 0 Getting SpeedDial buttons to work on a Cisco 12 SP+
3:09PM 1 Script for load testing
3:07PM 3 Cisco DHCP and Polycom boot server
2:42PM 3 Problems with HINT
2:13PM 0 RE: Asterisk-Users Digest, Vol 16, Issue 60
12:21PM 1 dial during greeting to access another extension if busy or not available?
11:52AM 1 Asterisk OH-323 module-Inbound Call dropped due to in-call-rate violation (1.55)
11:46AM 1 Areski Can you Help ??? We are stuck
10:45AM 3 Test environment for a Predictive Dialer
10:36AM 0 Automatic testing of my DIDs?
10:31AM 1 Kapanga SoftPhone HOWTO
10:29AM 4 Realtime Voice Changer Patch
10:28AM 2 Cisco 7940 - TFTP
10:19AM 2 CVS HEAD - app_muxmon
10:12AM 3 Zaptel T1 Timing Source
9:52AM 5 Receptionist phones
9:40AM 5 force to expire a sip registration
9:22AM 0 Call forward to cell phone and X100P
8:55AM 0 long calls on same channel
8:21AM 1 Intel Desktop MotherBoards Unsuitable for DigiumBoards
8:10AM 3 problem with g729 and CME-Asterisk
8:01AM 1 [Asterisk-User] Festival help
8:00AM 2 TDM400 FXO Screech
7:58AM 1 Sending DTMF tones after answering on an IAX channel
7:50AM 0 Asterisk 1.0.9 + TE210 + SpanDSP
5:22AM 0 Zaptel: chan_zap.c:6514 mkintf: Unable to open channel 1 : Operation not supported by device
5:05AM 0 SIP/H.323 suggestion
4:49AM 0 New asterisk web gui for small/medium sizedbusinesses
4:06AM 7 dell and digium hardware
3:51AM 2 ast_streamfile failed
3:43AM 0 extension and overlap
3:39AM 0 bysy tone when dialing out via SPA-3000 in the netherlands????
3:23AM 0 queue_log and mysql support
2:53AM 1 PRI pass-through
2:11AM 1 Asterisk Fax support using T.38
1:43AM 2 MeetMe invite another user
12:51AM 1 how to setup Agent dialing in multiple asterisk servers
12:50AM 1 iax2 config sanity check
Tuesday November 8 2005
11:35PM 0 Double Transfers.
11:17PM 0 Iax config areskiCC
10:41PM 0 Privacy Manager Application
10:37PM 4 Intel Desktop MotherBoards Unsuitable for Di gium Boards
10:21PM 0 DTMF problem in * -> ZAP/g1 calls(ZAP/g1 is a PRI group)
10:00PM 2 maximum concurrent conference peers in asterisk
9:43PM 0 OT: Atlas 550 Caller ID interoperability wit h Digium TE110P?
9:38PM 0 callfile to a ring group
9:21PM 2 Avaya 4612 IP phones with Asterisk?
9:17PM 3 sip_message_support.patch
9:15PM 1 Unsuccessful Native Bridge Between Zap Channels
9:14PM 1 SNOM360 & Monitoring Extension States
8:55PM 2 Asterisk 1.2.0-rc1 Released!
7:36PM 0 7970 How-To
6:24PM 1 Zap/TDM400p with old phone.
6:08PM 1 How do I show that a message is waiting on a Zap channel?
5:59PM 1 t38 for asterisk
4:20PM 8 Intel Desktop MotherBoards Unsuitable for Digium Boards
4:17PM 2 sipphone for freebsd
4:03PM 1 strange tone is droping calls
4:03PM 0 Please help me diagnose/fix this echo
2:43PM 1 A2Billing PIN does not get registered - keeps getting prompted
2:18PM 1 1.2b2/mpg123 and memory usage
2:15PM 1 Re: [Openvpn-users] v 1.5 is a web 2.01
1:47PM 1 Fading echo on Zap<->SIP channels...
1:45PM 1 Play message and dial extensions simultaneously
12:48PM 3 Agent Call Recording
12:43PM 1 Help with SER
11:53AM 1 Asterisk Consultant
11:51AM 5 Extension Ring on Multiple Phones
11:39AM 5 ATA-488 FXO
10:18AM 0 FW: New package posted to Sourceforge
9:59AM 4 libbluetooth
8:06AM 0 Lost Cisco SIP phones after reboot
7:59AM 1 how to use #include to all files in /etc/asterisk/customdir ?
7:38AM 2 [Asterisk-User] Estension s don't start
7:37AM 1 OT: Atlas 550 Caller ID interoperability with Di gium TE110P?
7:22AM 6 New package posted to Sourceforge
7:19AM 1 Sipura 2000
7:01AM 1 LCDProc for Asterisk?
6:57AM 2 BRI cards, HFC, and bristuff - a general question to clear up my understanding.
6:51AM 3 Cisco 7960 Password Recovery
6:32AM 3 Playtone on answering the phone
6:12AM 4 groupware + unified messagerie +Asterisk
6:08AM 2 Hiss
5:53AM 3 Sangoma 102 installation problem
5:14AM 1 Problem dialling multiple SIP devices
3:24AM 0 Detect registered peers
3:09AM 3 Sip provider problem or?
3:01AM 3 sangoma a104d install
2:49AM 0 SRTP proxy
2:34AM 4 Sensing fax with txfax
2:27AM 1 CallerID via chan_capi-cm-0.6 possible?
2:24AM 0 Softphone to show the activate sip user and their sip number
2:17AM 1 Which Wildcard?
2:17AM 0 Bristuff 0.2.0-RC8o or 0.2.0-RC8n (* 1.0.9)
1:36AM 4 Fwd: differences between chan_capi and chan_capi-cm
Monday November 7 2005
11:42PM 2 How to configure LineJack
10:13PM 1 ad hoc conferencing-reg
9:51PM 2 How to make write and read formats equal to native format?
8:41PM 3 Choppy Audio in Echo Test and Music On Hold (1.2.0-b2)
7:56PM 0 FW: Error building res_perl
6:08PM 0 several beginner questions
5:44PM 2 how to send fax using Spandsp
4:12PM 3 SIP domain support for authentication and virtual hosting
3:06PM 1 zaphfc not generally compatible with kernels >= 2.6.13
2:51PM 2 libmfcr2 - spandsp.h: present but cannot be compiled
2:40PM 0 new sip domain support and REGISTER requests
2:36PM 3 Problems with DTMF on Polycomm Phones
2:11PM 4 [OTAnn] Feedback
1:53PM 2 Can't make calls from Asterisk IAX to other IAX
12:23PM 1 asterisk 1.2b2 compiling problem
12:18PM 3 Stopping Asterisk from forwarding calls?
11:27AM 8 asterisk-1.2-bêta2 | presence/subscription support in the SIP channel driver
11:23AM 1 Speex codec problems
11:17AM 4 CentOS vs. Vanilla Kernel
11:14AM 0 AGI environment dump callerid
10:20AM 4 Help with dialplan to allow breakout to DISA
9:28AM 2 MP3 or OGG
9:27AM 5 Change Asterisk User
8:46AM 0 Help needed for Onhold calls
8:35AM 1 Re: Asterisk-Users Digest, Vol 16, Issue 44
7:36AM 0 Use of Queues and agents to provide office phone coverage.
6:03AM 1 FXS problems
4:55AM 0 h323 nat externip
4:04AM 3 asterisk as SIP gateway
3:50AM 0 Outgoing and incoming call of LineJack
3:42AM 2 meetme conference getting error using codec g729
3:33AM 1 queues in 1.2-beta2
3:21AM 1 Asterisk Addons linker's error....
3:07AM 0 SJphone "Awaiting ACK" after updating Asterisk to CVS-HEAD of September
2:38AM 6 Dropping last digit from phone number
1:29AM 0 Festival Sound Quality
12:42AM 1 What's the purpose of the "username=" line?
12:00AM 0 how to configure adhoc conference in Asterisk
Sunday November 6 2005
11:41PM 0 Cisco 3640 as * FXO GW using MGCP?
11:11PM 0 meetme conference pbm using g723.1 codec
8:37PM 0 Slightly OT: Firefox search plugin for
8:24PM 0 sangoma a104d
5:56PM 1 Testing with X101P
5:31PM 0 Stuck getting SIP trunk to work with 404 error.
5:30PM 0 Problem with Aterisk 1.2.0 beta 2 and sip dtmf
3:28PM 0 Help with SIP Phones inside a NAT with * inside another NAT
1:57PM 0 ABE - Are you happy with it?
1:41PM 0 Grandstream HandyTone 386 HT386 Distinctive Ring with _ALERT_INFO
12:09PM 1 Problem ringing multiple extensions when one is forwarded
10:54AM 2 escaping to an extension while listening to voicemail message
10:49AM 1 Re-invite don't always work
10:30AM 5 DNS Server Failure wreaks havoc
6:03AM 8 Meetme Conference-reg
3:20AM 1 how to conferencd in Asterisk
1:08AM 1 limiting incloming call on sip phones to 1
Saturday November 5 2005
9:30PM 0 Can't Access Amp
9:22PM 1 TDM400P hangup detection on Bell Canada PSTN
7:35PM 0 Fw: Fw: Inbound Calls on Asterisk from VBuzzer
7:25PM 0 Fw: Inbound Calls on Asterisk from VBuzzer
7:23PM 0 Inbound Calls on Asterisk from VBuzzer
6:48PM 0 General questions about register and nat traversal
5:54PM 1 anyone using for termination?
4:05PM 1 Voipjet - No one is available to answer at this time
2:26PM 0 Realtime IP peer with static IP won't load
2:19PM 0 Registration time
2:18PM 1 PHP error setting up AMP
1:55PM 6 TDM400 FXO vs FXS Interrupt performance
1:37PM 1 "Hand-over" phone connections
12:35PM 1 Asterisk & Lucent TNT w/11.0.2
11:27AM 0 Timing out on Registration
11:11AM 0 How does Nightly Downloads work at
10:36AM 5 sill looking for a provider
9:27AM 2 chan_capi-cm-0.6 can't be loaded with latest asterisk version from cvs
5:42AM 1 How uniqueids are formed - possible race conditions for linked channels?
5:12AM 0 Looking for DIDs in Dubai
3:35AM 0 How to messure PDDs, how to detect fast hangup?
2:53AM 2 all circiuts busy now. resolution?????
Friday November 4 2005
11:57PM 0 Sipura 2000 could not show incoming call's number
10:34PM 0 10/28 head > 10/29 head capi issue
9:51PM 1 Snom 190 Vmail setting
9:31PM 1 HDLC errors on PRI
6:59PM 4 SIP extension calls itself intermittently
5:24PM 0 MFC/R2 - unicall
4:21PM 1 Different answering policies for two zap interfaces
4:14PM 0 RE: Your message to Asterisk-Users awaits moderator approval
3:38PM 0 TDM2420E Availaibility
3:05PM 1 [OTAnn] Groups:New Developments at Roomity
2:51PM 1 Cisco phone firmware
1:43PM 5 GSM sound player for windows?
1:26PM 4 Moments of silence - take2
1:20PM 1 Asterisk 1.2beta2 and UIP200
1:20PM 0 manual transfer to automated operator.
1:09PM 2 User language switching in dial plan
12:57PM 1 What do I need to setup Asterisk with an H323 client?
12:37PM 2 Moments of silence
11:27AM 1 R2-Digital (Q.421)
10:44AM 1 Problem on Data-Connections through Asterisk
9:57AM 2 Te100 Digital vs Analog
9:53AM 5 Uninstall AMP
9:20AM 2 Meetme: Sending DTMF to other users in a conference
9:12AM 0 Beta2 problems with DTMF with "T" option in Dial Command
8:46AM 1 Can´t compile asterisk1.2beta2
8:20AM 1 SIP phones supporting early dial
8:19AM 1 Does AEL support arrays?
8:17AM 1 Dial in via pstn , out over IP
7:44AM 0 re: Attempted to delete nonexistent schedule entry...
7:36AM 0 COREDUMP in actual CVS
7:34AM 1 Polycom IP 600/601 microbrowser specs
7:14AM 1 SCCP: ServiceURL and Mailbox Notification
6:33AM 0 one way audio on oh323 channel, there's no rtp traffic
5:52AM 0 $29.95 unlimited...and no catch in T &C is anyone using them
5:44AM 1 CVS HEAD Broken?
5:44AM 1 Route call based on CallerID
5:44AM 0 2 Dial plan questions
4:55AM 2 Asterisk connected with CAPI
3:57AM 2 Zaptel: Hz != 1000 causing ztdummy compilationerror?
3:34AM 0 Zaptel: Hz != 1000 causing ztdummy compilation error?
3:27AM 2 Called number (Destination Number)
2:10AM 4 IAX2.FWDNET.NET not responding?
2:07AM 1 Hold Music is breaking up
1:33AM 2 Every SIP on its own FXO
1:04AM 1 Forward call without answer
Thursday November 3 2005
10:38PM 0 ztmonitor usage
6:52PM 2 How to dial direclty from PBX extension to IP phone
5:32PM 1 Invalid/Timeout handlers in ael?
4:57PM 1 One Touch Record in 1.2
4:08PM 0 Voicetronix OpenPCI , anyone using this?
3:37PM 0 Multiple zaphfc cards (for ISDN BRI) in a single machinemachine
3:00PM 0 Cisco smartnet to download firmware
2:41PM 1 T38 not compiling with today CVS
2:28PM 0 chanisavail - queuing
1:56PM 0 sip to asterisk?
1:26PM 1 Asterisk and SER for Call Center Application
12:59PM 3 SIP Disconnect Supervision
12:52PM 1 References?
12:42PM 0 Problems with meetme dropping audio during call
11:57AM 1 chan_agent.c fails to compile
11:47AM 0 Unicall
11:31AM 1 TDMoE problem
10:12AM 2 Basic question...
10:02AM 0 spandsp changelog
9:54AM 1 timed allow functionality of 'include =>'s
9:46AM 3 How to configure Asterisk through webmin
9:34AM 1 How to call each other for dynamic ip hosts
9:22AM 1 Ignoring Incoming RFC2833 DTMF?
9:03AM 9 Looking por a provider to work with asterisk
7:41AM 0 Include statement options docs .
6:34AM 3 How to detect AGI script failure?
6:26AM 5 call from asterisk to SIP cisco 5300
6:00AM 0 Getting started, how to :D
5:13AM 1 curious bandwidth usage (incoming taking 3x more)
4:13AM 3 IAX test service
3:57AM 3 Multiple zaphfc cards (for ISDN BRI) in a single machine
3:16AM 0 Re: Asterisk and reverse DNS
3:14AM 1 Starting our own ip-telephony service?
3:13AM 0 Re: [Serusers] Accounting
2:35AM 0 Is this PRI INTENSE DEBUG correct (long)
2:06AM 0 Asterisk GUI/web interfaces that don'tchangeconfig files
2:03AM 3 Distinctive Ring Detection in AU
12:58AM 1 app_followme
12:08AM 1 Skinny.conf and sccp.conf
Wednesday November 2 2005
11:48PM 1 Call Disconnect problem
11:21PM 8 Response time of TDM04b
9:04PM 0 tweak rxgain to prevent premature hangup (and hungup) by fax machine when communicating with rxfax
8:45PM 0 Problem with an AGI script. Going bald on this one.
8:22PM 1 Send text to Cisco 7960?
7:53PM 5 Anyone know who is in this picture?
7:32PM 2 Sipura password not working
7:04PM 2 Very basic switching application -- bounty?
6:22PM 1 New asterisk web gui for small/medium sized businesses
5:36PM 2 1.2-beta2 odd CLI output
5:26PM 1 A2Billing Authentication Refused
5:25PM 2 RealTime extensions - why so many SELECTs per call?
5:13PM 1 2 Asterisk boxes
5:13PM 2 Queue Strategy problem or advice
4:43PM 2 listening on multiple port #'s
3:47PM 0 Masquerade a call
3:38PM 1 RE: [Asterisk-biz] Asterisk as a VoiceConferenceServer
3:28PM 0 Re: Anyone aware of a current Dell servermodelwith 3PCI slots
3:22PM 1 cli output issue
3:11PM 0 Error with loading an FXS module
3:02PM 2 Warning -- chan_iax2.c: ast_sched_runq tasks
2:47PM 1 MG1 echo canceller results
2:29PM 0 Cadence, distinctive ringing and zapata.conf
1:19PM 1 E1 PRI card 17:31 channels problems
1:18PM 0 Problems with some channels on PRI E1 card
1:16PM 5 PRI E1 Problem only chan 17-31
12:51PM 5 RE: [Asterisk-biz] Asterisk as a Voice Conference Server
12:26PM 1 Possible Issue With Meetme Conferencing in 1.2.0b2 and latest CVS HEAD (02/11/2005)
12:16PM 4 OS for ABE
12:11PM 3 faster transcoding possible
12:10PM 2 firmware update polycom 500 / dial problem
11:53AM 1 Fax between Asterisk SIP clients
11:33AM 1 A few Zaptel BRI questions...
11:11AM 0 Re: Re: intel e7230 chipset (Kevin Hanson)
10:25AM 4 Time based call direction
10:01AM 6 Satellite WAN
9:45AM 2 TDM0xB vs. SIP for FXO
9:17AM 1 How to bridge fax from pri to fxs
8:56AM 1 Voicemail in Realtime mode
8:14AM 1 intel e7230 chipset
7:17AM 1 extension
6:32AM 1 Zap Polarity Reversal
5:04AM 2 Options for 3-way or Conference Calling
4:15AM 0 REGEX() 1.2beta2
3:51AM 0 Noise in Echo()
3:29AM 1 is it possible to connect to Asterisk from an external application?
3:22AM 0 Ericsson MD evolution and asterisk
2:07AM 0 Fritz!Card PCI ver2.0
1:47AM 4 Installing beta2
1:14AM 0 [Voicemail] Quota
12:30AM 0 SV: dial-out gives always "not found" (dial-in worksfine)
Tuesday November 1 2005
11:12PM 0 CDMA gateway
9:38PM 1 changing email text based on voicemail user
9:20PM 1 Missing audio from Zaptel channels
7:05PM 1 Echo on TDM - Solved!
6:32PM 8 server hardware
5:09PM 2 Polycom files
3:40PM 0 IAXy Ringback Issues
3:25PM 3 How do you handle situation with Grandstream occasionally losing registration with Asterisk ?
3:10PM 0 Asterisk Beta 2 Possible Bug.
2:41PM 1 Double DTMF sent on T1 to T1 Native Bridge
2:36PM 2 Anyone aware of a current Dell server model with 3 PCI slots
2:19PM 3 Slightly OT: Cisco 7960/7940 and AsteriskReg istration Issues ove r a WAN
2:04PM 0 2 AgentCallbackLogin Questions
1:10PM 1 Slightly OT: Cisco 7960/7940 and Asterisk Registration Issues ove r a WAN
1:10PM 1 HT-486 Voice Nat Problem
1:05PM 0 New version (0.6) of Queue Statistics released
12:42PM 2 shared lines
12:30PM 2 Caller ID lookup via
12:07PM 1 format_mp3 error on 1.2b2
11:50AM 0 PRI to SIP D-channel Red Alarm
11:50AM 1 TDM dial in question
11:35AM 1 Error with one of my Zapata channels
11:25AM 4 feature.conf in 1.2beta2
11:12AM 2 Asterisk Extension Language -- what's it's "status"?
11:01AM 0 Latest CVS just noticed this warning for the first time. Bis
10:59AM 1 Delays in sip invites.
10:56AM 4 Latest CVS just noticed this warning for the first time.
10:16AM 1 PAP2 and ringing issues
10:01AM 0 BLINDTRANSFER and Referred-Byand Referred-By
9:33AM 2 mesreading echocancel vs. echocancelwhenbridged?
9:30AM 1 Asterisk 1.2.beta2 and chan_capi
8:41AM 1 1.2.0-beta2 and realtime sip
8:38AM 0 process ID in log file?
8:19AM 1 Incomming calls
8:07AM 1 Problem with call files
7:44AM 1 Adding caller name / ID to outbound meetme calls
7:37AM 6 inband dtmf on ploycom ip501?
7:19AM 3 chan_exosip2
7:10AM 2 Blind transfer from queue into another queue
6:52AM 0 User Permission
6:45AM 2 Fresh checkout Zaptel will not compile?
6:07AM 1 Voicemail Limits and Auto deleting
6:06AM 0 Forward sip messages to a proxy
5:25AM 0 UK BT Caller ID patches for X100P
4:25AM 1 problem with CME on 12.3(11)T6 and later (MWI)
3:36AM 0 Asterisk + Ser + Music on hold
3:19AM 6 1.2beta2 and spandsp
2:58AM 1 Polycom IP600 and micro-browser...
2:44AM 1 IAX2 trunking not work with slinear
2:26AM 0 How to program Phone "Configurable line indicators" for some PSTN lines
2:24AM 0 Different Ringing Tones depending of the call
2:23AM 0 No Media for Ringing Indication