asterisk users - Jan 2006

Tuesday January 31 2006
11:51PM 1 Leftover sound on isdn modem channel
11:41PM 0 About Meetme and CDRcustom
11:06PM 0 Asterisk and Fax ?
10:57PM 2 Comedian Mail Wont Take Password
10:12PM 1 T.38 patch instruactions
10:10PM 1 Strange echo phenomenon (double tandem)
9:32PM 0 Cisco Gateway - Context Issues
8:31PM 0 Ast<->Ast: IAX2 error w/no audio
8:07PM 4 Asterisk Registering with SER question
7:58PM 0 problem loading zaptel drivers
7:19PM 0 AirCard GSM and Asterisk PBX
6:37PM 0 continuing the context after caller hangsup
6:19PM 1 playing audio to one of several bridged channels
6:03PM 0 RE: [Asterisk-Announce] Asterisk 1.2.4 and Zaptel1.2.3
5:47PM 0 RE: [Asterisk-Announce] Asterisk 1.2.4 and Zaptel1.2.3
5:41PM 3 MOH sourced from a sound card?
4:18PM 0 Help with sip setup because can't receive calls!!!!!!
3:07PM 1 Networking voicemail
2:19PM 7 Teliax - Codec Preference effective?
2:11PM 0 Lucent MAX TNT faxing
1:57PM 3 ZAP <--> sip(polycom301) can not hear each other
1:53PM 1 Test Test
1:19PM 0 feedback on grandstream budgetone
1:08PM 1 Fw: Codec preference selection?
1:00PM 1 Channels Codecs
12:57PM 4 broadvoice??
12:23PM 1 R2 implementation problem
12:17PM 1 E1 PRI Error: Provider error messages.
12:13PM 3 Linking Asterisk Boxes with Sip
11:54AM 0 Snom 360 Message Waiting indicator
11:49AM 0 idefisk 1.31 - Voicemail Button
11:41AM 0 What is causing this error?
11:04AM 2 Asterisk hangs on 1.2.1
10:56AM 2 Canadian Termination $0.0039 / Minute
10:31AM 1 RE: Euro-ISDN
10:06AM 0 How to start a playback after the called partypicks up?
9:55AM 1 Asterisk 1.2.1 + TDM400P + fax machine unreliable ?
9:52AM 1 international caller id on UK (BT) PRI
9:20AM 1 Forwarding issue.
9:09AM 0 dialing 2 channelsatthesametimewithdifferentcaller ID number?
9:03AM 1 Voicemail greetings
8:54AM 0 unable to register using SIP
8:23AM 1 dialing 2 channels atthesametimewithdifferentcaller ID number?
8:18AM 5 Polycom IP501 Endless Loop
8:14AM 0 Asterisk 1.2 1 FXO Problem
8:09AM 1 dialing 2 channels at thesametimewithdifferentcaller ID number?
8:02AM 0 dialing 2 channels at the sametimewithdifferentcaller ID number?
7:52AM 1 Polycom IP301: Pass-through ethernet port unusable?
7:23AM 2 Gain adjustment
6:45AM 1 How to start a playback after the called party picks up?
6:35AM 1 Voipbuster incoming
6:20AM 0 newbie dial problem,
5:49AM 5 Queue() with timeout=0
5:03AM 3 Individual SIP account how to make it Trunk
4:35AM 2 Preventing Asterisk from transfering the call
4:28AM 1 meetme and dtmf
4:25AM 0 information on how to use asterisk for telephony boards other than given ones
4:10AM 2 Asterisk hardware.
4:03AM 0 New GXP-2000 Beta firmware available
3:08AM 1 missing pre pattern matching feature
3:01AM 3 Default value for ASTERISK_VERSION_NUM
2:12AM 7 Interface card for Euro-ISDN (BRI)
2:09AM 0 SV: Set caller id on Swedish PRI (euroisdn)
1:57AM 1 Forward a call from AGI/PHP script
1:38AM 2 R: Kirk IP600
1:32AM 0 Sharing a dialplan
Monday January 30 2006
11:48PM 0 dialing 2 channels at the same timewithdifferentcaller ID number?
11:20PM 0 dialing 2 channels at the same timewithdifferent caller ID number?
11:06PM 0 dialing 2 channels at the same time withdifferent caller ID number?
10:50PM 1 Grandstream Budgetone BT-101 audio problems
10:37PM 0 dialing 2 channels at the same time with different caller ID number?
10:11PM 0 Meetmee weirdness
9:47PM 2 RE: [Asterisk-Announce] Asterisk 1.2.4 and Zaptel 1.2.3
9:33PM 0 Asterisk 1.2.4 and Zaptel 1.2.3
7:47PM 1 Asterisk SIP phones to Cisco Unity via CCM4.0SIP Trunk
5:57PM 3 polycom ip601 attendant console
4:10PM 0 mISDN errors on asterisk CLI
3:40PM 2 Asterisk Evening in Melbourne: Feb 2!
2:28PM 0 sip domain
1:51PM 0 load balancing
1:37PM 0 Dlink DVG-3004S ?
12:45PM 0 re: help with redirect from SER
12:43PM 0 Codec preference selection?
12:43PM 1 help with iaxy - one way sound
12:42PM 8 Analog with channel bank - Inbound works, outbound doesn't
11:56AM 2 Cisco 7940 not reading SIP image file
11:40AM 1 Kirk IP600
11:35AM 0 About Extensions
11:17AM 1 Connecting the two servers
10:23AM 1 Live CD?
9:55AM 5 How many TDM2400P's will a server take?
9:42AM 1 Cant compile asterisk #error "You need newer libpri"
9:34AM 2 Most Popular FREE SoftPhone for Windows
9:34AM 1 Need to recompile * after changing zap echo method?
9:21AM 4 DID over analog?
9:13AM 0 Question on SIP Domains and registration
8:50AM 0 Unable to do anonymous outbound calling
7:43AM 3 Set caller id on Swedish PRI (euroisdn)
7:18AM 1 Manage api- Matching 'Newchannel' event with the 'Originate' command
7:14AM 1 Gateways
7:13AM 1 Help configuring Asterisk server
6:37AM 1 Asterisk and LCS ?
6:26AM 0 backgrounddetect and busy
6:23AM 1 SIP-H323 translation
4:33AM 0 Realtime Queue not realtime anymore in Asterisk 1.2.3?!
4:30AM 1 Playing music while transfering
3:59AM 3 adress book
3:33AM 1 app_snmp
2:42AM 5 Grandstream Budgetone mass deployment?
2:08AM 0 intel 536 EP as x100p clone?
1:59AM 3 How many digium cards per server ?
Sunday January 29 2006
9:29PM 0 dialing 2 channels at the same time with differentcallerID number?
9:08PM 0 dialing 2 channels at the same time with different callerID number?
8:41PM 0 Transfer (SIP REFER) - AccountCode not available?
8:38PM 0 Dialogic / Voip Forum
6:29PM 10 Web interface
3:58PM 2 Access Codes
3:40PM 4 Asterisk + XEN does it make sense?
3:00PM 0 Cisco VG200 as FXO for * ?
12:43PM 1 New C7960 won't tftp?
11:40AM 1 Unable to get IP of eth0
11:36AM 1 HandyTone 488 ata?
11:18AM 2 simulating a few thousand SIP clients?
11:04AM 0 strange performance issue
8:32AM 2 username not stabled?
7:39AM 4 How to remove first ring tone on FXO?
7:01AM 0 Modprobe Zaptel error
6:58AM 1 file.c:509 ast_openstream_full: File 100 does not exist in any format
6:46AM 1 Moprobe Zaptel error
6:16AM 3 Wifi phone set-up
5:49AM 0 Real-time: username
4:11AM 1 changing displayed call info on snom 360
3:25AM 1 Asterisk as SIP endpoint ?
Saturday January 28 2006
8:54PM 0 Adjusting gain, Milliwatt and ztmonitor
7:51PM 3 Urgent: Unable To Execute after updating from SVN
5:29PM 3 Multiple Subscriptions to SIP accounts at Same Domain
4:18PM 1 english snom support forums ?
4:14PM 2 RoadRunner
3:53PM 0 voicetronix FXOs with * ?
3:06PM 0 AutoDialing with VOP USING SIPURA 2100'S
2:08PM 0 Help with Music on Hold during transfer
1:42PM 1 Installing the none commercialintelg729codecsinto Asterisk@Home 2.2?
12:03PM 0 How to Unregister?
11:21AM 1 Looking for 150 SIP desktop phones with power over ethernet that will work with Plantronics HL-10 Handset Lifter for Remote Answering
10:28AM 2 Best CoDec for high network latency
9:46AM 2 VOIP carriers and asterisk
9:45AM 0 Other side disconects when using TxFAX
8:51AM 1 Can't send DTMF transfer code from called SIP phone
8:00AM 0 Re: 5, 000 concurrent calls system rollout question
7:01AM 0 Re: Lockups since upgrade 1.2.3 - anyone else? Anyideas?
6:18AM 1 regarding connecting to AMP>>
5:39AM 1 double ringing tone on asterisk 1.2 (workaround)
2:32AM 2 Trunk is not released
1:34AM 3 No IN and OUT on ISDN line at the same
1:27AM 3 Simple question about ringing multiple phones (extensions)?
1:09AM 3 (Un)PauseQeueMamber usage
Friday January 27 2006
10:44PM 1 Installing the none commercial intel g729 codecsinto Asterisk@Home 2.2?
10:13PM 2 Name/username (sip show peers)
10:13PM 1 shared fxo line
10:07PM 3 G729 Commercial Licenses.
9:28PM 23 5,000 concurrent calls system rollout question
8:40PM 2 DTMF's indescipherable, but voice clean!
4:45PM 1 Agent counts
4:38PM 3 sip qualify=yes interval
3:33PM 0 Re: Asterisk-Users Digest, Vol 18, Issue 181
3:16PM 2 fxo/fxs cards with 8 ports
3:02PM 2 VOXEE Caller ID..
2:58PM 0 FlashTransfer to Bridge
2:55PM 0 Page() and Asterisk 1.2.3 Problems?
2:46PM 0 moh & clock
2:46PM 1 How's the best way to set up agents...
2:22PM 4 CDR reporting between two Asterisk servers
2:20PM 1 Help with Congestion error
1:45PM 0 Good provider of Polycom Phones (mostly for accessto latest/greatest firmware)
1:35PM 0 starvox communications
12:41PM 6 Lockups since upgrade 1.2.3 - anyone else? Any ideas?
11:48AM 0 SIP channel not diconnecting on hangup
11:35AM 1 chan_bluetooth: successful compile and outbound cell calls: Still tweaking inbound setup. WAS: Cannot compile chan_bluetooth on Asterisk 1.2.1
10:33AM 1 802.1p
9:45AM 1 SIP incoming calls
9:29AM 0 Digium Wildcard TDM400P call pickup timing
9:24AM 1 No IN and OUT on ISDN line at the same time?
9:19AM 0 Problems with MFC/R2 in Brazil
8:41AM 5 SER redirect
8:26AM 1 Good provider of Polycom Phones (mostly for access to latest/greatest firmware)
8:09AM 5 External IAX2 phone defined as internal behaving as from PSTN
8:06AM 3 OT?: International number parsing
7:16AM 0 Caller Presentation
7:13AM 7 AAH out bound routing problem
6:48AM 0 Newbie SIP trunk question...
6:27AM 1 T38 providers
5:46AM 0 ATA's ???
5:00AM 0 wcfxo md3200 problem...
4:43AM 0 pb with callerid
4:29AM 2 Spa3k and ISDN
4:13AM 1 Outgoing FXO and CDR
3:48AM 2 WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38
2:04AM 0 ODBC Problem with voicemail.
1:57AM 0 No matching peer or user based on IP address
1:32AM 0 Offtopic: Auto provioning Snom 360
1:31AM 1 Packeting multiple GSM frames in one IP packet - Help needed.
12:49AM 3 Max concurrent calls
12:23AM 3 paging agi
12:05AM 0 How to put peers into Realtime
Thursday January 26 2006
11:58PM 0 Alex Tew interview made possible because of Simon @ Simwood eSMS
11:47PM 1 S100-FX v2.0
11:42PM 3 Chan_capi on builds 7955>8320 strangeness
8:15PM 2 Shared Line Appearance
8:08PM 1 Current viewpoints on the Sayson/Aastra 480i
7:59PM 0 Using Flash
7:31PM 1 ISAC Codec Support
6:07PM 4 extension to extension dialing
5:55PM 2 PRI restarting each hour?
5:17PM 0 Webphone with Asterisk??
4:43PM 5 Is Voxee down?
4:30PM 0 Anybody will to share a sm bus polycom dialplan to share?
4:20PM 1 Round Robin Call Distribution
3:43PM 0 Re: OT: Legacy systems / fax
2:36PM 0 To which context a "registered peer" is sent?
2:29PM 0 How set up "call forward on no answer" for allextensions?
2:27PM 2 Transferring Using Flash
2:19PM 0 Hamachi with Asterisk
1:25PM 2 checking voicemail via trunk
1:08PM 1 Manager API mailing list
1:00PM 1 CDR problems
12:43PM 5 Skype-to-Asterisk(SIP): progress
12:33PM 0 Aastra 9112i or 9133i Ring Tones
12:02PM 1 fax detect DID variable
11:41AM 0 Asterisk 1.2.3 CentOS 4.x RPMS <--SpanDSP Ba ckport?
11:29AM 2 Announcement: Snom 360 with integrated XML O bjects
11:23AM 0 Pause/UnpauseQueueMember
11:06AM 5 Asterisk 1.2.3 CentOS 4.x RPMS
11:06AM 3 snom 320 echo problems
11:02AM 1 Announcement: Snom 360 with integrated XML Objects
10:40AM 0 app_background and app_cepstral
10:13AM 0 Local Channel Call Looping
10:00AM 2 Dynamically disabling echo cancellation (Zap).
9:50AM 1 CDR logging in /var/log/asterisk instead of MySQL DB
9:42AM 0 Plea to support a much needed function for Call Centers in Asterisk.
9:38AM 0 [Fwd: Asterisk as an Ascend box]
8:52AM 6 Fail over to Pri on VoIP connection failure
8:31AM 2 Snom360 Sidecar & Asterisk
7:58AM 6 * point to point t1 solution? / alternatives
4:51AM 0 0h323 - one way audio
4:26AM 1 Calls pickup
3:54AM 2 using sangoma cards as a timesource?
3:52AM 0 Good switchboard solution?
2:24AM 0 codec selection based on call prefix
2:08AM 1 TDM400 pinout
1:30AM 0 Missing meetme recordings.
1:02AM 7 Bootable CD?
12:42AM 3 VOIP Router
12:16AM 1 Asterisk Setup Question -- Please Help
Wednesday January 25 2006
11:53PM 20 * point to point t1 solution?
10:49PM 1 asterisk 1.2.3 call problem
10:46PM 5 transfer, recording ...
10:24PM 1 Adding number prefix on Polycom SP300 phone
10:19PM 2 Voipbuster/voipstunt -- what a crap service
10:18PM 0 Free calls to UK, US and Germany???
9:33PM 0 Want to automatically park call and have callerhear ring tones
9:14PM 0 Received fax "offset" in tif file?
8:54PM 2 Changing Asterisk install location...
8:53PM 0 Re: Asterisk-Users Digest, Vol 18, Issue 158
8:37PM 0 Re: Asterisk-Users Digest, Vol 18, Issue 158
7:45PM 3 Fast AGI Options. Eeeek!
7:30PM 1 Want to automatically park call and have caller hear ring tones
6:58PM 1 Speech playback getting cut off
6:18PM 0 include from database
5:44PM 0 Monitor and * 1.2.3: Sync issues?
3:23PM 1 Dial String Questions
3:03PM 2 Best FXO hardware for home use
2:37PM 0 asterisk 1.2 with grandstream ht-496 2nd port registration issues
2:04PM 0 Steal with MusicOnHold
1:39PM 4 VoIP in India
1:15PM 1 Disregard: Looking for the .xml file format for idleURL for Cisco 79xx
1:03PM 0 SIP register vs SIP with a fixed IP
12:47PM 10 Asterisk 1.2.3 Released - Critical Update
12:46PM 0 Parking from external PBX
12:31PM 0 SIP re-invites ignored by other end
11:10AM 0 feature transfer on PRI
11:05AM 4 Setting ringtone on Polycoms
10:58AM 1 BroadVoice subscribers and Asterisk 1.2.3
9:22AM 0 Looking for the .xml file format for idleURL for Cisco 79xx
9:11AM 0 chan ooh323 choppy sound
9:10AM 0 Echo while using Headset with Polycom IP 501 / 601 Asterisk 1.2.1
8:21AM 1 NEAX 2000 IVS Integration
7:55AM 0 Polycom 601 Bricked?
7:47AM 0 Problem in auto dialing through call files
5:58AM 2 Help with sip setup because can't receive calls
5:41AM 0 ISDN / Analog
5:25AM 1 Asterisk + Ericsson PBX
2:59AM 14 No audio? Update your Asterisk
2:49AM 1 ISDN D-channel disconnects for a minute every 5 minutes
2:49AM 5 trunk to trunk forwarding
2:10AM 0 need an bench-marking tool
2:08AM 0 dailplan questions
1:22AM 0 (no subject)
1:15AM 1 ACT-P104S SIP Firmware
12:52AM 1 jitterbuffer causes no sound?
12:49AM 2 Suddenly No audio
12:25AM 0 Very low audio levels after asterisk answers inbound calls
Tuesday January 24 2006
11:55PM 0 (no subject)
7:27PM 0 Re: Anyone using verizon fios ftth foranalogvoice?Any echo?
5:55PM 0 Re: Anyone using verizon fios ftth for analogvoice?Any echo?
5:51PM 0 Asterisk and IP Aliases
5:34PM 0 Astbill and Wholesale
4:53PM 0 Including files in AEL file
4:40PM 3 Linksys SPA-941 multiple line appearences
4:14PM 1 E1 -> T1 native bridging for fax, will it work?
4:04PM 0 No ringing
2:57PM 1 Hunting for DIDs in Kenya/Nigeria
2:51PM 0 SIP call failover
2:38PM 0 analog channels answer detection anything new in1.2.X
2:15PM 1 AAH 2.0 fax problems continued
2:00PM 1 Paging HardPhones
1:05PM 5 Looking for Q.Sig success story
12:52PM 1 oh323 and asterisk v1.2.2
12:38PM 3 ZAP - Can't pickup calls on Analog Trunk
12:18PM 1 Mini frame before first full voice frame (IAX)
12:08PM 4 which gui for asterisk on web
11:54AM 0 Disable music on hold per user
10:38AM 1 CAPI crash/lockups?
10:31AM 1 Re: Anyone using verizon fios ftth for analog voice?Any echo?
9:50AM 1 Call Parking - Set ID on return
9:36AM 0 OT: testing email routing
9:12AM 0 How to keep Asterisk (1.2) out of the media path
8:36AM 0 pulsedial on fxo signalling
8:35AM 2 txfax application problem
8:26AM 0 PhpAgiTutrial
8:07AM 13 Nortel Meridian Opt 81C and PRI
7:54AM 0 Microsoft Office Communicator 2005 as SIP client?
7:26AM 4 Asterisk with SuSe 10
7:15AM 0 Help compiling bristuff on FC3
6:54AM 2 Re: Asterisk-Users Digest, Vol 18, Issue 134
6:54AM 1 cannot change distinctive ring polycom phones
6:34AM 0 Nortel IP2000
5:51AM 0 Re: Asterisk-Users Digest, Vol 18, Issue 144
5:48AM 0 Problem: have no RTP streams from Asterisk
5:01AM 6 iax provider
4:35AM 0 H.264 and AAC codecs
4:09AM 1 Voipbuster problem
3:36AM 3 Simple setup ...
3:16AM 5 Is it possible ?
2:39AM 1 iaxphone for ubuntu 5.10
2:17AM 1 suggest a gsm router
2:13AM 8 UK Provider
1:47AM 1 MOH begin behavior
1:44AM 0 Anyone using verizon fios ftth for analog voice? Any echo?
1:20AM 0 What happens to global and channel variables?
12:58AM 1 need help asterisk and AS5300
Monday January 23 2006
11:18PM 3 MOH Server
10:18PM 0 Jumping on the asterisk bandwagon
9:03PM 2 weird zttest result
6:46PM 14 Polycom 501 horrible echo
6:42PM 3 Config File Storage
6:01PM 0 asterisk fax to pdf, blank pdfs?
5:19PM 0 DTMF not working on overseas cellphone calls
4:48PM 1 chan_capi - B3 Error
4:24PM 3 canreinvite always =no * no matter what we try :-(
4:18PM 0 Firewall/Embeded System/CF/etc
3:52PM 1 OFF TOPIC: Core router upgrade for a voip colocation center
2:40PM 2 Newer version of Zaptel with 1.0 branch of *
2:26PM 2 Polycom phones and dynamic IP for NAT
2:09PM 4 make linux26
2:07PM 0 Help with bad audio using MPC..
1:48PM 1 Asterisk SIP phones to Cisco Unity via CCM 4.0 SIP Trunk
1:34PM 2 Fw: setting outgoing caller ID by the queue an extension is logged into
1:26PM 1 background SayDigits()?
1:21PM 3 SPA-3000 - the party's over :-(
1:06PM 0 user not seen
1:01PM 2 analog channels answer detection anything new in 1.2.X
12:47PM 1 Video Conferencing.
12:29PM 1 Installing the none commercial intel g729 codecs into Asterisk@Home 2.2?
12:27PM 5 dial out and message playback
12:19PM 0 Call Waiting CallerID
11:47AM 1 Answering Service Add-on?
11:44AM 0 Odd asterisk behavoir
11:40AM 0 H.323 videoconferencing with asterisk?
11:26AM 3 How to view Q.931 Disconnect code
11:08AM 2 Home Test!
10:48AM 0 Polycom videoconferencing with asterisk?
10:20AM 0 Problem with Codecs
10:16AM 7 G729a Pass-Through and Recording/Monitoring
10:06AM 1 SIP over TCP: latest news?
9:58AM 3 Announcing PodMail 1.0 (GPL)
9:31AM 1 not able to start asterisk
7:55AM 0 openH323 from cvs
7:30AM 1 Testing List (JUST A TEST)
6:47AM 1 debug with ser
5:30AM 0 Sip Extensions
4:59AM 0 SIP response 300 "Multiple choice" ???
2:38AM 0 Xlite set-up program
2:26AM 1 How to set-up LCR
1:53AM 0 Re: Asterisk-1.2.1.tar on Suse Linux 9 (Atif Nadeem)
12:53AM 5 Bug in attended transfer or as expected?
Sunday January 22 2006
11:16PM 0 changing agent passwords without reloading asterisk
9:45PM 0 Forwarding out to cellular phone's voicemail with AMP
8:42PM 1 SNOM 190 Daylight Savings
8:33PM 3 how to set caller id?
7:56PM 1 macro-faxreceive
7:19PM 6 spandsp Error
7:13PM 0 Finding good, objective reviews of major VoIP phones
4:43PM 0 IP SIP Phone/2.0.6
4:05PM 4 Detection of Answering Machine
2:40PM 0 SIPDiscount inbound number
2:19PM 4 Snom 320 and message retrieve key
12:44PM 0 Interrupting ring to go to voicemail pickup -- How to ring after Answer()?
12:39PM 1 asterisk 1.2.2 and zap channel voice detection
12:11PM 1 Fail over using CHANAVAIL
11:45AM 1 Gen. Question
11:40AM 3 Installing the none commercial intel g729codecs into Asterisk@Home 2.2?
10:29AM 1 Distinctive ring detection using SIP - Broadvoice addon line detection
9:42AM 1 wildcard matching in dialplan
8:52AM 0 Thanks for all your messages
8:04AM 0 RE: Asterisk-Users Digest, Vol 18, Issue 131
6:42AM 5 T3 Mux and Asterisk Question
6:37AM 1 Installing the none commercial intel g729codecsinto Asterisk@Home 2.2?
5:09AM 0 Asterisk cut offs on TE110P
5:00AM 1 Asterisk TS-1
4:59AM 1 Gateway TIMEOUT
4:30AM 1 Asterisk-1.2.1.tar on Suse Linux 9
2:46AM 2 Disposition codes in CDR
Saturday January 21 2006
11:52PM 0 Extensions for in-bound faxes w/o properly-provisioned T1.
10:56PM 1 Can you disable Forward on a Polycom phone?
9:21PM 1 Compiling app_cepstral.c into Asterisk - failing
6:14PM 3 cvs asterisk compile failed (newer libpri)
5:54PM 2 Tellabs 2572 EC Photos here.
3:57PM 2 How to disable WARNINGS in CLI
3:36PM 0 [Announce] Mark Spencer interview
3:01PM 0 sip outgoing calls over proxy
2:52PM 1 h323 configuration
2:12PM 1 TE110P + PRI incoming + outgoing extensionsquestion
1:49PM 0 Anyone interested in getting a basic training course together for the greater NYC area?
12:42PM 1 Is corking reliably for others on list?
12:30PM 0 Dialstatus Oddity in 1.2
11:56AM 1 SIP and NAT - best practices?
11:30AM 1 Caller ID and Sipura Router
11:02AM 4 asterisk + usb celular
10:42AM 0 jitterbuffer on zap channel
9:35AM 1 Asterisk 1.2.2 - Double Quote on CallerID Causing SIP Problem (7940)
7:47AM 0 Rotated Logs Per SIGXFSZ every few seconds
5:37AM 0 Providers with jitter buffer
4:53AM 7 MeetMe Dialplan question
3:26AM 3 Asterisk always uses address
Friday January 20 2006
10:29PM 0 Activate Call Waiting by default
10:27PM 0 Custom cdr trouble, help this newbie
9:49PM 1 SIP problem picking up the call
9:31PM 1 Need a good extensions.conf sm bus config w/polycom phones
8:56PM 0 Japanese J1 question
7:28PM 2 TE110P + PRI incoming + outgoing extensions question
6:06PM 0 Need a good extensions.conf sm bus config w/ polycom phones
5:10PM 3 OT:Snom 360 prompt for registration pwd?
4:45PM 1 Why is agents.conf not utilized? (aka: can't find good info on agents and queues for AMP)
4:41PM 1 Teliax Down?
4:40PM 7 Asterisk Development and Release Cycle
4:15PM 5 When/whether to use SER?
3:40PM 1 How to Clear SIP Channels
3:26PM 1 Calling MySQL 5 stored procedures from app_mysql
2:56PM 1 SPA-941 auto-answer capability
2:51PM 0 Queues & All Agents Busy
2:51PM 1 applicationmap
2:34PM 1 cisco 7940g, 7960g phone screen sizes?
2:09PM 1 Dell PowerConnect 2724 Switch and QoS for VOIP?
1:48PM 5 Asterisk in SPA9000?
1:46PM 1 AIX calls with sipdiscount
1:40PM 0 sip notify on sipura?
1:36PM 1 HardPhone Dilemma
1:36PM 2 Agressive echo cancelation
1:33PM 0 Problems with incoming PSTN calls
1:28PM 0 Cisco 7912G SIP phone and Asterisk double RTP packets
1:18PM 1 SIP, NAT and Firewalls
1:15PM 2 ztdummy on opteron
1:14PM 1 Hardwiring a Tellabs echo canceller - help req
1:10PM 0 h extension
12:46PM 0 Help with poor audio using SIP
11:58AM 1 Can TE406P provide PRI to other VoIP gateways?
11:47AM 2 Asterisk bounty PRI 2B channel transfer for NI2 PRI line
11:09AM 2 Conversation interrupted by fax
11:02AM 5 iDEFISK (mac iax2 softphone) release
10:49AM 1 How to have a phone ring another extension as soonas off-hook?
10:41AM 1 Dial command not executing following priority when caller hangs up
10:32AM 2 How to have a phone ring another extension as soon as off-hook?
10:13AM 1 quality and delay test
9:43AM 0 Realtime - reading values from registred family name
9:40AM 0 newbie cdr_custom and cdr_csv2 problem, please help
8:50AM 0 Double Progress Tone
8:36AM 2 no nat, but one way only audio (more info)
8:22AM 1 IAX and call transfer
7:53AM 0 can wengophone or gizmo be used directly with asterisk???
7:45AM 0 multithreading for res_perl
7:24AM 1 2400P Pinouts
7:21AM 0 Mapoing extensions to specific trunks
7:19AM 1 Connecting a TE to a NT BRI isdn
6:52AM 0 R: Dect to SIP PCI card
6:40AM 2 no nat, but one way only audio
6:28AM 0 Asterisk and Cisco GW
6:16AM 1 instant fallback to zap in case of sip/iax/xyz-failure
5:54AM 0 No translator path: iax2 calls not possible
5:51AM 2 AVM C4, asterisk-1.0.8, /etc/asterisk/capi.conf
5:44AM 1 more voicemail frustrations (was: realtimevoicemail)
5:36AM 0 AstTAPI and CallerID Popups
4:50AM 1 SIP phone receiving but not transmitting
4:49AM 3 Detecting a PRI failure from dialplan
4:35AM 1 How to Confiure Voicetronix V4PCI16 in asterisk
3:15AM 1 No congestion
2:55AM 2 'h' in CDR
2:12AM 0 [ANNOUNCE] Asterisk::LCR released on CPAN
1:52AM 3 Dect to SIP PCI card
1:43AM 1 RTCP XR support (RFC 3611)
Thursday January 19 2006
11:54PM 0 sipura 3000 help needed
8:41PM 1 Port forwarding on a DLink Di-604
5:14PM 1 Problems with Module
3:53PM 1 TDM400P zttest not working
3:48PM 0 PRI
3:32PM 1 Cannot compile chan_bluetooth on Asterisk 1.2.1
3:03PM 0 New astGUIclient/VICIDIAL release: 1.1.9
2:13PM 13 Polycom FW
1:28PM 0 DTMF not recognized on overseas call from cellphone
1:04PM 0 AudioCodes Unreliable DTMF Detection
12:45PM 2 Dial() Jumping behaviour and Vesrsion 1.2
12:01PM 0 Fw: chanspy
11:56AM 0 AGI Tx error 510
10:52AM 4 Disabling zap echo cancellor from dialplan
10:45AM 1 Sound issue with Asterisk
10:40AM 1 Problem with rxfax - Dropping incompatible voice frame?
10:40AM 0 transfer and zap
9:30AM 1 (newbie) using dtmf during a call
9:11AM 1 DTMF # ?
9:10AM 0 Incoming fax on voipbuster
7:23AM 0 Asterisk least cost routing expert needed
6:34AM 0 sipTAPI and usernames
6:24AM 0 Connection pooling
6:23AM 3 Processor Size
6:02AM 0 Loud Tone issue, still having problems
5:33AM 0 Problem configuring Asterisk
3:48AM 1 CDR Accounting Question
3:14AM 0 A problem in recieving voice on one side
3:12AM 2 Brief silences during calls
3:03AM 1 DTMF Simultaneous Inband and RFC2833 performedby Asterisk => Duplicate tones
1:30AM 0 spandsp-0.0.2pre22 not working!
1:13AM 2 Asterisk and Linux-HA
12:36AM 0 Fax sending with asterisk 1.2.1
12:10AM 0 SoCal Asterisk Users Group Thursday Evening
12:03AM 0 agi_callingpres: 0
Wednesday January 18 2006
11:45PM 1 SIP RTP Negotiation
9:37PM 1 speex in asterisk 1.0.10
9:12PM 1 bug in Authenticate application ?
9:08PM 5 SMS to fixed phone line
8:13PM 3 asterisk 1.2.2 RPMS for CentOS 4.x
7:34PM 0 IAX2 between two * server not working
7:24PM 1 Sip phone with Bluetooth - does it exist?
7:19PM 0 Asterisk with GnuGK
6:58PM 0 OT: Network Wire Brand
6:43PM 0 Re: Asterisk-Users Digest, Vol 18, Issue 115
6:33PM 1 Iaxmodem and Efax?
5:58PM 1 howto change language?
5:13PM 1 I see Asterisk 1.2.2 into the ftp or was a vision?
5:02PM 1 Asterisk 1.2.2 Released!
4:57PM 2 SipAddHeader bug?
4:56PM 0 RTP error problem
4:18PM 1 Still the LDAP Realtime extension
4:10PM 0 Re: Asterisk-Users Digest, Vol 18, Issue 114
3:35PM 1 chan_sip.c:5262 sip_reg_timeout Probably a DNS error for registration
3:10PM 1 Polycom 301 DTMF
3:06PM 0 O'Reilly's Etel Conference
2:33PM 1 DTMF Simultaneous Inband and RFC2833 performed by Asterisk => Duplicate tones
2:31PM 2 modem simulation
2:00PM 4 sipura ata 3000 UK (BT) CAllerid
1:26PM 1 Bugs that Need Your Input!
1:02PM 0 Call Waiting CallerID not showing up
12:26PM 5 SAN Devices
10:54AM 1 Disconnecting call 'SIP/X.X.X.X-085340d0' for lack of RTP activity in 11 seconds
10:36AM 0 Atcom AT320: SIP or IAX?
10:22AM 2 1.2 in production w/100+ phones?
10:13AM 0 misdn svn
9:48AM 0 Problem with Vonage and Asterisk, Please help me
8:53AM 1 detect when grab up the phone
8:47AM 1 Dial Rules in localprefixes.conf
7:57AM 1 Web Conferencing
7:44AM 2 Asterisk Sound Issue
7:30AM 0 PING users of Manuel Guesdon's LDAP extensions
6:32AM 0 asterisk 1.2 bristuff and sms
6:20AM 1 LDAP direct authentication Problem
6:10AM 0 Force Port Number on INVITE
5:16AM 0 Asterisk Fax part 2
4:32AM 1 Australian Asterisk Job Listing
3:49AM 0 Problem with DIAX and Asterisk and Vonage
3:00AM 0 CPU utilization in general
2:49AM 1 PRI D-channel errors
2:48AM 0 get only GHOST fax
2:48AM 1 SpanDSP not sending to fax extension.
1:54AM 0 rtcachefriends and REALTIME + MWI
1:20AM 0 SIP IP Phone is not registering [urgent]
12:12AM 1 Attended transfer reconnect when goes to voicemail?
Tuesday January 17 2006
11:00PM 0 Problem with Asterisk and DIAX, Please help me
10:41PM 1 Qwest can't/won't
7:25PM 0 A few straightforward questions about 1.2
7:00PM 0 Loud Tone When Key Pressed
6:50PM 0 ruby-agi 1.1.0 released
6:33PM 0 rx/txgain per device?
5:07PM 2 MeetMe Listen Only flag (|m)
4:55PM 2 How do you deal with subprefixes with LCR?
4:39PM 0 Line transfering calls back to asterisk system from another pbx
4:21PM 2 chan_sccp crashes Asterisk on startup
4:21PM 0 How to compile and install just one module?
4:18PM 3 [Asterisk-Dev] WAS: click-to-call cleint NOW: XML Manager I/F str aw poll
3:46PM 1 Make SIP calls go back out a zap trunk
2:55PM 2 Re: Choosing an FXO card, Asterisk-Users Digest, Vol 18, Issue 100
2:36PM 1 Asterisk and Fax part 2
2:35PM 1 Asterisk & Genesys integration
1:41PM 1 Voicebroadcasting with Asterisk?
1:25PM 1 Slightly OT: Plantronics headset quick connectorwiring
1:07PM 2 idefisk 4 linux now available for download
1:02PM 1 red alarm?
12:34PM 0 Includes affecting menu, zaptel transfers
12:05PM 0 RE: Building from scratch would like the benefit of (TOO LONG...)
11:35AM 2 Problem with ISDN HFC-S card
11:10AM 1 Asterisk LDAP Authentication Problem
11:08AM 0 asterisk.ctl limitations
11:04AM 0 RE: Building from scratch would like the benefit of (TOO LONG...)
10:41AM 3 Fritz card technology & German *
10:22AM 4 How to find out if a new voicemail exists
10:07AM 0 1.2.1 canĀ“t register with SIP-Provider, 1.0.9 could
9:54AM 0 call fails first time, then succeeds
9:51AM 6 nwebmail
9:44AM 6 OT: DCAP Certification
9:31AM 0 Verizon DTMF Recognition
9:03AM 1 ACD announce-holdtime
8:52AM 0 Slightly OT: Plantronics headset quick connector wiring
8:22AM 2 auto load SIP peers on startup
7:32AM 0 Problem with installation of rpm's, Please, help me.
7:28AM 2 Building from scratch, would like the benefit of everyone's experience
7:24AM 0 SIP hardphones with xml/html/xhtml/microbrowsersupport?
6:46AM 2 Is Asterisk the right tool?
6:44AM 2 Problem configuring Asterisk, Please help me
6:18AM 0 FYI - Cisco IP Phones SYN Flood Device Reload Vulnerability
4:44AM 3 experiences with teliax, voipjet or junction networks?
4:40AM 1 Call Center sofphone
3:56AM 0 SVN Compile Error
3:39AM 3 Phone still rings while on a call
3:36AM 1 Asterisk under SUSE 9.2/VMWARE 5.5.1
3:00AM 1 Is there a key sequence to stop a call as its ringing?
2:10AM 2 IAX/SIP and openser problem. IAX bug?
1:54AM 1 Hold on with Asterisk Manager
12:53AM 0 Possible Job
12:01AM 0 Asterisk RELAY
Monday January 16 2006
10:47PM 0 Quadra software - Changing to Opensource
10:44PM 3 CVS HEAD chanisavail not working for sip channel?
10:14PM 1 Problem with installation of rpm's, Please help me.
9:04PM 1 RTP redirect system usage
7:00PM 2 question about zttest
6:45PM 1 Incoming call: Got SIP response 503 "Server error" back from
5:55PM 5 SIP hardphones with xml/html/xhtml/microbrowser support?
5:50PM 1 modify a cdr values..
4:59PM 2 automon - one touch record
4:56PM 1 I've sent a message to the list 6 hours ago and it's still not showing up
4:41PM 1 setting Cisco 7940 to factory default
4:36PM 2 Agents getting logged off agressively
4:08PM 2 Call Center and Predictive dialing
4:01PM 1 cisco 7940 firmware upgrade
2:22PM 0 Looking for Full Time technicians
2:20PM 0 asterisk 1.2.1 crashed
1:39PM 1 Zapata.conf and Realtime
12:46PM 5 Dundi Examples
12:41PM 1 TE210P Trade
12:36PM 1 making wakeup feature call phone number, not extension?
12:23PM 0 FW: Exited non-zero
12:20PM 0 Asterisk RTP Bridging
11:53AM 1 IAX voice distortion with full upload channel /SIP ok
11:10AM 0 Set(LANGUAGE()=language throwing warnings
11:05AM 2 New RPM packages for CentOS4.0
11:02AM 2 MeetMe greeting message.
10:47AM 1 Support for RFC3323?
10:30AM 2 Problem with calls starting from a legacy PBX
10:03AM 0 How to put someone on hold with Astersik Manager
9:43AM 2 ztdummy inaccuracy on linux-2.6
9:36AM 1 Asterisk for Call Center (missing reference)
9:26AM 0 Asterisk for Call Center
8:34AM 0 Asterisk with Cisco
8:33AM 2 AGI variables
7:57AM 3 Max Number of #include statements
7:56AM 4 new in asterisk world
7:43AM 2 Pickup Button
7:33AM 0 FW: confirmation
7:28AM 0 strange voicemail issue
7:25AM 2 cmd Dial parameters
7:20AM 0 OT: ignore me, just a test
7:13AM 1 Dynamic conference - add participants
6:51AM 2 agi debug - unable to set normal priority
5:02AM 3 asterisk down because of cdr
4:27AM 3 distorted native music on hold
4:05AM 4 problems with a pri (E1)
3:28AM 1 chan_capi-cm and DID
3:23AM 0 Pre-made E1 crossver cables for the UK
3:16AM 1 Test to see if I'm still on list...
2:58AM 0 SIP Error 401 Problem
2:37AM 0 dnid
1:20AM 0 asterisk1.2.1/PRI-E1 outbound call issues
12:07AM 0 Zapata.conf Realtime?
Sunday January 15 2006
7:20PM 6 uplink call quality issues
5:18PM 2 Choosing an FXO card
1:37PM 1 Oooh / ahhh . . . 5 tellabs boards on ebay.
9:06AM 3 MoH trouble with latest bristuff (0.3.0-PRE-1f)
7:40AM 2 RX/TXgain on bristuff/zaptel ?
6:25AM 2 Save the Quintum before I throw it out a window....
5:16AM 3 Detecting Long PDD
12:51AM 0 passing user information problem
12:10AM 2 zaptel echo canceller preload patch
Saturday January 14 2006
11:35PM 0 codec_gsm.c:194 gsmtolin_framein: Invalid GSM data
11:00PM 1 No "native bridge" on outbound SIP channels
10:54PM 3 Reducing echo on FXS port
8:56PM 4 Ugly echo cancel, with Bristuff/Zaphfc
7:59PM 3 SIP RTP
4:41PM 0 Mediatrix windows-based setup?
4:20PM 2 Advice Of Charge (AOC) ?
4:16PM 4 tuning an x100p in Australia for echocancellation
4:10PM 4 echo tail stats
12:24PM 0 oh323 h245 tunneling not working
10:36AM 2 1.2.1 "Silence suppression is disabled" whatthehell?
10:22AM 1 I need feed back on how an Aastra VentureIP 4FXO
9:55AM 0 RE: read.what else to do ?
9:26AM 3 1.2.1 "Silence suppression is disabled" what the hell?
8:38AM 1 call file result
8:37AM 1 Problem with just one number!
8:11AM 0 RE: Mediatrix Unit Manager Express needed
7:07AM 3 rxgain/txgain test numbers in Germany?
3:47AM 2 IAX voice distortion with full upload channel / SIP ok
2:03AM 4 "Catch all" extension
1:48AM 0 class 5 softphone
Friday January 13 2006
11:03PM 0 Asterisk/Zaptel 1.2
9:46PM 2 "auto fallthrough" hangup on 1.2.1
8:14PM 1 linksys pap2 automatically connect on liftinghandset
8:09PM 1 CALLERIDNUM::3 do not working on 1.2.1
7:50PM 0 Extensions.conf error - 'MaximumInclude level(10) exceeded'
7:30PM 1 linksys pap2 automatically connect on lifting handset
6:18PM 0 configuring asterisk to send and recieve fax using hylafax
5:50PM 9 loading zaptel drivers automatically upon reboot
5:36PM 1 tuning an x100p in Australia for echo cancellation
5:07PM 2 Extensions.conf error - 'Maximum Include level(10) exceeded'
4:46PM 0 Extensions.conf error - 'Maximum Include level (10) exceeded'
2:57PM 4 PHPAGI daemon/background task?
2:50PM 1 uip200 transfer calls
2:40PM 1 ZAP Digit Timeout
2:17PM 2 ILBC to G711 transcoding experince ?
1:58PM 0 SIPDiscount credit card details
12:24PM 2 zapata.conf for non pri T1?
12:14PM 1 [Fwd: SipDiscount END-OF-LIFE announcement for old IAX2/SIP servers]
10:57AM 1 MINNESOTA: TwinCities Asterisk Users Group - Saturday 01/14/2006
10:34AM 1 queus & agents
10:23AM 1 Re: <Ben Higley> Can you send us your AGI CDR logging application?
9:54AM 2 AEL2 -- The Future --
9:40AM 1 TDMoE - best signalling method?
9:39AM 0 NOTIFY authentication
9:21AM 2 Asterisk echo & fxotune
8:28AM 1 Calls through madiatrix with incorrect disposition
8:16AM 3 FastAGI Command Execution
7:41AM 2 Use Grandstream ATA as trunk
6:58AM 0 Low Speed Software Modem
6:36AM 1 double ringing tone on asterisk 1.2
5:59AM 1 pause between queue calls for agents
5:19AM 1 Cepstral in AGI problem
4:56AM 0 R: RE: RE: Spandsp
3:39AM 1 Agent monitoring - join files
3:05AM 0 How to configure (make) Fax over IP with T38 and Asterisk?
3:00AM 1 dnid support?
2:26AM 0 resinstall
2:15AM 0 Variable
1:38AM 0 Voicemail indication fails
1:20AM 2 X-web Lite
12:54AM 2 PrimuX Cards with chan_capi-cm
Thursday January 12 2006
11:30PM 3 Asterisk Prepaid Solution
10:07PM 0 Configuration of SIP Mysql peers.
9:14PM 2 Random Disconnects
9:14PM 0 SOLVED: SIP phones can't pick up incoming call on analog (PSTN) trunk - signalling problem?
7:50PM 0 SIP phones can't pick up incoming call on analog trunk - signalling problem?
6:10PM 0 Re: Asterisk-Users Digest, Vol 18, Issue 77
4:54PM 3 D-Link announces Asterisk on Router/DSL-Modem
3:45PM 0 Sending commands to Asterisk via FastAGI
3:42PM 1 Why can Asterisk Auto Attendant pick up on firstring?
3:38PM 0 latest openh323...still compile error
3:34PM 0 latest pwlib...still compile error
2:42PM 0 trigger event on asterisk warning/error
2:33PM 1 PRI and QSIG
2:03PM 0 safe_asterisk not working?
2:03PM 1 spandsp and page orientation
2:01PM 3 linksys SPA-941
1:56PM 2 interfacing w/ a legacy InterTel PBX
1:24PM 5 [Announce] Web-MeetMe v2.0.0
1:09PM 2 Server Specification
1:05PM 2 dimensioning: Where is the CPU vs Asterisk load table
12:53PM 2 SIP phones unbeatable echo
12:50PM 2 Easy to Access Telephone Directory AGI
12:48PM 3 Bridging app
12:33PM 0 Second edition of my * book has been release d
12:20PM 0 Re: Transfer issue with a Cisco CCM/phone (Peckham, Christopher)
12:09PM 0 How to register a SIP phone on Asterisk behind NAT
11:43AM 3 Using an extension to send a linux command
11:02AM 0 (Trunk) in production
10:41AM 0 cisco as5400, sip, asterisk. cisco won't detect that the call is answered
10:33AM 2 Where do I find *asterisk-capi*
10:03AM 0 PBX making ENUM lookups
10:02AM 2 Asterisk crossed lines?
9:58AM 1 Problem with an automatic responder
9:58AM 1 R: and
9:38AM 2 Company directory not finding names... sometimes.
9:32AM 2 DTMF Issues With Asterisk 1.2 IVR
9:25AM 2 Adit 600 and echo
9:01AM 0 Transfer issue with a Cisco CCM/phone
8:39AM 6 and
7:33AM 2 conditional canreinvite
7:26AM 4 dCAp
6:47AM 2 Build Error - ZT_EVENT_DTMFDIGIT
6:35AM 0 (no subject)
6:15AM 1 No D-channels available! Using Primary on channel 16 anyway!
6:01AM 1 GSM codec problem - Windows messenger 5.1
4:57AM 0 Fwd: voip - forwarding ports
3:56AM 0 Avoided initial deadlock
3:56AM 3 read .what else to do ?
3:41AM 0 Catv ATA problem
2:25AM 0 Re: Issue calling other PBX systems using VoIPwithPolycom 501
12:45AM 2 Zaptel SVN
Wednesday January 11 2006
11:25PM 2 Dial application newbie help
10:04PM 0 Turning off 2100 Hz tone detection without editing zconfig.h and recompiling
8:46PM 0 patton smartnode 2400 with ic-t1v
8:16PM 1 chan_bluetooth problems
7:16PM 0 problems with installing app_odbcexec into dialplan
6:53PM 0 AlarmReceiver?
4:46PM 0 Remote Trunk setup
3:58PM 1 a2blling billing system
3:19PM 1 OOH323 Configuration with Cisco FSX ports, no Gatekeeper
3:17PM 1 Zaptel modules load, but Asterisk fails at s tartup
3:01PM 0 FW: Enchance Me 1.004 Released!
2:48PM 0 Enchance Me 1.004 Released!
2:20PM 1 where to get app_cepstral.c
2:06PM 21 FXS or VOIP
1:47PM 1 asterisk with an external predictive dialer
1:43PM 0 Execute command on pickup the phone
1:08PM 1 Re: setting up asterisk to handle incoming SIP URI
12:55PM 1 OT- Sangoma Question
12:30PM 1 Asterisk and Radius
12:21PM 1 Fax RX and SIP/IAX
11:35AM 1 Zaptel modules load, but Asterisk fails at startup
11:21AM 1 Issue calling other PBX systems using VoIP with Polycom 501
11:21AM 17 Nested MySQL Commands
10:53AM 0 ruby-agi-1.0.2 released !
10:45AM 0 China DID Wanted
10:28AM 4 Echo on phones...
8:59AM 0 Asterisk Manager API and ZapBarge or ChanSpy
8:59AM 0 Asterisk doesn't detect answer for some numbers
8:59AM 1 patching asterisk with tzafrir patch for voicemail permission does not work
8:31AM 1 Call Parking...
8:20AM 1 Signaling the status of the line on the phone
7:57AM 1 Better solution to mysql reconnect timeout
7:46AM 4 Why remotely reboot SIP phones?
6:59AM 3 Web based SIP client
6:58AM 1 SIP standard for flash
6:52AM 0 Connecting to a legacy PBX extension
6:45AM 1 [suse-isdn] Major Problems UTStarcom F1000 registering -- pls help
6:43AM 6 Failover Device?
6:42AM 0 Errors with bristuff-0.3.0-PRE-1e and asterisk cores
6:26AM 3 video development
6:01AM 5 Recommend Fax Hardware for T1 PRI
4:46AM 2 Transfer sounds - notifications
3:24AM 0 Incoming PSTN Calls - Can't interrupt Main Menu
2:31AM 1 Transfer to meetme on different server
2:27AM 1 Asterisk REGISTERs
2:12AM 0 does anyone know how to use 1.2 CVS setgroup in CAGI script
Tuesday January 10 2006
11:52PM 3 IAX & CallerID
10:03PM 0 New Freelance Site for Asterisk Consultantsand Those who Need Projects Done
9:10PM 1 SOLVED: Hung Zap channels connected to old key system
7:50PM 3 New Freelance Site for Asterisk Consultants and Those who Need Projects Done
6:38PM 0 register to a peer register => from database
3:40PM 0 TDM04B odd problem
2:38PM 2 TE405p -- loopback for the phone company?
2:26PM 0 The second edition of my Asterisk book is nowavailable
2:21PM 0 testmail
1:47PM 4 Help with amportal: asterisk ended with exit status 127
1:28PM 3 The second edition of my Asterisk book is now available
1:24PM 1 Still an open Seat in London for Next Weeks Signate intro to Asterisk Course
1:19PM 1 Eid Mubarak
1:19PM 0 Sipura SPA-2100 / 3000 provisioning .xml examples / xml variable list
12:59PM 1 GrandSTream 488/Asterisk
12:16PM 0 outboundproxy issue
10:56AM 1 Asterisk voicemail support
10:49AM 1 pattern mach doubt
9:58AM 0 Sacramento Asterisk Users Group
9:54AM 0 Re: 32 e1's with asterisk
9:53AM 1 Another cisco question
9:02AM 0 Besides the ISDN Guard what options?
8:42AM 0 ASTCC Voice Prompts in Spanish
8:38AM 1 avoided deadlock/channel already in use
8:25AM 4 web sip client
8:10AM 1 32 E1's in one Asterisk 'box'
8:04AM 0 Need help testing IAX based web conferencing tool
8:02AM 1 FW: Re: hangup detection
7:56AM 1 VMauthenticate always asks for mailbox
7:48AM 1 busydetect
7:37AM 0 Asterisk configuration using Database..!
7:28AM 1 Disconnected calls
7:20AM 0 Austin User Group
6:04AM 0 Live Demo of DRUID Asterisk Management Interface
5:56AM 0 Max calls & IAX2 trunking
5:17AM 1 Sip Behind Proxy
5:09AM 0 Setting up Asterisk using Mysql.
4:58AM 3 CDR problem - incorrect time
4:06AM 2 Asterisk Archives: BUG?
12:10AM 2 Problem with Action:Originate with ASterisk Manager
Monday January 9 2006
10:10PM 0 (no subject)
10:09PM 1 Second edition of my * book has been released
9:50PM 0 Call Rules
9:22PM 1 How does the PCI bus effect latency and echo?
9:20PM 1 Tellabs echo can, can someone wire mine up for $?
8:06PM 3 Incoming Zap channels not behaving as expected. Rejecting call on channel....
6:45PM 7 Presence support on GrandStream GXP-2000
6:44PM 9 Recommendations on a WiFi phone for *?
6:29PM 1 to another country
5:30PM 0 RE: Help needed ("...Broken pipe" error
5:15PM 2 mpg123 removal
4:36PM 2 TDM400 (TDM11B) configuration
4:26PM 1 how to adjust volume
3:46PM 2 ZAP - configure not to answer?
3:31PM 15 MTU and Voice Delay (latency??)
3:14PM 0 Stanaphone Configuration
2:56PM 1 OT: IAXModem in inittab causes modem to be u nres ponsive, running from console it's OK
2:20PM 8 Pri Gateway Hardware
2:08PM 0 Asterisk 1.2 - sip_buddies restrictid problem.
2:06PM 0 zaphfc and T0 ISDN to Alcatel PBX
1:47PM 2 Cisco phones 7940
1:44PM 1 Unable to connect to Asterisk
1:28PM 7 "Decent" sub-$100 SIP phone.
12:58PM 3 Problem Compiling Zaptel 1.2.1
12:53PM 1 OT: IAXModem in inittab causes modem to be unres ponsive, running from console it's OK
12:50PM 1 Zaptel errors (power alarm?)
12:46PM 1 Asterisk featdmf signalling.
12:26PM 0 Answer call waiting / flash with Zaptel POTS and VOIP
12:16PM 0 ectoolkit
10:54AM 1 SPA-841 spontaneous voicemail problem
9:52AM 1 Voicemail emailed volume
9:32AM 0 asterisk stops unexpected, no crash, but " clean" exit
9:32AM 1 ATA failover between datacenters
8:52AM 1 PrivacyManager & CallerID not passing
8:44AM 3 Same Zap channel in multiple groups
8:23AM 1 PSTN line quality
8:10AM 0 Asterisk over 3Com
8:02AM 1 Chanspy options in Asterisk Manager API
7:41AM 0 Dialtone detection help needed
7:36AM 0 Agents in 1.2.1
7:00AM 0 Snom Idleline XML
6:33AM 2 Lost my Zap's
6:06AM 1 snom programmable buttons
5:32AM 1 Is it Wildcard 406
4:40AM 2 dual IP connections
4:21AM 2 call files, fax
4:21AM 3 SNOM Hotdesking...
3:17AM 0 GradStream Budge Tone - 100 / PLease help
1:35AM 1 ISDN beronet: cannot send digits during outbound calls
12:57AM 0 SIP-SIP transfer via the REFER/NOTIFY method
Sunday January 8 2006
11:32PM 0 Re: Asterisk-Users Digest, Vol 18, Issue 46
11:15PM 0 problems with app_odbcexec
11:04PM 1 Asterisk crashing system
10:28PM 0 Re: Asterisk-Users Digest, Vol 18, Issue 46
10:19PM 0 Call forwarding for particular extension when line 1 is busy
9:20PM 1 Successfully Ported Asterisk On ARM Platform
8:13PM 1 FastAGI available?
8:05PM 1 spandsp, rxfax, TDM400/zaptel, missed frames, any help?
7:18PM 1 JiveMessenger HOWTO
7:17PM 0 DialPlan for Call Limit, Call Duration, And Group Call
7:11PM 0 FWT - LSP-350T - Asterisk
4:42PM 0 DTMF relay problem
3:35PM 0 spandsp for 1.2.1 - cannot openshared object file: No such file or directory
3:24PM 0 spandsp for 1.2.1 - cannot open shared object file: No such file or directory
1:52PM 1 PolyCom phones with blinking clock and wrong time
1:51PM 0 Fwd: Problems with R2 Support
12:39PM 3 Monitor Logged in Agent's conversation
11:41AM 1 new AMPortal and Asterisk debs
8:01AM 1 Dialogic VFX/41JCT-LS found i a drawer
6:50AM 2 Zaptel make install error
5:22AM 8 Cisco 801 and rcapi
5:08AM 2 3 PSTN lines, 3 IP Phones
4:01AM 0 2 small issues with Cisco 1760 gateway and Asterisk
2:11AM 0 Advice on fax support
12:28AM 1 Processor Update?
Saturday January 7 2006
11:37PM 0 Line Sharing or Better Call Pickup
10:12PM 0 Agi Perl Talk Time
9:27PM 1 Immediate routing on "0" (DNIS)?
9:03PM 1 Some advice on routing DID's
8:33PM 0 Up to 4 seconds delay to play prompt?
7:45PM 1 Kudzu and Zaptel Cards
7:23PM 0 cisco 8xx ISDN router
7:11PM 0 Asterisk Market Share
5:11PM 2 how to configure iax account for iaxmodem?
4:47PM 14 Asterisk Jobs
4:42PM 1 Possible bug with GotoIfTime
2:23PM 1 choppy music on hold - only on PRI PSTN
6:00AM 4 Draytek Vigor 2900 & Asterisk
5:51AM 1 Problens to link 2 * servers
5:19AM 2 wich IAX soft client allow to specify a different server port?
1:26AM 1 re: where can i find all .C files
Friday January 6 2006
8:47PM 0 --- AEL 2 --- Try it out!
6:18PM 1 Aastra 9133i and NAT: Can it work?
6:02PM 0 h323 gatekeeper??
4:39PM 3 transfer application
4:15PM 7 Fax, txfax -bizarre thing
3:56PM 1 Latency
3:30PM 0 Bristuffed asterisk 1.2.1 on Suse 10 - problem with zaphfc module
2:56PM 6 Non-PRI T1
2:52PM 3 Help Connecting server districts
2:24PM 3 Asterisk initialization
2:24PM 2 Voice mail messages aren't sent to e-mail
2:16PM 0 How to properly use GROUP
2:06PM 0 Not Able to Connect Two Asterisk Servers Usi ng IAX2
1:57PM 2 Using local\number
1:05PM 0 SJPhone with external ringer
12:57PM 2 controlling SIP subscriptions from SNOM phones
12:20PM 2 Not Able to Connect Two Asterisk Servers Using IAX2
12:06PM 2 SPA-3000 is translating vocal sounds into DTMF
11:33AM 1 Alphanumeric pattern match in extensions.conf
11:24AM 1 server recommendations
9:54AM 0 Problem with Call Monitoring
9:28AM 1 Annoying Notice Message: "Don't know what to do with control frame 15"
8:51AM 4 Problem with show channels
8:47AM 5 3RD REQUEST - Any Help Is Appreciated
8:28AM 2 Incoming PSTN Calls - Stumped
7:57AM 0 IAX2->SIP dropped calls
7:46AM 3 Announcing a call transfer
7:37AM 3 Recording Calls at the phone
7:35AM 1 How To - Building a VoIP-PSTN Gateway with Asterisk
6:53AM 0 Xs4all VoIP service - SIP config?
5:51AM 2 Call forwarding for particular extension
5:23AM 1 Problem with integrating ISDN PBX using NT mode
4:57AM 0 RE:how many calls Asterisk gateway can handle
4:10AM 2 Budge Tone-100 as a Ext in the LAN
3:27AM 3 Macro DialPlan
2:27AM 3 experience
1:45AM 0 cisco/asterisk interop issues?
12:01AM 1 Sharing SIP Info with Realtime
Thursday January 5 2006
10:50PM 0 DNIS dropping digits.
9:05PM 1 CD (call deflection) on Bristuff/zaphfc?
7:22PM 1 bristuff/zaphfc disturbing other ISDN phones
6:50PM 2 Screening incoming calls.
6:49PM 0 Call Limit[Local, InterLocal, International, Group, Time, Duration, Etc]
5:42PM 1 open h323 compile error
5:41PM 1 fax detection on TE406P
5:01PM 2 Integrating with Toshiba Strata DK40i KSU
3:36PM 0 phpagi stream_file
2:56PM 1 In search of Headset Compatible Analog Phone
2:48PM 3 TE110p and pri_cpe signalling not recognized
2:47PM 1 Re: Has anyone tried using flash() in features.conf (applicationmap) - RESOLVED
2:37PM 1 Polycom 501 netboot not working.
1:32PM 1 troubleshooting hangups?
1:31PM 0 Trailing silence in voicemail messages
1:10PM 0 PRI deadlock problem is 1.2.1
12:50PM 0 Meetme user join/leave
12:42PM 0 Bizarre Answering Problem - 2ND REQUEST
11:49AM 1 Call logging
11:38AM 1 ChanSpy via external application
11:28AM 0 Hardware Manual
10:43AM 1 UserEvent() with multiple body lines
10:19AM 8 Asterisk Debugging
9:52AM 0 Problem when i make a DATA CALL
9:41AM 0 red alarm when modprobe wcte11xp
9:14AM 5 OT: SIP aware firewalls?
9:02AM 2 Call Group Limit
8:24AM 3 Fax with Asterisk and Sipura 2100
8:10AM 0 Reading sound and recognizing DTMF sounds in eagi script ?
8:05AM 1 zaptel does not compile with kernel 2.6.15
7:50AM 1 Bizarre Answering Behavior
7:29AM 0 Re: Problem with blind transfer and Polycom phones !! more info
7:11AM 3 Remotely reboot SIP Phones ?
6:59AM 2 Asterisk CLI | more
6:15AM 1 Iaxy Ringtone
6:07AM 3 Problem with blind transfer and Polycom phones
5:35AM 1 Incoming PSTN Calls
4:44AM 0 Regular Crashes - Partially Solved
3:30AM 1 Virtuozzo - G729
3:23AM 0 Incoming calls grind to a halt
2:22AM 0 SIP/IAX softphones for use in callcentre environments
2:17AM 1 Bind asterisk to multiple IPs (reply problem)
12:48AM 1 TDM400P modules not found
12:39AM 4 Zap channel instances
Wednesday January 4 2006
11:32PM 0 RE: how many call an Asterisk gateway can handle
10:37PM 3 SIP/IAX softphones for use in call centre environments
10:00PM 0 Some WARNINGS
8:51PM 0 Has anyone tried using flash() in features.conf (applicationmap)
7:21PM 2 Cisco phone issue
7:00PM 1 New Mail Message Waiting
6:52PM 0 Sip peers got disconnect
4:35PM 2 H323 compilation Help needed
3:11PM 2 [Web-MeetM] Seeking Beta testers
2:18PM 0 Asterisk Dial problem
1:48PM 1 local exchange dialtone on ISDN/bristuff?
12:53PM 3 Email2fax big problemo
12:22PM 0 NOTICE[3589]: frame.c:128 __ast_smoother_feed: Dropping extra fr ame of G.729 since we already have a VAD frame at the end
11:59AM 5 Grandstream web configuration utility
11:27AM 0 Strange AGI behavior on Mac OS X
11:26AM 0 confusion about contexts - SER
11:24AM 2 Dial(Console/dsp) and option g doesnt appear to work
10:19AM 0 TE411P in a HP DL360 - which BIOS settings work?
9:59AM 1 M0n0Wall traffic shaping rules
9:59AM 1 freeze my box on unload
9:43AM 2 VoiceMailMain Pass Mailbox
8:43AM 0 SUSE 10.1
8:13AM 1 AMP: Losing backslash characters in config files
8:01AM 0 how do i play a prerecorded message in the middle of a conversation ?
7:41AM 1 FYI new aricle on asteisk
7:39AM 0 Unknown digits
7:22AM 2 Using *RT for HA purposes was: RealtimeMultipleAsterisk boxes, iaxusers
7:20AM 0 Entry level IP phone
7:08AM 2 suddenly iax calls don't work anymore
6:32AM 0 Can i compile Asterik on Fedora 4 x86 64 and which hardware could you support ?
6:30AM 0 Compilation of OpenH323 libraries under CYGWIN...
6:08AM 1 RxFax : Change FAX Resolution
4:23AM 0 Anybody successfully using vISDN on A@H?
3:43AM 0 USB fxo/fxs devices
3:31AM 2 remote sip client fail to register
3:28AM 1 RBT enable/disable
2:42AM 2 call monitoring from 3th phone
2:42AM 0 Can I call another S0 bus device (BRI) locally without taking 2 channels through Telco provider ?
1:57AM 1 SIP security
1:47AM 0 Is it possible to get caller and callednumberwith Asterisk Manager
1:32AM 0 MusicOnHold don't start at begin
12:19AM 2 Ominiis Asterisk TAPI driver
Tuesday January 3 2006
11:16PM 1 Detect a forwarded incoming call?
10:33PM 0 Asterisk::LDAP
9:31PM 1 Raw Hangup messages with IAX2?
9:18PM 1 IAX2 channels denoted as '(None)'
6:54PM 5 Start recording after call started
6:14PM 2 integration with Meridian/Norstar ATA2
5:51PM 0 How do you check whether a channel is active andthe number of calls
5:33PM 0 (Fwd) bridging two active calls
5:11PM 1 bridging two active calls
4:43PM 4 Problems Upgrading to 1.2.1 on Fedora 3
4:31PM 7 Dialer
4:30PM 1 confusion about contexts
4:17PM 1 Resolving timing issues with dual PRIs in a TE411P
3:26PM 0 Uvox streams
2:41PM 1 Using *RT for HA purposes was: Realtime MultipleAsterisk boxes, iaxusers
2:35PM 1 Sipbroker?
12:56PM 1 Asterisk 1.2.1 Type of Service
12:43PM 0 Re: Asterisk-Users Digest, Vol 17, Issue 109
12:43PM 3 OT: XML Content Manager for Cisco 79XX Phones
12:27PM 2 How do you check whether a channel is active and the number of calls
12:14PM 0 Experience with SetTransferCapability
12:10PM 4 iax2 wireless and Multicast
11:42AM 2 Looping Problem With Call Forwards - Do you have comments on my solution?
11:29AM 2 Heavy Static on incoming calls
11:26AM 0 cannot register whit sip client when i'm outside the PBX LAN
10:51AM 0 Recording Agent Calls
10:39AM 0 DTMF dialing
10:33AM 3 IAX termination services
10:21AM 5 Asterisk on Dell blade servers
9:48AM 0 Re IAXTEL
9:37AM 0 Sipura SPA-1001 question
9:33AM 9 FC3 or FC1 (or something else?)
8:55AM 0 Anyone heard of this company?
8:50AM 1 AEL - Using a Macro in the Dial Command in AEL
8:18AM 0 Asterisk realtime mysql connection
8:16AM 1 Problem with date & time on Aastra480isincerelease 1.3
8:05AM 2 Question on SPA-2002
7:41AM 1 E1 with CAS but no call signalling?
7:38AM 2 Problem with date & time on Aastra 480isincerelease 1.3
7:13AM 2 Problem with date & time on Aastra 480i sincerelease 1.3
7:01AM 1 Howto compile chan_h323 on macosx 10.3?
6:06AM 4 Where is the Prefix() application in Asterisk 1.2.1 ?
6:01AM 1 Three Way Calling with HFC PCI Card
5:52AM 0 outbound sip calls on asterisk
5:45AM 0 SetCallerPres
4:41AM 0 Txgain & Rxgain
3:47AM 3 cisco 7960 registration fails
2:59AM 1 AgentCallbackLogin pre-# announcement?
2:56AM 0 dhcp auto-provision spa-3000 like hardphones?
2:06AM 4 call-limit kills hints
12:44AM 0 How to get To-header and From-header display name
Monday January 2 2006
11:25PM 0 Mute a given channel.
7:52PM 1 offline g.729 transcoding
7:40PM 2 Phantom Call on hangup SPA 2000
5:06PM 1 Limit concurent calls per MSN on BRI (bristuff/zaphfc)?
4:23PM 2 Asterisk Upgrade to 1.2
2:41PM 0 Memory PB.
1:38PM 1 Festival clicks instead of sound and disconnects.
1:28PM 1 Ignoring too old SIP packet packet problems
1:23PM 6 @Home overwrites configs on startup
1:08PM 0 Re: Asterisk-Users Digest, Vol 18, Issue 6
12:32PM 5 SIP through freeBSD NAT
9:40AM 0 RE: simulator for asterisk gateway
9:35AM 0 R: Client SIP fo Windows Mobile
9:10AM 3 USB phone
9:05AM 2 Client SIP fo Windows Mobile
8:35AM 5 E911 And Routing
8:27AM 1 sip realtime and setvar
8:20AM 2 Translating between different codes
7:52AM 4 Asterisk PRI problems.
7:46AM 1 Echo after asterisk has been running for several days
7:44AM 2 Q: How to dial out / transfer calls with manager
6:13AM 2 Is it possible to get caller and called number with Asterisk Manager
6:06AM 1 CAPI unable to handle busy()
5:03AM 6 connect more the one phone to ONE sip Acoount
4:30AM 2 unable to execute call file
4:05AM 0 Happy New Asterisk Year!
2:40AM 1 Newbie Problem With Agents
Sunday January 1 2006
10:43PM 2 (Fwd) hi there
4:28PM 1 Asterisk 1.2.1 segmentation faulting!...
3:01PM 6 Cell phone dock/switch as Asterisk FXO source
12:42PM 5 Having major issues with TDM2400
9:55AM 1 Need HT488 FXO example for both inbound andoutbound.
9:29AM 1 Snom 190 occasionally NR, SIP 401
8:13AM 3 CrystalFontz LCD display
6:50AM 2 Recommendations on web interface for IT staff