asterisk users - Jul 2004

Saturday July 31 2004
11:10PM 3 Asterisk on Sparc64
7:10PM 2 480i User Feedback With Asterisk (fwd)
5:05PM 1 Hiring Setup
12:44PM 2 different pridialplan for different channels in zapata.conf
12:27PM 1 learning from the audio folks
10:38AM 3 MGCP & Cisco ATA 186 Help
8:48AM 0 Trunk doesn't work Adit 600/T100P
8:30AM 1 Asterisk does not disconnect SIP call
8:24AM 2 Asterisk scalability?
7:59AM 2 Which version of MySQL works with cdr_addon_mysql?
6:36AM 3 one extention, multiple phones
Friday July 30 2004
11:59PM 2 Sipura 3000 PSTN disconnect in the UK
11:56PM 0 G.729 <-> ZAP ?
6:59PM 3 VoiceMail Not releasing
6:33PM 1 Compiling * on OpenBSD 3.5
4:48PM 1 VoIP gateway (2 FXO, 2 FXS)
1:58PM 1 SIP connections do not hang up
1:09PM 2 asterisk-oh323-0.6.3a
1:05PM 0 Transfer call help needed
1:04PM 0 X100p / OEM X100p In MACs
11:44AM 9 Rodopi Billing
11:39AM 3 New to IP-PBX
10:58AM 0 Two different voicemail messages
10:08AM 0 AstriCon 2004 Early Bird Period Ends Saturday!
8:40AM 1 FW: Limit incoming calls to SIP Channels
8:28AM 0 Transfers- Please Help ASAP
8:20AM 3 Connecting Asterisk and Avaya Definity By E1 . Incoming work, but not outgoing
8:15AM 2 zaphfc hardware & sound trouble
8:04AM 2 Outgoing *-initiated calls from spool directory not working
7:44AM 2 audio delay over time on Zap to SIP
6:50AM 1 cisco ubr924
6:14AM 1 Connecting Asterisk and Avaya Definity By E1. Incoming work, but not outgoing
6:05AM 1 Outgoring spool
4:22AM 1 Running AGI script on answer.
3:42AM 0 Help for VoIP Gateway with 2 x FXO & 2 x FXS
3:34AM 0 Cisco PRI CLID outbound fails
2:40AM 0 How to detect the sent status of the Fax
2:18AM 5 Non standard usage of X100P card.
1:49AM 1 Asterisk+zaphfc: how to answer incoming calls from a BRI (EuroISDN)?
Thursday July 29 2004
11:46PM 2 Astricon Dev Meeting On Line
11:41PM 5 playing a sound during a call
11:13PM 0 G.729 between Zap and SIP
7:09PM 1 Re: Zaptel doesn't see remote hangup ?
6:40PM 2 chan_sccp2 testers needed
5:49PM 2 Astricon Conference Call?
5:14PM 0 DISA and notransfer/reinvite?
3:27PM 1 permit/deny in sip.conf not working
2:53PM 0 CREATIVE Critisim wanted for new application
2:06PM 1 Limit // incoming calls to Queue Agents
1:47PM 1 snom 200 and call parking?
11:13AM 2 Zultys Zip 4x4
11:10AM 1 OH323 and codec selection
10:49AM 1 Experiance of shipping into the UK
9:01AM 1 Unauthenticated calls from a specific IP
8:32AM 1 Asterisk and festival
8:30AM 0 *** Asterisk Summer News: The heat is on!
7:43AM 6 Zaptel doesn't see remote hangup ? euro-isdn
7:23AM 0 Winbond drivers
7:14AM 4 One More IP Phone for interoperability with Asterisk
7:14AM 1 (no subject)
7:14AM 2 Aastra 480e phone ADSI config
7:09AM 1 incoming caller doesn't hear rining.
6:59AM 5 Astricon Conference Call?????????
6:59AM 3 queue_log question: which endpoint was connected?
6:26AM 2 BugetTone Bug Showstopper,
6:14AM 1 Quadbri in NT Mode against PBX.
6:10AM 2 where to start asterisk sourcecode
5:29AM 0 SIP and RTP / 302 after 18x / Call forwarding after announce
5:11AM 0 New app: Consultative transfer for each phone
4:56AM 1 call center with *
4:46AM 3 Polycom IP Soundpoint 600 & early dial
3:52AM 1 PSTN tone simulation
3:38AM 0 ParkAndAnnounce command !!!
3:21AM 3 IAXy config samples
3:15AM 0 Asterisk PID file
2:50AM 10 Asterisk GUIs at Astricon * REMINDER *
2:04AM 0 Compatible E1 card
1:57AM 1 SIP Outbound Proxy Support
12:35AM 1 ToS flags for VoIP
Wednesday July 28 2004
11:10PM 0 sip phone, receiving calls but not placing any call
10:58PM 1 rate-engine loading error
10:55PM 1 Please share your Solaris experiences on the Asterisk Solaris Wiki page
9:36PM 1 using round-robin dns for sip registrations
8:38PM 2 call waiting, * and FXO
8:29PM 4 X-Lite to Asterisk through NAT?
7:33PM 3 faxing
6:57PM 1 false busy using sipura spa-3000 with asterisk on solaris
5:06PM 0 X101P problem with latest FreeBSD zaptel drivers
4:53PM 1 Reverse Battery Disconnect Supervision in X100P or TDM400P FXO
4:53PM 0 D-Link DG-104SH H323 problemm
4:34PM 0 broadvoice/asterisk incoming calls problem
3:40PM 0 SipTone 4 Sale...
3:13PM 3 Workaround for BroadVoice and possibly others...
2:59PM 1 is chan_skinny broken?
2:54PM 0 "Re:" is ok
2:29PM 2 Collect recording before sending to extension or queue
2:12PM 4 Cisco 7960 backlight and list etiquette?
1:13PM 1 Zap hanging up others zap.
12:48PM 0 source for zultys zip phones?
12:08PM 3 Changing Transfer key
12:02PM 1 ParkAndAnnounce Doesn't Release when Call is Picked-up
11:52AM 3 Problems with * and Grandstream Budgetone 100
11:44AM 3 New Zealand DIDs
11:40AM 0 VoiceMail Problem or bug?
11:20AM 3 Asterisk, PBX, VoIP and PRI
10:56AM 0 Unix ODBC Segmentation Fault
8:43AM 2 Music On Hold - not working for me...
8:16AM 4 MS SQL & Free TDS
7:52AM 0 Asterisk-and-MacOSX News
7:31AM 0 only one call at the time
7:30AM 2 Desired Install in MotorHome
6:36AM 2 Rate Engine Compile Error
6:24AM 8 Best Linux for Asterisk
6:23AM 5 Trouble compiling asterisk-addons MySQL
6:13AM 3 MGCP & Caller ID
5:35AM 2 Asterisk voicemail from mysql no longer working
5:22AM 1 Access voicemail from Cisco 7960
5:22AM 0 Commercial Asterisk Support
5:09AM 1 Outgoing works, incoming doesn't...
4:27AM 1 Problems Compiling Asterisk-oh323-0.6.2
4:27AM 3 shan:Needed help
4:23AM 2 IAX transfer bug in last CVS ?
4:23AM 0 Shan:Help in configuring Dialplan
1:04AM 1 problems with digium cvs????
Tuesday July 27 2004
11:34PM 0 E-mail address for DIAX support has been changed
9:44PM 0 Example Polycom XML applications, anyone?
8:05PM 1 re: asterisk/incoming calls attn:James
8:04PM 0 converting gsm file to g729 format
7:50PM 6 Successfully Using $135 Avaya sip phone
6:43PM 1 Broadvoice - incoming calls problem
6:10PM 2 Polycom IP-600 leasing?
5:53PM 6 zt_pri_error: PRI: Warning: unknown/inappropriate protocol discriminator received (00/0)
3:52PM 5 Has anyone tried using a Sipura-3000 as an FXO device for *?
3:27PM 1 Dial out problems with Digium TDM400P card.
1:31PM 1 Problems connecting xlite phone
1:01PM 6 Cisco 7910 Setup
12:57PM 0 Local voice feedback
12:52PM 1 "Broadcasting" Calls?
12:11PM 0 Touch Tones not working after call is made
11:40AM 0 Strange RTP audio errors on console
11:38AM 0 Strange I4L troubles
11:27AM 1 GrandStream BudgeTone 100 Firmware?
10:27AM 2 Open for beta testers - free calls in us/canada
10:01AM 0 Comparing * with SIPxchange
9:17AM 2 g729 + GSM + g723
9:15AM 0 Re: [Asterisk-doc] New Hardware Support
9:09AM 0 Off Toppic-ish Telephony question
9:03AM 0 switchhook flash / link
8:03AM 0 Re: Nat...again...
6:51AM 0 safe_asterisk and odbc
6:40AM 2 Problems with variables
6:25AM 5 User-Oriented Management of Asterisk
5:50AM 1 VoicemailMain Issues
5:32AM 0 h323.conf - host= multiple ip addresses
5:20AM 1 asterisk <-> stanaphone?
4:30AM 1 Puzzled by CapiCD (call deflection to mobile phone)
4:14AM 5 sip over h323
3:44AM 1 Hook-flash timing
3:27AM 0 and a eNUM
3:27AM 2 Enum
3:12AM 0 Re: Nat...again...
3:07AM 3 2 cards
2:06AM 6 Asterisk to CCM
1:56AM 2 Using rxfax over SIP
1:49AM 0 How to allow softphone to dial thru with full SIP URI?
1:27AM 3 Pickup an unanswered line
12:59AM 2 Transfer / Park issue
12:20AM 7 broadvoice/asterisk
12:14AM 2 Decent Asterisk Compatible VoIP HardPhone - Australia
Monday July 26 2004
10:57PM 6 New Beta version of Grandstream Firmware
10:20PM 0 Suggestion on reporting provider problems (was [RANT] Boadvoice...)
7:54PM 0 rtp.c:487 ast_rtp_read: Unknown RTP codec 72 received
7:49PM 1 voicemail+g729
4:32PM 0 Timing pips ?
4:10PM 4 Pickup zap channel already in use?
3:44PM 0 callgroup pickup
3:19PM 1 Integrated Networks IN1002 SIP Phones?
3:16PM 0 Sample extensions & SIP Conf files
3:10PM 1 Voicemail from MySQL and Directory
2:07PM 1 Global Variables Scope
2:03PM 5 Upgrade from Altigen
1:59PM 4 IRC Etiquette
1:07PM 2 Broadvoice problems again Attn: James
1:07PM 4 Asterisk for a large scale implementation
1:03PM 4 Echo in asterisk phones.
12:34PM 0 Sip usage and busy? (BroadVoice related)
11:47AM 2 IAX2 to IAX2...i'm obviously an idiot!!
11:03AM 3 ResponseTimeout, Straight to operator?
10:54AM 1 drivers, kernel 2.6 and distribution
10:42AM 3 Source for 9-911 Labels to attach to phones?
10:00AM 2 Display and UUS IEs on PRI - Q.931 question
8:59AM 0 Can't dial SIP<->EuroISDN (HFC-S based PCIISDN card): Unable to create channel of type 'Zap'
8:35AM 0 Voicemail Hangup detection issue
8:29AM 1 snom 105 Attended Transfer does not work
8:14AM 1 Nat...again....
7:35AM 6 Can't dial SIP<->EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'
7:18AM 0 Having * show lines in use on hard phones.
6:48AM 0 PacketCable & Asterisk
6:42AM 0 How to use SER together with Asterisk and to preserve Asterisk features like Dialplan, etc.
6:30AM 0 Problem with T1 signalling mode
6:30AM 0 X100P Red Alarm after echo on the line
6:26AM 1 Feature question
5:40AM 5 GrandStream CallerID
5:36AM 0 Can Asterisk register through a sip proxy with a username incorporating a different server?
4:51AM 0 Call Forwarding/Redirection?
4:17AM 0 Astricon news :: The conference agenda now published
3:03AM 0 IVR help
2:40AM 3 TE405P and E1
1:16AM 1 bri-stuff NT mode
12:58AM 0 Asterisk and AOC?
12:30AM 1 Nested/cascading "switch" statements: possible?
Sunday July 25 2004
10:04PM 1 Asterisk CDR & UniqueID
7:59PM 1 Busydetect problems
7:55PM 17 Broadvoice problems again
7:15PM 1 pound key tone generated after call answered?
5:55PM 1 Can not make progdocs
5:52PM 1 X100P Inbound Issue
4:47PM 2 Incoming SIP gateway context?
4:43PM 1 Web Based Admin Interface
3:33PM 1 how do I play congestion tone when Zap channels are full?
11:17AM 0 trunk line usage report
8:24AM 2 which DB to use? or Why Berkley DB vs. MySQL, etc
7:23AM 3 FXS vs. FXO
1:09AM 1 sip ua---------asterisk-------h323gw
Saturday July 24 2004
8:52PM 3 Help with T1 PRI Configuration
8:46PM 5 VoiceMail Group Broadcasting
8:02PM 4 Layer 3 VPN Question
4:45PM 1 Autologout of dynamic agents
3:22PM 2 John Vogel
11:25AM 1 Do not buy "Asterisk for Small Office Setup"!
11:03AM 1 Please help I fear I have missed something very important! but what?
10:43AM 1 Play CD!
7:44AM 1 Hack to make * -> (H323) -> CCM -> IOS GW work
7:26AM 0 PBX functions and different channels grouping
7:25AM 1 Attendant configured AutoAttendant
7:23AM 0 Should configured devices show up with "show channels"
7:06AM 2 Need to block incoming collect calls
6:50AM 0 Question when using a Cisco as a PSTN GW
6:33AM 2 yes shady dial running now but not dialling
3:31AM 6 h323 to SIP Server Load
Friday July 23 2004
10:44PM 3 "Asterisk for Small Office Setup"
10:44PM 0 Cisco 7940 hook-flashing blind xfer.
4:53PM 0 Deltathree and Go2call registration ? Anyone ?
4:11PM 0 SIP302 Redirect Problem (Moved temporarily) fixed, Asterisk working now with Nikotel!!!
3:06PM 3 Wildcard T100P in 1U
2:51PM 2 Reverse number lookup
2:40PM 2 Queue Statistics Line
2:32PM 6 Robbed Bit T1 Configuration
2:19PM 0 Reinstalled FRom CVS - Things are really different now...
1:34PM 0 Pipecall problem
12:26PM 1 chan_alsa record problem
11:54AM 1 Norstar ATA2 signalling protocol?
11:15AM 1 addmailbox
11:02AM 1 No channel type registered for 'ZAP'
10:07AM 0 AstriCon Update: Very Low Priced Ground Transport Available
10:01AM 0 Problems calling a phone number through a X101P card
9:21AM 3 DTMF stops working w/ Voicemail
9:00AM 4 Doublehash transfers
8:33AM 4 hang up when going to voicemail
8:12AM 0 MGCP and one-way audio
7:50AM 0 cisco 7940 audio problems to PSTN
7:34AM 0 using the older iax1 protocol
6:53AM 6 Priorizing of packets
6:44AM 3 Status of Q.SIG on Asterisk?
6:09AM 0 * poll
6:08AM 4 still can't load oh323 - Are we not supporting H.323 any more?
5:03AM 0 SIP - Cancel request fails with "481 no such call"
4:13AM 0 qudBRI and transfering calls with the latest RC2.
3:55AM 0 ringing tones for E100P (like early B3 in chan_capi)
3:43AM 3 Grandstream Budgetone 101 channels don't disappear on hangup.
2:27AM 0 AW: Large Enterprises using asterisk
1:12AM 2 Cisco 12sp firmware... Anyone got it???
12:37AM 0 Configuration help
Thursday July 22 2004
9:35PM 2 NAT + iConnectHere Broken in 1.0RC1
8:16PM 2 D-Link 1120M
7:35PM 0 Re: VSP? Looking for advice
4:27PM 0 Outbound Dialing Format
2:19PM 4 VSP? Looking for advice.
1:49PM 0 Connecting more Asterisk Servers in Cluster to works as one IP PBX
1:06PM 0 still can't load oh323
11:09AM 6 D-Link DPH-80S vs *
11:07AM 7 Asterisk and Linejacks
10:53AM 0 Future installation questions - what do I need
10:19AM 1 app_dbodbc URGENT
10:05AM 1 RAID/SCSI/IDE/SATA and a TE405P (or T100P) c ard. Should I expect problems?
9:22AM 1 How to calculate the price for Asterisk based Solution
9:18AM 2 Nortel SL1 protocol and *?
9:04AM 1 no incoming pstn ring tone
8:49AM 0 (no subject)
8:39AM 0 Call Quality - Factors and Config Values
8:38AM 0 New astGUIclient version released 1.0.3
8:35AM 0 Application Hangup not hanging up, possible dialplan cockup?
8:09AM 1 Faild Echotest
7:41AM 1 Asterisk-oh323 on fedora Core 2 - Anyone has a working install?
7:06AM 1 Echo Canceller Wiring (Tellabs.. HOWTO..?)
6:11AM 1 Voicetronix Openswitch
6:08AM 0 Re: Astricon costs
6:07AM 1 Can anybody recommend a good T1/PRI provider ?
6:02AM 0 ODBC.conf
5:57AM 1 Daytime - Nighttime
5:53AM 1 Can anybody recommend a good T1/PRI provider?
4:31AM 8 debian install zaptel
2:45AM 0 Re: h323ep----gnugk-----astersik------h323ext
2:45AM 0 ZAP Channel doesn't hang up - X100P
2:42AM 2 error while compiling asterisk-oh323
2:20AM 1 Symbian Softphone
1:48AM 1 Sip -> H323 using oh323 and G729
12:08AM 0 SIP Providers Setup.- Go2call, deltathree, etc...setup files ? any
Wednesday July 21 2004
11:12PM 0 Asterisk sees inbound call, but won't answer
10:28PM 0 X100P only dials a single digit
10:24PM 11 Large Enterprises using asterisk
9:42PM 2 Anyone heard of BroadVox direct?
9:15PM 2 TDM04B Dead?
5:44PM 2 Anyone else having Broadvoice Problems?
4:08PM 0 sangoma AFT A101 works with Asterisk?
3:20PM 6 Astricon costs...
3:14PM 1 Install problems
1:52PM 1 Zaptel - delay before dialing last DTMF digit?
1:49PM 0 Adding another channel to a Dial() already in progress
1:24PM 2 Compiling Asterisk (zaptel) failed on Debian 2.4.18-686
1:00PM 1 h323 call flow fails
12:57PM 0 Mac OS X installer for Asterisk - Missing Files Patch Now Available
12:33PM 1 Error in compilation [URGENT].
12:23PM 0 go2call setup ?
12:19PM 3 X100P panic
12:14PM 5 RAID affecting X100P performance...
12:14PM 2 ENUM lookup help
12:04PM 3 Help needed for Seting Up Asterisk
11:58AM 4 Future installation questions - what do I need?
11:58AM 0 NAT table expiration
11:48AM 1 roblems with Junghanns QuadBri
11:45AM 3 Asterisk Server gives 403 forbidden
11:45AM 0 SIP Hard Disconnect Detection
10:58AM 1 TDM400 dropping loop current 10 seconds after answer
10:37AM 1 E1 card with R2
10:24AM 3 echotraining on T1 circuits
10:09AM 2 fonction Getvar
9:46AM 1 bare minimums
9:36AM 0 AW: Asterisk RC1 and bristuff
9:14AM 0 S100I-IAXY
8:48AM 1 rxgain - txgain values
7:55AM 1 Bri solution for Asterisk
7:53AM 0 extensions.conf variable declaration
6:43AM 1 chan_capi-0.3.4b and asterisk last cvs
6:13AM 0 Cisco 7960, multiple registrations, and NAT
5:58AM 2 Caller based routing
5:38AM 0 Queue Monitoring
5:04AM 2 queue stats
4:25AM 0 chan_capi busydetect
4:07AM 0 Asterisk RC1 and bristuff
4:00AM 1 libr2 completion staus
3:59AM 0 music during conversation
3:17AM 1 Digium card x100p
3:05AM 0 Voicemal error
2:47AM 0 Senao SI-7800
12:51AM 0 Zaptel Hung Up on x101p and cisco analog line
12:27AM 0 Integrated management tool?
Tuesday July 20 2004
10:51PM 2 No Ringing.
10:39PM 1 * CLASS codes
9:09PM 0 libr2 status
7:03PM 3 # Transfer Context
5:54PM 1 Latest CVS (7/20/2004) stops answering SIP calls after 5 min
5:53PM 0 Festival 2.0
4:32PM 0 BRI dead in USA?
4:02PM 1 Voice Pulse And Incoming DID
3:30PM 1 hold then transfer...
3:01PM 0 Linux sparc64 conferencing?
2:50PM 2 SIP Registration issues
1:58PM 0 Phone numbers in SPAIN
1:12PM 1 RC1 and advanced voice mail options
12:55PM 1 Strange behaviour using 7960
12:01PM 0 R: Dial plan errors
11:04AM 0 Error on Zaptel install
11:02AM 0 Call Queue: strategies and penalties
10:54AM 2 FREE (305) and (786) termination. Anyone interested?
10:54AM 10 Installing X100P
10:31AM 2 question regarding Asterisk. X-Lite, and firewall
9:34AM 3 how to configure my cisco 7960?!
9:29AM 1 Up to date?
9:14AM 1 DID VoIP trunk provider for metro Chicago, LA and/or Orlando.
9:01AM 1 quadBRI
8:36AM 10 PRI dead in USA?
7:43AM 0 received a call waiting CONNECT_IND
7:26AM 1 SIP 2 ISDN
7:20AM 1 Sound files - uncompressed versions available?
6:58AM 3 New CVS version
6:41AM 0 Asterisk CVS compile error YDL 3.0.1
6:39AM 1 Random Dropped Called
6:29AM 0 NAT problems with ZIP 4x4
6:26AM 4 Wireless SIP Phones
6:24AM 0 Modem chipset Intel
5:23AM 1 what is :
3:47AM 0 Grandstream transfer button
3:44AM 2 Calls from H323 to SIP unsuccessful
1:39AM 1 chan_vpb
Monday July 19 2004
8:39PM 1 isdn cli
7:36PM 2 codec translate
6:45PM 2 callparking vs calltransfer
6:26PM 0 Setup for Go2call ? Or any SIP provider using phonejack or linejack g729 g723
5:53PM 6 Problem Starting RC1
5:33PM 0 Hospitality Industry
5:12PM 11 Echo on a PRI
4:34PM 0 MWI - Config Stupidity or Notify Issues?
4:20PM 1 Unable to launch asterisk and connect to console. ?????
3:19PM 0 (Asterisk-Users] Affordable SIP Phone - Stiil a Myth?
2:33PM 0 Asterisk RC1 and advanced voice mail.
2:22PM 0 Cant compile Zaptel at all
2:09PM 1 Occationally SIP ext apparently is busy and goes to VM
2:04PM 0 dropping g729 frames
2:03PM 2 Affordable SIP Phone - Stiil a Myth?
12:33PM 6 collect calls
11:24AM 3 PSTN gateway implementation?
11:00AM 0 CTR21/CTR37 Gigaset phones and GS HT286
10:43AM 6 Codecs - Advantages
10:17AM 2 Mac OS X installer: missing files fix
10:09AM 1 Flash Zap trunk from a Sipura
10:02AM 1 MAC OS X Panther :?
9:33AM 0 POE Switches and QOS
9:29AM 5 Cisco 7960 SIP V6 and distinctive ring.
8:26AM 1 Channel banks, voicemail, and immediate=no
8:19AM 2 BroadVoice problems?
7:40AM 5 Cheap PoE switches/injectors?
7:01AM 4 FATAL: Module zaptel not found.
6:57AM 1 uip200 clips audio prompts
6:52AM 0 AGI Dial, Extension dial SIP Loop
6:31AM 1 Help w/ SIP response 481
6:07AM 2 Unavailable/Withheld identification
5:28AM 3 Numbering Plan and Siemens EWSD
5:13AM 4 TDM400P Internal Extenion Config
5:08AM 0 *** Asterisk Sun/Monday News: Time to download, Scotty!
1:53AM 0 ast_data compile problem in asterisk CVS Asterisk CVS-HEAD-07/14/04
12:46AM 0 sip-h323
Sunday July 18 2004
11:30PM 0 Loud echo with answer before dial
10:24PM 0 GR-303 and _FXS_ support!
9:51PM 0 Asterisk Control Script
9:40PM 0 GUI based.. or ??
7:58PM 3 Adding voice mail box
7:41PM 1 TE405P
7:09PM 3 LAN Switch w/ QoS
6:38PM 2 call progress detection
6:13PM 1 CID, international style?
5:07PM 0 ChanIsAvail issue
4:53PM 18 Polycom IP 500 Voicemail
4:33PM 0 chan_capi-0.3.4a
4:33PM 0 Help. New SIP hardphone.
4:08PM 1 chan_capi won't compile
2:52PM 4 Brain-dead Grandstream BT102?
12:57PM 1 Help! Unable to create channel of type SIP.
10:50AM 4 New G.729 codec and VLANS
8:16AM 4 quadbri NT_mode S-Bus Problem
8:12AM 1 PhoneGaim?
8:12AM 6 PSTN Gateway X101P
7:45AM 0 Asterisk and zaptel on Fedora Core 2
6:36AM 2 Hotline
5:59AM 3 zaptel issues
5:13AM 1 sent into invalid extension 's'
5:13AM 4 Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
4:48AM 1 Asterisk NAT spa-2000
12:12AM 0 Polycom IP 500 Phones - Button Assignment
Saturday July 17 2004
9:57PM 0 sip-oh323
8:06PM 1 Using a group variable for a groupofextension to dial
7:05PM 1 Using a group variable for a group ofextension to dial
5:30PM 2 Parking renamed to feature in 7/17/04 CVS
4:24PM 1 Using a group variable for a group of extension to dial
4:18PM 1 overlapping extensions
2:27PM 1 Asterisk at OSCON?
1:21PM 1 voicemail broadcast feature
11:09AM 6 Mac OS X installer for Asterisk
9:46AM 1 Question about Asterisk Installation
9:31AM 0 Re: asterisk echo problem ever go away???
7:12AM 0 Updated RPMS for Asterisk-1.0 RC1
3:51AM 1 MYSQL_FRIENDS and IAX problem
1:35AM 1 Wo uses H323-phones with asterisk?
1:32AM 0 RC1 Mirror, was Re: Asterisk-1.0 RC1
1:27AM 3 chan_capi: sending incoming calls to different contexts
1:16AM 4 E100P and Colt Telecom (Europe)
Friday July 16 2004
11:17PM 8 Asterisk-1.0 RC1
8:23PM 1 Pressing digits on SNOM phone results in letters on display
7:36PM 0 I already have a VAD frame?
7:31PM 0 Transmitting a hook-flash down an E&M DS-0?
6:43PM 1 Need configuration sample for VoIP(SIP) -> PSTN Gateway
4:41PM 7 7960 Dynamic DNS?
3:11PM 3 PSTN/phone/FXO/FXS cabling issue
3:05PM 0 Sipura 3000 user guide is now available
2:28PM 1 SIP register and unregister events via Manager API
2:13PM 1 Looking for WiFi phone recommendations
1:34PM 6 Asterisk + NEC Electra Elite IPK Integration
12:16PM 0 outgoing calls over SIP
11:54AM 0 Patch to test: Mailbox path changes
11:42AM 1 Problems with festival
11:31AM 0 Cisco Call Manager and Asterisk (AVVID) - Comparison
11:02AM 0 zaptel red alarms with e&m wink
10:54AM 1 Patch to test: Never say goodbye to an agent :-)
10:35AM 0 SIP module with radius authentication support
10:28AM 1 SIP channels UNKWN
9:58AM 0 Hardware platform / features
9:46AM 1 and relaymail to a smtp server
9:43AM 3 Echo problem update - POSSIBLE SOLUTION
9:14AM 0 How to configure Asterisk as a VoIP(SIP) to PSTN Gateway?
9:12AM 0 Path to test: Sending HTML virus, no, VOICEMAIL!
9:10AM 0 Path to test: Czech localization
9:08AM 0 Patch to test: Dynamic queues
8:10AM 1 3COM 3102 SIP Phone
8:09AM 0 How to handle a macro that dials both international (011) and national
8:03AM 1 DTMF issue --help
7:48AM 2 where to sign up for fwd
7:36AM 2 Offhook tone in channel OSS/dsp
7:28AM 0 RESOLVED: 'Dropping voice to exceptionally long queue on IAX2'
7:18AM 1 Feature Group D
6:57AM 1 MWI on Grand Stream ATA-286
6:47AM 0 Problem with asterisk and zaphfc
6:42AM 1 Astersik with g729 and 120 active channels with digium card ISDN PRI
6:02AM 1 VoiceMail fails to delete messages after emailing them
5:56AM 0 Asterisk sales materials
5:45AM 0 Subject: Re: SoxMix - Fails to Execute
5:40AM 1 Using Asterisk with fiber optic
4:54AM 1 iaxy server issue
4:09AM 1 When does the PUC become an issue?
3:30AM 1 Anyone experience with early dial?
2:19AM 1 Line Display
2:02AM 7 some questions on uniden uip200
12:03AM 2 Flag Bad PRI Channel
Thursday July 15 2004
11:59PM 2 sip phone configuration problem
11:36PM 0 fax still fails, ideas sought! Re: rxfax/spa ndsp fails to decode
11:23PM 0 app_rpt
11:00PM 0 What happened to ?
7:24PM 0 Unable to create chanel of type SIP
7:14PM 3 G.729 codec doesn't seem to work *even* after installing the license
5:21PM 0 accountcode problem
5:20PM 0 sphinx2 how-to
5:17PM 1 "Reverse Hold" feature prototype...
4:41PM 1 Spectrum Analyizer software
4:17PM 3 SIP to H323 call timeout
4:13PM 3 Current echo status?
3:36PM 4 Kernel panic with two Fritz cards
2:48PM 1 Call Queues help
2:35PM 3 Important note for AGI with PHP newbies
2:34PM 1 Fedora Core 2 softphone
2:15PM 2 SoxMix - Fails to Execute
2:14PM 2 Cisco phones and Messages and Forward ToVM keys
1:50PM 1 Polycom IP 500 and Asterisk
1:19PM 1 bristuff 0.0.3 ?
12:54PM 4 freenode #asterisk IRC channel identd problem
12:19PM 0 Hangup FXO line detecting & PSTN Tone Signals Detecting
12:05PM 8 Directory
11:48AM 6 [OT] The stories people tell to support.
11:21AM 3 Database App
11:13AM 2 Really long first ring, then normal
10:12AM 0 SIP registry forwarding top SIP connections
9:45AM 0 astcc database configuration
9:38AM 1 Using SIP phone to dial out using ISDN ?
8:27AM 0 Grandstream Budge Tone 100 No Ringtone
8:23AM 4 ZyXEL 2000W
7:23AM 17 VoicePulse changes
6:01AM 1 random disconnect with hfc ISDN card and sipura
5:21AM 0 TE4XXP Signaling
5:01AM 0 DTMF and Voicemail issues
2:53AM 1 zapras - and kernel ??
1:27AM 1 *, NAT & STUN
1:03AM 1 Problem loadin oh323 solved
12:27AM 0 Incoming SIP calls as asterisk@...
12:11AM 2 Small setup
Wednesday July 14 2004
11:54PM 0 Originate to IAXComm problem once again
9:35PM 1 looking for 802.11 SIP phone
7:57PM 1 why ata stop working after 10 mins after registering from mysql
7:56PM 0 Voice Numbers in Spain (SIP)
6:10PM 1 oh323 dial structure and oh323 debug?
6:03PM 0 changed ip now * demo call not working.
5:19PM 1 Noob Service Provider T1/T400p physical interfacing question
4:58PM 0 Voange with asterisk settings
3:52PM 2 different port setting
3:32PM 3 Vonage working with asterisk
3:16PM 0 Errors connecting to FWD
2:56PM 8 Directed Call Pickup
1:20PM 1 SMDR/CDR - Asterisk integration - Clarification
1:16PM 2 Chan_Capi 0.3.4a error
1:02PM 0 Status of ALERT_INFO and Cisco 7960?
12:02PM 0 CHAN_H323 bridge SIP no audio
11:41AM 13 Where can i get an UK SIP account with UK number?
10:30AM 1 Digium X100P card to a brazilian analog line
10:17AM 1 Starting up considerations.....
10:15AM 2 Increase transmit volume on IAX channels
10:13AM 0 random red alarms with t100p
10:06AM 0 Windows Messenger Problem
9:54AM 4 can you trust CDR for billing information?
9:50AM 0 Distinctive Ring availability on an IOS sip gateway
9:43AM 8 spa-3000 review?
9:26AM 2 SIP only
9:21AM 0 asterisk as a SUA together with SER
9:09AM 2 RE: [Asterisk-User] asterisk compile problem
8:09AM 0 who knows asterisk/libpri source code interaction
7:51AM 1
7:30AM 1 zaphfc ptp & blocked incomming calls
7:19AM 5 Getting an USA phone number
7:09AM 1 error 1 and 2 during make of asterisk with SUSE 8.2 and 9.1
6:51AM 3 Using a DNS name for externip in sip.conf
6:43AM 1 Onhold Music
6:14AM 2 GSM adapter + Automatic Routing function
6:12AM 0 Having serious problems with AGI
5:25AM 1 invalid extension -> missing the original ${EXTEN} value
5:01AM 5 ACD Issues
4:53AM 0 forward_msg: no 2nd via found in reply
4:46AM 0 ISDN PRI "calling number" for outgoing calls
4:27AM 1 Questing regardning dialplans on a Cisco 5350
4:08AM 3 Voicemail/autoattendant not working
12:25AM 0 Audiotel - Premium, call proceeding ?
Tuesday July 13 2004
10:24PM 2 ASTCC: Asterisk Calling Card Solution
8:47PM 1 SIP authentication bug with insecure= lines?
7:57PM 0 Unable to place more then 1 call in or out.
7:50PM 1 0h323/ h323-registration
7:29PM 3 Bounty! For help with echo cancellation code.
5:46PM 1 asterisk compile problem
3:31PM 1 Re: Applications of TDMoE "critch"
3:31PM 1 Problem with multiple phones behind firewall
2:35PM 0 Meetmee feature - Possible.
2:34PM 0 Looking for US ISDN card...
2:26PM 1 bad sound quality, also the ringtone
1:28PM 0 RE: Problem with multiple phones behind firewall
1:24PM 0 Registration Refresh Per REGISTER line in sip.conf
1:09PM 0 will digium hardware and asterisk function in asia (korea)?
1:08PM 0 Upgraded to CVS HEAD 7/12/2004 and calls very bad now
12:58PM 2 Swiss IP10S using SIP
12:46PM 1 fax still fails, ideas sought! Re: rxfax/spa ndsp fails to decode
12:36PM 3 Bandwidth requirement with G729A
12:26PM 1 Mailing to the list
12:22PM 4 Rotary phones? (No, I'm serious)
12:19PM 2 How to 'Dial' a Parked Call ?
12:08PM 1 Asterisk ML archive down?
11:55AM 0 One way audio when the BT-100 is behind Firewall
11:42AM 1 integrating ser with asterisk
11:11AM 0 Any way to change ring back behavior for call park?
10:50AM 0 Echo, DTMF, issues
10:36AM 1 codec issues between linphone and *
9:41AM 0 WARNING: Deprecated incominglimit and outgoinglimit
9:37AM 0 Asterisk and Swissvoice
9:19AM 0 'Dropping voice to exceptionally long queue on IAX2'
9:06AM 5 WiSIP and Zyxel Prestige 2000W
9:00AM 1 Broken pipe in remote exeute
8:49AM 0 "unclean hangups" can I turn off hook flash?
8:45AM 0 zaphfc does not indicate congestion!?
8:44AM 1 G729A and GSM - newbie question
8:14AM 2 Help Needed in configuring Cisco 7940
7:46AM 1 Dial Fail - Send Email
7:46AM 2 Local Call Problems
7:26AM 3 Cann't load oh323 0.6.3a
7:15AM 0 Possible Asterisk Notify Bug
5:38AM 2 IAX2 calls through
5:33AM 1 Asterisk don't listen to my phones
5:24AM 1 HFC-S card and Unable to create channel of type 'Zap'
5:11AM 1 Meridian Option 11c Asterisk Expert Needed
5:07AM 0 X100P ring/off-hook in strange state 6
5:03AM 0 Local Calls Not Working
4:30AM 3 Applications of TDMoE
4:26AM 0 zaphfc TE -> NT problems
4:19AM 0 how to use direcotory from Voicemail
4:07AM 1 caller id problem on incominc call to x100p
12:27AM 1 segmentation fault on asterisk startup
12:07AM 2 SIP simultaneous registry possible workaround (was Re: New Asterisk bounty: SIP simultaneous registry)
Monday July 12 2004
8:37PM 0 No Compatible codecs? Got license
8:21PM 0 "Follow Me/FInd Me" functionality?
7:50PM 1 Asterisk as plain PABX in call centre
6:57PM 0 Using a SwissVoice IP10S with Asterisk
5:57PM 2 Oz ISDN
5:53PM 0 Announce of Cisco 7914 Operator Console Support in chan_sccp
4:54PM 1 Manager help
3:53PM 1 Problems with chan-capi
3:13PM 0 Running SIP on multiple ports
2:53PM 3 HELP: One way audio... continuously and randomly
2:37PM 1 incoming calls on Cisco 7960
2:21PM 3 notransfer
2:02PM 1 No voice bet/ ext with Polycom
1:54PM 1 Errors when compiling app_radius
1:47PM 1 CID not appearing via X100P
1:31PM 1 Problems Compiling asterisk-oh323 0.6.3a
12:59PM 3 voicemail setup guide?
12:45PM 3 Audio filters (was: feature - VM gain adjust?)
12:43PM 4 call Intrude
12:39PM 0 "help"
12:26PM 0 GnuGK + SIP Provider + Asterisk
12:17PM 1 rxfax/spandsp fails to decode
12:06PM 5 Digium Cards in Boxes without Power Connectors
12:05PM 6 Asterisk crashing with no indication why.
11:53AM 2 OH323 and G729
11:42AM 1 SIP client to IAXTel 800/888/877/866 dialing thru Asterisk
10:54AM 0 Transfers (sip or asterisk "#' base) broken in certain scenario
10:33AM 0 IP Soft Phone with FAX
9:58AM 1 asterisk T1 question
9:50AM 1 Cheap ISDN interface + Asterisk what to choose?
9:39AM 1 Sort of OT: Recommended USB handset for use with iaxComm?
9:09AM 1 zaptel debugging tools
9:00AM 3 dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 2 frames
8:33AM 0 SIP => PSTN Pri Causes
8:28AM 0 Problem with Capi Channel
8:15AM 0 IAXy prov. using DNS
8:03AM 2 Indications missing on Cisco FXO -> ATA-186 (SIP)
8:01AM 0 GnuGK + Asterisk + SIP Provider
7:51AM 8 Gogoif with variables acting funny?
7:41AM 0 DTMF warning message in log while using SJPhone
7:28AM 0 Cisco Remote-Party-ID / Bug #2012
7:19AM 1 ZapBarge and SIP Channels
7:02AM 0 Using MD5 to encrpty PIN
6:24AM 1 R: How to make * don't strip the leading 0
5:57AM 3 How to make * don't strip the leading 0
5:36AM 1 PRI numbering plan
4:36AM 1 E100P and T1 channel banks
4:19AM 1 Can I hear voice messages from diax phone button directly ?
4:18AM 3 permission problem
1:54AM 0 Problem with character encoding in SIP channel (ISO vs. UTF-8)
12:57AM 0 RE: MeetMe Improvement
12:20AM 0 RE: How to differentiate a *busy* call from not available?
Sunday July 11 2004
9:08PM 0 wcusb dialing problem and line noise
7:09PM 1 Stopping reinvite with IAX2?
6:45PM 1 mediatrix 1204 hysteria
6:00PM 1 Please ignore my last message...
5:38PM 1 Echo issues (again...)
4:00PM 6 feature - VM gain adjust?
11:21AM 3 QoS in asterisk
10:52AM 0 DIALSTATUS variable and oh323 channel
10:38AM 4 Asterisk on FreeBSD 4.10 dies
9:16AM 0 Hardware for sale / donate
9:11AM 0 VoiceMail + Forwarding + Directory Dial by Name : How?
8:07AM 0 iax2 - peer 2 peer - asterisk?
3:14AM 0 Wh uses H.323-clients with call transfer?
1:15AM 20 New Asterisk bounty: SIP simultaneous
Saturday July 10 2004
8:35PM 0 VoIP provider for 2 site enterprise deployment??
7:44PM 0 Two server
7:10PM 2 Looking for a patch that was post May 1 2004
4:58PM 1 NuFone Error
3:59PM 2 New Asterisk bounty: SIP simultaneous registry
2:38PM 0 Does the SPA-3000 get rid of echo that the X100P can't?
12:05PM 0 How to use freeradius for Asterisk billing
10:04AM 1 tdm 400p noise
9:33AM 5 Three (quick?) questions...
8:11AM 1 Asterisk + g.726
7:40AM 3 X101P FXO with RED alarm
7:29AM 2 German Asterisk Site
6:53AM 0 (no subject)
6:46AM 0 bad clicking sounds with Diva+capi+asterisk
4:17AM 0 Asterisk Support for ISDN National-3?
Friday July 9 2004
1:50PM 0 Did I get booted?
12:52PM 0 Asterisk and CDRTool
12:51PM 9 using asterisk voicemail with a class 5 softswitch
11:56AM 3 SMDR/CDR - Asterisk integration
11:55AM 1 RE: ATA 186, firmware SIP 3.1 and codec g.726 + now SIPURA SPA-2000
11:54AM 1 No data when recording a Meetme conference with Monitor
11:31AM 1 x100p and groundstart signaling - help
11:28AM 7 Predictive Dialers
11:21AM 1 QoS - Routers
11:14AM 1 IVR Menu and VoiceMail quality
10:59AM 3 E1 config help and guidance
9:50AM 1 zaphfc - TE mode - callerid trouble
9:31AM 3 ATA 186, firmware SIP 3.1 and codec g.726
9:26AM 1 Fwd: Problem of loading the oh-323 module
8:36AM 3 Debian Unstable Claims Asterisk 1.0-1
8:02AM 0 GSM to iLBC one way audio :-(
7:41AM 1 Re: SNMP Monitoring (Andrea Fino)
7:28AM 2 * integration possible?
7:09AM 2 T1 Hardware Echo Can
6:27AM 1 Help needed regarding Grandstream phone
6:20AM 4 Cisco MC3810 -> Asterisk
6:18AM 0 chan_mISDN test release....
5:46AM 2 SIP Regiter config question
5:27AM 4 strange echo problem
5:01AM 4 Dell 6450 / TE405p
4:42AM 0 Overlapdial on PRI
3:54AM 7 IRC channel #asterisk on
3:17AM 2 Problems with cdr_csv
3:03AM 1 Asterisk and Audiocodes MP124
1:59AM 1 sound quality IAX client GSM to ALAW with oh323
1:06AM 1 bristuff - hfc card + x100p
Thursday July 8 2004
9:50PM 1 ok
8:52PM 1 Two outbound calls at once
5:32PM 1 Re: Asterisk-Users digest, Vol 1 #4460 - 14 msgs
5:10PM 1 Asterisk receives TMC Labs Internet Telephony Innovation Award
4:42PM 3 asterisk to asterisk config
3:35PM 0 GS & DTMF in voicemail with CVS of today!?
3:00PM 1 Intermittent SIP 404 Not Found response?
2:45PM 6 Updated Grandstream configurator
2:22PM 2 SNMP Monitoring
1:39PM 1 displaying call progress with SendText on a Snom
12:59PM 2 internal & external SIP
12:13PM 1 advanced audio recording agi help
11:47AM 0 IAX2 problems transfering back and forth between pbxes
11:47AM 0 outgoing caller id from SIP to isdn (p2p)
11:09AM 5 Using Cisco AS5350 as pstn GW .. one-way audio problem
11:02AM 0 rxfax - mISDN - missing logs
9:49AM 1 Re: tdm400p static - out of ideas (Jorge Mendoza)
9:48AM 3 Interface to generate Statements?
9:19AM 1 Access Bank 2 <---> T100P T1 Cable.
9:06AM 0 Turning off RFC 3265 in Asterisk
8:26AM 1 Using Windows Messenger+Video in *
8:09AM 3 i or s or whatever the invalid_exten is HELP !!!!!
7:48AM 0 WellTech Wellgate 5250 E1 trunk gateway
7:22AM 3 Audiocodes -> Asterisk Implementation
7:05AM 8 FINALLY! a good book about Asterisk.
7:04AM 2 Shady dial anyone??
7:01AM 2 Slackware 10.0 and asterisk and 2.4 vs 2.6
7:00AM 0 Problem SIP no audio just noise
6:42AM 1 Sip Peer Status
6:31AM 4 sample config file for GS BT101?
6:07AM 0 Meetme and IAX
5:21AM 2 Question about Cisco IP Phone 7960
5:02AM 0 Minimum install required for Asterisk + voicemail & SIP friends from mysql
4:00AM 2 Cisco 7960 NAT question
2:55AM 1 Rollover oddity
2:52AM 1 x100p and two hfc isdn cards
2:50AM 0 Call placed towards a trange called number 'h'
12:47AM 3 ISDN, AVM C4, HFC-cards and echo
12:39AM 2 pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' WARNING
12:35AM 0 R: VoIP hackers gut Caller ID
Wednesday July 7 2004
10:22PM 0 X100P bad sound after period of time
10:20PM 0 Intermittent cidname lookups
7:49PM 4 Small Linux Distro
7:28PM 1 patlooptest output
6:51PM 1 RE: What is the difference between queeu members and queue agents
6:44PM 0 NEWS from the chan_sccp developers.
6:39PM 1 OH323-COMPILE
5:45PM 5 E100P
4:32PM 3 Mandrake 10, Request for comments.
4:14PM 1 Problem when using asterisk + gnugk
3:52PM 0 rxgain and txgain not effective
3:06PM 2 Parking call problem
2:38PM 0 Perl libaray for manipulating .conf file
1:43PM 1 Cisco, Sip, Linux, ISDN
1:40PM 2 Perl library to manipulate 'ini files'
1:38PM 0 GR-303 configuration options?
1:23PM 0 solved - Audio cuts off 10 minutes into calls
12:56PM 4 VoicePulse Connect DID Problems
11:44AM 1 Ringinbacktone even without 'r', and inexistant codec
11:42AM 0 Audio cuts off 10 minutes into calls
11:25AM 1 res_odbc not working
11:16AM 0 Conf files doubt
10:46AM 0 :: Astricon :: Registration now open!
10:42AM 0 Modem support via Cisco FXS ports?
10:23AM 1 Software SIP fax client
10:23AM 1 RE: What is the difference between queeu members and queue agents
8:31AM 2 zaphfc and ASUSCOM working in the US
7:13AM 4 tdm400p static - out of ideas
7:09AM 1 UDP Ports scan on firewall
7:03AM 0 kdeconsole and gtkconsole
6:52AM 0 IP Dialog Hangup problem
6:50AM 2 Problem SIP Register
6:43AM 2 Asterisk Article
6:34AM 7 New PBX Help
6:34AM 0 Asterisk not populating nonce count
6:26AM 1 Sangoma cards
6:11AM 1 X100P donĀ“t answer sometimes
5:40AM 0 Simple H323 Test
5:37AM 0 Language
5:29AM 8 VoIP hackers gut Caller ID
5:24AM 0 sound quility
3:41AM 4 HFC- Colongne TE Mode
2:37AM 1 Call files timeout on Flash command
1:35AM 3 Problems installing asterisk.
1:19AM 8 Voicemail volume
1:14AM 1 recording an on-going call
1:00AM 1 CDR records into SQLite
12:32AM 0 app_flash Flash command - flash lasting too long
12:26AM 0 FYI: David Isenberg on IAX and SIP
Tuesday July 6 2004
11:43PM 2 AGI - No audio
9:45PM 4 Newbie's doubt on sip.conf
9:22PM 1 compiling mysql addon
8:47PM 3 odd behavior - adtran ta 850 + t100p
8:22PM 3 Cisco 7960 and Voice Mail
7:22PM 1 Identify incoming 800 number
7:14PM 0 CDR and EXTEN
6:52PM 1 g729 codec compatibility voiceage vs Digium
5:55PM 1 asterisk grandstream aleatory error
5:52PM 2 Kerry/Edwards campaign and VOIP
5:34PM 2 TDM FXO port remains offhook
3:52PM 1 Hangup's not detected correctly
3:13PM 0 Channel bank or IAD with message light capability?
2:09PM 1 Problem related to TPX100 card installation
1:18PM 3 multiple days on a GotoIfTime command?
12:56PM 2 GR303
12:44PM 1 zaptel DTMF delay
12:31PM 0 Sound card troubles with asterisk resulting in no sound
12:19PM 3 Zap Channel error using 4-port FXO TDM400P
12:17PM 1 FYI House bill exports analog phone regs to VoIP
12:01PM 4 Odd Zap dialing problem
11:37AM 2 Mediatrix 1102 Problems
11:34AM 1 SIP and H323
10:25AM 1 rh9, asterisk HEAD, & asterisk-oh323-0.6.3a working
10:12AM 0 Numbering range
10:02AM 1 quantumvoice
9:43AM 2 ztdummy running, but moh & meetme don't work
8:53AM 2 Uniden consult transfer
8:42AM 3 New CVS for patch...
7:48AM 1 G.723.1 and Asterisk
6:47AM 3 SPA-2000 and time of day
6:27AM 3 Dialing out of a voicemail message?
6:13AM 1 zaphfc 2 cards working with P2P Mode ?? - massive Problems
5:48AM 1 * and Innovaphone
5:39AM 0 Music on hold error since CVS update
5:00AM 1 missing .gsm in VoiceMailMain(2)
4:37AM 0 Asterisk config on PostgreSQL
3:36AM 0 AW: AW: Junghans Quad-BRI card and asterisk cvs-head
3:23AM 3 H323 channel
2:35AM 1 AW: H323 Call Transfers
2:29AM 3 AW: Junghans Quad-BRI card and asterisk cvs-head
2:13AM 0 ZyXEL P2000W - working conf example
2:09AM 1 H323 Call Transfers
1:58AM 1 Junghans Quad-BRI card and asterisk cvs-head
1:38AM 2 How do I disable '#' to transfer a call?
1:11AM 6 is srv lookup being done when REGISTERing?
12:39AM 0 isdn to sip callerID pass
12:33AM 0 How to connect to cellular phone beside analog interface card?
12:27AM 1 2x analog interface (1 ISDN and 1 door phone) recomendation for Europe ?
Monday July 5 2004
11:28PM 0 Any experience with Citel Link 3300 and Asterisk
10:02PM 2 What happened to the CVS asterisk_stable branch?
9:29PM 3 asterisk, fwd, and grandstream?
7:51PM 7 Calling an outside phone number as part of a hunt
7:30PM 1 Clean compilation
5:50PM 0 Playback/Background over Console/dsp
5:48PM 2 Playback over Console
4:05PM 3 Randy Bush is a destructive force with a hidden professional agenda
4:01PM 1 Voice channel could not be established.
2:46PM 2 Wake Up Call AP
1:14PM 0 outgoing Sip-call problem URI and Phone-number
1:11PM 9 iax or sip
11:29AM 2 No RED/GREEN alerts on TDM400P?
11:12AM 1 Cut off after 8 secs?? Help
10:58AM 2 T1 configuration, getting help via IRC?
10:11AM 1 FireFly client and echo problems with IAX
9:56AM 4 IAX Call Pickup
9:27AM 2 fax detection and X100P
9:19AM 0 Voicemail plays back at very low volume - how to make it louder?
9:11AM 0 SMS on TE410P
8:59AM 0 Rederecting an incoming CAPI call to SIP soft phones
8:11AM 3 dialing # on a crisco (was: Divert to arbitrary number)
8:04AM 0 chan_misdn HFC-NT dialtone
7:25AM 4 Question about x100P and zap
7:19AM 2 Again Sip Registration Fail
6:23AM 0 [Asterisk] 2 T100p and and Panasonic
6:00AM 1 *8# into invalid extensions
5:30AM 2 Problem with BRI_STUF / direct connected ISDN-Phone
5:12AM 3 *** Asterisk Sunday (hrrm) News: Moving ahead at CVS Warp 5
4:09AM 1 Divert to arbitrary number.
12:34AM 0 Penalty in queues.conf
Sunday July 4 2004
8:50PM 1 PTHREAD_MUTEX_RECURSIVE in appradius-1.0
5:23PM 0 LCS multiparty conferencing commercial opportunity
4:13PM 1 cdr and edit dst field
12:12PM 2 music on hold question with asterisk
11:53AM 3 looking for newbie resources
11:15AM 4 Asterisk Book
10:48AM 1 conf from pgsql database
10:03AM 2 I wanna kill FWD.... GRRR!!!
7:06AM 0 FWD/SIP audio suddenly stopped working
5:03AM 1 How to use return value in extensions.conf
1:31AM 1 Using call redirection numbers
Saturday July 3 2004
7:47PM 0 PRI unknown signalling on TE405P
5:56PM 1 Caller ID and DNIS Problems (Non-Pri T1)
5:09PM 1 Size of asterisk internal database
11:22AM 11 Music on hold problem
10:44AM 2 Multiple E1s over TDMoE?
9:48AM 2 saydigits/background
4:42AM 0 Support for Snom 200 Extension Monitoring
Friday July 2 2004
10:31PM 0 TDM400P GroundStart Problems
4:30PM 0 DISA and AGI: authenticate by caller ID? (resolved)
2:48PM 2 Zaptel dacs / dacs
2:48PM 2 H323 -> IAX
1:43PM 3 Termination for Asterisk Users - Inter-Asterisk Exchange
1:37PM 1 IAX to IAX call with really bad echo
1:19PM 0 TA750 + T100P configuration. flashing red alarm on t100p after running modprobe wct1xxp, ztcfg
12:49PM 3 Inter-Asterisk Exchange
12:18PM 1 Compiling Gastman for Win32
10:54AM 0 do_monitor: Bad file descriptor
10:26AM 0 Channel Bank Newbie Problem
9:43AM 3 IRQ Misses and Dropped Calls?
9:28AM 1 mysql voicemail
8:50AM 3 Suggestions for 96 tip/ring lines?
8:36AM 1 Problem with CHAN_SCCP
8:22AM 1 RTP Source IP Address
8:17AM 3 CDR shows billsec=12 for all bridged calles.
8:07AM 0 Cisco 7960G and *
6:36AM 0 ZAPTEL FXO debuging - Tones, Voltages, Ampers, etc.
5:18AM 2 Zaptel, Line Impedence and Echo
4:47AM 0 Have problem install via cvs
4:42AM 1 TE410P PINS
4:27AM 0 ip10: config setting "PackageNotify"?
3:21AM 0 Status of Australian approval for E100P...???
3:15AM 0 CBMySQL
3:14AM 2 Monitoring Asterisk
1:11AM 0 Problem locating stream files
12:33AM 1 Grandstream available
12:31AM 4 Delay when dialing with Sipura 2000
Thursday July 1 2004
11:07PM 2 Grandstream HT286 & Meetme
8:43PM 5 Inter-Tel Eclipse2 (IP PhonePlus)
6:52PM 1 commercial implementation
5:23PM 0 Invalid context
4:36PM 1 Hangup on transfer...
4:36PM 0 Re: Hello
4:06PM 1 two sip clients on one server
3:12PM 1 Cannot install module Bri-Stuff-0.0.2 zaphfc.ko does not exist.
3:02PM 9 Config Files
1:14PM 1 SPA-2000, call for help testing echo issues...
1:00PM 5 Sip to Sip
12:56PM 0 Updated version of Grandstream cfg file generator
12:34PM 2 IAX2 to IAX2 connection problems
11:55AM 0 Weird LAN VoIP Echo
11:03AM 4 Pager Notification
10:30AM 0 zaptel wont compile errors on zttest
10:08AM 2 DISA and AGI: authenticate by caller ID?
9:13AM 0 Sound: Record Overrun
9:00AM 1 How to track (log file) Dial Plan events to fix unsteadily states like opened FXO port
8:51AM 5 Zultys 4x4 or 4x5 ip phones?
8:26AM 0 sccp to sip call signalling
8:23AM 0 simple AGI script
8:11AM 0 Strange behavioir on a exten
7:36AM 5 voicemail notification?
7:07AM 4 1800 number with colo
7:02AM 0 R: Asterisk Docs
6:58AM 3 R: execute a context from cron
6:46AM 5 execute a context from cron
6:42AM 1 Asterisk Docs
6:24AM 2 Registration failed for SIP
3:28AM 1 Help with Welltech 2FXO gateway, GS BT100 and Asterisk
2:58AM 0 Simple gateway SIP <--> ISDN
1:57AM 2 Providing Telewest in the UK with per extens ion outbound callerID
12:08AM 0 2 T100P and a Digital PBX