asterisk users - May 2005

Tuesday May 31 2005
11:20PM 0 A newbie question - SIP to Trunk
7:51PM 2 pbx -> fiber -> network media converter -> wifi -> network media converter -> fiber -> pbx ???
7:37PM 0 Google Summer Of Code
6:36PM 0 * with mfcr2 and libunicall success stories ?
5:32PM 1 Phone always busy after caller hangup
5:14PM 1 Suppress "Missed Calls" 7960 SIP
4:58PM 4 Karl
2:39PM 1 Sipura 3000 - fax passthrough?
2:31PM 1 `hint` priority and Polycom 500
1:54PM 4 Asterisk@Home 1.1b1 has been released
1:06PM 0 Re: chan_sccp / 7960: reproduceable semi-lockup
12:21PM 0 High CPU in Asterisk, chan_unicall and dtmf
12:01PM 3 Opinions of Sphinx?
11:36AM 0 Re: [cisco-voip] SIP Authentication problem between Cisco router and Asterisk when calls are forwarded
10:56AM 0 Codec ordering?
10:30AM 1 SIP Authentication problem between Cisco router and Asterisk when calls are forwarded
10:18AM 1 rxfax application - doesn't work properly
9:58AM 1 Re: chan_sccp / 7960: reproduceable semi-lockup
9:56AM 0 Re: chan_sccp / 7960: "End call" softkey: "That key is not active here"
9:35AM 0 Session Registrations
9:07AM 0 Campon feature?
9:02AM 2 ISO Suggestions for Multiple Inbound Voicepulse Lines
8:36AM 0 Receive calls with Aastra 480i phone problem
8:26AM 4 AreskiCC - DOES IT REALLY WORK??????
8:24AM 1 # Transfers
8:21AM 2 R: R: R: R: AT-320 + supervised transfer
8:18AM 5 CIsco 7960 SIP Image
8:13AM 1 Built-In Transfer Questions
8:09AM 1 asterisk acts as media gateway for existing pabx ?
7:36AM 0 Connecting a peer to a dynamic ip asterisk b ox ???
6:59AM 1 Re: astpp database creation failed...please help...
6:43AM 0 asterisk sip register with no username and password.
6:40AM 4 Chan_sccp / wiki
6:37AM 0 Sipura 3000 Analog Line No Answer, No Audio
6:28AM 4 Asterisk with another Asterisk
6:26AM 0 Polycom IP500 with Video
6:08AM 7 Tools for effectively manage Asterisk
6:00AM 0 'beeps' while recording..?
4:51AM 1 monitoring oh323 calls
4:41AM 3 Automatic Codec change for different communication channels!?
4:22AM 2 Ztdummy usage
4:21AM 0 Auto-generated incoming calls X100P
4:08AM 1 Asterisk compailation Error Chan_zap.c
3:06AM 2 Sipura 2000 behind NAT issue, Vonage is working
2:54AM 1 How does ISDN really work?
2:51AM 4 Extension context question
1:26AM 1 Uniden UIP1868 - any sightings or users?
1:20AM 0 Asterisk: HelpDesk / CRM type of Application in Asterisk
1:04AM 2 handytone 486
1:00AM 10 UPS rating for SOHO asterisk box
12:56AM 0 UK NCFA calling
12:56AM 0 ipchains for firewall, QOS howto?
12:53AM 0 Re: Asterisk-Users Digest, Vol 10, Issue 234
12:14AM 2 Problem with asterisk+gnugk
12:11AM 1 MGC on asterisk
Monday May 30 2005
11:37PM 1 astpp database creation failed!
11:23PM 0 newbie problem with registration of sip client
10:58PM 1 Chan OH323 and overlapping digits
10:44PM 2 Sipura 3000 dialing "noise"
10:18PM 1 RE: Invalid login/password with AreskiCC V2
9:51PM 0 Dialplan structure
9:45PM 1 Speed dial number and actually dialing combined with ASTCC
9:39PM 0 LCR and ASTCC
9:10PM 1 Codec - pay for and to whom?
8:44PM 1 Remote phone: Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from
8:18PM 1 Where to start to solve hardware problem?
7:53PM 0 Sipura 3000 does not dial out
6:22PM 1 Sipura ATA and Asterisk No Answer Issue
4:47PM 3 Connecting a peer to a dynamic ip asterisk box ???
2:46PM 1 Urgent Help neededt!! Asterisk 1.0.7 CPU
2:06PM 1 I865, HFC-S etc.
1:42PM 2 Error in Zapata Config?
12:48PM 3 Asterisk install error ...
12:22PM 2 Multiple Ext on IP500
12:06PM 0 transcoding prevention
11:41AM 2 pridialplan & prilocaldialplan
11:13AM 3 Urgent Help neededt!! Asterisk 1.0.7 CPU at 99%
11:07AM 0 where can i get a vanity DID?
10:21AM 4 R: R: R: AT-320 + supervised transfer
9:34AM 0 perl agi : get_variable problem
9:32AM 1 nntp access
9:22AM 2 R: R: AT-320 + supervised transfer
8:53AM 2 Meridian 808 Function
8:19AM 3 R: AT-320 + supervised transfer
8:03AM 2 choice of processors
7:58AM 1 asterisk compatible, hot swappable PRI card
6:42AM 0 Gradwell UK DID + DTMF
6:21AM 0 Areski Calling Card
6:14AM 5 Asterisk on Soekris
6:05AM 0 ISDN RAS and data calls
6:00AM 0 Pls, find me a VoIP Supplier/Reseller in Dubai-UAE
5:59AM 0 Flash Operator Panel 0.21 released
5:48AM 1 AT-320 + supervised transfer
5:38AM 0 MGCP and missing digit map
4:26AM 1 Serious ZapRAS problem!
3:43AM 0 @home to @home
3:40AM 0 IAX2 to H323
2:27AM 1
2:25AM 0 Asterisk with PrimuX 1S2M ISDN card
2:10AM 1 How to configure Inter7's Asterisk Fax with Postfix
2:08AM 0 IAX2 registration period
1:48AM 2 Problem with SIP clients
1:36AM 2 IAX encrytion
1:01AM 0 asterisk integration with Quintum Tenor AXT800!
12:58AM 0 Voicemail make crash
Sunday May 29 2005
11:10PM 0 [Serusers] QOS of VoIP
8:29PM 0 Asterisk Multi Tenant setup
8:27PM 1 voice is coming only from one side
8:05PM 2 Recording does not stop.
7:21PM 1 Pre paid Card
5:59PM 4 Re: Digium Website Update: Asterisk Business Edition
5:14PM 1 ANNOUNCEMENTt: GPL Asterisk Billing Software
3:26PM 1 chan_unicall and dtmf problem
3:20PM 0 Custom Extension on AMP
2:32PM 2 Peer to Peer calls
11:50AM 0 chan_oss.c:572 oss_write: Unable to set device to input mode error
11:43AM 2 CallerID of calls FROM queue
10:27AM 1 60 second time out
9:22AM 0 Digium Website Update!
7:51AM 1 Upgrading my HOP-1002 software
7:44AM 3 BT100 Phone Died During Call
6:22AM 0 Asterisk 1.0.7 on VIA EPIA 5000
6:06AM 0 How to Define Asterisk Behind a Nat
5:24AM 1 Database Usage with Asterisk
3:21AM 1 Error attempting to make Zaptel on Red Hat linux 9.0
12:37AM 1 LCR
Saturday May 28 2005
8:51PM 1 Recommendations are highly appreciated -SIP HARDWARE phone
8:46PM 1 Pictures of the Digium booth at ISPCon 2005
8:10PM 0 parsing extension name in a command
7:06PM 1 3 goes and your out
5:47PM 0 Asterisk on a Linksys wrt54g
3:54PM 3 CallerID when transferring calls.
3:31PM 0 chan_sccp / 7960: ALERT_INFO?
3:19PM 1 cmd curl crashes asterisk:
2:32PM 0 Re: chan_sccp / 7960: "End call" softkey: "That key is not active here"
12:40PM 0 Re: chan_sccp / 7960: "End call" softkey: "That key is not active here"
12:37PM 0 Re: chan_sccp / 7960: 7960-font.xml
12:33PM 1 CallerID for UK
11:55AM 1 Pick up on first ring
10:44AM 0 Help with New SIP phone.
9:02AM 1 Enum or Dundi?
8:09AM 1 Fax and SIP Device
7:41AM 0 newbie asterisk SIP config question (using VoicePulse Connect!)
7:34AM 0 newbie asterisk SIP config question (using VoicePulse Connect)
7:34AM 0 Re: Asterisk-Users Digest, Vol 10, Issue 222
6:31AM 2 UK DID providers
5:45AM 0 MWI - One mailbox, multiple extensions, lots of lights!
3:27AM 0 chan_sccp patches
2:40AM 2 xc-ast 0.9.0 is out today
2:28AM 0 TDM zap channel Exception on 15, channel 1
2:03AM 0 Asterisk@home rejecting nufone incoming calls (iax2)
12:20AM 1 Quintum Tenor AXT800!
12:17AM 1 7960 / chan_sccp: Less than three lines / more than three speeddials possible?
12:01AM 1 ivr not working?
Friday May 27 2005
10:48PM 1 Re: Asterisk-Users Digest, Vol 10, Issue 221
10:45PM 0 Remote Server IAX Configuration
10:03PM 0 Asterisk vs pingtel?
9:10PM 1 Changes on CVS HEAD
8:26PM 3 Polycom phones, UNREACHABLE
8:20PM 1 Re: Asterisk-Users Digest, Vol 10, Issue 114
7:52PM 0 you can bid on this very small project
6:53PM 0 How to timeout using AGI.
4:47PM 1 static linking
3:36PM 1 Asterisk stopping to respond and CPU at the top
1:37PM 0 Switch from NBX to Asterisk
1:34PM 3 Wacko Distinctive Ring Patterns being detected??
1:24PM 0 DVG-1120S does not show callerid Name and resets time
12:46PM 0 sip phone behind nat connecting to an asterisk box that has one port on the open internet
12:36PM 0 SIP REFER: Trying again
12:26PM 1 VoiPSupply Dot Com: Epilogue
12:18PM 1 Soyo G688
11:57AM 0 asterisk and nortel CS1000 using SIP
11:54AM 0 asterisk and nortel BS1000 using SIP
11:42AM 1 Fwd: Newbie here. Tips on setting up 100 phones wanted.
11:37AM 1 Upgraded firmware on Polycom 500, digit-order problems
11:36AM 1 How to Connect Netphone IP phone with ASterisk
11:21AM 0 Call waiting on TDM-400 FXO
11:10AM 0 Another OH323 Problem
10:51AM 2 Polycom IP 500 SIP bootrom and firmware upgrades
10:18AM 3 Newbie here. Tips on setting up 100 phones w anted.
10:00AM 6 Newbie here. Tips on setting up 100 phones wanted.
9:01AM 1 Temporary unavailable -????
9:01AM 2 CRM integration (was RE: CallerID)
8:12AM 3 Recommended Network Latency
8:01AM 2 Grandstream GSX-2000 - dead :-(
7:57AM 2 PRI "Actual-HookState" not showing offhook on inbound
7:56AM 1 Unable to create channel of type 'Zap' with zaphfc driver
7:48AM 2 Interco H323 : IPNx (from WTL) and *
7:40AM 0 compiling new module; conflicting function
7:22AM 0 bristuff-0.2.0-RC8e priindication=passthrough problem
6:55AM 0 Cisco 7960 tftp "not null terminated"
6:48AM 3 Call waiting?
6:35AM 1 VoiceMail with Polycom 500
5:08AM 0 problem about client authorisation
5:08AM 0 Can't transfer calls on polycom 500 after new firmware upgrade
3:33AM 0 Problem with SIP peer registration
3:28AM 0 spandsp <--> analogue modems?
3:12AM 3 G729 vs. gsm
3:03AM 2 5000 sip clients (voip phones)
2:51AM 0 H323 setup problem
2:35AM 0 CRM integration (was RE: CallerID)
2:26AM 0 Re: MoH: mgp123 problems
2:23AM 0 ASTCC/ 'L' option hangup wackyness
2:08AM 1 SIP SoftPhone for debuging
1:42AM 0 Re: Areski Calling Card Download locations
12:22AM 0 Re: Asterisk-Users Digest, Vol 10, Issue 215
12:20AM 2 DID - B8 Message
12:02AM 2 chan_zap.c:8534 pri_dchannel: PRI Error
Thursday May 26 2005
11:23PM 0 capi dial in/out configuration
10:47PM 1 does Jitter calculation in chan_iax2.c work???
9:30PM 0 Asterisk crashes with sipp
8:47PM 4 International Caller ID?
4:40PM 1 TDM400P in 2U server?
3:56PM 3 Analog Telephone Adapter
3:00PM 1 How do I diagnose the problem in this Asterisk test session with FWD?
2:39PM 0 dhcp vars, mediatrix 1204's
1:34PM 1 Asterisk con X-lite : Register Ok but no calls (404 Not found)
1:16PM 4 tds_CDR and MS SQL Server troubleshooting
1:10PM 2 Limiting maximum runtime of echo test
12:52PM 1 Asterisk connecting to Nortel 1000 with SIP
12:18PM 0 Asterisk on 64 bit Linux
12:12PM 0 ICD Usage Examples
12:12PM 0 TDM > * rings once and goes fast busy
12:03PM 5 Looking for inexpensive phone to use with Asterisk with message light and a button that will let me play new messages
11:57AM 0 Connecting a couple DS0's to a wildcard
11:38AM 1 Dropping frame of G.729 since we already have a VAD frame at the end
11:17AM 0 app_dial_rev5
10:58AM 1 YET Another echo issue PRI CARD Any help acc epted :-)
10:49AM 1 Using zap channels on 2 different servers
10:14AM 2 Asterisk@home - mysql login
10:14AM 4 YET Another echo issue PRI CARD Any help accepted :-)
9:37AM 0 jitter buffer recomendations
9:13AM 1 How do I know that my machine will support APIC?
8:28AM 0 DNS protocol flaw in Cisco client products announced May-24-2005
8:25AM 4 multiples broadvoice lines
8:03AM 1 SIP V2 Support
7:52AM 0 SetCDRUserField not working in cvs 5/25/05
7:43AM 0 PSTN->SIP->PSTN transfer problem
7:30AM 0 Scalability issue: P2P connection w/o passing through asterisk
5:52AM 5 SIP Soft Video phone for Asterisk usage
5:49AM 1 Echo with two IP phones through Asterisk using SIP
5:41AM 0 Q : registering sipXphone
4:39AM 1 Re: Asterisk-Users Digest, Vol 10, Issue 188
3:59AM 1 Speakeasy as a VOIP provider?
3:59AM 1 VIDEO ON 1.0.7 stable
3:24AM 2 voicemail comprehension
3:05AM 0 Looking for wall mountable cases in the UK.
3:03AM 0 SV: Little Php question
3:01AM 1 Little Php question
2:47AM 0 video conference feature
2:45AM 0 Prad or V5.2
12:54AM 2 chan_capi with an AVM C4 connected to 4 BRI-lines in PTP-Mode
12:44AM 1 deadlock
12:38AM 2 static database config gui
12:31AM 1 Size of extensions.conf
12:14AM 1 AS5300 + Asterisk
Wednesday May 25 2005
11:30PM 3 Asterisk Versions
11:01PM 6 new cisco ip video phone?
10:31PM 0 Port 6057 blocked on firewall
8:07PM 7 Survey: E1 prices
7:24PM 0 Choppy audio
7:02PM 0 AMP 1.10.008 released!
6:51PM 0 Tying together two Asterisk servers
6:25PM 1 Legacy Toshiba integration
5:51PM 0 FAST BUSY on Back to back ZAP outgoing calls
5:18PM 0 MeetMe Announce User feature
3:29PM 1 Default caller ID
3:25PM 5 SER Config for Asterisk
3:13PM 4 SER Help
2:02PM 0 bounty: app_queues.c with mysql support
1:53PM 5 Asterisk Crashing; Not getting Core dumps
1:52PM 0 Excellent Article explainng what is up with Broadvoice
1:46PM 1 Asterisk and SER on Same Box
1:30PM 0 Cisco 7960 Firmware help pleas
1:10PM 0 Sipura 3000 sound problems
1:04PM 0 Correctly handle two extensions for the same phone (one with voicemail one without)
1:04PM 1 astcc no billed cost
12:46PM 1 Problems with Public IP
12:44PM 1 Remote Voicemail Notifier / enter Dialplan on SIP Register
12:05PM 0 Remote Voicemail Notifier / enter Diaplplan on SIP Register
11:59AM 0 CRM integration (was RE: CallerID)
11:53AM 0 Getting firmware
11:18AM 2 Conferences using Manager API
10:58AM 8 What does Asterisk need in the way of a GUI?
10:34AM 13 Cisco 7960 Firmware help please.
10:25AM 0 oh323 problems - Solved
10:13AM 2 Nortel i2004 firmware upgrade.
10:11AM 2 Budgetone 102 and voicemail problem
10:08AM 4 Polycom IP501
9:54AM 7 zaphfc: empty HDLC frame or bad CRC received
9:40AM 4 Asterisk's MultiProcessor Ability
9:18AM 2 Manager and Callerid problems
9:04AM 1 Looking for list with asterisk default extensions
8:40AM 0 May Twenty Fifth SayUnixTime
8:39AM 0 Attended Transfer failing with Agents
8:34AM 1 Asterisk, 2 x network interfaces and traffic shaping on same box?
8:12AM 2 CRM integration (was RE: CallerID)
7:59AM 1 LiveVoip does not like customers anymore, ....
7:58AM 0 CRM integration (was RE: CallerID)
7:39AM 0 CRM integration (was RE: CallerID)
7:25AM 0 - new Swedish user forum
7:14AM 1 CRM integration (was RE: CallerID)
6:52AM 3 sip extension logon failed problem
6:11AM 0 Meetme - any way to stop a participant receiving audio?
6:07AM 1 Can Ztdummy be used in production environment
5:32AM 2 MoH: mpg123 problems
5:15AM 0 G.729 disappears from h.323 codecs. Help, please!
5:09AM 2 Asterisk and Monwall - comments
4:07AM 0 Is SKYPE a threat orshould wedo something(together)
3:14AM 5 how to dial extension with menu
2:47AM 0 configuration asterisk zap module
2:41AM 1 Possible to send Calling Number as TON: international ?
2:39AM 15 PHP/AGI Problem
2:36AM 2 HiPath 4000 and Asterisk
2:17AM 1 Polycom IP 600 DHCP problem
2:15AM 1 Asterisk SIP cannot restrict call from softphone before registration
1:41AM 2 RTP path with Cisco CCM
1:12AM 0 Segfaults on Asterisk HEAD
12:56AM 5 C files of Asterisk
Tuesday May 24 2005
11:53PM 3 rxfax(spandsp-0.0.2pre18) and HT488
10:58PM 1 OT: cisco ip phone security problem
10:14PM 3 New Grandstream phones.
9:38PM 6 echo problem
9:14PM 1 General AGI Question
8:35PM 0 [***POSSIBLE SPAM***]RE: Help Configuring CiscoATA 186
8:13PM 2 Help Configuring Cisco ATA 186
8:02PM 0 silence in virtual extension
7:28PM 0 Firmware for Cisco ATA 186
6:26PM 0 IPswitch brings up a message daily
5:09PM 4 audio message delivery
4:46PM 1 Cisco Config
4:02PM 0 Xeon server board for TE405P
2:00PM 3 Budgetone and NAT not working
1:53PM 3 PHPAGI problems
1:42PM 0 asterisk take 99% of CPU resources
1:38PM 0 Re: origination providers (mike castleman)
1:35PM 0 Redirection
1:30PM 0 Re: IAX Firefly config (Jeromy Grimmett)
1:28PM 0 G729 and XTen Pro
12:53PM 0 Key Rotary Lines ?
12:28PM 1 realtime static
12:24PM 2 RE: Firefly config
12:00PM 1 origination providers
11:58AM 3 How To Connect an IP phone with asterisk
11:50AM 1 CTI
11:23AM 5 Red Alarm TE110P
11:23AM 0 Sipura SPA-3000 call progress, and interdigit delays
11:12AM 0 IAX Firefly config
10:45AM 0 302 redirection issue
10:12AM 5 MySQL Support For OS X
10:03AM 0 CallerID with Lucent TNT
9:54AM 1 Fax detection: Problem with extension number
8:16AM 2 Dial to a SIP fone ends up at Voicemail Busy
7:31AM 2 capi.conf
6:43AM 1 5ESS central office question
6:28AM 0 Problem with FXO taking a call
6:28AM 4 Rings - How to set number
6:06AM 0 txfax return code
6:01AM 1 Random Sound File
5:51AM 2 spandsp issue
5:18AM 1 Digium Wildcard X100P Error
5:12AM 0 Digium T100 Error
5:07AM 0 Echo with Digium TDM02B
4:59AM 0 How do you prevent a 3-way conference if an extension is busy ?
4:36AM 2 writing to MYSQL database
3:51AM 1 Silence supression
3:33AM 0 Problems installing TDM22B
3:31AM 1 Asterisk processes
3:18AM 0 DIAL with FastAGI and Answer Supervision
2:01AM 0 Remote-Party-ID handling
1:47AM 2 How to setup Dundi in Asterisk?
1:35AM 0 record message during dial
1:35AM 1 BudgeTone 101 doesn't register with FirmWare 1.5.23
1:34AM 1 Early B3 connects on zaphfc
1:33AM 0 H323 integrated Asterisk support
Monday May 23 2005
11:19PM 4 Broadvoice delivers CID even when restricted?
10:31PM 0 Callerid problem with identapop pro
10:29PM 0 App_odbcexec
7:42PM 1 Basic newbie questions
5:40PM 5 Inbound call center - reliability \ scalabil ity with queues
5:12PM 0 Inbound call center - reliability \ scalability with queues
4:27PM 3 ISPCON Mini-emergency: ATA186 Power Cube OK on SPA841?
3:52PM 1 Please explain this streaming example to me
3:11PM 0 OT: recover complete AES keys
1:32PM 0 spa-1001 not getting a dial tone on my pbx
1:31PM 3 Junction Networks
12:24PM 5 bluetooth headset/handsfree
12:15PM 2 E&M Tie Line
11:52AM 0 How to detect DTMF and change if needed
11:44AM 4 Digium FXS modules too fragile?
10:23AM 1 t38modem
10:11AM 4 How do you transfer a call to a busy extension ?
9:55AM 0 Sip reg problem
9:39AM 1 ZyXEL Prestige 2000W - cant make a call?
9:29AM 0 SIP authentification? Any ideas?
9:08AM 9 Windows IAX Softphone
9:07AM 1 E1 PRI Warnings
8:29AM 4 Programs to parse queue_log
8:01AM 0 Message in event_log.
7:34AM 8 play gsm files in windows
7:30AM 0 Drag and Drop with IPS
7:26AM 1 SendDTMF into a conference room
7:16AM 1 two isdn cards
6:48AM 1 Astersik vs. Pingtel
4:38AM 1 Grandstream GXP-2000 headset
4:37AM 7 Cisco 7960 & v7.4
3:37AM 0 Two or more asterisk servers, shared dialplan. Please help
2:59AM 0 Modifying the RTP and SIP protocol
2:53AM 0 Modifying Asterisk's C files
1:52AM 0 All channels on PRI stuck "Resetting"
1:33AM 1 How to connect to IPTEL.ORG
1:05AM 4 CallerID, TAPI and CTI
Sunday May 22 2005
11:31PM 1 spa-1001 with asterisk?
11:00PM 2 Looking for people to test calls
10:25PM 1 Which H.323 for Stable?
9:53PM 4 Cisco 7940g Firmware load problems
9:16PM 0 Using patch -p0 <meetme-diff-cbmysql_1.txtproduces 'malformed patch' message
9:02PM 0 how to forward a call to mobile?
8:01PM 2 Not answering/script.
7:48PM 4 Hangup Issues on TDM40B FXO Australia
3:47PM 0 Allied Telesyn AT-VP504E and asterisk
2:45PM 3 more than one company hosting their PBX on the same machine?
2:45PM 0 Using patch -p0 <meetme-diff-cbmysql_1.txt produces 'malformed patch' message
1:47PM 0 Digium and IPsando announces agenda for Astricon Europe - register now!
1:37PM 0 Trouble using two Fritz ISDN cards in one machine
1:31PM 2 asterisk with vonage linksys adapter?
12:59PM 1 Polycom IP600 Questions
10:09AM 2 (another) cisco 7960 question
9:19AM 1 asterisk-oh323: Max simultaneous calls ?
8:22AM 1 Asterisk Project Consultant/Parner Wanted
8:18AM 0 Fax and Voice VoIP services
6:49AM 2 2 Asterisk boxes sharing dial plans.
6:25AM 0 Pri doesn't accept Zap/g2 to call
6:16AM 0 Questions about TE410P card
5:55AM 4 Getting a Cisco gateway to work with Asterisk
5:12AM 0 *@home 1.0 FWD inbound problems, 2 calls generated
3:53AM 1 Upgrade cause's no Audio on IAX
1:50AM 1 realtime excessive database queries
Saturday May 21 2005
9:23PM 1 IP header Bandwidth Reduction
8:25PM 0 IAX provider using Broadvox's network?
8:15PM 0 Re: failure notice
6:34PM 2 realtime app data formatting
5:20PM 2 IAXTEl down
1:26PM 3 having asterisk play music on hold to callers while phone rings?
12:41PM 2 Working Xten, Asterisk, double-NAT configs out there?
11:53AM 1 Asterisk on NetBSD
11:32AM 1 Uncommon callback
9:59AM 1 Affecting overhead with Runlevel?
9:01AM 0 I call an USA MOBILE phone and it is registered as ENUM => failed
8:35AM 2 Spanish Voice Messages
8:35AM 1 Confirmation Of Extension Before Transfer?
4:56AM 0 PRI doesn't call cellphones
3:25AM 1 LiveVoip setup
3:15AM 0 Asterisk-Hylafax
3:14AM 1 Help Understanding ISDN Channels
2:47AM 0 IPSwitchBoard now supports CAPI
1:49AM 1 ISDN data connection through Asterisk
1:00AM 2 Set CallerID in zapata.conf with QuadBri or other solution with parallel call signalling
12:17AM 0 acd with mysql or ast_data support
12:16AM 1 PSTN->voip/sip echo
Friday May 20 2005
11:42PM 1 Voicemail With No Messages?
9:59PM 1 How can you keep agents logged in across a restart?
9:31PM 4 Boosting Internet Bandwidth for VOIP
8:56PM 0 Developer Needed!
7:49PM 0 I am looking for a programmer (VoIP, Linux)
7:27PM 1 Local Testing
7:21PM 0 Annoucement in MeetMe and segmentation fault
3:24PM 5 Who knows where voicepulse has their asterisk servers?
3:00PM 0 Looking for asterisk consultant with H323 configuration experience
1:56PM 1 Dell Poweredge 1850 and Zaptel
1:14PM 1 which cvs versions are being used in production systems?
12:52PM 2 MGCP 1.0 / NCS 1.0
12:49PM 3 Help with follow me
12:46PM 0 TDM04B and spandsp can't send a fax
12:46PM 2 Dell PowerEdge SC420 for Office Implementati on???
12:44PM 0 Displayed CallerID on Polycom 500 shows CALLERNAME only
12:19PM 1 Displayed CallerID on Polycom 500 shows CALLER NAME only
12:01PM 5 Dell PowerEdge SC420 for Office Implementation???
10:44AM 2 How to get in touch with sixTel?
10:22AM 2 Trouble getting a SIP phone to dial out through TE100P
10:08AM 1 MFC&R2 Venezuela with libunicall
10:07AM 1 Valet Parking and SuperValet Parking - back level
9:13AM 0 re:Digital Phones
8:55AM 0 Asterisk RealTime & asterisk configuration files through DBMS
8:22AM 0 RE: [Asterisk-biz] Asterisk at ISPcon
8:08AM 10 Stange question...
8:08AM 0 ToneCommander
7:44AM 0 Auto Answer BEEP
7:42AM 0 Registering with second SIP service causes error every 2 seconds - what is going on?
7:38AM 1 Raw Hangup
7:27AM 2 SecureTelephony
7:14AM 0 Anyone done the Cisco 7960 FW migration path programmatically?
7:14AM 0 X100p cards
7:06AM 1 H.323 Gateway
6:43AM 4 paging thru sipura-841
6:42AM 1 RDNIS (DNID) Call Routing
6:41AM 2 Polycom takes long time for reboot to access web page
6:34AM 4 Sipura 3000 Question
6:20AM 0 ref: Cisco 7960 question
6:00AM 1 does not load due to KRB5 symbol.
5:11AM 5 Newbie on IVR
4:56AM 2 call barring
4:23AM 1 chan_capi error2
3:27AM 1 Unable to create channel of type 'IAX2' (cause 3)
3:00AM 0 Offloading all user/peer autentication to SER?
2:17AM 0 lookup for extensions on another SIP Proxy
1:49AM 0 Greetings
1:26AM 0 Hint with snom 220 - call pick up
1:13AM 0 why can't my asterisk restart?
Thursday May 19 2005
10:53PM 2 NVFaxDetect on Gentoo
8:30PM 0 I am looking for some features,
8:15PM 2 IPswitch cannot delete lines & double lines
7:44PM 2 cisco 7960 question
7:25PM 0 Zaptel on pSeries
6:32PM 2 Voicemail wav49 format problem
6:11PM 1 Alsa and lag
5:16PM 2 MusicOnHold Loudness/Distortion
3:46PM 2 DHCP available?
3:02PM 3 Konftel
2:52PM 1 HasNewVoicemail not being called if user hang up after leaving VM ??
1:57PM 2 How do you put someone on hold on a zap channel?
1:55PM 1 Asterisk at ISPcon
1:51PM 0 Default time zone for asterisk
12:42PM 6 Boosting Shared Internet Bandwidth for Asterisk
12:37PM 0 Configuring a Grandstream 486 Device with AOL Internet connection
12:02PM 1 no music on hold
12:00PM 1 TE110P without router ???
11:50AM 0 Phone keypad input not working during "menu's"
11:21AM 1 Re: Grandstream ATA 286 and ilbc (Anton Krall)
11:14AM 0 Re: Asterisk-Users Digest, Vol 10, Issue 154
11:06AM 1 RHEL 3
10:58AM 1 Expression in Extension
10:08AM 7 Cisco Call Manager & Asterisk for Voicemail
10:02AM 0 Connecting an External Extension
9:46AM 0 Can't make outgoing calls
9:43AM 1 ACD Methods
9:33AM 1 (no subject)
8:45AM 2 Two TDM04 with Poweredge
8:26AM 0 Selling: E100P interface card
8:23AM 1 User cannot dial
7:47AM 1 New IAXy from Digium
7:24AM 0 tdm400p fxo not working
7:14AM 1 chan_capi patch eicon
6:52AM 1 Do Both! :) Re: Telecom SIP termination vs. DS3
6:50AM 3 Public vs. Private Network
6:09AM 1 OT: carrying a router, firewall, switch, ser ver, some phones with me on flight to Europe
6:03AM 5 MusicOnHold probelms
6:02AM 1 Random Blip
5:51AM 5 Deleting Monitor Files After 2 Months
5:38AM 1 retail unit for cards
4:56AM 3 asterisk-oh323 build problems
4:55AM 0 asterisk-oh323 building problems
4:12AM 1 Manager Port
3:53AM 1 Asterisk real time extensions problem...
2:54AM 1 Newbie X100P question
2:54AM 1 GOTO statement in Realtime-Extensions not working like expected
2:15AM 2 Forbidden - wrong password on authentication for NOTIFY
1:26AM 1 ser+asterisk problem
1:16AM 0 dail out with SIP through a second server
Wednesday May 18 2005
11:16PM 0 Telecom SIP termination vs. DS3
10:51PM 2 Spanish TTS
10:50PM 1 Grandstream ATA 286 and ilbc
10:39PM 0 Cisco ATA question
9:55PM 4 OT: carrying a router, firewall, switch, server, some phones with me on flight to Europe
9:38PM 0 MeetMe -1 return Code - howto
9:25PM 0 Tellabs Consultant Wanted
9:18PM 0 FCC Will Force VOIP E911 in 120 days ?
7:43PM 1 Most stable HEAD
7:42PM 1 Invalid sip contract
7:24PM 5 SIP Phone Recommendations?
7:23PM 0 Missing Transfer Command (asterisk CVS 20050518)
6:26PM 1 realtime versus static
6:01PM 0 send a text message from a phone get - 405method not allowed error
5:55PM 0 asterisk hung up the line after 10 minutes rightafter a beep beep beep sound
5:00PM 0 FW: No ASA statistics from call queue and CTI screen pops.
4:42PM 1 Extensions Issues.
3:54PM 2 FREE music for downloading
3:49PM 4 Pickup other ringing phone
3:46PM 0 Ideal Machine
3:06PM 2 Run Script when originator hangs up the phone
2:19PM 1 Follow Me solution
1:59PM 0 Asterisk and H323 vs OH323???
1:53PM 4 Outbound dialing issue with FXO
1:52PM 1 need 7960 power cubes
1:50PM 1 Nearing my wits end....bad switch???
1:36PM 0 Zaptel on Dell Poweredge 1850 with RH Kernel 2.4.21-15
1:22PM 0 Forward calls from PSTN to PSTN very choppy
12:30PM 1 asterisk hung up the line after 10 minutes right after a beep beep beep sound
12:25PM 4 FXO Gateways
12:25PM 1 send a text message from a phone get - 405 method not allowed error
12:22PM 2 Grandstream GXP-2000 and good support
11:57AM 7 Soft Phone
11:52AM 1 PBX integration call status-Calls do not show as connected
11:13AM 3 Recommend a good SOHO NAT Router
11:11AM 0 Call classification with Asterisk
10:46AM 0 question about VoIP headsets used by other call centers
10:39AM 1 No ASA statistics from call queue and CTI screen pops.
10:36AM 1 Issues with Polycom 1.5.2
10:23AM 0 SIP: Failed to authenticate
10:13AM 1 Agent Queues and Sending URLs
9:46AM 3 connecting a sipura sip device to asterisk before dialing any digits
8:43AM 1 Small office setup with Asterisk @home, IAX and analog termination
8:30AM 0 [Asterisk-Dev] Re: SigSeg in channel.c / chan_mISDN problem ?
8:05AM 0 Re: SigSeg in channel.c / chan_mISDN problem ?
8:03AM 0 Integrating Asterisk into our Legacy PBX <-- Newb (correction)
7:53AM 2 Best Compression Available
7:22AM 0 Softphone Requirements
7:17AM 0 RTFriendsCache=yes help Voicemail MWI help
7:12AM 1 Mysql cmd with Asterisk Problems
6:19AM 1 Audio flutter on OH323 output?
6:07AM 0 Integrating Asterisk into our Legacy PBX <--Newb
6:01AM 0 listening at 5070
5:42AM 5 Polycom Instant Messaging
5:22AM 1 SIP/nat situation
5:16AM 2 Traffic shaping for IAX and SIP calls through Asterisk?
4:31AM 0 Asterisk not recognising "On Hold"
4:24AM 2 Asterisk and Ericsson PBX
4:01AM 1 Asterisk H323 Trunk Zone
3:57AM 0 IVR/Voicemail, No Sound from Asterisk
3:54AM 0 Asterisk with Intel modems 537 or MD3200
3:49AM 2 FWD to Asterisk stops after 3 seconds
3:38AM 1 2 x Eicon BRI ISDN devices (UK)
3:34AM 3 DHCP, PoE, FXS, FXO and ONE power adapter ONLY???
3:31AM 0 HELP ME!!!! Asterisk don't do calls
3:26AM 2 Call forwarding...
3:21AM 1 eicon fdc3
2:34AM 6 zaphfc troubles
2:28AM 0 find free e1 channel
12:23AM 2 DEBUG output on sip extensions
Tuesday May 17 2005
11:07PM 2 Ubuntu Migration
10:51PM 1 OT: Multi-Format Sound Conversion Utility (and NOT sox, etc)
10:15PM 0 can't compile zaptel
9:37PM 10 VoiPSupply Dot Com
9:23PM 0 SIP, NAT and Asterisk
9:19PM 0 can't compile zaptel..
9:16PM 0 Asterisk Aborted
9:13PM 0 Debugging voice cutoff problems
9:08PM 3 Guest
8:55PM 2 fax soft client
8:49PM 2 how to get remote extensions to work correctly with a zap channel?
6:53PM 1 Display SIP useragents
6:14PM 0 Music On Hold problem: Read 392 bytes ofaudiowhile expecting 1600
5:37PM 0 Linejack
5:24PM 18
4:50PM 0 Agent Queues/XTen X-Pro/Multiple Call Appearance
4:20PM 1 Agent Login/Logout
4:10PM 2 Asterisk - Spandsp: fax header
3:49PM 0 Dropped calls with TDM400P - 4 FXO
2:34PM 1 Music On Hold problem: Read 392 bytes of audio while expecting 1600
2:13PM 0 Can't connect to SIP provider
2:02PM 1 call waiting signal
11:01AM 3 How much CPU power needed for asterisk
10:19AM 2 Compile Error - MySql addon
9:59AM 4 multiple sip accounts from same sip registrar
9:43AM 0 PRI Providers in San Francisco?
9:34AM 0 cdr from operator initiated calls
9:31AM 0 overlapdial timeout [bristuff]
9:04AM 0 Help on ODBC & Asterisk usage
8:52AM 11 Asterisk Fax
8:45AM 0 Asterisk and rfc2833
8:42AM 1 callgroup and callwaiting for IAX clients
8:37AM 3 Call Forwarding / Redirect with PRI
8:37AM 0 asterisk tapi
8:25AM 1 fdc3 no gsm
8:07AM 0 no audio for voicemail
8:05AM 1 Digium and Asterisk
8:00AM 1 One * server unavailable when multiple servers connected together
7:45AM 4 Is SKYPE a threat or should we do something (together)
7:42AM 1 Asterisk and rfc2833 help
7:33AM 4 peering with friendly networks, ...
7:13AM 0 Asterisk Realtime extensions configuraton..
6:51AM 0 jitterbuffer stability and use with meetme
6:49AM 0 Asterisk + digium
6:16AM 0 Failed to grab lock, trying again...
6:01AM 1 Obtaining Cisco Firmware painlessly?
6:00AM 0 Problem with getting the value of variable DIALSTATUS in AGI script
5:59AM 0 Background AGI
5:55AM 0 "Failed to grab lock, trying again..."
4:47AM 1 spandsp + HFC poor fax quality?
4:32AM 0 SMS & Grandstream ATA-286
3:58AM 1 sip show registry empty ?!?!!?
3:50AM 0 Using US Robotic router for 60 calls
3:35AM 1 Background() problem (with queue(), etc.)
3:09AM 0 Asterisk with PINs
3:02AM 2 Asterisk and Credit Card Machines
3:00AM 0 does NOT rtptimout work configrued localy for a peer ???
1:27AM 0 Junk at the beginning, Warning, flexibel ratenot heavily tested!
1:07AM 2 Junk at the beginning, Warning, flexibel rate not heavily tested!
Monday May 16 2005
10:14PM 0 Asterisk's clients logon failed if asterisk cannot register on its own sip proxy
10:13PM 2 Telephony keypad
8:55PM 4 Web Client with IAX2 and ilbc
8:08PM 0 Using prepaid calling cards to dial out with Asterisk - extensions.conf
7:50PM 1 Warning[3817] and REGISTER
6:30PM 0 DTMF asterisk-2-asterisk using SIP w/ dtmfmode=rfc2833
6:21PM 2 Asterisk and a D/42NS
6:08PM 1 GSM bandwidth
5:51PM 2 Help with extensions - can't dial 700
5:43PM 3 CLI and DNIS presented to Analog extension
5:00PM 1 Callerid not passing across IAX2 trunk
3:36PM 11 H323 to SIP
3:31PM 0 Queued calls
3:16PM 0 FW: Static on TDM Zaptel FXO
3:13PM 1 Dial plan - does not stop after first match
2:49PM 0 spandsp in 64 bit Linux on AMD64
2:04PM 10 Static on TDM Zaptel FXO
1:39PM 4 Forwarding To Cell Phones with Asterrisk PBX
1:25PM 2 Transfer of Calls Between Legacy PBX and Asterisk
1:06PM 0 Using PAP2 with g723
12:53PM 0 Outbound Faxes with spandsp
12:40PM 0 mysql debug
12:25PM 2 Pass variable to Authenticate?
12:20PM 0 Asterisk Fax On Demand using SPANDSP?
12:05PM 3 voicemail.conf from DB
11:34AM 2 outlook express intregation
11:02AM 2 Broadvoice Toll-Free IVR issues
10:53AM 3 Error running Make config on Debian Sarge
10:39AM 1 ShoreTel 210 MGCP phone drops calls with MGCP RSIP
10:33AM 1 Setting DID info for analog Zap channels
10:11AM 4 IAX jitter
10:04AM 0 WIP-5000 SIP Settings
10:02AM 4 Lucent TNT & ASTERISK
9:34AM 0 Asterisk - fax - spandsp <--older threadlet from Jean-Yves about fax corruption, *not* timing
7:51AM 1 Vonage users with Asterisk in UK?
7:40AM 5 xbox asterisk?
7:33AM 0 .call file
7:24AM 1 problems with asterisk starting from init.d
7:06AM 3 Need off-the-shelve PC for Asterisk Server
6:41AM 1 2 servers via PRI
5:41AM 2 Broadvoice: No Service, No Email reply but charging the credit card still works
5:18AM 3 cisco 3620 setup (newbie cisco alert)
5:12AM 0 IPS can now print and chartc
5:06AM 2 NAT and sip issues
5:05AM 1 pickup timeout
4:29AM 1 Always Ringing
3:37AM 0 chan_misdn and passive BRI cards
3:35AM 1 SIP-->h323 conversion
3:27AM 1 zaptel.conf in /etc not /etc/asterisk - historical reason?
3:19AM 2 callback problem
3:14AM 4 Asterisk@home 1.0 + Sipgate UK/SIP Provider
2:31AM 1 relocation error
2:09AM 0 Number Portability Details
1:23AM 1 Re: SpanDSP TXFax and multipage faxes problems (aditional info)
12:45AM 1 A hook flash sent using RTP for telephony signals (RFC2833) does not flash zap channel
12:08AM 0 Re: Asterisk-Users Digest, Vol 10, Issue 117
Sunday May 15 2005
10:34PM 1 Custom SIP messages
9:56PM 0 Bridge stops bridging channels SIP
8:43PM 0 Re: Asterisk-Users Digest, Vol 10, Issue 114
8:26PM 5 zttest
8:21PM 0 Hang up error: Didn't get a frame from channel
7:59PM 0 Bridge stops bridging channels
7:57PM 1 Old DBGet/DBPut vs. new Set(var=${DB(...
7:19PM 14 POE hub
7:18PM 0 Multiple Questions -- Please Help
6:26PM 2 Do sipura 200 and linksys pap2 ATAs send their mac address in REGISTER message?
5:52PM 1 Modprobe wctdm hang at command prompt
5:45PM 0 No Such host - IAX2 channel problem
5:29PM 4 Callerid on PC and more
4:53PM 5 FXO/FXS suggestions:
3:58PM 1 Compile problem on last CVS
3:20PM 1 can't CLI> STOP NOW by zombie MOH
2:23PM 3 knopsterisk
2:01PM 0 Known Working Motherboard/CPU for TE410P
12:55PM 1 Scalability of chan_oh323
12:08PM 1 Re: SpanDSP TXFax and multipage faxes problems (aditional info)
11:52AM 1 Asterisk@home backup/restore question
11:43AM 0 Several questions. Please help
11:22AM 2 Road Warrior phone config
11:17AM 0 SIP or IAX2 Web UA
11:14AM 4 Outgoing spool file ignored
11:02AM 2 SIP Gerenal settings conufsion
6:24AM 1 Re: SpanDSP TXFax and multipage faxes problems (aditional info)
4:40AM 0 AreskiCC doesn't log in
4:30AM 2 Voip Provider in Brazil
3:36AM 1 Problem with extensions and when channel is unavailable
12:22AM 5 AreskiCC
Saturday May 14 2005
11:29PM 6 ASTCC does not count all calls
8:58PM 2 Asterisk Guru help needed for DISA troubles
4:44PM 5 ser and asterisk
3:34PM 0 Cannot create a personalized unavailable message
10:21AM 0 Building OPENH.323 ERROR HELP PLEASE
9:50AM 2 Broadvoice outage times?
8:18AM 0 Transferring a call, IAX2->SIP, DTMF/RFC2833 doesn't work?
6:56AM 0 pbx autodiscovery
6:02AM 1 Help Please Multiple Users for Broadvoice
5:58AM 0 installing linksys pap2 and welltech lp302
1:02AM 2 How to connect two Asterisk servers
12:15AM 0 How to beark Queue() and jump to voicemailMain
Friday May 13 2005
11:11PM 1 IAX2 and FWD - Wrong context?
7:13PM 1 A@H Email Relay
6:57PM 4 Polycom configuration
6:42PM 0 asterisk dials random number when receiving incoming call
5:36PM 3 Audio quality
5:06PM 0 Echo problem on SPA-841
4:07PM 4 1-800 with FWD
3:54PM 1 Fax service (instead of tdm card)
3:12PM 1 CVS HEAD - FATAL: Error inserting wctdm
2:03PM 0 Caller*ID failed checksum?
1:20PM 0 Spawn extension -----what does this mean ?
1:17PM 0 Formatting problem in cmd sip show peers
1:14PM 3 Other memory stuff
1:12PM 1 MusicOnHold "zombie" mpg123 processes
12:54PM 0 wanted - asterisk reseller / integration consultant in NC
12:15PM 2 TDMoE emulates a T-1= Is there a product to simulate a PRI trunk? (Robert Goodyear)
11:56AM 1 Polycom IP 500 caller id
11:53AM 3 Poor volume on SPA-2100 due to asterisk?
11:38AM 6 64 bit
11:36AM 1 DTMF problems with International Calls
11:05AM 5 Is there a product to simulate a PRI trunk?
10:51AM 6 voip encryption options
10:51AM 0 Re: Interrupting voicemail with "*", dropping to "a"
10:28AM 4 Asterisk - fax - spandsp
9:48AM 0 Chanspy crash
9:02AM 1 broadvoice replacement
8:58AM 0 delay before call file execution
8:25AM 0 My experience with our VS-1 Asterisk server
8:17AM 0 Dropped Calls between Sip and Zaptel
8:16AM 1 Tyan Transport GX28 with TDM400
8:08AM 1 Why always getting "max retries" error during idle?
8:07AM 0 ISDN passive card (HiSAX driver) / Fax reciever
8:03AM 0 Zaptel and zttest
7:13AM 0 Extension never ring, goes straight to VM
6:44AM 0 Autodial and autoanswer
6:26AM 2 Asterisk extensions from Mysql
5:38AM 2 In/out calls from/to same sip provider
5:09AM 1 Help needed on setting up realtime
4:13AM 1 ASTCC Compilation Error
3:56AM 0 Unchanged sound through Asterisk
3:49AM 1 Re: SpanDSP TXFax and multipage faxes problems
2:17AM 0 Problem with IAX trunking
1:30AM 0 [Asterisk-Dev] Re: oh323 compile problem in FreeBSD
1:27AM 3 2 minutes pause before ring on H323 channel
12:50AM 2 About Voip Technology : RTP over TCP
12:00AM 0 Problem with calls on hold
Thursday May 12 2005
10:55PM 1 sipsak with asterisk
10:34PM 14 voipjet anyone?
9:43PM 0 Asterisk, SIP and NAT: Help needed!
9:16PM 0 Voicemails not deleting
7:32PM 1 Can the originator of a call transfer it?
7:01PM 1 realtime sip show peers no nat
6:24PM 2 ISDN Clock Source
6:22PM 2 Polycom IP4000
6:15PM 1 Queue/Agent recording and configuration
5:51PM 2 UNREACHABLE messages
5:26PM 3 Dead Polycom ip500
5:05PM 5 1-800 free calls
4:26PM 3 Interrupting voicemail with "*", dropping to "a" extension. Does it work?
3:52PM 5 French SIP or IAX phones
3:47PM 0 Chicago users/implementations
3:16PM 1 IPVolution release info....
3:05PM 0 Fix for increasing delay over time on non-Zap channels in MeetMe
2:25PM 0 FW: Incoming calls picked-up then simply hanged-up
2:20PM 0 SER Asterisk and NAT
2:18PM 2 Problem with Polycom SP 500 and Cisco PIX
2:16PM 4 Sound card Line-In as MOH source
2:13PM 0 Incoming calls picked-up then simply hanged-up
1:19PM 1 Asterisk with ShoreTel 210 (MGCP)
1:04PM 0 One sided sound outgoing only
12:57PM 0 Escape context and queue application
12:52PM 1 Re: Headset for Cisco 7960?
12:43PM 3 Giving user progress in an voice menu system
12:42PM 0 "Called" ID question - Trying again
12:21PM 2 VoiceXML
12:17PM 0 SIP authentication on outgoing call
11:50AM 0 "Called" ID on local extensions
11:36AM 2 AMP and dialparties.agi
10:54AM 2 Problems with Simpletelecom and *
10:51AM 4 Polycom Bootrom 2.6.2 and SIP 1.5.2
10:37AM 3 How to decrease Asterisk load
10:30AM 3 * Server
10:17AM 3 Something every TDMP user should know
10:10AM 0 Asterisk as a fax/voice switch
9:26AM 2 Cisco 7960 Can't be unlocked
9:21AM 2 IAX to FWD?
8:53AM 2 Best CPU config for dual-Xeon?
8:49AM 1 cdr!
8:47AM 2 Immediate Answer
8:39AM 1 Incoming context problem
8:27AM 2 GSM gateway for Asterisk
8:26AM 2 Voice Recognition - Cases of success
8:24AM 0 switch in extensions.conf
8:10AM 4 gnugk
7:45AM 0 ${BLINDTRANSFER} variable
7:38AM 0 Cellsocket with @home
7:20AM 5 chan_capi, chan_misdn and chan_modem
7:00AM 2 SIP and FastStart
6:43AM 2 Inbound ANI & DNIS format
6:38AM 0 Open Source MGCP Softphone
6:21AM 1 chan_capi and chan_misdn
5:53AM 1 FW: failure notice
5:48AM 1 ast_yyerror - 'space' in Caller-ID - string comparison
5:07AM 0 Show useragents?
5:03AM 0 delay before execution of call file
4:59AM 0 Connecting * to a PBX throught a PRI.
4:42AM 4 VoiceBlue GSM
4:32AM 0 Making Asterisk run on Mysql backend
3:25AM 1 Wrong password on authentication for Notify
3:13AM 6 Cisco contract for 7940/7960 firmware access
2:47AM 0 Wrong password on Auth for Notify
2:16AM 2 Voice mail - "Extension at" vs "Phone Number" OGM
1:43AM 0 SV: beginner in Asterisk configuration
1:17AM 0 Snap, Crackle and Pop with Dell 1850 and TE410P
12:58AM 5 beginner in Asterisk configuration
12:52AM 2 GXP 2000 Conference Button and ILBC
Wednesday May 11 2005
11:48PM 3 octtel SP 4220 gateway and Asterisk
11:31PM 2 Icecast
10:41PM 0 Dial out on ZAP channel
10:27PM 1 outgoing-call-logs to a text file
10:23PM 1 HELP: ASTCC (AGI) meets call forward ERROR
9:35PM 1 Anyone ever implement an *outbound* dial-by-name??
8:44PM 0 TDM400P with 2 FXS module fail to Hangup
8:15PM 1 re:oh323 driver compiling problem
8:03PM 0 digium card and fax
7:56PM 0 Re: Asterisk-Users Digest, Vol 10, Issue 83
7:34PM 1 ?????sip channel & AGI problem
7:31PM 3 Astlinux & AMP
6:19PM 0 Database of actve calls (as per astguiclient)
4:45PM 1 Echo from a mail loop in list
3:26PM 0 Seshu, on April 20, you said this about the Astcc & AreskiCC --> Re: AreskiCC installing assistance for seshu.kanuri @
3:15PM 1 New provider needed - any recommendations
2:59PM 1 Asterisk Video Conferencing Bounty bumped to $3, 000
2:51PM 3 Grandstream GXP2000 firmware update
2:15PM 4 Problems with VIA Chipset
2:12PM 12 Snom 360
2:07PM 0 TDMoE vs IAX2
2:01PM 0 dialparties.agi and @home
1:57PM 0 Mediatrix 1204 caller ID
1:55PM 5 IAX.CC/SixTel
1:50PM 0 Fw: pinout for"standard"telephoneheadsetrequired.?
1:26PM 0 Vegastream assistance?
1:20PM 1 high availibilty (heartbeats) - a good way to ensure automatic redundency?
1:18PM 5 Status of FAX
1:05PM 0 outbound proxy field in sip.conf
1:02PM 0 softphone buzzing
12:52PM 1 Asterisk @home with IAX termination...
12:42PM 1 Forcing Asterisk to not bridge/transcode RTP traffic
12:39PM 2 AreskiCC Install Problems
12:33PM 1 AreskiCC - Install Problems
11:56AM 1 ITSPs with good phone support
11:19AM 1 IAX and calls on hold
11:13AM 0 snom190 and SUBSCRIBE failures with 407
10:34AM 0 Sipgate incoming DTMF
10:23AM 7 Satellite Providers
9:59AM 0 Audio delays during file playback and zap channel activity
9:51AM 2 PRI QSIG and legacy toshiba intergration
9:48AM 0 Inbound Calls Codec
9:41AM 1 Trouble Connecting Xlite to Asterisk
9:26AM 0 Mass Deployment
9:25AM 1 Channelized T-1 (NOT PRI) Voice and IP mixed
8:53AM 3 Live Voip
8:42AM 1 oh323 driver compiling problem.
8:33AM 2 Realtime voicemail login incorrect
8:08AM 1 CDR and Postgres
7:47AM 0 Outgoing calls log in a text file
7:35AM 0 [SPAM] - RE: Grandstream-Budge tone - Email found in subject
7:27AM 0 wip 5000 and using write msg on the phone - anyone?
6:57AM 1 Grandstream-Budge tone
6:57AM 5 Voicepulse down?
6:52AM 0 T1 Card ------ Adtran ------- FXS BUG???
6:44AM 2 forum
3:16AM 1 Gateway service under Asterisk
2:23AM 2 Asterisk and Cisco AS5300 or 3600
2:07AM 0 TDM400P for UK
2:06AM 2 Sip or IAX2 eb Client
1:16AM 0 Predictive Dialier
12:50AM 2 Log Output
12:18AM 0 how to detect a hang-up in the first 5 seconds
12:14AM 0 SIPURA SPA-2000 webserver dead after firmwareupgrade
Tuesday May 10 2005
10:54PM 0 rtp.conf not working as expected
10:43PM 1 asterisk-addon
9:31PM 3 is it allowed to install 2 TE405P cards at same P.C.?
8:21PM 0 VoIP A-Z Carriers
7:33PM 1 Call Queue Priorities
7:28PM 1 Restricting connection of unauthorized phones.
6:50PM 4 SIPURA SPA-2000 webserver dead after firmware upgrade
6:16PM 1 MF and DTMF tones in the same AGI script
6:13PM 3 Voicemail Passwords
5:55PM 2 RE: Writing To Multiple MySql Tables
5:47PM 13 What do you name yours
5:06PM 0 Registering phones with the same/invalid extension number
4:53PM 0 Transfer from/to a queue
3:55PM 3 Setting Variables
3:38PM 3 Sizing a machine
2:52PM 3 Phone attached to Sipura SPA-1001 has no ring
2:15PM 0 AstLinux 0.2.6 Released!
1:26PM 0 Flashing DVG-1120M to DVG-1120S
12:47PM 2 Warning of the Asterisk server
12:40PM 0 problem with ControlPlayback
12:34PM 3 Asterisk and Avaya 4602 SIP phone
12:09PM 2 AAH 0.9
12:08PM 2 DS3 (T3) Card for Asterisk?
12:06PM 1 Limiting outbound calls
11:52AM 1 Zaptel problems on Debian
11:46AM 0 Re: Sipura 841 and headset (Josiah Bryan)
11:44AM 1 Asterisk PRI problems (Crashing when full)
11:41AM 0 outbound PSTN numbers over SIP failing
10:10AM 0 Incoming 800 Number
9:51AM 2 DISA
9:50AM 2 Manoj Shetty is out of the office. [Email checked- EMEA]
9:38AM 2 Cisco 837 router config
9:25AM 1 Spa3000 doesn't hangup after a conversation
9:23AM 0 how to get extension for ivr
9:14AM 0 ISDNguard
8:53AM 0 IPSwitchBoard version 0.115
8:52AM 0 extensions logon failed problem
8:08AM 1 AreskiCC + MySQL
8:07AM 0 AGI (LCR) within AGI ( possible???
7:55AM 2 Stun & codec
7:34AM 0 Cisco IP Phone 7912
7:32AM 3 MGCP : chan_mgcp.c:1509 find_subchannel
7:13AM 0 zt_rbs errors!?! never seen before.
6:56AM 1 Re: E1 (Digium E100P) problem : B-channel succesfully restarted
6:54AM 0 Asterisk Upgrade Path
6:45AM 2 Sipura 841 and headset
6:38AM 2 skype channel
6:16AM 1 Redirect to an application on other asterisk server
5:50AM 3 Interconnecting two lans using Asterisk over a PSTN
5:16AM 0 problem with mysql
4:04AM 2 E1 (Digium E100P) problem : B-channel succesfully restarted.
3:58AM 2 BYE from Cisco gateway
3:38AM 1 SIP transfers failing
3:07AM 2 outsourced pbx functionality- distributing calls evenly amongst agents
2:56AM 1 Problem developing my office
2:40AM 1 Cisco 7912G DST
2:15AM 0 Ericsson FCT f251m and polarity reversal
1:46AM 1 Group dial, first phone cannot pickup call. Cisco 7905 hangs.
1:42AM 0 cvs stable with db support in extensions.conf
1:32AM 1 BRI and PRI together possible?
1:13AM 0 Atcom AT-320 call forwarding - how?
12:34AM 0 spandsp configuration
Monday May 9 2005
11:31PM 1 Kphone-->asterisk<--Kphone
11:29PM 0 alcatel voip phone 4038
11:01PM 1 Asterisk + SER and NAT
9:00PM 4 Multiple Calls with Asterisk?
8:16PM 0 Central Asterisk Server and Asterisk VoIP Gateway
5:48PM 6 livevoip
4:51PM 0 [Feedback request] Web-MeetMe authentication
2:44PM 0 New IAXy available?
1:19PM 7 Will Asterisk do well in this application?
12:58PM 0 Consultants - Sydney Aust
12:24PM 0 Manoj Shetty is out of the office. [Email checked - EMEA]
12:17PM 1 Personal Communications Assistant
10:19AM 0 HELP... SER + Asterisk as feature server
10:08AM 0 RE: Asterisk at home with Broadvoice?
9:25AM 0 RE: Asterisk at home with Broadvoice?
9:19AM 0 AstriCon Europe: June 15 - 17 in Madrid Spain
9:18AM 1 Configuring SPA-3000 As A Trunk
9:04AM 0 New script: /usr/bin/asteriskdial + Kontact
8:43AM 1 asterisk lock up
8:36AM 3 ANNOUNCEMENT : AreskiCC V2.2 - Asterisk CallingCard Application
8:12AM 1 Interfacing AT&T Spirit System to Asterisk
7:50AM 1 Caller Name Database
7:48AM 0 SIP and MD5 passwords.
6:17AM 0 2 accounts on one Snom 220 with a queue
6:06AM 1 Asterisk DIDs configuration
5:49AM 0 SV: Re: Sangoma A102 cards testing FIXED
5:45AM 3 qozap(!) problem
5:40AM 1 Cisco ATA 186 with *70
5:33AM 2 sangoma fdc 3?
5:29AM 1 extension based on a dialed number?
5:21AM 0 Re: Sangoma A102 cards testing FIXED
3:25AM 8 Connecting 20+ asterisk servers together
2:20AM 0 SV: Re: Sangoma A102 cards testing FIXED
1:49AM 0 transfer queues agents
1:46AM 0 How to config call out pstn by sip in a meetme bridge application?
1:30AM 0 Mediatrix APA III-4FXO configuration
1:28AM 0 How can I send e164 ID to my gatekeeper?
1:19AM 3 Zyxel 2000W (WI-FI) Problems
1:14AM 0 AW: CAPI on ptp with variable length digits inphonenumber: SOLUTION for EICON
12:23AM 2 AGI - How to Make Calls and Bridge to Original Incoming
Sunday May 8 2005
11:32PM 0 help needed for PSTN
11:07PM 2 Background command noanswer option
10:25PM 2 Sangoma card !
8:34PM 2 HELP: how to get "To:" from AGI?
5:52PM 1 Help with Realtime & Seeding
5:16PM 3 Grandstream firmware
5:07PM 1 Cisco Mass Deployment
3:42PM 0 Restrict RTP ports - inbound and outbound?
3:09PM 5 8+ line receptionist only setup
12:42PM 2 detaching console from background asterisk
11:48AM 2 Just added snom Mass Deployment
10:00AM 4 Cellsocket help needed
9:30AM 2 Suggested Reading for VOIP
7:55AM 0 RSA question
5:38AM 0 Heavy CPU Usage During SPEEX Calls
5:26AM 1 RE: Asterisk at home with Broadvoice?
2:39AM 0 AreskiCC installation
Saturday May 7 2005
11:48PM 4 Setting variable for a context for all extensions?
11:32PM 2 What is the Polycom 301, 501 & 601?
10:20PM 0 Asterisk@Home on OnComputers Show Sunday morning
8:44PM 1 Setting the jitter buffer in AIX
8:30PM 0 Getting DTMF to work with SIP?
8:04PM 2 At home Asterisk via Broadvoice?
7:32PM 0 Problem Dialing out via external SIP account.
2:55PM 2 Cisco ATA 186 and Asterisk
2:31PM 0 Cisco ATA & Call Waiting
2:08PM 0 Polyco ip600 incoming ring time
1:48PM 1 WIP-5000 and DTMF
12:58PM 1 two questions about the Sipura 841?
12:15PM 0 cant connect
11:29AM 2 h323.conf - Asterisk not routing incoming calls based on IP - Ignoring type=user + host= + context=
9:37AM 0 Termination South America
8:52AM 2 Inexpensive FAX and 800 Number retail service
8:50AM 0 end user gui
4:18AM 0 IAX service provider with account balance announcement
3:11AM 0 DTMF generated from phone or from gateway?
1:58AM 0 DTMF detection with Adit 600
1:53AM 0 ChanIsAvail for MGCP
1:34AM 0 MWI Suggestion
1:32AM 5 Good NAT Pnp Hardphone
12:08AM 1 Echo Madness
Friday May 6 2005
11:46PM 0 Cisco ATA186 Fax problem solved:
9:40PM 1 Qos betwenn WIFI machines in LAN? oh323?
9:22PM 5 Who's happy with their voip service?
6:05PM 0 Setting ANI
4:54PM 0 Migrating to ODBC Voicemail
4:43PM 1 Upgrading to 1.x from 0.7 on Linux
2:31PM 1 Am I on the right track, and consultants
2:16PM 0 Wildcard TE110p initial setup
2:01PM 0 MWI on Cisco 7905
1:42PM 1 ZapBarge a PRI DDI
11:24AM 2 broadvoice NCFA numbers
11:17AM 1 sendtext to a phone that is off
11:06AM 0 Asterisk 1.0.7 and VIA EPIA
11:02AM 3 Review Outgoing VM Messages
11:00AM 0 Zaptel kernel modules correct?
10:50AM 0 RE: [Asterisk-biz] voip VPN solution requirement
10:13AM 1 Make error on ZT_EVENT_DTMFDIGIT
10:11AM 2 my_zt_write
9:57AM 0 Need your HELP: Avaya SIP phone 4602 and Asterisk
9:29AM 0 IPS version 0.114
9:26AM 0 WG: Newbie *@home + Xten.
9:19AM 0 To receive faxes on a dedicated extention and to forward them to a dedicated e-mail
9:12AM 2 HINT
8:57AM 2 Newbie *@home + Xten.
8:19AM 1 SIP NOTIFY retries exceeded.
8:08AM 0 ADTRAN Total Access 624 Work???
8:00AM 0 My Sangoma Experience in Asterisk: Followup
7:50AM 0 anyone experiencing half connections
7:36AM 0 OT: NAT traversal with SIP Paper
7:20AM 1 SEND TEXT to an extension?
6:54AM 1 Polycom 600 rollover
6:46AM 3 Web GUI
6:25AM 1 Operator Monitoring...flash operator panel?
6:17AM 1 CAPI on ptp with variable length digits in phonenumber: SOLUTION for EICON
5:53AM 1 Mitel SX200 integration
5:51AM 2 how do I register my Asterisk with oh323 on gatekeeper?
5:29AM 3 good bri card not junghanns
5:12AM 2 Transparently Routing German pri through Asterisk
5:09AM 0 Chan_misdn - cannot get the channel driver to load...
5:03AM 4 3 x TDM400P in one PC ??
4:43AM 1 Zapata.Conf Sanity Check
4:04AM 2 CAPI on ptp with variable length digits in phone number
3:59AM 1 oh323 compile problem in FreeBSD
3:40AM 1 DTMF oddity with OH323
3:35AM 1 Re: Sangoma A102 cards testing FIXED
2:46AM 0 OT: 911 service
1:55AM 1 Is such a thing as a analog (or even IP) video door entry system available?
1:26AM 1 IAX hint
1:11AM 1 IAXy Firmware Upgrade
Thursday May 5 2005
11:52PM 0 Polycom IP 600 not ringing
11:35PM 0 sccp transfer question
10:27PM 1 unknown RTP codec 72
9:38PM 1 Cisco device for gatewaying SIP to H323 suitable for ~50 simultanious calls
8:48PM 1 Connecting to provider
8:32PM 0 cdr_pgsql amaflags are always 3
8:20PM 2 CNAM lookup: new method for Caller ID Name delivery
6:19PM 0 Fax hangup causes incoming ring to be generated
6:18PM 5 snom mass deployment (probably off topic)
5:08PM 0 Cisco XML Parking Lot
5:00PM 4 Asterisk on Fedora Core 2 startup script
3:19PM 3 1800 DNIS and asterisk (HOW TO?)
3:15PM 0 iaxy dial out automatically
3:09PM 0 Zap Channel: CallerID feed failed
2:57PM 0 What is better? 2 lines of 128kbps or 1 line of 256kbps
2:16PM 0 Silly version question
1:19PM 3 can't create Zap channel
1:19PM 2 7777 (simulate incoming call) not working
1:06PM 1 Any thoughts on why I can't dial out my PRI?
1:04PM 0 How Two Asterisk Boxes Behind A Nat Initiate Calls
12:27PM 0 (res_)Monitor: wav - no sound; wav49 - sound
11:54AM 0 Alepo VoIP Billing with Asterisk
11:52AM 3 Account Code in all cases?
11:51AM 2 Question PSTN->VOIP forwarding and # of inbound calls
11:42AM 0 Voicemail call/dial out notification
10:30AM 2 Sayson caller id
10:24AM 0 Asterisk, Cisco 837 router and 79xx phones
10:23AM 1 Help needed with PSTN line
9:48AM 0 SIP forwarding to ZAP and call files
9:29AM 0 Optimal prompt format (gsm, ulaw, wav) for quality effeciency space
8:52AM 0 IAX2 monitoring
8:35AM 6 Opinions on Cisco 7960G, Polycom IP-600, and Snom 360
8:29AM 0 Re: Asterisk-Users Digest, Vol 10, Issue 39
8:26AM 1 MOH per User
7:34AM 11 Broadvoice "Issues"
7:24AM 0 Can I hide caller id on the fly (per eachusesetting) on Bristuffed * and quadbri
7:21AM 3 load_module fails: Fedora Core 3: SMP
7:17AM 1 Why switch from Asterisk@Home? was: Re: 7960'multi-line' configuration
6:36AM 0 Can call but not getting voice response with Cisco ATA186 behind nat
6:06AM 0 softphone for ipaq h4350
5:37AM 2 (OT) Interesting Product Vocera
5:23AM 2 Polycom 300 setup and AMP
5:20AM 1 Problem with Manager Originate and SIP extension
5:17AM 1 Realtime and Asterisk Database
5:07AM 11 FXO ATA?
4:43AM 1 Asterisk + GNUGK
4:22AM 1 test - ignore
4:11AM 2 Did nufone change allowed codecs?
3:32AM 0 problem with H323:Gatekeeper could not find user
2:33AM 5 Registering/Unregistering
2:15AM 2 mpg123 zombie processes ...
2:03AM 1 E1R2 Route
1:48AM 2 PRI debug
1:32AM 0 Compiling with GCC 4
12:23AM 2 Fritz Card sound quality
12:20AM 0 UniCall bugs
Wednesday May 4 2005
11:58PM 2 RED ALARM on PRI channel takes Asterisk DOWN
11:31PM 1 Working exten=> fax...
9:53PM 0 TE410P Drops Calls after many touch tones fromcaller
9:41PM 4 Problem with realtime SIP
9:12PM 2 TE410P Drops Calls after many touch tones from caller
9:04PM 0 max call rate (ingress direction) 1.00/30
8:15PM 0 Looking for Log parse for CDR's
7:46PM 2 10 digit dialing in Ft Lauderdale, FL?
6:42PM 0 SCCP and channel question
6:23PM 2 Connecting 2 * Together-Pulling hair out
6:20PM 3 QoS for improvements
5:58PM 0 Music on hold for agents and queues
5:25PM 2 Strange problem with G711/G729, Cisco and Grandstream
4:55PM 5 TE410P does not fit in motherboard
4:41PM 1 broadvoice not hanging up
3:43PM 0 Multitech SIP BRI Gateways
2:59PM 0 DTMF callerid does work
2:06PM 0 res_features: builtin monitor probs
1:51PM 0 Running explicit codecs between two hosts using distinct peers?
1:51PM 0 RANDOM command
1:48PM 0 Asterisk and prepaid sip clients.
1:38PM 0 Philips IS3030 Qsig
1:24PM 5 CDR for PSTN
1:10PM 4 Can I hide caller id on the fly (per each use setting) on Bristuffed * and quadbri
1:03PM 4 Voice mail Greetings
1:02PM 0 Passing CallerID outbound
12:13PM 0 parking in a specific parking spot with Asterisk 1.0X Is it possible????
12:11PM 1 FXO ports
12:11PM 7 TE410P on Dell 2650
12:05PM 2 IP500 Registration
11:23AM 0 TxFax and Tiffs
10:48AM 5 Aastra 480i
10:06AM 1 Company Signed Letter of Intent to Acquire LiveVoip, LLC
10:03AM 3 MEETME core uses ulaw?
9:53AM 3 Voicemailbox on Queue?
9:50AM 1 Attended Transfer using wrong Context
9:36AM 3 [Fwd: Call forwarding]
9:05AM 1 PRI timing problems: Fax & Voice
8:53AM 1 HFC: zapata + bristuff - how to set an outgoing number
8:38AM 0 new production server for SOHO installation
8:35AM 1 Cisco 7960: Builtin CFwdAll working?
7:37AM 1 TDM04B in a Mac
7:26AM 3 Philips - [QSIG] - Alcatel - [H323] - Asterisk -[SIP] - Users.
7:15AM 1 ackcall
7:04AM 0 Philips - [QSIG] - Alcatel - [H323] - Asterisk - [SIP] - Users.
6:28AM 4 Put a wait in a .call file.
6:21AM 0 GXP-2000 review..
5:55AM 1 ISDN transfer, handoff to masterswitch
5:14AM 0 IPSwitchBoard version 0.113 released
5:09AM 1 SetCallerPres problem
3:36AM 0 Newbie setting up LineJack Card
3:11AM 0 Zap (or carrier) issue ?
2:35AM 0 bristuff-RC8b-CVS
1:30AM 1 Data calls trough IAX?
1:29AM 0 RE:oh323 compile error
1:18AM 1 Difference between Asterisk and Asterisk@home?
1:12AM 1 Mysql/Radius Authentication
12:38AM 0 dial analog phone with sip
12:18AM 1 Asterisk and Post Paid Billing
12:11AM 1 oh323 compile error.
Tuesday May 3 2005
11:52PM 0 CODEC Allow statement help
10:49PM 0 broadvoice setup
10:40PM 0 sccp question
10:10PM 0 Messages while on hold was:RE: Digium MOH
9:10PM 0 iax native bridging
7:49PM 1 Asterisk dialplanner
7:38PM 0 IAX Dual Servers
7:31PM 0 MEETME core uses ulaw...
7:29PM 0 Grandstream, Asterisk and codec mismatch
7:26PM 0 MOH Core uses ulaw...
7:21PM 1 Hardware Capacity/Configuration
7:15PM 0 TE4XXP and /etc/zaptel.conf
6:38PM 0 asterisk not detecting call hangup
6:21PM 0 wip 5000 hitachi crossing subnets question
5:53PM 3 Audio quality problem recording calls using gsm codec
5:15PM 1 Is there any chance to bring Skype andAsteriskUser together?
3:35PM 0 chan_vpb Verbose Logging
3:28PM 0 Asterisk crashed
3:20PM 0 app_dbodbc or current recommendation for odbc methods
2:58PM 0 Thanscoding and MoH questions
2:48PM 1 ztcfg at boot time
2:37PM 0 Queues Member Types
2:20PM 0 Re: LiveVOIP
1:52PM 0 Cisco 7970 blank screen timeout
1:35PM 4 Good web interface for the enduser
1:30PM 0 ast_readstring replacement for res_perl
12:55PM 0 99% Usage buy Asterisk?
12:00PM 1 Directory for Polycom 600
11:55AM 0 Netweb 401 short review
11:17AM 1 How do "take away" do not disturb from certain phones
11:15AM 1 Any useful results?
11:05AM 0 Fwd: SIP over IAX2
10:59AM 7 Digium MOH
10:47AM 1 IAX2 attended transfer on 1-0-6 Stable
9:58AM 0 some soft phones only talk to default context of asterisk
9:54AM 3 bad CLI colors? bad terminal?
9:44AM 4 zttool: BLU/RED Alarm
9:22AM 1 monitoring which IVR extension is pressed
8:58AM 0 qozap message error
8:48AM 12 TDM users: modified zttest.c for testing
8:40AM 8 Freak incidents, who's to blame?
8:32AM 1 Mediatrix 1204 Help
8:25AM 6 Light weight and slimmed Asterisk
8:18AM 1 30 button vip 1 way audio
8:01AM 1 invalid frame size for G.729( 2 bytes)
8:01AM 1 xpro codecs and asterisk
8:00AM 0 Forwarding incoming calls via SIP
7:51AM 7 Voice Quality
7:31AM 8 IP Phones for home use?
7:20AM 1 Multi-tenant Setup
6:48AM 2 [SPAM] - Re: cisco7940 upgrading problem - Email found in subject
6:40AM 0 Re: LiveVOIP
6:03AM 2 zaphfc dialout problems
5:40AM 2 cisco7940 upgrading problem
5:37AM 4 asterisk to analog pbx
3:17AM 1 Strange area codes when dialing outgoing calls on EuroISDN E1
3:15AM 1 Is there any chance to bring Skype and Asterisk User together?
3:01AM 1 Very weird behaviour of Asterisk and SIP
2:33AM 1 Group redial
2:28AM 0 Voice Transfer of a Call Works only in One Way
1:52AM 0 Asterisk connects to ISDN via Fritz!Box Fon 7050 anyone?
1:43AM 1 How to display info from Asterisk on/to the phone ?
1:02AM 2 SIP NAT Polycom
12:30AM 1 Fwd: config for call pstn from voip
12:04AM 2 SIP and CVS Head
Monday May 2 2005
11:36PM 1 email notification when leaving a message
11:18PM 4 DELL 2800 : PCI Parity error
10:18PM 3 Ast 1.0.7, IP-500's with unmanaged switch...remote end missing bits of audio
10:02PM 2 how stable is oh323 ?
9:58PM 1 Wiki Trouble?
9:09PM 0 Regarding asterisk-dev list
9:01PM 3 BSD Compatability
8:57PM 0 DTMF in Voicemail
8:20PM 0 how do you get rid of Spawn's
8:17PM 2 Re: LiveVOIP
7:42PM 1 signaling table of E100P Digium Cards
7:21PM 1 Detecting Fax and bad CDRs
7:02PM 1 Sip Group
6:35PM 1 Asterisk CDR - Mysql
6:21PM 2 LiveVOIP troubleshooting
5:48PM 1 External Voicemail Access
5:18PM 4 Anyone else having Broadvoice issues today?
4:56PM 0 Queue Event
4:20PM 1 automated availabilty testing
4:19PM 0 mp3 problems
3:11PM 2 processing power measurement?
3:08PM 0 Re: Your message to Asterisk-Users awaits moderator approval
2:18PM 0 Adtran 600 config
2:14PM 2 Fedora Core 3 & Shorewall Install
2:12PM 0 hint does it work?
1:41PM 1 Taking asterisk out of the media path - SIP - how is it achieved
12:10PM 0 codecs, asterisk, xpro
11:46AM 7 voicemail volume with sipura 3000
11:44AM 4 Debuging SIP
11:43AM 1 zaptel 1.0.7 problems (again)
11:35AM 2 Need help getting zap trunk to work
11:28AM 2 Things to backup:
10:52AM 4 7960 "multi-line" configuration
10:36AM 0 Re: Asterisk-Users Digest, Vol 10, Issue 10
10:02AM 2 Outgoing calls, X100P
9:59AM 0 Asterisk CDR Bug Or Not?
9:23AM 3 Choppy Sound on PSTN End
9:04AM 0 Bug found in SJLabs SJPhone concerning dialpad
8:56AM 0 Polycom Sip TEXT Messaging
8:44AM 0 oh323 codec order
8:40AM 7 Please find me a IAX provider
8:24AM 0 Asterisk, h323
8:07AM 3 Asterisk as VM for Nortel System
8:07AM 2 IAX Timeout
8:02AM 0 large scalable voip setup
8:02AM 0 How to cancel a transfer in progress:
7:52AM 4 Chan_sccp - status
7:43AM 0 ExtensionState problems using Asterisk API
7:42AM 1 Pb SIP and port
6:49AM 0 Phonejack PCI-card
6:23AM 0 Problems with TDM400P card -correction to last post
5:56AM 1 X-Lite and callto:// syntax in webpages
5:36AM 0 Putting in an Application
4:50AM 2 extensions.conf dial plan
4:15AM 0 Loading ztdummy Stops MoH but Conference Works - VmWare
3:58AM 0 increasing delay in meetme conference
3:03AM 0 config for call pstn from voip
2:35AM 0 calling out through second server.
2:11AM 5 Diffrence bewteen FXO and FXS
2:10AM 0 Can anyone recommend some hardware for UK use?
2:06AM 0 Meetme and a timing source
1:45AM 1 chan_h323
12:36AM 0 unable to use addpac-ap200 (sip | h323)
Sunday May 1 2005
9:50PM 1 Sipura SPA2000 dialplan vs Asterisk dialplan
8:35PM 1 Caller Hears Ring During Attended Transfer?
7:47PM 1 which port is used when "asterisk -r "
7:45PM 1 how to disconnect a call manually
6:18PM 1 Set up SIP, now I'm getting a busy tone and weird (to me) messages on the console
6:14PM 1 post-dial variable for whoever answered?
5:32PM 1 asterisk and USRobotics Courier V.Everything
4:44PM 0 IAX channels do not disconnect
3:56PM 0 Latest CVS Head Nukes Server
3:10PM 1 Centos - Hylafax Install
2:24PM 0 Sipura 401 Unauthorized.
12:54PM 0 Problems in new implemenation....
12:24PM 0 TDM400P does not detect hangup on UK BT analogue line
12:20PM 1 Pre-Parse Extensions.conf?
12:11PM 1 Make Webvmail Error
11:39AM 1 Audio cut off at beginning of call
11:05AM 1 4 - 8 port w/QOS switch for Asterisk
10:35AM 2 TDM400P Power Connector
10:17AM 1 zaptel.conf multiple devices
10:00AM 0 Asterisk 1.0.6 stable IAX2 Firefly supervised call transfer?
8:49AM 2 TFTP question
8:31AM 1 Playback() stops working.
8:23AM 4 Dutch SIP or IAX numbers
7:38AM 0 sip based fax client software
5:30AM 0 IPS Version 0.112 released
2:27AM 1 [Announce] New chan_sccp release adds support for Cisco 7970
12:54AM 1 mISDN error while compiling
12:43AM 2 Cisco 7960 SIP Reject Call Option
12:16AM 1 Sip calling errors