asterisk users - Sep 2006

Saturday September 30 2006
8:54PM 6 Dial-9 (was Extension Numbering)
6:08PM 1 Install issues on Freebsd 6.1 with Asterisk 1.4.0-beta2
12:36PM 2 Where are the kernel sources?
12:27PM 3 Google talk and Asterisk 1.4
9:49AM 0 Asterisk Fax Server
9:42AM 0 PRI channel becomming unavailable
9:12AM 3 Issues with Monitor in 1.4?
4:10AM 5 How do I reset a password?
1:30AM 1 Realtime; User not registering {Urgent}
Friday September 29 2006
8:07PM 5 480i phone: Is there a trick to registering with *??
5:22PM 18 SIP phones not talking
3:44PM 0 FW: 480i phone: Is there a trick to registering with * ?? <--forg ot my sip.conf
3:39PM 0 480i phone: Is there a trick to registering with * ??
1:13PM 0 Fiber Outage: LA->Washington
12:44PM 0 native sounds
12:05PM 0 A Step back with AddQueueMember() ?
11:54AM 2 Problems with DISA
11:50AM 0 Re: asterisk-users Digest, Vol 26, Issue 172
11:20AM 2 recommended application for salesman using asterisk
11:12AM 0 Queue and Pickup/DPickup
10:25AM 0 manager api redirect dropping calls
10:20AM 1 Asterisk IVR .wav issue
10:10AM 0 [Fwd: [Fwd: [Fwd: asterisk-users Digest, Vol 26, Issue 166]]]
9:49AM 0 SPA3000 register in asterisk
9:42AM 3 Any Suggestions for Election Polling Application?
9:13AM 6 real time billing system
7:49AM 0 What minimum required packages for 1.4
7:32AM 0 Off-hooking Snom hanset doesn't answer incom ing call
7:31AM 0 [Fwd: asterisk-users Digest, Vol 26, Issue 166]
7:23AM 10 Extension Numbering
6:33AM 0 attended transfer unreliable
6:11AM 0 Polycom IP430 HUM and Echo
6:07AM 4 Sip answer one side , ring other side
5:54AM 3 Replacing mpg123 with madplay under Solaris?
5:16AM 2 VMware and Digium TDM400P card
3:30AM 1 Off-hooking Snom hanset doesn't answer incoming call
2:51AM 0 re: asterisk/SER integration - HELP
2:33AM 0 Difference between "SIP Server" and "Outbound Proxy"
2:30AM 1 Sirrix to Legacy PBX
1:08AM 10 pstn failback
12:13AM 0 Is this phone any good
Thursday September 28 2006
9:03PM 3 t1-pri or sip trunk?
6:32PM 0 SPA3000 and asterisk
3:22PM 0 Digium G.729 codec binaries updated for Asterisk 1.4 on Solaris
2:46PM 0 7970G SIP8-0-4 not registering with asterisk
2:10PM 7 Forcing Transcode
12:47PM 0 Asterisk lockup and boot error after one day?
12:23PM 4 Queue AddQueueMember()
12:17PM 2 WAS: 64 analog phones NOW: Selection criteria and recipie for a good Asterisk install [long]
11:33AM 0 You are not the next caller
11:18AM 1 Yet another processor question
11:02AM 1 Polycom Queues, Login, Logout etc
10:56AM 0 AstriCon 2006 Reminder / Hotel Selling Out
10:37AM 7 Cisco CAll Manger and H323
9:27AM 0 Voicemail callback bug?
9:11AM 2 T1 incoming connects, but no sound
8:14AM 0 Early rtp bridge and reINVITE in 1.4b2
8:12AM 4 Asterisk -> Tekelec T6000 (Vocaldata, voiss)
7:45AM 11 WAS: 64 analog phones NOW: Selection criteri a and recipie for a good Asterisk install [long]
7:32AM 6 Polycom 501 One-way Audio
7:29AM 0 SV: txfax reliability on TDM cards
7:05AM 11 extensions.conf strangeness
6:50AM 0 txfax reliability on TDM cards
6:36AM 3 unable to call AT&T audio conference bridge
6:24AM 2 asterisk on 2.4 kernel ... scheduler problem?
6:03AM 1 New ptlib dependency-requirement in SVN-trunk?
6:02AM 1 Master.csv has stopped writing call logs.
6:01AM 0 X100M location in circuit requirement?
5:28AM 0 Asterisk <=> E1 <=> Alcatel OXO
4:54AM 2 Is this phone any good?
3:55AM 2 How to enable jingle in 1.4beta2?
3:40AM 3 7940 vs. 7941
3:24AM 0 No answer time
3:04AM 0 Sangoma a301
3:03AM 0 get the value from CDR
2:34AM 0 How does SIP work?
2:06AM 4 quadbri + tdm400p + modem-fax
1:23AM 1 importance of crc4 in zaptel.conf?
12:52AM 4 Multiple asterisk same GUI
12:35AM 0 corrupt faxes
12:15AM 2 pay as you go t.t38 fax termination and origination
Wednesday September 27 2006
11:44PM 0 media stream count
8:08PM 6 Asterisk Hangups on PRI Interface
5:34PM 2 problem with trying to use two extensions for different announcements
4:22PM 1 SIP on Asterisk, new install
3:37PM 0 MWI on 1.4 Beta
3:34PM 2 SMS Text Send working with BT Text in the UK??
3:24PM 1 How to detect dial tone on ZAP channel before dialling using TDM2400P
2:06PM 2 UK Colocation services
1:27PM 2 Are you using app_meetme or app_conference
1:20PM 6 Spurious hangups on zaptel interface
1:13PM 9 RPID
12:40PM 1 TDM400P Problem or Not?
12:15PM 4 RE:T1 timing errors Nortel 61C with TE110P
12:12PM 0 Cisco ATA escaping # into %23?
11:41AM 0 txfax question
11:34AM 0 Linksys/Sipura 3K, Calls Timing Out
10:00AM 0 Any suggestions about VoIP provider?
9:53AM 0 FW: Re: asterisk to cell phone network
9:45AM 1 Queue Status via Dialplan
9:42AM 0 Zapata.conf
9:02AM 13 SER with multiple asterisk deployment
8:04AM 0 TDM04B Installation Problem
8:03AM 2 IAX phones?
7:42AM 0 Outgoing DialPlan
6:29AM 4 Good Book on Asterisk
6:06AM 7 Right way to prevent analog channel from answering the phone?
6:05AM 1 T1 timing errors (Frame Slips) on Nortel 61C to TE110P
2:56AM 2 ISDN30 and digital phones
2:26AM 0 High CPU usage when Internet goes down
1:08AM 0 Re: Advice of charge
12:49AM 0 How can I unistall Asterisk?
12:48AM 0 Configuring Asterisk 1.4-beta2 to work with jingle
12:44AM 11 Voip Buster - CID
12:30AM 0 my (SIP) INVITE is ignored
Tuesday September 26 2006
8:50PM 0 WARNING: chan_sip.c add_realm_authentication: ???
8:43PM 2 Anybody have the opvx1200.c driver?
7:57PM 0 queue information
7:53PM 7 max number of devices in hint
6:31PM 9 How to change pager notification message
4:18PM 1 Priority "n"
4:12PM 0 Grandstream GXV 3000
4:10PM 1 Context default & incoming ENUM
3:43PM 0 Problem with "Background" DTMF detection with A200D
3:33PM 10 se├▒alizacion te110p, signaling te110p
3:12PM 1 mISDN, 2 Billion HFC ISDN cards, cannot dial or receive
2:46PM 1 speaker phone echo
2:26PM 1 IAX2 & SIP Monitoring Solution for Asterisk
12:05PM 1 TE406P not working on Intel D101Ggc motherboard.
11:50AM 0 voicemailmain menu
11:35AM 0 Rewriting CID number w/o changing CDR src field
11:11AM 0 H323 IP phones
10:50AM 3 SIP Gateway
10:41AM 2 Included context
10:23AM 16 PRI Outbound CallerID Question
9:48AM 11 Asterisk behind Sonicwall firewall
9:18AM 0 X100P Clone card in JAPAN
8:04AM 0 Is there T.38 support on asterisk 1.4 beta2 ???
7:55AM 0 Play wav file during conversation
7:30AM 1 Asterisk 1.4 mohsuggest
5:32AM 5 How can I stop lost DNS from killing Asterisk?
3:14AM 0 error 407 authenticate on INVITE
3:06AM 3 asterisk 1.4 branch and chan-capi-0.7.0
2:40AM 5 asterisk - alcatel
1:55AM 2 Cannot make outbound calls to some numbers with PRI line from legacy PBX thru Asterisk with Sangoma
1:34AM 1 core dump with and chan-capi-cm 0.6.5
12:54AM 0 Asterisk and UMTS phones
12:53AM 0 The best way to track no-audio calls
12:52AM 0 Changing the recording dir of MeetMe recordings.
Monday September 25 2006
10:49PM 0 Unrecognized frames
9:31PM 1 Ericsson MD110
8:26PM 1 rtc: lost some interrupts at 1024 when loading ztdummy
4:30PM 3 "does /var/run/asterisk.ctl exist?" -- but Asterisk *is* running.
3:55PM 0 fw: Uniden - TVUNIDEN_UIP300
3:32PM 3 MOH in 1.4 - Still Broken?
3:01PM 1 Shared Line Appearances in 1.4
2:56PM 2 Extensions busy/congested and "circuit-busy"
1:12PM 1 Network impairment tools
1:02PM 1 include "context"
12:56PM 0 How to stream audio to external app for speech recognition and recognize dtmf in parallel ?
12:46PM 7 TDM2400P vs Sangoma A200
12:07PM 9 trixbox t38 pass through
10:19AM 0 can someone recommened a reliable, cheap t38 origination/termination provider
10:06AM 0 Asterisk 1.4 autoconf and /etc/asterisk directory
10:02AM 3 Got SIP response 415 "Unacceptable Content-Type" back from
9:57AM 1 DUNDi Servers
9:53AM 0 PBX TDA620 AND TE110P
9:38AM 23 Running Multiple Instances of Asterisk
7:35AM 7 asterisk to cell phone network
7:31AM 1 Queue failover and wrap time
7:25AM 15 OT: Opinions on Aastra 480i CT?
6:54AM 4 ztcfg / X100P question
6:15AM 2 progress problems from SIP to PRI
5:45AM 0 Asterisk Trunk with Alcatel 4200 PABX
4:05AM 3 Line Pickup Problem
3:16AM 0 A Strange doubt and problem
3:13AM 2 Cisco 7970 - DTMF
3:02AM 7 voicemail greeting
2:40AM 0 AgentCallbacklogin in Asterisk1.4 beta2
1:06AM 0 ougoing calls problem
12:43AM 1 Snom MWI not turning off when message picked up.
Sunday September 24 2006
11:37PM 0 High utilization with SIP registration
8:06PM 2 Need a recommended T38 FOIP solution
4:00PM 1 Rpath PoundKey 1.2
1:58PM 10 spandsp (foip)
11:31AM 0 how to configure a sip service
11:05AM 7 2 CPU's, Only 1 taking IRQ's
10:33AM 5 Missing sound in spanish from 1.4 beta2
10:24AM 1 iaxy: one way audio
8:37AM 0 Asterisk+Astbill
6:43AM 0 dialplan for confrencing
5:46AM 0 running ooh323 on asterisk-1.14beta2
Saturday September 23 2006
9:04PM 3 Segmentation fault on Asterisk startup: problem?
8:45PM 2 e911
7:26PM 1 fax over ip
1:00PM 0 lumenvox speech recognition
12:23PM 4 Problem with zaptel 1.4b2 and X101P Wildcard
12:12PM 0 Conference Call Delay & Quality
12:08PM 0 Debugging and Outbound SIP Trunk
11:46AM 0 One server SUBSCRIBE for information on multiple voicemail boxes
11:44AM 0 Connecting Motorola VT2442 Device
11:24AM 2 libpri current extracts as beta1
8:43AM 9 1.4 Beta 2 Config Problem
5:18AM 12 Trixbox Documentation
4:57AM 3 Cisco 7960 Double Natted
Friday September 22 2006
11:38PM 1 Polycom phone help needed
11:36PM 7 OT But So Ungodly Important
8:02PM 2 iaxy will register, but doesn't detect POTS line
5:10PM 2 Comments on new system plan.
5:03PM 1 make error
3:44PM 1 OT: Anybody remember this from last Dec?
3:22PM 3 Leased line interconnect
2:00PM 0 Polycom (and others) digitmap info
1:27PM 13 Very high ping times from 7960 phones
11:24AM 6 Digium G.729 codec binaries updated for Asterisk 1.4 beta
10:46AM 1 channel.c: Nobody there, continuing...
10:38AM 0 ZAP: psuedo camped on channel 1?
10:12AM 0 Asterisk 1.4-beta2 Spanish Sounds missing vm-youhaveno?
9:37AM 2 SNOM 320 - 404 "Not Found"
9:34AM 1 Re: [asterisk?users] Integrating Asterisk with LDAP Realtime
9:12AM 0 chan_isdn / chan_sip problems
8:31AM 2 Display message on voip phone...hint?
8:23AM 0 Asterisk & MSN ?
8:16AM 1 dialout-trunk vs. dial group
6:54AM 0 experience with phones locking up uniden and cisco
6:27AM 0 Re: [asterisk-dev] To bweschke regarding app FollowMe
6:23AM 15 hint status from dialplan?
5:48AM 1 Help with Tieing Outbound calls to Zap Channels
5:47AM 6 Re: [asterisk-dev] To bweschke regarding app FollowMe
5:41AM 0 Asterisk ramdonly crash using Realtime Static
5:39AM 1 Question about SVN-trunk-r43322 and Asterisk Recording Interface
4:46AM 1 Polycom phone references needed
4:10AM 0 How can the User Know he has voicemail in the Databases.
3:48AM 1 Does Asterisk 1.4 going to support realtime ex-girlfriend logic?
2:42AM 0 Where to find error codes
2:31AM 3 freepbx dial plan, add and remove at the same time
2:27AM 18 64 analog phones
2:17AM 3 ATA with wireless client
2:04AM 0 INVITE re-try interval
1:52AM 0 Iax2 show netstat
1:46AM 3 alternatives to mpg123: format_mp3, rawplayer or madplay?
1:39AM 6 new in 1.4?
1:04AM 0 E1 - PCI-Express
12:42AM 15 Dual core
12:16AM 3 Fax Detection on outbound call
12:00AM 0 Picking up a call from queue?
Thursday September 21 2006
11:39PM 1 Re: Can you explain why multiple registration isan important (missing) feature ?
11:19PM 2 Dynamic DNS asterisk server?
10:28PM 2 Application of Asterisk Packetization Patch
10:04PM 0 iaxy configuration problems
9:40PM 1 multiple zaptel cards
8:24PM 0 zttest output
8:19PM 3 iaxyprov downloading problems
4:21PM 3 SOS building fastagi C
4:16PM 5 Any one knows if I can connect SS7 to Asterisk, (with a TE405P or other) ???
4:12PM 0 TDM2400 problem isolated with POLYCOM IP301 phones!!!
2:45PM 3 Integrating Asterisk with LDAP Realtime
2:13PM 0 Dual asterisk, CallerID(name|number) problem
2:11PM 8 DSL router with integrated SIP proxy?
1:30PM 4 IAX or SIP termination provider that reaches6421xxxxxxx?
12:46PM 0 Extensions problems
12:23PM 2 SPA941 -> Asterisk -> Voip provider -> PSTN -> ShoreTel garble
12:16PM 0 Asterisk, Asterisk-Addons, Zaptel and Libpri 1.4 betas released!
11:49AM 0 Mini call center only 15 seats fxs to sip suggestion
11:39AM 0 mISDN problem: no version for "capi_cmd2str" found
11:39AM 0 Asterisk 1.4 Beta Uploaded!!!
11:21AM 2 asterisk and PowerEdge 1950
10:51AM 3 TDM2400 wired description and skiping frames
10:38AM 0 Asteisk plays music on hold startingfrom randompoint
10:04AM 0 Asterisk and Panasonic D500
10:00AM 3 Linksys SPA400
9:41AM 0 Polycom 650 Question
9:39AM 2 Call is dead after featuredigittimeout
9:36AM 3 notransfer local channel on redirect
8:41AM 9 CURL
6:48AM 1 Asteisk plays music on hold starting from randompoint
6:44AM 12 Looped message playback
6:33AM 8 Setting QOS settings in asterisk and/or CentOS?
6:04AM 0 Calls between IAX2 Clients don't work correctly
5:13AM 2 Using Asterisk with IVR connected with legacy pbx via rs-232
3:19AM 15 asterisk, iaxmodem, hylafax quality problem
3:16AM 7 Two phones, same number
2:58AM 0 Invite issues
2:55AM 4 asterisk / chan_capi problems
2:47AM 1 asterisk skills in the philippines
2:24AM 0 Habitual set of number
1:40AM 7 Iax Netstat Output
1:26AM 1 Unexpected delay: problem with outgoing calls
12:21AM 6 RTCP and RTP packetization in 1.4
12:19AM 0 How much SIP calls can I squeeze from this box
12:01AM 1 Help in Reloading of Asterisk...
Wednesday September 20 2006
11:27PM 1 Re: Can you explain why multiple registration isan important (missing) feature ?
7:30PM 0 Call/Voicemail Screening
6:50PM 4 Polycom 2.0.1 Software
6:24PM 2 Configuring Codecs
4:45PM 0 X100P compile problems
4:10PM 0 No Sound from VoicemailMain - Device Linksys PAP2T-NA
3:54PM 0 voice detection during playback
3:35PM 0 Asterisk Bussiness Edition and Realtime.
3:07PM 0 Re: Can you explain why multiple registration isan important (missing) feature
1:45PM 0 macro-dialout-trunk without agi or manager
1:04PM 1 A Caller ID question (UK)
1:02PM 3 Cisco 7970 behind NAT
1:00PM 6 PRI Backup
12:56PM 0 ztdummy installed but choppy audio warning
12:11PM 0 Round Robin + Ringall
12:11PM 0 RTT in rtcp debug
11:25AM 0 SPA-3102 PSTN->VoIP Gateway (quest for one stage dialing)
11:06AM 0 Asterisk Voicemail with Sonus?
9:18AM 3 Asteisk plays music on hold starting from random point
8:56AM 1 Getting Music On Hold working in * with Fedora?
8:54AM 0 A-Z termination
8:45AM 0 No channels available after reloading config
8:39AM 2 (no subject)
8:39AM 7 HINT problems with SVN-trunk-r43322
8:32AM 0 How to register from asterisk server to an xlite.
8:26AM 0 Available channels
8:16AM 0 Zap channel digit.
6:59AM 1 Asterisk capabilities, was University dumps CISCO VoIP for Asterisk
6:41AM 0 Unexpected delay
6:38AM 0 tx_fax over sip to TDM card
6:27AM 4 Sip configuration using mysql
6:23AM 2 MOH distorted on Pound Key Linux on asterisk1.2.8
6:16AM 1 Realtime madness
5:48AM 2 Incoming calls, identify
5:34AM 0 Forwarding the Ring Group and Calls coming in to Queues
5:16AM 2 enumlookup - deprecated working - but appreciated one duznt :-(
5:03AM 10 University dumps CISCO VoIP for Asterisk
4:29AM 0 Register doubt
3:55AM 0 Registration doubt
3:33AM 0 Mediant 1000
3:09AM 4 Channel kept busy when creating ssh tunnel via AGI
2:44AM 1 BRI: Asterisk disconnecting on 'call diverted' message?
2:39AM 2 stress a server with a tool
1:07AM 15 Uninstalling Trixbox
Tuesday September 19 2006
7:07PM 2 SkypeOut with Asterisk?
5:34PM 2 MOH distorted on Pound Key Linux on asterisk 1.2.8
4:58PM 1 Polycom 500 power supply
4:29PM 3 Cisco 7960 part numbers ...
3:29PM 2 IAX or SIP termination provider that reaches 6421xxxxxxx?
2:39PM 6 g729 and polycoms problem
2:37PM 3 Grandstream SX2000 attended tranfer
2:29PM 2 DTMF Detection Problems with certain phones incoming zap channels
1:29PM 2 Asterisk AGI question
1:00PM 0 clustering asterisk is possible ?
12:15PM 1 SPA 3102 does not even attempt to register
11:55AM 3 Pri Event 6 and 8
11:07AM 18 grandstream gxp 2000 does not display names when calling out
10:44AM 1 gTalk no audio issue
10:33AM 0 Repost: Register message received from realtime peer crashes Asterisk
10:16AM 5 SIP "Lines" Example Citel
10:10AM 5 codecs/voicemail/DTMF
8:44AM 0 Call forward with CFU?
8:36AM 1 Semi-OT: SIP or IAX provider in the Boston area?
8:35AM 3 When does Scalability requests Asterisk
8:23AM 3 Polycom default handset volume
8:00AM 1 transcoding error?
7:18AM 3 fast SIP failover (outgoing sIP requests) wi th 1.2
7:03AM 1 fast SIP failover (outgoing sIP requests) with 1.2
6:18AM 8 Problem with # locking up call
5:57AM 1 polycom 501 digitmap
5:46AM 8 Format_MP3, Streaming, File Formats, MOH
5:12AM 3 Aastra 9133i and Atcom AT-320 - Comments please
5:00AM 0 Clustering architecture and echo cancellation issue
4:25AM 0 Query ,NEED help regarding MWI
4:06AM 0 Anyone Using a Patton (Inalp) SmartNode 2400 for T.38?
2:07AM 8 Alcatel OXO Sip
2:01AM 23 When does Scalability requests Asterisk to Use SER ?
1:28AM 0 Wrong call handling
1:14AM 7 How to Dial a number with Sangoma PRI card?
Monday September 18 2006
11:34PM 0 prompt playing problem
11:16PM 0 Query on MWI
10:53PM 3 488 Not acceptable here sent by Asterisk - SIP debug follows
10:24PM 1 Accounting and re-invite
8:34PM 0 spandsp fax using Asterisk 1.2.X
6:19PM 0 create_addr: No such host:
6:09PM 2 Starting Asterisk PBX: FATAL: Module ixj not found.
5:10PM 1 Asterisk Appliance, will Asterisk Business Edition be mandatory?
5:07PM 4 Enabling Second Processor Trashes Audio Quality
4:42PM 16 Digium GUI?
4:36PM 3 How to learn or teach VoIP QoE
4:21PM 0 Periodic announcements & MySQL Realtime
2:29PM 1 ANI and Meetme...
2:19PM 2 sip.conf for talking to other Asterisk machines
12:32PM 2 Asterisk / Audiocodes annoying issue - Seeking Suggestions
11:57AM 0 CSR introduces UniVox reference platform
11:34AM 8 Fedora
11:08AM 2 How to make Polycom 501 go off hook when pressing any digits
10:54AM 0 X100P and zaptel 1.2.8
10:28AM 0 Changes in extensions.conf handling between 1.2 & 1.4
9:14AM 1 LDAP athentication
8:41AM 2 Dial and Timeout
8:12AM 3 Cisco 7940 Problem (Mess)
8:09AM 0 Chanspy crashing server, again
8:02AM 3 Polycom SoundPoint 2.0.1 SIP firmware?
7:39AM 1 unable to change the emailbody for email notification
7:33AM 0 RE : Re: [asterisk-dev] open letter
7:31AM 2 FOP Installation help
7:30AM 10 Re: Mediatrix 1204 trix
7:03AM 5 Chanspy crashing the server, again
6:36AM 1 Asterisk Design Question
6:32AM 1 User authentication
6:02AM 3 Playtones
5:46AM 0 Problem with Asterisk Realtime (MySQL)
5:38AM 1 Variable that gives the SIP channel
5:34AM 5 pickup call little complicated
4:28AM 0 Queue - Agent language
3:13AM 0 disconnect code in featuremap doesn't work on unanswered calls
2:20AM 1 Log out an Agent on RNA
1:53AM 3 is chanisavail command reliable?
12:15AM 3 Xorcom Astribank
Sunday September 17 2006
6:44PM 0 Noob question: Packet size
6:06PM 3 Polycom Expansion Module
5:55PM 0 problem installing func_odbc on asterisk 1.2 ...
1:12PM 3 Termination Rates
11:59AM 0 Register message received from realtime peer crashes Asterisk
9:07AM 5 A1200+fxo, anyone using this?
6:27AM 7 Asterisk Server Down
4:45AM 2 Does a "HST Saphir III ML PCI" work with Asterisk?
4:41AM 0 IAX2 audio problem
4:25AM 2 Starting out
1:26AM 0 How does Asterisk determine an incoming SIP Channel name?
Saturday September 16 2006
10:13PM 2 system cmd
5:34PM 1 Wrong outgoing port
1:54PM 4 Polycom programmable buttons
1:16PM 1 Calling to PSTN newbie question
12:57PM 0 RE: [Asterisk-video] VXIasterisk is available !
9:52AM 0 USA Regulatons
6:12AM 7 SHSU asterisk installation?
4:44AM 3 "Ping" a phone
4:15AM 1 read variable from shell script
Friday September 15 2006
11:51PM 1 Integrating the Openser for VoiceMail and PBX with Asterisk, For Account
11:05PM 1 Asterisk as a gateway to SER
7:33PM 2 call across 2 asterisks
7:30PM 6 Scaling/Loadbalancing a Call Center and Redundancy
7:28PM 7 amr codec
6:47PM 10 saved.gsm -> Voicemail greeting ??
6:37PM 0 voxee, callerid and trixbox
4:50PM 0 Help spread the word about Asterisk!
4:46PM 0 AEL2 patch for Asterisk
4:02PM 0 pickupgroup 1
3:43PM 1 DTMF Tone Not Passing Help
2:20PM 3 FollowMe question
2:18PM 0 Digium G.729 codec now available for Solaris/SPARC
1:30PM 1 Asterisk and Zaptel Released
1:03PM 1 Asterisk variables
12:39PM 10 Reliability of the newer IAXy's
12:17PM 0 New astGUIclient VICIDIAL Release: 2.0.1
11:45AM 2 Attended transfer and parking calls
11:09AM 5 Bristuffed asterisk 1.2.10 on Suse 10 - problem with module versionmagic
10:57AM 0 app_txfax segv fault
10:26AM 1 Voicemail adjustments
10:18AM 12 Asterisk with cisco 7935
10:11AM 1 Internal message being heard on pstn line
10:02AM 1 Cisco GW & CID Name
9:50AM 0 Branch office interconnect - IAX :vs: SIP?
9:10AM 0 inbound call from GSM gateway: handle_request_invite: Failed to authenticate user
9:09AM 1 ZT_SPANCONFIG failed on span 1: No such device or address (6)
8:18AM 2 4-wire analogue interfaces?
7:57AM 0 [asterisk-dev] open letter
7:20AM 0 Polycom 501 - message waiting LED manipulation
6:50AM 3 Issues with AGI+Dial command
6:39AM 1 where download app_txfax?
6:16AM 0 Section '12345678' lacks type
4:48AM 5 Cisco Distinctive ring using alert-info
4:11AM 0 Compile error in Asterisk
3:04AM 2 CDR question with SIP/IAX trunks
2:27AM 0 Setting up imap based voicemail / invalid remote specification
2:16AM 0 491 request pending [2]
2:14AM 0 trying to understand siprealtime & nat/MWI issues
1:55AM 0 Anyone using Voicemail with IMAP Support?
1:34AM 0 AOC - advice of charge
1:18AM 1 Shared Line Appearance, Snom and trunk
12:51AM 8 two safe_asterisk processes on the same PBX???
12:48AM 3 non-technical, dealing with users giving feedback
12:07AM 1 Bri Card for Asterisk ?
12:05AM 0 Cisco 7961 "dropouts"
Thursday September 14 2006
11:26PM 14 Can you explain why multiple registration is an important (missing) feature ?
11:21PM 1 Hangup on Panasonic KX-TEM824
11:10PM 3 How to download asterisk 1.3 development version
11:09PM 1 Cisco 79xx and vlan
9:02PM 0 Urgent !!!! Unable to make calls from cisco callmanager to asterisk
8:32PM 6 Why not g726-32?
8:12PM 2 G729 and Tribox
6:23PM 0 Asterisk with Addpac 2120
5:47PM 0 Login user
2:04PM 19 problems with Polycom 500 boot up
1:05PM 1 Asterisk / Patton SmartNode SN2400 Strangeness FYI
12:25PM 4 mISDN versus ZapHFC with BRIstuff
10:45AM 3 how to transfer a caller out of a queue ?
10:17AM 0 OT, Definity G3 Problems with Asterisk (Any Avaya
10:10AM 0 loosing Sipura 841 almost exactly on the hour
9:58AM 1 (Off-topic) Voip number "tracert"
9:57AM 7 How to send DTMF down a channel
9:42AM 3 [asterisk-dev] 491 request pending
9:24AM 0 Incoming SIP provider goes unregistered and never recovers
9:14AM 2 Page() paging application problem
9:12AM 0 Zork & Asterisk; zoip 0.2.0 released
8:58AM 0 491 request pending
8:52AM 5 Asterisk 1.4 Docs
8:41AM 0 incoming call h323 cdr
8:21AM 4 WAIT FOR DIGIT not working
8:03AM 4 Forcing Marker bit, because SSRC has changed
7:55AM 4 BLF across asterisk trunks
7:55AM 2 Controlling the channel
7:39AM 5 Getting 'i' functionality on internal extensions
7:37AM 3 Detect PBX vs Network message
7:26AM 0 Attended Transfer Asterisk 1.2.11
7:20AM 2 Asterisk and peers behind nat with port forward, how to proxy?
6:49AM 0 Dealing with FINAREA redirects
6:42AM 0 VoiceXML browser for Asterisk available !
5:45AM 5 voicemail access thru apache on another server
5:13AM 6 asterisk server to server using sip question
4:35AM 10 9 becomes 99 ? And other strangeness
4:21AM 2 Silence Call {very very urgent plz}
3:53AM 1 Thomson 2030
3:36AM 0 SOLVED: ringback on box with E1 and premicell
3:17AM 3 Correct settings for UK (BT) FXO
2:39AM 0 DIAL and automatic/manual co line acces
2:30AM 0 voicemail ,MWI problem
2:17AM 0 ASTCC: change from no pin to pin request?
1:51AM 0 MWI problem on asterisk
1:44AM 5 One way audio problem on gateway to PSTN after some time, no NAT involved
12:53AM 1 urgently requires help regarding MWI
12:17AM 0 changed behaviour of status indication on incoming pri lines in asterisk 1.2?
Wednesday September 13 2006
11:38PM 1 Flexible Wrap Up Time for Queue
10:48PM 9 How to install HUDLite Server
7:14PM 1 QuadBRI and Zyxel Wifi phone stop working togetherafter 3 calls
5:35PM 0 QuadBRI and Zyxel Wifi phone stop working together after 3 calls
3:40PM 0 Asterisk as a B2BUA and/or a FXS-SIP gateway?
3:37PM 15 stopped working after upgrading CentOS
3:27PM 4 Copyright issues with libcurl and OpenSSL
2:56PM 2 Via Epia platforms and asterisk
1:30PM 2 OT -- echo cancellation of an audio file
1:10PM 1 Polycom 501 display bug
12:57PM 0 Jitter Buffer on SIP
12:52PM 1 Astmanproxy authentication problems
12:19PM 0 success
12:02PM 1 Asterisk crashing when monitoring SIP device with Chanspy
10:01AM 4 Dropped Calls on TDM400p
9:52AM 11 University switches to Asterisk
9:46AM 4 callback without agi
9:41AM 0 Third Lane PBX Manger Multi-Tenant
9:00AM 7 Building Zaptel 1.2.9 with Octasic
8:32AM 2 set global variable
8:20AM 0 audio drop out half channel
7:31AM 0 ip address incoming call
7:08AM 6 I need to record 30 conversations and have other 30 with music on hold, all at the same time, can a PIV handle it?
6:51AM 2 Streaming MoH Problem, starts and then stops immediately
6:30AM 0 Customize host in INVITE's Contact header?
6:27AM 0 sample configuration
6:12AM 2 HFC isdn card and bristuff 0.2.0 rc8n
6:11AM 0 Problems with call parking
5:33AM 0 Problem withfeature introducer as first digit on a call
4:46AM 3 Queue - static members
4:27AM 3 voicemailmain errors on CLI
4:24AM 0 Asterisk and "305 Use Proxy"
4:08AM 0 (no subject)
3:54AM 0 Adding own info in AMI
3:53AM 11 IVR not able to Play the Balance.. need some help here
3:45AM 2 Calls on hold
3:31AM 20 rxfax, spandsp and lack of ecm
3:20AM 2 Kirk IP600 V3 DECT Wireless server
3:18AM 11 OT, Definity G3 Problems with Asterisk (Any Avaya Definity Experts out there?)
3:14AM 3 "Too many files..." error - best way to fix?
2:55AM 1 Polycom IP430 sound level too low?
2:25AM 0 Queue - persistent members
1:06AM 2 Anyone working on VXML, CCXML support for asterisk?
12:04AM 0 Long Delay in IAX Calls
Tuesday September 12 2006
10:58PM 0 Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?
9:27PM 2 fxotune failure! "Could not fill input buffer - got -1 bytes, expected 4000 bytes"
6:10PM 4 Polycom Firmware
4:58PM 0 Bad number - is not in inbound speed dial
4:22PM 1 Makefile.moddir_rules: No such file or directory
4:14PM 2 Virtualise asterisk on Xen
3:50PM 0 INX ( Outgoing problem
3:04PM 3 All circuits are busy now???
3:04PM 6 sound file length
2:07PM 2 sip origination and termination
1:53PM 1 Switch Experiences
1:17PM 0 strange problem with calls between MGCP and SIP clients(ATA's)
1:15PM 0 consitent half channel loss after 6 minutes
12:01PM 0 Please help with a telular mod. SX5e
11:03AM 0 RE: [asterisk-biz] Come see us at VON
10:31AM 0 Trouble connecting to my telco with fonebridge
10:12AM 0 AEL if/else/IFTIME fun.
10:00AM 1 A simple goal, help me please!
9:57AM 2 Calling Card and Billing
9:28AM 1 Verizon ISDN service in NY & Hunt Groups
9:22AM 2 Polycom MyStat
8:34AM 4 Problems getting 7970G upgraded to SIP
8:08AM 0 Conference bridge problem
8:07AM 1 Dropped call question - "Maximum retries exceeded on transmission"
8:03AM 0 IAX phone recommandation
7:57AM 1 RE : Re: [asterisk-dev] Forwarding sip requests from none localdomains
7:39AM 3 Suggestion for directed pickup in bristuffed 1.2 Asterisk
7:37AM 0 Grandstream Budgetone phones don't show
7:33AM 1 RE : Re: [asterisk-dev] Forwarding sip requests from none local domains
7:27AM 1 [BULK] Re: Prompts recording for Asterisk
7:17AM 0 Deploying an IVR - direct extensions.conf or AGI scripts?
7:00AM 1 How to setup announce attibute in queues.conf
6:43AM 1 WG: Asterisk and Agents
6:33AM 1 Features.. phone vs. asterisk?
6:25AM 4 WARNING[21314]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/1 already in use on span 1. Hanging up owner.
5:32AM 4 Rack for Asterisk with TDM2400 Digium board
5:14AM 0 Which SIP hardphone implements RTCP XR (aka RFC3611)
2:55AM 2 Junghanns BRI cards and misdn
2:23AM 0 Samsung OfficeServ 500 + Asterisk(Tormenta 2) via PRI
12:48AM 1 asterisk logging per day
12:38AM 0 SIP/2.0 403 Relaying denied
12:34AM 1 about 'zap show channels'
12:17AM 0 The best way to design local-only off-hours ringing
Monday September 11 2006
11:41PM 0 Browsing distant missed call list
8:25PM 1 DID not getting passed?
8:23PM 0 Polycom HD Voice - 16 Khz - Asterisk support ?
8:02PM 0 Digium at Ohio Linuxfest
5:58PM 0 GXP2000 - Blind Transfer Hangs Up Call
4:41PM 0 Change Payload
4:30PM 1 Forward recorded voicemail message to more than one extension using sendvoicemail=yes
4:27PM 0 SIP 415 messagse
4:25PM 0 BLF via metermaid on and aastra 9133i
4:01PM 8 question...
3:13PM 7 Dell hardware ...
1:47PM 4 Static RealTime - SIP.CONF
1:21PM 0 experience with
12:41PM 1 More Zaptel build problems
12:15PM 0 Cable Systems ICS-G302, Anyone have an Admin Guide Please?
12:14PM 6 Weird (bri)stuff 0.3.0-PRE-1s
11:59AM 9 Polycom Soundpoint Key Remap
11:57AM 0 IAX2 trunk problem
11:41AM 2 How to configure Fritz ISDN2 card with Trixbox?
11:30AM 1 PRI channel hangup
11:25AM 3 Remote tone access
11:13AM 0 updated zaptel tarball
10:45AM 1 sip and iax over the internet (asterisk to asterisk) drop outs normal???
10:38AM 0 Getting Incoming called from
10:30AM 0 Realtime Queues and Postgres.
9:30AM 0 --- Dlink DVC-2000 VideoPhone (H.323) with Asterisk ---
8:54AM 0 [Serusers] MS LCS 2005 / SER / Asterisk Integration
8:45AM 11 Grandstream Budgetone phones don't show alphanumeric caller right
8:44AM 0 Is anybody using autofill option in queue.conf?
8:42AM 1 help connecting cell phone, chan_bluetooth
8:17AM 17 PRI: sometimes Asterisk drop calls
8:05AM 2 Verify Database Installation
7:38AM 0 Support for Intel Boards On Asterisk
7:05AM 0 Register 2 times with same host
6:56AM 3 Problems Unpacking tarball For Asterisk Application
6:20AM 0 switching from IAX to SIP
5:18AM 1 realtime static config include contexts
4:47AM 0 Handling incoming calls from VoIPbuster
4:30AM 2 SIP trunk
3:55AM 0 Ringtones
3:47AM 0 Can Asterisk bind on multiple ports?
3:36AM 2 modifying the INVITE headers
3:19AM 0 I am not getting 302 redirects...
2:51AM 1 TE411P or TE412P?
2:50AM 0 SIP hardphones and BLF monitoring keys
2:48AM 0 Outgoing callerid in AMI
2:07AM 2 SIP parameter to prevent a call from being added in missed calls logs
2:06AM 3 Asterisk Realtime Arch - static or realtime?
12:43AM 3 How to integrate freepbx with a2billing?
12:01AM 1 beginners question....
Sunday September 10 2006
11:12PM 0 Looking to hire somebody to setup a SER load balancer
8:35PM 3 QUINTUM TENOR ASM200 Configuration
7:42PM 3 Hrmm.. OK, what am I missing? sendmail: Cannot open mail:25
4:53PM 3 Setting system time via Asterisk
4:42PM 12 using residential voip for business?
4:05PM 4 Max Size of Conf Files
3:49PM 0 data pass through
2:02PM 2 music onhold choppy music problems
1:38PM 3 can someone recommend a voip provider that...
12:43PM 7 Voip providers and sip origination and termination?
12:02PM 3 su - postgres -bash-3.00$
11:10AM 0 Accounts registered, but call is not going
10:24AM 7 Polycom related question
8:54AM 3 Take 3 -- Trying to get SIP firmware on a 7970G
5:09AM 0 How could i get bridged channel partner
4:40AM 1 Satellite link-IAX Jitter Buffer.
3:24AM 0 call notification for queues?
Saturday September 9 2006
10:09PM 2 Quintum tenor configuration with asterisk help
10:03PM 0 Streaming audio for MoH
9:55PM 2 Grandstream GX-2000 Remote Login Problem
6:58PM 0 What really happens between Asterisk and an SPA-3000?
3:52PM 16 Whcih phones are better for mass deployment
3:47PM 1 Using option 'r' in queue doesn't announce frequeny etc.
2:25PM 3 Scope of contexts
1:54PM 2 ztdummy installed but choppy audio warning on load
1:46PM 0 Problems configuring Polycom 301
12:48PM 0 RE: asterisk-users Digest, Vol 26, Issue 54
10:10AM 10 Intel Based G.729 and SVN-trunk-r42453
7:19AM 0 DID Provider in Thailand
7:17AM 0 ISDN / Multiplink PPP (ZapRAS)
6:05AM 1 Another (quick) Polycom 501 question
5:15AM 6 Call Processing Slow 11 seconds
2:19AM 1 Call Forward Problem
Friday September 8 2006
11:32PM 2 Receive Fax with rxfax on asterisk with debian
10:40PM 14 Zaptel-1.2.9 compile error
8:56PM 0 zaptel 1.2.9 won't compile
7:29PM 0 Asterisk 1.2.12 and Zaptel 1.2.9 released!
4:54PM 0 How to play a sound on a periodic basis during a call?
4:42PM 3 MSSQL connection
3:44PM 2 Stupid question about FXS/FXO
3:05PM 3 No such device -> TDM13B
2:54PM 3 Little help for a newbie configuring a TDM13B - ztcfg fails on channel 4
2:06PM 1 I'm I wrong - No 3-way calling for Single line sets?
1:26PM 0 help chan_bluetooth
12:47PM 0 RE: Peter Dicks Chairman ofSportingbet PLC is arrested at JFK!!
11:44AM 6 No dialtone, just directly busy
11:32AM 24 What don't I get about SIP?
11:11AM 0 ISDN HFC card cannot 'detect remote answer'
11:06AM 0 Want to support a better SIP stack in Asterisk?
10:47AM 4 Use PauseQueueMember
8:22AM 1 Asterisk and SIP Redirect message
7:31AM 1 Reload question
7:22AM 16 FW: Peter Dicks Chairman of Sportingbet PLC is arrested at JFK!!
7:04AM 15 Tracking the source of a disconnect?
6:43AM 2 Grandstream, how to use the configuration tool
6:36AM 0 How can I set CDR data in dialplan? Set(CDR(src)=foo)
6:35AM 2 Grandstream GX-2000, doesn't send calls to free lines
6:04AM 0 Problems with KG1000 voip gateway and DTMF
4:39AM 3 distinguishing users by their domain
3:39AM 2 Asterisk and "Maximum retries exceeded"
3:04AM 4 Trouble with rxfax multi-page printing with cups
2:59AM 3 Digits are played in english in french voicemail
2:38AM 0 Problems with app_directed_pickup
2:35AM 2 sip peer question
2:00AM 3 codecs translation in Asterisk SVN-trunk-r41990
12:46AM 0 dialplan applications
Thursday September 7 2006
11:07PM 5 Asterisk 1.2 and SATA drives
10:39PM 0 app_amd and voicemail
9:47PM 1 Intel 945G and Digium TE110P compatibility issue
4:04PM 0 te110p and te205p behavioural differences
2:43PM 0 RE: asterisk-users Digest, Vol 26, Issue 39
2:14PM 11 Call Forwarding in SIP.conf
1:57PM 1 Speex Codex - Eyebean to Asterisk
1:32PM 0 Open source G.729 and G.723.1 release for 1.2 and 1.4
1:32PM 1 TDM400 and T100 config on same asterisk
12:37PM 4 Experiences, Tips on Voicemail storage using ODBC or IMAP?
11:06AM 5 Asterisk Outgoing Spool Failed
9:47AM 3 How to Install H323
9:15AM 0 Sound (or lack of it) problems
8:55AM 3 uConnect Voip device
8:32AM 9 Asterisk hangs up after 10-15 minutes when SIP Phone is on mute
8:30AM 4 Asterisk and NAT ?
8:07AM 0 Voicemail Delete Bug?
8:04AM 2 g729 failover when out of licenses
8:00AM 1 svn trunk or branches ???
7:17AM 6 Capacity for transcode G711 to G729
5:54AM 3 Asterisk "Clusters"
5:48AM 0 Incoming call problem-calling part is busy(I PKall)
5:25AM 1 Incoming call problem-calling part is busy(IPKall)
5:11AM 7 Response to KP Flemming...
4:54AM 4 Cisco 7970 directories and services xml
4:48AM 19 Softphones IAX vs. SIP, remote connectivity.
4:27AM 4 bristuff compile problems with kernel
3:53AM 0 WG: mobile refusing call
3:50AM 3 netmask
3:14AM 2 New polycom firmware / presence
2:51AM 0 Configuring new IAX2 Jitter Buffer for IVR application.
2:39AM 0 How to send and receiving fax with asterisk?
1:03AM 0 ast_parse_allow_disallow: Cannot allow unknown format 'h264'
Wednesday September 6 2006
10:43PM 0 Re: [asterisk-dev] UUI in calls
8:10PM 2 the sounds quality of IAX2 channels are not good as SIP channels?
7:46PM 0 How to check which rtp ports my firewall let through?
7:21PM 13 Polycom new firmware and bootrom
6:27PM 7 using SIP to connect remote other VoIP server
4:08PM 0 Garbled (quality probs) IAX2 & SIP calls Asterisk-to-Asterisk
4:00PM 0 Digium's response to posting of G.729 and G.723 source code
3:25PM 1 Digium G.729 codec binaries updated
3:21PM 0 faktortel
1:35PM 6 Volume events causing talk off on Asterisk with Digium 411P
12:50PM 5 Linksys SPA-942 + Asterisk 1.2.10 = Inability to transfer calls
12:17PM 1 Call parking and RTP traffic
11:16AM 1 Is asterisk's mgcp support(NAS) Network access server package
10:46AM 4 Conditional IF based on IP address?
9:47AM 7 Cisco MWI
7:45AM 0 Which SIP hardphone with embedded VPNClient ?
6:56AM 8 app_rxfax Only Receives One Page
4:57AM 5 cmd SET time value
3:33AM 1 how to setup poxy sip server
2:39AM 0 mobile refusing call
1:59AM 0 flag 'g' in Dail() is'nt working with agentcallbacklogin()
1:56AM 0 sangoma A104d echo canceller and fax
1:29AM 1 core dumps
1:17AM 5 Budgetones - multiple phones losing IP address during day
12:50AM 2 How to test TE405P T1
12:29AM 0 Asterisk AGI and Firebird
Tuesday September 5 2006
11:55PM 3 Dell Poweredge SC430 and Digium cards compatability enquiry
11:23PM 1 Asterisk + Samsung OffServ 500
10:57PM 2 macros in Realtime
9:38PM 5 Really bad phone line.. possible causes?
9:00PM 4 Has anyone tried to install both digital card and analog card in one machine
8:26PM 0 Need somebody for video phone testing
7:34PM 5 Native Chinese speaker needed
4:53PM 3 Merlin Legend - Working Now!
2:46PM 6 How to notify an ACD agent before he/she picks up
2:36PM 9 Wildcard X100P Disconnect Problems
2:36PM 0 PCI FXO disconnect problems
2:36PM 2 musiconhold.conf problems
2:32PM 8 Asterisk Cygwin Port.
1:51PM 0 Is this a warning or not...MYSQL Fetch
1:44PM 2 Adding custom fields (more than one) to CDR DB
1:40PM 0 Linking Asterisk with PBX through E1
12:52PM 4 asterisk t.38 fax failed
12:50PM 3 config include issues
12:39PM 0 Meet-me recording formats
11:37AM 3 IAX and rsa
11:35AM 3 Different MOH between waiting calls and transfer calls
10:01AM 1 Faxing ..
9:52AM 1 Find-Me/Follow-ME
9:42AM 2 Asterisk vicidial question
8:38AM 1 ATA being used as a SIP Trunk to connect LegacyPbx to Main Asterisk Server
8:26AM 12 monoBRI + install-misdn-mqueue: no inbound calls but strange messages
8:03AM 2 Different MOH in waiting calls and parked calls
7:58AM 1 Unable to make calls from CallManager to Asterisk
7:01AM 0 R: Re: LinkSys PAP2 ATA & Siemens Cordless 3010
6:55AM 1 Zero length queue
6:29AM 1 ISDN config EWSD
6:28AM 1 LinkSys PAP2 ATA & Siemens Cordless 3010
6:28AM 0 telco error message on PRI and BRI
4:54AM 1 Experience Patton BRI gateways and Asterisk?
4:05AM 2 latest CentOS-asterisk-freepbx installation procedure
3:21AM 5 How to manipulate a plus in a phone number
2:04AM 7 why executed Hangup doesn't exit DialPlan?look my dialplan...
1:26AM 0 A couple more interviews with Digium staff
12:21AM 0 connect with two servers multiple time
Monday September 4 2006
11:47PM 1 End of call
11:25PM 1 Re: [asterisk-biz] Re: G729 Replacement Codec - FREE or may ne cheaper than existing one.
11:19PM 0 Reading the raw E1 channels ?
10:42PM 0 HITBSecConf2006 Final Call !
10:36PM 0 Warning about using PAP2-NA ATA recent firmware 3.1.12 LS
7:13PM 0 SNMP with 1.2.11 stable
6:56PM 8 Digum g729 and g723
3:13PM 0 Re: Nufone making changes
1:58PM 2 Call center reports
1:33PM 1 app_conference not working for me
1:17PM 1 Looks like Nufone is changing around...
12:26PM 2 Grandstream and H.264 !
12:19PM 0 playback some digits to the caller from the callee (involves DTMF) prob
11:58AM 0 Astbill DIALSTRING doesn't work
11:26AM 6 blf aastra 9133i working but can't pickup calls
10:27AM 1 missing pri connect (wwomera to pri)
8:55AM 2 Dropping extra frame of G.729 ?
8:47AM 3 Submenus
8:26AM 0 Handling Disconnection Causes
8:10AM 0 PAP2-NA + Asterisk
7:35AM 7 FAX handling
6:52AM 8 Asterisk 1.2.11 and # key
6:27AM 0 usereqphone=yes seems to don't work
5:37AM 13 includes in realtime ??
5:21AM 1 External calls from Asteris over a legacy Siemens BusinessPhone 250 PBX
5:00AM 0 External calls from Asterisk over a Siemens(legacy) RDSI PBX
4:55AM 4 Any Hardphone with VPNClient embedded?
3:04AM 0 Architecture:MainOffice(AstServer)-SmallOffices(ATA.-LegacyPBX)
2:14AM 3 Zaptel-1.2.8 compile problem
2:12AM 0 "Asterisk Developers Mailing List" <>,
1:20AM 10 Prompts playback changing tempo w/ SMP kernel
12:52AM 0 No more linux/compiler.h in Fedora Core 6.
Sunday September 3 2006
10:26PM 4 Asterisk calling through FWD?
4:52PM 0 Please help route incoming PSTN calls to Asterisk
3:12PM 1 PBX -> VoIP migration
10:54AM 0 AgentCallBackLogin and cdrupdate
7:13AM 0 Query about Call Detail Record in Asterisk
4:10AM 2 Asterisk+ser+docs
3:05AM 8 UK (BT) Problem with TDM 400P
2:47AM 8 Asterisk not sending RTP
1:53AM 2 SER+Asterisk integration
Saturday September 2 2006
10:18PM 2 What I always get asked in SME * deployments
4:12PM 4 Queue timeout problems
4:11PM 12 How to use Grandstream GX-2000 phones for paging
4:07PM 5 Roundrobin not working on PRI
4:05PM 4 SIPP problem
3:58PM 8 How to send correct Caller ID on PRI
3:37PM 5 Caller ID has extra digits to strip
3:11PM 0 res_osp.c not compiled
8:01AM 0 Nokia N80
1:28AM 14 Keys pressed not registering ...
1:21AM 7 G729 Replacement Codec - FREE or may ne cheaper than existing one.
1:18AM 1 Asterisk mysql cdr
Friday September 1 2006
10:37PM 19 Blind transfer 3/4 digits
8:37PM 4 Help with blind transfer
7:12PM 12 Asterisk & Linksys PAP2 ATA
5:42PM 2 What does 'trunk' mean in outgoing and incoming?
2:52PM 2 Can QUEUE member be assigned from a GlobalVar set in EXTENSIONS.CONF?
1:39PM 1 Cisco 7960 won't download dialplan.xml
11:51AM 1 Callback + dtmf problem
11:49AM 6 Asterisk speaks Italian!
9:15AM 5 Hardware ? Analog DID trunks (ILT)
8:44AM 12 Sipura 3000 and Asterisk
8:18AM 0 incompatible hardware?
7:22AM 0 Asterisk Core dump
7:03AM 7 vim syntax highlighting( for Asterisk.conf files)
6:45AM 0 phpagi syntax and SendDTMF
6:27AM 0 Different MOH in parked calls??
6:27AM 6 balance anouncement
6:11AM 2 Asterisk as a SER client
5:42AM 0 looking for GXV-3000 users
4:37AM 2 Probelm with incoming calls to my DID-Please help me
2:27AM 4 [Slightly OT] Grandstream configurator tool
2:16AM 2 Outgoing Call group ?
12:36AM 5 Any way to go from factory reset 7970 to SIP without Call Manager?