asterisk users - May 2008

Saturday May 31 2008
6:34AM 1 IAX2 hardware video phone
3:02AM 0 SLA & Polycom
Friday May 30 2008
6:18PM 2 cdr_odbc not working for some reason
6:12PM 1 How to have a difference dial tone after dialing the ignorepat digit
4:45PM 2 Dial() and the timeout to jump for next priority
3:33PM 2 Calls to Teliax dropping...
3:21PM 3 CDMA Phones
1:49PM 1 SPA 3102 unable to detect hangup
11:46AM 1 3Com Phones
11:24AM 1 Recommend a speaker/conference room SIP phone?
6:57AM 1 Problems CallerID in Call Out
5:27AM 0 Friday @12 PM EDT VOIP Users Conference
12:02AM 2 OT: Good phone to work with Asterisk/NAT
Thursday May 29 2008
10:09PM 0 FIXED @1.4.11 - zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
8:55PM 1 polycom 501 call waiting. can hear it but can't pick it up
4:29PM 1 New York based Asterisk Users event this monday the 2nd of June at 6.30pm
2:32PM 2 Ideas for call back when all lines are busy
2:26PM 1 Channelized T1
12:42PM 0 Skype connectivity? Siskyee or chanskype?
10:25AM 0 ILBC sh script throwing error
9:44AM 0 Asterisk Cnfiguration requirements?
5:38AM 2 Polycom 601 BLF multiple servers
5:30AM 2 Dialplan questions...
3:58AM 0 Zaptel 1.2.26 and 1.4.11 Released
2:27AM 2 Wait for input
Wednesday May 28 2008
9:09PM 2 Does Digium support the B410P?
6:13PM 1 [Extensions.conf] Interval + discrete extension?
5:56PM 1 Zambia Africa
4:53PM 2 Calling '**1' through Asterisk
4:23PM 7 Cisco Gateway sending call to * without CID Name
4:11PM 4 transferring a not yet answered call
3:13PM 4 iLBC and Asterisk
1:49PM 3 Asterisk VoIP in Dubai/UAE?
1:25PM 1 Callgroup phone Notificacion
10:31AM 2 any pointers to mute/unmute a channel
8:57AM 3 Making remote calls via my Asterisk server
8:56AM 1 About Meetme
7:31AM 1 Linksys 942 and pickup function
7:21AM 1 length voicefile
5:26AM 1 why is CLIP(CallerID presentation) facility charged?
Tuesday May 27 2008
11:05PM 0 callerid.c callerid_feed: Unknown IE
10:02PM 1 CDR "forking" for DUNDi calls
9:43PM 2 Trunk/Peering provider in Canada
8:05PM 2 Problem with Asterisk to Asterisk IAX trunk
6:12PM 2 ForkCDR
6:09PM 1 asterisk 1.4 QUEUES Variables
5:11PM 4 Asterisk just stops working...
3:55PM 1 SIP to SIP calls
2:46PM 4 Can Asterisk run fine on 64 bit CentOS 5.1?
1:02PM 0 Max SIP Registrations with and without Realtime
12:37PM 1 Max number of Registrations with and without Realtime
10:05AM 0 Is it possible to specify a timeout value to AGI or FastAGI applications ?
9:28AM 2 Why "should not run FastAGI on the same server" ?
7:04AM 3 Group pickup and CDR Logging / TAPI
Monday May 26 2008
11:19PM 9 Voice low on ZAP
9:36PM 1 TE2XXP and modems
9:23PM 3 Card loading order...
8:58PM 1 Loud, obnoxious screech heard during ring
7:44PM 0 SIP Debug information
4:15PM 0 queues MEMBERINTERFACE variable
4:10PM 1 Long Distance Access
4:06PM 5 Skype Howto
3:57PM 3 Registration of multiple SIP-clients for the same extensions
2:15PM 1 Kevin Flemming live from Berlin NOW
2:11PM 1 Asterisk and Linksys SPA8000?
1:01PM 0 callback
12:47PM 1 Require a Virtual Private Server (VPS) to terminate calls
12:00PM 2 Installing a TDM410 card
9:52AM 1 Zaptel does not hangup
9:15AM 1 How to do not use Asterisk internal DB for SIP register?
8:40AM 0 realtime problem with two Asterisk servers - UPDATE
6:30AM 0 realtime problem with two Asterisk servers
6:25AM 0 Question about ISDNguard
3:59AM 1 Ubuntu as remote MySQL server
12:36AM 0 problem with e1 connection
12:35AM 1 AMI Redirect between MeetMe rooms
Sunday May 25 2008
9:51PM 0 I invite you to join my Ziki Network !
5:06PM 0 Incoming SIP call ring timeout
4:58PM 3 End call behaviour
3:29PM 1 No Audio on Meetme
11:17AM 1 Logical AND (resent due to bounces)
10:47AM 2 Logical AND
9:45AM 3 trying directrtpsetup
8:49AM 0 Cmd Dial (a group) and 1) who pick up the call 2) How to use the G option
6:46AM 1 Asterisk Tag Berlin live notes page
1:33AM 1 Error after upgrading from 1.2.18 to 1.4.20
Saturday May 24 2008
6:09PM 0 (Newbie)How to reduce security risks inopening IAX & Sip Ports
6:04PM 2 One way Speech issue - only in PAP2T to SIP device attached to Asterisk but not PAP2T to Voip service provider
2:03PM 1 Asterisk not answering calls
1:52PM 3 excessive bounces???
11:55AM 2 Asterisk semi-hangs
10:53AM 2 digium cards with sangoma cards
10:09AM 2 Incoming calls not being answered by asterisk
Friday May 23 2008
10:43PM 1 Strange ring or moh quality
9:27PM 2 *#&%! Polycom...
9:08PM 1 dialplan syntax error: need new eyes
5:42PM 1 [asterisk-dev] Asterisk 1.6 Realtime Database must use ', ' not '|'
4:51PM 0 OOH323 to Avaya S8500?
4:24PM 2 Strange State 6 on Channel X
3:28PM 0 Proposed changes for queue timeout
3:26PM 0 Asterisk chan Skype
3:17PM 1 Transfer
2:15PM 1 (no subject)
1:56PM 0 Asterisk/OpenSER users in Porto, Portugal?
12:56PM 2 New York Asterisk Users
12:41PM 2 B410P install
9:02AM 0 Todays at 12 Noon EDT: Mike Trest on large volume calling with asterisk
8:05AM 3 H.323 video support
4:45AM 3 forwarding pots lines
4:02AM 1 application sendtext
Thursday May 22 2008
8:36PM 0 SIP configuration issues
7:59PM 1 Handling multiple fax machines and the fax extension, and general call routing
7:35PM 0 Continued NTL ISDN issues with Vox D210E card
5:49PM 7 reload stopping EVERYTHING on CLI and causing havoc.
5:48PM 1 Telco intercept prompts
5:48PM 1 Asterisk Wackyness
5:42PM 1 Adit 600 password reset
5:37PM 1 asterisk-addons 1.6.0 "Command 'realtime mysql status'
5:23PM 0 Call Park Logic?
4:53PM 0 asterisk-addons 1.6.0 "Command 'realtime mysql status' failed"?
2:56PM 2 More fun but with Wireshark capture
2:54PM 0 /home/putnopvut/asa/AST-2008-007/AST-2008-007: AST-2008-007 Cryptographic keys generated by OpenSSL on Debian-based systems compromised
1:48PM 2 Grandstream
9:33AM 3 call pick up
8:48AM 2 upgrade of asterisk .... to what?
7:28AM 1 Require Customization of SugarCRM or Vtiger
7:02AM 0 Require software like StarTel for CTI Pop Up ?
7:00AM 1 Fwd: - Asterisk Local channel
6:59AM 0 -
6:57AM 2 Meet you at Asterisk-tag in Berlin!
6:44AM 0 iax test
12:46AM 1 Call Placed through Manager connecting before the call connects.
12:39AM 2 1.4.20 delay
Wednesday May 21 2008
11:55PM 1 Asterisk 3 way calling
9:57PM 1 using gtalk received instant messages in the dialplan
9:37PM 11 Asterisk Database Handling
9:06PM 9 ASP web phone
7:31PM 4 addons-1.6 not seeing installed MySQL packages
4:22PM 0 AsteriskNow Wizard not running?
4:13PM 0 Language Change on Polycom S IP 300
2:54PM 3 3 ways
2:39PM 1 T38 fax solution with Asterisk possible?
2:01PM 0 Gentilini, Paul is out of the office.
1:55PM 6 Fax solution for Asterisk
1:00PM 5 asterisk and sipura 3102 (pstn to sip/sip to pstn calls)
12:23PM 0 iax2 received mini frame before first full voice frame
11:28AM 1 speex, ilbc and g729 codecs, GSM with IAX
6:22AM 1 T.38 w/ MAX TNT & ASTERISK
5:00AM 0 Require US Toll free number
Tuesday May 20 2008
11:35PM 2 Asterisk 1.4.20 Released
8:33PM 7 Busy out a zap channel?
8:25PM 2 Error Counters on PRI Circuit
8:05PM 0 Troubleshoot in-bound DTMF over PRI
7:35PM 0 asterisk black holing h263
5:41PM 5 Digium announcement: new community manager - John Todd
4:09PM 0 183 Session Progress
3:51PM 2 At whit's end was 'DHCP Failure screws up system '
2:55PM 1 Problem with Polycom forwarding
1:03PM 2 AsteriskNow: No Analog Hardware Detected
12:55PM 5 Server recommendation help
12:14PM 1 IVR for callee (called party)
12:12PM 0 [svn-commits] file: branch 1.4 r117081 - /branches/1.4/channels/h323/ast_h323.cxx
12:10PM 3 Newbie Voicemail: Just use one [context] invoicemail.conf?!
9:58AM 2 karaoke functionality
8:41AM 2 (Newbie)How to reduce security risks in opening IAX & Sip Ports
7:58AM 2 Newbie Voicemail: Just use one [context] in voicemail.conf?!
3:44AM 0 mute a call/ re-invite mid-session?
3:27AM 1 Fax Machine Options
12:39AM 1 Zaptel project being renamed to DAHDI
Monday May 19 2008
11:19PM 3 Concept Clarifications
8:45PM 0 Not hearing first prompt
8:33PM 2 Recording problems, reinvites
8:01PM 0 Asterisk Jobs Spring Special - Free Postings for Employers!
7:24PM 0 VoIP Termination To Togo and Cameroon
7:21PM 1 Asterisk first time user
6:45PM 1 Dialplan Visualization, Rating System Unveiled!
6:42PM 1 DHCP Failure screws up system
6:12PM 3 Understanding Incoming sip DID handling
6:09PM 0 outbound calls on PRI all congested
5:38PM 3 Fedora 9 + Asterisk
5:29PM 3 Which best practices to build and deploy Asterisk on different hardware ?
5:28PM 1 nokia 770 has a build in mic, asterisk and iphone
4:47PM 3 Wireless headsets for Polycom phones
2:17PM 2 Is this the forum for Business Edition of Asterisk
11:38AM 0 zaptel errors update
10:59AM 1 (Newbie)Security Risks in opening IAX & Sip Ports
10:59AM 1 where did the switch statement come from?
9:39AM 0 - Failed to authenticate user
9:20AM 0 callerid
8:58AM 0 Dutch Asterisk mailing list
6:44AM 2 Extension not found
5:08AM 6 time limit
4:53AM 0 max retry
4:42AM 1 Recall: Newbie Asterisk: Install Asterisk as non-root
Sunday May 18 2008
9:17PM 3 Inbound Answer not working
8:46PM 2 Dutch Asterisk mailing list?
7:33PM 1 strange name alt.asterisk.canary.tweet.tweet.tweet
10:57AM 1 Bridging a call on hold with an active call
7:43AM 0 Asterisk users and Twitter
12:40AM 1 Paging
Saturday May 17 2008
9:50PM 1 Using a Loopback Plug for an RJ-45 Ethernet Interface for testing a Digium Card
8:15PM 1 Hangup issue
7:21PM 0 Local loopback vs SIP/IAX2
7:01PM 1 zaptel 1.4.10 doesn't build on debian etch epia itx system
6:18PM 1 asterisk virtualization on VMWARE SX infrastructure
5:40PM 0 sipbroker CLI
5:38PM 2 Googles 411 services
1:11PM 0 trixbox, sangoma a200, dell poweredge
11:47AM 1 One way sound when Using Dial cmd without "t" option (SOLVED) Need explanation
10:08AM 3 Implementation of Video Conferencing using Asterisk
6:40AM 1 More dialplan visualization (neat graphs!)
3:35AM 0 CFP For HITBSecConf2008 - Malaysia Now Open
12:17AM 0 Asterisk peer definition with multiple host ip addresses
Friday May 16 2008
11:50PM 3 Queue Stats
9:10PM 2 Fetching Binary data from SQL Server
8:42PM 1 Alternate names in Directory (dial-by-name)
7:40PM 0 Digium TDM4xx CID problem
5:48PM 1 trixbox, sangoma a200, dell poweredge 2550 issue
4:35PM 1 PRI debugging ...
2:28PM 2 Connecting a PSTN gateway to Asterisk using PRI
1:46PM 1 Problem with cisco 7970G []
9:37AM 3 PBX deployment big problems: Voip traffic analysis
7:47AM 7 Asterisk concurrent calls count
7:36AM 3 Problems passing variables from a macro
3:39AM 1 queue autopause
3:38AM 1 A couple of newbie questions
Thursday May 15 2008
11:26PM 7 playing .gsm sounds through a web browser
9:42PM 3 Not hearing first prompts
9:41PM 0 Asterisk for Larg (Al Baker)
9:39PM 2 QOS and Asterisk
9:04PM 1 Received SIP subscribe for peer without mailbox
7:14PM 1 Where does menuselect save your choices?
6:59PM 1 Citel Gateways
4:23PM 0 how to find the logs for this problem
4:11PM 7 Asterisk dropping around 2% of ALL calls ever since we moved to EM_W signalling?
4:06PM 1 *72 Telco Call Forwarding
3:47PM 1 ChanSpy not working - "transmit frame type 64" warning
1:37PM 1 caller-id on X100P fails frequently
11:05AM 4 are channel names unique
10:57AM 1 Problem while running Flash Operator Panel
8:17AM 4 Newbie Asterisk: Install Asterisk as non-root
5:50AM 0 Friday May 16th @1 Noon EDT: VoIP Users Conference is about Click 2 Call
4:49AM 0 Listen And Talk mode differentiation of meetme() conference
4:08AM 0 Fw: voicemail not sending emails
3:31AM 0 Polycom IP300 Language
1:54AM 0 monitoring names
Wednesday May 14 2008
11:45PM 9 Polycom XML Files / asterisk
7:52PM 3 Question about SS7
7:39PM 6 anyone from Joplin, MO
5:02PM 0 Setting CallerID UNKNOWN on an outgoing
4:16PM 1 Asterisk 1.4.20-rc3 and 1.6.0-beta9 Now Available
2:31PM 3 Announcing the first North America Druid Meetups happening Chicago 22 May 2008 and Altanta 27 May 2008
1:57PM 2 Understanding Asterisk
9:32AM 0 [asterisk-dev] UWB Codec / Command-line softphone help
8:48AM 0 No sound with Playback() and Background()
6:08AM 2 Setting CallerID UNKNOWN on an outgoing call
2:22AM 3 No-mobo PC for USB Drives Enclosure?
Tuesday May 13 2008
11:51PM 1 Zaptel Install Error
11:48PM 6 voicemail not sending emails
9:55PM 2 [asterisk-biz] ANI
8:56PM 2 Installation Question
8:44PM 0 Call retard from a softphone to a hardphone
7:50PM 7 BLF Compatible Phones
6:31PM 3 Call only for registered sip users...
6:18PM 0 Asterisk Released
5:27PM 1 queue problem
4:38PM 1 More one way audio...
3:47PM 0 Fwd: [asterisk-dev] Paging intercom extensions
3:46PM 0 conference, May 26th/27th, Berlin, Germany
3:45PM 2 How to test dialplan w/o a trunk
12:44PM 3 Queuing if no one available to answer
12:35PM 0 Paging for analoge devices
12:35PM 1 Control of individual call legs
11:55AM 0 Extension Auto Fall through help when matching fails.
11:17AM 1 New Asterisk Deployment - Need some tips
10:29AM 1 cannot get calls with Tellfree brazilian provider
10:11AM 2 Asterisk stops MOH on transfer
7:44AM 1 chan_mobile install with Asterisk 1.4.19
6:18AM 0 Newbie Polycom: Cannot Disable Services button
Monday May 12 2008
11:43PM 2 Newbie Dialplan: Best Practice in using Context - Do not use Default??
10:42PM 0 Require a Touch-Tone to Connect? proof of concept with meetme()
10:21PM 0 business class sip provider with a SIP proxy server in India ?
8:57PM 2 Which sound file formats?
8:44PM 1 Is there a way to have Manager Bridge Channel without being connected
7:05PM 1 Crappy sound on Console (chan_oss)
6:07PM 1 Problem with SIP Subscription Status
5:32PM 0 externip not working...
5:30PM 1 "module reload" CLI Asterisk question
4:19PM 1 Using multiple variables in SIP.CONF setvar
3:29PM 2 3U server chassis & Digium TE405P?
3:02PM 1 exten => pattern match query
2:42PM 1 x100p card or similar in India
2:36PM 1 Sangoma and Voicetronix cards
2:28PM 1 Lone worker system
1:34PM 0 "module reload" question
1:09PM 0 test message please do not reply and clog up the list
6:36AM 1 Escape characters or replace function
Sunday May 11 2008
5:51PM 0 Digium AEX410
4:24PM 3 Anyone Know How to Have Asterisk Work Like GranCentral and Require a Touch-Tone to Connect?
7:23AM 2 Use safe_asterisk manually, you get colors in CLI. Crontab it, you don't.
Saturday May 10 2008
6:26AM 1 G.722 for polycom
4:33AM 1 ztdummy problems
Friday May 9 2008
10:36PM 1 Unable to build Hpec on Zaptel
9:32PM 3 Calls on E1 TDMoE span are dropped at random
8:24PM 2 Polycom Advanced Features
3:52PM 1 Polycom causes conference to fail
3:36PM 2 DTMF lose with TE-121F
3:19PM 11 Best Linux distribution to use in Asterisk server
2:14PM 1 Asterisk ZRTP?
8:49AM 2 app_queue New Function ToggleQueueMemberUse()
8:09AM 0 Dial Command with the L switch again
6:44AM 0 VUC Friday May 9 @ 12 Noon EDT: Asterisk 3rd party licensing platform
5:07AM 1 -zapg729toulaw did not update samples 160
4:09AM 0 t38modem
3:07AM 0 Zaptel ring voltage detection
2:38AM 1 Text for built-in recordings
Thursday May 8 2008
11:43PM 0 I hear noise in the line
10:17PM 0 Lucent Max TNT PRI Agg --> * --> SIP DEV (PHONE or ATA)
9:46PM 1 MOH and Licensed G729 codec
9:28PM 2 Out-Going Callerid
9:22PM 1 Zap Channels Collide (Incoming & Outgoing)
8:54PM 3 Which Cepstral Voice to license
6:06PM 0 chan_sip Maximum retries exceeded on transmission
4:39PM 0 (no subject)
4:22PM 1 Lucent Max TNT PRI Agg --> * --> SIP DEV (PHONE or ATA) Excessive Echo (only to the sip party)
3:46PM 3 help with rotating number plan
2:39PM 1 Manager API - Setvar not working
10:00AM 2 SLN File Format
8:25AM 3 Newbie Queue: tricky problem with MOH
5:31AM 3 Looking for a Snom expert
5:22AM 0 md5secret for IAX?
2:08AM 2 dundi network - redundancy / fault tolerance ?
Wednesday May 7 2008
10:38PM 1 Big difference in CPU utilization with MeetMe
10:07PM 1 RE:Asterisk 3rd party developed commercial software sales licensing platform
9:39PM 1 show CODEC in CDR
9:36PM 1 Mediatrix 2102's
9:11PM 2 Realtime status feature - user feedback needed
4:49PM 4 VOICEMAIL OPTIONS help needed
2:24PM 0 Problem using the sip_header-function
1:39PM 0 SLA in 1.4.18: i'm going crazy.
12:57PM 1 voice mail indicator on phone
12:39PM 0 reINVITE with Dial() options -- bug 0010647
12:20PM 1 Ubuntu 8.04 + Astribank
11:59AM 2 Setting the TOS using IPtables screws up the DSCP field
10:08AM 1 update DB on ringing/ catch ringing event
9:36AM 1 cdr question
8:48AM 0 IAX IP Trunk + GSM Codec and Noise in the Polycom IP Phone 320
5:34AM 0 queue
5:10AM 4 Newbie alert: VoIP hardware
3:42AM 0 phone status question
2:34AM 3 better enumlookup handler
1:44AM 1 Melbourne Asterisk night
Tuesday May 6 2008
10:54PM 2 Receptionist SNOM-360
7:33PM 2 Storing voicemail on samba share
7:21PM 3 asterisk queue cluster
5:45PM 1 how do I set callerid for incoming iax?
5:24PM 2 Mixmonitor recording issue
4:01PM 2 PRI D-Channel reconfiguration = crash asterisk?
3:57PM 0 finding Asterisk user group users and enthusiasts in the (Salt Lake City Utah, Chicago IL, Boston MA, Tampa FL)
3:03PM 0 Asterisk in Production
2:20PM 3 Performance issues
1:59PM 2 Basic modules of Asterisk
1:55PM 1 DUNDi call impossible in one direction
11:38AM 11 Asterisk in Production ?
1:29AM 1 using cell phone as an FXO port
12:18AM 0 Passing Dial URL argument through Asterisk
Monday May 5 2008
11:59PM 4 Running Asterisk as root
11:09PM 2 AGI - Choppy Sound
8:32PM 1 Call manager using Asterisk as voicemail server (SIP) not working ...
7:59PM 2 DTMF
7:04PM 0 - ribbit mashup
6:47PM 0 Adtran TA-750 channels go onhook
4:54PM 2 T38 Passthrough Verification
3:53PM 1 Asterisk & Bluetooth
3:29PM 0 FW: [Red5] Open-source SIP phone with Red5 and Flex3
2:51PM 5 Asterisk Restarting due to segfault
2:19PM 0 TDM410 and TE205P in a dell 2850
2:14PM 2 Unicall - How to automatically block collect calls
1:59PM 3 TDM410P driver?
1:26PM 0 Problem with transfer (and asterisk -r)
12:29PM 4 [OT] wireless headphone that can answer a call?
11:24AM 4 FW: Asterisk 3rd party developed commercial software sales licensing platform
9:44AM 3 MeetMeAdmin() working problem
8:47AM 2 ResetCDR() - v
8:41AM 1 Playback don't play the beginning if a sound file
8:40AM 3 simple realtime question
8:07AM 0 hangup occurs after 2 rings with B410P
Sunday May 4 2008
7:38PM 2 Astricon?
5:55PM 3 Dialplan, Extensions, etc. Worksheet
5:09PM 1 segmentation fault
3:55PM 1 externnotify php script
5:49AM 0 Director Server
12:26AM 1 UK BT ISDN30e PRI Problem
Saturday May 3 2008
10:29PM 1 AGI asterisk high balance
6:48PM 2 RTP and Sip Provider
5:05PM 0 Apple iPhone visual voicemail
4:19PM 3 Anonymous statistics collection tool for Asterisk servers?
3:27PM 2 China vaults past USA in Internet users - now 220 million users in China
3:18PM 2 toggling recordings on / off using MeetMe()
10:32AM 0 Zap channels hang
7:54AM 0 Attended transfers with original CID information - Polycom
1:51AM 0 Photos and Presentation Materials from HITBSecConf2008 - Dubai Released
1:13AM 1 Asending or Round robin with trunks sip
Friday May 2 2008
10:34PM 0 OT: VoIP 911 Call Misrouted; Child Dies
8:14PM 1 DTMF issues in 1.4.19 with missing digits
6:28PM 0 Problems with a BRI Line
6:22PM 0 One Way Audio After Dial
6:15PM 0 voice mail indicator command
5:25PM 1 rebooting newer cisco phones
4:46PM 0 SRTP between 2 asterisks
2:57PM 0 12 Noon Friday VoIP Users Conference "Weak-ly" Reminder
2:51PM 2 sip show peers
1:58PM 1 Asymmetric codecs in IAX2 trunk
1:46PM 0 Macro
1:45PM 0 Connecting Elite H323 gateways
10:07AM 7 Digium Card: Power Connector, from SATA to NORMAL
5:04AM 1 Asterisk - get Caller String(as per key action)
2:27AM 2 call pickup - Asterisk -
1:35AM 0 sound quality drop after call transfer
12:34AM 2 Stupid Timeout Question
12:25AM 4 e164 Format Numbers
Thursday May 1 2008
11:47PM 0 Asterisk 1.4.20-rc1 Now Available
10:00PM 8 New generic sounds
9:35PM 1 Sound Prompt 'per'
8:41PM 0 Providers
8:19PM 3 Minimum upload speed for Asterisk?
8:10PM 1
5:07PM 3 Zaptel Released
3:39PM 3 Digium PRI card hi-Z for sniffing?
3:38PM 3 How i know the version of my vpmadt032 firmware
2:05PM 0 Background ring
11:59AM 0 PCI ISDN as a PSTN gateway card
9:45AM 1 ast_indicate_data: Unable to handle indication 3
8:49AM 0 Penalty based Cascading Queue - possible ?
7:17AM 3 Newbie: How to remote test a call prolem in an Asterisk site?
6:54AM 1 Remote host can't match request NOTIFY???