asterisk users - Mar 2008

Monday March 31 2008
10:01PM 3 Need some input for Quad T1 and channel banks.
8:45PM 7 Cisco 7965 SIP Firmware
8:00PM 1 Gentilini, Paul is out of the office.
7:33PM 1 How to give user a prompt before connecting thecall
6:54PM 0 CDR Timestamps (cdr-custom)
6:50PM 1 Control of RTP open ports
6:23PM 0 How to give user a prompt before connecting the call
5:12PM 1 AsterPas ObjectPascal Based FastAGI Server goes Open Source
4:52PM 0 SIP proxy screwing up peer addresses.
3:47PM 3 Simple Question
3:08PM 1 asterisk-users Digest, Vol 44, Issue 104
2:53PM 0 Problem with VoiceMailMain
2:33PM 0 Tests in VMWare (was: Re: asterisk-users Digest, Vol 44, Issue 104)
2:25PM 0 How to customize voicemail greeting
1:23PM 0 transfer call
12:32PM 2 UK FXO hangup detection with a twist
12:06PM 0 ENUMLOOKUP question.
9:07AM 0 Broken calls during conversation
9:02AM 1 The most efficient way to know SIP phones IP addresses ?
8:44AM 0 No voice in one direction, SIP, call manager
8:14AM 0 applicationmap in features.conf Asterisk 1.2 is ignoring DIAL tT options
1:36AM 0 Advice on queue setup needed please.
12:50AM 2 Tests in VMWare
Sunday March 30 2008
6:35PM 1 audio disappeared after ztdummy install
5:14PM 0 ztdummy / RTC error
8:56AM 1 breaking DNID into country code, area code, and local code
8:31AM 2 How many maximum SIP Registrations can Asterisk Handle
Saturday March 29 2008
10:38PM 1
2:00AM 2 Finding iaxy's (iaxies?)
Friday March 28 2008
11:04PM 0 Asterisk 1.4.19-rc4 and 1.6.0-beta7 Now Available
9:36PM 2 New Tutorial: Asterisk on EPIA VIA C3
7:03PM 1 how to register IAX user without password for any user
6:06PM 0 More info on my previous dynamic queue question
6:00PM 2 voicemail custom greeting
5:22PM 1 Question on Dynamic Queue and Agent
5:09PM 1 Grandstream BLF and Call-limit
5:09PM 0 overlap calls from NT-BRI timeout problem
3:31PM 1 jingle with Asterisk + PSTN
2:34PM 1 how to register IAX user without password
1:40PM 1 recommendable softphones / X-Lite / Zoiper for amd64?
1:27PM 3 Two phones fail to agree on codec, asterisk at fault?
1:02PM 2 wrong extension status when call-limit=1 is used
11:30AM 1 sip.conf setvar option
10:46AM 1 PRI error cause hangup calls
10:13AM 1 IAX user register problem
8:25AM 2 Call deflection on ISDN PRI in Sweden
6:54AM 1 Need help with voicemail odbc
4:38AM 4 Newbie Polycom: DHCP/boot server supporting 2 models of phones
2:12AM 6 SPA-962+ SPA-932- blf function
Thursday March 27 2008
11:13PM 0 Developer Conference, Aug 5-7, Chicago
9:22PM 1 Problem when leaving voicemail
8:23PM 3 Star Wars Echo Sound
8:00PM 1 Help with cisco 7960 phone
4:40PM 1 Asterisk not picking up (some) calls due to zaptel detecting and clearing alarms
4:32PM 2 callers in queue passed to agents who accept only one call at a time
3:56PM 0 Had it with Dell Garbage - HP Question
12:14PM 1 Unable to establish handshaking with fax machine
12:02PM 0 Astlinux Friday Mar 28 @12 Noon EDT VoIP Users Conference
11:02AM 1 Cisco 7971
10:08AM 5 Problem with socket_process: Call rejected by Busy
8:04AM 2 IAXy device
7:58AM 4 SPA400 vs Rhino/Digium card
7:47AM 2 Calling users to the external domain using Asterisk
7:36AM 1 ADPCM codec and IAXy device
7:28AM 1 Asterisk not hanging up after voicemail
4:00AM 1 Zapata Tormenta 2
3:45AM 3 problem about voice when using TDM2400p with VPMADT032 echo canceller module
3:31AM 0 How to change Internal and external callerid
Wednesday March 26 2008
10:59PM 1 Got SIP response 406 "Not Acceptable"
10:55PM 0 Strange RTP problem...
10:06PM 2 DTMF suddenly stopped working on SIP channel
7:10PM 1 AGI-python script
6:21PM 2 customizing faxrcvd in PHP
3:02PM 1 Playing sound while talking
1:18PM 2 Dialing off-hook with Polycom SoundPoint IP 430
12:12PM 0 Test
11:30AM 5 Asterisk parking hold and transferdigittimeo ut
10:26AM 2 UK GMT/BST settings
7:44AM 0 Fax to DB
7:25AM 1 what's a softphone can activer web browser
6:26AM 0 SOLVED: Avantfax on Debian
6:14AM 2 Avantfax installation on Debian
3:27AM 2 Broadcast/Announce app
12:33AM 2 Realtime replication!!!!!
Tuesday March 25 2008
10:46PM 0 Automatically reload/restart asterisk following IP change (dynamic IP)
9:11PM 0 Distorted Audio for incoming DTMF
7:31PM 1 SIP Domain Authentication
6:44PM 1 Send received fax to different email account
6:31PM 2 RTP Payload Problem
5:43PM 1 Have problem with realtime sql
4:19PM 2 Slightly OT: Getting VOIP number into phone book
3:54PM 2 voicemail.conf fromstring, emailbody - per context?
3:28PM 0 Auto-congest time for sip peers
1:50PM 0 To what degree can Asterisk replace Cisco Unity?
1:48PM 1 Asterisk parking hold and transferdigittimeout
1:35PM 2 CCM and multiple trunks
1:19PM 1 Delete voicemail messages on asterisk by replying to email
12:51PM 1 How to obtain SIPCHANINFO variables within custom application?
10:32AM 1 force soft hangup
6:05AM 0 New Release asterisk 1.6 Beta
4:04AM 1 Sip exten matching based on contact: sip header?
3:16AM 0 Redirect and free the channel
2:59AM 1 Audio Problem...
1:47AM 2 Menuselect?
Monday March 24 2008
11:40PM 4 SIP carrier billing technicalities
7:59PM 0 FYI about my Mona Vie business venture - apology and rethink
7:12PM 1 Passing variables over IAX2 -- IAXVAR patch?
6:56PM 7 FYI about my Mona Vie business venture
6:14PM 4 estimation on phone network capacity
5:03PM 0 Slow compilation speed
3:56PM 3 Dynamic meetme conference creation with Authenticate (Asterisk 1.4.0)
3:36PM 1 G.729 Copy Protection
2:03PM 1 app_sms and smsq in germany
1:05PM 1 Gold Mine CRM + Asterisk
11:41AM 1 Calling extension from CLI?
11:29AM 0 Queue - Dynamic association
10:48AM 0 Greetings
9:42AM 1 g729 license for debian etch
9:02AM 2 Getting Exec Format Error when running AGI call
6:38AM 3 Unable to obtain dialed number through ZAP
3:02AM 3 How to capture destination number when receive call through ZAP
Sunday March 23 2008
6:51PM 0 Problems with calls in asterisk.
6:41PM 1 zap callerid problem
6:22PM 1 Storing voicemail in mysql
3:08PM 1 How to detect if a call is fax or not
2:54PM 3 Unable to capture CallerID through Zap
10:05AM 6 Access rights between AGI and Web server?
7:45AM 2 More Broadvoice woes. Who's fault could this be?
7:28AM 1 No audio on Sangoma A104.
6:28AM 1 voicemail and needed language to be selected
Saturday March 22 2008
7:21PM 3 G723 on asterisk 1.4.1
3:21PM 2 Problem: Digium TDM400 with XOptionsFlex - Line Busy
9:08AM 2 Anyone used Siemens SIP/Dect phones?
8:59AM 1 how to detect redirect fax call
1:22AM 0 --POSIBLE SPAM-- Peticion on line Parque de atracciones mcdonals
Friday March 21 2008
11:36PM 1 wholesale voip provider
9:13PM 4 Calls to sip extensions not defined
5:37PM 1 wholesale voipprovider --starting at 1.1 cent per min
4:08PM 3 Problem with user regsitration and ldap on SVN version
3:20PM 2 wholesale voip provider --starting at 1.1 cent per min
3:12PM 1 TxFax in asterisk 1.4
3:04PM 0 Hardware supporting groundstart signalling
2:57PM 2 Digium registration utility version 3.0.3 released
10:38AM 1 Which command line is used to send emails to notify incoming voicemail ?
10:07AM 0 DRUID/Voiceroute on VoIP Users COnference today Friday 2008-03-21 at 11:58 AM EDT
Thursday March 20 2008
11:10PM 0 BLF on Cisco 7970
7:09PM 1 polarity in zapata.conf
5:53PM 8 BLF and Snom phones
5:40PM 1 423 "Interval Too Brief" and expiry settings in sip.conf
4:18PM 1 Polycom 650
3:45PM 1 More DTMF issues
3:07PM 1 Dialplan Help
1:10PM 0 question on app_conference()
9:46AM 0 Asterisk re-invites and billing
8:32AM 2 hint status unavailable
6:48AM 1 Unable to build smsq on beta6 and x86_64.
5:13AM 1 Newbie: Two problems with Asterisk Config, Please Help
4:16AM 0 AMD timing issues
3:48AM 6 Newbie IVR: How to read() before playback() is finished?
1:06AM 1 Newbie Asterisk: Disaster Recovery Proof Asterisk
Wednesday March 19 2008
10:32PM 1 Call All
9:01PM 1 Multiple sites, same extension
8:14PM 0 Chanspy from a call file
7:41PM 2 Is Asterisk ready for Prime-Time?
7:06PM 0 Inband SIP DTMF
6:12PM 0 question on meetme
5:48PM 3 phpagi
4:46PM 1 Ribbit Demo
4:05PM 0 Can't play recording message wav file
3:13PM 3 How to configure Voice mail for multi users.
3:10PM 0 How configure Voice mail for multi users.
1:41PM 1 Bug in voicemail's serveremail setting in 1.4.18
1:28PM 0 question how to play wave file in meetme and exit
11:49AM 1 fxo tdm400p issue
11:34AM 8 Limit calls when using autodial
9:04AM 1 Query about Bluetooth Head phone
6:11AM 1 Newbie Queue: greetings when first joining queue
5:45AM 0 Deadair in queues.
5:36AM 1 Getting config from SPA-941 or 942 phones
5:01AM 1 Call Screening feature using asterisk
3:03AM 6 Hardphone SIP phone costs
1:30AM 2 Asterisk with lumenvox
12:02AM 0 Asterisk and Avaya 4610 handset
Tuesday March 18 2008
11:37PM 0 AST-2008-005: HTTP Manager ID is predictable
11:32PM 0 AST-2008-004: Format String Vulnerability in Logger and Manager
11:29PM 0 AST-2008-003: Unauthenticated calls allowed from SIP channel driver
11:26PM 0 AST-2008-002: Two buffer overflows in RTP Codec Payload Handling
9:35PM 0 AEL2 Hint & Parking
8:54PM 2 (Critical Updates) Asterisk 1.2.27,, 1.4.19-rc3, 1.6.0-beta6 Released
8:22PM 1 Sip Line Status/Pickup
5:55PM 3 capacity
5:37PM 1 Call forward on Telco line
4:11PM 2 How is uniqueid computed
3:28PM 2 ztdummy problem causing playback () to fail
2:06PM 6 Call signalling on BT FeatureLine Compact (Sangoma A200)
1:29PM 1 Using dedicated eth2 card for SIP trunk line to ISP provider - how to setup ?
12:02PM 1 Sangoma FXO/FXS config
9:40AM 6 Asterisk 1.4 reliability problems
7:20AM 3 Newbie Queue: Simple Queue Problem
6:24AM 2 call screening feature
1:31AM 0 Way to hangup a call between two asterisk machines
1:14AM 0 Subexpression usage in Asterisk Dialplan Regular Expressions
Monday March 17 2008
10:29PM 2 Pre-pending certain digits (like 9) to an outbound call number
8:44PM 2 Order of queue member list
7:42PM 1 Core dump?
6:45PM 1 asterisk.conf uniquename or sysname for uniqueid field in CDR
5:50PM 2 php web chat + asterisk -> callcenter
5:08PM 1 Turn off MusicOnHold for individual User
2:57PM 4 MeetMe option b
12:47PM 0 The switch statement in extensions.conf
11:17AM 1 ldap for sip users.
9:26AM 0 NVFaxDetect and Asterisk 1.6
7:14AM 6 Handling 3 different call ending causes
7:08AM 1 Desperately need help with Asterisk setup
6:06AM 1 update_call_counter: Call to peer '2509' rejected due to usage limit of 1?
6:00AM 3 Newbie Polycom: DND answered as "on the phone" instead of "not available"
1:09AM 1 Redundant Voicemail
12:46AM 2 Newbie ASTDB: cannot replicate among Asterisk servers?
Sunday March 16 2008
10:11PM 1 using the System() command to call a script
8:10PM 1 Problem with incoming calls on Broadvoice after upgrade to 1.4.18
7:53PM 0 DUNDi or ENUM
6:16PM 0 Telemarketer Torture.... (was: Re: asterisk-users Digest, Vol 44, Issue 49)
5:13PM 0 cordless usb handsets: Uniden Win1200?
1:49PM 1 LDAP (was: Re: asterisk-users Digest, Vol 44, Issue 48)
3:08AM 4 Telemarketer Torture....
Saturday March 15 2008
10:25PM 1 Calling a Macro with arguments in AstApplication/AstApplicationData
9:53PM 2 Asterisk VOIP Jobs version 2 Launched!
9:19AM 1 Sip phones for call centers
9:00AM 0 IAX ATA that can be set up and drop shipped to a US address?
12:13AM 2 DID T1 PRI
Friday March 14 2008
10:43PM 1 Callerid Error- Causing All Zap Channels Busy
8:57PM 1 Looking for a cheap SIP termination point.
7:27PM 0 FW: [asterisk-dev] Hardware and CentOS tweaks.
7:27PM 0 FW: [asterisk-dev] Call failed, reason 0 explanation.
6:12PM 1 Trouble with Incoming Callerid on Trixbox
1:56PM 2 Logs for Call generated by Manager API
10:46AM 0 CallerID(num) not showing on cli
9:28AM 0 VoIP Users Conference for Friday March 14th @ 12 Noon EDT
7:29AM 3 Dialing patterns and "GSM" format numbers
6:02AM 1 Group Listen on SIP Phone
5:59AM 3 Anyone know of a pass through ATA
4:51AM 2 Asterisk 1.6
Thursday March 13 2008
10:22PM 1 Multiple clients registering on same definition in Realtime
9:09PM 1 Hardware Platform
9:04PM 5 Mail Server
8:54PM 1 Help: DTMF problem
8:32PM 2 RedFone foneBRIDGE2 2e1 - anyone used it oranother TDMoE bridge?
7:38PM 2 RedFone foneBRIDGE2 2e1 - anyone used it or another TDMoE bridge?
6:19PM 4 Application registration on Asterisk 1.4 and 1.6?
6:01PM 1 CallerID setting issue with withheld numbers and mISDN ...
5:38PM 0 Error in Callback CDR
5:17PM 0 need * consultant in st louis area
2:28PM 2 SNOM on "Do Not Call" list????
2:17PM 2 queue log vs. cdr
2:13PM 1 sip.conf help, inbound calls fall to last specified context
11:22AM 0 OT: RTP - NAT - SBC
4:59AM 3 Newbie One-touch Recording: Does not work (more info)
4:39AM 5 Newbie One-touch Recording: Does not work
3:47AM 3 How to find out the IP of the calling party?
3:16AM 1 T.38 SIP Issues
1:31AM 1 asterisk out of service
Wednesday March 12 2008
9:27PM 0 test please ignore
8:12PM 0 Problem sending CallerID Name to Dialogic based phone app
7:13PM 4 does the meetme module still require an external timing source?
6:20PM 0 SIP Registration!
6:20PM 0 user web interface
5:43PM 9 Druid Open Source Edition
5:40PM 2 DUNDi
2:54PM 0 snom official english forum
9:51AM 0 Expected iax user behaviour
8:39AM 1 include context on [globals]
6:48AM 3 DTMF problems while greeting is playing (Background())
6:22AM 4 authentication number at the end of the number before calls go through.
5:40AM 1 S100I IAXy - Why is it discontinued?
4:39AM 0 chanspy doesn't work properly
1:02AM 2 TXFax/RXFax/AGX-Addons/SpanDSP Crashing
12:18AM 1 Asterisk not transcoding between installed codecs
Tuesday March 11 2008
11:19PM 2 AGI - calling functions, CHANNEL STATUS broken?
10:07PM 1 Sending Incoming Calls straight to park
9:58PM 0 Gartner Article (was: Re: asterisk-users Digest, Vol 44, Issue 32)
9:32PM 0 Asterisk 1.4.19-rc2 Now Available
8:48PM 2 Polycom IP 330 w/VLAN?
8:21PM 7 Best alternative for getting prompts recorded.
7:25PM 1 setting callerid across servers
7:16PM 0 T.38 passthrough in asterisk 1.4
4:12PM 0 Gartner Article
1:48PM 2 Unison
1:41PM 3 Call tracing - Asterisk 1.4
1:00PM 1 Meetme application on AstriskWin32
12:14PM 3 E1 Card emulator?
12:11PM 4 CCM 6 and Asterisk routing again
11:19AM 0 Central Asterisk with remote 'trunking' asterisk gateways
9:25AM 1 WARNING[4294]: chan_zap.c:8851 zt_pri_error: 1 copying 5 bytes LLC
9:19AM 1 Sending SMS
7:37AM 1 Newbie Polycom: IP601 console with expansion module
4:30AM 0 Little help with Conference
1:40AM 3 need * consultant in houston area
Monday March 10 2008
10:36PM 2 About CID with DTMF in Asterisk
8:14PM 0 Disable SIP notify for peers
8:13PM 1 Want to know Frequency and lenght of Frame
6:48PM 0 Audiocodes MP124-FXS replying BUSY when line is not.
5:59PM 5 display time on Cisco 79xx
5:42PM 1 Strange problem
5:02PM 1 Redirecting channels?
4:55PM 1 Asterisk and Packetcable
4:27PM 1 Queue Pickup
3:54PM 1 Intermittent DTMF Problems
3:00PM 1 Shared Extension
2:09PM 2 Global Variables on Reload
1:30PM 1 FaxBack Service with Asterisk
11:12AM 2 dialstatus and cancelled calls
10:53AM 0 call forward facility in INDIA
10:52AM 2 Call forwarding-in india
9:31AM 1 dialplan logic questions: macro not reaching DIALSTATUS based extension (s-ANSWER)
4:25AM 1 1.6.beta5 (format 0x40 (slin))
3:05AM 11 Microsoft Office Communications Server
2:59AM 2 What replaces Macro() now? And how do you do the equivalent?
1:38AM 1 Local music on hold -- mohinterpret=passthrough assymetrical ?
Sunday March 9 2008
11:00PM 2 Dead Air on PF firewall
5:59PM 0 phones start ringing randomly with Grandstream GXW-40XX - solution!
4:23PM 0 replace astdb with a cluster-capable sql database engine (was: Re: asterisk-users Digest, Vol 44, Issue 22)
2:45PM 0 Realtime error - PgSQL with 1.6 beta 5
6:34AM 10 Read function
2:00AM 1 How Do I continue after Dial Command ??
Saturday March 8 2008
7:53PM 1 PRI suppliers in Switzerland
6:36PM 1 I am now on Refriendz!
6:01PM 3 replace astdb with a cluster-capable sql database engine
2:20PM 2 Experiences with grandstream GXW 4024 FXS?
Friday March 7 2008
10:23PM 1 Sync Problem (astribank)
7:46PM 0 Motorola SBV5220
7:36PM 1 sip show channels - gives a growing list of dead channels
7:18PM 0 VoiceMail dialout context
7:01PM 0 How to get call back during attendant transfer?
4:10PM 1 WirelessIP5000 SIP registration problem
12:24PM 2 Background: reading the digits correctly, buffering it, waiting the sound message to complete
10:47AM 0 chan_sip.c:2918 auto congestion
10:36AM 0 Call flows of sequence
10:14AM 3 Asterisk Realtime and SIP configuration
8:34AM 3 Silencing VoiceMail() app in * 1.4.10
7:31AM 1 Newbie MeetMe: How to control max users in conference?
5:49AM 1 Call flows of conference
2:43AM 0 How to return the status of a call to the calling server?
2:14AM 3 is this possible..
Thursday March 6 2008
11:02PM 1 OT: Upgrade Addpac AP200C
10:26PM 0 Receiving double DTMF "if I pressed 1, then asterisk box recognize it 11"
10:06PM 2 Cool New Website
9:10PM 0
8:39PM 0 Asterisk 1.4 w/ realtime static zapata
8:32PM 0 Net Neutrality
8:19PM 2 zaptel compile question
6:51PM 0 Allowguest=yes & language
6:25PM 2 Provider recommendation in USA
4:54PM 1 Call Manager as trunk
4:14PM 0 Asterisk in the call center - how do you do it?
4:07PM 2 format of UNIQUEID variable
4:01PM 1 Best Java library to interact with Asterisk
3:21PM 14 FXS channel banks
2:03PM 0 bristuff qozap support for beronet cards
1:50PM 0 IAX user identification.
1:18PM 0 chan_iax2.c:3904 iax2_trunk_queue: Maximum trunk data space exceeded
12:38PM 0 Tentative: [Invitation] VoIP Users Conference @ Fri Mar 712:00 - 13:00 ()
12:04PM 1 AEL - SQL and TIMEDIFF()
10:31AM 0 Asterisk authentication by SIP Proxy
9:42AM 0 Tentative: VoIP Users Conference
9:40AM 5 Declined: VoIP Users Conference
8:44AM 0 New Time Proposed: [Invitation] VoIP Users Conference @ Fri Mar 712:00 - 13:00 ()
8:42AM 1 Accepted: [Invitation] VoIP Users Conference @ Fri Mar 712:00 - 13:00 ()
8:36AM 0 Accepted: VoIP Users Conference
8:35AM 0 Fax in AGX Extra Addons for Asterisk causes Asterisk to die
8:29AM 2 VoIP Users Conference for Friday March 7th @ 12 Noon EST
8:29AM 0 [Invitation] VoIP Users Conference @ Fri Mar 7 12:00 - 13:00 ()
7:27AM 2 Newbie Polycom: IP600 Headset Problem
7:13AM 0 RFC for conference
5:58AM 0 Message sequence of a conference
2:37AM 1 LDAP
2:30AM 1 Newbie Polycom: how to effect change in sip.cfg?
Wednesday March 5 2008
10:46PM 6 Asterisk based UNIX
10:44PM 1 Voice quality is bad from one side and good from another side
9:50PM 1 Asterisk 1.6.0-beta5 Now Available
9:41PM 0 Asterisk 1.4.19-rc1 Now Available
9:09PM 0 Aastra-Asterisk: 6 beeps then voice quality degrades
8:20PM 4 OT How to Change Polycom Web Admin User:Pass via Web
6:39PM 1 Codec Preferences
6:35PM 1 g729 to GSM translator is needed for voicemail to work fine, how?
6:19PM 3 codec_g729-v34 Builds Now Available
6:17PM 4 Problem between Asterisk and an Aastra 57i
5:50PM 5 C compiler cannot create executables when building zaptel
5:41PM 2 Transferring Unanswered Calls
5:10PM 4 NIN Ghosts music (free download) safe for MOH?
3:39PM 0 DNS Changes never picked up with Asterisk 1.4.18 chan_sip?
3:35PM 0 fonality new version
3:15PM 0 SIP REFER Message, over NAT
2:17PM 0 no audio between two asterisk servers
1:49PM 1 How to restrict a Polycom from receiving unauthorized calls
1:33PM 2 Passing variables between two DUNDi/IAX2 peers
12:49PM 1 Linksys SPA devices and CID
10:12AM 4 {s} - extension
4:51AM 1 Newbie dialplan: dial 0 for outside line
Tuesday March 4 2008
9:32PM 0 missing ${DIALSTATUS} in hangup extension?
9:04PM 1 console dsp
8:34PM 4 Mitel SX-200 + *
8:27PM 1 speaker volume on Polycom SIP phones
7:10PM 2 Asterisk and Avaya...
7:04PM 2 Problems configuring Astribank
6:11PM 1 Cisco 7960 SIP Upgrade
5:38PM 3 PPP dialout via * server
4:20PM 0 Page() command
3:16PM 1 Clustering Meetme over multiple boxes?
1:53PM 1 astmanproxy and core dump
1:48PM 3 incoming call popup
9:52AM 0 [Fwd: OT - CEBIT next week!] - updated list
9:22AM 1 Aastra Park Softkey
Monday March 3 2008
11:00PM 1 ekiga sip registration fails; externip no help
9:35PM 2 Switchvox feedback
9:20PM 0 OT - Mime-construct, exim4 and job numbers
7:45PM 1 Aastra phones and park/pickup feature
7:19PM 2 T1, Rhino, & Nortel
6:26PM 0 Polycom VSX 7000e Series & Asterisk
2:14AM 5 Newbie on VoIP
Sunday March 2 2008
10:54PM 1 Speex: complexity, VBR, ABR, CBR, quality
2:42PM 0 Cisco 79xx users/consultants, 7970G color in particular share information (was: Re: asterisk-users Digest, Vol 44, Issue 3)
2:08PM 0 Cisco 7970 - register with NAT phone
1:40PM 0 OT - CEBIT next week!
11:33AM 3 override/redefine asterisk DB function
9:21AM 5 DID number
12:04AM 0 Cisco 7910 and Asterisk
Saturday March 1 2008
10:48PM 0 scripts to convert .conf files to SQL for realtime
7:47PM 2 I need the least expensive way to do this
6:51PM 2 Page app, Polycom IP 601, 60 SIP peers, Interesting Issue WORKING NOW
5:25PM 0 Mediatrix 1124 and Audiocodes MP Series gateways
4:55PM 4 Cisco 79xx users/consultants, 7970G color in particular share information
2:12PM 7 ASTCC installation error install: invalid user `apache'
9:13AM 0 iax2 reload does not update channel config for removed key from config file
6:17AM 1 Help asterisk connectivity with MS SQL
5:10AM 1 "callpark" feature in ABE?