asterisk users - Jul 2006

Monday July 31 2006
11:06PM 0 Re: If you prefer to read this mail list asa forum ...
10:33PM 0 VM integration to panasonic kx-td500
8:15PM 1 FC4(2.6.16-1.2108_FC4smp) meetme 404 Not Found
7:31PM 0 Selective Router (route based on Caller-ID) configuration
7:03PM 1 Asterisk SIP problems with Nokia E61
6:45PM 0 SIP response 400 Bad request
4:58PM 0 Disconnection During Incoming Call
3:13PM 1 AGI Scripts and CDR
2:51PM 1 RemoveQueueMember isn't working.
2:33PM 0 Goldmine sip client revisited
2:32PM 4 sip phone networking question [possibly OT]
2:11PM 3 IAX over two T1 connections bad quality
1:32PM 5 MWI from Asterisk to Meridian
1:11PM 0 Do zttest results matter without telephony hardware?
12:03PM 0 Playfile waiting for N digits
11:59AM 1 Automatic deletion of voicemail messages older than N days?
10:14AM 1 Problems with supervised transfer and agents
10:08AM 1 music ring (CRBT)
9:30AM 1 not reaching at the destination number I dialed
7:51AM 0 Call Disconnects
6:34AM 0 Port doubler for 8 port BRI cards
6:12AM 2 Voice mail limit
6:05AM 1 Asterisk Polycom_acd_functions and G729
6:03AM 0 Can't load ztdummy
4:52AM 1 app background
4:25AM 1 Testers for ISDN AOC (Advice of Charge (Gespraechsgebuehren)) needed
4:15AM 1 Compiling zaptel on CentOS x86_64
3:41AM 3 asterisk 1.4 download
3:20AM 0 Cisco 2610 RTP port forwarding
3:15AM 1 DNS lookups failing for SIP register
3:10AM 3 Canreinvite and remotely registered devices
2:35AM 2 Voicemail dial pattern from old pbx
2:17AM 0 Postpaid
2:14AM 0 Invalid Conference Number - Meetme Created via FreePBX GUI
1:37AM 1 SIP channel problem
12:22AM 1 Multiple dialing
Sunday July 30 2006
11:38PM 1 Disable native bridge between two zap trunks
11:14PM 2 MeetMe recordings in mp3 format.
10:22PM 1 freepbx and a2billing
10:17PM 0 VoiceMail Name Variable in Dial Plan
8:50PM 2 New Asterisk GUI
3:33PM 0 Server for Asterisk PCI
2:59AM 0 Hangup detection with Sangoma A200 in the UK?
2:51AM 1 Zap Problem
2:35AM 0 Error On brdging Call
Saturday July 29 2006
11:05PM 2 FYI - first release of alarm response code.
6:02PM 0 voice format changed to 4
3:37PM 2 Polycom 1.6.7 Firmware Messages Button
11:26AM 0 agentcallbacklogin Asterisk V1.210 and v1.4
10:07AM 1 How do you recompile individual source modules?
3:07AM 1 where to read stderr.out from an agi script
Friday July 28 2006
5:43PM 1 can't retake call after dialing through Zap/E1 wich doesn't answer
5:10PM 1 Asterisk AGI cmd Record
1:56PM 3 AEL2 Looping
1:12PM 0 Hairpin Detection issues
1:02PM 3 need a pointer regarding scripting asterisk
12:45PM 1 Announce queue?
11:49AM 11 VoipNow 1.2.0 Beta
10:30AM 0 Clicking noise when load XP100 zaptel driver (at boot time)
10:26AM 1 SendText() & displaying text messages on a SIP handset's screen
8:53AM 2 Source Directory of ASterisk
8:45AM 0 asterisk cdr shows "FAILED"
8:37AM 1 wav49 for voicemail attachment not playing
8:02AM 0 R: Canreinvite
7:55AM 0 Extending call parking to display park extension on the handset display
7:54AM 0 SMS functionality of bristuff (0.3.0-PRE-1r) with a Junghanns "duo GSM PCI" card
7:34AM 2 One extension to ring on multiple outside lines
6:58AM 0 Which card do you recommend for heavy load application?
6:47AM 3 Asterisk VOIP / Mikrotik
6:40AM 1 Change the from@ using the voicemail.conf
6:18AM 1 Install asterisk-bristuff for Debian Linux
6:17AM 1 sendtext or sip message - where in RFC
5:52AM 4 Grand stream 2000 will not dial *xx
5:04AM 1 Zaptel trunk failed to compile
4:45AM 0 Weird E1 problem
4:41AM 10 cmd DIAL - Who picked up the call?
4:19AM 1 stream file outputs only silence, even with asterisk example gsm file
3:45AM 1 Voicmail Question
3:35AM 1 registration process
2:57AM 1 Transfer call in SIP
2:34AM 2 CDR IP Authorization
1:59AM 0 Sending email after voicemail
1:55AM 0 asterisk+ooh323.. one way audio issue
1:11AM 2 PAP2T always busy on incoming calls with zaptel
12:34AM 1 FreePBX Inbound Route
12:25AM 4 Fritz!Box Fon ATA
Thursday July 27 2006
11:12PM 0 asterisk with CSTA using VAIL SIP TIM
11:03PM 0 CSTA support for astersik
9:34PM 0 Asteriskguru switchboard
6:44PM 2 Looking for carrier grade redundant solution
6:41PM 1 Rate engine AGI?
6:18PM 9 Strange behaviour Panasonic KX-TD1232
4:26PM 2 Trunk transferring?
4:02PM 1 accessing dialplan global variables in agi
3:50PM 1 Asterisk 1.4 Schedule and Features/Changes
2:39PM 3 long distance ethernet & Asterisk
2:06PM 5 Getting no Audio with G729
1:59PM 2 gxp-2000 configure line appearances
1:43PM 1 IAX2 Connection fails over time...
12:56PM 1 Detecting voicemail from CO on FXO port and passing to H.323 phone. Possible?
12:36PM 3 Anyone tried vitelity?
12:14PM 0 Re: Goldmine SIP client/softphone questions continued: (Dan Elder)
12:02PM 0 DTMF Dial Tone
11:49AM 0 SIP phone w/ 'modem/data' port?
10:54AM 0 Goldmine SIP client/softphone questions continued:
9:41AM 0 OFF-TRACK: Is VOIP -PSTN integration legal inChina
8:09AM 2 SIP client with video???
6:33AM 1 Linksys SPA-3102
5:36AM 1 Problem with call receiving (Asterisk+PSTN+Digium TDM04B)
5:35AM 1 playing a sound into a meetme conf
5:26AM 1 Rxfax and squashed TIFF
4:06AM 3 dropping calls in the middle of conversation
3:47AM 6 Manager interface
3:11AM 3 alcatel ip touch 4068 ... sip?
3:07AM 1 Nokia E61/E70 not always answering voip calls
2:40AM 0 Malformed/Missing URL Problem with Cisco Callmanager 4.1
2:36AM 0 [oh323]FastStart/H245Tunnelling/H245inSetup
2:35AM 2 Mobile SIP Client
2:00AM 1 SV: Sip phone settings set when user registers
1:39AM 3 Sip phone settings set when user registers
1:32AM 1 french promt
1:13AM 0 CDR dest question
12:35AM 1 Multi Asterisk Server to relay call request
12:04AM 2 Reload of wct4xxp without restarting of Asterisk?
Wednesday July 26 2006
10:48PM 1 Determining what gets written to the dst field for a CDR
10:43PM 1 OFF-TRACK: Is VOIP -PSTN integration legal in China
9:30PM 1 Top Users in a MeetMe room???
8:01PM 0 Polycom 501 provisioning : how to secure valueslocated in plein text files
7:22PM 0 playing a sound into a meetme conference
7:03PM 2 Polycom 501 provisioning : how to secure values located in plein text files
6:58PM 0 Developing VoIP with Asterisk (hardphones & softphones)
6:50PM 1 Cisco 7960 Call Waiting Beep
5:06PM 3 HP DL380 and the TE4xxP cards
2:40PM 5 Strange Error when calling
1:09PM 0 problems with IAX, extension recognition and Asterisk
12:49PM 2 Developing VoIP with Asterisk
12:07PM 0 Sip phone settings according to logged in user
11:44AM 1 Sony Ericsson F250m, Sipura 3000 and Asterisk
11:11AM 0 Just bought a Polycom 501 - I feel likemyGXP-2000was better...
10:18AM 1 Which ATA to test T.38 ? What about Linksys 3102
9:58AM 1 SIP is not working sometimes. IAX is working fine. Why?
9:55AM 1 Dial exited non-zero, only if PSTN/ZAP/E1 hangs up first. not if voip hangs up.
8:24AM 2 2 * servers, IAX connection between to dial extensions across IAX - not working
8:15AM 0 FS: 2 x Asterisk X100M (red) daughterboard cards - brand new.
7:19AM 2 Message waiting question...
6:48AM 4 CSTA support for asterisk
5:50AM 0 wip-300 question on audio dial out with tdm2402e
4:29AM 3 Zip code, city and area codes
2:38AM 8 Ringing timer
2:17AM 3 E1 connectivity question
1:40AM 1 Asterisk with Linksys SPA-3000
1:32AM 1 Extension planning
12:05AM 1 Fwd: Problem with
Tuesday July 25 2006
11:58PM 2 Queue announcement issues
9:13PM 2 MWI from Octel to Asterisk
8:58PM 0 Just bought a Polycom 501 - I feellike myGXP-2000 was better...
7:04PM 2 odd sound between SIP & IAX clients
6:37PM 1 T.38 call with t38 in original SDP fails
6:33PM 1 Play sounds to the callee and the caller synchronously when call begins
6:06PM 1 Change current working directory to /tmp
3:26PM 0 sounds format
3:24PM 0 How to send a signal via E1/T1 ISDN to asterisk, to ask the call to be moved.
2:34PM 0 PRI died and Asterisk crashed
2:34PM 3 Rookie voicemail user question
2:12PM 4 Sangoma Stops Receiving Calls
2:03PM 2 sip realtime
1:42PM 1 Voicemail Forwarding
12:44PM 0 sdp multipart information nortel
12:25PM 4 PRI vs "Digital Trunk"
11:49AM 0 All Extensions Dropped
11:25AM 19 Caller ID on Transfers
10:53AM 1 transfers from an E1 using 2b-channel or similar anyone? (QSIG?)
10:06AM 0 netstats like command for sip , Is there one ?
9:41AM 0 MoH clicks and pops
9:27AM 0 Call transfer asterisk + with SPA-1001
8:59AM 2 Connecting branch offices through IPsec tunnel --latency effects?
8:28AM 2 New message
8:27AM 2 SIP and podcasts
8:25AM 5 Connecting branch offices through IPsec tunnel -- latency effects?
8:23AM 2 Recommend hard phone which supports IAX2?
8:23AM 3 Still voice with echo
8:13AM 1 RE: Just bought a Polycom 501 - I feel like myGXP-2000 was
8:02AM 1 FW: IP CDR
7:47AM 0 IAX2 Variables
7:43AM 3 vegastream 50 FXO DTMF Problem
7:36AM 1 G729 License to Bridge calls through VOIP provider?
7:19AM 3 One way "screech" or tone
6:15AM 0 IAX ATA with FXO
3:18AM 0 SIP user deny and permit for calls through Asterisk
3:12AM 6 Binary/unreadable configuration files?
2:45AM 0 Double Ring on Asterisk 1.2.x (fwd)
2:21AM 0 Conference help
1:19AM 0 Re: FW: meetme application doubt
1:01AM 1 Force peer source ip
Monday July 24 2006
11:53PM 2 TDM01B -1 FXO card not working.
10:32PM 1 Asterisk/GPL and G.729 licensing
10:28PM 0 kernel: Error! while loading wct4xxp module
10:22PM 1 Just bought a Polycom 501 - I feel likemyGXP-2000 was better...
7:52PM 1 Unicall reload problem
5:50PM 3 Voice with echo
2:02PM 1 ERROR 1045 (28000): Access denied for user
2:02PM 0 sipbuddies realtime fields and latest documentation
1:56PM 0 sms on wifi phones
1:55PM 3 Just bought a Polycom 501 - I feel like my GXP-2000 was better...
1:47PM 2 RDNIS and IAX2
1:40PM 1 create custom cdr's
12:51PM 1 Urgent source code changes needed
12:03PM 2 Goldmine CRM softphone + asterisk
11:31AM 0 Zap DMTF detect error
11:24AM 3 Polycom_acd_functions SIP trouble
10:42AM 0 Lots of Asterisk child processes
9:59AM 0 core dumps when phpagi script ends?
9:47AM 2 Asterisk Realtime Macros
9:46AM 2 Intercom feature on Polycom phones
8:41AM 3 G729 Softphone
8:29AM 0 Odd SIP timeout
8:20AM 3 Clocking Multiple T1 Cards
7:59AM 0 Asterisk and Vigortalk problem
7:58AM 3 Circuit/channel Congestion
7:38AM 9 Transfers - No ringback or moh
7:37AM 4 Operator in Voicemail
7:33AM 1 reboots itlself
7:30AM 1 AstLinux 0.4.2 Released
7:29AM 0 Astrisks compatable cards
7:24AM 0 playback / stream file
6:54AM 2 How to receive a phone call each time you receive an email ?
6:50AM 1 Asterisk, IAXModem and Hylafax
6:24AM 1 H.323 an IAX
6:16AM 1 asterisk extra sounds: what for?
6:06AM 0 PRI got event: HDLC Abort (6) on Primary D-channel of span 1 (fwd)
6:05AM 0 compain
5:57AM 0 Asterisk and Phonesystems ...
5:20AM 1 Connecting Asterisk to a Metaswitch
5:01AM 5 Voicemail not sent via email
4:11AM 2 Regular expression problem
4:10AM 4 Mitel 3300 + *
3:16AM 0 Multiuser and analog port
2:35AM 0 Transfering Problem
1:29AM 0 Asterisk Updates
1:14AM 8 overlapdial and DID not always working
12:09AM 1 Solution init.d scripts for CentOS 4.3
Sunday July 23 2006
10:08PM 1 Missing close quote in CallerID breaks SIP. . .workaround?
10:01PM 0 MeetMe in Realtime
9:46PM 0 (no subject)
7:30PM 4 Asterisk autoloading of card modules
4:32PM 0 Problems with freePBX and Fax reception
11:30AM 1 How to connect XLite with another public IP?
7:50AM 2 SIP Woes
5:52AM 1 G726 codec softphone
1:05AM 0 Request for some help....
Saturday July 22 2006
11:24PM 1 Trouble configuring TDM400P on Dell SC420
7:23PM 0 SNOM missed call.
4:20PM 3 newbbie question
3:45PM 3 Operator Console(s)/Shared Call Appearances
1:00PM 6 Asterisk Dial Plan to Play Message
12:02PM 2 X100P clone not working
8:47AM 0 SIP reinvite _and_ NAT
6:47AM 1 Upgrading my office - Need help
3:00AM 1 cannot received calls in pstn line
2:34AM 0 NAT and externip problem or bug
Friday July 21 2006
10:06PM 1 Cyberdata paging speakers - anyone use them?
4:49PM 2 Information about Softphone support G729 ?
3:37PM 1 Digium TE110P IRQ
3:20PM 0 China
2:43PM 0 Yate client
2:26PM 5 Associate manager events to a previous Originate action
1:13PM 0 Weird Hold Problem
12:57PM 0 MFCR2 Patch
12:53PM 2 Invalid module format (ztdummy)
12:08PM 0 AGI record_file
11:36AM 7 Germany VOIP provider
10:24AM 1 [OT] Windows softphone with handset support?
9:37AM 1 Error in ubuntu dapper
9:24AM 7 ftp setup for Polycom phones
8:38AM 3 How to connect 2 AAH
8:23AM 8 Sipura ATA's Forwarding PSTN Calls to Asterisk
7:49AM 2 Queue Persistence with queue.log
7:43AM 0 MySQL question
7:20AM 1 did sometimes not working
7:07AM 0 Asterisk internal extensions caller ID
7:02AM 5 question about asterisk DB
5:49AM 0 problem with iax -> sip across 2 asterisks
5:36AM 0 help for SPA-2100
5:28AM 0 Transfering a caller_in_queue to a conference room
5:26AM 0 I: ooh323c - cdr problem
4:51AM 1 asterisk-1.2.9 /
4:02AM 1 How to connect XLite with public IP?
3:47AM 1 Problem with NAT
Thursday July 20 2006
11:47PM 0 Voicemail volume patch
8:46PM 0 Asterisk / Avaya 70XX
8:35PM 10 Typical Asterisk Company
7:53PM 9 If you prefer to read this mail list as a forum ...
7:26PM 0 Re: OT: Project Management & Collaboration Software
7:16PM 1 Asterisk dead-air issues with Digium TE110P and IVR/meetme/internal directory-
7:10PM 0 realtime function
4:41PM 1 OT: FOP examples
4:08PM 0 Asterisk and PacketCable PROJECT
3:39PM 4 A very lost newbie.
2:43PM 2 Asterisk fails to register, when the full logging is turned on
1:56PM 0 Re: [Asterisk-java-users] A newbie introduction
1:40PM 2 Source Clock
12:46PM 1 Overriding # at the end
12:30PM 0 Re: asterisk-users Digest, Vol 24, Issue 116
12:30PM 1 Re: asterisk-users Digest, Vol 24, Issue 116
12:27PM 3 Unicall, not HOW but WHY
12:04PM 1 Interested in IVR information
11:47AM 1 Cisco 7960 - automated send DTMF digits after dialing?
10:43AM 1 Aastra 9133i w/NAT and Asterisk
9:49AM 0 regexten / Realtime WAS DUNDI / regcontext
9:45AM 8 Redundant Ethernet
9:10AM 0 Polycom IP301 and Queue questions, deployed environments
9:01AM 1 Agent Attended Transfer Without DTMF
8:55AM 0 PRI channels filling up
8:44AM 0 Problem handling agents and queues vía RealTime
8:05AM 2 Two phone numbers, one SIP provider
8:02AM 0 Calls waiting announcement with two or more queues?
7:50AM 1 Fast busy after one digit dialled? - 7970 SIP 8.0.3
7:05AM 0 agentcallbacklogin (logging out of)
6:26AM 3 ACD Queues Agents logout
6:10AM 7 *****SPAM***** Load balenced (ADSL) network connections, is it possible?
5:33AM 1 Has anybody in here created their own softphones?
5:16AM 1 Voismart GSM - no billsecs
4:51AM 0 Has anyone programmed their own user\client software for asterisk?
4:06AM 1 Automating the registration process
4:05AM 2 Writing own applications for asterisk - read CALLERIDNUM
3:57AM 1 meetme application doubt
3:53AM 1 Macro help needed!!!!
2:31AM 1 all call forward
1:10AM 1 setting call-limits
12:23AM 2 IP CDR
Wednesday July 19 2006
11:48PM 0 !! Got a UA, but i'm in state 1
11:41PM 0 [Fwd: [Fwd: polarityswitch: no ringback]]
10:30PM 0 Polycom Silent ring
9:29PM 0 question about function realtime
8:17PM 2 Unicall in Australia
6:29PM 1 Help with sip debug?
6:25PM 0 Asterisk process run amock
5:20PM 0 Realtime, ODBC Voicemail, and multiple asterisk servers?
4:45PM 1 RE: [asterisk-dev] How to send DNIS(B-party number) in IAX trunk
4:37PM 0 RE: [asterisk-dev] How to send DNIS(B-party number) in IAX trunk
3:27PM 3 Warm transfer issues in 1.2.10
3:10PM 1 RE: $3,000 server
2:37PM 1 OH323 registration with gatekeeper problem
2:19PM 1 SIP Registration conundrum
1:52PM 0 SipAddHeaders Question
12:40PM 1 Identifying invoking party for a feature
12:39PM 0 Server locking up again
12:13PM 1 Is dmtfmode used/valid in iax.conf contexts?
11:17AM 1 MoH from Sound Card: Does it actually work?
11:09AM 0 inbound sip rtcp hangup
10:39AM 1 Can't get blind transfer to work
10:28AM 1 Stuck ACD Agents
9:58AM 0 Asterisk patches for packetcable
9:49AM 6 Simple But important question (for me)
9:34AM 1 SV: Queue hold position in other language?
9:16AM 0 asterisk core dumps on a Sipura forwarded to a queue/moh
9:15AM 1 Zaptel Problem - Unable to create channel of type 'Zap'
8:48AM 1 Queue hold position in other language?
8:45AM 2 Zap channel faxing in or out fails but phone calls work.
8:37AM 0 Choppy/Jittery playback at beginning of calls
8:35AM 0 Problems after upgrade asterisk
8:27AM 1 Callback: Dial(dummy) 10 seconds rining without costs?
6:27AM 5 Don't Hit # after 9 to get PSTN line
6:05AM 0 emulating key system - pick up so and so on line1
4:59AM 2 QueueMetrics 1.2.1 released today
4:48AM 0 -- Going to extension s|1 because of immediate=yes, but immediate is 'no'
3:19AM 1 Issues with MeetMe
3:12AM 1 Finding far end echo in Verizon network
3:11AM 0 Dynamic Queue Members never called
2:53AM 1 BudgeTone BT-102 not registering to Asterisk
2:50AM 1 QuadBRI + TDM + GSM hangup problems
2:34AM 1 Issue with g729 codec
1:33AM 1 Unicall libmfcr
1:14AM 1 Alternative (?) ways to handle G.729 and annexb
1:10AM 1 Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?
1:05AM 1 Zaptel Compilation Error
12:46AM 4 header replacement
12:19AM 0 Uni Call
Tuesday July 18 2006
11:21PM 1 Keep Zap Channel from answering
10:58PM 8 Please suggest me Best VoIP Service Provider
9:57PM 0 FW: How to send DNIS(B-party number) in IAX trunk
6:48PM 1 Problem with MFCR2
5:28PM 0 zapata.conf pri
4:32PM 0 AW: Using dproxy to solve "no DNS hangs everything"problem?
4:22PM 2 Using dproxy to solve "no DNS hangs everything" problem?
3:11PM 1 Install H323
2:04PM 1 Polycom 601 and Paging
1:37PM 1 Macro call uniqueid
1:34PM 1 ISDN Protocol
1:09PM 1 Hints to help me debug cdr_odbc not inserting
1:05PM 1 Serveremail Setting Does Not Work for Text Messages
12:24PM 3 emulating key system - pick up so and so on line 1
12:02PM 1 Astribank?
12:00PM 0 Ignore This
11:50AM 0 SPA-2000, Asterisk 1.2.4 & Incoming call success? Anyone?
11:36AM 1 Error: Dropping incompatible voice frame
11:10AM 1 - Broken?
11:01AM 0 rxfax Got hangup
10:20AM 0 Net::CSTA on CPAN
10:08AM 2 extensions.conf 4 digit dialing question
9:51AM 3 Asterisk Crashing
9:41AM 1 Examples of handeling input from phones with PHP
9:22AM 0 call-limit and problem with freezy phones. also freezy zap channels with x101p card.
8:46AM 2 Reload clears queue stats
8:27AM 0 Buch Bekanntmachung: Der Weg zu VoIP Asterisk von A bis Z
7:39AM 1 PAP2 TUI Configuration Menu
7:28AM 3 GSM gateway flooded cell - how to detect?
7:03AM 0 Asterisk Trunk Name Problem
6:49AM 0 External call press 1
5:46AM 0 Reinvite and NAT -> Problem
5:40AM 0 ooh323c - cdr problem
5:40AM 0 Called party cannot hear caller
5:16AM 0 Other phone continues to ring when pick up a call with *8 on SVN HEAD
4:16AM 0 GSM Module not picking up DTMF digits from VOIP FXO Gateway
3:30AM 2 CentOS 4.3 and Zaptel-1.2.7
3:16AM 1 realtime oracle dialplan select
3:06AM 0 usage of ast db
2:35AM 0 Asterisk v/s other Telephonic plants
2:17AM 0 how to enable users on other iax server call my iax users
1:56AM 6 call forwarding to mobile phone
1:33AM 1 SIP ATA Channels for outbound calls - How to select in dialplan
1:28AM 0 Forward call
1:15AM 2 don't hear start/begin of voiceprompts
12:19AM 1 link quality is poor
12:05AM 3 polycom 601 manual config?
Monday July 17 2006
8:15PM 0 app-callforward
6:40PM 0 Hardware/Software suggestions, Supermicro 6024H-TR?
6:29PM 4 PRI and Asterisk
4:55PM 1 Dlink DVG 1120S/Asterisk VoIP to PSTN
4:42PM 1 phpagi problem
4:26PM 1 asterisk and spandsp and rxfax
3:31PM 1 Asterisk H323 and Alcatel 4400
3:26PM 1 Voicemail and Polycom ip301
3:13PM 1 Two security holes fixed in latest versions of Asterisk
2:52PM 0 Unable to find extension in context ''
2:24PM 1 RE: asterisk-users Digest, Vol 24, Issue 86
1:03PM 1 Extensions Register but don't ring when called, can call others though
11:30AM 2 Passing Variables with IAX
11:29AM 0 Queue Penalty
11:19AM 1 show channels
10:31AM 1 an ATA with lamp support
10:14AM 0 Call information on blind transfers
10:10AM 2 Setvar=var=val in sip.conf
10:00AM 2 ooh323c - cdr
9:40AM 2 How many users on an asterisk box behind a dsl can you have
9:06AM 1 Cisco 7960 SIP 8-3-0
8:36AM 0 Cisco 7960 SIP 8-3-0 getting "Got SIP response 400"
8:35AM 0 R: R: Called number on ISDN
8:29AM 0 Queue Transfers
8:25AM 0 MOH With Asterisk Controlled Transfers
8:03AM 0 Current radius patches
7:56AM 0 One extension can transfer internal calls, can't transfer incoming external calls
7:25AM 1 What is ZapRas used for ?
7:25AM 2 can no more compile zaptel !!!
6:17AM 1 [Fwd: where is the error?]
5:46AM 0 asterisk sending connects when it shouldn't (is there a q931-INFORMATION equivalent in IAX2 ?)
3:12AM 0 question ast db
3:03AM 1 asterisk sending connects when it shouldn't
2:16AM 4 problems to call brazil from germany
1:24AM 1 DTMF in QUEUES dont work
1:22AM 2 Parked calls
Sunday July 16 2006
11:05PM 6 Testing 911?
10:05PM 3 zaptel on dual processor, How?
10:02PM 2 Sphinx and Asterisk Integration, How?
6:51PM 1 Polycom phone cycles between UNREACHABLE and REACHABLE
6:18PM 2 Vicidial + Unicall mfcr2
4:15PM 5 Polycom IP301 and Queues
3:31PM 3 Regression testing dialplan changes
1:54PM 1 Queue RoundRobin
1:29PM 0 Automation of call initiation
11:36AM 1 Setting a threshold for asterisk to take ZAP line off hook ?
9:54AM 1 Injecting prerecorded audio into active call
9:41AM 1 sending flash using DTMF
8:56AM 1 OT: Skype protocol cracked?
8:08AM 1 7970 SIP configs
1:53AM 4 SRTP enabling
Saturday July 15 2006
6:13PM 1 How to create or test tone configuration to include them in zaptel
2:59PM 2 DUNDI / regcontext
1:00PM 1 compiling zaptel 1.2.7 error: stray '\194' in program
5:22AM 4 PRI dropouts - solution I hope...
4:18AM 1 Blog about asterisk and voip techology
1:28AM 1 Manager action "hold" missing?
Friday July 14 2006
8:40PM 1 Tough time getting Polycom phones to register after router reboot
8:18PM 3 SIP configuration by group
7:19PM 0 489 Bad Event
6:04PM 2 PRI dropouts
3:33PM 0 THOR-2 support
3:21PM 14 Hitting # to Transfer out of a Queue
3:14PM 7 Asterisk 1.2.10 and Zaptel 1.2.7 released!
2:20PM 0 Can not check voicemail from outside
2:11PM 0 Transferring out of Queues
2:01PM 2 Clearing variables in the dialplan?
1:50PM 0 Update for trunk?
1:33PM 1 ATCOM's AG-188
1:28PM 1 Polycom - simpler transfers?
1:13PM 4 Snom 300 headset with static noise
12:11PM 0 Install Asterisk on VPS
10:43AM 0 Linksys SPA941 - low Static Noise? or some parameter in hands
10:16AM 4 Polycom config file location
8:14AM 0 Transfer ACCEPT followed by DECLINE
8:09AM 1 Cisco Gateway & CallerID Name
8:07AM 3 "Legacy" analog data modems and Asterisk
8:04AM 0 Caller ID on a Sangoma
7:36AM 1 Can incoming alternate rings be discriminated?
6:59AM 0 SIP-> H323
6:44AM 2 Contacts for Chan_gsm_bt maintainer?
6:34AM 2 Again on ISDN - MSN in Italy
6:34AM 3 R: Called number on ISDN
5:55AM 1 Called number on ISDN
5:07AM 1 Call queue drops call after 1 min
4:01AM 4 asterisk + centos 4.3
3:57AM 2 astbill + mysql 5
3:54AM 0 Nokia Primicell and Asterisk ? Hangup and Answer detection ?
3:47AM 1 billed calls when cellullar phone is unreachable
2:15AM 0 ACD rejected calls with out going to Voicemail
Thursday July 13 2006
7:56PM 2 need a pointer about scripting asterisk
7:42PM 1 No ringing on outgoing SIP calls.
6:35PM 0 Faxing over CCM SIP trunk to asterisk
5:20PM 0 SPA-3000 XML Config File
5:10PM 0 CT3 cards
4:35PM 1 DUNDi 'Unable to Find Key'
2:44PM 1 Wrong account code from iax_buddies
1:43PM 1 Asterisk instances on VPS
1:29PM 2 FW: Are FreePBX Extensions not being created in asterisk? & FOP question.
1:18PM 0 Delay on ring after dial Out
12:37PM 0 Extensions not busy showing as busy
12:28PM 0 Voicemail Getting Cut Off after 5 seconds
11:04AM 0 SIP adapters questions
10:47AM 3 quad T1 pri
10:11AM 0 Mediatrix 1204 and Asterisk 1.2.9 stops working intermittently
9:50AM 4 How do you harden an Asterisk install?
8:44AM 2 New York city Asterisk consultants
7:56AM 1 cdr functions change between * 1.2.4 and (agi)
7:51AM 1 Voicemail & CallerID
7:17AM 0 Asterisk Console Colorization Question
6:53AM 1 Can I register multiple TERMINATORS to a single account on IAX?
5:59AM 1 Connect to 'agi://blablabla' failed: Operation now in progress
4:11AM 3 SIP To: header
3:35AM 2 asterisk dual servers through iax: Accepting UNAUTHENTICATED call
3:02AM 0 H323 implementation
2:55AM 2 Using DUNDi with TrixBox mini HOWTO
2:43AM 1 sending out fax using asterisk
1:45AM 1 Very bad quality withAVMFritz!cardPCIandchan_capi
1:12AM 1 IAX2 vs TDMoE
1:09AM 2 Channel Redirect
1:05AM 0 Cisco 7912 IP Phone - Convert SIP to SCCP
1:03AM 1 CDRTools please help
Wednesday July 12 2006
11:17PM 8 priority problem
10:20PM 2 Names Mark Spencer of Digium to its “30 Under 30: America’s Coolest Young Entrepreneurs”
8:49PM 0 console/dsp and autoanswer
7:20PM 6 Polycom compatible phone for Asterisk
6:42PM 0 IGNORE: test email
3:59PM 1 Recording/Monitor after xfer
3:32PM 2 founded
3:17PM 2 DTMF detection and Sangoma cards
2:53PM 1 Cisco 7940 dialplan.xml
1:34PM 1 sip, dbsecret, and dundi
1:28PM 0 ttp question getting connection timeout.
1:16PM 1 Trouble with call file
1:09PM 1 FW: $3,000 server
12:55PM 0 Very OT: For the Record
12:51PM 4 RE: $3,000 server
12:37PM 0 Agent login problem with MP 124
12:34PM 0 (no subject)
10:39AM 2 FXS adapters and Polycom phones
10:33AM 1 an operational scenario
10:20AM 1 where the bottleneck lies ? (was: Serverredundancy)
10:17AM 1 Exclude a certain route from using a trunk
10:13AM 3 PCMCIA card support
9:34AM 0 Call Parking breaks suddenly
8:06AM 0 Hardware... dimensioning ??
8:05AM 1 Email notification of voicemail
7:45AM 4 comcast info -- somewhat offtopic
7:11AM 0 Lets All Get Smart...
7:08AM 0 Option D in dial doesnt seem to be working
6:48AM 0 Echo on PRI
6:31AM 5 Asterisk version: or older?
6:29AM 0 waitexten only provides one digit in chan_zap
6:22AM 3 Problem with making outgoing calls
6:15AM 8 1000s of extensions in one context?
5:39AM 0 Automatic Hangup problem on IAX2 communication to Asterisk
5:28AM 0 Problem incoming calls from sipphone/giztmo
4:46AM 0 IVR with LDAP query for phone number and mobile number??
3:49AM 2 Queue menu
3:34AM 3 Possible polycom_acd_functions BUG
3:12AM 1 Urgent context
2:39AM 1 asterisk + nite affiliates
1:59AM 0 dial plan -- help
1:58AM 1 Polycom ACD, Asterisk, Kernel 2.6 - now SIP does not register
1:30AM 2 IAX2 trunking problems
12:56AM 0 Urgent call forward
Tuesday July 11 2006
9:55PM 0 TE110P configuration problem
7:16PM 1 Polycom, TFTP, and DHCP
7:16PM 0 Problem - Can't pickup call
6:27PM 0 register process flow
5:03PM 0 multiple authentication realms
3:40PM 14 NuFone, please send the log file
3:29PM 2 Intercom mode on Polycom and/or SPA9xx
2:54PM 0 taskset with asterisk
2:26PM 0 2 legs and cdr's
2:21PM 0 Question on event AgentComplete of Manager API
1:33PM 0 CDR Call Status
1:26PM 3 Polycom ACD, Asterisk, Kernel 2.6
1:15PM 0 Inconsistent call detail records
11:51AM 2 So many configuration files!
11:09AM 0 several asterisk servers questions
9:49AM 2 MFC/R2 country and carrier specific protocol variants
9:13AM 1 what single PRI interface, from which manufacturer
8:45AM 0 [announcement] kansas city asterisk user group
8:29AM 3 Issues with making Transfers
8:20AM 4 Asterisk stops abruptly
8:13AM 0 IPKALL direct to asterisk bypassing FWD
8:04AM 1 RE: [Asterisk-video] Asterisk as an MCU
8:02AM 2 How to do load balancing (1:1) with IAX and two different ISPs
7:48AM 1 WARNING[30954]: chan_sip.c:2734 sip_indicate: Don't know how to indicate condition 9
7:44AM 2 Server Optimization and Load Balancing
7:17AM 1 Yet another problem with incoming SIP calls and 407
6:40AM 1 Rate or rank ITSP
6:23AM 4 New Asterisk server crashes daily
6:14AM 0 stuck/phantom zap channels
5:59AM 1 Having trouble to receive fax from samsung sf3200
4:54AM 6 Provider UNREACHABLE
3:57AM 0 WG: CDR ist getting wrong status
3:55AM 1 Anyone out there using Junghanns ISDNguard?
1:41AM 0 SRTP or zrtp
12:47AM 0 sip_poke_noanswer: Peer xxx is now unreachable
Monday July 10 2006
11:52PM 1 Asterisk Servers problem?
11:13PM 1 2 NICs; Asterisk receives on eth1 and replieson eth0
9:31PM 2 2 NICs; Asterisk receives on eth1 and replies on eth0
8:54PM 3 Text priority labels not working for me
7:29PM 2 Asterisk and NEC NEAX 2000 IPS
6:40PM 2 Problem with GotoIf in dialplan
6:30PM 8 Server redundancy
3:47PM 0 timing sources
1:53PM 1 Blended?
1:31PM 1 Dialing timeouts
12:14PM 0 Keeping stable updated with patches
10:15AM 2 Mutiple Homes one asterisk box
9:30AM 5 OT: 3Com 3C10222 POE 24 Port Ethernet
9:20AM 7 Mandriva 2006 Cooker RPM for Asterisk 1.2.9
8:40AM 0 I need help patching source
8:25AM 0 multiple calls
8:05AM 0 loading graphic on a Cisco 7960
8:03AM 1 zaphfc - problem
7:24AM 1 Very bad quality with AVMFritz!cardPCIandchan_capi
7:17AM 0 Sip No Audio Both Side
6:46AM 1 QueuePauseMember(|Agent/) question
6:38AM 0 IAX2 failed to authenticate as priv (DUNDi)
6:37AM 0 Dial command option D(digits)
6:05AM 3 outgoing call problem
5:45AM 2 Unable to configure my DID number
5:41AM 1 Call-limit and internal transfer
5:05AM 0 channel bank log
3:09AM 3 Certain fax types cause problems
2:51AM 1 Which Fax Solution really works on IAX or SIP?
2:50AM 2 Encrypting the Conversation
2:39AM 0 Error on dial_exec_full
1:59AM 0 CDR calls started via AstManProxy
1:31AM 5 AGI tutorials
1:23AM 1 FXS: No ringtone
1:15AM 0 SV: setting up an email to fax with asterisk
12:35AM 7 setting up an email to fax with asterisk
12:07AM 1 spa941 call pickup?
Sunday July 9 2006
11:47PM 1 Urgent Upgrade
11:37PM 7 IVR DTMF
10:35PM 0 PRI Random Disconnected
9:33PM 0 spandsp and app_*fax.c
7:28PM 0 How to transfer other sessions
5:04PM 1 NuFone suggests to use Vonage!!!!
2:07PM 2 Global variables and AGI
11:49AM 6 Choppy MOH (Cisco gateway)
11:17AM 2 2 Handsets, Same extension
7:01AM 2 Can one SIP extension be used for two phones?
6:59AM 1 zap and fax
1:22AM 4 What's the story with X10*P FXO cards?
Saturday July 8 2006
11:56PM 1 Suggesstion Required
11:52PM 0 packet8 dta 310 power supply question
7:04PM 1 Help with router setup on new asterisk box
1:37PM 1 PHP AGI
1:09PM 1 Freeware sip/iax client windows mobile
12:30PM 2 trouble with * and # infront of a phonenumber
10:45AM 1 setting of volume
7:30AM 0 voicemail realtime and MWI
6:59AM 3 Asterisk with ISDN Fritz PCI card
2:59AM 1 CallerID in UK on TalkTalk - different to BT?
1:29AM 2 Outgoing MSNs and chan_misdn
Friday July 7 2006
11:46PM 1 Uninstalling Asterisk? No make uninstall?
7:28PM 0 Play sound mid way through call
6:42PM 1 Disable the flash hook hold capability on a SIP-to-SIP or SIP-to-ZAP call?
5:57PM 1 Asterisk with Analogue cards
3:34PM 1 zaptel errors
12:57PM 2 test tone
12:39PM 6 Fonality vs TrixBox UI
12:12PM 3 prob with debian and chan_zap
11:50AM 0 Re: Feasability of using * for smallappartmentbuilding?
11:44AM 1 Re: Feasability of using * for small appartmentbuilding?
11:39AM 5 [tip]semicolon trouble: System($(sleep 4; cp out)&) not working, but System($( sleep 4 && cp out)&) ; )
11:35AM 4 Voicemails randomly not deleting in ??
11:33AM 1 Asterisk and NFS
10:52AM 0 Re: Feasability of using * for smallappartmentbuilding?
10:50AM 2 Re: Feasability of using * for small appartmentbuilding?
10:36AM 2 Help with MusicOnHold!!!
10:31AM 0 E1 additional calling party number
9:31AM 1 Metermaid phone compatibility
9:20AM 2 ASTCC: inuse flag still hangs!
9:17AM 0 ASTCC: how can I limit to xxx minutes per week?
9:14AM 1 Incoming Call matching to peer
9:11AM 2 New GTK Gui for Monitoring and Administration
8:47AM 1 Asterisk stops accepting calls
8:36AM 6 Feasability of using * for small appartment building?
8:16AM 0 SIP account not available in queue ringall
7:57AM 3 ztmonitor in numeric mode
7:48AM 2 qozap w/
7:33AM 1 OT: Sipura SPA-3000 ATA Directing Calls toAsterisk
7:20AM 0 Best method for detecting state of a sip trunk
7:08AM 4 Do you need a licence to connect a Cisco hardphone to Asterisk ?
7:04AM 1 mgcp trouble
6:38AM 2 Test E1 channel
6:28AM 3 Dell PowerEdge 830
6:15AM 4 IVR - Automatic Attendant database query
5:46AM 0 Multiple issues
5:44AM 3 Problem With Transfering Calls.
4:40AM 3 SV: How to collect Call duration, Dialout Call files?
2:22AM 0 How to collect Call duration, Dialout Call files?
12:11AM 2 Best practices with Asterisk
12:05AM 0 2.6.18 Kernels
Thursday July 6 2006
11:33PM 0 sip.conf, extensions.conf
10:57PM 2 menu system - configurator
9:08PM 0 Please ignore ...
4:46PM 5 Help with MusicOnHold
4:35PM 0 Help troubleshooting "deadlocked" Asterisk
4:19PM 0 Dropped Calls Need Help
3:38PM 2 Tadiran Coral IP PBX to Asterisk
3:27PM 0 fxo lines bridged on a new call once!
3:00PM 2 OT: Sipura SPA-3000 ATA Directing Calls to Asterisk
1:57PM 3 NOT logging Callerid/Call Data?
1:55PM 2 asterisk and sip nat problems
12:57PM 2 Zap Channel not hanging up on Telco side
12:44PM 10 for you guys setting up customer offices...
11:58AM 0 How to plot/graph fxotune -d data
11:53AM 0 xlite softphones: Got SUBSCRIBE for extensions without hint. Please add hint to 1001 in context
10:33AM 2 Phones cutting out.....again - PLEASE HELP!! !
9:58AM 1 audio session start delay
9:57AM 0 Asterisk Home on 64bit?
9:51AM 6 Phones cutting out.....again - PLEASE HELP!!!
8:55AM 3 Cisco SIP Firmware
7:26AM 1 spa941 and sip "bye"
6:38AM 3 Cisco 7941/7961/7971 wont register with asterisk
6:25AM 0 SOLVED: Re: Calling Extensions generates congestion when call answered
6:22AM 0 SOLVED: Re: Extensions dialing but fails on pickup
6:19AM 4 mISDN configuration
6:17AM 0 Using outboundproxy in sip.conf
6:09AM 0 WG: CDR Accounting wrong
6:03AM 3 Invite someone to Conference
4:00AM 2 Sip voip call termination in Nigeria
3:04AM 2 Unable to find good link to configure Polycom 501 with Asterisk (Plz send good link)
3:01AM 0 SIP connections
2:57AM 11 Tired of fax calls... :-/
1:47AM 0 (no subject)
1:46AM 0 SV: B2BUA Webbased and Click 2 dial apps
1:40AM 4 B2BUA Webbased and Click 2 dial apps
1:32AM 1 Rockwell Modem
12:59AM 0 Polycom with Asterisk
12:01AM 1 control during registration process
Wednesday July 5 2006
10:46PM 0 Help! Zap Startup failure: why is libpri not defined ?
10:11PM 0 fax to HP machine
9:01PM 2 Cisco Buddies
8:48PM 0 Echo cancellation doesn't work after inbound calls are transferred to another extension
8:46PM 0 Voicemail Contexts
7:09PM 1 SIP conf
6:25PM 3 buy X100p card in singapore
6:11PM 0 Got Mediatrix 1204 to work! now MWI and Poly com
6:10PM 1 PRI issues with telco access codes
5:31PM 1 sip codec convertion on the fly
4:52PM 3 Any Polycom dealers willing help out?
4:14PM 0 tormenta2 drivers
3:54PM 0 Weird transcoding error (SIP, local channels): sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 64/256)
3:27PM 1 Got Mediatrix 1204 to work! now MWI and Polycom
2:12PM 0 New mailing list: asterisk-speech-rec
2:04PM 0 Zaptel For new TE412P
1:29PM 0 test to see if I can get any message through
1:14PM 2 Possible Bug?
1:03PM 1 Looking for an asterisk guru
1:00PM 7 Asterisk in Seattle
12:57PM 0 strange Fax or modem like tone when tdm400 answers pstn
12:40PM 0 is ooh323 RAS/ASN.1 broken?
12:15PM 1 Caller Prompts in a Queue??
11:42AM 0 delay and jitter issues..
10:47AM 4 'sip debug'
10:05AM 2 [Asteirsk-Users]TE110P configuration problem
10:02AM 1 Cisco 7960 Softkey templates
9:55AM 1 CFWD Status with PHP
9:34AM 0 Sangoma A200 and hangup detection with Asterisk.
9:08AM 1 Agent penality for dynamic agents
9:05AM 1 Asterisk UAc / Request-URI
8:41AM 0 Performance of Database Storage Vs Clustered File System
8:35AM 2 Troubleshooting Random PRI disconnects
8:22AM 0 AGI: Channel status
8:11AM 0 meetme issue with high cpu usage and "hung" conference rooms
7:44AM 1 Queues and script
7:39AM 0 DEBUG[13314]: Didn't get a frame from channel: SIP/
7:11AM 2 International Dialing setup in extensions.conf
7:07AM 5 intel vs amd motherboards
6:58AM 0 Extensions dialing but fails on pickup
6:26AM 0 zaptel Disabled echo canceller because of tone (rx) on channel 2 work?
5:24AM 0 ZAP channel for outbound calls.
4:56AM 0 g729.1 + g723.1 codec conversion
4:46AM 1 Bug in chan_sip mysql support and canreinvite?
4:00AM 1 SV: SV: Nokia E61
3:45AM 0 0000491...
3:18AM 3 Skype gateway
2:29AM 0 Hanging SIP Channels
2:18AM 2 Intel E7220 chipset?
2:12AM 2 SV: HP Proliant server?
1:48AM 7 HP Proliant server?
1:39AM 0 Dynamic realtime with MWI working
12:34AM 4 SV: Nokia E61
12:05AM 0 Bridging Prob:::I guess
Tuesday July 4 2006
11:57PM 0 Asterisk Shutdown !!!
8:53PM 1 tdm04b strange noise when answering calls
8:10PM 3 RE: Is there a search feature?
7:18PM 2 H.264 and Asterik?
4:51PM 2 More g729 calls than licenses?
4:13PM 0 Sample PRI and FXS channel bank zap files for zaptel and asterisk.
2:21PM 1 Sangoma A200 woes
2:03PM 0 vserver with no /dev/tty* how to run "asterisk-c"for a colored CLI?
1:40PM 1 Page() command and file playback
1:34PM 0 vserver with no /dev/tty* how to run "asterisk -c"for a colored CLI?
1:10PM 2 vserver with no /dev/tty* how to run "asterisk -c" for a colored CLI?
12:32PM 7 MediatrixclientauthenticationfailedEFAILURE_REASON_AUTHENTICATION
12:09PM 0 Mediatrix clientauthenticationfailedEFAILURE_REASON_AUTHENTICATION
11:53AM 0 Mediatrix client authenticationfailedEFAILURE_REASON_AUTHENTICATION
11:27AM 0 Mediatrix client authentication failedEFAILURE_REASON_AUTHENTICATION
11:25AM 0 please remove the autoresponder
11:12AM 0 Mediatrix client authentication failed EFAILURE_REASON_AUTHENTICATION
10:39AM 11 SOLVED: IAX jitter / clocking problem
10:36AM 11 voip-magazinearticle"UsingDUNDiwithaClusterofAsteriskServers"
10:22AM 1 voip-magazinearticle"UsingDUNDiwithaClusterofAsterisk Servers"
10:19AM 0 voip-magazine article"UsingDUNDiwithaClusterofAsterisk Servers"
10:16AM 0 voip-magazine article "UsingDUNDiwithaClusterofAsterisk Servers"
10:13AM 1 voip-magazine article "Using DUNDiwithaClusterofAsterisk Servers"
10:07AM 0 voip-magazine article "Using DUNDiwithaClusterof Asterisk Servers"
10:03AM 0 voip-magazine article "Using DUNDi withaClusterof Asterisk Servers"
9:59AM 0 voip-magazine article "Using DUNDi with aClusterof Asterisk Servers"
9:55AM 0 voip-magazine article "Using DUNDi with aCluster of Asterisk Servers"
8:26AM 0 Recommendations for best Voicemail application manager?
8:10AM 2 vserver (Debian) - no tty: howto use /usr/sbin/safe_asterisk with "-c" for color CLI?
8:01AM 0 I am looking for a (graphical) statistic program
7:41AM 0 SV: SV: Running 40 active calls (too much för CPU?)
7:36AM 0 how to send flash command from asterisk to old pbx when pressing button on phone
7:35AM 0 SIP <--> H323 RTP Questions (1 WAY Audio only)
7:11AM 1 H323 Asterisk best practices
7:06AM 3 Zaptel 1.2.6 / Upgrade Problem
6:52AM 0 Quintum A400 Configuration
6:31AM 2 Libpri + Zaptel + Asterisk polycom_acd_functions error message
6:30AM 2 Help getting International Dialing setup in extensions.conf
6:05AM 0 Quintum A400 Call Establishment Prob
5:54AM 9 time variable
5:43AM 4 Need help with config-files
4:09AM 14 Does asterisk support outbound fax?
3:49AM 1 Calling Extensions generates congestion when call answered
1:40AM 3 trixbox 1.1 download
1:36AM 1 AW: Putting a call recording into a mailbox
1:17AM 1 Putting a call recording into a mailbox
12:52AM 0 FW: SRTP
12:50AM 0 Qsig-Link * to Meridian 81c
12:49AM 3 SV: Running 40 active calls (too much för CPU?)
12:41AM 1 Running 40 active calls (too much för CPU?)
Monday July 3 2006
8:57PM 0 Howto: Gentoo + Hudlite + Scratch Asterisk Install
5:11PM 1 Nokia E61
1:53PM 1 Trouble Setting Up International Dialing in extensions.conf
11:04AM 1 The Asterisk console on a Dell D820 with Intel High Definition Audio.
8:50AM 1 SV: SV: SV: How to configure NOKIA N70 with Asterisk?
8:12AM 2 TDM Installation error
7:59AM 0 PacketCable and Asterisk
7:14AM 1 can't dial Scotland ...
7:11AM 3 Polycom Soundpoint IP 301 w/ MGCP
7:01AM 0 file.c: Unexpected control subclass '14'
6:25AM 5 flash button on asterisk + legacy pbx system
6:22AM 9 SRTP
6:09AM 1 callwaiting
4:41AM 2 Aastra phones - disable call waiting
3:56AM 2 Queues and annoucements
3:48AM 2 Help with IVR menu.
1:12AM 1 Call waiting using free PBX
12:57AM 1 SV: SV: How to configure NOKIA N70 with Asterisk?
12:44AM 1 Duration for billing
12:30AM 2 SV: How to configure NOKIA N70 with Asterisk?
Sunday July 2 2006
11:28PM 1 performance & reliabulity of asterisk voicemail using odbc storage
11:24PM 0 How to configure NOKIA N70 with Asterisk?
8:55PM 1 SIP debug logging
8:41PM 0 What does it mean?
6:50PM 1 Motorola and Asterisk
6:36PM 1 Latest SVN of asterisk-addons doesn't compile
12:42PM 2 how to ask for number to dial and then dial it?
12:12PM 0 H323 to SIP Gateway
11:52AM 0 to.gsm and the.gsm
11:37AM 0 setting cdr userfield in .call file
9:59AM 3 dtmfmode=inband but SDP also indicates rfc2833
8:44AM 1 channel shows to be in use
8:24AM 2 How to continue after a match in an include
Saturday July 1 2006
9:21PM 0 ooh323 svn updated
1:48PM 0 Cant seem to send cidname to snom 320
12:45PM 1 can't run "cat $filename" inside scripts with system()
8:33AM 3 Nufone Tollfree Port
5:00AM 1 svn trunk and call hold / transfers
3:22AM 1 IVR menus on different DIDs
2:02AM 1 callwaiting in queues
1:33AM 0 Asterisk and HiSax