asterisk users - Jul 2005

Sunday July 31 2005
TimeRepliesSubject
11:59PM 1 binding asterisk-h323 on two interfaces
11:47PM 0 [OT] MyOSS(Open Source) Magazine - Edition 4 Available Now!
9:46PM 0 Asterisk fax problems with spandsp
7:45PM 0 A configuration question
7:31PM 0 Voicemail envelope time is 4 hours ahead(?)
6:59PM 1 Questions on Asterisk and CallerID
6:48PM 4 Gmail and the list
3:07PM 0 Cisco 7960G firmware backup
2:13PM 0 Sipura 841 vs Grandstream GXP2000
1:02PM 0 Mailing list down?
11:49AM 0 Sound Card help
9:13AM 0 Sipura support down the tubes
6:05AM 0 Zaptel and SUSE - the real story
5:27AM 0 BETA Testers wanted
5:00AM 0 Building zaptel.1.0.9 on Suse 9.3
5:00AM 0 Zaptel Compile related question
2:10AM 0 Searching for suitable Distro for Asterisk
 
Saturday July 30 2005
TimeRepliesSubject
8:05PM 0 How to create a secret code to use my IAX server's long distance plan from a public phone for instance
7:02PM 0 # to initiate transfer on outbound call
6:51PM 1 Record() permission problem
2:29PM 0 Using AGI, how do you clear a variable?
1:45PM 0 Asterisk Compilation Failure
12:53PM 0 Windows Client for spandsp txfax
11:38AM 0 Broadvoice- 404 not Found
10:25AM 0 Alcatel phones
7:18AM 0 OT Skype almost being sold
2:48AM 0 Register Every User ?
 
Friday July 29 2005
TimeRepliesSubject
11:40PM 0 X100P/Caller ID: clidtest works, * complains
11:23PM 0 chan_sccp 20050730 release
10:48PM 0 mailing list vanished
10:40PM 0 Simbion or palm os sip clients
10:04PM 0 Free or cheap PSTN to IAX gateway
5:47PM 0 IAXy Caller ID Issues
3:22PM 0 Sipura 841 vs Grandstream GXP 2000
3:11PM 0 Realtime and Voicemail, not updating passwords?
2:18PM 0 ReInvite X Broadvoice
1:28PM 0 IAX Huge Delays after Hold or Transfer
1:27PM 0 asterisk knows best? softphones
12:37PM 1 Asterisk 1.0.9 and PostreSQL DB
12:18PM 0 new TDM04B
12:02PM 0 Feature requests :: For now, not in the bug tracker - please!
11:04AM 0 IAX Music on Hold Classes
10:11AM 0 SIP calls no longer hangup [1.0.8]
10:07AM 0 Music On Hold pain - suggestions?
9:23AM 0 YAACID V0.95 Released
9:13AM 0 OT: Asterisk and ISDN cards = ISDN simulator for Cisco lab?
9:11AM 1 Can Asterisk & Shoretel systems talk to each other?
9:07AM 0 Does asterisk support call hunting?
9:07AM 0 Cluecon Next Week!
8:54AM 0 CVS-Head from Wednesday - libpri errors in Chan_zap
8:42AM 0 OT: Lucent, Cisco, DS3 and SS7
5:47AM 2 New digium TE406 & 411
5:17AM 0 Snom 360 not dial with direct buttons
5:17AM 2 SIP phone procedural question
4:42AM 0 Auto loading of qozap module
4:26AM 0 problem calling SIP accounts
1:10AM 1 FastAGI problems
12:28AM 0 25 second delay, then busy...?
12:18AM 0 How to change default music on hold class
 
Thursday July 28 2005
TimeRepliesSubject
11:51PM 0 H323 problem
10:51PM 0 PC requirement
10:47PM 1 Potential reboot problem with Polycom IP600 phones
9:44PM 0 oh323 compile problem
9:25PM 1 IP-ID in RTP/UDP/IP packets
8:53PM 0 Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 10.10.10.123
8:37PM 0 How to compare VOIP providers
8:14PM 0 Braodvoice Line Busy
7:31PM 0 Asterisk@home and OpenLine4 issue
7:12PM 0 grandstream budget tone 100 ip-phone just one call
6:14PM 0 IP Phone Advice ??
6:14PM 0 CallerID Advice ??
5:46PM 0 Messaging - Asterisk presence
4:00PM 0 Need suggestions on solution for central Asterisk server and multiple private networks.
2:29PM 2 SIP Debug
2:20PM 0 Sound Cards, ALSA, and Asterisk
2:19PM 0 How do you dial an alternate line on busy with several multi-line phones?
1:48PM 0 Zaptel files for New Zealand
1:31PM 0 SIP and consultative transfer
12:33PM 0 List extension in directory without mailbox?
11:14AM 1 Problem with BT100 behind iptables firewall
11:07AM 1 Querying Nagios users...
10:34AM 2 [Asterisk-Dev] Digium to Sponsor a Pizza party at Cluecon
10:23AM 3 Cisco Call manager
10:21AM 5 Nat Transversal
9:55AM 0 Unicall Dialing problems
9:47AM 10 most stable linux to build business
9:43AM 2 How to adjust codec voice detection? Changin RxGain does not help me...
8:48AM 4 TAPI Interface
8:44AM 6 delay on pri dialling when asterisk is spliced between E1-Pri and legacy pbx
8:24AM 1 A problem with queues
7:53AM 0 How to: recompile asterisk 1.0.7 beta 3
7:37AM 15 help Windows messenger configuaration
7:21AM 4 What wrong with asteriskathome.org
7:07AM 15 SIP WEB Phone (Wanna implement Click to Call)
7:03AM 0 Zaptel rpm spec file with udev support
6:59AM 1 Suggested System Specs - 20 ext, 8 Incoming Lines
6:58AM 1 probing a SIP device for redirection information?
6:47AM 0 Call Status from a IAX trunk to a Zaptel trunk
5:48AM 3 Asterisk fails to start
5:31AM 7 strange dial problem with polycom 501
5:07AM 23 Can you caculate with me?
4:03AM 4 Public phone
3:48AM 1 Monitor IAX
3:47AM 0 New feature in V1.2: attended call transfer
3:22AM 2 different _source_ addresses for registrations?
2:55AM 0 Wrong cdr records
2:42AM 0 Asterisk version 1.2 :: What's new?
2:30AM 26 dialplan defenition
2:17AM 0 pre/post-paid billing system
1:19AM 2 how to loop E400P card to test ?Any help will be appreciated.
1:05AM 1 realtime: sip show users/peers
12:52AM 1 help on linux version
12:41AM 4 CDR disposition field always says ANSWERED on inbound calls
12:12AM 1 how to configure E400P card?
 
Wednesday July 27 2005
TimeRepliesSubject
10:53PM 0 announce to caller in queues (asterisk for art!)
10:13PM 1 call forwarding without answer
9:03PM 50 Full T38 sip Faxing now Available
8:48PM 0 [PLEASE RESPOND] Supervised transfer over SIP to outside POTS lines
8:42PM 0 Who is using asterisk on large scale consumer sites in london?
7:40PM 1 RE: Asterisk fax problems with SPANDSP
7:05PM 5 Cisco 7940 - Disappearing Clock - SOLVED
6:50PM 2 Cisco 7940 - Disappearing Clock
4:20PM 7 Get older CVS version
3:43PM 0 erros while updating the latest CVS
3:38PM 3 oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6
3:33PM 12 Snom 360 record button?
1:57PM 2 Toshiba Integration - MWI Light
1:19PM 0 Queue Statistics on 1.0.7
1:04PM 1 Question about Nextone softswitch
12:53PM 1 Motorola A910 WiFi + GSM phone
12:39PM 3 "Received packet with bad UDP checksum" - whats the real problem?
12:36PM 0 IPSwitchBoard Updated
12:26PM 1 No Extensions
12:14PM 4 Read Back Caller ID Using Number Announcement in Digital Receptionist
11:48AM 0 Latest CVS HEAD and the new wct4xxp card
11:31AM 2 Zaptel Problems with 1.0.9
11:28AM 1 Install failed on Asteriskathome
11:26AM 1 call failed: 499 Not acceptable here
10:33AM 3 SIP ATA's as "house" phones
10:28AM 0 Polycom gain settings
10:09AM 0 Erro
10:08AM 2 CVS Head No ringing on calling end?
9:51AM 0 Sending DTMF Tones Offhook
9:24AM 1 Recording suddenly stopped
9:19AM 0 voicemail ODBC storage question
9:07AM 2 Dial through IAX to FWD
8:51AM 2 Random Behavior on Trunk Lines with TDM Card
8:34AM 0 Playtones not passing sound to incoming SIP connection
8:22AM 5 does not implement 'PUBLISH'
7:40AM 1 H323 Configuration file
7:29AM 8 Music on Hold: CPU Intensive Monster
6:23AM 6 Attended transfer not working (atxfer)
6:20AM 0 Call rejected/No application dial
6:17AM 6 Klicking sounds in background
6:13AM 0 Fax detection on isdn
6:08AM 0 R: Sound Quality Problems
6:01AM 2 R: Zaptel error: Unable to create channel op type'Zap'
5:50AM 0 Agent penalties and busy status
5:42AM 0 Port Restricted CONE NAT Error
5:30AM 1 Zaptel error: Unable to create channel op type'Zap'
3:51AM 0 I found problem with TE110P and the new kernel offedora, "kernel panic"
3:38AM 2 ISDN & ASTERISK Cabling...
2:50AM 2 Zaptel error: Unable to create channel op type 'Zap'
1:43AM 1 Why is sip saying NO NAT
1:34AM 0 I found problem with TE110P and the new kernel of fedora, "kernel panic"
1:26AM 0 R: TE110P Cable Pin Out
1:26AM 0 Does Asterisk need to know where the call is comming from?
1:12AM 2 Regarding Call Hold
12:44AM 11 cdr_mysql does not write to mysql db
12:29AM 0 echo capi AVM fritz card
 
Tuesday July 26 2005
TimeRepliesSubject
11:29PM 0 re: switch statement in dialplan
11:08PM 2 Supervised transfer over SIP to outside POTS lines
8:01PM 1 What does pbx-wilcalu.so do and why does it keep crashing my * box?
7:42PM 1 Are busy and congestion behaving differently than documented?
7:21PM 10 TE110P Cable Pin Out
7:14PM 1 Real-time for H.323?
6:27PM 3 Melting TDM card
4:41PM 0 Upgrade to *@H 1.3 "Problems with Background files"
4:23PM 1 Good day everyone, i need firmware for the ATA186.
4:12PM 1 cannot find channel_pvt.h
4:02PM 3 ASTCC: different incriments
3:43PM 1 problems with compiling asterisk-oh323-0.6.5
3:24PM 6 mpg123 - two processes
3:21PM 0 Channel restarted en E1 Card
3:01PM 0 E-911
2:53PM 1 Registration failed problems/Polycom 500/maybe nat problem?
2:39PM 0 What are SIP proxies and H323 Gatekeepers
1:56PM 1 Generate ring while waiting for SIP connection to initiate
1:30PM 1 TO: M.G. Ref: Dial using URI(web) or using FORM(web)
11:38AM 5 Polycom digitmap question
10:49AM 2 Automatic setup of calls between two external lines
10:41AM 0 If voice volume level too low, it is been cut
10:39AM 2 [Asterisk-Dev] CRITICAL PATCH for anyone using the L option in dial.
10:30AM 4 [Asterisk-Dev] CVS HEAD behavior change: Beware!
10:23AM 2 sip+oh323 - no voice at sip side
10:12AM 0 Sound Quality Problems
9:59AM 0 Load Balancing with SER
9:48AM 3 Dial using URI(web) or using FORM(web)
9:33AM 1 Polycom 600 Presence indications - ALWAYS OFF-HOOK?
9:31AM 3 7960 from SIP to SKINNY
9:23AM 0 SIP INVITE and caller id / proxy-authorization strange behaviour
9:15AM 8 [Asterisk-Dev] Asterisk 1.2 Release Plans
9:10AM 1 existing ISDN PBX <-> asterisk <-> 2xBRI for IVR and SIP
8:26AM 4 qozap junghanns errors
8:00AM 0 Polycom 501 indicated -1 Urgent and 1 new for new voice mail
7:33AM 2 Perl AGI
7:28AM 12 7960 SIP Firmware Upgrade Strange Problem
7:08AM 0 Call quality degradation after time
7:02AM 0 queue members with multiple devices (bug 4759)
6:30AM 4 Any experience with Sixtel--tollfreedirect--iax.cc?
6:17AM 0 AGI why oh why?
6:15AM 3 Billing works but i do no get full calling cost...!
6:07AM 0 Problem with SIP
6:04AM 0 RE: VM on * for CME Install - Solved
5:06AM 7 Stumped on vMail problem, any ideas?
4:29AM 1 how to compile asterisk-oh323
4:15AM 1 Eyebeam Video+Nat
3:32AM 0 ABI manager - redirect
2:27AM 0 how to config E400P card?
1:26AM 0 CLI messages that are hard to understand
1:19AM 0 include not working in bristuffed Asterisk 1.0.7 extensions.conf
1:01AM 6 function declaration isn't a prototype
12:43AM 0 Re: Asterisk-Users Digest, Vol 12, Issue 144
12:20AM 7 Method not allowed error
 
Monday July 25 2005
TimeRepliesSubject
11:55PM 7 Some more VOICEMAILMAIN issue...
11:12PM 0 Latest batch of CVS changes
11:11PM 3 why zap call transfer fails?
10:25PM 0 SJ Phone NAT/Firewall Blocked
10:07PM 0 Multiple language problem
9:57PM 0 Keys ???
9:41PM 1 Proper Jitter Buffer Settings?
8:26PM 6 To anyone seeking 911 Service Providers "stay away!!!"
8:05PM 0 Jyran Glucky is out of the office.
7:28PM 1 CAPI Eicon Server bri, extreme noise or gain
7:26PM 0 serrctl add : HA1 calculation failed error
7:01PM 0 Fw: /bin/sh: build_tools/make_version_h: not found
6:32PM 6 How can I use MySQL in the dialplan?
5:02PM 0 No Audio with T100P Enabled
4:51PM 3 DISA disconnects
3:47PM 0 ClueCon Giving Away Voice Hardware (even more than before)
3:30PM 0 [Asterisk-Dev] We are giving away 3 A101 single-port T1 cardsduring Cluecon!
3:07PM 4 chan_sccp release 20050725
2:38PM 0 Grandstream 488 - VoIP-to-PSTN Calls
2:22PM 0 MWI on Siemens Hicom switches
2:18PM 1 911 Service Providers
2:03PM 4 Voicemail and musiconhold sound stopped working
1:53PM 2 problems with compiling asterisk-oh323
12:54PM 0 [Asterisk-Dev] We are giving away 3 A101 single-port T1 cards during Cluecon!
12:36PM 0 RE: Voicemail Send Message (Options 3, 5) Patch
12:30PM 10 A TDM issue..
12:13PM 0 slightly OT: firefly won't hang up!
12:05PM 2 cisco 7920 makes 7940 reboot
11:55AM 1 exten => fax in [macro-blah]
11:29AM 1 TE410P (2nd gen) red alarm
11:23AM 1 Nufone inbound
11:04AM 3 MozIAX phone on FC4/Firefox 1.6
10:48AM 2 Re: Asterisk-Users Digest, Vol 12, Issue 171
10:44AM 9 [Asterisk-Dev] Zaptel update, Asterisk 1.2 janitor projects
10:19AM 3 re: realtime caller id extensions matching
10:01AM 0 Problem - jittery.
9:41AM 0 realtime caller id extension matching
9:34AM 1 Playing sounds while dialling
9:21AM 6 Fritz PCI card in ptp mode with chan_misdn
8:37AM 0 RE: Asterisk-Users Digest, Vol 12, Issue 171
8:29AM 7 [Asterisk-Dev] Cluecon - Who's going ?
8:21AM 0 chan_agent / manager API / SIP - possible bug?
8:11AM 10 Should this work?
8:05AM 0 Hangups transferring call from Intertel system
7:55AM 4 US CallerID and TDM04B
7:42AM 0 SER & Asterisk & SIP =513 "Message Too Big"
7:38AM 1 Voicemail: could not stop recording
7:31AM 1 Re: Marco and Realtime Extension Problem [SOLVED]
7:21AM 0 CDR Accounting/Billing Advise
6:47AM 4 100% CPU with Unicall and * head
6:44AM 4 Zap channel configuration problem
6:15AM 11 Polycom IP600 - Flashing clock and date?
5:55AM 0 variables from before call entered queue
5:42AM 2 Operating AAH v1.1
5:07AM 12 Asterisk Configuration
5:07AM 1 asterisk + i4l problems
4:30AM 2 network problem -- echo
4:25AM 0 Which mix of VOIP services do we need?
3:28AM 3 Wengo config and G729(a)
3:15AM 1 Voicemail : Unable to create lock file: No such file or directory
3:02AM 1 Meetme and option c for announcing user count
2:10AM 2 "Cannot native bridge" on licensed G729
2:06AM 0 Outgoing SIP to SER causes LOOP BACK message
1:42AM 1 sendDTMF at pickup
1:42AM 4 Polycom 600 one-touch message access?
12:48AM 3 VoiceMailMain issue..
12:09AM 1 does h323 exists in astcc trunks
 
Sunday July 24 2005
TimeRepliesSubject
10:03PM 1 Caller ID, Called ID and Forwarded ID
8:54PM 1 Unlimited land line calls in Australia.
8:51PM 27 super high bandwidth codec
8:25PM 1 asterisk with ser project, , , , here we go! ready or not!!!
8:06PM 3 Polycom 600 Ring-Answer (but not ring!)
7:55PM 1 HFC-S cards in Australia
6:38PM 1 Disconnecting a call on asterisk
4:30PM 9 Busy Lamp Field SIP Phone
4:02PM 12 Need to ztcfg every time I reboot *
3:31PM 0 [Asterisk-Dev] sip messaging (tested on eyeBeam) support
3:30PM 1 Incoming call prob
1:24PM 0 FS: Zhone Channel Bank
12:54PM 12 TNT and SIP problem
12:34PM 8 Help with Asterisk@home and Broadvoice incoming calls..
12:20PM 14 DID + 800 Providers
11:27AM 5 success story: TE406P (quadspan with hardware echocan)
10:51AM 14 Why can't sip/200 call sip/202
9:07AM 3 Zap PRI load testing
9:04AM 0 E&M wink start patch
8:36AM 1 Do I have to worry about interrupt sharing here?
7:41AM 0 ASTERISK-ITA mailing list is back
3:39AM 0 does astcc support h323
 
Saturday July 23 2005
TimeRepliesSubject
8:11PM 0 dead pg connection with voicemail
7:27PM 0 callgen323 & ohphone!!
12:48PM 3 IAX phone not hear the other phone ring when calling
11:45AM 0 Question about the latest CVS and Zaptel
11:13AM 1 Analog extensions behind E1, how to create them?
10:45AM 1 Outgoing SIP Problems with Asterisk and SER on same PC
9:30AM 1 Need to start from somewhere
8:13AM 0 spa-2100 3.2.1 firmware
7:29AM 1 astcc timestamps
5:52AM 7 Asterisk 1.2 is getting closer - please help
4:07AM 0 Running Asterisk on a Dell PowerEdge 2850 ServerRe: Dell Hardware
3:28AM 6 ASTCC gives me only the time, but no cost
2:46AM 2 Asterisk users from Turkey?
1:51AM 2 (cause 66 - Channel not implemented) -- IAX?
 
Friday July 22 2005
TimeRepliesSubject
11:51PM 0 announce hold time issues
11:34PM 1 Running Asterisk on a Dell PowerEdge 2850 Server Re: Dell Hardware
10:20PM 6 XML or Push Info
8:29PM 1 CVS-HEAD v release 1.0.9
7:08PM 0 IAX2 attempts native bridge when notransfer=yes
5:26PM 2 Modules fail to load after kernel update
4:49PM 2 CVS-HEAD dies signal 11 after incorrect vm password
4:35PM 1 Voicemail passwords located in #include file
4:32PM 1 Chan_capi MSN problem
3:58PM 0 unknown rtp codec
3:39PM 2 X100P not answering
3:33PM 0 Outgoing SIP causes error Got SIP response 482 "Loop Detected&#9; " back from.....
3:03PM 2 Caller logging in to call out IAX line?
2:17PM 4 Opteron Hardware with Asterisk
2:13PM 0 Oytbound Proxy Support in Asterisk
1:09PM 2 Lost in AAH Setup
12:36PM 0 Please help!! ASTCC logs only the first record !! What is wrong?
12:36PM 0 No caller ID, straight to voicemail
11:24AM 0 extension matching using includes...errornous results
10:57AM 0 T1 signalling on Bahamas
10:06AM 0 How to set the SMSC sender = VoIP provider 10-digit #
9:47AM 1 *@Home: SIP for testing?
9:05AM 2 web managment
8:58AM 5 Need Advice
8:50AM 0 no active channel but one active call???
8:24AM 0 WAS: Stupid hold music NOW: list gripes
8:13AM 1 zaphfc: sync lost, pci performance too low. you might have some cpu throtteling enabled.
7:24AM 1 Form IAXY to DIAX: No sound.
7:15AM 4 Asterisk and Norstar MICS
7:00AM 1 Interconnect with Mitel PBX
6:48AM 28 Dell Hardware
6:20AM 0 R: Quadbri trouble
6:19AM 2 YAACID - 0.91 new release
5:50AM 1 SIP extension auto busy's itself
5:38AM 2 --- Problem with queues.conf and extensions.conf ---
4:40AM 1 low profile FXO card
4:37AM 0 capi or mISDN for passive Fritz!Card PCi
3:55AM 0 all zap channels get RING signal when starting *
3:48AM 0 Marco and Realtime Extension Problem
3:47AM 0 unable to disconnect a bridged channel
3:43AM 2 Asterisk operator functions
3:17AM 1 asterisk captures sound device
2:47AM 1 SATA
2:00AM 0 GXP2000 and Headsets, Call Center phones.
1:47AM 2 Problem with Zaptel FXO..
12:31AM 1 Re: zaptel make problems
12:07AM 0 Xorcom Rapid 1.1
12:02AM 0 No D-channel available
 
Thursday July 21 2005
TimeRepliesSubject
11:17PM 1 queues and roundrobin/rrmemory
10:59PM 1 Re: zaptel make problems
10:57PM 0 Re: zaptel make problems
9:36PM 1 Re: Asterisk-Users Digest, Vol 12, Issue 143
9:08PM 1 IAXY & Voicemailmain problem
8:58PM 20 Stupid hold music
7:16PM 1 SOLVED: TE410P card in an HP-Compaq DL380 G4 server
7:04PM 0 COVAD voipr movie clip - A MUST SEE
5:55PM 0 chan_capi or chan_mISDN with passive Frtiz!Card
4:43PM 0 YAACID update
3:26PM 3 DNS SRV supported phones
3:18PM 2 cat 5 'joiner'? (polycom 500 problem)
2:07PM 4 caller id on a T1 PRI
2:02PM 6 [Asterisk-Dev] ClueCon in 2 Weeks!
2:00PM 0 re: DTMF woes, continued
1:16PM 4 Semi-Ot - Cisco IP Phone Password Reset Procedure
12:58PM 20 IAX over HTTP
12:41PM 0 Busy Extensions
12:22PM 4 Busy Extensions.
12:14PM 1 OT: Potential reasonable solution to the 911 problem, integrate t o Asterisk?
12:13PM 0 Can't hear auto-attendant
11:54AM 5 initiate call with asterisk
11:54AM 2 Question on VoipJet
11:29AM 4 picking a cvs-head version
10:59AM 3 Queue agent wrap up time.. .any ideas?
10:49AM 1 Disable Console Audio
10:31AM 1 account code missing in csv cdr
10:30AM 3 Routing by DID
10:18AM 0 chan_sccp new release 20050721
10:04AM 0 Iaxy call waiting problems
10:01AM 2 Looking for Thai DIDs
9:48AM 0 New features for e164.org
9:15AM 0 error while writing audio data: : Broken pipe ... segmentation fault
9:04AM 1 HOW TO RECEIVE FAX
8:10AM 13 Did anyone else get spammed by GIZMO?
8:08AM 0 Call queue advice
7:49AM 1 Asterisk and IP500 / IP600 Boot RoM
7:35AM 0 MeetMe Enter & Exit Sounds
7:30AM 8 Queues and timeouts
7:23AM 0 Asterisk, tdm card and BT line:
6:50AM 0 Queues Messages not Playing
6:41AM 0 Dropping call
6:24AM 0 kphone & Asterisk CVS HEAD: no audio
6:09AM 1 Queue issues: timeout and leastrecent strategy
5:42AM 5 attended transfert
4:47AM 0 Anyone have experience with Asterisk under Solaris 10 X86?
4:20AM 0 hwo can i manage TDM04B incoming calls
4:19AM 12 Problems installing asterisk-addons
3:41AM 7 a ne pas voir
2:33AM 3 Thailand DIDs
2:28AM 0 DIDs in Thailand
1:49AM 4 zaptel make problems (long)
1:08AM 3 SIP & messengers & video phones
12:52AM 0 DTMF with Asterisk as SIP client
 
Wednesday July 20 2005
TimeRepliesSubject
10:42PM 0 How to use Audiocodes MP-108 with Asterisk in Singapore
10:41PM 2 Play Dialtone - get digits
9:41PM 2 Last two digits getting cut off?
8:05PM 1 /dev/zap/channel missing?
6:12PM 2 SIP phone failover using DNS SRV?
5:28PM 2 iconnecthere
3:55PM 3 New voiceovers for Allison Smith: submit today
3:45PM 5 OT: Hottie ?!?
3:09PM 2 Asterisk and MRTG
1:39PM 1 Enter numeric value to use as a parameter
1:05PM 0 Freshtel.net - Spamming?
12:33PM 0 g729 codec for Windows Media Player?
12:24PM 2 Force SIP peers to Re-Autheticate
12:05PM 4 Anyone have success with BRI in Italy?
12:00PM 0 musiconhold in sip.conf
11:53AM 0 SetVar(PEER_IP=x.x.x.x) - after Dial PEER_IP is not available.
11:47AM 3 Test CVS HEAD Voicemail ODBC Storage
11:47AM 16 Alternatives to Digium 729
11:13AM 4 HOWTO capture digits
10:41AM 1 "That is not a valid conference number.." with ztdummy running
10:34AM 10 Scottsdale Arizona DID
10:31AM 3 Fedora Core 3 + AVM Fritz ?
10:18AM 13 T1 - incomplete calls
10:15AM 0 Cisco Call Manager with Voicemail on Asterisk Problem
9:54AM 0 FXO hangup delay...
9:36AM 1 Announcement: YAACID (Caller ID for Asterisk)
9:06AM 5 Grandstream GXP2000 resetting all the time
9:00AM 3 Firefly 3rd party - it hangs on "Initialising" and exits with error
8:56AM 1 Problem with CDR web page
8:51AM 1 Anybody has one SIP minimal configuration and one working Softphone?
8:47AM 2 Free Music
8:32AM 1 can asterisk send Remote-Party-ID header ???
8:24AM 1 ceptral (swift)
8:12AM 3 AstLinux creator to speak at Cluecon
8:04AM 1 where i put the astcc config? In the extensions.conf or in the astcc-exten.conf?
7:45AM 0 Re: The issue of negative timestamp is Fixed
7:29AM 0 Sipura 3000 x special dialling pattern (pin code)
7:23AM 13 Extension Lights Patch
7:07AM 1 getting problem in Picking up the parked call
6:57AM 1 Unattended Agent Login
6:56AM 1 Agent Penalty
6:56AM 18 Mahler's Book - New Project
6:49AM 7 Asterisk and flash disks
6:03AM 0 protocol application invalid cisco 7940g
5:39AM 0 IPivr
5:10AM 6 How to send Fax from Asterisk
4:22AM 3 Problem while configuring two TDM400P cards
3:39AM 13 GSM gateway hardware
3:35AM 1 Asterisk Real Time (Users/Peers)
3:03AM 10 /bin/sh: build_tools/make_version_h: not found
2:59AM 0 Polycom phone echo question
2:31AM 15 Junghanns quadBRI on Dell PowerEdge
2:21AM 9 ATXFER discussion, what's your opinion ?
2:04AM 3 [Asterisk-Dev] Memory Leak in Stable?
1:49AM 7 Working with an ongoing call
12:24AM 4 IAXY with DNS name, not IP
12:10AM 3 Zap channel(s), meetme and codecs/licences
 
Tuesday July 19 2005
TimeRepliesSubject
8:59PM 0 test message - for checking
7:51PM 0 When Incoming Caller-ID is Blank Dialparties.agi is shoving incoming IP Address into it.
5:47PM 0 Cisco Video Confernce and Asterisk Video Conference
5:14PM 2 Free Music for MOH from Digium?
4:56PM 0 Adding a Button Template to a CISCO 12+SP
4:27PM 0 AutoAnswer .. not to useful ?
3:24PM 3 Linksys PAP2-NA failures...
2:54PM 0 How to minimize the VoiceMail interface options?
2:52PM 1 two extensions for same phone
2:51PM 3 Call Problems W/CVS Head
2:34PM 1 NoOp does not seem to be printing messages on the console...
2:31PM 0 How to play voicemail greetings?
2:29PM 2 Installation
1:51PM 0 Timing out issue whenusing AGI
1:24PM 5 new to Asterisk, is it possible to call two external lines and connect them using two channels
1:20PM 2 cisco 7970 sccp
12:35PM 5 Asterisk Quit Registering with Broadvoice
12:01PM 7 Asterisk bounty: email TTS
11:59AM 1 Re: So you all think VoIP sypply is warm andfuzzy
11:49AM 0 "Looking for username in default"
11:21AM 0 Using AMP on remote machine with concurrent Manager API connection to Asterisk.Ideas, comments?
11:00AM 1 Information setting up asterisk with an ISDN NT
10:38AM 31 Best VoIP provider
10:09AM 2 spandsp - fax is just blank pages
9:22AM 1 Why so many attempts to native bridge?
9:02AM 0 CVS Build from 16-7-2005 Crash! bug or what? ; -D
8:56AM 2 Register list instead of just one
8:39AM 1 Asterisk Fake Tone
8:14AM 3 Which ATA adapter to use with an analog fax maschine?
7:32AM 1 SIP Phones with Asterisk
7:11AM 1 Strange PRI lockup
6:57AM 3 No sound when bridging two single FXO cards
6:37AM 3 Help DBdel is not working.
6:19AM 2 bandwidth cosume - iax
4:55AM 2 presence in cvs head - how does one map extension to sip user?
4:52AM 1 Things about Mail Notification
4:36AM 3 Remotely Access an Extension
4:33AM 0 Polycom phone configuration script available for download
4:02AM 3 No voice for SIP to ISDN?
3:58AM 0 A-law distortion
3:21AM 2 SIP CANREINVITE
2:40AM 6 CID Matches On Incoming BroadVoice???
2:15AM 0 Has anybody installed Sphinx?
1:17AM 0 Asterisk with Realtime registration problem
 
Monday July 18 2005
TimeRepliesSubject
11:16PM 0 MeetMe application without ZAPTEL INTERFACE
10:02PM 1 AW: Concurrent users
8:36PM 23 Polycom IP600 - Worth the extra $$
8:16PM 2 Asterisk 1.0.9
8:00PM 0 re: Asterisk will destroy the call if no answer
6:53PM 4 Bulletin Board for Asterisk is Now Available
6:17PM 0 Zap channel is never realised
6:08PM 3 Vizufon Video Phone
5:58PM 3 Polycom 501 Configs
5:46PM 1 Concurrent users
5:44PM 8 Streaming MP3's from Asterisk with Ices
5:35PM 12 G.729 licensing - Hardware Devices rather than software
4:56PM 4 ztdummy (again)
3:19PM 8 Panasonic KX-TD500
3:13PM 27 So you all think VoIP sypply is warm and fuzzy
2:04PM 6 TDM04B - Takes long to initialize...
2:03PM 0 Zaptel noise
1:51PM 5 snom 360 audio garbled
1:25PM 0 call pickup with a variable pickupgroup/callergroup based on context
1:19PM 8 Crazy stuff in latest CVS HEAD
1:16PM 2 Mail Notification
1:10PM 0 Statics per Server
12:53PM 2 Restricting outgoing calls by extension / Multiple providers
12:20PM 1 one-way IAX trunking
11:43AM 6 does asterisk support FAX t38 protocol?
11:27AM 0 Help in setting up MGCP from asterisk@home
11:04AM 0 chan_sip.c:939 __sip_xmit warning
10:28AM 1 Sending an SMS out of Asterisk via Kannel
9:36AM 0 Ring requested on channel already in use?
9:22AM 8 Codecs and bandwidth
9:17AM 7 Comments on Areski Calling Card Solution plz
9:14AM 1 Transcoding problems
9:08AM 6 long pause on dialing..
9:03AM 0 IAX register confusion
8:57AM 0 IP Trunking for LD?
8:40AM 1 massive outbound calling...
8:38AM 3 Asterisk @ Home incoming CID
8:29AM 1 Asterisk Comedian Web page login
8:25AM 2 Iaxy and Echo
8:23AM 0 Attended transfer with original CID info?
8:21AM 2 swissvoice
7:58AM 0 Crash on reload only with autoload=no
7:40AM 2 TDM04B + Voicemail poor Quality
7:02AM 1 Stale nonce received from
6:33AM 4 Teliax to VoIPJet
6:23AM 0 Multiple Appearances of Extension on Multi-line SIP Phones
6:15AM 1 telecomFM CellRoute GSM with Asterisk?
6:04AM 0 Astricon 2005 :: Call for speakers and Asterisk projects
5:33AM 1 SIP reinvite on calls over multiple Asterisks
5:30AM 3 CVS Build from 16-7-2005 Crash! bug or what? ;-D
5:27AM 2 Passing DTMF Transparently
5:20AM 0 why $cdr{'CALLERID'} and $cdr{'DNID'} are empty in perl agi connected with asterisk manager
4:42AM 2 Asterisk/Hylafax <=> Receive/Send faxes
3:47AM 6 unsolicited NOTIFY messages from Asterisk
3:29AM 4 configuring trunks
3:15AM 1 FastAgi ...fastagi-mapping missing error
2:59AM 0 test mail - please ignore
1:29AM 0 [bristuff] returning a Busy to the telco?
1:28AM 1 ISDN cards that support nt mode
 
Sunday July 17 2005
TimeRepliesSubject
11:33PM 0 Cisco ATA186 Internal Dialplan: How to send *8?
10:19PM 0 Debugging Realtime Asterisk
9:48PM 2 OT Number of Agents for Tech Support Call Center
9:38PM 2 Problem while capturing DTMF digits in AGI
9:02PM 1 * CVS-HEAD and ASTCC Intermittent issue
8:38PM 1 FreeBSD 5.4 (Asterisk 1.0.9) - Playback , MP3Player and Musiconhold not working
8:33PM 1 Asterisk@home not accepting IAX calls from outside
8:16PM 15 Difference between Asterisk and Asterisk@home
8:08PM 2 Panasonic KX-T7665 and Asterisk?
6:52PM 3 System Jsut hangs Up
3:47PM 0 negative timestamp error
3:07PM 0 asterisk and TTS ( text to speech)
12:55PM 6 HFC BRIstuff woes
12:37PM 4 HOW TO make xten eyebeam incoming video start before you send yours
12:16PM 0 [Asterisk-Dev] Please, excuse me
12:09PM 6 modprobe wcfxo fails.
12:03PM 0 wcfxo fails to find Sweex CA000022 - X100P clone
9:10AM 0 Voipjet test account - unable to make calls.
9:10AM 0 oh323.conf ... how to regitster users ... tell me PLZZZZZZ
8:46AM 0 Dialing via sipgate - remote answer does not stop asterisk internal ring until cycle finished?
6:30AM 0 Have some latency problems.
4:01AM 2 DNS SRV
3:09AM 0 Queue Log Analyser Build into IPS 0.123
2:39AM 0 Xten does not want to dial
2:24AM 1 chan_sip.c:5606 check_auth: stale nonce received from
2:07AM 0 Queue Log Analyser Build into IPS
1:47AM 5 Read error om sound device
1:24AM 0 Pingtel hardphone config' requested
 
Saturday July 16 2005
TimeRepliesSubject
10:20PM 2 beginners question about extension context
9:15PM 3 Multiple ISDN BRI Units with Asterisk usingBristuff zaphfc in NT mode?
6:14PM 6 Implementing a ISDN home PBX
2:47PM 0 VoIP with asterisk and x-lite
1:32PM 5 Sip registration question
11:55AM 8 Asterisk Interface with mobile phone
11:38AM 1 Voicepulse connect - unable to dial out, asterisk says "9696"
10:53AM 1 Cisco 7960 Auto Answer (SIP)
10:50AM 0 DTMF transparancy
10:45AM 0 [ANNOUNCE] chan_capi-cm-0.5.4 release
9:58AM 3 FreeBSD 5.4 (Asterisk 1.0.9) compile error
9:21AM 8 InfoWeek Article on VOIP
9:12AM 0 Hangup Detection with busydetect
9:09AM 3 Memory leak in asterisk CVS
8:55AM 11 Any Ideas??? 3rd time posting => Sipura SIP Phones Multi-Line Appearance... How to use? |----->WAS----> NEWBIE Question: Asterisk with multiline/button phones
8:44AM 0 Asterisk International Carrier Buildout - Create our own International networks for BEST pricing!
8:07AM 0 Paging (I know, AGAIN)
7:47AM 2 howto on ISDN HFC cards with AAH v1.1
7:23AM 2 Voicemail management
6:53AM 1 Voicemail macro?
5:11AM 1 BT / X100P impedance matching
3:54AM 1 Got SIP response 406 "Not Acceptable" back from 10.0.0.10???
3:51AM 0 Server side call waiting for SIP
3:17AM 0 nathelper vs. asterisk
2:00AM 0 why $cdr{'CALLERID'} and $cdr{'DNID'} are empty in perl agi with asterisk manager
1:22AM 3 Multiple ISDN BRI Units with Asterisk using Bristuff zaphfc in NT mode?
1:04AM 2 PRI got event: HDLC Abort (6) on Primary, D-channel of span 1
12:27AM 0 G729 with 2 channels
 
Friday July 15 2005
TimeRepliesSubject
10:35PM 0 Can Asterisk pass through SIP status codes transparently?
9:27PM 0 Error Broken Pipe
8:44PM 2 How to 'read' ztmonitor and set gains
5:28PM 1 Re: Asterisk-Users Digest, Vol 12, Issue 103
4:24PM 0 Help - Lost All Calls
3:44PM 2 SYMBOL NETVISION II NP-3010
3:25PM 1 Manager API commands QueueStatus and Queues
1:36PM 11 arrgg! www.voip-info.org down again (or too busy)
1:00PM 1 OT: cisco voip vulnerability
12:58PM 0 Unicall and Asterisk HEAD
12:55PM 0 VM on * for CME Install
12:21PM 2 seems-to-be-inexpensive source of polycom 301 and501
12:21PM 0 Cisco 7920 WLAN Phone
12:01PM 0 seems-to-be-inexpensive source of polycom 301 and 501
11:30AM 0 Channels being lost/disconnected using Q.SIG
11:23AM 10 VPN's
10:16AM 0 How to get _out_ of an attended transfer?
10:11AM 2 2 TDM04B In Asterisk at home
9:22AM 1 [Asterisk-Dev] call pickup with snom function keys now working with cvs-head + patch sipsubscribe-20050715.rev779.txt
9:14AM 0 OT (kinda): Justification for adding Asteris kto the business plan
9:07AM 2 [Aserisk-Users]no audio inside the net
7:57AM 0 Queue_log stats
7:31AM 0 Grandstream SIP phones across NAT
7:13AM 0 Asterisk and Cirpack, REGISTER patch ?
7:11AM 1 make problem.
7:01AM 0 No ringing using SIP or IAX phone, ringing using ZAP channel
6:27AM 0 Memory Leak in CVS-HEAD 11-22-04?
5:48AM 0 double dtmf in incoming SIP call?
5:20AM 0 Differences between System 75 and Asterisk
5:15AM 2 OT (kinda): Justification for adding Asteriskto the business plan
5:03AM 9 WG: Cisco 7920 WLAN Phone
4:49AM 4 RES: Meet Me - this is not a valid conference number, please try again
4:17AM 1 OT (kinda): Justification for adding Asterisk to the business plan
3:30AM 0 08** presentation numbers in the UK
2:24AM 1 Asterisk+errision PBX
2:12AM 2 Strange problem with SIP and CAPI
1:52AM 1 channel.c:41:31: asterisk/transcap.h: No such file or directory problem
1:19AM 9 RE: 2 asterisks connected but trying to bridge
1:10AM 6 Meet Me - this is not a valid conference number, please try again
12:59AM 0 2 asterisks connected but trying to bridge call and this is not wanted
12:46AM 2 problems with tdm11P
 
Thursday July 14 2005
TimeRepliesSubject
11:41PM 4 Vonage to IAX DID to IVR => Poor DTMF
11:24PM 0 H264 Passthru
10:18PM 0 How to start with SER and Asterisk?
10:14PM 1 RTP not thru asterisk
9:22PM 4 PSTN to SIP gateway
9:08PM 4 * behind NAT and local subnet
8:30PM 14 Polycom configs?
8:28PM 1 Building zaptel on x86_64
7:36PM 0 dialplan for monitoring outbound calls
6:32PM 1 Any way to authenticate SIP peers using SRV?
6:31PM 4 Phone manual..
6:28PM 2 LED went off after loading wct4xxp
6:14PM 1 How to change Port for SIP users
6:13PM 0 Zap channel billing on busy tone!
5:30PM 1 Asterisk SIP to extenal PBX extension
3:41PM 0 Sorry
1:20PM 8 Polycom Auto-Answer problems
1:11PM 0 Seperate RTP server, voice not being received.
12:49PM 22 SoftPhones: Bad, or just bad QoS?
12:22PM 0 Setting Callerid in callout file problem
12:08PM 3 Latest Stable
11:45AM 0 Cisco CME Integration - IOS Version known to work?
11:36AM 0 Asterisk (or generic telecom) Stencil's for Visio 2003
11:35AM 0 dialplan logic, logical not
11:28AM 0 PRI Channel Question
11:17AM 1 MOH Class in MeetMe
9:57AM 0 Monitor command stop on call transfer
8:46AM 5 Systems Admin; Telecom Newbie - What do I ne ed?
8:34AM 1 Re: <asunto_mensaje_entrante>
8:31AM 0 Re: <asunto_mensaje_entrante>
8:29AM 0 Re: <asunto_mensaje_entrante>
8:26AM 0 Polycom behind firewall issue
7:54AM 0 HFC + DECT sync
7:54AM 0 MeetMe + CONSOLE
7:41AM 0 SMS in Belgium
7:35AM 0 Changing the voice in Asterisk
6:38AM 5 asterisk number of calls
6:35AM 0 AgentMonitorOutgoing question
6:34AM 0 Plzzzzz tell me how to register users in oh323.conf
5:59AM 4 Cisco 7960 on Asterisk?
5:42AM 0 No more sound on MOH after adding TE405P
4:53AM 6 SpanDSP rxfax, no tiff
3:33AM 0 bandwidth of gsm and g729
3:33AM 0 bandwidth og gsm and g729
2:46AM 3 Sangoma A104c vs. A104u
2:26AM 0 SMS transmit to analog device
2:08AM 1 auto dialing - call file - channel variable question
2:00AM 4 Wire Tapping on Asterisk
1:55AM 0 [Asterisk-Dev] PRI Q.921 problem
1:22AM 1 *** install error
12:06AM 3 CVS HEAD voicemailbox full error
 
Wednesday July 13 2005
TimeRepliesSubject
11:53PM 7 Panasonic PBX -to- Sirrix BRI: Numbers getting echoed/duplicated
11:16PM 1 VOIP phone, how to use with asterisk ??
10:30PM 8 Is soekris good?
9:59PM 0 Are chinese voice files available?
6:45PM 2 DBput from the web?
6:03PM 2 SMS on my own possible?
5:19PM 5 CONSOLE/dsp
5:14PM 1 Manager API quit working for no apparent reason
3:56PM 0 call forward / and voicemail setting
3:55PM 2 time includes
3:45PM 0 Jukebox
3:29PM 0 tiny audio drops (blips)
1:57PM 0 AW: SpanDSP rxfax, no tiff.
1:54PM 0 Sipura SIP Phones Multi-Line Appearance... How to use? |----->WAS----> NEWBIE Question: Asterisk with multiline/button phones
1:31PM 0 Limit Minutes / Mensual?
1:28PM 3 NAT Asterisk Peering
1:07PM 72 Business Edition
12:47PM 6 Festival questions
12:40PM 3 Intermittent Silence
12:31PM 0 Problem with inbound/outbound calls bridging on Zap lines
12:29PM 4 Mixed Voice/Data T1
12:24PM 2 Minutes Limits
12:20PM 9 chan_sccp new release
12:06PM 1 How to get a beep on a Auto Answer Intercom - Cisco 7960
11:58AM 5 Support needed
10:52AM 3 SMS over SIP and Asterisk ??
10:48AM 0 R2 Digital
9:57AM 2 SPA3000 to Asterisk Server - Asterisk server not answering calls
8:40AM 15 Multiple NICs on Asterisk box
8:35AM 3 Meet Me Configuration
8:29AM 0 Re: Asterisk-Users Digest, Vol 12, Issue 65
8:27AM 0 SIP calls to 'BUSY' or OFF HOOK PSTN numbers do not return busy indicate to sip phone?
8:19AM 6 No channels after starting asterisk
8:17AM 4 SpanDSP rxfax, no tiff.
8:06AM 23 OT: DS3 -> VoIP Hardware Recommendations
7:39AM 4 Anyone signed up with Galaxyvoice lateley?
7:39AM 3 asterisk E1 in europe
6:43AM 0 [Asterisk-Dev] CVS HEAD behavior change warning
6:36AM 7 Faxing Suggestions
5:43AM 1 Polycoms and paging
5:18AM 0 how to connect to asterisk via perl agi
5:01AM 2 Can I introduce sql sentences in the DialPlan (Asterisk Realtime)??
4:39AM 2 extension mobility and CDR logging questions
4:01AM 1 help needed-call SMS
3:56AM 0 Cisco ATA186 + Dell 1600n printer-fax
3:09AM 0 Running commands from dialplans
2:34AM 0 Two ISDN cards on same machine
2:18AM 0 how many g729
1:55AM 1 Suddenly a problem with outgoing calls made from Cisco phones...
1:39AM 0 h323 still no success to dial out via GK
1:31AM 0 Call file ][ Unable to request channel ZAP/g1/0123456789 ][ Call failed to go through, reason 0
12:52AM 2 OT: proliant fedora asterisk
 
Tuesday July 12 2005
TimeRepliesSubject
10:34PM 0 Problem with modem and asterisk
8:55PM 8 SpanDSP+astfax with multiple fax pages
8:38PM 0 personal voicemail , and call transfer --- howto
8:36PM 1 Re: Asterisk-Users Digest, Vol 12, Issue 79
7:58PM 1 Skip Announcement Confirmation in MeetMe
7:38PM 6 NO calling tone
6:03PM 10 Unable to call certain 800 numbers through Teliax
5:58PM 3 AgentCallbackLogin Question
5:34PM 1 Little doubt on Asterisk and EyeBeam
5:28PM 0 Manger-command Getvar?
4:22PM 1 Compile failure on Mac OS X Tiger
3:15PM 3 Polycom 600 phone
3:02PM 0 TDM400P FXO callprogress doesn't detect remote answer
3:00PM 0 IAX2 ping confusion and unreachable soft phones
2:53PM 0 H323 email address
2:33PM 25 Systems Admin; Telecom Newbie - What do I need?
2:29PM 3 SNOM 360 and parking
2:12PM 2 Cisco 79XX Jitter Stats Question
1:47PM 0 Cisco SIP Frimware for 7940/7960 v7.5
1:09PM 0 Pushing new firmware to Snom 190 <--solved
1:07PM 1 help needed-call recording
11:53AM 5 ASTPP
11:52AM 2 Odd MOH problem...
11:23AM 5 Having Trouble Creating an IVR
11:15AM 0 Returning values from macro inside Dial command
11:07AM 0 FYI: BT Caller ID.
10:59AM 4 Asterisk and Dell SC420 Server
10:19AM 0 Network Configuration Question for Asterisk Server
9:44AM 7 Help Configuring TDM04B
9:21AM 3 Cisco 7940/7960 interdigit timeout
9:16AM 39 Any suggestions for an IP phone?
9:13AM 0 meetme an customized menu
8:28AM 0 TDM22B - asterisk and seimens hipath 3750
7:59AM 6 PRI problem
7:33AM 2 New Cisco 7960 Firmware 7.5
6:53AM 1 How to integrate the "Call Pickup with CID info" feature in the release tree of Asterisk?
6:39AM 0 Referrals/Success Stories would be greatly appreciated
5:42AM 3 how to debug perl agi
5:35AM 2 choosing a softphone
5:11AM 2 Will pay for asterisk help...
5:00AM 2 Help: TE100P connecting to non PRI, ISDN interfaces
4:31AM 0 asterisk as media proxy
3:48AM 2 Possible Agent Bug found while upgrading from Asterisk 0.7.1 to 1.0.9
3:46AM 0 IPSwitchBoard shows Call Charges
2:40AM 0 chan_capi-cm-0.5.3 and ${DNID}
2:37AM 11 Last CVS -> High Load
1:54AM 5 bristuff patches and realtime mysql
1:32AM 0 asterisk PBX and Siemens Hipath 3750
1:31AM 12 asking again
12:52AM 3 monitor using incorrect path
12:42AM 0 Asterisk not accepting user input .. pls help !!
12:41AM 0 Asterisk realtime failover problems
12:21AM 0 Asteriski misses the table
 
Monday July 11 2005
TimeRepliesSubject
11:17PM 0 Modem Connection from TDM card to TE4xxP card
8:58PM 3 SIP NAT + m0n0wall 1:1 mapping
8:50PM 3 asterisk and seimens hipath 3750
8:48PM 4 h323 and asterisk
8:39PM 1 Rating application for Asterisk
8:29PM 0 CallBack Retries
7:31PM 24 IP Phone with Standard Power Ethernet
7:01PM 0 sphairon ADSL with FXS
6:12PM 5 Unable to dial certain calls
6:12PM 6 Enabling rtcachefriends prevents phones from calling each other
5:29PM 8 Zaptel won't compile under Fedora Core 4
4:08PM 1 Looking for a consutant in France about Asterisk.
3:37PM 1 G729 - What versions can Asterisk support?
3:36PM 2 Question about Polycom SoundPoint 500
3:18PM 0 Which H323 for Video and how to setup
3:08PM 0 Grobill 0.1 - Asterisk Prepaid Billing
2:27PM 8 Pushing new firmware to Snom 190
2:07PM 0 zaphfc / incoming call - error 6
1:51PM 1 VoIP services
1:32PM 2 TDMoE and callerID
1:06PM 11 DTMF not sending properly via IAX
1:03PM 0 Forward the ALERT_INFO
11:33AM 0 Help !!! astcc
11:27AM 1 OT- USA reseller list required
11:27AM 2 [Asterisk-Dev] Peter Nixon to Speak at Cluecon
11:17AM 1 RTP traffic
11:16AM 1 Snom 360 NOTIFY syntax
10:54AM 0 Some refer transfer questions / issues!
10:13AM 1 Compile Error chan_sccp-20050705 on asterisk 1.0.9 (tarball)
9:37AM 1 chan_cornet status
9:28AM 0 FW: Retrieving dtmf, passing to shell and getting the result
9:21AM 1 Zaptel configuration for Argentina
8:31AM 0 No sound when dialing out over SIP Proxy
8:12AM 0 error related to the native formats
8:00AM 0 [Asterisk-Dev] Vikrant Mathur lead developer for the open sourceOSP Toolkit to speak at Cluecon.
7:51AM 2 Asterisk @ Home Voicemail
7:47AM 0 [Asterisk-Dev] Vikrant Mathur lead developer for the open source OSP Toolkit to speak at Cluecon.
7:21AM 0 Trunk number (SMDI)
6:30AM 1 Valgrind effects
6:18AM 1 ASterisk@home + Broadvoice = Almost working installation...
5:00AM 2 2.6.13 Kernels
3:13AM 1 How to force RTP through Asterisk PBX.
3:01AM 0 Calls dropped upon 'native bridging' after IAX2 transfer
2:43AM 2 Dial SIP extension
2:42AM 8 asterisk and h.323
2:35AM 0 DIAL Event, who picks up?
1:40AM 4 Sharing variables between contexts
12:41AM 18 Video phone settings???
 
Sunday July 10 2005
TimeRepliesSubject
11:59PM 1 searching for assistance
10:59PM 5 asterisk cluster
10:22PM 4 Howto get streaming mp3 at an extension?
10:02PM 1 how to download chan_sip2
8:17PM 0 NEWBIE Question: Asterisk with multiline/button phones
7:45PM 5 Problems with a new box of asterisk@home 1.3
2:24PM 0 SIPGetHeaders for chan_sip (derived from chan_sip2)
2:11PM 5 VM Outcall: Rube Goldberg Edition
11:39AM 0 Re: Asterisk-Users Digest, Vol 12, Issue 63
10:44AM 0 iax fwd - calling twice
10:16AM 7 Incoming calls from BudgetPhone.nl
9:55AM 0 (no subject)
9:21AM 4 SMS Handler in Asterisk
9:18AM 0 Tormenta 2 / E400P cards in AMD 64 bit machines
9:01AM 0 Problems with firefly connection via SIP
8:12AM 0 How to properly handle incoming SIP and IAX calls, so user can call back and how to properly make outgoing sip/iax calls through Asterisk ?
7:41AM 1 Retrieving dtmf, passing to shell, and getting the result
7:24AM 1 IAX2 softphone for Pocket PC
7:11AM 2 GXP-2000 MWI
6:46AM 0 SIPGetHeaders for chan_sip (derived from chan_sip2 )
5:09AM 8 iax.cc opinion request
3:06AM 3 chan_capi ASTCC trouble
1:37AM 0 Mitel 5220 Hold?
12:11AM 0 Time out not working from php agi...
 
Saturday July 9 2005
TimeRepliesSubject
11:34PM 2 MeetMe problem - some parameters ignored
11:29PM 0 Strange softphone issue - audio open before answer
9:44PM 2 TDM04B Outbound calls
8:27PM 0 FS: Digium TDM04B (PCI with four FXO daughterboards)
7:27PM 0 Meetme recordings
4:59PM 1 Remote SIP Connection using Asterisk // Cisco7940's
3:27PM 3 polycom soundpoint 300 sip phone and hold music
12:50PM 4 Modifying astcc
11:03AM 3 Remote SIP Connection using Asterisk // Cisco 7940's
10:20AM 0 Agent Queue, Silent Calls Problem
6:21AM 4 can we register users in oh323.conf ?
6:15AM 4 Asterisk + spandsp
5:52AM 1 chan_bluetooth, no voice
5:15AM 0 make available again a zap channel after a red alarm...
5:01AM 6 SIP phone w/ XML browser
2:54AM 0 Closest dialplan language equivalent for dialparties.agi ?
1:28AM 2 how to edit ring time
12:48AM 0 About the using of astmanproxy
12:03AM 2 Polycom SP300 config files
 
Friday July 8 2005
TimeRepliesSubject
10:37PM 10 editing ring time
10:16PM 10 phantom incomming calls from asterisk
7:26PM 0 Agent Silent Call Issue (seems like an asterisk bug / SjPhone Bug)
7:03PM 0 Leave Message - IVR don't work
6:31PM 0 All Circuits Busy instead of Busy Signal when calling a busy number using a PRI
3:45PM 5 McLeod Integrated T1 - no PRI?
2:42PM 0 dialling in from analog line -> only get 2 of 3 digits extensions
2:36PM 2 admin password does not work on APM in a new box
2:01PM 6 Can Asterisk ring a specific extension based on the number the outside caller dialed?
1:14PM 0 FW: Routing DID calls to external lines
12:19PM 5 best Fax board?
12:17PM 0 INVITE/REFER with only 2 ends on asterisk
11:42AM 3 Soft-switch.org is out?
10:58AM 3 Dial 9 to PBX to PSTN pattern question
10:04AM 0 GnuGK Nufone H323 -HEAD - Prefix issue
8:56AM 0 Exception flag set on 'UniCall/2-1', but no exception handler
7:12AM 1 Serial port DTMF Caller-ID reciever /w scripts for X100P/X101P/Clones....
6:37AM 5 Unknown numebrs to a context/extension
6:18AM 1 Loading configuration files from the database.
6:04AM 0 IAX - newbie question
5:06AM 0 call pickup with snom phones
4:43AM 0 asterisk-oh323: New version 0.6.6
4:27AM 1 Two TDM04B
3:15AM 1 WARNING[3240]: chan_oss.c:305 sound_thread: Read error on sound device: File descriptor in bad state
3:04AM 0 Testing Asterisk with PROTOS
3:02AM 0 PSTN -> E1 -> Asterisk -> T1 -> Rhino Channelbank -> Faxes --- works ?
2:19AM 0 beronet and tdm400p cards conflict?
1:42AM 1 kascaded call queue
1:14AM 4 changing "Nobody picked up in 30000 m"
12:50AM 1 DSL Provider
12:43AM 8 Definitive CallerID Format and anonymous?
12:11AM 1 Help needed - Zap Transfer Failing...
 
Thursday July 7 2005
TimeRepliesSubject
11:33PM 0 h323 how to ?????
11:03PM 0 Re: Braodvoice - UK Non Geographic Numbers
9:00PM 2 Routing DID calls to external lines
8:56PM 0 wholesale DID's?
7:21PM 3 Extension Problems
5:34PM 1 Problems to leave messages in Asterisk
5:28PM 20 Asterisk@home and Cisco 7910
4:23PM 0 CDR Record question.
3:52PM 8 Sipura SPA-841 Volume Oscillation Problem
2:56PM 2 Announce incoming callerID via paging/intercom?
2:24PM 2 MeetMe hardware dimensioning
2:07PM 2 IAX2 Trunking - CVS-Head
12:55PM 0 zapata.conf reload
12:49PM 0 Parial Hang with cvs-HEAD and queues/agentcallbacklogin
12:46PM 2 Asterisk Crashes after update
12:33PM 6 Long Distance
12:07PM 2 changing "Nobody picked up in 30000 ms"
11:24AM 0 TDMoE bandwidth and load
10:49AM 3 Newbie Question: Type of card
9:13AM 3 IAX Transfers
9:09AM 63 IAXphone -> ip address -> extension number.
8:46AM 2 FXO hangup Problem.....
8:20AM 1 How to slow down dialing
8:09AM 4 res_config_mysql.so in CVS asterisk-addons broken?
8:07AM 1 Logging SIP response codes
7:46AM 0 [Asterisk-Dev] Cluecon, A mix of leading Open Source VoIP devlopers...
7:41AM 0 rxfax/txfax
7:12AM 3 Asterisk/Grandstream Budgetone disconnect issue
6:50AM 0 [Q]: Asterisk + gnugk + BRI ISDN as H.323/ISDN gateway?
6:50AM 4 isdn30 / pri lines in the UK
6:45AM 2 Using G729 in pass through mode
6:42AM 0 app_rxfax and app_txfax - Asterisk CVS HEAD
6:06AM 0 mISDN transferring a call
5:54AM 1 Queues and busy agents problem
5:36AM 1 Calls with oh323 with no sound
5:01AM 0 disconnect with various codecs
2:27AM 1 experience with analog channel banks in E1 land
2:21AM 3 asterisk and wireless on site personal paging system
12:59AM 0 Senao WiFi SIP Phone SI-680H
 
Wednesday July 6 2005
TimeRepliesSubject
11:05PM 2 Teliax Passing Audio?
11:02PM 0 Dropped calls if transferred across servers into MeetMe with mobile source
10:34PM 0 oh323...getting incoming calls ... but how to do outgoing ????
8:05PM 6 SIP Xten eyeBeam Video Problems
7:18PM 0 Restart DISA from the beginning
6:34PM 0 verbosity in log files
4:50PM 2 "Set" syntax equivalent of DBDel?
4:47PM 0 E1 Channel Bank Recommendation
4:07PM 0 I need somebody who has a video phone for testing
4:04PM 0 Call Pulver communicator to an asterisk box
3:59PM 0 Dialplan help needed: How to avoid wakeup call in the voice mail box?
3:07PM 12 Emergency Asterisk Guru Help needed EMERGENCY
2:59PM 1 /etc/asterisk/manager.conf
1:55PM 2 ATA not sending data to asterisk?
1:53PM 5 NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1
1:36PM 0 cdrtool anyone ?
1:33PM 1 Putting AGI applications in their own directories
1:33PM 0 Busy tone (German/Dutch/French)
1:08PM 3 asterisk perl radiusclient
1:02PM 0 Using DISA when dialing into a Zap interface
12:47PM 8 cdrtool
12:27PM 15 Connect 30 phone lines to asterisk how to
12:12PM 0 [Asterisk-Dev] Bounced mail apologies
12:08PM 7 quadBRI form junghanns.net
12:04PM 4 Incoming 800-number over IAX - first few words are cut-off
11:45AM 1 [Asterisk-Dev] Retrieving number of messages in a mailbox by an application
11:44AM 0 FW: Maximum Number of Mailboxes in Asterisk
11:41AM 0 Maximum Number of Mailboxes in Asterisk
11:10AM 4 converting windows .wav to .gsm
10:58AM 1 Some problems setting outgoing PRI Origination Number
10:56AM 5 zaptel missing /dev/zap after FC3 update
10:42AM 3 ISDN PRI No Audio
10:03AM 4 OT: Congrats, Europe!
9:51AM 12 problem with iax2 and 2 peers behind nat
9:31AM 10 Snom phones - any advice
8:49AM 0 re: help debugging dialplan
8:44AM 2 Crash without "make valgrind"
8:43AM 0 Send Variables over IAX
8:40AM 7 Asterisk 1.1
8:23AM 0 Asterisk voicemail
7:46AM 0 I was mistaken about Areski: he does relply to mails and help people.
7:32AM 2 FW: ETSI or QSIG
6:51AM 3 how to set language in capi
6:50AM 8 DECT VoIP Gateway
6:33AM 0 getting Incoming but unable to dial out using oh323
6:21AM 7 app_conference and AGI
6:20AM 0 UK asterisk
6:08AM 4 phone comparison matrix
5:59AM 0 MWI from SIP provide
5:27AM 3 Polycom distributor in the UK ?
4:32AM 2 chan_capi 0.5.3 & asterisk HEAD 2005/07/04 undefined symbol error
4:29AM 2 Dialplan configuration with Realtime
3:30AM 0 newbie asterisk-addons installation
2:46AM 1 app_rxfax does not receive
2:03AM 0 can we use asterisk as a SIP Redirect Server?
1:35AM 0 Can we use Asterisk as a Redirect server??
1:33AM 1 g.729 codec -- open source?
12:49AM 2 SIP/2.0 403 Forbidden
12:48AM 0 SIP dialout
12:22AM 0 Re: Asterisk-Users Digest, Vol 12, Issue 25
12:20AM 4 cisco 7940 + sccp issue
12:09AM 11 URGENT: hardware spesifications needed
 
Tuesday July 5 2005
TimeRepliesSubject
10:07PM 3 Asterisk addons install problem
9:00PM 1 Help with Cisco 7905G corrupted image!!
8:36PM 0 Polycom IP 500
5:52PM 8 Users handbook
5:16PM 0 Re: [Serusers] NAT considerations...
3:22PM 9 Remote SIP Connections
3:08PM 0 sip peer dinamically
2:52PM 2 Previously: Queue + optional URL
2:48PM 0 chan_h323 passes no audio?
2:21PM 3 Stale nonce received?
1:48PM 2 Any SIP hardphone recommendations?
1:03PM 1 (no subject)
11:44AM 1 app_conference, CVS HEAD, SIP and Xen
11:28AM 0 MYSQL alises
11:18AM 0 Re: MOH - request to schedule in the past SOLUTION and New Asterisk Queues Tutorial.
11:06AM 9 Uniden UIP 200 and Asterisk.
10:55AM 1 Early media dectection problem
10:02AM 3 Asterisk-CVS-HEAD locks up on 'reload' from CLI (sometimes)
9:58AM 1 Queue + optional URL
9:28AM 7 How to prevent log files from eating my hard drive?
8:55AM 11 TDM04B problems
8:49AM 1 AT-320EE
8:43AM 5 PRI or Trunk monitoring
8:35AM 1 external IVR
8:14AM 0 chan_bluetooth.so
8:04AM 0 [Asterisk-Dev] Craig Southeren to speak at Cluecon!
7:32AM 17 How does Vonage support fax machines?
7:07AM 5 new Asterisk@home installation
6:53AM 1 voicemail.conf overwritten
6:24AM 0 meaning of parameters
4:43AM 1 Newbie question reg. Asterisk and Channel Access Bank I and TE110p
3:23AM 8 Asterisk on Linksys WRT54G
2:48AM 2 Cmd MusicOnHold works, but no sound when a call gets holded
1:54AM 4 Calls authentication by IP address
1:41AM 5 asterisk box after an analogic pbx
12:49AM 0 Problems installing AMP
12:07AM 0 About AgentMonitorOutgoing
 
Monday July 4 2005
TimeRepliesSubject
11:49PM 0 Transfer and CDR's
11:46PM 0 Dialogic D/300 E1
11:43PM 3 calling shell scripts from within *
11:24PM 1 Problem in connecting Arekiscc and asterisk using sip channel!
10:02PM 2 Line sharing
7:14PM 1 QoS settings of the SIPURA ATA
7:10PM 4 Colocation/Telehousing
6:12PM 1 Anyone written a call recording interface
4:33PM 26 Simpletelecom dead?
4:26PM 6 HDLC bad FCS
3:56PM 0 Asterisk stop working with HiSAX ISDN
3:56PM 0 no sound. "Failed to write frame" (2nd post)
3:51PM 3 Restricting SIP trunks to extensions
2:43PM 6 VOIP Providers Problems
2:29PM 4 Long delay via Teliax
1:40PM 1 Unresolved symbols - Zaptel 1.0.9 and Linux 2.4.31
12:55PM 4 Proper way to start * and load modules on a RedHatbox
12:13PM 2 Enable verbose output for TxFax/RxFax
12:06PM 7 Proper way to start * and load modules on a RedHat box
11:33AM 1 How to know what happend after dial
11:25AM 2 Hardware sizing
11:16AM 1 Asterisk and Cisco 5300
10:22AM 1 Zaptel and 2.6.13-rc1
10:01AM 2 [Asterisk-Dev] presence and IM again, want to develop a working "hack"
9:59AM 1 Asterisk 1.0.9 and FreeTDS
8:52AM 0 [Asterisk-Dev] New Astmanproxy 1.1 now available!
7:53AM 6 X100P FXO PCI Card + Incoming Fax
7:11AM 9 Call Transfer using SIP clients
7:07AM 2 voicemail (gui vmail.cgi) patch
6:58AM 2 Asterisk on Virtual Machine
6:45AM 6 G729 licencing with asterisk, how does it work ??
6:07AM 2 Extensions will not go to voicemail
5:48AM 0 radius client for portaone with asterisk-1.0.9
5:19AM 1 annoying static when calling from legacy PBX -> * ZAP interface
4:55AM 3 Asterisk with Intel Blade Machine...
4:40AM 1 mgcp fon behind NAT gw
4:20AM 3 Dial *97 to pickup voicemail buts says my password incorrect
3:01AM 3 Fax DETECTION with CAPI
2:34AM 0 OT Mark Spencer lunch in Paris Fri July 8th
2:26AM 0 RE: Asterisk-Users Digest, Vol 12, Issue 17
2:14AM 0 MAKEing zaptel and ztdummy on SuSE 9.3 - Repost
1:52AM 0 Idefisk iax2 softphone - new version
1:49AM 1 OT : Wengo sucks
1:39AM 1 No Sound (2nd post)
1:05AM 0 SV: Epia C3 Linux
12:05AM 28 #include not working with *1.0.9
 
Sunday July 3 2005
TimeRepliesSubject
11:55PM 1 Repost: how to configure asterisk user and group rights
9:38PM 0 no sound. "Failed to write frame"
8:43PM 5 TDM01B card configuration
7:25PM 0 H323 with GSM codec is not working
4:10PM 2 play message to callee beforeconnecttoincomingcall
3:54PM 4 Connecting two servers - dial string
3:49PM 0 how to configure asterisk user and group rights
2:43PM 5 12 seat call centre with Asterisk, VoIP only, UK - possible?
1:42PM 0 MAKEing zaptel and ztdummy
11:45AM 0 Quintum & Asterisk w/ Realtime
9:47AM 0 Example of user authentication?
9:44AM 2 How to keep track of who is doing what?
9:31AM 2 Questions about real-time voicemail, foreign languages and voicemail folders...
9:29AM 2 re: another database question
9:05AM 9 Bind port
8:33AM 1 Swedish CallerID?
6:47AM 17 Buy IP address
6:42AM 1 raising the sound volume on zap
5:11AM 3 asterisk strips off trailing digit from incoming calls
4:10AM 1 editing time in astcc
3:44AM 1 grandstream sip phone to analog not working
3:21AM 2 play message to callee before connecttoincomingcall
1:03AM 0 Asterisk as Media Gateway (was: ATT CallVantage & Asterisk
 
Saturday July 2 2005
TimeRepliesSubject
11:45PM 21 Linux Distribution for Asterisk server use
9:02PM 4 Problem registering Asterisk Dual Server
8:28PM 0 Festival long starting time
5:43PM 0 Connecting * to a Ericsson BP250
5:37PM 0 Audiocodes MP-108 FXO to Asterisk HELP
4:18PM 4 HW Capacity plan - How many Digium is recomended per server
3:50PM 0 play message to callee before connect toinco mingcall
3:17PM 3 LDAP search application for Asterisk
1:30PM 0 Enum or DUNDi
1:25PM 5 Colored asterisk -R?
1:12PM 3 Sipura SPA2000 behind NAT
1:05PM 5 Telephoning Announcements -- Suggestions?
1:00PM 1 play message to callee before connect toincomingcall
12:55PM 1 play message to callee before connect to incomingcall
12:19PM 1 play message to callee before connect toincoming call
12:06PM 4 call forwarding, most basic case
10:23AM 1 play message to callee before connect to incoming call
10:17AM 0 Is it possible to setup group voicemail inAsterisk?
10:03AM 1 Is it possible to setup group voicemail in Asterisk?
9:53AM 0 (Simple?) ENUM Question
9:13AM 3 What to use h323 or oh323 ???
5:41AM 0 Audio delay w/ call forwarding
5:16AM 0 Snom -> Asterisk -> Vegastream
3:04AM 0 PortaOne's Radius client for Asterisk
3:03AM 0 editing time to say astcc-noanswer
2:09AM 0 Operators Panel for Asterisk
 
Friday July 1 2005
TimeRepliesSubject
10:00PM 4 Does PCI Developer Kit work with kernel 2.6
5:23PM 5 MOH - request to schdule in the past
4:46PM 9 asterisk showing more than once on ps
3:14PM 5 Provider Survey
2:33PM 1 Errors Question
2:33PM 0 Got SIP response 481 "Invalid CSeq Number" backfrom
2:31PM 2 How to Configure a H323 Phone (newbie here)
2:15PM 0 Got SIP response 481 "Invalid CSeq Number" back from
1:26PM 1 Got SIP response 481 "Invalid CSeq Number" backfrom X.X.X.X
1:20PM 0 Got SIP response 481 "Invalid CSeq Number" back from X.X.X.X
12:24PM 1 SIPGetHeader application in asterisk-1.0.9
11:56AM 0 new chan_sccp release
11:40AM 1 asterisk newbie and phones which don't want tocomunicate
11:22AM 1 Re: [Asterisk-ss7] Asterisk - ss7
11:21AM 0 Got this error after my installation when i doztcfg -vv
11:15AM 3 pattern matching based on callerid, not working
10:47AM 1 Unable to forward frame/voice
10:33AM 2 how to send voicemail notifcation every 15 minutes until message is checked
10:32AM 1 how does pattern routes works
9:53AM 24 Epia C3 Linux
9:33AM 0 RE: [asterisk] VocTel service provider
9:09AM 15 Visual ring notification
8:45AM 5 Problem with DTFM and complex international setup
8:43AM 0 asterisk newbie and phones which don't want to comunicate
7:34AM 0 (en|dis)able CW within /etc/asterisk
7:27AM 0 voicemail and mysql
7:19AM 0 Catch Autodial failure
7:05AM 0 how to PortaOne's Radius client for asterisk
6:59AM 2 Problems loading asterisk .
6:31AM 3 Source for Sangoma or Digum 2+ port T1 Card near NH??
5:42AM 0 Voicemail storage
5:24AM 1 no voice
4:51AM 8 make error for zaptel
3:41AM 2 astmanproxy
3:36AM 8 Ambient MD 3200 (X100P Clone)
3:22AM 2 Sip.conf problems
3:19AM 6 E3 card
3:07AM 0 Groupcall problem
2:07AM 0 IAX DTMF Challenges...
1:47AM 10 Sometimes yes - sometimes no (dialplan)
1:37AM 0 Asterisk and Alcatel 4400
1:25AM 1 Attended transfer works for caller, not for callee
12:39AM 0 ${BLINDTRANSFER} in *-1.0.X
12:15AM 4 Asterisk and DHCP