asterisk users - Sep 2004

Thursday September 30 2004
11:37PM 0 Please help - jitter problem
11:04PM 1 problem in using sip-communicator with asterisk
7:28PM 1 V1.0.1 - Segmentation fault in ?
7:11PM 1 Queue Setup almost got it
6:43PM 0 register expire
6:35PM 2 Queue Setup
4:26PM 1 Monitoring SIP with Whats Up Gold
4:06PM 2 Timing Problem
3:38PM 4 Caller ID Info from Cisco router to Asterisk
3:13PM 7 Asterisk hardware
2:54PM 0 cisco 7940 in mgcp
2:02PM 2 Time of Day Routing
1:31PM 4 Voice mail
1:17PM 2 FXO/FXS card
1:02PM 1 Channel banks that you've used
1:00PM 0 BRI cards for Asterisk in US
12:52PM 3 Debian: loading Zaptel modules at boot time
12:49PM 0 CLI color using -r
12:36PM 0 Re: Re: Re: Confused of London - How to associate zapchannels to extensions
12:15PM 1 Sip over cisco VPN
12:13PM 1 Re: Re: Re: Confused of London - How to associate zapchannels to extensions
12:11PM 0 autodial question
12:00PM 0 * doesn't answer
11:15AM 0 UK Caller ID - todays CVS update knocks out a channel
10:42AM 1 Anyone in digium?
10:26AM 2 OT: Kphone installation problem
10:21AM 0 Asterisk seems to have more jitter than a hardphone with SIP
9:28AM 5 Confused of London - How to associate zap channels to extensions
9:17AM 7 asterisk 407 Proxy Authentication Required
9:03AM 4 Setting auto-attendant to answer immediately
8:29AM 0 Asterisk server suddenly fronzen after many zapter errors
8:09AM 4 Ring Multiple SIP client at the same time
7:54AM 3 Sipura-3000 - silent dial out on FXO port
7:25AM 6 No Asterisk Sounds on SUSE ES 9/Linux 2.6
7:06AM 0 Re: Asterisk-Users Digest, Vol 2, Issue 329
6:42AM 1 CVS best practice
6:22AM 1 Strange Quality problems with Asterisk, Gentoo, Redhat and Kernels - /dev/dsp]
6:13AM 0 Refer Method
4:08AM 1 Sip Channels.
3:02AM 0 Oops, a seg fault =(
2:16AM 1 sipfriends in MySQL question/request
2:06AM 4 No Audio
1:56AM 2 cli command to check the codec use for the connected calls
1:15AM 1 easy way of add 100 extensions
12:44AM 0 Video mail
12:36AM 1 how to hung up a call immediately if it SIP response 486 "Busy Here" received
Wednesday September 29 2004
11:29PM 2 Asterisk 1.0.1 Released
10:51PM 0 CDR Disposition for call queue's
10:37PM 1 Asterisk 1.00 Call quality problem
9:31PM 0 Big zaphfc issues
9:24PM 3 Calling Waiting on PSTN line
8:52PM 0 Query on DMA Config
8:50PM 1 Budgetone problem
7:21PM 1 mid-call echo
6:37PM 3 7912G SCCP only?
6:34PM 0 TDM400P (TDM20B) 'Freshmaker failed register test'
5:52PM 3 Dial Delay
5:47PM 1 I am newbie with asterisk
5:36PM 0 astersk-oh323 compile error make
5:33PM 0 no sound from jsphone, the voicemail ok
5:07PM 4 Wooksung Video Phones
3:33PM 0 Queue member still rings even if on call
2:34PM 1 Zaptal and Fedora Core 2 and losing GSM playback
2:15PM 3 HELP: Asterisk - SIP to H.323 translation
1:54PM 0 Stanaphone DID : user not available
1:49PM 1 VoicemailMain GSM
11:30AM 0 SIP signalling from Gateway
11:26AM 0 Re: Asterisk Performance
11:14AM 1 Channelized E1 - help !
11:03AM 0 Grandstream BT101 stops ringing
10:58AM 0 PRI Failover Success?
10:49AM 0 chan_sip2 broken with FWD
10:42AM 0 PRI D-channel signalling error? "Ring reques ted onchannel 0/1 a lready in use on span 1. Hanging up owner."
10:32AM 4 Cisco 3620 PRI and Asterisk
10:19AM 3 X100P Unstable.
10:01AM 1 Call waiting does not ring phone
9:40AM 0 Asterisk - SIP to H.323 translation
8:33AM 1 E1 in Iran
7:56AM 2 Strange Quality problems with Asterisk, Gentoo, Redhat and Kernels - /dev/dsp
7:54AM 0 Compiling Zaptel on Suse ES 9.0 and torisa module errors
7:29AM 0 stateful vs. stateless
7:21AM 7 Credit Card machines / interop
7:09AM 0 Ang: Re: Dutch (DTMF) caller-ID
6:44AM 5 music on transfer
5:43AM 0 Anyone uses Lucent i2021 BRI Phone with Asterisk.
5:17AM 2 secure
4:01AM 4 * and Fax
3:50AM 1 Call Forward, DND and other standard features
3:03AM 0 sound dropouts during SIP re-register
2:38AM 1 Help termcap error
1:17AM 1 iax connection and 1 way distortion
12:56AM 4 No Sound between H.323 Gateway & Asterisk
12:40AM 0 PRI D-channel signalling error? "Ring requestedonchannel 0/1 a lready in use on span 1. Hanging up owner."
Tuesday September 28 2004
11:46PM 1 too many ex-(boy|girl)friends
11:18PM 2 Nat Traversal help!
11:18PM 2 GSM phones, bluetooth and general hapiness
9:55PM 1 ASTCC : card generation problem
9:42PM 1 Newbie 2 PBX VOIP, protocol ?'s using Cisco 827 7910
8:38PM 1 CIsco Gateway recommendation
8:19PM 4 Gatekeeper registration failed
8:01PM 2 7960 upgrade
6:12PM 0 TXfax: return code
5:53PM 1 binding to two IPs among five
5:26PM 1 ASTERISK loosing it.. stops responding to commands on TERMINAL
5:00PM 0 Understanding codecs and transcoder
4:44PM 2 Speex config
3:49PM 0 Leader IP10S
3:39PM 3 Retrieve voice mail message from outside
3:09PM 1 asterisks queues with static members
2:41PM 1 Help with Call Waiting!
2:31PM 20 Polycom IP500
2:27PM 0 Intel Chipset MD-3200. | Works as 100XP- Clone. :)
2:18PM 1 Help with fax dial plan
2:08PM 0 PRI hiding caller id
2:02PM 1 Codecs and negotiations
1:42PM 0 Asterisk and other codecs
1:00PM 2 PRI D-channel signalling error? "Ring requested on channel 0/1 a lready in use on span 1. Hanging up owner."
12:30PM 0 Creating native bridge takes long time
11:27AM 0 Subscribe 403 forbidden
10:46AM 1 Sep 28 17:52:28 WARNING[163850]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call 2bda648025dbf8c52fd293515d98d2c2@ for seqno 102 (Non-critical Request)
10:14AM 1 IAX softphone issues
10:06AM 3 CODECs and sip.conf and voice quality
10:03AM 1 Voicepulse quality problems
10:00AM 1 Looking for whoever wrote cdr_mysql
9:53AM 1 Is app_icd ready to replace app_queue?
9:11AM 1 Fast AGI
9:03AM 1 cisco 7960 7.1 -> 7.2 upgrade problem
8:43AM 0 xc-ast public beta
8:37AM 2 Variable Codecs Order
8:18AM 1 ZT_CHANCONFIG failed on channel 1
8:13AM 1 chan_oh323 and DTMF
7:32AM 1 CDR and transfer
7:14AM 0 SpanDSP mail to fax
7:07AM 0 FW: FXO question
7:01AM 3 building a phone recorder
6:43AM 0 custom emailbody per any voicemail (mailbox)
6:38AM 0 HELP with IAXy provisioning........... newbie
5:48AM 2 SMDI Bounty - where?
5:45AM 0 H323 dropping connections
5:21AM 7 UK (British Telecom) Caller ID again
3:52AM 1 CAPI channels
3:34AM 2 Asterisk, Hylafax and T38Modem - help!
2:46AM 0 srv records
12:11AM 2 Unable to open pseudo channel
Monday September 27 2004
10:42PM 1 Peer Review - Linuxfest Presentation Outline
10:17PM 1 (no subject)
10:05PM 1 asterisk with subnet 172.16.x.x
9:05PM 11 sipura over heat
8:20PM 2 Asterisk 1.0.0 rpms available
7:11PM 0 Cisco 7940 -60 firmware upgrades
7:07PM 1 Fedora2 and zaptel - using the udev
5:45PM 0 Predictive dialing and ICD
4:53PM 0 ASTCC with oh323
4:47PM 3 CDW Part# for Cisco Software upgrade contract
3:40PM 0 recording a meet-me - only my voice?
3:14PM 3 Mutliple Line 'Buttons'
1:52PM 0 Re: Asterisk-Users Digest, Vol 2, Issue 281
1:49PM 2 BudgeTone 100 & Call Transfer
1:23PM 5 Sending DTMF after recording new voicemail
12:48PM 0 Aastra/Sayson 480i
12:45PM 0 Passing DIALSTATUS between servers
12:35PM 2 Pass-thru port on X100P Clone
12:23PM 1 FC1 and FC2 RPMs now available
12:02PM 3 Agent Call Back LOGOUT?
11:56AM 3 Simple question
11:50AM 0 Flashing a POTS Line when you are on a call in Asterisk
11:21AM 1 New to Asterisk, questions about IVR and MySQL integration
11:07AM 0 Incomming calls are fine. No sound on outbound.
10:52AM 7 X100P knock-off price jump
10:04AM 2 How to revert back to autoattendant from a voicemail box?
10:00AM 2 Cisco Downloads --> was --> Re: Cisco 7960 andAsterisk...not working...
9:51AM 0 SNMP instrumentation and/or talk path health monitoring?
9:31AM 1 Beeping on messages and calls
9:14AM 0 Re: Complete newbie seeks start
9:11AM 0 spandsp question
8:40AM 1 Cisco Downloads --> was --> Re: Cisco 7960 andAsterisk...not working...
8:21AM 1 Manager QueueAdd
7:51AM 6 FXO question
7:48AM 1 Call Center Reporting Tools
7:28AM 1 PRI fields
7:17AM 8 Complete newbie seeks start . . .
6:58AM 0 Non-PRI T1 configuration - Asterisk-Users Digest, Vol 2, Issue 263
6:39AM 1 G729 Private Licensing ??
6:23AM 3 chan_capi, Eicon Diva server BRI, kernel 2.6?
6:20AM 1 Prepaid function with MySQL
5:54AM 2 stable OS
5:44AM 1 Dutch (DTMF) caller-ID
5:31AM 0 Faxdetect does not detect certain Fax machines
5:29AM 2 Brand New to List, requesting assistance
5:27AM 0 New Chipset from TI for VoIP with 802.11g/b
5:23AM 0 chan_sip.c 183 / 180 handling, unexpected results & playtone bug ...
5:23AM 1 ASTCC installation problems
5:00AM 3 TDM400 FXO outgoing call only
4:53AM 1 VoIP South Africa
4:33AM 1 Cisco IP phone G.723
2:47AM 1 IP phone programming,
2:29AM 0 Asttapi - Version 0.04
2:28AM 1 GS 101/102 Reboot
2:17AM 9 Question
1:47AM 1 make update and upgrade failed with `ZT_EVEN T_POLARITY' undeclared
12:57AM 1 music on hold file
12:51AM 0 ISDN line
12:20AM 0 Speex/ILBC buggy with * 1.0 and X-Lite/Pro?
12:15AM 3 Asterisk Compile error
Sunday September 26 2004
11:22PM 1 FWD's Ed Guy notes from Astricon
11:17PM 0 is it possible to 'Auto Call Back' in Asterisk ??
9:53PM 0 Error Compiling libunicall for MFC/R2 with spandsp
9:35PM 1 pri to voip
8:38PM 1 ASTCC Terms : Help
7:24PM 1 Autodial on off-hook?
7:01PM 0 RE: What about [TCL as ] a higher level configuration language?
6:29PM 1 Strange problem: can't fit into subject
6:06PM 1 Background call forwarding?
4:02PM 1 H323 with Tenor CMS Gateway
3:45PM 2 spandsp with TDM fxo card?
2:38PM 6 SIP Registration Timeout, No FW
2:17PM 0 zqaptel and hdlc
11:35AM 0 iLBC modes
11:17AM 6 Digium and mailing lists
11:05AM 1 How Do I configure zaptel for PRI in
11:04AM 4 IP Phones ?
9:19AM 6 Looking for a commercial version of an IAX2 Softphone
8:31AM 2 Proper Syntax
7:42AM 1 spandsp patch help
6:52AM 0 HELP with IAXy provisioning
6:24AM 1 voicemail /w asterisk - voicemail() problems
6:03AM 3 What about a higher level configuration language
5:08AM 1 pthread problem
1:55AM 0 Got SIP response 400 "Bad Request" ; Cisco 7940 inbound station/station call problem.
12:41AM 2 Asterisk <-> WellGate 3502a : ulaw/alaw only?
Saturday September 25 2004
11:39PM 0 Digits being dropping when dialing from certain analog phones
7:50PM 2 Cisco Downloads --> was --> Re: Cisco 7960 and Asterisk...not working...
4:46PM 0 Cisco Soft IP Phone
4:24PM 2 * works, but after a few seconds audio always stops.
2:53PM 0 Dropping numbers on dialout through tdm400p
1:32PM 1 German Termination and DIDs
1:15PM 1 Application almost there..Dialplan challenges
12:24PM 1 How can I dial one unbusy channel of 4 available?
11:47AM 0 Ring delay
11:40AM 2 Cisco 7960 and Asterisk...not working...
11:34AM 3 Queue and Agent functionality
11:01AM 3 Non-PRI T1 configuration
8:16AM 4 Cisco PIX and Asterisk
8:02AM 1 Whoa.... I'm owned but found ??
7:50AM 0 Codecs Problem?
7:09AM 0 getting variable using agi
7:00AM 1 Astricon Developers Conference Recordings
6:28AM 3 Help with dialing out with TDM400P
6:18AM 1 ilbc problem
5:31AM 4 Absolutely minimal Asterisk PSTN gateway
5:24AM 0 G.729 and Asterisk intellectual property issues
5:03AM 2 Asterisk 1.0 & Zaptel 1.0 -- False Hangup Disaster
5:01AM 0 Put Asterisk 1.0 mirrors into the Wiki
3:48AM 1 Only Accept Call After Pressing a Key '#' or '*'
3:43AM 1 chan_capi install problem
2:58AM 1 TDM400P Newbie configuration hell :-)
1:12AM 3 Debian Sarge, ISDN, CAPI and Asterisk blues
12:19AM 2 How to get Call Details Records
Friday September 24 2004
9:48PM 1 _use_ast_pthread_create_instead_
9:22PM 0 getting dtmf in between an active conversation
9:01PM 0 PAP2 vs. PAP2-NA
8:16PM 2 What type of PRI setup is best
5:52PM 0 Astrison - many thanks - lots of enthusiams
5:10PM 0 Re: Setting [rx/tx]gain for spandsp/fax
4:29PM 1 Support
2:59PM 0 Two questions for Asterisk setup (Definity G3R and NFAS Trunk Gro ups)
2:56PM 3 ISDN (point to point) questions
2:51PM 2 CTI development
2:44PM 2 Call Groups
2:11PM 1 help with skinny
2:05PM 0 Asterisk MySQL CDR - Destination Number
1:10PM 2 Re: [Asterisk-Dev] Free G.729 ready for download
11:51AM 0 Intel IPP licensing and G.729
11:28AM 1 Cisco Support Agreements
11:27AM 0 Calling to Broadvoice via Linux MASQ (NAT)
11:18AM 2 kernel: Power alarm on module 1, resetting!
11:01AM 0 Re: Thank you Mr. Mark Spencer and Asterisk
10:53AM 2 1.0 Libs
10:50AM 3 SMP support
10:12AM 2 Digium Closed Today?
10:05AM 0 Verso Call Manager
9:50AM 2 Free G.729 ready for download
9:38AM 0 app_queue
9:31AM 0 SER -- Asterisk , RTP Question.
8:53AM 5 TDM channel shows Offhook when I plug it to the telco
8:42AM 1 Cisco SIP Files
8:09AM 2 dev meeting bridge
8:06AM 5 Local Outbound Calls on PRI
8:00AM 0 how to put extension on hold? using h323 phones and gnu gatekeeper
7:52AM 1 No sound into asterisk???
7:39AM 0 Asterisk skinny or sccp as softphone
6:51AM 0 How to transfer a call before the called party to answer
6:42AM 0 is the feature list online somewhere?
6:38AM 2 Asterisk as PSTN gateway
6:27AM 0 SIP - how does * decide codec order of preference
6:25AM 0 Fax Status
5:17AM 0 Fw: latest cvs / spandsp
5:12AM 0 latest cvs / spandsp
4:13AM 0 Cisco Phone 7960
4:11AM 2 Case studies for 120 simultaneous calls on IVR
4:00AM 1 Call redirect with *
1:54AM 1 dynamic config
Thursday September 23 2004
10:18PM 0 7960 Backlight project status?
9:57PM 0 problme with astcc
8:48PM 2 CallerID on Channelized T1 not working with 1.0.0
7:54PM 2 How to set up a server compatible with Windows apps ?
4:57PM 1 IAXTel and Telesthetic
3:18PM 0 Duplicated INVITE in SIP session?
2:15PM 5 Billing Fun - anybody know where to get a NPA/NXX db?
1:51PM 2 viewing fax tiffs?
1:37PM 1 running 1.0 on macosx
1:33PM 1 Cisco 7960G, SIP, NAT, Qualify and Unreachable
1:05PM 1 Alternate MP3 Player
12:56PM 4 Asterisk 1.0 RPMS RH73 and RH9
12:53PM 1 Diva Server PRI/E1-30
12:42PM 1 new user From Canada
12:07PM 1 CDR (cdr_odbc)
11:41AM 2 New Mirror
10:40AM 2 Random Intermittent Noise for SIP to FX0 calls plus echo
10:10AM 8 GSM phones, bluetooth and general happiness
9:22AM 0 asterisk-1.0.0 woes
9:20AM 1 TDM400P FXO and Primus TalkBroadBand
9:12AM 1 send Flash via FXO
8:07AM 0 Asterisk 1.0.0 Mirror
7:57AM 3 app_valetparking / parking in general
7:40AM 0 Re: [Asterisk-Dev] Softphone for PocketPC or iPaq
7:28AM 12 Asterisk 1.0 released
7:21AM 10 MFC/R2
7:21AM 11 1.0 Mirrors
6:51AM 3 Help with strategy for echo cancellation.
6:40AM 0 Monitor w/ m flag - Doesn't mux in some cases - Advice?
6:37AM 3 eyebeam
5:32AM 1 PRI(E1) Call recording with Digium cards?
5:28AM 3 GnomeMeeting and h323
4:30AM 2 Cisco 2610XM and Asterisk
3:52AM 0 Redirecting incoming PRI to PSTN
3:22AM 0 MusicOnHold and Mp3 threads
2:22AM 2 Modem[i4l]/ttyI0 sent into invalid extension 's'
2:03AM 1 I can't solve mi problem compiling CAPI, please help
1:59AM 1 video via IAX or SIP
1:56AM 0 (no subject)
1:14AM 0 Eicon Diva PCI 2.02 (not server)
12:33AM 0 Cisco 30 VIP
12:07AM 0 Asterisk and Siemens HiPath
12:06AM 0 RE: An old problem still hanging around?
Wednesday September 22 2004
8:28PM 1 News From Astricon
8:03PM 4 Softphone for PocketPC or iPaq
7:33PM 7 Some photos from Astricon 2004
6:55PM 1 TDM400 synch issue
4:10PM 1 (euro)ISDN: complete silence / can't hear a word.
3:36PM 3 American vs English
3:00PM 2 SIP soft phones
2:46PM 1 T100P PRI Problem
2:36PM 4 zaphfc NT-mode can't dial outgoing
2:25PM 3 Galaxy Voice changed their SIP proxy
2:06PM 0 How an application could dialog with an external ivr ?
12:44PM 1 7960 SIP 7.2 keypress (not DTMF) problem
12:28PM 1 TE405P hardware question
11:05AM 1 IAX Config
10:28AM 0 make update and upgrade failed with `ZT_EVENT_POLARITY' undeclared
10:15AM 2 Problems compiling CAPI
9:37AM 2 MS SQL
9:16AM 0 SIP Clients Dont Clear
9:07AM 0 app_queue & cisco transfer
9:05AM 0 Siemens Optipoint 400 and Voice Mail
8:44AM 2 Cisco IP phone
8:32AM 1 'asterisk' displayed on my Cisco 7960 & 7912 ...
7:37AM 0 Re: Asterisk-Users Digest, Vol 2, Issue 216
7:20AM 2 Transfering incoming calls using same line
7:18AM 4 PRI messages while running
7:14AM 8 Digium Hardware
7:07AM 0 IBM releases speech code as open source
7:06AM 18 Linksys PAP2-NA
6:08AM 1 OT: Hardware solutions to tie two offices together
5:27AM 1 Help: global (India/US) connection too expansive
5:06AM 1 'asterisk' displayed on my Cisco 7960 & 7912...
4:52AM 0 No Echo Cancellation with echocancel=yes
3:45AM 0 The previous reload command didn't finish yet
2:46AM 1 Opteron vs Xeon?
2:05AM 2 Grandstream bin cfg.txt generator
1:40AM 0 bug 2462
1:06AM 1 Status of conference calls at Astricon ?
12:13AM 3 Optus Australia Multiline SHDSL service
Tuesday September 21 2004
7:23PM 1 Asterisk , ISA Firewall/VPN , STUN and other
6:03PM 3 Uniden uip200
5:40PM 2 Asterisk(OS X) & X-Lite
5:20PM 0 Astguiclient problems
4:46PM 1 iax2 notransfer=yes ignored
4:43PM 0 No call progress from * to E1
2:58PM 3 Cisco 7905G
2:53PM 0 Asterisk + GnuGK :::: Unreachable Destination.
2:38PM 2 Asterisk , ISA Firewall/VPN , STUN and other issues
2:01PM 1 More than one OH323 Gatekeeper Registration
1:40PM 12 Astricon pictures
1:35PM 1 Cisco 7940/7960 and voicemailmain not able to press keys after a hold.
1:33PM 0 SIP Phone dropping calls, SIP Softphones working fine
1:13PM 1 asterisk voip only solution
12:59PM 0 Chan_capi: using both b-channels
12:26PM 1 Astricon meets?
12:13PM 1 Auto Attendant How To ?
11:19AM 0 Sayson/Aastra PT390 in Canada
11:16AM 1 Cisco 7905/7912 SIP image location (on Cisco's site)
11:00AM 1 HELP on AVM Fritz with CAPI drivers for SMP RH 9
10:43AM 1 IP phones AT-723 or AT-323
10:20AM 0 Zyxel P2000W or WiSIP with asterisk?
10:16AM 1 Polycom IP500 problem updating bootrom
10:15AM 0 Sanity Check --Zapras With T-1
10:05AM 2 Anti Ex-Girlfriend feature for entire area codes?
8:53AM 0 ZAP problem / Strange State
8:53AM 0 Queue position and thankyou message plays even when queue is empty?
8:44AM 0 New astGUIclient version released 1.0.4
8:42AM 0 T100P lost D channel
8:23AM 3 chan_sccp/SEP<mac>.cnf.xml
8:17AM 1 sipura registration problem
8:16AM 1 Need Help !!
8:03AM 0 more on spandsp and partially received fax
7:56AM 4 Voicemail forward to a remote server?
7:41AM 2 RC1 still broken with Cisco 7960?
7:30AM 0 Segmentation Fault TDM22B & TDM04B
6:53AM 0 Queues & Transfers
6:52AM 1 RDSI vs Analogic
6:28AM 0 Question/Future Request for Call Queues
6:22AM 2 Can someone suggest
6:01AM 2 Basic ISDN Access
5:32AM 2 ISDN problem: lacking dialtone
5:20AM 2 SIP termination in Brazil
5:20AM 3 FreeBSD 100% cpu
4:28AM 1 zaptelrtc for 2.6.x
1:55AM 2 TDM400P: RJ45 to RJ11
12:38AM 1 Faxing thru freshtel
Monday September 20 2004
11:45PM 9 panasonic KX-TD1232
10:08PM 0 how do I get R2 signalling working with a Digium
10:08PM 1 spandsp / I get only garbage in my faxes
10:01PM 0 Notes from Newbie
6:38PM 3 fax autoanswer
5:09PM 3 asterisk install in a home with regular phones and a x100p
3:33PM 0 SIP peers in MySQL Database and accountcode lack
3:15PM 0 [QUAR] How can I make a rotative board?
2:54PM 1 ZapRTC loading problems
2:46PM 3 Question about the 'fax' extension
2:22PM 4 How can I make a rotative board?
2:07PM 2 1 extension entry for multiple purposes?
1:58PM 0 MySQL Directory Patch
1:40PM 0 Connect asterisk on classic pbx withisdn card
1:35PM 1 Nortel/Bell Canada Vista-350 ADSI
1:28PM 0 Can't Dial using perl.
1:21PM 1 Configure an TDM04B & TDM22B
1:09PM 0 Error compiling astersik-oh323
12:25PM 6 Newbie has a few basic questions please.
12:17PM 1 wcfxo load problem
12:11PM 2 Voicemail Directory
10:02AM 1 spandsp core dumps asterisk receiving fax
9:54AM 1 Cannot do international dial with E1 in Germany
9:43AM 0 Call failed to go through, reason x
9:29AM 0 Spellcaster ISDN BRI card
9:11AM 0 Unable to request channel Zap/r1/...
9:04AM 2 Garbled voice on long distance calls
8:39AM 0 Installation problem; collect2: ld returned 1 exit status
8:24AM 6 SER + Asterisk
8:20AM 2 Polycom ip500 dial prob
7:54AM 0 Manager redirect action does not appear to work in some cases.
7:17AM 2 Cisco 76XX - How to ignore a call (silence ring)
7:12AM 4 spandsp / compilation errors
7:04AM 2 H.323 call problemm (no sound)
6:59AM 0 problem with dialing
6:43AM 0 connect asterisk to isdn bri connection
5:50AM 5 iax2_read: I should never be called
5:41AM 0 add iax user
4:45AM 1 can't compile chan_capi 0.3.5 under SuSE 9.0
4:37AM 6 PBX CallTransfer
4:06AM 0 virtual T38modem with hylafax over asterisk OH323 chans
3:15AM 2 Update: Welltech Wellgate 3504A registration problem
3:10AM 1 Wait()
2:46AM 0 Dialplan Configuration with MYSQL
1:37AM 2 CallerID in Queue
Sunday September 19 2004
11:41PM 1 how do I get R2 signalling working with a Digium E100P E1/PRA Card
10:04PM 1 Cisco Phones (7960, 7905G) and tdm400p with FXS for sale
9:59PM 2 Effectively using a telco Type 102 Milliwatt Test line with ztmon itor -v to set txgain/rxgain in zapata?
7:31PM 0 Timing source on SMP system - DisableRTC forzaprtc
6:46PM 0 Re:Asterisk and Red Hat 9 (Henry Devito)
5:56PM 2 passing octothorpe
5:51PM 0 PC Requirements
5:41PM 2 Setting time on ADSI phones from Asterisk
5:25PM 0 How does Asterisk interact with an h323 gateway
4:01PM 1 How To get response of command from another socket
3:30PM 7 Asterisk and Red Hat 9
2:48PM 1 Timing source on SMP system - Disable RTC forzaprtc
2:42PM 1 Using queue app with external members/destinations
2:21PM 1 Re: X100p on VIA EPIA-V
1:11PM 0 New Zealand Supplier
11:58AM 2 Timing source on SMP system - Disable RTCforzaprtc
11:13AM 2 Timing source on SMP system - Disable RTC for zaprtc
8:22AM 6 new ATA box for sale by Linksys
7:12AM 1 openh323 compile for Asterisk
5:29AM 4 X100p on VIA EPIA-V problems
2:46AM 1 Is Dual CPU machine solution for using Asterisk with other general apps (like home automation, web server, ...) in home environment ?
2:14AM 1 Dial 0 to outbound
1:12AM 1 RE: [Asterisk-Dev] Hardware details for the Digium TDM400P
1:11AM 1 vim ftplugins for asterisk?
12:53AM 1 [OT^2]: Getting at the fan on an IBM Thinkpad 600E Laptop?
12:03AM 0 problem to reach sip client which is connected to asterisk when t he call coming from another sip server.
Saturday September 18 2004
8:23PM 2 Timing source on SMP system
8:03PM 0 New MOH stream for each queue member?
7:06PM 1 Asterisk stopped answering the calls
1:54PM 1 13 sec. delay what is causing it?
1:23PM 1 NEWBIE - No Audio on ISDN BRI (Teles PCI)
12:03PM 3 uk caller id
11:41AM 2 IP Intercom's
10:56AM 2 Asterisk as an outbound call machine?
10:39AM 0 Quintum A800 and asterisk
9:28AM 1 First time asterisk installation problem
3:30AM 9 No sound
2:47AM 0 RSS Feed Added to Asterisk News Site
1:31AM 2 call center application
Friday September 17 2004
11:42PM 2 Caller ID with DTMF
10:52PM 1 Canreinvite=???
9:05PM 3 how to get caller ID
8:13PM 1 Connecting SPA-300 to Asterisk
7:53PM 1 ?
7:52PM 0 anyone can see response of a request from other connections?
5:35PM 1 Zaptel compile error - unresolved symbols
3:52PM 5 Background() command
2:50PM 1 ZAPTEL Compile Problem?
2:30PM 0 new Cepstral voice's TEST number
2:18PM 3 Medium volume 100% SIP/IAX PBX.
2:04PM 0 MySQL Voicemail and Directory Patch
1:56PM 3 MySQL Voicemail Problems
1:02PM 0 AnnounceOveride
12:51PM 3 Astricon
12:40PM 3 FC2 zaptel compile failure
12:27PM 0 using Astriskfor MGCP
11:32AM 3 Cisco 7940/7960 QOS?
11:17AM 0 Re: Asterisk forum created http://ASTERISK.XVOIP.COM
11:05AM 1 Ackcall works for sip, not for zap
10:13AM 2 AB1
10:11AM 1 No sound from IVR scripts, yet calls placed without any problem.
10:06AM 9 Asterisk forum created
10:02AM 2 New User Help
9:20AM 2 Re: Asterisk-Users Digest, Vol 2, Issue 163
9:09AM 6 Agents and Queues
9:00AM 2 Suppressing CallerID in .call files
8:30AM 8 cisco 7960 CTLSEP
8:20AM 1 Permanently logged in agents?
7:01AM 1 AGI Python Clear or Channel Failure?
6:26AM 4 SS7 E1 cards
6:13AM 2 Transferring Calls
6:08AM 1 Asterisk and Norstar 0X32 MICS
5:41AM 0 dtmfmode with kphone
5:35AM 1 let incoming callers contact a certain extension...
4:52AM 2 Error in zapata/zaptel configuration
4:21AM 1 How would you handle a fax without T.38orG.711uLaw?
3:21AM 1 AW: dial '0' for outside line and get a dialtone...
2:51AM 0 paly answering sounds
2:43AM 2 dial '0' for outside line and get a dialtone...
1:50AM 1 caller id?
1:35AM 1 Silently Wait for DTMF Input
1:25AM 0 OT: FWD Iax
12:22AM 8 English vs American voice files
12:17AM 1 Issue with TE405P and Adaptec U160 SCSI
Thursday September 16 2004
11:17PM 0 Problem in Dialing
8:56PM 0 Predictive Dialer, Web & Inbound Phone System
8:02PM 3 Creating conference calls from within Astman.
8:00PM 0 ISDN BRI termination via Cisco?
7:49PM 3 SIP Phone -> PBX Phone
7:23PM 2 FW: Polycom IP500
6:51PM 5 reverse the selection order of zap channels for outgoing calls
6:49PM 0 apologies if last message was sent multiple times...
6:03PM 1 SIP channel stuck after registration
5:00PM 0 Conf file for an Avaya 4624 ip phone
4:17PM 0 Dial command r option
4:12PM 1 Non-PRI T1 showing red
2:48PM 3 Playing GSM files
2:27PM 1 How would you handle a fax without T.38 or G.711uLaw?
2:09PM 2 H323 dialing makes Asterisk crash
1:57PM 0 No Caller Name sent from Asterisk over Natio nal or DMS100 PRI to a Norstar MICS?
12:32PM 1 Sprint PCS -> Asterisk through Digium TDM400P
11:42AM 5 Earthlink Releases SIP Based P2P File-Sharing App
11:22AM 1 Static noise and server locked when using two 4FXO tdm400p pci cards
10:36AM 1 IAX2 only asterisk scalability
10:18AM 1 Unable to dial using SIP using FWD and iConnectHere
9:41AM 1 Transfer and Release of a call out to PSTN
9:40AM 3 IAX- FAX
9:16AM 1 ID for outgoing calls from DDI (DID) line
9:06AM 0 spandsp on current cvs?
9:06AM 0 What can you do with Asterisk in Brazil following the law
9:02AM 0 call parking & forwarding
8:54AM 1 Beyond T1
8:20AM 0 Re: No Caller Name sent from Asterisk over National or DMS100?
7:20AM 2 Uniden UIP-200 Multiple line appearances
7:08AM 1 ${CONTEXT} variable
6:55AM 1 ERROR[16384]: chan_h323.c:1987 load_module: Gatekeeper registration failed
6:23AM 0 Language settings Cisco 7960
6:11AM 0 3 Way Calling on Snom Phones and Asterisk
5:58AM 0 problem connecting to icallglobe
4:23AM 2 Help with E1 configuration
4:22AM 2 Audiocodes Mediant 2000
4:17AM 2 Current bristuff error report
3:25AM 1 ZAP Hook flash / recall on active zap interface
2:48AM 2 No Caller Name sent from Asterisk over National or DMS100 PRI to a Norstar MICS?
2:13AM 0 Thoughts on Adding Locking to db.c?
1:34AM 0 H323 - Control Protocol Error (Master slave Determination)
1:27AM 1 Problems with native h323 channel on Asterisk RC2: No early audio and codec negotiation issues
1:11AM 1 quality of musiconhold...
12:43AM 0 Receiving queue urls
12:33AM 0
12:20AM 0 transfering a call
Wednesday September 15 2004
11:31PM 0 codec trouble?
10:02PM 1 Zap: busydetect & busycount
8:20PM 3 SIP Options
5:39PM 0 Re: question on VoIP setup
4:13PM 0 No Audio in Voicemail
3:10PM 4 Fax and Asterisk
2:51PM 1 Asterisk is not "picking up the phone" with a x100p card
1:43PM 1 Channel H323, RH9, OpenH323_1.12.2, pwlib_1.5.2 +GnuGK
10:38AM 1 Extension based call forwarding using capiECT
9:19AM 1 Sending IAX2 calls back to a registered client
9:05AM 0 Static Problem... Ahhh!
8:56AM 3 Cisco 79xx + asterisk + some functions Q
8:25AM 0 Asterisk and Cisco MC3810 Help needed
8:16AM 2 Results of 13 month study on reducing telemarketing calls
7:55AM 0 Question calling number
7:40AM 0 Asterisk SIP gateway --> SCCP Phone
7:23AM 1 Transfer / Music-On-Hold
7:21AM 0 design check
7:19AM 0 chan_capi and outgoing calls
7:10AM 1 voicebox
6:06AM 3 call recording and CDR "feature" discovered?
4:45AM 1 RC2 zaptel compile problem
4:39AM 1 phone line "roaming"
3:29AM 1 * and Philips IS3090 PBX
3:22AM 1 Not register
2:45AM 4 IAX to IAX connect question
2:14AM 0 IAX2 call drop
2:06AM 0 What FXO cards (1,2 or 4 channels) in Europe ?
1:43AM 1 capiHOLD and capiECT
1:02AM 0 incoming calls to a soft phone
12:44AM 0 AGI didn't get var from Asterisk?
12:32AM 3 ztdummy on Fedora Core 2
Tuesday September 14 2004
11:48PM 3 Fw: Asterisk R2 Signaling
11:26PM 4 One Question:CLI dial cmd
9:06PM 4 Sending Caller ID info in MD/USA
6:25PM 2 Press 9 to dial by name
6:13PM 2 Warn before Absolute Timeout
5:54PM 1 Polycom IP600 and instant messaging
4:57PM 0 Problem with hangup
3:04PM 2 3-way calling
2:13PM 0 Cepstral available
2:06PM 1 ERROR: cannot load module kernelcapi
12:56PM 1 Patching UK Caller ID
12:56PM 1 i4l "1 second patch", anyone got it?
12:25PM 1 cannonicalizing phone num in macro
12:16PM 1 Agents on zap channels must acknowledge calls even with ackcall=no
11:32AM 0 Chanspy updated
11:20AM 2 Spawn extension.....exited non-zero
11:10AM 1 Clarification - FAX on local network
10:52AM 1 Detecting DTMF tones
10:25AM 1 Using Asterisk as a replacement for a Merlin Legend.
10:21AM 1 asterisk does not start...
9:58AM 2 Asterisk not outputting real time display
9:53AM 0 video softphones
9:32AM 1 Openswitch12
9:30AM 1 Manager events logic depends on channel type?
8:35AM 0 MeetMe - waiting for marked user
8:00AM 0 SIP registrations CVS Head
7:57AM 0 *called* id name display?
7:48AM 1 Comparisons between * and sipXpbx (PingTel's open source product)
7:47AM 0 RE. compiling zaptel
7:27AM 0 Detecting DTMF reliably
7:20AM 0 Get Connected With Kingston A How To Guide
7:15AM 1 Setting up Asterisk with fwd
6:54AM 0 SIP call server- Too many hops
6:33AM 2 Mitel 5010 +5220
5:48AM 2 Use ISP's SIP account for IP-PSTN gateway
5:24AM 1 Cheap Sams computer good for tdm400?
5:17AM 3 OH323 Trunking
4:35AM 1 cvs stable
3:47AM 1 Newbie question: X101P card - Asterisk - /dev/dsp0
3:26AM 1 What does 'Forbidden (From header is not a Trust host or gateway)' mean?
3:05AM 1 Requested device 'ttyI1' does not exist
2:30AM 1 Wrong ID going out...
1:01AM 3 how to route these outgoing calls?
12:23AM 0 softphone crash?
Monday September 13 2004
10:37PM 0 Making the Old PABX work with new * box
9:36PM 4 asterisk make
8:56PM 1 Read command without #
8:00PM 1 agents and *8 pickupgroups
7:08PM 3 Aasterisk SIP<->SIP No audio
5:34PM 4 PABX & VOIP Gateway
4:42PM 1 chan_sip2 Install Question
4:04PM 0 Sipura-3000 Assistant for Asterisk on MacOSX? Well, maybe, with your help!
3:41PM 1 Extending E1's over a Satellite link
2:59PM 0 voicepulse problems since new configs
2:44PM 0 Codec usage in iax.conf
2:42PM 2 Sip Outbound Proxy
1:55PM 0 Arrgh, Broadvoice, SIP.conf
1:33PM 0 IAXy loud static problem
12:37PM 0 Registering asterisk with FWD
12:35PM 2 allowing/disallowing codecs in dialplan?
12:31PM 7 festival
11:43AM 0 Dialplan transfer. (h323 transfer)
11:43AM 0 Dial-plan transfer
11:09AM 1 Caller ID "forwarded" to analog phone?
10:38AM 0 WhoIsIt -- a contributed utility
10:27AM 3 Alchemy branch integration, one way audio
10:08AM 1 Server load capabilities
9:10AM 3 Astersk as AVAYA IVR
9:00AM 0 iax2 transfer and CDRs
8:50AM 0 CVS lock directory still not fixed?
8:23AM 2 Astricon tutorials :: Open for registration again
8:12AM 0 Zaprtc help
7:52AM 0 test membership
7:45AM 0 CDR database.
7:45AM 0 Post to list
7:42AM 8 Playback Fileformats
7:35AM 0 Asterisk daemon start errors
7:25AM 1 IAXy DHCP lease not renewing
7:22AM 1 Simple Caller-ID match and block and/or play voice file saying you're calling too much or don't call!
7:02AM 2 unavail and busy.
6:42AM 0 Agentlogin incorrect
6:19AM 0 IBM to Open Voice Recognition Software
4:53AM 5 music on hold not strting
4:13AM 4 Unknown RTP codec 72 received
3:40AM 1 Red Alarm - Config Zaptel card
3:39AM 1 (no subject)
2:57AM 1 ProSLIC and measuring of PSTN parameters like Voltage, Polarity, Power (A) and Frequency (Hz)
2:30AM 1 SIP Remote-Party-ID
Sunday September 12 2004
11:38PM 4 One Question
10:22PM 0 RE: No subject by Steve M
10:01PM 1 (no subject)
9:54PM 0 iconnecthere DTMF detection
8:41PM 2 New BudgeTone
7:46PM 1 IAX2 crash course wanted
5:12PM 1 detecting fax and passing it to Hylafax
3:37PM 1 Monitor and AGI - doesn't record much!
1:50PM 2 Multiple MD 3200 (Intel 537) cards on a single system.
1:27PM 1 SetGroup Limitation!!!
12:29PM 2 Overriding SIP From Header
11:35AM 2 (no subject)
10:59AM 3 Final Help on setting up x100p
9:02AM 1 TN405P running but with errors
7:28AM 2 Grandstream Budgetone 100 Caller ID shows extension, not incoming Caller ID
5:27AM 2 sipphone dial out problems..
1:54AM 0 sip does not bind all addreses
1:05AM 1 Voice from one call carried on to next call
12:18AM 2 GSM / Radio
Saturday September 11 2004
11:36PM 1 mknod /dev/phone0 c 100 0
10:08PM 2 VoIP Telephony with Asterisk by Paul Mahler
9:19PM 25 Broadvoice
8:52PM 1 Audio from GS to asterisk double speed
8:17PM 0 Problems with Call Progress and fax detection on PRI
7:41PM 2 TDMoE questions
6:51PM 1 IAX not binding to the right port
6:51PM 0 How to make a call from command line
6:34PM 1 IAXy intermittent sound problem
5:47PM 0 h.323 Transfer
5:43PM 1 creating device=/dev/phone0
4:30PM 2 Leading '0's and what do 'pri_dialplan', 'pridialplan' and 'prilocaldialplan' in zapata.conf do?
4:20PM 0 Grandstream x Asterisk 1.0 RC1 x VOIPJet
1:40PM 2 Questions about PRI lines for modem banks and Asterisk
1:13PM 1 Compilation error with 2.6 kernel
11:44AM 1 Final status of the call
11:40AM 2 Help!!!!!
11:39AM 2 Audio level in compressed wav files
10:41AM 0 Spandsp garbage
9:29AM 3 FWD
9:00AM 0 DTMF signaling with GSM codec
5:19AM 1 call park question
2:46AM 0 call forwarding when busy - single pots blues
2:30AM 0 Call Queues, CallerID, SIP and AutoDial
12:57AM 1 Questions about cdr
Friday September 10 2004
7:47PM 2 Suggested Motherboard for TE410P
6:57PM 1 (Resend) Trouble with all linux sip softphones.... And asterisk/linphone/kphone SRPMs
5:39PM 0 chan_agent and SIP UA transfers fail
5:03PM 1 Can't get ChanSpy to work
3:21PM 4 SIP on Handhelds
3:03PM 8 Organization wide
2:05PM 0 Definity <-> Asterisk w/callerid
1:53PM 0 Proposal regarding the "*80" vertical service code
1:41PM 0 RDNIS and Q.931
1:06PM 1 moh cell phones
1:04PM 0 SIP Dropped Calls
12:53PM 2 What would be required for this?
12:46PM 1 Sangoma S508 Rev-B
12:09PM 2 SpanDSP/RxFax anomalies...
11:37AM 3 call quality monitoring
10:44AM 1 Net2Phone, Asterisk, and "404 Not Found"
10:21AM 0 Re: Problem with Openh323 channel driver
9:46AM 0 Re: Asterisk-Users Digest, Vol 2, Issue 94
8:58AM 1 Problem with stuttering on TE410P
7:27AM 1 Call Parking Problem
7:21AM 0 pridialplan & nationalprefix
5:38AM 14 Asterisk newbie questions
5:32AM 1 No DTMF or Audio
4:55AM 1 Problems with 0penh323 Channel Driver
4:50AM 4 sip.conf from mysql
3:35AM 2 Asterisk and VoDSL
1:58AM 1 Netmeeting i can't hear voice
1:26AM 2 Snom 200 updates
1:06AM 3 Asterisk testbed for teaching connecting to a PRI-ISDN
Thursday September 9 2004
11:09PM 0 Asterisk server keeps crashing
7:54PM 3 Store data from call to database
5:57PM 1 DevKit TDM400P module won't load
2:29PM 1 Debian Sarge -- cvs vs. apt-get
2:23PM 1 Unknown IE 40 (cs6, Unknown Information Element)
2:13PM 2 Conference Phone
2:06PM 1 Uniden UIP 200
2:04PM 0 What are ZOMBIES and why am I getting them?
1:27PM 0 two asterisk boxes and using outgoing spool file on second box to call out on first box.
1:19PM 1 astcc not working
1:07PM 4 IAX2 dropping call?
12:59PM 10 Cepstral
12:55PM 0 rate_engine substitue db field?
12:40PM 0 Polycon IP 300 SIP vs Grandstream BT-101Deployment
12:34PM 1 UIP-200 conference call
12:06PM 1 Festival Speech Synthesis 1.95:beta July 2004 Eval
12:04PM 0 Astricon News :: Tutorials are now fully booked
11:29AM 12 SNOM 200 can't conference.
11:16AM 3 Caller-ID name lookup via
11:10AM 2 Dial Out w/ OH323
10:36AM 0 Dialing Out through Provider with Authentica tion
10:11AM 0 OT: how much are polycom phones in the UK
9:56AM 2 Legacy Toshiba Phones
9:55AM 0 Aruba Origination
9:53AM 3 Polycom IP500 vs Cisco 7940
9:34AM 0 Re: Asterisk-Users Digest, Vol 1, Issue 5082
9:33AM 1 Virtual queue member
9:26AM 0 Telcordia TBCT
8:53AM 3 Simple question about SIP community
8:15AM 0 Asterisk not playing sounds after Kernel upgrade?
7:48AM 2 Fax relaying with T.38
6:38AM 0 Queues : Rings even when the agent is on a call
6:20AM 3 Dialing Out through Provider with Authentication
6:16AM 3 weird routing(?) problem with 2 Asterisk servers
4:53AM 0 ser+ asterisk
3:44AM 0 russian sound files
2:30AM 1 Problems to setup ast_data with asterisk.
2:04AM 0 zaphfc errors
1:28AM 0 Ordinary phones can call into asterisk - but * does not recognize the dtmf signals
1:09AM 1 Dialing pstn-asterisk
Wednesday September 8 2004
11:50PM 0 transfer on a zaptel FXO port
11:26PM 0 Asterisk & the Micronet SP5210 anyone?
6:51PM 2 Zaptel and Linux Distros
6:01PM 1 New ChanSpy and MOH Patch
5:13PM 0 IAXy/S100I reomote PBX extension provision
4:40PM 0 Spontaneous Hangup occuring
3:05PM 0 T100P calls with playback starts speaking be fore pickup
2:35PM 0 Disa extension entry timeout
2:29PM 4 Cisco GW and DTMF problems
2:18PM 0 T100P calls with playback starts speaking before pickup
1:03PM 2 My AGI is not detecting hangups on outgoing calls
12:42PM 2 'Hangup' not hanging-up, is this intended behaviour?
12:40PM 0 Changed * server to static non-nat IP from nat
12:39PM 2 How do I get DIDs for remote areas in Canada
12:35PM 1 successful echo cancellation!!! (multitech)
11:44AM 0 Driving MWI on Norstars (was Maximum tollera ble lag/jitter...)
11:38AM 1 Polycom SIP 1.3.1 & Reject Button
11:28AM 2 Asterisk with Primus Talkbroadband
11:06AM 1 zap: reroute incoming calls to dedicated channel
10:53AM 1 Intertex IX66
10:18AM 0 stale voicemail messages / greeting
8:40AM 0 Directory command assistance
8:14AM 1 OH323 Ignoring PROGRESS indication
7:50AM 2 PRI issue
7:41AM 3 Where to post SuSE 9.x startup script?
7:31AM 1 Polycon IP 300 SIP vs Grandstream BT-101 Deployment
6:28AM 2 Help needed!
6:21AM 1 Problem playing file with G729A
6:14AM 1 accept DTMF while beeing in a queue
5:55AM 0 asterisk+chan_h323+redhat9 troubles
5:29AM 4 WellGate 3504A with Asterisk SIP authentication and config
5:16AM 1 SIP and */#
4:54AM 1 asterisk console from xinetd?
4:10AM 3 sendmail&hostname
3:37AM 0 zaphfc strange errors
3:27AM 1 Assigning a higher irq to a digium card
2:04AM 0 re: asterisk, SER and autocreatepeer
1:43AM 2 X-Lite & Meetme problem
1:31AM 3 Newbie: Only allow authenticated users to call
1:08AM 2 'connecting' voip-numbers to our Asterisk
12:20AM 2 Answer confirmation on non-Zap channels?
12:14AM 3 astwind has any one got this thing to work?
Tuesday September 7 2004
9:50PM 0 Monitored outbound dialing via Zap interface ?
9:16PM 1 QSIG against a Nortel/Meridian PBX
8:43PM 4 Caller id and the number of rings
8:41PM 1 Got *80 working ... now some Blacklist questions
8:12PM 6 Problems with length of voicemail
6:44PM 1 astcc dont write to the table cdrs or cards
5:55PM 2 Maximum tollerable lag/jitter for IAX2 w/o j itterbuffer enabled?
5:48PM 1 REPOSTED: Problems patching Makefile in apps directory
5:04PM 1 Compiling on Mac OS X (10.3.5)
4:54PM 4 Maximum tollerable lag/jitter for IAX2 w/oji tterbuffer enabled?
4:26PM 3 Maximum tollerable lag/jitter for IAX2 w/o jitterbuffer enabled?
2:52PM 1 Monitored outbound dialing via Zap interface?
2:43PM 0 chan_h323: remote ip address -> context
2:36PM 0 T100P problem with LD T1
2:14PM 0 Bristuff wackyness - not answering
2:11PM 0 OH323 return call from openphone to sip?
1:48PM 1 Checking Return Codes
1:37PM 6 MySQL on another host?
1:25PM 3 DTMF Caller ID w/o polarity inversion
12:54PM 1 MeetMe without ZAP?
12:13PM 1 Tormenta & Asterisk
11:56AM 0 FWD registration sort of expiring
11:10AM 0 new Asterisk resources site
10:49AM 0 OT - Experience using Gmail for AsteriskMail ingList
10:13AM 0 Cisco 7960G SIP Registration Timeouts
9:46AM 1 MOH/mpg123 broken when running asterisk as non-root?
9:27AM 2 TE410P in Germany
9:16AM 1 Nortel PC client
8:34AM 2 OT - Experience using Gmail for Asterisk Mailing List
8:29AM 1 Performance Specs
8:18AM 0 GRQ
8:16AM 0 phones and atas timeout..force * restart
7:44AM 0 extension mobility with cisco phones
7:21AM 1 Asterisk + NetJet (ISDN4Linux)
6:43AM 0 Problems patching Makefile in apps directory
6:27AM 1 Cisco 7912 issues
6:24AM 2 Crossed lines - a worrying problem.
5:51AM 0 sending SIP Message 404 out of extension.conf
5:10AM 0 Country specificals-- Incomplete
4:57AM 0 Country specificals
4:37AM 0 Conference Call Query?
3:56AM 0 voip gateway connect to a pbx
3:45AM 3 H323 Control Protocol Error
2:11AM 0 SRV lookup fails after DNS update
1:47AM 0 [Asterisk-User] Problem with bus s0 and analog lines
12:51AM 0 problem with E100P
Monday September 6 2004
11:46PM 1 forwarding calls thru Freshtel
10:38PM 0 carrier connection options
10:35PM 3 iaxy vs sipura
10:01PM 0 Zaptel errors with E100P + TDM40B
9:42PM 1 [patch] allow the transfer keys from app_dial's't' and 'T' and hangup key 'H' to be configured via features.conf
9:19PM 1 [patch] allow the transfer keys from app_dial's 't' and 'T' and hangup key 'H' to be configured via features.conf
6:51PM 0 IAX2/GSM VOIP troubleshooting
6:12PM 9 Zaptel 'Under the Hood' Project
5:18PM 1 Wait for Dialtone syntax in Dial cmd?
4:12PM 0 Horrible noise instead of indications
3:16PM 1 Problem Loading asterisk_oh323-0.6.3b eith last *cvs...
3:09PM 2 Codecs for fax traffic
1:29PM 6 RES: Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual.
12:26PM 0 only hear a few ring tones
10:24AM 3 multiline IP hardphone w/ FDX speakerphone?
10:15AM 3 SIP rtp port forcing
10:04AM 2 DTMF information?
9:48AM 3 VM access
8:38AM 2 Placing Asterisk between existing PBX and PSTN
7:37AM 2 spouse-friendly spa-3000 pstn interface
7:03AM 0 x-lite and pound key
5:05AM 1 UK Callerid bug #1719 & TDM400p
4:02AM 5 Newby question. Basic structure
3:45AM 0 Wildcard TE410P still making trouble
3:22AM 1 cvs server problem
1:44AM 2 Four single-port FXO Cards in one * box
1:08AM 0 SIP-Channels cannot be created after a while of running asterisk ...
1:01AM 1 Voicetronix OpenSwitch12
12:40AM 1 T.38 "pass-thru"
Sunday September 5 2004
9:28PM 5 Asterisk Conferencing using g729
3:22PM 0 DTMF with HFC-S, not supported yet?
2:52PM 4 Asterisk & sudo from httpd
2:03PM 1 res_perl
1:51PM 1 internal s0 using chancapi
11:19AM 1 Pause or Wait character in Dial command?
10:49AM 2 ZAP channell Dial timeout
10:35AM 0 iconnect and Asterisk
10:26AM 3 ChanSpy by anthm and more...
9:32AM 1 Number of digits
9:07AM 2 GRQ / RRQ
8:14AM 1 Any asterisk echo demo servers ?
7:37AM 0 My Cisco 7940 is not registering with Asterisk
7:36AM 2 FXO/FXS with T.38 over SIP
1:10AM 2 offtopic - channel banks
12:02AM 1 need help configuring dlink dvg-1120M
Saturday September 4 2004
10:25PM 5 Wildcards and variable number of digits
3:11PM 1 Oh323, Please Help Newbie ;(
1:57PM 3 Question on echo's for Canadian Asterisk users ...
12:18PM 5 Free WWT (WorldWideTelco): Utopia, or just a matter of organization?
11:45AM 0 Terminating Multiple incoming calls on *
11:18AM 3 Help Running Perl/CGI on Apache Server
10:13AM 0 Wall-mounting UIP 200 and SoundPoint IP600 keepscoming off hook
8:42AM 0 current cvs - zap failure with tdm?
6:53AM 0 PBX --- > Asterisk Connection
5:43AM 1 call back on failed transfer or dial?
4:35AM 8 Linux distribution
2:44AM 0 Which proto. Is more stable/less resource demanding for Asterisk, SIP or H323?
2:12AM 1 How do you avoid or reduce false hangups on X100P?
1:34AM 1 SIP/IAX2 phones with builtin magnetic stripe reader
Friday September 3 2004
6:04PM 0 D/600JCT-2E1
5:14PM 1 Voicemail Size on Disk
4:13PM 3 Putting a call on hold
2:41PM 2 Wall-mounting UIP 200 and SoundPoint IP600 keeps coming off hook
2:21PM 0 Re: Re:New to *
1:55PM 0 Lower cost router suitable for VoIP?
1:37PM 0 Rejecting Calls in Cisco 7960 --
1:16PM 2 Using AVM Fritz!PCI as zap interface
12:02PM 3 Help setting 2 Offices in US and India
11:19AM 2 X100P blows up after a while (really loud noise)
10:42AM 0 Sending multi-line sms text
10:26AM 0 Slow Robotic or like underwater voice
9:22AM 7 Dropping incompatible voice frame
8:37AM 0 Call Parking with Queues
8:21AM 0 Dlink Video Phone & Asterisk
8:07AM 1 BIG ISSUE with SIP, not sure where to go but it's killing asterisk.
7:04AM 2 mpg123 - multiple instances, taxing CPU
6:38AM 0 I forgot to add my email please contact me offline we have around 300, 000 to 1/2 million minutes per month for India and Pakistan .. can ztdummy help trunk mode?
6:17AM 1 RES: Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual
5:49AM 1 SIP / Keep alive...
3:59AM 0 busy signalling on PRI doesn't work...
3:31AM 5 Digium E100P and PMX in Germany
2:35AM 1 one doubt
2:30AM 5 Lower cost router suitable for VOIP ?
1:53AM 0 RC2 with OH323 or H323
1:27AM 2 OH323 0.6.3b compilation problem with 1.0 RC2 on RH9
12:20AM 1 zap barge restrictions
12:13AM 0 Kphone Can't register to ser via Asterisk
Thursday September 2 2004
11:37PM 3 digitnetworks card issues?
11:07PM 1 Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual?
10:24PM 0 Message waiting / stutter dialtone ?
8:40PM 1 GSM codec bandwidth
7:42PM 5 Any way to _always_ execute certain commands in a dialplan context?
7:33PM 2 Sorry, Newbie here
6:46PM 5 Polycom SIP INFO & Changing Ringers
6:03PM 0 diff between SNOM & Asterisk
5:53PM 0 Quad E1 card that support R2 signaling
5:27PM 0 Weird CallerID question
2:24PM 1 Analogue call answer detection
2:16PM 1 WG: Digum TE410P
2:05PM 0 MeetMe- on demand recording
1:21PM 2 Phone numbers for testing
1:20PM 2 Polycom Microbrowser
11:27AM 1 Hard Ground (On Ring)
11:20AM 2 uniden Uip300
11:16AM 0 GUI VoiceMail directory question:
11:06AM 0 Audio dropouts w * and 7960's
10:58AM 0 line feedback, no dial tone
10:43AM 0 UIP300
9:47AM 1 Incomming ring on POTS line kills ongoing voip call?
9:15AM 2 Please help to config tdm11b
8:52AM 0 ROBO-8712VLA SBC
7:53AM 0 isdn, pbx and *
7:40AM 1 Asterisk + ISDN BRI - gateway or card?
7:06AM 1 call back on failed transfer?
6:27AM 0 Problem with supply of pin number from SJPhone
6:03AM 1 Commercial CID spoofing system
5:51AM 1 BRI&DDI
5:24AM 2 ${CALLERID}
5:05AM 2 How let SIP clients connect directly?
5:04AM 1 no dial tone when dialing out on vonage
4:51AM 1 Problem with HasNewVoicemail()
4:01AM 0 Webmin module.
3:30AM 1 asterisk config and root
3:16AM 1 why do i get this message emailed to me everytime i post?
3:10AM 1 Any UK PipeCall/PipeMedia users?
2:08AM 3 BT Easicom - Andy Powell
1:46AM 0 oh323 <> sip
1:43AM 0 X-Lite from Home
12:32AM 1 voicemail email problem
12:21AM 2 Searchable Archives?
Wednesday September 1 2004
11:38PM 2 Hung SIP channels
10:57PM 3 Distinctive rings
10:47PM 0 Meetme delay issue
10:35PM 0 Audio Delay in Meetme
9:57PM 0 Whats the '411' on echo cancellation?
9:33PM 1 Odd PRI Behavior
7:43PM 1 HFC cards and Asterisk
6:56PM 4 Why are you guys promoting a Rippoff
6:00PM 1 Really Wierd softphone problem ... must read
5:28PM 0 Newbie: Asterisk Config to Replace Lucent Partner
5:08PM 1 Festival TTS & mbrola ?
3:40PM 0 Analog -> ip sip softphone on Fritz Capi - strong reverb ?
3:36PM 2 zaphfc crashes Linux
3:09PM 0 h323 - forcing user authentication
2:48PM 1 X100P + Call-Waiting - Flash how-to.
2:17PM 0 TDM40B hangup on fax or data modem carrier
12:39PM 2 Migrating Asterisk
12:31PM 0 Asterisk, newbie, fwd and is this jitter?
12:19PM 0 Using an analog modem through asterisk (zap channels)
11:49AM 1 latest CVS build won't load
11:43AM 1 Broken sound in VoiceMail
11:01AM 1 MWI light on Cisco Phones
10:59AM 2 Help Me - SIP Phones ( No Voice) !!!!
10:41AM 2 Rebooting Linux / Asterisk
10:26AM 1 FXO Disconnect supervision
10:24AM 1 Dynamic dialplan
9:13AM 1 NEWBIE: PWLIB Build Failure
8:48AM 0 Newbie - Troubles after installing e100p
8:33AM 0 CLI variable not set on incoming call
7:35AM 1 Agents Log off
7:19AM 4 Group Dial
5:28AM 5 dtmf problem
4:59AM 2 Lucent iMerge
3:07AM 0 Ring tone when busy in trunk scenario
1:07AM 1 international caller id support
1:05AM 0 has anyone the capiCD() funktion in chan_capi running?
12:20AM 6 Mitel 5010