asterisk users - Sep 2003

Tuesday September 30 2003
TimeRepliesSubject
9:20PM 0 SIP REGISTER channel leak fixed
7:09PM 1 Help with Zhone z-plex 10 reset
1:05PM 0 can't load x100p card
11:30AM 7 [Fwd: Re: Voicemail: Timestamp suddenly reverted to GMT!!]
11:29AM 0 T1(not-PRI) problem with ANI(callerID)
9:38AM 4 ADSI only works with what?
8:57AM 11 * not logging CDR to MySQL - anyway I can debug this?
8:15AM 0 x100p bridged - detect voice?
6:38AM 2 SIP Registration Difficulties
6:11AM 2 Application Flash
6:06AM 4 SPEEX bitrate?
4:56AM 0 Asterisk and Vocaltec
1:29AM 0 Missing ring indications with DPH-100H/chan_h323
 
Monday September 29 2003
TimeRepliesSubject
11:01PM 1 Voicemail: Timestamp suddenly reverted to GMT!!
10:47PM 0 Busy detect and Hangup with VoiceMail Problem
9:52PM 4 How to use vmdb.sql in voicemail.conf/extension.conf
8:47PM 0 Digium hardware distributors near Budapest, Hungary
8:40PM 0 FW: Identify the channel ID within a SPAN
8:17PM 2 Needed: Configuration Examples for VoIP Providers Asterisk can Register With
2:32PM 0 Fax detection when dialing out
2:16PM 5 Nortel M Series phones support
1:57PM 0 PRI protocols
1:50PM 15 RE: SIP i.e. Is something broken?
9:06AM 3 Is somthing broken?
5:38AM 54 Help with GPL license of Asterisk
5:17AM 4 cisco AS5300 : problem configuration
4:46AM 2 SIP Channels
2:22AM 0 RE: Asterisk list a SPAMer (uol.com.br), I t hink not ...
2:15AM 0 How to prevent echo ?
2:07AM 1 Can't place a call with MGCP Phone
1:11AM 0 Netopia SDSL modem w/ IAX2 calls...
1:10AM 1 RE: Asterisk list a SPAMer (uol.com.br), I think not ...
 
Sunday September 28 2003
TimeRepliesSubject
10:18PM 0 tdm400p + x100p config problem
8:06PM 5 FYI-New ATA clone out
5:35PM 1 Forwarding SIP over IAX problem: No One Available
4:02PM 5 Outgoing call spool
2:43PM 0 Flash and Call Transfer
2:12PM 33 Google newsgroup or Forum setup.
1:58PM 0 TE410P timing and multiple, different spans
1:56PM 0 (no subject)
1:55PM 0 Latest CVS breaks sound
12:17PM 2 Help with PHPconfig setup??
11:17AM 12 NAT/SIP solution?
10:26AM 0 "i8253 counter to high. Resetting..." message
10:19AM 0 Asterisk CVS viewer on line
 
Saturday September 27 2003
TimeRepliesSubject
8:16PM 0 More Sip/Grandstream issues
7:23PM 4 Installation counter
4:01PM 1 SIP/ Grandstream Issues
3:11PM 4 how stable is dynextendb
1:16PM 2 Continuing Budgetone woes
8:49AM 0 sipphone .com config
2:46AM 4 IAX and NAT
1:33AM 6 Budgetone + NAT: Firmware Version?
 
Friday September 26 2003
TimeRepliesSubject
8:58PM 1 Question about codecs and interoperability with cisco AS5350
8:13PM 3 Creating a SIP gateway for use behind NAT
8:12PM 5 Set context based on CID...
7:49PM 4 Configs for IAX <> IAX trunk
7:16PM 0 G729 experiences.. (fwd)
6:20PM 2 outbound IAX calls bog down DSL
3:50PM 3 RE: Asterisk license (fwd)
1:09PM 3 number detection problem.
12:55PM 1 Cisco 2600 and ASTERISK and calling out
12:41PM 0 Unable to find a path from ULAW to G723
12:28PM 0 Incomming call management
12:03PM 1 Gastman and SIP?
10:35AM 4 dialing out with the outgoing queue problem.
10:25AM 0 X100P: Can I detect/react to CLASS "you got voicemail" signals?
9:28AM 1 Config TE410P + TDM400
9:10AM 9 X-Lite for Linux
8:54AM 8 the g729 situation
7:33AM 2 X100P - Busydetect / calls being disconnected - Australia; tip.
7:16AM 2 ATM support?
7:14AM 0 difference between nufone chan_h323 and openh323 chan_oh323
7:13AM 1 Wildcard for Conferencing (VoIP)
7:02AM 0 trouble with MGCP Phone
6:44AM 0 E1 Card Signaling
6:37AM 0 chan_capi newbie questions
6:37AM 3 FCC/Euro/Aussie approvals on TE410P
6:20AM 7 An interesting call path observation..
5:49AM 9 Newbie: Crossing my fingers
5:30AM 3 RES: RTP routing..
3:25AM 0 Strange warnings from chan_zap
2:58AM 0 Voice Port to IXP425 Network Processor Board
2:36AM 8 RTP routing..
1:47AM 3 IAX calling number
12:45AM 0 Sound file script
 
Thursday September 25 2003
TimeRepliesSubject
8:17PM 0 problem with dtmf detection or ioctl with READ mask
7:57PM 4 SIP Problem
7:37PM 2 VoIP Support for Symbian OS Devices
7:07PM 4 '.' pattern and non-SIP phones
7:05PM 0 X100P not passing DTMF through?
6:00PM 0 My linux machine thinks its running windows!!! It crashes... (Problems with my x100p card.)
5:48PM 4 AGI: getting the return code from an exec()'d application?
2:43PM 1 additional ports for devkit tdm400p(tdm10b) ?
2:06PM 1 Choppy communication issue
1:34PM 0 (no subject)
1:01PM 6 SIP codecs Errors
12:49PM 0 SJPhone and Asterisk
12:48PM 2 Greetings..
12:38PM 4 ztdummy loading: unable to specify channel 1
11:48AM 0 Help installing FXS card
11:12AM 1 Grandstream and G729
10:22AM 119 CDR Web Search Frontend
9:13AM 2 VoiceMailMain skipping extension and password prompting
7:06AM 4 configuring TE410P for four E1 PRI lines
6:26AM 0 IAX ---> PRI ---> PSTN call disconnected after a few minutes
6:26AM 8 FW: RE: AntiSpam UOL
6:15AM 9 E1 in Brazil
5:57AM 0 Problems with cdr_mysl
5:10AM 2 SNOM + GSM
5:01AM 13 G729 experiences..
4:50AM 2 Cannot compile channel h323 from yesterday's CVS?
3:52AM 0 AntiSpam UOL [andersoncbr.sspam@uol.com.br]
3:40AM 1 Sometimes pri channels restart during * is runnig ?
1:30AM 1 sipphone.com configuration
12:43AM 10 Meetme question
12:36AM 0 Dlink 104S
12:34AM 0 SIP problem with asterisk
 
Wednesday September 24 2003
TimeRepliesSubject
11:55PM 4 Starting Development Perl or Python
11:33PM 2 best low-bandwidth strategy
11:10PM 0 Group pickup codes, etc.
7:27PM 8 Purchasing Grandstream Phones
6:15PM 1 Voicemail doesn't hangup
5:39PM 0 Removal of anti-spam responder
5:04PM 0 (no subject)
4:56PM 6 echo for 15 seconds
4:54PM 0 help asterisk call waiting X100P -> MGCP ata 186
2:00PM 3 Dlink DG-104S (chan_mgcp) and configuration w/Asterisk
1:53PM 2 Chan_capi accountcode.. (repost)
1:50PM 0 # No ring tone while dialing out with AVM PCI2.0
1:36PM 0 Re: Asterisk-Users digest, Vol 1 #1380 - 15 msgs
1:14PM 1 Packet8 sans DTA310
1:09PM 0 More on"Callprogress"
12:17PM 2 No ring tone while dialing out with AVM PCI2.0
11:23AM 8 RedHat 9.0 and 100 percent CPU utilization
10:43AM 5 Prebuilt Asterisk
10:41AM 17 VIA vs Intel
10:13AM 0 Adding a DELAY to an ADSI script
8:01AM 14 Check and restart script..
7:55AM 2 Snom 200 errors?
7:53AM 5 Call transfert with dial plan
7:50AM 0 netconsole - bad file descriptor?
6:47AM 3 X100P incoming calls - "hangup" delay
6:45AM 23 Cisco 2600 and ASTERISK
6:35AM 27 SIP / GrandStream Configuration
6:22AM 12 Festival Problems
6:01AM 2 Using Asterisk in an netted scenario
3:51AM 2 CPU Optimisations For asterisk
3:24AM 10 Does SIP work?
3:03AM 2 Meridian Option 11 and asterisk
1:28AM 7 list of voice prompts
 
Tuesday September 23 2003
TimeRepliesSubject
11:05PM 4 festival problem
10:45PM 0 LIST MOM / DAD PLEASE READ [andersoncbr.sspam@uol.com.br:......]
10:16PM 2 initial review of Grandstream HT-286 ATA device
6:54PM 1 PROBLEMS WITH IAXATEL AND DIGIUM IAX
4:20PM 0 Example callback/call out AGI script
4:17PM 0 Grandstream HT-286 review on the way
3:46PM 1 App_festival crashing
11:09AM 4 IAXTEL
10:43AM 3 Port problem
10:27AM 0 Cisco Callmanager 3.3 Asterisk OpenH323
10:04AM 9 Segmentation Fault on reload (gdb output included)
10:04AM 1 FW: asterisk call waiting X100P -> MGCP ata 186
9:40AM 1 Windows Media Player Error
8:43AM 0 TDM400P port does not operate
8:39AM 0 Cisco Sip Gateway Register with *
8:32AM 14 dialing codes..( You can help! )
8:21AM 0 Interesting product
8:20AM 1 Cisco 7960 SIP Firmware.
8:09AM 1 New Cisco "Color" Phone
7:59AM 4 Advantage of Cisco 7960 with 5.x firmware?
7:55AM 10 Dial over IAX ahngs up after 3 rings
6:30AM 1 Question about dialogic hardware
5:54AM 3 New kid on block
5:51AM 3 error message playing .mp3
5:37AM 6 dlink104S->asterisk->OH323->OpenPhone
3:08AM 0 FW: RE: iaxtel and iax.conf (HTML CONTENT, FYI)
1:53AM 0 pingtel phones
 
Monday September 22 2003
TimeRepliesSubject
11:34PM 3 iaxtel and iax.conf
8:38PM 2 Can't get simple config working!
7:39PM 11 Recommended OS
3:52PM 0 Last call: Asterisk BoF in Boston, Tuesday 23rd
3:06PM 1 Speaking of Outlook
2:42PM 3 Call Pickup problem with cisco 7960 (SIP)
2:34PM 0 Example weather report AGI by Zip Code using Festival available
1:33PM 2 Re: Anyone looking for IP Phones?
1:26PM 1 Undocumented variables in chan_sip.c
1:08PM 1 app_festival volume problems
12:44PM 1 Status of shipdate on the 4 port FX0 card?
12:30PM 2 failed to load chan_zap
12:25PM 2 re: Anyone looking for IP Phones?
12:14PM 3 Voicemailmain2 user docs?
11:49AM 2 Also CR Spam filters
11:45AM 0 ISDN and Whisper
11:29AM 10 MS Outlook
11:25AM 0 3 fritz-cards pci
10:51AM 3 THIS IS STRANGE
9:25AM 11 how to dial a h323 destination ?
9:16AM 2 Chan_capi account code..
7:27AM 1 Switch between calls without initiating a threeway converstaion
4:50AM 2 Meetme Admin menu
3:33AM 0 Chan_h323 config
3:29AM 4 Setting up MySQL CDR??
12:57AM 0 Warnung: File dsp.c, Line 1198 ???
12:10AM 14 G.729A + Cisco AS5300
 
Sunday September 21 2003
TimeRepliesSubject
7:48PM 3 Skinny
7:46PM 4 Incoming phone line rollover / hunt?
7:42PM 1 SIP NAT QUESTIONS
3:40PM 0 outgoing limit in chan_sip not working as described
3:02PM 12 Very bad echo (appears that...)
2:43PM 5 ISDN BRI hardware
2:38PM 1 Calls being interrupted, analog signalling problems
6:43AM 0 SIP segfault, problem loading modules, gdb output included
5:41AM 0 Extract header(s) of SIP signalling messages
 
Saturday September 20 2003
TimeRepliesSubject
11:56PM 0 Asterisk with Samsung SKP 816H PBX !
11:50PM 2 Oops!!! Current CVS crashes
8:53PM 0 Digium FAQ.
7:24PM 0 Michael Van Donsel
6:23PM 4 iptables rules that work?
4:28PM 1 SIP segfaults and problems loading modules
3:30PM 0 Asterisk fax detection
12:29PM 8 how many production systems are there?
11:05AM 1 sip tone question
9:07AM 10 Maximum retries exceeded w/SIP
3:06AM 9 MY Sql CDR
2:55AM 5 False RING (incoming call) on Digium X101P FXO
1:31AM 0 Configuring Dialogic JCT 2E1 Card with Asterisk
 
Friday September 19 2003
TimeRepliesSubject
10:55PM 1 Cisco ATA 186 / FXO card problem
10:25PM 0 Multiple Asterisk Servers
6:52PM 6 VoiceMail fromstring?
6:28PM 1 Budget Hotel PBX
6:14PM 0 IAXTel calls coming into wrong context
6:03PM 0 When ISDN is busy, asterisk hangs
4:12PM 3 built in dial functions?
2:12PM 1 regexp problems
1:39PM 1 TDM400P question.
1:15PM 0 chan_capi 0.2.5b released
1:11PM 6 SIP + NAT Howto?
1:04PM 6 Recall doesn't seem to work
12:03PM 2 How do you get registered to IAXTEL?
11:18AM 11 IAX vs SIP
10:10AM 0 exit from conference
9:41AM 5 GSM player or plugin for XMMS
9:11AM 1 Aastra 390 w/ADSI - Doesn't automagically use "Asterisk PBX" script
8:58AM 1 Dial out from script. Mini predictive dialer
8:43AM 2 Interface with PBX
8:34AM 0 Equipment listing
8:34AM 5 Can Asterisk automatically initiate a call?
7:40AM 4 No sound on PSTN --> */PRI
7:38AM 0 phonecore, gnophone from CVS.
7:36AM 13 Identify call router? How?
7:35AM 0 Fw: hangup problem Brazil
6:52AM 3 IAXTel registration rejected
6:35AM 3 Voicemail2 crashing on replay
5:28AM 0 ringing tone on analog Zap channel question
4:11AM 1 codec probs wit g723.1
2:54AM 4 SIP registration between *'s
12:41AM 14 AGI problem
 
Thursday September 18 2003
TimeRepliesSubject
9:58PM 2 * website needs a place for
8:32PM 6 TDM400P??
3:47PM 2 Asterisk segfaulting with chan_sccp+7920
2:03PM 3 Adpcm quality
1:08PM 0 Cisco 7910 w/SCCP
12:39PM 0 Hanging up one call when you have call waiting
12:37PM 6 New message 0 in mailbox 7606
10:20AM 2 SIP, X-Lite
9:18AM 5 h.323 - success
9:12AM 2 SIP error messages
8:38AM 1 e100p and E-bit alarm indication
8:20AM 2 CDR of calls transferred via IAX[2]
8:07AM 1 Possible FAQ: IAX2 -> SIP with G729 and no licence
7:44AM 2 VoicePulse offering IAX2 services
7:36AM 2 Need help with H.323
3:55AM 3 Disconnect Problem
3:04AM 0 gastman executable for Win32?
2:45AM 0 no ring tone analog Zap --> I4L
2:03AM 2 Skinny + XMLDefault
 
Wednesday September 17 2003
TimeRepliesSubject
9:33PM 1 End Hide
8:39PM 1 core dump back trace of chan_oh323
8:20PM 13 Nufone 800 numbers working?
6:21PM 3 Cisco 7960 + 5.x Firmware + *
5:51PM 1 Prices for new channel banks, patch panels, cables etc.. etc..
3:18PM 2 ITFS & VoIP
2:59PM 0 CAPI & AVM Fritz DID question
2:04PM 2 Web Based Management App
1:36PM 5 Programming 976 numbers from dialing out.
10:57AM 0 scalabilty
10:01AM 5 documentation?
9:10AM 0 Voicemail2 and time stamps
8:46AM 0 chan_h323 as a gatekeeper?
8:09AM 3 Sip call waiting
7:24AM 1 A WORKING EXAMPLE
7:21AM 0 WebVoiceMail forward message error
6:15AM 3 NEW Asterisk Security vulnerability report ...
5:39AM 0 Does the X100P Support...
4:59AM 0 Aleatori PSTN number with SIP.
4:22AM 2 using pci modem cards as fxs/fxo ports in *
3:21AM 1 Re: Asterisk-Users digest, Vol 1 #1279 - 16 msgs
2:30AM 5 help jeremy
 
Tuesday September 16 2003
TimeRepliesSubject
11:13PM 4 calls terminating abnormally
8:48PM 0 tdm400p loading solved : thnx
6:48PM 4 Follow Me
6:27PM 6 problem loading chan_iax2.so and chan_zap.so from latest CVS
6:14PM 1 Using IAXTEL with RSA authentication. MD5 works, RSA not. [2]
5:55PM 16 iaxComm - IAX client for Win32
5:39PM 0 tdm40b
4:56PM 1 (no subject)
4:47PM 16 Hangups after voicemail
4:32PM 3 Dialogic Hardware (Take 2)
4:23PM 2 Re: [digium.com #860] Fw: x100P: Ring/off-hook in strange state 6 on channel1
3:50PM 0 VTGO! Skinny PocketPC Client fails with Skinny Register
2:56PM 0 NOTICE[81926]: File chan_sip.c, Line 5144
2:32PM 5 Adpcm, 6KHz codec
2:28PM 0 Dialogic channel pricing
12:33PM 4 Installation Configuration Questions
9:38AM 16 call center design question
9:35AM 0 C7960 distinctive ringing sample config?
9:24AM 0 Asterisk voice mail to PBX
6:52AM 3 Cisco Gateways
5:24AM 0 X100P and TDM400P
5:13AM 0 No correct IP in RTP media stream
3:09AM 8 h323 gatekeeper registration failed
12:55AM 0 audiocodes mp-104
12:38AM 2 Channelized T1 Question/Request
12:20AM 3 Any Universiry using Asterisk ??
 
Monday September 15 2003
TimeRepliesSubject
10:52PM 0 advice sought on possible rollout
7:35PM 9 Talking to other SIP hosts, wrong IP
2:10PM 2 Anyone using National ISDN (NI-1) BRI under Linux?
1:50PM 5 T1 PRI
1:31PM 2 Online ordering for Grandstream phones, now available
1:11PM 49 Grandstream Source?
1:01PM 4 SOME QUESTIONES (LOG, MySQL, Extensions)
12:49PM 5 Hardware Vendors for Asterisk other than "DIGIUM"
12:47PM 3 X100P & T100P knock-off boards
12:00PM 4 T/E410P motherboard requirements ?
11:41AM 7 Cisco 7905
10:49AM 3 extension parser
10:15AM 0 bug in Directory app??
9:35AM 14 Radio for Music on Hold?
9:13AM 5 TDM400p loading errors
8:19AM 2 echo cancelation
8:08AM 2 Asterisk - Different Subnet to phones (Cisco 7960)
8:00AM 1 User interface issues (was voicemail menu structure)
7:27AM 19 Analog FXO Card
2:46AM 0 Codec problems only with AGI ?
2:42AM 12 ISDN BRI active adapters with NT mode - any alternatives ?
 
Sunday September 14 2003
TimeRepliesSubject
10:46PM 0 SNMP development ideas for Asterisk
8:50PM 2 FXS Card Recommendation
8:26PM 1 gnophone and RH9 comments and questions
8:03PM 0 QTelNet Products : Anyone used before?
7:39PM 6 Outgoing SIP trunk
7:33PM 0 web based interfaces
6:36PM 4 cdr_mysql: cannot connect
4:58PM 2 ATA v2.16 Register Update problem
4:37PM 8 AGI question
3:44PM 1 Architecture Advice
12:07PM 0 asterisk monitor
9:26AM 4 Today's CVS is good (Version Asterisk CVS-09/14/03-08:48:21)
8:57AM 7 chan_capi
8:45AM 0 debugging callerid help
1:22AM 2 Ztdummy not loaded
 
Saturday September 13 2003
TimeRepliesSubject
10:14PM 6 MusicOnHold (MOH) silent on BudgeTone-100 only.
10:03PM 2 Caller-ID name delivered in double-quotes
9:37PM 9 [Release] Skinny Support in cvs
7:10PM 0 Submit commands thru web & php
7:01PM 1 * <--> FWD
4:27PM 0 # during ringing causes Asterisk to crash!
4:06PM 3 VoiceMail2 mysql table structure
4:05PM 14 Does * machine need a sound board for MOH?
2:47PM 0 cvs tags / release process
2:41PM 1 ZT_CHANCONFIG failed on channel 1: Invalid argument (22)
1:00PM 14 LineJack + Asterisk HELP!
12:51PM 5 Voicemail to a commercial PBX/key phone system
11:08AM 9 SJphone DTMF?
10:41AM 2 install help - make error
7:18AM 4 Source for 50-pin amphenol cables?
5:22AM 2 How to test * ?
5:01AM 3 UK Suppliers
4:13AM 0 Monitoring an active channel
 
Friday September 12 2003
TimeRepliesSubject
10:17PM 2 problem with * and Howlink CL-100 ip phone
9:20PM 3 Caller ID Problems
8:41PM 0 zaptel wcfxs and wcfxo recent cvs drivers bad ??
8:24PM 0 NETjet ISDN Card
7:43PM 1 Re: Asterisk-Users digest, Vol 1 #1273 - 10 msgs
5:20PM 0 Xten SIP client for Macintosh OSX
2:48PM 8 asterisk and defunct perl procs
2:06PM 0 Auto-detect of fxo vs. fxs channels?
2:02PM 0 First seconds of outgoing SIP call are cut-off
1:26PM 1 Dect Phone
1:05PM 6 IAX, IAX2 and authenticatyion
12:51PM 0 (no subject)
12:47PM 1 TDM40B Installation problem
12:37PM 12 (no subject)
11:54AM 0 getting a * to work through at NAT to an outside proxy
11:47AM 3 register => w/MD5?
11:24AM 2 X-Lite + Asterisk
11:19AM 0 NAT support idea
11:08AM 65 Music on Hold
10:24AM 5 7206 as SIP->PSTN Gateway?
9:36AM 0 New Implementation Questions
9:30AM 6 Asterisk using a h323 gateway
9:12AM 2 Voicemail time limit?
9:02AM 0 Newbie (unfortunately =)) q regarding BRI
8:52AM 0 Asterisk SIP DNS srv records
8:43AM 5 E400P woes
8:34AM 5 Voicemail 1 and 2
8:30AM 8 h323 v oh323
8:21AM 3 say number question
6:56AM 0 Q. on key sniffing/spoofing
6:20AM 4 Voicemail menu structure
5:30AM 1 Call volume on ATA 188
2:09AM 1 UK based guy, wants card for machine.
 
Thursday September 11 2003
TimeRepliesSubject
10:58PM 10 Cisco 7960 + SIP
10:46PM 1 Problems dialling US numbers with asterisk
9:50PM 0 Simple script to get asterisk from CVS
7:36PM 3 phpconfig README and INSTALL
6:57PM 17 phpconfig is out in CVS
6:52PM 0 E1 config - Telekom Malaysia
6:06PM 2 Incoming calls from IAXTEL over NAT
5:50PM 8 Start of all recordings cut off
3:23PM 5 PROBLEM RECIVING CALLS AT FXO
1:32PM 3 SIP busy
1:29PM 1 Segmentation fault due to SIP registration N UMBER 2
12:41PM 1 how to make sip uri work
12:14PM 2 Segmentation fault due to SIP registration NUMBER 2
12:05PM 14 Legal Interception - tapping
11:40AM 2 How much to charge for Asterisk installations?
11:33AM 1 Final version of ZapScan
10:54AM 0 Asterisk with 300-400ms latency
10:45AM 1 * with cisco 7960G
10:21AM 3 Is my card bad?
10:16AM 0 Hangup Detection and BUSYDETECT_MARTIN
10:05AM 0 QOS LINUX
9:56AM 0 RV: WARNING[5126] Maximum retries exceeded on call
9:54AM 5 Some Question of extension.conf
8:55AM 3 SIP client<->NAT<->Asterisk<->NAT<->SIP client. only works with canreinvite=no.
7:07AM 1 newbie - sip, pxb, ata, nat
7:03AM 0 AOC
6:52AM 2 TDM400P Problem
6:33AM 1 Dialogic/4
6:19AM 3 Is there any MFC-R2 implementation for asterisk?
4:53AM 0 CLI remove dynamic members from queue
4:52AM 2 autologoff dynamic agents
4:39AM 0 PPP over ISDN BRI (modem_i4l) ?
2:11AM 1 UK Asterisk user, please pick up the white courtesy phone
1:59AM 1 g729 codex experimentation
1:13AM 9 SIP to SIP monitor and record?
 
Wednesday September 10 2003
TimeRepliesSubject
11:07PM 0 Transfer button on BudgeTone (Re: Transfer of queue call)
10:44PM 5 Cisco 7940/7960 XML application hint
5:30PM 1 MOH - White noise, static
4:37PM 6 # of T400Ps in a machine
4:01PM 0 config for X100P card?
1:47PM 16 G729
1:15PM 2 NO TONE ON ZAPATA FXS CHANNEL
11:32AM 1 New RFC: How to specify a phone number
10:59AM 1 Request for best practices
10:46AM 5 running * on a VPN gateway
10:12AM 4 Voicemail notification email with no attachment despite attach=yes
9:55AM 5 TDMoE and codecs
9:52AM 4 ADSI Programming
9:17AM 0 incoming ringmaster distinctive ring identification
8:47AM 1 newbie help.
8:37AM 10 Having problems with S100U
8:15AM 2 Prompts and sound quality of the X100P card (FXO card)
8:09AM 1 Linejack Dialout (FXO)
7:39AM 13 Free World Dialup (FWD).
7:02AM 0 Request for consulting time wanted for project
6:50AM 1 ADSI & Vista/Aastra 350
4:28AM 2 LEDs on E100P card
3:34AM 0 Noise over iax2 and FXO
1:28AM 4 CAPI silence detection
12:44AM 2 Unexpected Call Termination!
 
Tuesday September 9 2003
TimeRepliesSubject
8:28PM 3 Transfer of queue call
8:19PM 10 Dial + disconnect
7:06PM 2 Nortel i2004 and asterisk ??
5:16PM 7 help on MOH config, pretty close?
3:25PM 2 UK Caller ID and X100p
3:15PM 3 Pushing data to a 7960
2:55PM 0 * Picks up line during outgoing call
2:02PM 2 Is there a web interface to the asterisk system?
1:53PM 2 ADSI phones?
12:54PM 27 Asterisk Security vulnerability report
11:49AM 1 Is t.38 fax relay supported in Asterisk?
10:41AM 2 ISDN TA
10:26AM 2 Has the "allow=all" function changed in sip.conf?
9:44AM 13 Request for comments on queue statistics
8:48AM 0 Snom polling..
7:49AM 6 delay problem in h323
7:36AM 0 Asterisk @ SMAU
7:25AM 0 Snom200 -> C7960 noisy?
5:17AM 1 Getting a local number abroad - Newbie question
4:01AM 2 Most Basic System
4:00AM 2 DBPut and DBGet performance
2:27AM 1 Dynamic SIP outbound usernames?
2:24AM 1 grandstream-budge tone message button
2:07AM 0 User Guide
1:27AM 0 incomplete address response SIP 484
1:15AM 7 Xlite = no sound
1:02AM 1 zultys 4x4 ip phone
12:32AM 0 Comparison- SNOM 4S Soft Switch and Asterisk
 
Monday September 8 2003
TimeRepliesSubject
11:56PM 13 Callgroup, Pickupgroup and SIP
10:33PM 7 Channelbanks
10:22PM 3 Monitor an active channel?
10:13PM 0 Is this use of DISA secure?
8:26PM 3 Asterisk as a GW or PBX?
4:47PM 1 Urgent help - File size limit exceeded error
4:32PM 0 Browsing CVS Online
3:41PM 6 Help needed with IAX behind NAT
1:48PM 1 Double dialing with asterisk and Grandstream BudgeTone-100
1:29PM 4 Adtran TA750 MWI problem
12:17PM 1 extension.conf and SIP phones.
11:33AM 1 Mixed FXO and FXS on one Adtran, T1 card?
10:52AM 56 Fax
10:37AM 3 Using a Cisco 7960G
10:35AM 1 SIP Status Codes
10:13AM 3 Cisco 7940/7960 ethernet ports
9:58AM 0 Question about using AGI - over network (with Java)?
9:47AM 13 *78 *72 and sip?
9:46AM 1 WARNING[81926]: File chan_sip.c, Line 432 (retrans_pkt): Maximum retries exceeded on call 56f9689d0bf0b8ca2c9f5b675119089c@209.107.188.218 for seqno 102 (Request)
9:10AM 0 chan_oh323 delay
8:53AM 0 The old versus new TDM400P board
8:13AM 0 asterisk-oh323 question
8:05AM 0 Determine who picked up after dialing multiple lines
6:52AM 0 CDRs and zap channels
5:44AM 1 ephone hunt group?
5:07AM 9 live monitoring
3:11AM 17 Maximum number of X100P cards in the same * box
2:51AM 1 cisco 7960 G with *
2:10AM 2 cdr_mysql question: accountcode
 
Sunday September 7 2003
TimeRepliesSubject
11:51PM 2 exten sent with MWI??
11:23PM 0 Asterisk Call Flows
9:40PM 7 Call Time out Problem-Very Urgent!
9:24PM 11 how to connect 2 TE410P
9:22PM 2 Problem Installing Open H.323 Channel Driver
9:11PM 0 SayNumber patch for spanish
9:01PM 2 TE410 - 3.3v?
7:06PM 0 Conference Leader
3:02PM 1 CallerID through the GnuGK - does this work?
2:15PM 0 chan_local environments: unexpected results
2:13PM 41 freebsd and asterisk ?? anyone yet
2:07PM 5 Is 3 minutes a magic number of some kind??
12:59PM 3 Sound error during launch
10:04AM 5 New cvs compile; basic operational question, please.
7:34AM 1 ISDN problems (Take 2)
4:17AM 1 Asterisk Application Documentation
3:39AM 0 FYI: Perl module for Cisco 79x0 phones...
1:59AM 0 Problem calling from IAX client to SIP phones
 
Saturday September 6 2003
TimeRepliesSubject
10:39PM 3 GrandStream Phones... White,Black or Green?
8:01PM 0 ZAPATA_R2
5:57PM 8 MP3 streams for MOH: idea
3:52PM 0 Fwd: Asterisk BoF: Boston, Sept _2[2-4] - interested?
2:54PM 5 digium dev kit - X100P & TDM400P
1:46PM 2 google search of asterisk archives?
12:32PM 1 Limiting the number of SIP/IAX "lines"
11:49AM 9 MOH other than mp3 ??
11:28AM 2 SIP Phone -> Asterisk -> SIP LD Provider question
11:03AM 5 Ser vs Asterisk?
10:07AM 0 Voicemail Indications
9:50AM 0 NuFone.net Was:VONAGE or IP Dialtone
7:42AM 10 OT: Creating documentation using a web interface
4:40AM 15 What is the best IP phone?
 
Friday September 5 2003
TimeRepliesSubject
8:47PM 2 Bug in my head or bug in the code?
7:03PM 1 ISDN Primary Rate Interface (PRI) - 2B Transfer
6:48PM 0 SIP + NAT question
5:46PM 0 SIP and NAT traversal
5:09PM 1 Fax tone detection problem on bridged (PRI -> SIP) calls
4:20PM 7 VONAGE or IP Dialtone
2:40PM 1 SIP Phone to use with *
2:36PM 17 Moh
2:31PM 1 T1 - A little guidance needed to get started, What order to do zaptel, zapata...
2:09PM 0 chan_zap "Cannot handle frames in 2 format"
2:08PM 1 Noisy/Clicky hangup
1:08PM 0 Asterisk phone system plan - for review!
12:21PM 1 CDR not recording SIP username
12:05PM 13 app_queue input needed...
9:12AM 0 Polycom IP Phones
9:08AM 0 Windows 2000 call viewer!
8:50AM 1 Can you turn up the gain on sip calls?
8:37AM 0 Manager / Windows Apps / Line Appearances
8:31AM 0 IAX sound probs
7:22AM 1 oh323 call segmentation fault
7:16AM 0 Voice prompts, do we have to use GSM?
6:31AM 0 Problems setting asterisk environment varibles
5:58AM 5 call parking -- what was the key combination?
1:24AM 5 Transfer (again!)
12:59AM 0 Realm..
12:56AM 10 Hardware IAX phone (please read and reply!)
 
Thursday September 4 2003
TimeRepliesSubject
11:05PM 0 DTMF CLIP
10:08PM 3 Cant locate my X100P
10:05PM 7 The sounds of silence: silent soundfiles available
9:56PM 2 cisco ATA186 I2 vs I1
5:12PM 1 Regular expression matching for ":" - examples needed
3:22PM 3 Modems and Tivos? Oh my!
3:15PM 0 * and Zap on AMD64/Opteron
3:10PM 5 7960 backup proxy registration
12:48PM 2 Traffic Modelling (was IVR only system with scalibility...)
11:22AM 9 I don't think I understand "Call pickup"
10:43AM 7 IVR only system with scalibility with asterisk???
9:02AM 1 Scalability of the Asterisk
8:40AM 12 update re. Grandstream + SIP + Echo problems ..
8:39AM 0 First steps with asterisk and Voip (SIP, MGCP, H323, IAX).
8:10AM 2 Question about cdr_sql fields
7:43AM 1 Asterisk vs. Vocal (Vovida) vs. Bayonne
5:53AM 7 Call script after hangup
4:31AM 0 oh323 <-> sip communication problem
3:28AM 0 Problem in generating CDRs
3:23AM 1 Arraycom voip phone
3:01AM 1 SIP - DTMF Payload type
2:33AM 4 can't use 2 controllers
2:30AM 0 E100P in Switzerland
2:21AM 2 Help configuring E400P cards
2:04AM 1 remotely picked-up extension keeps ringing
1:03AM 4 Incoming CallerID management
 
Wednesday September 3 2003
TimeRepliesSubject
11:18PM 1 FAX over SIP
9:29PM 5 IAX2 ports usage
6:58PM 8 Pointer to upgrade 7960sip beyond v3.2.0?
4:48PM 1 Packet8 Users
3:48PM 14 Asterisk Jitters
2:35PM 0 SCCP Protocol and Asterisk
1:03PM 5 telantek.adsi
12:58PM 1 SIP to PSTN gateway
12:51PM 6 Newbee Question
11:49AM 3 resend: * newbie: overhead paging and nbsd
9:20AM 4 IAX and frames/packet
9:13AM 5 MusicOnHold and MP3Player not triggering "answer"
8:57AM 1 SIP to CAPI problem
8:32AM 11 SIP on TCP
8:23AM 1 Error Making zaptel.o
6:01AM 6 OT - Headsets for Cisco 7940/7960
5:47AM 2 E1 PRI's in Australia
5:36AM 7 E1 problems
4:23AM 0 * max & limits
1:27AM 3 g729 codec + kernel upgrade
12:56AM 1 Traversing the NAT
12:29AM 0 Problem with Asterisk -> Welltech Wellgate 3502
 
Tuesday September 2 2003
TimeRepliesSubject
11:25PM 2 One way voice through NAT
11:14PM 14 frames/packet
10:51PM 5 STUN server from Vovida
10:14PM 0 IXJ card doesn't want to dial out (see previous thread, asterisk won't answer pstn ring)
10:13PM 0 Designing a lab for a telecommuncations course using Asterisk
6:48PM 4 DTMF Tones During Call
6:46PM 5 Outgoing call answer confirmation
6:13PM 25 ISDN
2:30PM 1 Configure DID Numbers with T1 Line & T100p
2:17PM 3 Openbsd PF firewall ?
1:50PM 10 Still no audio on SIP phone
1:31PM 2 Cisco IP Phone 7905G
12:54PM 5 extensions.conf issue
12:45PM 0 exiting Voicemai for VoiceMailMainl
11:11AM 3 problems with mediatrix 1204 FXO
10:12AM 5 Stuck On ISDN
9:25AM 0 vmail.cgi forward problems
8:36AM 1 Connecting to an Ericsson AXT121 with a Digium Wildcat E100 card
6:30AM 0 =?iso-8859-1?Q?Re: =A0RedHat=A0Distribution?=
6:11AM 5 Low bit rate codec (speex)
4:29AM 2 RedHat Distribution
2:08AM 0 Incoming phone dialing / IXJ
12:21AM 1 Matrix of Hardware in use by Asterisk Community
 
Monday September 1 2003
TimeRepliesSubject
11:39PM 12 TE410P - one way audio, after "rpm -qa" on RedHat 9
8:23PM 14 Sip Software from Nero Folk?
1:40PM 6 Unified Messaging Support ?
1:18PM 2 IP Phone compatible with Asterisk
12:52PM 0 RAS
12:25PM 0 Warnings on IVR
10:23AM 7 MGCP question
8:38AM 1 Non Traditional PSTN Trunking
6:01AM 1 Quickstart Guide
4:51AM 0 X-Lite and iLBC
4:06AM 0 Problem with SIP: Maximum retries exceeded
2:19AM 3 gnuGK + h323 Caller ID
2:07AM 3 some pri questions...
1:51AM 14 Change include contexts runtime