asterisk users - Jun 2004

Wednesday June 30 2004
TimeRepliesSubject
10:05PM 2 Session timer
8:26PM 0 CVsup up
8:05PM 2 Can't transfer with Zap and SPA-2000
6:39PM 1 Digium cards supporting E&M signaling
5:04PM 0 Dial plan errors
4:36PM 16 Echo cancellation, when software doesn't cut it. Whats next?
4:11PM 0 Answering Service Auto Login
3:13PM 2 Ring tone changes when asterisk answers the call
3:10PM 3 Patch for call queues?
2:57PM 24 Special Delivery from China
2:41PM 0 asterisk: problems with connecting to a (german) sip provider
2:40PM 0 Asterisk Wish List - Can We do it? Can you add to it?
2:23PM 0 Double DTMF digits
2:05PM 21 Asterisk Causing Cisco 7200 Router to Crash?
1:28PM 0 Ipeya iGate-4-FXO-SIP
1:25PM 1 SIP Notify contents showing 0/0 on VoiceMail
1:17PM 9 strange problem with oh323 loaded!
12:59PM 1 Spam
12:03PM 1 asterisk-addons unable to compile
11:49AM 4 Answering Service Agent Auto Login
11:39AM 10 Bugfix for CVS-HEAD-06/26/04-21:56:45
10:57AM 1 Providing Telewest in the UK with per extension outbound callerID
10:40AM 5 Using Asterisk as H323 gateway
9:31AM 8 Remote SIP client HACK JOB
9:25AM 1 Sound not working?
8:41AM 2 IAX2 IP Address memory
8:21AM 1 Null Pointer Reference h225_1.cxx
7:17AM 3 10:10am CST - VoicePulse appears to be down
6:35AM 0 Hold Button on FireFly does not launch MusicOnHold on *?
5:49AM 4 Support for CENTOS-3.1
5:16AM 4 Anyone using gr303?
4:06AM 0 Compile error with CVS HEAD zaptel
3:46AM 2 AGI Diad number
1:58AM 27 zaphfc - hfc pci based ISDN card : point2point & DDI
 
Tuesday June 29 2004
TimeRepliesSubject
10:28PM 0 Vonage Softphone/resolved
8:58PM 8 SIP->Asterisk->GnuGK->Cisco 5300
8:22PM 0 VoiceTronix OpenLine4 FXO Setup
8:18PM 4 [RFC] New Wiki page on IAX2 authentication
6:54PM 2 TDM411B configuration
5:22PM 0 timestamp in the future (linphone)
1:17PM 4 Registration of H323 Endpoints?
1:03PM 1 Spandsp and rxfax
12:59PM 0 Ring Voltage
12:54PM 1 RTP Binding Address
12:24PM 0 chan_dialogic
11:55AM 11 cisco phone and parked calls
11:28AM 3 incoming cid translation tables
11:15AM 6 ISP: AT&T or Sprint
11:04AM 0 Caller ID software v1.1 available
11:00AM 1 Update Mysql with DTMF
10:55AM 4 t100p configuration troubles
10:34AM 0 Play Music on hold until a ZAP channel frees up.
10:10AM 5 Getting Asterisk to automatically dialout
9:57AM 0 ldap-lookup
9:56AM 2 T100P-E100P circuit board differences
9:31AM 0 Complaining Emails
9:24AM 6 nat problem
8:53AM 0 MGCP and call waiting, doesn't work.
8:51AM 0 Playing the invalid extension input
8:23AM 2 Asterisk and dial-up modems
8:14AM 0 DLink mgcp phone and CVS HEAD
7:13AM 10 Outgoing CallerID on PRI problems
6:36AM 4 Call dropping out after 5s: Solution!
5:32AM 0 h323 audio problem (next)
5:10AM 0 Routing incoming H.323 calls to specific contexts.
4:35AM 0 Compiling libiax2 on windows
4:17AM 0 Voip Account over H323
3:42AM 4 Ruggedised IP Phone
3:07AM 4 1 user 2 VM boxes?
2:43AM 3 cvs log archive
2:06AM 2 Customized Call Parking
1:59AM 3 How to test E1 interfacing?
1:36AM 1 Asterisk and Sipura SPA-1000 configs
1:23AM 0 P32mxi
12:43AM 18 linux kernel 2.6.6
12:14AM 3 * Busy-Redial ??
 
Monday June 28 2004
TimeRepliesSubject
10:12PM 2 New Firefly release - 1.9.3
9:34PM 3 cannot make app_prepaid
8:04PM 8 Adit 600 - Getting Dial Tone
7:16PM 7 Polycom IP600 stops to send/receive calls
6:34PM 1 Cisco 79XX Ringers & chan_sccp
5:07PM 2 Incoming IAXTel/IAX2 issue
5:00PM 0 SpanDSP Scrunching incoming faxes
3:47PM 35 Modems behind Asterisk - how?
3:46PM 0 New VoIP deployment.
3:37PM 0 Suggestions for Outbound Proxies?
1:48PM 1 Asterisk & Festival, not a happy couple
1:26PM 0 Context for Incomingmsn
1:18PM 1 Asterisk and hyperthreading
12:55PM 0 Weird 7940 issue
12:51PM 8 Security Vulnerability in Asterisk
12:01PM 2 Would this work?
11:47AM 0 Queue hold time in seconds
10:50AM 12 Vonage and Asterisk integration
10:34AM 5 Chan_Capi Down
9:43AM 4 Dial Command
9:09AM 5 zaptel compile error
8:48AM 3 Asterisk Flah Operator Panel show iax2 trunk
8:26AM 7 Zap X100P oscillation
7:55AM 3 (no subject)
7:45AM 2 asterisk-oh323, new version 0.6.3
7:35AM 3 AGI->Exec Problem
6:00AM 1 TE410P -> Dialogic D240SC
3:45AM 3 Protocol Error (6) using Zaphfc
3:13AM 1 SetGroup and CheckGroup
3:11AM 3 Unable to forward voice
2:11AM 1 Disappointed
1:52AM 2 sip to isdn-capi call problem
 
Sunday June 27 2004
TimeRepliesSubject
11:01PM 1 New idea
7:02PM 4 Re Cron
6:55PM 0 ip10s Sip Firmware released
6:52PM 2 Re: I never get to hear more than 5s of the demo channels
6:49PM 1 IAX Phone Issues/McAfee Virus Scan vs. IAX Phone
5:23PM 0 Hangup Issue
5:16PM 1 Source for MD3200 modem cards?
3:20PM 17 H.323 Audio problem UPDATE
1:39PM 14 Re:Latest Echo changes
12:55PM 1 Asterisk on 64 bit... and testing e164.org
11:43AM 11 Optipoint 400 Standard Sip
10:55AM 2 H323 audio problem
10:20AM 1 Asterisk on 64 bit... and testing e164.org's stuff
9:46AM 2 Dead Budgetone-101?
8:10AM 6 Multiple X100P in Asterisk box?
7:55AM 1 General advice on confs and setup for new users
3:06AM 15 Asterisk on 64bit ?
2:20AM 8 Hong Kong VOIP Exchange
2:09AM 3 Why? oh why can't I dial out?
1:17AM 2 Confused with CallerID when using the iax chanenls
12:49AM 4 asterisk addon mysql
 
Saturday June 26 2004
TimeRepliesSubject
8:52PM 3 Newbie needs help
7:35PM 0 Broken Pipe?
12:03PM 1 Echo worse after new echo patch
12:01PM 6 ZyXEL Prestige 200w - should I return it ?
5:40AM 0 I need DIDs in Canada and USA with roll over option
4:58AM 2 Asterisk Eating Digits
4:49AM 3 IAX & FWD, No authority found?
2:37AM 2 Nightly / Daily, Weekend / Weekday and Holiday regime of the Asterisk
2:29AM 1 How to transfer call in case that I am the originator
2:09AM 1 Setting up your own menu like voice mail
 
Friday June 25 2004
TimeRepliesSubject
9:08PM 5 Problems Compiling and Loading asterisk-oh323 0.6.2
7:48PM 0 3-way calling woes... Nasty static and inconsistent flash detection?
3:31PM 2 chan_sip.c max number of retries
2:17PM 5 Can one send CLID NAME over PRI?
2:05PM 0 - eezeeFone.com - Need to connect 500 Simultaneous users - An opportunity from test bed to a product.
1:49PM 0 Using *0 with Asterisk
1:18PM 0 Ring voltage on a TDM400
1:16PM 0 Asterisks RTP source address binding
12:51PM 4 Using Soxmix on extensions.conf
12:31PM 3 panic() panic() panic()
12:28PM 0 RE: H.323 - NO AUDIO IN BOTH DIRECTIONS
12:25PM 1 SER and NAT
12:09PM 0 ATA186 (sip) in * dynamic mode
11:43AM 2 503 "Unavailable"
11:26AM 1 SS7 status report 2
10:40AM 0 Stable branch usable? Development branch better?
10:18AM 3 Termination Provider
9:50AM 1 Polycom IP 500 - Quality Issues
8:49AM 1 SIP extension outside of IP tables firewall
8:32AM 2 Asterisk & SIP
7:55AM 6 IAX2 authentication confusion
7:54AM 19 NO AUDIO IN BOTH DIRECTIONS
7:52AM 2 forced ring on dial?
7:24AM 17 SS7 to Pri
7:00AM 0 SIP/IAX to PSTN setup time
6:56AM 0 problems compiling shadydial-asterisk on gentoo
5:00AM 0 HT286, fax and FXS impedance for Europe?
4:00AM 2 Howto: Use setgroup, checkgroup to check incoming and outgoing client limits
3:57AM 4 Failure in RTP streaming
2:56AM 0 Provide Open Settlement Protocol capability
2:16AM 3 Latest CVS fax detection & grandstream bug
1:59AM 0 wcfxs CPU usage
1:36AM 0 Bridging two calls together with Eicon card - help please :)
 
Thursday June 24 2004
TimeRepliesSubject
10:31PM 3 Latest CVS, Grandstream and Zaptel bug?
7:40PM 6 Problem with music on hold...
4:55PM 0 Zaptel ZT_CHANCONFIG failed on channel 1
4:37PM 0 Difference between Tormenta/Zapata and Digium Hardware
4:15PM 0 Inbound call handling in Asterisk
4:12PM 0 Wildcard X100P dial out troubles
1:47PM 6 Leave one call to pick up another
1:42PM 0 New changes
1:28PM 9 chan_capi problem - hangup???
12:56PM 0 Detect more than one type of DTMF for calls to voicemail
12:18PM 5 host=dynamic vs host=xxx.xxx.xxx.xxx
11:54AM 2 Record call from switch using service observe? (execute command after dial?)
11:15AM 1 Cisco ATA 186 from iconnecthere, locked?
10:37AM 5 toll access - account code
9:10AM 16 X101P on a UK BT line ---- txgain issue
9:08AM 4 Pulver's WiSIP with Linksys WAPs
6:05AM 14 Asterisk with PostgreSQL
5:56AM 1 Asterisk Manager Commands - Timeout
5:53AM 2 Asterisk bypassed for name but not number - softphone
5:39AM 4 R: R: R: How to force G729
5:36AM 2 Dead air on 7960 sip at start of call.
5:03AM 4 Video/H323/SIP
4:35AM 0 Anonymity and Privacy headers
4:30AM 0 ZAP hangup not working with siemens HICOM
4:02AM 2 ZyXEL Prestige 2000W and DTMF
3:58AM 0 SIP clients, H323 client as gateway?
3:37AM 2 Help with chan_capi
2:47AM 1 R: R: How to force G729
2:41AM 0 -- Serious issues with current CVS?
2:19AM 10 R: How to force G729
1:52AM 4 Swissv oice IP10 behind NAT
1:35AM 2 Delay in Zap Calls?
1:24AM 0 2 E100P cards on one asterisk
1:22AM 2 How to force G729
12:10AM 0 false hangups
 
Wednesday June 23 2004
TimeRepliesSubject
11:49PM 1 Read this if you use CVS HEAD (Unstable) and chan_capi and jitter buffering in IAX
11:48PM 1 R: Which Linux ?
8:53PM 18 Which Linux ?
8:50PM 0 cdr_mysql: Unknown connection error
7:26PM 0 #1 Asterisk and Locustworld
6:09PM 4 FW: No dial tone after installation
5:54PM 20 Serious issues with current CVS?
4:39PM 26 tdm (and x100p?) echo - fix is coming!
4:12PM 0 Problem with Unavailable Message Creation
2:43PM 1 New VM feature: broadcast and delete=yes
2:31PM 0 Asterisk info needed for new application development.
1:56PM 0 UPDATE Patch for postgres enabled app_voicemail.c
1:55PM 8 Really basic stuff :(
1:48PM 0 Patch for postgres enabled app_voicemail.c
1:32PM 1 SIP and audio delay
12:39PM 8 CDRs, Conferencing, and MeetMe
12:39PM 4 Codec G729 Registration problem
12:15PM 0 Digium/Asterisk in Paris
11:07AM 0 tdm fxo users - new bug tracker entries
10:52AM 0 Three Way Calling and External Flash Hook
10:11AM 2 problems compiling zaptel X100P on Redhat Fedora 2.6.5-1.358
10:02AM 0 SNOM 200 using GSM Codec dtmf problem
9:36AM 8 Conference application !
9:22AM 5 Voicemail Password Changes Lost on Asterisk Restart
9:03AM 2 asterisk + appradius & freeradius
8:52AM 0 Asterisk as a SIP UA and voicemail with SER not working anymore
8:47AM 0 Conference calling
8:12AM 6 help needed with read()
8:04AM 0 clarent hardware
7:42AM 2 Call Generator for ISDN (PRI/BRI)
7:17AM 6 X100P Noise
6:49AM 1 Problem with incominglimit and outgoinglimit
6:06AM 0 Busy message and extensions are hanging.
5:12AM 8 Outgoing CLI
4:33AM 1 Asterisk user/host registration
3:51AM 4 Codecs and pauses
3:50AM 0 connecting to Iconnect here using asterisk
3:33AM 1 Problem when dialing in manager terminal
3:26AM 1 capi.so problem on startup
3:13AM 1 USB handset for IAX softphone ?
3:08AM 0 general install??
2:56AM 6 Skype 4 Linux
2:46AM 3 Iax unable to transfer
2:30AM 2 Ireland PSTN Number
2:22AM 0 Accountcode missing in log
2:19AM 0 CSV log stopping
2:06AM 0 Réf.: Call generator
1:46AM 7 Call generator
1:14AM 1 cdr_mysql compilation error
12:42AM 4 Future WinCE IP Phone
 
Tuesday June 22 2004
TimeRepliesSubject
11:31PM 2 Asterisk -- PBX Do Not Disturb
9:29PM 2 Cisco ata-186 port died
6:49PM 0 No dial tone after installation
6:46PM 0 DID in Fiji
4:17PM 6 sidetone noticeably loud on analog handsets on T100P
3:33PM 2 Multiple DTMF digits on 7960
3:19PM 3 Asterisk Caller ID Application (win32)
3:09PM 5 Asterisk answering only one (dialed-) Number on a PTMP (German "Mehrgeräteanschluss")?
3:09PM 2 Problem with Asterisk
2:58PM 2 Two SIP servers communicating without IAX
2:49PM 1 Core Dump on app_dial.c
2:28PM 0 Accessing ISDN with avm bluetooth hardware
2:21PM 2 AstriCon Registration Opens Next Monday, June 28th
2:01PM 1 No such extension ...
1:47PM 3 Unify Incoming and Outgoing sound files
1:12PM 10 FXO impedance matching
1:00PM 4 Eicon Diva 2.0 PCI ISDN Card
12:55PM 0 Modified Prepaid App Database error
12:25PM 0 Ringing to some numbers...
12:18PM 1 AgentCallbackLogin - invalid extension
11:22AM 0 Queueing and parked calls
11:16AM 4 Failover of IAX or Spillover as the case may be
11:00AM 0 Users do not disconnect
10:38AM 1 Weired Probelm with Asterisk
9:48AM 6 *69
8:21AM 2 iax.conf : what is the purpose of trunk ?
7:44AM 6 CISCO 7960 Goes missing
7:43AM 2 Unable to find libiodbc.so.2
7:36AM 0 patlooptest
7:35AM 0 Tricks for Multiple TMD0xB cards?
7:22AM 1 Problems compiling cdr_odbc.so
6:59AM 4 Call forwarding and voicemail
6:52AM 0 2 T100P cards - 2 switch types
6:41AM 0 zapata initial context question
6:34AM 11 IAX2 Trunking help!
6:23AM 4 License and Commercial Use
6:06AM 0 swissvoice ip10s firmware?
5:45AM 0 Re: [Asterisk-Dev] Skype support
5:21AM 1 Eliminating silence suppression(?) on IAX2 calls
4:56AM 1 Unable to create channel - CVS Broken?
3:54AM 0 exten => i ????????
3:46AM 6 Any echo issues with phones from TDM400P > X100P
3:26AM 0 Site changes
3:23AM 1 using 2 single pri cards on 1 server
1:25AM 10 No Caller ID from FXO Problem
12:30AM 4 pwlib compile error
 
Monday June 21 2004
TimeRepliesSubject
10:20PM 0 Call forwarding code
9:26PM 34 Busy message
9:02PM 2 Failover Trunking Won't Fail Over
6:29PM 1 OpenSS7 T400P-SS7 and Digium T400P
6:26PM 0 dialplan help!-RESOLVED
6:04PM 2 Problems with Zaptel
5:41PM 3 VoiceXML support and integration
5:19PM 0 SLC-96/TR-08 Support with T100P?
5:06PM 1 IAXTel Help
4:21PM 0 IAXtel questions
1:16PM 3 Asterisk<>X100P<>Packet8
12:57PM 2 Connect 16 E1/T1 between * and other switch...
12:42PM 0 SpanDSP Fast carrier Failed
12:19PM 4 integrating with existing PBX
11:46AM 0 Strange * hangup issue
11:32AM 0 call forwarding question
11:23AM 4 Siemens Optipoint 400 SIP Problem
11:21AM 0 Asterisk As A Career?
10:17AM 0 Directory dial by name
9:23AM 0 Error compiling festival
9:19AM 4 Caller ID double quotes
9:13AM 4 PRI & immediate=no
7:48AM 0 A Callback AGI script
7:40AM 0 R: Re: cdr_addon_mysql compiling error
7:25AM 3 using # to end a number
7:13AM 0 mandrake and zaptel
5:22AM 2 app_dial broken
4:42AM 1 R: Re: cdr_addon_mysql compiling error
3:42AM 9 Channel bank problem via long cable
3:00AM 0 Queue Stats - Management App?
2:56AM 0 Restricting outbound dialing on a specific p hone
2:53AM 2 Restricting outbound dialing on a specific phone
2:48AM 4 disabling ALERTING message
2:14AM 1 Problem compiling fax applications
12:15AM 0 Re: Asterisk-Users digest, Vol 1 #4230 - 13 msgs
 
Sunday June 20 2004
TimeRepliesSubject
10:49PM 0 Modified Prepaid database
8:52PM 4 please mail me wave.cc and tts.scm
6:28PM 5 Sipura config
6:09PM 0 Question - TDM40B - Hunt Group Possibility??
6:03PM 3 Need different contexts for 2 X100P FXO Cards and forwarding calls
5:45PM 1 Data over Voice through Asterisk
4:03PM 2 Harald Baron/EBAROH/CH/Ascom ist nicht anwesend.
3:55PM 1 No config file?
3:16PM 1 asterisk console mode
3:14PM 1 Grandstream HT-286 // Custom Ring Tones
2:51PM 2 Channel Bank Frustrations
10:59AM 23 One way audio
9:22AM 19 call waiting from PSTN
7:45AM 9 Date Time Stamp with Caller ID
7:23AM 0 Grandstream HT-286 and DTMF
5:49AM 1 chan_oh323: busy not correctly signalled
5:33AM 0 Asterisk rxfax(): One page gets two pages
3:11AM 0 BT Broadbandvoice ATA186 and *
1:14AM 1 Softfax/spandsp Makefile.patch rxfax/txfax
 
Saturday June 19 2004
TimeRepliesSubject
6:40PM 2 RxFax problems
2:59PM 3 HST Saphir with Asterisk
12:10PM 0 Mediatrix 1204 Incoming calls
9:54AM 0 Directory function is not working
9:32AM 0 Hard Coded CLASS Codes (was 11 instead of Star)
9:27AM 0 Fw: #asterisk is +r now, meaning register your nick with nickserv
5:28AM 0 chan_modem dialout
4:40AM 0 Busy when not registered
12:34AM 7 Big problem with Flash
 
Friday June 18 2004
TimeRepliesSubject
11:15PM 8 Fax with SPA-2000's?
9:10PM 15 WaitExten substitute
9:05PM 2 current code release & chan_sip problem/question rport
7:40PM 0 New Skinny/chan-sccp release
5:23PM 0 #asterisk is +r now, meaning register your nick with nickserv
5:12PM 0 not getting sound from chan_oss paging setup
4:53PM 1 using asterisk as sip registrar is not working for me
4:26PM 0 SIP error 407 - can't make outgoing calls
3:06PM 7 Testing UK emergency dialing and LCR.
3:03PM 2 app_prepaid NAT issue
1:58PM 0 enhanced privacy manager AGI
1:27PM 9 Problems with faxing via TE405P/Asterisk
1:16PM 2 cdr_addon_mysql compiling error
1:14PM 0 R: Thousands of contexts?
1:13PM 8 Grandstream CFG file generator
1:10PM 12 Asterisk References
12:53PM 0 cdr mysql amaflags field
12:00PM 2 Iaxy issue
11:53AM 0 Fwd: Re: Disable IAX1 Registrations
11:48AM 0 cisco 924 config
11:31AM 1 Grandstream HT-286 and NAT
10:04AM 1 Asterisk as Media Gateway (was: ATT CallVantage & Asterisk)
9:43AM 12 Thousands of contexts?
9:27AM 1 Lingo and *
9:06AM 2 trouble compiling zaptel-0.9.1 on YellowDog (PowerMac)
8:36AM 10 UK install
8:18AM 0 Possible chan_skinny problems - no ringtone, no moh and no queue messages
8:12AM 4 C7960 g729 question
7:52AM 1 X100P in Switzerland
7:30AM 0 ATT CallVantage & Asterisk
7:16AM 1 Hwo to get CallerID: SIP -> ISDN
6:57AM 8 Problems with X100P
6:23AM 0 FXO Issues - Sorry
6:19AM 2 FXO Issues
5:03AM 31 TE410P / Eicon PRI
4:02AM 0 bri-stuff with current CVS head
4:00AM 0 Asterisk does not start when cdr_odbc ist configured
3:52AM 0 Asterisk and CISCO Gateway
2:37AM 0 Problems reciving fax with Asterisk
1:06AM 0 Poopy errors on quad wcfxo
1:04AM 0 Asterisk command
1:03AM 0 problem number analize
12:21AM 1 Draytek Vigor 2600Vi as SIP client on Asterisk
 
Thursday June 17 2004
TimeRepliesSubject
11:46PM 4 IAX Jitter Buffer
6:09PM 0 Mediatrix 1204 Mibs
5:45PM 0 Zap Dial Problem ---- Erroneous dash
5:43PM 3 IAXy and bandwidth requirements
5:42PM 17 7960 straight through?
4:53PM 0 dialtone stop
4:22PM 0 zaptel - make config
3:16PM 4 trying to set an internal ivr
2:30PM 7 Compiling problem on Debian
1:37PM 2 How can i get the last codec_g729.so
1:19PM 0 Re: SJphone registration problem - Help!
1:05PM 0 snom phone with asterisk and vocal
12:56PM 2 Having problems with Agents and calls going to voicemail
11:57AM 11 TDMoE Question
11:24AM 2 Disable IAX1 Registrations
10:21AM 0 mgcp/T1 interface/alternatives
10:20AM 15 BT Caller ID - From Patch ?
9:57AM 1 VOIP to Cellular
9:55AM 1 Zap dropping calls
9:13AM 0 Terminating VoIP calls with Asterisk
9:09AM 0 Resend to correct graphic - Internet Talk Radio use Talk Show PBX
8:40AM 1 VOIP wiretapping article
8:36AM 0 Port numbers for traffic shaping
8:20AM 2 How to let users change Voice Mail password in Asterisk
8:18AM 0 Problem with bridging two external lines
8:03AM 4 SJphone regestration problem - Help!
7:44AM 8 Blank faxes with RxFAX
7:32AM 4 Asterisk as Internet Talk Radio PBX system
7:07AM 6 asterisk-addons compilation error
5:23AM 0 Zapata.conf & Signaling for Bulgaria (PSTN: Siemens PABX)
4:35AM 5 SFTP
4:19AM 2 HFC ISDN card with bristuff from jung hanns.n et?
3:54AM 12 Cheap (US$120 or less) SIP Phones
3:49AM 1 Anyone have experience with chan-capi in Australia?
3:42AM 1 HFC ISDN card with bristuff from junghanns.n et?
3:40AM 2 HFC ISDN card with bristuff from junghanns.net?
2:28AM 6 Problems with PRI with T410 messages
1:41AM 7 LDAP synchronization script
1:12AM 1 Calling the firefly network?
12:43AM 0 Accepting SIP calls from unregistered gateways
12:27AM 8 pri with TE410P not working (Austria)
12:12AM 0 no audio with sip
 
Wednesday June 16 2004
TimeRepliesSubject
11:23PM 4 IAX2 no compatible codecs
10:57PM 0 Disable authentication on outgoing SIP calls
10:14PM 3 RxFax - Fast carrier training failed
9:49PM 0 S100U USB FXS problem
7:55PM 0 D-Link DVG-1120M and *.
7:35PM 8 911 emergency service and VoIP
5:16PM 0 (no subject)
5:15PM 0 Changing the asterisk timezone
4:13PM 1 Modified Prepaid Error
4:05PM 9 embedded Asterisk
3:01PM 0 Size of box for 4xE1 conf bridge?
2:19PM 8 UIP200
1:55PM 3 ZAPHFC - only for * 0.7.2?
1:49PM 6 Failed to authenticate on INVITE
1:23PM 2 playinterruptibletones
1:05PM 1 festival with asterisk problem
1:03PM 3 X-Lite/Firefly behind NAT connecting to Asterisk not receiving RTP
12:26PM 4 Soekris Engineering net4801
11:23AM 2 IAX registration
11:20AM 1 limitations ?
11:18AM 15 Cost of IP Phones, or Isn't It Just Software?
10:20AM 2 ATA186 v3.1 SIP - Attended transfer: NO JOY
8:40AM 0 Cisco DSP Modules and Linux
8:15AM 8 Status-info 1: Signalling C7 / SS7
7:56AM 1 NAT and Qualify Question
7:32AM 11 Invalid Extensions -- More like traditional PBX systems?
6:53AM 1 replacing cisco callmanager with asterisk?
6:14AM 3 BT101 and caller id and web interface
5:44AM 0 Re: Approved
5:18AM 1 VOIPTalk silver service
3:46AM 2 Remote rebooting a Cisco 7940
3:30AM 0 asterisk server hang up after conference
3:20AM 5 Digium X100P vs Dodgy Ebay X100P
3:16AM 0 Problems with Call Forwarding on a 7960
2:32AM 0 Problem with incoming calls from FXO
2:06AM 2 Asterisk hardware configuration and cost?
12:44AM 2 error loading meetme module
12:38AM 2 Fedora2 and Kernel 2.6 again!
12:19AM 4 asterisk/netmeeting works, asterisk/ohphone doesn't?
 
Tuesday June 15 2004
TimeRepliesSubject
11:33PM 0 how can I catch
6:11PM 1 Choppy sound ONLY when a voicemail is left
4:50PM 7 Voicepulse Down Again?
2:36PM 9 anyone use mailboxexists?
2:29PM 1 sip register and nat
2:23PM 0 sip.conf - register and peer groups
1:18PM 0 IVR Prompt errors (scratchy)
1:17PM 0 RE: send pstn calls to cisco gateway ?
12:53PM 0 TDM400P FXO problems
12:25PM 0 Excluding DIDs from telco long distance codes
12:06PM 0 app_conference Compile
11:35AM 9 Grandstreams randomly go busy with Asterisk?
10:58AM 3 Multiple X100Ps -- order?
10:19AM 0 making * more like a normal pbx (ciscoata-186)
7:39AM 4 using SetCDRUserField in an AGI script
7:28AM 0 SIP Registration with Entice Softswitch
6:55AM 0 Re: OT: fax obsoleted? Was: Re: Fax via email (Steve Underwood)
5:59AM 2 Cdr_addon_mysql.c compile problem.
5:59AM 0 Re: OT: fax obsoleted? Was: Re: Fax via email (Steve Underwood)
5:55AM 9 Polycom IP 600 Programmability
5:35AM 5 Capi problems
5:29AM 2 Re: OT: fax obsoleted? Was: Re: Fax via email (Steve Underwood)
5:09AM 10 building asterisk
3:38AM 0 how can I catch How to catch some incoming call
3:22AM 0 Siemens Optipoint 400 standard SIP
3:15AM 0 Simultaneous UA use of services
2:49AM 0 Trunk ?
2:27AM 6 PRI problems (telewest -> * -> LG GDK 186)
1:35AM 5 Queue then Voicemail
1:30AM 8 Outgoing DTMF when using BRI & i4l (Eicon Diva) - problems
 
Monday June 14 2004
TimeRepliesSubject
11:46PM 7 No B-Channels. PRI. E100P. HELP!
11:33PM 0 pulse dialing
8:50PM 5 IAX2 hangup on transfer
8:03PM 40 Asterisk-Users List Etiquette
4:34PM 1 making * more like a normal pbx (cisco ata-186)
4:33PM 3 inviting an spa-x000
3:46PM 2 chan_h323 no audio both ways
3:40PM 1 IAX and Reorder
2:50PM 1 Cisco SIP Phone Licensing
2:15PM 0 CLEC / SIP interconnection?
1:38PM 1 telephones to use with asterix
1:36PM 0 ast_data, mysql, md5 hashes for passwords
1:34PM 3 International Talking Clocks
1:10PM 1 MailboxExists application
12:25PM 1 Multiple tennants, two DIDs, One IAX provider
10:41AM 0 Nextel phone and mute on Asterisk?
10:41AM 5 Sipura 2000 not answering em_w calls
10:14AM 6 Number Portability and VoicePulse
10:08AM 0 compile error with asterisk-addons
10:02AM 0 do_monitor warning message
9:44AM 4 german localization for mailbox available?
9:24AM 11 Polycom IP 600
9:13AM 4 Asterisk real life examples and case studies ?
8:43AM 0 Canadian DID
8:33AM 0 If IAX client is not logged in/registered, Dial plan executes BUSY vs UNAVAILABLE
8:27AM 0 T1 - Adtran and SIP
7:42AM 1 Chan_Capi 0.3.4
6:55AM 7 ASTERISK V. SER
5:58AM 9 Prepaid application error
5:29AM 15 where can I get toll-free number?
5:17AM 6 <<< GSM Audio Files >>>
4:46AM 0 Asterisk as MGCP endpoint
3:58AM 1 Festival application: clipping start of sound?
3:37AM 4 <<< GSM AUDIOFiles >>>
3:10AM 6 TE410P in Austria
2:54AM 13 making * more like a normal pbx
2:49AM 0 FXS--->SER---><Asterisk>--->FXO--->PSTN
2:02AM 1 Install Question
1:55AM 0 Accepting post selection digits over isdn trunking
1:06AM 26 oh323
12:45AM 11 collaboration with Panasonic PBX
 
Sunday June 13 2004
TimeRepliesSubject
9:08PM 2 SIP audio cut off even with Answer, Wait...
5:53PM 8 Comfort Noise
5:46PM 2 Asterisk Agent Logoff?
5:37PM 0 errors on startup
5:30PM 2 Strange voicemail things
5:25PM 0 Help Wanted
4:11PM 0 Red alarm on T1 PRI but not on zttool
2:24PM 0 sip.conf => Configuration of Asterisk with siproxd ?
2:16PM 0 Spanish, Portuguese, other recordings for Allison
2:00PM 2 Wiki now based on CVS head
1:03PM 4 831/408 iax termination
12:42PM 2 Cisco 7960 Problem
11:05AM 6 Sayson IP Phones?
7:35AM 0 DIAX 0.9.8c available for download
5:30AM 1 Re : Newbie help !
4:58AM 9 *** Asterisk Sunday News: Off track with 1.0, moving forward
12:36AM 4 Is nufone web site down?
 
Saturday June 12 2004
TimeRepliesSubject
10:08PM 6 (no subject)
5:48PM 0 ASTTAPI 0.03 hangup not working
5:31PM 2 Junghanns QuadBRI stable?
1:55PM 1 Asterisk on FreeBSD News
11:52AM 2 DECT delay once hungup
10:41AM 1 Problems with Alcatel Speedtouch ST280
9:20AM 2 Call Relaying
8:57AM 2 Sending SABME continuosly. Urgent help needed!
8:47AM 1 'background' problem
8:40AM 1 Changed IP and subnet now no SIP Register 403
7:30AM 1 Capture user input
6:30AM 6 MWI on Cisco ATA-186 (SIP)
6:19AM 0 Problem with E1
6:12AM 2 Cisco ATA-186 Firmware upgrade
1:16AM 14 2 NuFone lines- which one to dial out on
1:14AM 12 Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
 
Friday June 11 2004
TimeRepliesSubject
10:19PM 4 DID/T1
9:59PM 2 * as conference server for shoutcast.
6:46PM 0 GUI Design Ideas Request
6:29PM 0 Service in 252-255
6:04PM 13 BudgeTone hold?
5:54PM 3 ssh key problem
4:38PM 1 Broadvoice conf
2:59PM 3 extensions question
2:16PM 1 oh323 0.6.2
2:03PM 23 Voicemail problem
1:56PM 5 Cisco 7940
1:36PM 0 Newbie to SJphone
1:06PM 2 catch when no voicemail configured
1:00PM 0 context of a transfer
12:21PM 5 cdr_addon_mysql.c
11:16AM 0 CDR not always correct / IAX clients unmonitored
10:30AM 2 Exit Voicemail to VoicemailMain?
10:03AM 1 Integration with SIEMENS HIPATH PBX
9:04AM 0 SIP->Application Codec debugging
8:54AM 8 Simplified Voicemail app / keeping peace with cohabitants
7:38AM 0 Problem with AGI
7:17AM 4 Cisco Auto Provisioning
6:21AM 1 direct dial-in (DDI)
6:16AM 2 Asterisk newbie help !!
6:06AM 24 Broadvoice and DTMF
5:51AM 1 QuadBRI outgoing call problem.
5:48AM 0 Aggressive Echo Suppression
5:37AM 9 phone calls betweens phones behind the same nat
4:38AM 6 Background Playback fails
4:25AM 1 trunk=yes with recent CVS head problems
4:22AM 1 CLI messages screwy?
3:49AM 0 R: VoipTalk down?
3:46AM 0 VoipTalk down?
3:05AM 3 Asterisk PRI calls to SER problem
2:48AM 3 7960 switch port / vlan issue
1:35AM 3 R: hide caller id
1:24AM 0 dialing several phone numbers in one call session.
1:16AM 1 "Caller ID" question
 
Thursday June 10 2004
TimeRepliesSubject
11:55PM 0 hide caller id
11:38PM 0 Re: Asterisk-Users digest, Vol 1 #4101 - 12 msgs
11:13PM 0 PC-to-PC call though SIP Proxy
10:07PM 5 Intel 537EP chipset, revisited
7:19PM 1 RE: question about prepaid app_prepaid
6:54PM 12 XML How To for Cisco 7960
6:37PM 2 Guest IAX with Dynamic IP
4:21PM 4 A couple of newbie questoins
3:46PM 0 Grandstream Ringtones on a per phone basis
3:30PM 1 Uniqueid changing with call parking
3:00PM 0 Missing connect indication on pri?
2:48PM 4 Cisco 7970 w/ 7.1 phones rebooting with asterisk
2:40PM 5 How to get the Called id with AGI
2:00PM 2 New faxdetect change in dsp.c
2:00PM 0 Would like to ask a * user some question over voice or a walk thru in the Hou,TX area
12:59PM 1 Manager logic to pickup a ringing extension
12:56PM 0 Asterisk on Sun Cobalt Qube 3-Ideal system for asterisk
12:49PM 1 Call originate with manager API
11:53AM 0 Outbound ZAP calls
11:18AM 0 oh323 0.6.2 q931 messages
11:00AM 0 BUG?: reinvite and nat
10:49AM 0 Asterisk as a VoIP Gateway to an Analog PBX
10:29AM 1 mysql errors
10:18AM 0 Re: Problem with * not detecting hangup on FXO and VM going into an infinite, loop
10:14AM 0 NAT and symmetric fw
10:11AM 1 Dialing delay when using Zap channels
9:43AM 9 incoming DTMF on iConnectHere?
9:18AM 2 BT is moving to IP ONLY
8:56AM 0 I can't get iaxComm to connect to guest@misery.digium.com
8:54AM 4 Problem with * not detecting hangup on FXO and VM going into an infinite loop
8:52AM 0 Cisco 7960 Tones
8:31AM 20 FW: question about prepaid app_prepaid
8:27AM 13 Automating calls
8:15AM 3 Asterisk on Apple PPC with YDL
7:49AM 12 GSM to ISDN or TAPI
7:09AM 0 isdn4linux and NT mode
6:41AM 3 Iax2 ringtone problem
6:27AM 1 FWIW- Cisco 1750 dropped packets and choppy audio
5:46AM 0 IAX Binding to 2 nic's for trunking two asterisk servers
4:26AM 0 Please help !!!! - IAX, MYSQL - Cant make calls
3:38AM 0 SIP Registration Failed !!(Need Help)
2:56AM 2 Using Asterix and Hylafax with Eicon DIVA E1
2:41AM 1 EU on VoIP
12:53AM 7 Primustel a.k.a. Lingo $20/month unlimited service
 
Wednesday June 9 2004
TimeRepliesSubject
11:44PM 1 Changes in VoiceMail
11:29PM 0 Introduction
10:36PM 3 Another Firefly update - now with SRV support
10:00PM 0 No ringing on outbound PRI calls
6:14PM 2 SIP Registration seems to timeout
2:52PM 2 PC Mag Online article on Asterisk
2:13PM 0 Call Pickup problem in Asterisk with SIP phones
2:11PM 1 IAX Peers from MYSQL
1:45PM 1 Seperate asterisk VM system possibility
12:34PM 0 any banks or financial institutions using asterisk
12:33PM 0 MeetMe and ztdummy problem
12:10PM 0 failover for voip providers (i.e. Dial() doesn't give enough options)
11:59AM 1 Using asterisk as voicemail system for SER
11:52AM 0 IBM T30, Redhat 9, Gnophone, mono PCM, Internet PhoneCard
11:00AM 5 Mine strangest asterisk problem ever ....
10:54AM 0 Asterisk voicemail problem
10:03AM 0 asterisk-addons mysql
9:51AM 0 Replacing a Cisco Call Manager
9:29AM 2 Hang-up Supervision (UK)
9:12AM 1 TE405P PRI B-channel resets
9:01AM 0 IAX, MYSQL - Rejected connect attempt from
8:54AM 0 Asterisk PRI messages
8:46AM 1 Asterisk Receptionist - Lite - CallerID Source code
8:24AM 9 ISDN BRI with National (north america) Signalling
7:47AM 24 Dyn Exten
7:09AM 4 NetworkWorld article on Open Source Telephon y
2:50AM 0 Zaphfc and Fedora core 1
12:17AM 0 curious (and incorrect) caller*id behavior
 
Tuesday June 8 2004
TimeRepliesSubject
10:20PM 2 Learn To build IVR
9:14PM 10 AS5300 and Asterisk
8:59PM 1 HOBIC
7:21PM 32 Sending # and Asterisk Transfer Conflict
7:06PM 23 NetworkWorld article on Open Source Telephony
4:54PM 1 Asterisk CallerID app (win32)
1:22PM 0 Cisco 7940 doesn't register
12:54PM 5 SMS in the UK
11:50AM 0 Call centers using Asterisk
11:30AM 0 TDM400P hangup / ringing detection problem
10:27AM 2 HOW-TO DIFF
10:04AM 8 Don't want a ring before voice menu
8:40AM 12 New version of DIAX (0.9.8a) available now for free download
8:38AM 0 Echo problems using AVM Fritz!PCI Card
8:24AM 11 iaxtel 1-800 gateway down?
8:19AM 16 CDR for transfered calls
8:13AM 0 Unable to call other SIP Phone
7:41AM 0 Camp On configuration?
7:05AM 6 Integration with a Siemens HiCom 150E / HiPath 3750
6:57AM 3 grandstream ringtones - makering.pl usage for 1.0.50
5:51AM 3 Outgoing call via Fritz!
3:42AM 3 E100P R2 signaling
2:19AM 2 Meetme2
1:39AM 12 makering.pl
1:17AM 0 Is there a problem with iaxtel?
12:03AM 0 How path latest CVS apps Makefile on order to compile app_rxfax and app_txfax
 
Monday June 7 2004
TimeRepliesSubject
11:37PM 1 illegal instruction - on Via board
10:43PM 4 Mediatrix 1204 Configuration
10:09PM 3 sip device discussion and reviews
9:34PM 6 dialplan experts needed
9:25PM 0 Asterisk Receptionist Lite version
9:24PM 0 asterisk to broadvoice?
8:27PM 2 Seeking Volunteers for an Intro to Asterisk Course
8:07PM 2 IAX Won't Pass Caller ID
5:41PM 0 re: Voicemail and Cisco Phones
5:33PM 2 chan_capi 0.3.3 compiling error
5:27PM 0 SIP registration issues - Ugly workaround
2:35PM 3 slightly OT: VoIP more expensive than Call-By-Call
1:58PM 4 meetme application
1:55PM 0 Application possibilities
1:49PM 1 pseudo zap channel - how to get rid of it ?
1:30PM 11 Network Sniffing Calls for recording
1:26PM 1 Voicemail missing playback options
12:21PM 5 hdlc setup routing question
12:15PM 5 Modem Calls
11:43AM 15 Compiling Asterisk with G.723.1
11:27AM 2 AGI + g729A
11:02AM 0 cisco reinvite
10:10AM 1 Grandstream Codec Order
9:57AM 1 control which * pbx to use
9:10AM 2 Problem with rxFax
8:10AM 2 videosupport = yes -- how to use it?
8:08AM 2 AVM B1 and PTP mode
7:42AM 0 DTMF X100p to sip GS
6:15AM 0 re: Voicemail and Cisco Phones
6:15AM 22 Fax via email
5:52AM 3 re: Voicemail and Cisco Phones
5:30AM 2 Module nonsense (zaptel, wcfxs and wxfxo)
4:58AM 1 Zaphfc and BRI problems in Portugal...
4:28AM 13 chan_capi and DDI (Anlagenanschluss)
3:11AM 9 Voip-talk?
3:10AM 0 Updated: Advanced German Configuration
3:08AM 1 Multiple DDI & Hunting on Analog Lines ( UK)
2:54AM 0 FW: Problem with Asterisk PRI forwarding to SER
2:42AM 0 (Redirected to -Users) Re: [Asterisk-Dev] load_module error with chan_oh323
1:29AM 2 IAX calls dropout on button press
1:24AM 4 Multiple DDI & Hunting on Analog Lines (UK)
12:19AM 2 isdn4linux, NETjet, chan_modem help needed
 
Sunday June 6 2004
TimeRepliesSubject
11:52PM 8 nat=yes
11:18PM 3 Dial plan help
8:00PM 3 illegal instruction -via c5
6:31PM 0 Incoming calls not showing up in user specific CDRs?
4:55PM 0 AM-Web working?
2:02PM 6 BRI In the states
1:56PM 3 Analog Bridged Calls Pulsate
1:38PM 5 Zapata?
7:47AM 1 Incoming call voice data
3:42AM 0 *** Asterisk Sunday News: The SIP NAT Special
 
Saturday June 5 2004
TimeRepliesSubject
11:39PM 3 FXO answering quicker
6:18PM 2 FWD network from Asterisk through NAT
12:38PM 0 FW: Meetme with moderator
11:22AM 9 Configuring cisco 7940
10:50AM 0 Re: Asterisk-Users digest, Vol 1 #4041 - 11 msgs
5:45AM 0 DSP Tools Technical Support
3:47AM 0 Immediate partial pattern match
3:13AM 0 change cisco ata 186 dial behaviour
1:34AM 1 ISDN and incoming MSN
 
Friday June 4 2004
TimeRepliesSubject
9:35PM 3 illegal instruction
9:09PM 0 CFDA from cell phone to SIP line in Asterisk PBX
8:08PM 3 * to Vonage Connection anyone?
6:55PM 2 Cisco 7960 XML/Configs
3:50PM 2 CODEC and Fax
2:59PM 7 Voicemail and Cisco phones: Dialplan example
1:40PM 0 Appradius Installation
1:32PM 0 (no subject)
1:18PM 15 Grandstream 1.0.5.0 Firmware: SIP Register option gone
12:16PM 0 bitnet niagara presentation - might interest anyone local
12:07PM 0 Supervision Issue With Asterisk/Sipura/Talkn
12:00PM 5 QoS in Cisco
11:53AM 2 Recommendation for sip phone
11:29AM 1 RE RE: Asterisk Receptionist manager program.
9:36AM 10 MYSQL asterisk configuration
9:16AM 3 rxfax crashing asterisk and YES I'm using an approved libtiff :-)
9:12AM 0 IAX termination in 602 or 520
7:25AM 3 Cisco 12 SP+ and Asterisk?
7:07AM 2 Mystery PRI NOTICEs & WARNINGs
7:07AM 0 miserable time with Cisco ATA 186
6:26AM 4 Help, Ideas and Ready for use Solutions
5:48AM 4 (possibly) new use for asterisk
3:00AM 1 ast_log(LOG_DEBUG
2:16AM 0 Newbie question about dialling PSTN numbers from SIP clients
1:30AM 1 Strange connection to the outside...
1:26AM 0 bri stuff Issues
1:16AM 1 Newbie questions about ISDN&zapata.conf, outbound dialing, TDMoE
 
Thursday June 3 2004
TimeRepliesSubject
11:22PM 2 Asterisk & fax-out
7:43PM 9 miserable time with Cisco ATA186
2:02PM 1 parking in multiple contexts
1:45PM 8 Hardware Transcoder
1:21PM 1 Call Originate from Manager application.
12:07PM 0 Problem with vmail.cgi
10:19AM 2 X100P hangup, not available 60 seconds
9:07AM 0 New ASTGUICLIENT released: 1.0.2
8:42AM 0 Agent Groups
6:43AM 19 Problem with T1 PRI line resetting/dropping calls.
6:42AM 0 7960 problem call to 7960
5:41AM 11 CALLERIDNUM not passed over?
5:34AM 0 zttest never get 100% accurancy
4:58AM 7 TE410P Q.931
4:40AM 0 Asterisk + E100P in Sweden
4:31AM 1 Small * issue
4:23AM 5 Time based calls charging and "reserved" numbers up to 999!
3:36AM 3 Asterisk & SER (www.IPTel.org)
3:34AM 12 DSP Coding
12:43AM 0 Text to speech on Asterisk - AT&T?
12:26AM 0 Preserving received digits during a fax match?
12:02AM 0 Any Idea why I am getting one Ring on my Analog Phone attach to Rhino Switch after Hangup
 
Wednesday June 2 2004
TimeRepliesSubject
9:55PM 1 Cisco VG200 & mgcp
8:54PM 3 Hot keypad on a Cisco 7960
6:18PM 2 (no subject)
6:14PM 0 WaitforDigit give ring on Analog Phone
5:54PM 0 MWI for Zaptel / libpri -> BRI
5:52PM 0 VON Developers Conference
5:23PM 1 X100P to hardware PBX
4:52PM 1 oh323: Failed to create smoother
4:00PM 2 IP Phone with multiple accounts on same instance of asterisk
3:20PM 0 mgcp.conf reference manual
2:23PM 0 Polycom SoundPoint IP 300 with Asterisk?
1:42PM 0 Stutter dialtone on TDM31B (TDM400P)
1:40PM 2 cisco ata-186 behind NAT
1:37PM 2 Problems with IAX Clients, HELP ME PLEASE.
12:14PM 0 3com SIP phone issues
12:09PM 4 Zapata FXO always answers call?
12:00PM 3 Where I can find Grandstream v 1.0.4.68 firmware?
11:52AM 2 HandyTone with Asterisk
9:40AM 2 H.323 and cause code 'user busy'
9:36AM 4 DTMF and SIP
8:45AM 0 SIP and multiple line appearances
8:15AM 0 ast_rtp_read: Unknown RTP codec
8:01AM 5 asterisk process respawn
7:57AM 5 Meetme with moderator
7:46AM 9 ZyXEL Prestige 2000W SIP hangup fails
7:44AM 2 Asterisk and Sip/IP Phones
7:22AM 4 Splicing audio clips into one stream
6:56AM 2 Problem compiling ZAPTEL on Linux 2.6.6
6:44AM 0 FireFly - no sound after first call
5:44AM 2 Feature request for integrating an OSS (Operations Support System) and Asterisk
5:41AM 1 isdn configuration
5:40AM 49 DNS SRV records
5:10AM 2 Fax Recognizion without Answer? How to Supress this?
4:30AM 0 Script to import Master.csv in the MySQL database - a short HowTo
2:43AM 1 Bluetooth headsets/phones.
2:28AM 3 Asterisk with Ericsson MD110 PBX
2:25AM 2 "403 Forbidden" between two softphones on same Asterisk
 
Tuesday June 1 2004
TimeRepliesSubject
10:33PM 2 problems with TDM400P
9:33PM 0 Asterisk Receptionist
9:04PM 1 determining cause of dropped calls?
7:46PM 5 Simultaneous ring internal extension and external phone number?
7:45PM 1 Feedback needed! FindMe/FollowMe FeatureSpec.
7:39PM 5 Multi process of *
7:34PM 0 Message light and paging on Zultys ZIP2, Uniden UIP200 time offset
7:06PM 2 extra FXS?
7:04PM 0 MOH From Line In on a sound card
6:29PM 2 VoIP phones in Australia
5:54PM 0 free sip termination
5:49PM 6 iax codec problem
5:16PM 1 Help in direction
3:44PM 0 SIP response 488 to special ext/pri?
2:34PM 6 Syntax for 2 ISDN Cards
1:14PM 68 Feedback needed! FindMe/FollowMe Feature Spec.
12:33PM 1 Zap and call pickup -- it don't work.
12:27PM 10 Adtran TSU 600
12:17PM 0 Re: Here
11:58AM 3 HDLC
11:38AM 0 Detecting Events in queues
10:24AM 1 SIP vs. SIP :-(
10:14AM 2 Router, Firewall, SIP Rewriter, and GnuGK
10:09AM 0 Presentation, Asterisk support in Montreal
9:57AM 0 Record Application Problem
9:51AM 18 BroadVoice usage?
9:13AM 1 Testers for chan_misdn searched
9:01AM 3 Difference between native and 3rd party h323 channel driver ?
8:49AM 0 changing the ip address of an asterisk pbx
8:27AM 0 System blocked when execute "asterisk -c"
8:27AM 15 Some (lack of) answers regarding the wakeup call application...
8:14AM 0 Unsupported Media error from iConnectHere
7:39AM 0 Call Transfer over Fritz!-ISDN Card with i4l does not work
7:37AM 0 MGCP Clients
7:17AM 2 R: Hyperthreading?
6:41AM 1 ISDN in Venezuela
6:22AM 0 Variable: in Originate via Manager API
6:04AM 0 Sipura-SPA2000 background noise
5:32AM 0 short delay before voice starts after ring
3:49AM 1 E100P isdn pri installation
3:01AM 3 Controlling SIP mobile extensions.
2:11AM 5 @mydomain.com
1:35AM 22 Hyperthreading?
1:22AM 1 D-Link DPH-100S
1:21AM 2 Stuck SIP channels? -> SIP show channels
12:36AM 0 Réf.: RE: SIPP Load testing
12:33AM 0 australian enum...
12:23AM 2 E1 Connection breaks