asterisk users - Oct 2004

Sunday October 31 2004
11:42PM 74 Linux and Windows
11:09PM 1 Inbound numbers question
9:53PM 1 record all calls
9:36PM 10 Amount of time asterisk take to pickup incoming call on ZAP interface
7:41PM 1 iax2_read: I should never be called!
7:35PM 4 goto() results in invalid extension
7:30PM 3 Embedded Asterisk Paper Complete
6:21PM 1 VoiceXML / Asterisk
6:19PM 0 Tool for viewing Message waiting status
3:57PM 0 UK Asterisk Consultant visiting San Diego
3:42PM 6 Asterisk and GnuGK on the same box?
3:36PM 12 UDP Fragmentation Problem
2:21PM 1 ISDN card advise
2:17PM 0 pri usage
1:56PM 3 I need help
12:01PM 3 Dialogic
11:47AM 0 norwegian sounds for Asterisk
10:01AM 1 Zapateller broken in ver 1.0.2?
4:04AM 1 Can't install the mfcr2 support correctly
2:56AM 0 make transfert and hold with FXS device
2:45AM 2 G.711alaw to iLBC
12:18AM 2 asterisk RELOAD option stability
12:09AM 4 video conferencing with sip
Saturday October 30 2004
11:13PM 9 Cannot start asterisk - CAPI issues
5:21PM 0 echo with long distance
5:01PM 0 voice delay with isdn
3:08PM 2 IAX2 bandwidth efficiency calculations from Farfon
1:37PM 51 How far is IAX to be a Standard
10:28AM 3 Wireless phones connected to VOIP DECT basestation
8:53AM 9 confusing info from Digium and asteriskdoc about setup of TDM11B
8:51AM 0 Latency/delay on IN1002 - PA1688 phone
8:44AM 1 re: asterisk SER and grandstream
8:35AM 1 DTMF and codec
7:25AM 3 iax registration & port number
7:22AM 0 SIP to SIP echo problem
6:24AM 4 loss concealment
6:16AM 0 Dialogic Card + TP100B
4:46AM 0 Ang: FXO flash from sip phone
1:50AM 0 FXO flash from sip phone
1:47AM 4 HELP: problem making calls from legacy pbx to cisco sip phone via asterisk
12:37AM 6 Wireless phones connected to VOIP DECT base station
12:08AM 0 g723 in pass-thru mode asterisk
Friday October 29 2004
10:00PM 3 This is VERY interesting -- A gateway between proprietary digital sets and SIP?
7:11PM 0 Asterisk works with SER
6:00PM 0 E1/R2 application in Brazil: Asterisk compilation with libunicall
5:50PM 2 Swissvoice IP10S opinions?
3:55PM 0 TDM channel shows Offhook when I plug it to thetelco
3:47PM 20 high-capacity systems / trouble with Tyan
2:34PM 0 My asterisk box is behaving funny!
2:14PM 5 Polycom failed registration - Cant figure out whats wrong
1:56PM 4 Rewriting a telephone number for remote dial out
1:17PM 8 non blind call transfers
1:10PM 1 FW: VoiceEclipse vePipe inbound config question - Authorization failed for user"####"
1:09PM 2 DISA() anyone?
12:55PM 2 Cisco PRI Gateway Problems
12:55PM 0 SIP Friends w/ MySQL
12:51PM 1 Suggestion re: SIP/NAT/*
12:39PM 7 Polycom IP 500 Config Files - searching
12:23PM 2 queue_log analyzer
11:39AM 6 Is NuFone messing up for anybody else?
11:22AM 1 Newbie question - pickup call waiting on an analog trunk
11:16AM 2 Ambient MD 3200+incoming problem
11:13AM 5 Anyone using Voipjet?
10:59AM 1 "Hiss" on Line, No ringing thru VoicePulse?
10:51AM 0 sip phones...
10:06AM 0 VoiceEclipse vePipe inbound config question - Authorization failed for user"####"
9:32AM 5 Snom 190/220
9:28AM 3 AddQueueMember and call distribution
9:06AM 6 Outbound IAX calls stop ringing remote phone, yet can still pick up
8:54AM 0 Suggestion re: SIP/NAT/*
8:07AM 0 Re: Asterisk-Users Digest, Vol 3, Issue 410
7:37AM 0 Nothing but static on new install with TDM11B
7:18AM 1 wake-up
7:13AM 0 voicemail transfer on busy fails
7:10AM 0 Security question (permissions)
7:02AM 1 Grandstream HT486 and FAX
6:48AM 1 Queue.conf, maxlen = 5 , but what happens with the 6. caller ?
6:38AM 1 sip <-> h323 audio problem
6:20AM 1 FOP 0.17 - Agent setup
5:41AM 3 ISDN EDSS1 protocol support
4:58AM 1 failed
4:35AM 2 eyebeam video
4:09AM 0 sip.conf registration
3:52AM 0 Asterisk Sipphone
3:49AM 2 chan_sccp and Cisco 7940
3:33AM 16 Echo in CAPI channels
3:28AM 0 Automatic codec selection
2:49AM 0 call another server
2:38AM 10 Asterisk with Nortel BCM
1:37AM 4 Modifying CDR data?
12:49AM 0 Problem with Dial (in v.1.0.2)
Thursday October 28 2004
11:54PM 1 Dropped call
11:48PM 8 Do I *need* to compile zaptel?
11:21PM 0 Mysql support
11:12PM 1 Snom200 strange sound problem
7:25PM 2 disa hangs up on me
7:05PM 4 question about asterisk
5:11PM 4 Analog answering machine hangs up early
4:05PM 1 E100P Call Deflection - Redirecting an Incoming Call with ISDN (Resend)
3:19PM 2 Automatic code selection
1:47PM 0 SMDI and Asterisk
12:52PM 0 729 -> 711 failover?
12:03PM 0 New astGUIclient version released 1.0.5
11:42AM 1 * connectionto home automation server
11:13AM 0 Suggestion re: SIP/NAT/*
11:09AM 8 Queue question
10:49AM 0 Registration Fail
9:26AM 1 Multiple Bandwidth Providers and Asterisk
9:16AM 0 Need Asterisk to generate ringing tone on inbound SIP calls
8:51AM 1 MFC/R2 Argentina variant ANI problems
8:08AM 0 how-to invoke the "Busy" voice mailbox menu in Asterisk
7:55AM 16 TDM400P hardware problems
7:28AM 0 Re: call progress - what are the sticking po ints?
6:42AM 0 Sipura 3000 tone table settings for Australia
6:27AM 0 Ex-girlfriend-logic
5:00AM 0 carrier deployment of SIP
3:58AM 0 mcedit syntax for asterisk conf files
3:31AM 1 disable second call / call waiting via SIP
3:20AM 3 Polycom IP 500 and DTMF
3:16AM 4 Using AVM C4 with fewer than four lines?
2:24AM 5 AW: AW: Firefly 1.9.6 released
2:20AM 0 RE: Why I can't hear anything from my sjphone asanh323 endpoint?
2:08AM 0 Getting result codes of SIP-dials
1:26AM 4 HiPath Wild Card T110P interface
1:25AM 0 integrating Asterisk to existing TDM-based PBX
1:02AM 0 Why I can't hear anything from my sjphone as anh323 endpoint?
12:39AM 3 Nightmare on disconnecting Zap and SIP channel
12:08AM 2 ISDN-Problem with Quadbri behind Tenovis
Wednesday October 27 2004
9:51PM 2 Problem with AstTapi
9:07PM 2 No dial tone from fxs port
8:38PM 17 WRT54GS zaptel timing device
8:15PM 2 [PATCH] DUNDi for 1.0.2
7:41PM 1 Why I can't hear anything from my sjphone as an h323 endpoint?
7:27PM 13 call progress - what are the sticking points?
6:47PM 0 IAX support added to AMP
6:39PM 4 New Strategy in App_queue
5:49PM 0 AudioCodes MP-108 (or MP-1xx) FXO gateway
5:44PM 2 Asterisk to Asterisk using SIP?
5:03PM 3 Asterisk-cvs does not compile on Red Hat 9
3:54PM 4 where do i find openssl-devel to mandrake 10.1
3:18PM 8 Type of T1 for T100P card
2:05PM 2 IAXy Call Waiting Disable
1:59PM 1 SRV lookup fails on dyndns wildcard domains
1:50PM 0 G.72[69]
1:46PM 0 Asterisk - Store and Forward Configuration with re-recording some part
1:32PM 10 Can bad person with SIPp attack Asterisk ?
1:19PM 6 Transfer caller
1:13PM 1 SIP vs MGCP
12:39PM 5 Motorola Vt1000
12:13PM 1 OT: The ideal switch for VOIP
12:05PM 4 Funny thing with LinkSys / IAX2
11:54AM 0 Call Waiting Via Sipura to X100P
11:42AM 4 AT&T Cordless VOIP Phone?
11:29AM 1 Zap issues...
11:28AM 0 [OT] How trustworthy is Yoda Communications in Taiwan?
10:30AM 7 Grandstream and CallerID - sorted
10:27AM 3 Sparco Office Supplies...
10:26AM 7 GSM Audio Files on Windows w/o Quicktime
10:08AM 0 Need help with extconfig (take 2)
9:52AM 0 neg txgain makes * oblivious to incoming calls?
9:48AM 0 Help needed with Extconfig, mysql
9:27AM 0 chan_sip2 won't compile
9:10AM 1 Zaptel channels
9:09AM 0 app_valetparking
9:01AM 5 RE: [OT] Sparco Office Supplies... (yeah right)
8:55AM 1 Remote Voicemail
8:33AM 0 How to install wakeup?
8:31AM 0 Agent Groups In Queues
8:25AM 1 Simple Asterisk Config Help withx100p
8:14AM 11 [OT] Sparco Office Supplies...
7:58AM 1 Multiple SIP gateway accounts
7:33AM 1 Directory () Problem --revisited
7:27AM 1 can't run ztcfg
6:35AM 0 RTP port mismatch and Astwind card support
6:18AM 0 Cannot Call IAX Softphones
5:43AM 2 OT COs/Providers Cannot Reach Others
5:41AM 0 Error when starting Asterisk (Loading module failed!)
5:01AM 1 pickupgroup and callgroup on zapata.conf , how they work ?
4:49AM 0 zaphfc bristuff ISDN transfer
4:05AM 0 RTP Ports mismatch & Astwind modem support
4:04AM 1 queue reports?
2:40AM 2 AW: Firefly 1.9.6 released
2:28AM 2 UK CallerID
2:27AM 8 TDM400P - TE405P- configuration issue
1:51AM 9 test telephone numbers
1:43AM 0 BillSec and CLID in CDR Problem
12:16AM 2 Firefly 1.9.6 released
Tuesday October 26 2004
9:55PM 1 Asterisk Intro for newbies
8:35PM 2 New card - TE110P?
7:39PM 3 Can I pick up a phone that rings from my phone?
6:56PM 0 queues.conf - Agent groups specified with : not @, difference in working?
6:51PM 3 Anyone playing with E1 channel bank?
6:50PM 1 Zaptel FXO channel picked up too early => no audio
6:38PM 1 Asterisk 1.0.2 (again)
6:22PM 2 Performance (Cisco AS5350) or Price (Wildcard TE410P)
6:20PM 2 Digits being lost going out POTS line?
6:14PM 0 torisa startup troubles
6:13PM 1 E100P Call Deflection - Redirecting an Incoming Call
5:05PM 2 Re: Asterisk and Broadvoice, no incoming voice (Brian Weaver)
4:42PM 5 Should I be worried? Newbie Warning
4:33PM 0 Phone Cutout Problems
4:04PM 0 feature request - sipfriends, iaxfriends...
3:44PM 2 7912G Ringers?
3:23PM 11 voicemail.conf
2:56PM 1 Dial timeout off by factor of two?
2:40PM 0 Determining that call was transferred
2:40PM 7 SUSE 9.1 and Zaptel
2:32PM 0 extensions.conf question
2:24PM 25 polycom IP 500/600
2:14PM 1 ASTCC no sound
2:02PM 2 Bandwidth Load Balancing / Dundi
1:31PM 3 Uniden UIP 200 not ringing
1:28PM 0 extension.conf on mysql
1:11PM 0 Graphing Date/Source/Destination nicely
12:59PM 7 De-Centralized / Distributed Conferencing App
11:10AM 0 outgoing spool dial local channels then actual dial from extensions.conf
11:03AM 0 Problems with Directory()
11:02AM 2 Asterisk + MGCP + Cisco E1 gateway
10:11AM 14 Gentoo
9:56AM 2 Begin to begin
9:48AM 7 ASTCC with password
9:47AM 21 Problem getting zaprtc installed on a mandrake 9.2
9:38AM 5 Can't connect even though its running..
9:05AM 23 G.726
8:41AM 1 Succesul outgoin calls using UniCall
8:40AM 0 Asterisk as a simple Message Store and Forward that Sends VoiceMail to a Group.
8:25AM 0 Where to catch events like Dial, Ringing, Transfer, Hold, Forward, Hangup, Park, UnHold, Answered, No Answer, etc.
7:56AM 19 Asterisk 1.0.2
7:10AM 2 SIP Conferencing Server
5:56AM 3 HANGUPCAUSE macro..
5:42AM 6 RDNIS
5:19AM 2 ASTERISK and VoiceXML
5:14AM 0 R: E1 configuration problem
5:02AM 4 cisco router & *
4:59AM 18 Need HELP to put * in use for good cause
4:06AM 0 Snom 200 SIP Settings etc
3:39AM 0 Once again: Problem compiling ZPAHFC with Suse 9.1, Kernel
3:38AM 4 E1 configuration problem
3:35AM 8 X100P noise on ADSL line.
2:52AM 12 H323.conf question
2:32AM 1 Little problem with AGI
2:23AM 2 IAX trunking clarification...
2:04AM 3 Can contexts have wildcards too?
1:17AM 1 snom200 & dial plan
1:15AM 0 PRI events
1:14AM 2 sip_xmit errors...
Monday October 25 2004
9:12PM 25 GPL thoughts
7:08PM 2 Re: [Asterisk-Dev] How to submit a patch?
6:43PM 4 Routing based on Caller ID
5:38PM 1 Re: Benjk's Question "Why FXS"
4:15PM 5 Re: Benjk's Question "Why FXS"
3:56PM 8 Agents allowed to transfer but * just hangs up!
3:47PM 2 Snom Phones and asterisk
3:34PM 1 Re: Benjk's Question "Why FXS"
2:56PM 0 Santa Cruz, Bolivia?
2:26PM 1 Company Directory/Dial by Name
1:48PM 2 GUI for Asterisk.
1:30PM 7 Auto-Login/Auto Answer
1:11PM 5 Transfering Calls
1:04PM 0 using @ vs : for Agent groups....
12:54PM 0 Augh!
12:52PM 2 SNOM 190 - strange voice problems
12:49PM 1 G729 Error. => No path translation.
12:10PM 2 Multiple Accounts on a Softphone
11:49AM 0 Softphone for QNX?
11:48AM 1 Call Parking + Agents (or queues?) does not work
11:36AM 2 echo questions
11:32AM 8 Error starting Asterisk.
11:00AM 5 Can home/office have same extension
10:54AM 1 Setup two Asterisk servers with MGCP
10:24AM 1 Rhino channel bank configuration with T100P
9:54AM 10 Nortel Phones.
9:40AM 10 Help Instalation
9:20AM 0 voicemail: fromstring and delete
8:53AM 0 Queue anounce time
8:46AM 1 ATCOM froze
8:20AM 1 SayNumber application - in spanish?
7:58AM 4 Multi-office topology suggestions
7:53AM 0 DNID in chan_sip.c
7:50AM 0 Digium TheVoice recordings' sound
6:53AM 23 Vonage Softphone--outbound calls work, inbound do not
6:46AM 1 sip.conf user withdefaultip=....worksbutcalleridnot settable (= ip)
6:46AM 3 Help for Newbie?
6:44AM 1 Problem with asterisk-oh323
6:36AM 2 Digium Wildcard T1 Compatibility (ethernet f or T1 cables)
6:25AM 3 sip.conf user with defaultip=....worksbutcallerid not settable (= ip)
6:06AM 1 sip.conf user with defaultip= .... worksbutcallerid not settable (= ip)
5:43AM 1 Fwd: IAX wireless problem
5:33AM 1 sip.conf user with defaultip= .... works but callerid not settable (= ip)
3:12AM 3 Bandwdith usage
3:03AM 2 AST doesn't start after update from 0.5 to 1.0
2:50AM 2 3com with Asterisk
2:47AM 2 protection
2:28AM 4 Quintum A800 OH323 problem
2:23AM 1 sip users registering fails
2:12AM 2 CDR Dokumentation
1:08AM 2 I have Asterisk & Hylafax on a server. What else do I need...?
Sunday October 24 2004
10:57PM 3 Snom200 & VMail (MWI)
9:49PM 1 G729 -> G723.1
9:43PM 0 Xlite works, Asterisk sometimes not
7:02PM 5 Several FXS Ports
6:48PM 3 Howto get voicemail $VM_ vars into externnotify script?
5:11PM 12 ACT Gateways
5:01PM 1 Connection to a H323 system
1:24PM 3 Iaxy authentication
12:20PM 0 Reload cause Sound Volumn becomes very loud
12:11PM 0 Asterisk Prepaid with MySQL
11:50AM 1 getting cid from spa3k pstn to *
8:11AM 1 Failed to authenticate on INVITE to '"601" ...
7:08AM 0 (iax|sip)friends in extconfig?
6:56AM 0 Error when compiling asterisk-oh323
6:44AM 7 random crash at startup
6:18AM 0 How to create Groups/members and do Conferencing?
5:20AM 0 Problem compiling ZPAHFC with Suse 9.1, Kernel 2.6.5
5:14AM 4 chan_sip CallerPres support?
3:37AM 23 Digium TheVoice recordings' sound terrible
12:59AM 0 bristtuff segfault
Saturday October 23 2004
11:08PM 3 Asterisk, ATA-186 & /
11:04PM 8 G.729 on YDL and MacOSX
10:42PM 0 One approach to SIP dialing through asterisk
9:53PM 0 Outlook reports internal error after using AstTapi
7:29PM 3 Fedora 2, Kudzu and X100P
7:14PM 0 Hardware (and apple YDL G.729)
4:46PM 14 Hardware
2:32PM 21 Asterisk and Broadvoice, no incoming voice
2:31PM 2 Re: Webmin for ASTERISK and QOS and call quality
2:18PM 5 Geotel integration with Asterisk
10:35AM 1 Zultys Zip 2 Setup
9:46AM 14 doublehash patch for 1.0.1
9:37AM 3 Cheap hosted servers and Asterisk
9:09AM 0 Quintum ASM400, ASM200 and ASTERISK
9:07AM 0 Digum board TDM to Phonejack --Quicknet --Trandsfering calls.
8:54AM 1 Support for reception of "send url" in SIP clients needed
8:54AM 0 Need help with RDNIS on ISDN PRI
8:10AM 1 IAX wireless problem
2:19AM 0 * dies with QuadBRI
2:06AM 30 iLBC/PCM16 Huge Cost
Friday October 22 2004
9:09PM 1 spa3k: cid vs authid
8:29PM 8 chan_sip changes affecting ACK? - Bug?
7:24PM 1 new quad T1 install
7:08PM 1 Asterisk-OH323 Invalid format RTP
5:28PM 0 iaxComm now supports iLBC,Speex
3:06PM 0 (no subject)
2:39PM 10 Direct SIP connection to Vonage service
2:12PM 1 Cannot send # to far end, asterisk intercepts.
2:02PM 16 res_config
1:45PM 0 "zt_get_index: nullok is not asserted" could led to freeze?
12:10PM 2 testing open ports 10000 - 20000
12:02PM 4 DTMF G729
10:52AM 5 One E1: 10 time-slots for voice (ZAP), 10 for Internet PPP (data) and 10 slots for Internet PPP (VoIP)
10:43AM 0 Best way to transfer incoming sip calls to other sip number?
9:26AM 10 How useful is the screen on IP phones?
9:06AM 3 Fw: SPA-3000 Disconnect tone detection in France ?
8:22AM 3 Queue / Agent Problem
8:06AM 1 Updates in the asterisk - cvs mailing list - Head or Stable?
8:01AM 0 Detecting Busy when dialing out on ZAP channel.
7:28AM 1 installing install isdn4k-utils from source ?
7:19AM 26 Hardware Recommendations
7:15AM 1 New. Testing?
5:26AM 1 IAXy echo avoidance/cancellation
4:01AM 11 Question about ISDN reason codes
2:56AM 7 MusicOnHold() - how to restart player from the beginning on each call? (fwd)
1:57AM 2 common numbers ?
1:46AM 1 Newbie General questions
Thursday October 21 2004
11:55PM 1 827-4V voice ports, asterisk and hookflash
11:48PM 0 problem with caller-id
10:52PM 3 Connecting to Commander NT132
7:50PM 3 Can I do that?
7:37PM 1 AGI comand channel status]
7:31PM 13 * and Verisign SIP-7 service
6:04PM 0 Anyone getting RDNIS on Lucent 5ESS
5:48PM 0 Help with asterisk-oh323 driver [resend]
2:49PM 23 Digium Wildcard T1 Compatibility
2:49PM 10 Fax detection in voip channel
1:22PM 6 automatically logging on/off agents
12:31PM 0 question about type=user in sip.conf
12:20PM 0 backgrounddetect command - what about busy
11:53AM 0 doubts regarding monitor command
10:47AM 9 answer on # key?
10:35AM 15 KSS/BLF on Asterisk
10:26AM 1 Load test IAX
10:19AM 3 sip call echo cancellation
10:17AM 4 modem question
9:34AM 12 Grandstream Flashing (different issue)
9:27AM 0 Polycom IP600 features
9:26AM 0 Local battery phones and Asterisk
9:16AM 0 Yoda SIP Devices: IAD100, IAD200, IAD211, IAD400 and other
9:05AM 1 Freshmaker failed register test
8:59AM 13 SER or not to SER?
8:43AM 1 AGI comand channel status
8:39AM 0 Queues Problems
8:37AM 0 Video Phone issues registering with asterisk
8:36AM 1 SIP - H.323 connection
8:18AM 0 gotoif regex?
7:58AM 1 MWI - Sip phones
7:54AM 0 SIP / H323 connectivity
7:41AM 3 Voicemail and ast_data
7:29AM 2 Asterisk and Nortel Meridian interconnection
7:09AM 8 first tries !
7:02AM 3 Press the * key to repeat
6:06AM 7 asterisk & ipv6
5:51AM 0 sip+iax+firewall
5:37AM 1 Performance with ASTCC.
5:34AM 39 G.729 licensing/patent?
5:32AM 1 beginners questions
5:06AM 2 Does use of riser cards in racks affects performance of PCI telephony cards ?
4:06AM 15 Calling IAX client behind NAT
4:05AM 0 Voicemail: Unable to open digits/hundred M
3:37AM 1 Manager API / Agents
3:17AM 0 Special Callback feature
3:03AM 0 Clients can login twice
2:19AM 4 2 * RxFax <-> TxFax
1:28AM 0 CISCO router closes the connection before starting conversation
12:18AM 0 SER + Asterisk Attended Call Transfer
12:15AM 0 "Number" of caller
Wednesday October 20 2004
11:37PM 0 Do any one have developed Asterisk ebuild for Gentoo
9:35PM 1 Flash Panel version greif with ming et al
8:30PM 0 Routing calls based on monthly usage?
7:57PM 18 grandstream 102 flashing
7:29PM 1 Help with asterisk-oh323 driver
6:58PM 0 Delay in outbound SIP call
5:24PM 1 H323 Connection to Splicecom Maximiser
4:35PM 0 codec problems with astcc and not with sip trhough aix
4:20PM 0 how to detect a busy line using analog ports TDM04B (station ports) and using outgoing spool to start the call
3:57PM 0 Grandstream phone - no dialtone
3:27PM 4 IP Phones -India
3:23PM 0 Received bad packet with bad udp checksum.
2:40PM 1 contexts based on time and date
2:34PM 0 Dialogic and TP card
2:10PM 10 app_conference
12:26PM 3 Newbie with new Project VOIp
11:34AM 1 Manger API flag from dialplan
10:58AM 22 IP Phone that OFFICIALLY support Asterisk
10:41AM 0 RE: Asterisk on a mid-sized flat corporate
10:36AM 3 ASTCC newbie
10:14AM 8 Graceful CLI/crontab reboot
9:48AM 7 X100P make phone ring on incoming sip call - possible?
9:37AM 2 Snom 190 "VMail Soft Key"
9:36AM 3 FWD via IAX2 -- anybody else experiencing timeouts?
8:48AM 5 manager interface to barge
8:38AM 0 Wildcard X100P/India
8:10AM 0 Trunking Gateway with E100P
8:05AM 0 RE: Asterisk-Users Digest, Vol 3, Issue 264
7:43AM 0 Personal Phone Gateway PCI and USB Phone.-
7:05AM 1 [OT] GSM patents
6:53AM 15 cannot call Grandstream
6:45AM 0 still riniging problem
6:30AM 3 grandstream handytone 286 problem
5:37AM 11 New Channel Driver: chan_bluetooth
5:14AM 7 Samsung DCS70 PABX
4:53AM 0 2 analog phone on FXS ports?
4:33AM 0 octoBRI problem
4:31AM 6 Attempt at country tones
4:03AM 6 X100P problems / UK Supplier of TDM400P FXO cards
3:36AM 0 SER problem?
3:06AM 1 SIP/SIMPLE, Jabber and Asterisk
2:21AM 0 Meetme room calls quiet for some lines/callers
1:08AM 23 cheap gig switch? smc, netgear, or 3com?
12:51AM 1 Load Balaning on 2 E100P cards
12:43AM 4 chan_mISDN problem
Tuesday October 19 2004
11:42PM 0 AW: CAPI and Asterisk (with AVM ISDN Card)
11:28PM 2 meetme latency
9:30PM 4 Sipura or X100P Option
8:38PM 2 Comments on proposed * setup
6:48PM 2 Anyone else seeing this?
5:29PM 0 OT: ATA 286 how to make the phone ring
5:25PM 0 7920 Help chan_sccp
4:51PM 2 Asterisk not sending full 11 digits dialed....
4:17PM 2 new here : logic of ser and asterisk all confused---longish
3:30PM 2 CAPI and Asterisk (with AVM ISDN Card)
2:23PM 0 PSTN -> PRI -> ASTERISK -> ASTERISK -> PRI -> Legacy switch ?
2:22PM 11 i extension
2:04PM 13 DUNDi in stable? (New subject)
1:37PM 39 Wonderful Success with PAP2-NA
12:55PM 7 X100P red alert
12:39PM 1 disabling "comfort noise", other odd thoughts
12:34PM 1 Asterisk on a mid-sized flat corporate network?
12:02PM 1 Some questions about channel banks signalling?
11:56AM 14 DUNDi on Slashdot
11:52AM 1 incorrect context called when receiving call on SIP channel
11:46AM 0 TDMoE Question?
11:23AM 7 Tranferring UniCall lines
11:18AM 8 Vonage with Nat - Working
11:13AM 11 How to ring internal extension?
10:53AM 0 Re: Asterisk-Users Digest, Vol 3, Issue 260
10:50AM 9 Asterisk on PowerPC v. Intel/AMD
10:10AM 1 Got SIP response 403 "Forbidden (From header is not a Trust host or gateway)" back
9:48AM 0 debug extension matching
9:36AM 20 Fax over IP doesn't works
9:02AM 0 Wellgate SIP product users - voice your concern!
8:56AM 2 chan_mISDN
8:18AM 17 Almost there--Remote connection
8:08AM 0 Problem with portaSIP provider
8:01AM 10 Problem with NFAS trunkgroups
7:52AM 0 Spandsp debug log question
7:47AM 8 test-driving G.729?
7:25AM 0 ExtensionState
7:01AM 1 Transparent SIP Server
6:37AM 0 txgain usage with T100P
5:50AM 2 SPA-3k & *
5:47AM 9 mISDN, CAPI, ISDN ???
5:31AM 0 Called number Callerid with Sip
5:16AM 0 Planet SIP Phone
4:23AM 0 I can't solve my problems with the IVR
4:15AM 9 Working Asterisk With Vonage
4:01AM 4 Setting CallerID on UK BRI line
3:18AM 6 About Supervised Call Transfert on GS BT100
3:10AM 2 AW: Follow me using a loop
2:54AM 1 record
2:20AM 1 Snom & Mass Deployment Config Problems
2:12AM 2 Follow me using a loop
1:17AM 0 Problem with DIAL command
1:07AM 0 Voicemail and AutoAttendant for a Nortel Option 11 PBX
Monday October 18 2004
11:19PM 2 SIP video support problem
9:16PM 1 ZapRAS from both sides
7:01PM 2 Anybody - please help me with this
6:52PM 0 Speex wideband mode
6:11PM 7 SMTP MTA suggestions.
5:34PM 1 SPAM Notice
5:32PM 0 anyone using a cisco 12sp+ or VIP 30
5:13PM 0 Cisco 7940 X IAX trunk
4:59PM 2 Specify location of ADSI Softkeys ?
4:31PM 7 verisign immitate e164
3:56PM 0 Routing over T100p Help Please...
3:20PM 3 IAX2 Nat issue, Any help greatly appreciated
2:52PM 0 SIP calls dropped (Ast 1.0 and Fedora core 2)
2:11PM 5 Quick question regarding daily restart of asterisk
1:49PM 14 VoIP over 1xRTT
1:02PM 0 Where to buy POLYCOM phones (forcing native bridge between SIP terminals)
12:41PM 0 Distorted Ringback
12:39PM 20 GSM to g729 Conversion
12:30PM 1 NMS AG4000?
12:19PM 0 Re: Asterisk-Users Digest, Vol 3, Issue 243
12:14PM 6 IAX2 Over Satellite => It works !
12:05PM 5 How to make asterisk send email notification of voicemessages?
11:58AM 4 Asterisk System Management User Interface
11:04AM 6 Where to buy POLYCOM phones?
10:58AM 1 Current Call information?
10:43AM 9 Voicepulse down for anyone else?
10:15AM 38 Polycom phones
9:25AM 14 New Realtime config and MWI
9:11AM 0 Problems with IVR digit recognition
9:04AM 0 Asterisk won't load some type of channel error
8:46AM 8 Call failed to go through
8:39AM 2 MWI for X-Ten Pro?
8:10AM 1 Transfer caller but on no answer, return to transferee...
7:49AM 0 Success with Swissvoice IP10S and SIP?
7:46AM 0 chan_iax2.c:5390 socket_read: Rejected connect attempt from
7:31AM 0 Sond problem on Second ISDN B channel
6:58AM 1 FireFly and GS-BT100 codec negotiation problem
6:31AM 4 Can't compile app_conference
5:44AM 1 Asterisk and video door phones?
5:37AM 1 Polycom IP-XXX with shared registration
5:21AM 7 mysql sipfriends and allowing individual codecs per user?
4:58AM 1 Svar: Re: Where to post SuSE 9.x startup script?
4:35AM 7 Capturing calls in asterisk
3:35AM 0 OH323 VoIP router connect debug question?
1:58AM 0 Xten eyeBeam Video codec
12:08AM 1 ACD/Queue Support with SIP Notification Messages?
Sunday October 17 2004
11:59PM 0 Sourcing H/W for Asterisk in India :: Digium/Intel Modems and IP Phones
11:54PM 0 cross-connecting dynamic channels
11:02PM 0 Asterisk AGI 'Get Data' escape digits not working on long records
9:56PM 0 Thailand
9:27PM 1 Problem In RTC Client When Used With Asterisk
9:05PM 3 chan_h323: forcing 20ms packetisation
8:42PM 2 Asterisk dropping last digit of phone number
6:50PM 0 Calling all Users to check out bug 2655
6:27PM 2 chan_skinny caller id.
5:55PM 0 Fax Redirection
5:02PM 0 chan_skinny callerID usage
4:59PM 1 SIP outbound dialing -- newbie alert.
3:23PM 0 OT - new SPA-3000 firmware out (v2.0.11a)
2:57PM 1 Cisco ATA-186 and Caller ID
2:16PM 2 Anyone else tried Speex 1.1 CVS?
2:04PM 4 DIAX 0.9.9b - now multi codec support
1:41PM 0 Wildcard X100P and Fedora Core 2?
1:14PM 2 Asterisk for a VOIP Provider?
1:12PM 6 can not compile chan_capi 0.3.5
11:16AM 0 IAX error messages
10:09AM 0 chan_skinny usage of callerid
9:39AM 2 Asked to transmit frame type 64, while native formats is 8
4:52AM 2 X100P, Dutch analong line, caller-id
2:59AM 1 Automated calling/Bridging and takedown in Asterisk?
Saturday October 16 2004
5:32PM 8 * Server behind a firewall - How To
5:17PM 0 Macro exmaple for saying digits in a more natural sounding way.
3:39PM 40 IAXy setup
2:17PM 3 Anyone using stanaphone? Having small problem
1:56PM 1 DTMF tones from CCME phone
12:52PM 2 Unusual protocols
10:05AM 7 FXO vs FXS question
9:02AM 1 Going to voicemail on noanswer
9:01AM 0 OT: Broadcom BCM1160 and BCM4318 released
8:39AM 9 Bandwidth control on a home office network
8:05AM 2 Asterisk Data Configuration Example 1
7:36AM 13 Sending broadcasts to all phones?
7:35AM 0 RE: G729 and Sipura
6:06AM 2 Zapata on PowerMac G4
3:39AM 1 sipgate cannot dial out / loop detected
3:27AM 8 G729 and Sipura.
1:47AM 1 Newbie question: asterisk and ser
12:35AM 1 Compatibility of Asterisk With Cisco 5350/5400Gateway
12:04AM 2 Japanese Translation of Asterisk?
12:03AM 1 Compatibility of Asterisk With Cisco 5350/5400 Gateway
Friday October 15 2004
11:34PM 3 help , chan_sccp wont build.
9:44PM 1 how can an AGI terminate a "STREAM file command"
6:01PM 6 sccp cisco 12sp HELP !!!
5:49PM 0 New Sipura Phone
3:49PM 1 Finding a gateway for home use (UK)
3:42PM 0 Congratulations to all comunity. Success: iaxy + openvpn + winxp routong.
3:34PM 1 Attempting native bridge .......
3:32PM 11 Cisco 7960 + 7914 - not worked
2:46PM 0 Looking for supplier for 7912G and 7940Gs reply in private
2:40PM 1 Help with Incoming calls
2:11PM 0 Avoiding deadlock
2:11PM 0 MySQL CDR addon problem
1:49PM 21 Cheap, Highquality IP Phones
12:50PM 0 Should ZAP channels pass CNAM to SIP?
12:46PM 4 HylaFAX v. spandsp
12:46PM 1 Unable to make E&M Wink work with T400P
12:44PM 2 [OT] HylaFAX and DID
12:42PM 0 My macros, etc.
12:39PM 3 Using my GrandStream remotely
12:29PM 10 res_odbc app_realtime
12:04PM 1 calling out from a remote * server
11:40AM 0 New asterisk user question
11:26AM 1 Sample advanced call routing standard extension
11:25AM 3 T100P Frame Errors
11:17AM 2 CHANUNAVAIL = CHANUNAVAIL doesn't eval properly
11:03AM 0 grandstream bt-486 can only dial with #
10:27AM 1 FXS port to use an Analog phone as a door phone.
10:19AM 0 RE: Cannot reach a SIP device (Sudhir Kumar)
9:56AM 19 New Open Source Project: Asterisk Management Portal
9:45AM 0 Cannot reach a SIP device
9:05AM 6 New Project - IP Phone Sources
8:56AM 0 Prepaid vs. Prepaid modified
7:56AM 2 app_queue & manager API
7:49AM 3 Cisco to * problem
7:22AM 3 Always get 401 Unauthorized..that normal?
6:47AM 0 RE: Cisco firewalls and softphones (Matthew Oulton)
6:34AM 0 Manager API and extension s
6:16AM 1 Invalid GSM data
6:05AM 0 Transmit re-INVITE before BYE is sent - why?
5:22AM 0 Looking for recommendations for a low-cost FXO toIP gateway.
5:13AM 0 Problem in DTMF Info message
5:07AM 1 Asterisk crashes on special Transfer with MGCP/ATA 186
4:51AM 4 SNOM 190 "Dial-Plan String" Settings
4:37AM 2 CID troubles...
4:06AM 5 FireFly w/ SIP
3:20AM 0 SIP <-> Asterisk <-> H323 Gateway
3:10AM 0 Path Replacement
3:08AM 0 Prepaid authentication and accounting using Asterisk
12:23AM 1 Newbie to Asterisk - VoIP end-to-end
12:17AM 0 Cisco firewalls and softphones
Thursday October 14 2004
6:41PM 2 Looking for recommendations for a low-cost FXO to IP gateway.
5:49PM 4 GPL Violations (Was: Advice on OS Choice)
5:00PM 0 notransfer=yes
4:35PM 4 Zap Channel wait for #
4:20PM 1 Bug in app_queue/AgentCallbackLogin
4:13PM 0 ODBCexec -Fixed-
4:10PM 0 asterisk seg faults
3:54PM 0 success with SIP on Swissvoice IP10S?
3:18PM 2 Limiting use of an account
3:12PM 4 how can I test canreinvite effectivness?
2:55PM 0 CNAM callerid from a T100p to sip cisco 7960 not working.
2:47PM 6 Dialogic D/300JCT-E1 support
2:36PM 2 Asterisk and Internet Phone/Line Jack
2:35PM 0 ODBCexec
2:14PM 3 MySQL CDR -- debugging
1:25PM 5 ast_data and dialplan in mysql
12:53PM 4 Intercept HOLD of Snom phones
12:44PM 1 IAX UDP packet dropped on incoming call
12:17PM 13 Running Asterisk on Linksys Router
12:09PM 3 FireFly SIP Registration Interval
11:39AM 0 Paging / 79xx cisco
10:46AM 1 Iaxy boot & provide...
10:36AM 8 SPA-2000's rebooting every hour or so...
10:24AM 1 Distinctive Ringing for SipToneII
10:11AM 5 Web stream from an extension?
10:03AM 1 searching for a nifty solution for different outgoing msn depending on the sip-user
9:07AM 3 About 3 Way Calling on GS BT100
8:19AM 1 Unable To retrieve DTMF tone from INFO message
7:51AM 0 Re: Asterisk-Users Digest, Vol 3, Issue 182
7:29AM 0 {SPAM?} Asterisk VIA SSH Tunnels
7:11AM 0 RE: Asterisk-Users Digest, Vol 3, Issue 185
7:10AM 9 (Another) Queue log analyser
7:06AM 0 authentification for H323 users
6:46AM 0 Xten eyeBeam Video
6:35AM 1 Hardware for 20 extensions (voip vs analog)?
6:10AM 100 Advice on OS Choice
5:41AM 0 no voice getting through
5:19AM 7 Configuring DIAX
3:40AM 1 cdr Logging - Postgresql
2:28AM 1 transfer call ?
2:22AM 1 Memory stuff
2:06AM 0 incoming ringsound
12:38AM 1 Call waiting trouble with 7912 cisco phones
12:25AM 0 [SIP] limiting the number of concurrent connections?
Wednesday October 13 2004
9:52PM 0 remote asterisk cannot register thru iax
8:10PM 1 3 way calling feature
8:05PM 0 New rh9/FC1 RPMS - v1.0.1 and CVS-10.11.04
7:46PM 5 Asterisk Post Paid Application
7:39PM 1 Embedded Asterisk System
6:27PM 2 No audio on incoming IAX calls
6:23PM 2 Cisco FXO
6:08PM 5 restricting access to outside calls
5:10PM 5 Uniden UIP200 Call Waiting Hold
3:16PM 4 SNOM 190: Good or crappy
3:02PM 3 Using Lucent/Ascend TNT as a PSTN Gateway?
2:13PM 1 spandsp-0.0.2 configure problem
1:45PM 1 Does Asterisk supports Sip Info Method?
1:17PM 1 TE405P and TE410P performance difference
1:06PM 0 TTS & Voice Rec (sphinx)
12:50PM 1 *8 on voicetronix OS12
12:47PM 5 Least Cost Routing
12:37PM 2 OpenSwitch12 install problems
11:53AM 11 DND on SIP
10:31AM 2 G.726/16kbps and Asterisk.!
10:26AM 3 quiet term
9:59AM 0 Asterisk answering calls - stupid newbie question
9:13AM 0 Pb H323 Connexion
8:58AM 0 Authenticate cmd with db
8:46AM 2 SayUnixTime(...,S)
8:26AM 0 Asterisk with wireless serial modems and multiple PC's
8:21AM 1 Asterisk (libpri?) and L1 Flags?
8:17AM 1 SpanDSP.0.0.2
7:54AM 6 Telco POTS -> FXO ?
7:47AM 1 Calling local extensions (also iax) directly from outside ?
7:33AM 10 Prerelease of DIAX 0.9.9a
6:46AM 4 ValetParking
6:42AM 0 CreateLogicalChannel Unknow Data Type
6:27AM 3 SIP 404 - circuit busy when dialing out
5:20AM 5 Backup POTS line
4:15AM 1 Dialing out with SIP phone problem
4:07AM 3 Not able to establish IAX call
2:29AM 0 IAX pretending to see unreachable hosts and other weird things
2:12AM 0 A question with voice Menu
2:11AM 2 quadBRI FAX problem
1:48AM 3 Where is the cheapest place to buy grandstream phones ?.
1:34AM 0 remote pickup
12:43AM 4 X100P sending out tone all the time?
12:00AM 6 RxFax multiple pages
Tuesday October 12 2004
11:11PM 5 Called name delivery
9:38PM 20 Passing CallerID to SIP phone from TDM400P
8:10PM 4 Bluetooth Bounty
7:42PM 35 Asterisk VIA SSH Tunnels
5:38PM 5 TDM01B Goes missing after reboot
5:25PM 18 mwi over serial port
4:19PM 3 G729 to G711 bridge
4:16PM 1 How many running instances (jobs) of asterisk
3:46PM 0 Canada Toll free
2:48PM 5 Channel Bank for T100P or E100P Digium Cards
2:26PM 11 Cisco 7960G "disk full error"
2:09PM 4 musiconhold will not start
1:14PM 2 SIP Connection to a Cisco AS5xxx gateway
12:53PM 13 Polycom Echo
12:44PM 0 In immediate need of Very powerful * for callcenter, ACD and outbound. Which consultant should I use?
12:41PM 0 Mesh Networking & SIP
12:10PM 1 Control Panel
11:28AM 0 In immediate need of Very powerful * for cal l center, ACD and outbound. Which consultant should I use?
11:21AM 4 In immediate need of Very powerful * for call center, ACD and outbound. Which consultant should I use?
11:16AM 5 Alternatives to the T100Ps?
11:11AM 0 Detect phone pickup, caller ID AGI
10:26AM 17 Chaining more than one zap echo canceller?
9:55AM 1 cdr make problem
9:26AM 4 Large Scale Asterisk Migration
9:02AM 0 Will an in-band 2100hz tone disable the zaptel (and/or other) Ast erisk echo cancellers?
8:57AM 0 ZyXEL P2602HW (WiFi + ATA Router)
8:19AM 2 calculating bandwidth on DSL?
8:01AM 4 Fast Busy
8:00AM 22 QoS Router/Software Suggestions
7:51AM 1 cvsup options file for v1-0
6:57AM 0 low bandwidth?
6:47AM 10 rfc3389 support in chan_sip?
5:52AM 0 4. Re: Quicknet Linejack Asterisk PBX (Lubomir Christov)
5:47AM 7 billing???
5:41AM 0 . Re: Quicknet Linejack Asterisk PBX (Jeremy McNamara)
5:28AM 0 RE: bt communicator`
5:19AM 0 Specifying different SIP packet destination from hostname in request line?
5:09AM 0 Echo Problem with IAX and Zaptel
5:04AM 6 How big .CONF files can be?
3:11AM 3 Zyxel P2000W web interface?
2:20AM 8 divert if not here
1:51AM 2 Redunance and failover
Monday October 11 2004
11:24PM 19 * box hangs after a couple of days...
9:26PM 1 CED tone and answering machine detection
9:08PM 0 Webmin modules for Asterisk
8:43PM 2 New Mailbox
8:33PM 4 Quicknet Linejack Asterisk PBX
8:30PM 3 cannot hear voice from phone
6:01PM 0 SNOM-105
5:51PM 6 SNOM 200 availability
5:27PM 0 OH323 and Mera Softswitch
2:25PM 0 Siemens Hicom / Digium TDM Card.
2:23PM 2 Echo problems polycom and x100p
1:42PM 2 linphone with *
1:41PM 30 Generic X100P's
12:42PM 0 Core files always appear in / ?
12:21PM 2 Zaptel with 2.6.9-rc4
12:05PM 1 CLI Destroy SIP channel?
11:21AM 2 G726 Codec Question
11:11AM 3 Disable flash hook hold?
10:57AM 3 Extensions.conf?
10:42AM 0 Database of world area codes
10:41AM 2 Sipura SPA-2000 / GSM or iLBC.
10:27AM 3 Unattended call transfer with IAX softphone or IAXy?
9:51AM 2 Re: Dial group continues to ring after answer -SNOM phones and solution
8:45AM 6 T100P to Verizon Smart Jack Question
8:38AM 0 FW: RTP timing issues
8:28AM 2 chan-sccp2
8:18AM 1 Re: Dial group continues to ring after answer -SNOM phones and solution
8:16AM 5 reading global vars from AGI
8:12AM 0 SOHO small or rack mount chassis and mobo for asterisk
7:48AM 2 System Hang Problem
7:38AM 0 7910 MWI
7:28AM 1 FWD incomming CALL won't authenticate in SIP
7:26AM 0 (no subject)
7:11AM 4 outgoing calls
6:51AM 4 Seeking a VoIP Solution for a big company
6:01AM 1 FYI - Zoom X5v built-in VoIP DSL router
5:55AM 1 Newbie OT Question - Hardware advise
5:22AM 0 re: ATA units: anyone have these working
4:45AM 3 Agent monitoring using fop
4:32AM 0 SetVar() with manager
4:05AM 3 Problems with voice menu
3:30AM 0 Re: Grandstream price in UK
2:32AM 1 SIP hangup issue
1:53AM 1 re: ATA units: anyone have these working with * or SER?
1:34AM 1 RE: bt communicator`
1:28AM 0 Request for IAX debug session transcript with IAXy
Sunday October 10 2004
11:39PM 6 Grandstream phone price
11:33PM 1 Where did USE_SIP_MYSQL_FRIENDS go?
11:19PM 0 Re: Asterisk-Users Digest, Vol 3, Issue 121
10:53PM 3 Error starting
10:31PM 0 C in Dial doesn't work (no cdr)
9:54PM 0 Conferencing -- app_meetme, app_meetme2, app_conference
7:05PM 2 newbie question - failed
7:00PM 1 How to Connect Fax to Dev PCi
6:51PM 0 Unable to locate sample sounds
5:45PM 0 microphone on localhost gateway
5:42PM 0 Flosys IT-550 GSM Gateway problems.
4:23PM 1 Broadvoice registration timeout
2:10PM 18 cisco ip 7905 legal ..
11:48AM 2 h.323 debian sarge problem - Could not open sound channel
10:18AM 2 Re: Asterisk-Users Digest, Vol 3, Issue 115
10:08AM 39 Intel Modem vs Digium Cards
9:56AM 4 TTS via text2wave
8:51AM 0 DIAX 0.9.8 and Windows XP SP2 problem
8:49AM 0 Problem with hearing 2nd call when call on hold hangs up
8:39AM 0 R2 update
8:11AM 1 ASTCC : rates based on incoming numbers
7:53AM 8 Zaptel 1.0.0. will not compile
6:24AM 7 DID trunk suggestions for Asterisk
6:19AM 65 SIP peers in MySQL Database
6:16AM 0 webmin module found on asterisk ftp site
3:41AM 3 Xorcom Rapid Asterisk distro beta 0.5.2
3:22AM 0 Why does incoming SIP call match a "peer" context in sip.conf?
3:16AM 3 Cost based Routing
3:08AM 3 SIP device not able to register but still able to make call
12:11AM 0 Channel Bank Suggestion. 100+ extensions system
Saturday October 9 2004
8:41PM 6 SPA-3k outbound calls...
6:54PM 2 Patch *
6:35PM 5 Access Bank II
6:30PM 4 Modprobe zaptel fails: Unknown symbol crc_ccitt_table
4:28PM 5 Asterisk Video Conference
3:26PM 0 Re: Vonage, PSTN, 911, and hardware question (Rajeef Sharma)
2:00PM 1 Asterisk - SIP -chan_sip.c:595 __sip_xmit: sip_xmit of 0x815008c (len 342,
12:56PM 1 iax2 w/ pa1688
11:51AM 1 ASTCC :: Strange Problem ast_openstream
10:22AM 2 nufone config
9:49AM 0 Unable to open master device - Fedora Core 2
8:45AM 0 web interface for meetme
8:44AM 0 How to check Asterisk status ?
8:43AM 1 Howtos on writting applications or modules ?
8:42AM 0 Re: if my Asterisk server is behind a FW ?.
8:16AM 14 Am I stupid or is my card DOA.?
7:23AM 0 Re: Sound Problem with * on VIA mini-itx M10K AC97' VT8235 (working)
7:20AM 26 Vonage, PSTN, 911, and hardware question
7:10AM 5 SIP SPA-3k & * Configuration
7:03AM 9 Slim Devices Sqeezebox Asterisk voicemail plugin.
1:37AM 2 Loopdrop
Friday October 8 2004
9:56PM 9 Can't compile chan_h323 in latest CVS...
8:51PM 1 Patch to queue.conf available for testing in Mantis
5:52PM 1 Cisco 7910 phones unlocking
5:24PM 0 VOIP provider in France?
3:07PM 4 * as sip proxy
3:03PM 0 TDM04B card connected to local switch, calling extension on local switch and playing message before answer
2:39PM 1 Application "Dial" option A
2:37PM 1 Need help configuring T1
2:34PM 0 Re: Asterisk Certification (was: Open-source VoIP 'will ne bigger than Linux')
2:34PM 3 SIP trunk: asterisk - callmanager
2:30PM 0 analog callerid private/out-of-area/...
2:23PM 4 NuFone & SetCIDNum not working since 10/5 - last Tuesday
2:11PM 0 Asterisk v1.0 CVS RPMS Available
2:03PM 1 MFC/R2 and Caller Id
1:30PM 0 Asterisk-OH323 (Couldn't transmit sound to and from ohphone)
12:59PM 0 Asterisk Certification (was: Open-source VoIP 'will be bigger than Linux')
12:57PM 0 After hangup phone rings
12:41PM 2 Disabling succeeding voicemail text
12:19PM 9 No sendmail on * server
11:45AM 3 user status in *
10:44AM 0 Asterisk ahead of old PBX problems (maybe PRI trouble)
10:34AM 0 Re: E1/R2 specs in Brazil
10:34AM 3 eyebeam and video
10:15AM 4 Answer()
10:04AM 0 IAX Connection problem with voicepulse
9:56AM 2 Flash
9:36AM 0 Hook flash hangup instead of hold?
9:35AM 0 Open-source VoIP 'will be bigger than Linux'
9:30AM 4 Registering to H323 Gatekeeper as client
9:14AM 3 BT ISDN30e and presentation numbers
8:24AM 0 oh323 channel : transport failure
8:06AM 1 connecting asterisk to existing pbx extension line
7:44AM 0 can I use these phones
7:34AM 2 MFC/R2 working with Ericsson MD-110 in Brazil
7:08AM 1 no ringing sound
7:06AM 0 COM-ON-AIR Dect Base as PCI/PC-Card
6:59AM 2 ${EXTEN} vs ${CALLERIDNUM} vs ??
6:45AM 11 Reload Asterisk from php or perl script
6:34AM 1 Sending CDR over permanent tcp port and how reliable is CDR information from Asterisk?
6:20AM 0 problems with asterisk-oh323-0.6.3b
6:16AM 2 Bypass VoiceMail Mailbox prompt
6:09AM 13 SPA3000 as a replacement for X100P
5:56AM 1 Zapateller Answering?
5:16AM 0 openphone congested link
5:14AM 0 BGT100 - Question...
3:10AM 3 open phone
2:43AM 1 grandstream bt-100 callerID not appear
1:11AM 0 Eicon DIVA Server 4BRI-8M
12:45AM 1 Incorrect ANI sent to PRI provider - CVS 9-29-04
12:25AM 0 re:uniqueid - how unique it is (Sathya Weerasooriya)
12:18AM 2 versions?
Thursday October 7 2004
11:39PM 1 dial out
9:37PM 0 chan_capi make issue
9:14PM 1 ATA & T.38 Fax
8:58PM 1 Adtran setup question
6:19PM 0 Cisco BTS 10200 G.729 problem
5:54PM 1 Call Parking with multiple contexts
4:55PM 0 uniqueid - how unique it is
4:36PM 1 IAX2 wait on channel
4:03PM 9 recent 's' and 'n' priorities and lables
3:46PM 1 T100P Pri Audio
2:58PM 1 PA168 ATCOM /Ezeephone Configuration
1:51PM 0 CallerID X100P
1:49PM 36 Nortel DMS250
12:42PM 0 Dialplan to Pick up calls that are ringingonother extensions?
12:10PM 7 Beginers Help - Hardware selection
12:09PM 0 Asterisk died on code 127
12:02PM 3 - Advice on NetFinity 5000 series
11:58AM 0 I need modify the time and cost the minute to second in application astcc ?
11:21AM 5 just getting started
11:16AM 14 Broadvoice problems
11:16AM 2 Dialplan to Pick up calls that are ringing onother extensions?
11:07AM 2 7912 Compatible SIP Images?
11:06AM 3 CVS branch v1-0 .vs v1-0-1
10:42AM 2 Dialplan to Pick up calls that are ringing on other extensions?
9:44AM 0 simple sip client
9:17AM 0 RE: Cisco and PRI IOS load
9:16AM 3 spandsp RxFAX problems.
8:57AM 6 TDM400P with FXO/FXS hangup problem
8:54AM 0 RE: Cisco and PRI
8:44AM 0 Remote Voice Mail
8:23AM 0 RE: Cisco and PRI
8:20AM 0 chan_h323 on latest CVS broken ?
7:59AM 0 Calling Card - GNUGK
7:36AM 1 'set debug' problems
7:34AM 0 Incomming calls on Eicon Diva 4BRI Card
7:34AM 0 Forcing a codec in chan_oh323
7:26AM 0 No incoming call on SIP Phone
6:59AM 0 Asterisk over NetScreen VPN/SIP protocol
6:59AM 3 SIP.CONF "Allow=All" do not work!
6:47AM 1 Confused about NAT and Authentication with FWD
6:37AM 2 openphone & Asterisk
6:30AM 1 bri-stuff for Asterisk v1.0
5:52AM 0 starting problem
5:51AM 6 RxFax - tiff problem
5:36AM 2 Asterisk ---- SER ----- GAteway and Reinvite
5:27AM 1 RTP timing issues
5:18AM 2 spa 3000 help
5:10AM 0 x-lite voicemail indicator
4:52AM 7 Display called Number or context on X-lite/X-Pro
4:23AM 0 Re: I'm thinking that FTP makes more sense for Volume One than CVS does
4:02AM 0 mod problem
3:54AM 1 Meetme conference across WAN
3:43AM 5 Vmail & Snom 190s
3:16AM 0 SIP header values in the dialplan
2:03AM 2 Higher level API on top of the Manager API?
1:43AM 0 ISDN4Linux early call progress tones & announcements from the PSTN
1:39AM 0 asterisk-oh323-0.5.10 problem
1:29AM 0 Missing Request URI in SIP message
1:10AM 0 phone autoregistration to *
12:34AM 0 Anyone using AddPac AP1200 VoIP Gateway?
Wednesday October 6 2004
11:56PM 3 AVM FritzCard
11:50PM 3 jabber clients
9:59PM 3 Transfer to Fax - 123 - Vonage
9:53PM 4 TDM400P stop responding
9:46PM 1 Asterisk Forums needs your input (
8:32PM 2 IAX2 Sporadic TX/RX retries
8:14PM 2 Cisco router for PRI termination?
8:05PM 0 Can asterisk unregister?
7:01PM 1 IAX2 to SIP
6:54PM 0 Zultys phones with encryption
6:48PM 2 REGISTER timeout problem with Broadvoice
6:46PM 2 No sound on 2 x-lite
6:36PM 3 Softphone
6:15PM 0 Grand Stream 486
5:04PM 21 Eezee phone?
3:40PM 3 Limit G729 concurrent calls
3:11PM 6 * to Cisco router with FXO's via SIP
3:10PM 4 T100p half-height PCI bracket
2:49PM 1 how does agent logoff if you supply extension?
2:19PM 4 Re 2 x100p cards H E L P (I have no hair left)
2:01PM 1 Asterisk to BabyTel VoIP SIP Provider
1:41PM 0 Remote mailbox in sip.conf
1:23PM 0 Rating IP phones from the users POV?
1:00PM 0 Asterisk and Festival, getting gethostbynamefailed error
12:59PM 3 OH323 compilation with updated Asterisk
12:48PM 0 Anyone using the Micronet FXS/FXO devices w/SIP
12:37PM 0 IAXy - Password protecting the IAXy device.
12:24PM 0 iax2, strange native bridge problem????
12:24PM 1 Asterisk and Festival, getting gethostbyname failed error
11:40AM 5 Astricon 2004 links collection
10:51AM 7 What would be needed ..
10:33AM 4 Tracking pressed keys
10:20AM 9 Comedian Mail User Guide
10:15AM 3 Setup problems
9:47AM 1 E100 Guide
8:59AM 3 Problem on getting wav file out of Festival-text2wave utility ?
8:21AM 0 mISDN channel
7:42AM 1 Cisco IOS SIP mime 1.0
7:15AM 0 Eicon ISDN to Voicemail audio dropouts
6:57AM 13 Asterisk and SIP phones
6:32AM 2 Issue with the channel drivers
5:56AM 7 Cisco Support for 7940, Is this Right?
5:53AM 2 Call Quality
5:40AM 0 Franklin Telecom Cards
5:29AM 0 GSM codec error in current CVS?
5:22AM 0 sip jitter
5:18AM 1 Cisco and ILBC
4:58AM 0 Pri E1 Timing - help ...
3:42AM 0 Echo / Adit 600
3:38AM 1 Queues/Agents
3:25AM 1 Hello - Simple SIP configuration
3:18AM 1 USE_MYSQL_FRIENDS=0, USE_SIP_MYSQL_FRIENDS=0 what is the difference
2:43AM 0 Can Asterisk provide Answer Supervision signalling to a channel b ank via T1?
2:29AM 4 Working Wellgate *SIP* 38xx/35xx hardware anyone?
2:28AM 3 no audio from asterisk
2:27AM 1 Anyone using VoiceMaster
2:16AM 0 Asterisk 1.0 -- Did the SIP dial syntax change?
2:06AM 0 asterisk generating alot of channels.. but for what ?
1:55AM 1 How much extensions will exhaust asterisk
1:40AM 0 start cracking
Tuesday October 5 2004
10:03PM 3 Queue() option not documented
9:53PM 5 Long pause between menus
9:41PM 0 loggedoff extension - why does * say "isonthephone"
9:12PM 1 Rate engine
8:41PM 0 loggedoff extension - why does * say "is onthephone"
7:15PM 3 odd configuration ... possible ?
7:06PM 1 Cannot compile Meetme2
6:34PM 1 Custom Monitoring Directories per queues
6:29PM 0 What kind of issues??????
6:23PM 1 loggedoff extension - why does * say "is on the phone"
5:25PM 0 real/wm/etc audio stream -> zap extension
4:57PM 0 How to force G.729 in H.323 calls
3:57PM 2 Dlink DVG-1120 Linksys PAP2 any Good?
3:52PM 0 MS netmeeting and *
3:38PM 2 Why I don't hear Call Progress
3:15PM 2 Howto change ACCOUNTCODE in extensions.conf
2:48PM 0 meetme caused 'RTP Read error: Bad file descriptor'
2:47PM 1 Dial group continues to ring after answer
2:45PM 0 looping back calls
2:39PM 4 broadvoice connection problem
2:33PM 1 asterisk compile with Fedora core 3 test 2 FC3T2
2:23PM 4 MeetMe MySQL Patch - Testers Needed
1:59PM 0 Using Macro's that cause loops, on purpose and using h, exten in default twice
1:57PM 0 SIP and symmetric NAT
1:38PM 7 its all 6's and 8's?
1:31PM 1 Newbie question ...
12:47PM 5 Problems installing app_valetparking
12:41PM 0 New Asterisk-CVS and Kernel/ALSA support RPMS Available NOW!
12:15PM 1 Pass a call to another switch
12:11PM 0 CMS
12:01PM 1 Popping and Clicking on Local WAN with X-Lite
11:52AM 1 Read error on
11:46AM 3 C flag in Dial command
11:31AM 11 Asterisk Perl AGI
11:28AM 1 Phantom calls on FXO
10:44AM 2 difference between dtmf digit 8 and 9
10:35AM 8 Long distance provider with access number and auth code
10:18AM 0 Asterisk and Alcatel 4200 -- comments, anyone?
10:15AM 2 SIPphone All-in-One: coments anyone?
10:01AM 1 Auto attendant dial an extenstion
9:57AM 0 Snom 220 Transfer Oddness
9:44AM 1 Low-Cost SIP Phones, ATA and Gateway excelent for Asterisk
9:28AM 1 For Sale Cisco IP Phones and ATA's
9:07AM 5 Re: RES: Working E1 MFC/R2 M?xico !!! (Steve Underwood)
8:50AM 0 Asterisk CLI Prompt : Small hack
7:26AM 2 hints lost after reload
6:57AM 1 Forcing a codec (take 2)
6:37AM 2 SIP multipart mime messages
6:12AM 1 Help 2 fx0 cards
6:09AM 4 books about ISDN/ss7?
5:41AM 4 Non-working module on TDM400P?
5:22AM 8 [OT] Has Sipura support been closed down?
5:21AM 1 Brazillian Caller ID: almost there...
5:13AM 1 OT: Can I use a SIPURA with Packet8?
5:07AM 0 Re: Firefly 1.9.5 released (gARetH baBB)
4:43AM 0 Polycom Echo using IAX2
4:27AM 1 asterisk with
4:25AM 0 Asterisk, Zaptel and Legacy Phones?
4:25AM 0 Paypal? Available in 44 of the world's approximately 190 countries
4:19AM 3 TDM20B and UK caller ID signalling
3:31AM 1 problems withX100P-Nochanneltyperegisteredfor'Zap'
2:49AM 1 problems with X100P -Nochanneltyperegisteredfor'Zap'
2:47AM 0 asterisk vic2-2bri NT/TE as gateway
2:19AM 0 Find a person
2:14AM 0 sipura 3000 , music on hold (playtones)
2:13AM 2 problems with X100P - Nochanneltyperegisteredfor 'Zap'
1:53AM 0 Just getting started with Asterisk
1:26AM 14 Special Meetme
1:15AM 0 Grandstream * Kingston Comms
1:04AM 2 Dialing a # in phone number?
12:48AM 1 problems with X100P - No channeltyperegisteredfor 'Zap'
12:26AM 10 Firefly 1.9.5 released
12:16AM 0 H.323: Inbound calls, incorrect remoteIpAddress
Monday October 4 2004
11:42PM 5 CallerID Question
11:37PM 0 Asterisk v1.0 sends incorrect invite to Sipura SPA-3000?
10:42PM 3 Cisco 7960G w/ SIP not working properly
10:33PM 38 IAXy - anyone using them yet?
10:33PM 7 problems with X100P - No channel type registered for 'Zap'
7:42PM 0 Avaya 4624 phones
7:30PM 1 some problems with OH323
7:24PM 0 How to check transcoding of audio codec
6:41PM 1 How to see CODEC which is in use?
6:38PM 0 Cisco ATA-188 w/502 Error on CallWaiting
5:19PM 3 3com NBX intergration
5:17PM 4 Limit extensions to single lines
5:12PM 2 IAX2 trunk mode not working
5:05PM 4 VoicePulse Connect Usage ??
4:41PM 6 Vonage just doesn't work?
2:17PM 9 motherboard for T100P
1:59PM 4 IAX/Grandstream.
12:11PM 4 Cisco XML 411 Interface
11:58AM 2 call/pickup groups
11:49AM 2 Will there be any support for iLBC in IAXClients soon?
11:45AM 0 OT: BudgetTone CallerID
10:48AM 5 Voice mail options/behaving change?
10:42AM 3 Queue/Agents problem with 1 agent
10:18AM 1 Parking calls
10:13AM 0 Ohio Linuxfest 2004 Presentation
9:34AM 8 Off Topic: Dead GS BudgeTone-100
9:25AM 0 echo cancellation: the never-ending quest fortruth
9:06AM 13 How to become an IP Service Provider?
9:01AM 0 Cisco 79XX Conference Call Issue
8:26AM 4 RES: Working E1 MFC/R2 M?xico !!!
8:25AM 0 Call waiting question for those who know the source
7:34AM 0 RES: Asterisk-Users Digest, Vol 3, Issue 25
7:20AM 1 What happened to my "Dial" command?
6:56AM 4 Re: Sound Problem with * on VIA mini-itx M10K AC97' VT8235
6:56AM 6 echo cancellation: the never-ending quest for truth
6:46AM 0 Lucent i2021 BRI Phone with Asterisk?
6:37AM 2 fxsmod cable length limit
6:28AM 2 Somebody using AS5350 CISCO?
6:27AM 2 exten patterns: how to match from XXX to ZZZ ?
6:16AM 8 budgetone-100 and handtone-286
6:08AM 1 Macro's and Var Scope's
5:53AM 13 Choosing a VoIP Phone
5:44AM 1 Asterisk CALLING CARD
5:25AM 0 using broadvoice and vonage hardware withAsterisk
5:24AM 1 Tones on a Cisco 7960 ?
5:21AM 2 FW: ASTCC: how to set quiet level
5:19AM 2 Re Problem with Asterisk 2 fx100 cards
2:29AM 2 SIP Proxy and use with Asterisk
2:14AM 0 Quassar 111 L
1:18AM 0 Appending a # to a dial-out number
12:57AM 3 300 extensions on Asterisk?
12:28AM 3 enhanced speed dial
Sunday October 3 2004
11:39PM 3 Conference by SIP phone
11:33PM 1 Help!!! Does Asterisk support call waiting in SIP phones
10:26PM 3 "#" sending
8:04PM 4 using broadvoice and vonage hardware with Asterisk
7:35PM 0 Sphinx 4
7:33PM 0 Issue with the oh323 channel driver compilation
7:24PM 29 ATA's
6:57PM 1 Sound Problem with * on VIA mini-itx M10K AC97' VT8235 chipset
6:04PM 7 Amazing, great protocol IAX
4:06PM 0 Tenor AS cancells calls through Asterisk
2:40PM 3 VoiceMail without password? How?
1:53PM 1 Asterisk + NCS patch
1:51PM 0 NCS and asterisk
1:22PM 6 asterix and phone system
1:01PM 5 Help with concept.
12:06PM 0 FW: Broadvoice
11:19AM 0 A problem with Asterisk-oh323
10:40AM 0 Looking to contract asterick setup
10:07AM 3 SIP-Provider who allows own Caller-ID
9:53AM 0 Call gets disconnected upon connect
9:21AM 0 Working E1 MFC/R2 México !!!
6:50AM 2 _asterisk-update
1:04AM 6 Hard phones that support ILBC
12:48AM 3 Where are the $500 24 port FXS gateways?
12:03AM 0 Presence Utility
Saturday October 2 2004
10:04PM 2 Second X100P card won't work
4:41PM 4 Getting Digium TDM card - what to watch for?
4:13PM 1 Latest ASTCC
3:56PM 2 Billing Applications - When does the bill start??
3:34PM 0 Asterisk- Cisco 7912G second call problem
3:25PM 0 ast_openstream: File your does not exist in any format
2:01PM 1 Fax passthrough
12:54PM 0 Packet cable NCS
12:38PM 1 H323 dial problem
11:46AM 1 RE: Random disconnects
11:20AM 0 IAX Ping for perl or python
9:28AM 0 Stability of the Asterisk platform
9:25AM 1 Compiling HDLC does not Produce hdlc0 for T100p
9:11AM 13 Callback
9:08AM 6 [OT] Sipura-3000 - Immediate hangup on inbound PSTN calls
6:25AM 4 voicemail attachment volume
3:04AM 0 S100U crashing server
2:35AM 4 Patch: Inbound-only busydetect
2:29AM 0 strange problem with a NT1 connected to * and an ISDN modem for data connection
Friday October 1 2004
10:31PM 8 Please, send me g723 & g729, pls
9:35PM 1 chan_sccp error
8:22PM 2 HT 486
7:47PM 29 Sipura 3000 FXO
3:14PM 6 Zaptel and ztdummy and timming question
2:53PM 3 Solution to my Grandstream lockups
2:31PM 0 Fw: OT: Opensource "Sipura Profile Compiler" for SPA2K, 3K
1:46PM 13 OT: Opensource "Sipura Profile Compiler" for SPA2K, 3K
1:46PM 1 upgrade goof up
1:23PM 6 CDR_Oracle anybody?
1:03PM 4 SMS in the U.S.
12:56PM 2 Forcing a codec
12:45PM 0 identify meetme participant by PIN
12:24PM 0 Random Call Disconnect
11:58AM 1 Unable to create Zap channels/IAX Warning
11:55AM 1 OT: Uniden UIP200 and NMAP
11:49AM 1 softphone over harphone
11:44AM 1 setting up more than one company on same * machine
11:42AM 2 BUG? no output from 'sip show users|inuse|active|subscriptions' when using MySQL auth
11:41AM 17 spandsp 0.0.2
11:27AM 4 IAX2 - Voice Pulse - slow choppy audio
11:12AM 6 Re: Problem with TDM400P
10:49AM 1 OT: Toll Free
10:07AM 5 MOH - 3 processes of mpg321 taking 20%CPU each -normal ?
9:56AM 3 Nuvox PRI - CCITT (ITU??) vs. ANSI
9:47AM 1 astcc question
9:46AM 1 How to contribute code?
9:14AM 2 H323 with 723.1
8:21AM 1 Agent Login Problems
7:58AM 0 More Reverb like Echo when calling for analog to ISDN - CAPI Fritz - what can I do ?
7:56AM 2 MOH - 3 processes of mpg321 taking 20%CPU each - normal ?
7:07AM 1 Help to connect to Mitel PBX via a T1 connection and a T100p
6:59AM 3 Cisco 7965 - New IP phone - Need Info
6:50AM 2 Maintenance Contract for a Cisco 7960 phone
6:49AM 2 DTMF relay
6:28AM 0 Re: [Asterisk-Dev] Use the Meetme application with another module thanUSB-UHCI
4:59AM 2 Hardware Compatibility Question
4:53AM 1 asterisk-addons on FreeBSD
4:31AM 0 E1 R2 MFC almost working on Mexico
3:43AM 0 S100U / wcusb Zaptel driver / Crash / Kernel problem maybe?
3:01AM 0 Cisco CM 3.3 and * via h.323
2:41AM 3 Intervivo sip.conf?
2:35AM 0 ASTCC with inbound
2:18AM 1 How to configure the voicemail message playback sequence
2:08AM 0 Is it possible to limit the number of voicemail per users?
12:50AM 0 Re: Asterisk-Users Digest, Vol 2, Issue 342
12:36AM 7 IAX busy signalling?
12:10AM 1 Configuring X Ten to make call using FX0