asterisk users - May 2003

Saturday May 31 2003
TimeRepliesSubject
10:25PM 8 Wildcard X100P question
6:26PM 0 SIP setup
4:24PM 0 adsi and voicemail application not working
3:34PM 7 Forcing intermachine codecs ?
2:00PM 0 Responder: Re: please help (reposted) - re. * connecting to a commercial call service
11:47AM 3 HELP ATA 186
10:00AM 0 Error Light
9:10AM 0 Von in London
8:36AM 0 register with outbound proxy from behind nat for freeworlddialup etc.
6:06AM 8 Passing audio stream through Asterisk or not?
3:59AM 1 oh323 problems
2:03AM 8 CAC ADIT600 / T400 config
 
Friday May 30 2003
TimeRepliesSubject
8:22PM 0 Another Problem!
4:57PM 2 receptionist application for asterisk?
3:44PM 0 call rejection with chan_capi
1:19PM 4 Whitenoise on TDM400p
11:10AM 2 SIP echo?
8:50AM 0 Disconnect Supervision/CPC/CPD from Sprint?
6:21AM 0 What I need to make this work?
6:17AM 6 nagios plugin to check asterisk
5:38AM 5 A Major Problem!
4:09AM 0 Question about Callerid
2:29AM 0 IAXTEL testing
2:26AM 3 External Directory Button and Dial tone on Cisco 7960 (SIP)
2:22AM 1 siemens optipoint 400 SIP
1:28AM 1 manager interface change request
 
Thursday May 29 2003
TimeRepliesSubject
5:58PM 3 aastra pt480 and adsi
3:58PM 3 Strange Issue with connected TA 750
1:50PM 0 Examples of using console as "normal channel?"
11:47AM 4 Asterisk IAX over VSAT satellite.
11:17AM 2 CalledID by channel difficulties
11:16AM 0 Outbound calls bridging
11:16AM 0 X100P call progress detection in UK
11:01AM 23 a beginner's SIP question ..
9:35AM 0 ACD
9:10AM 0 Messenger and DTMF
8:25AM 0 Fault tolerance?
8:04AM 14 T1-PRI deployment questions...
7:41AM 0 Transfer incomplete when MOH enabled
5:34AM 0 Large Asterisk installations
4:44AM 7 What is the going rate for the Snom 100 in the UK?
4:42AM 0 Would moving asterisk from behind NAT fix iconnecthere problems?
4:22AM 1 Setting up fax on *
1:39AM 8 External FXO device (USB or ethernet), supported by Asterisk?
 
Wednesday May 28 2003
TimeRepliesSubject
10:29PM 0 Lost in the Zhone
8:34PM 3 vmail.cgi contexts
5:29PM 1 SIP INVITE and ACK go to different ports
5:26PM 0 TDM40B Problems
4:26PM 7 ANI matching trouble
2:49PM 1 Cisco 827-4v SIP config
11:05AM 0 Question about Voicemail and Voicemail2
9:04AM 1 Voicetronix support
8:46AM 1 Newbie: Soundsdirectory
8:21AM 3 immediate on fxo
8:18AM 0 Testing tool for delay jitter and loss for constant bandwidth UDP streams
8:06AM 1 Disconnect options for X100P card
6:48AM 1 per virtual pbx VM storage
6:30AM 3 ILS - Internet Locator Service ?
5:35AM 0 calls between SIP and H.323 clients
5:30AM 5 dialogic DIALOG/4
5:27AM 2 Bridging two iconnect calls
5:09AM 2 chan_capi request
3:18AM 9 VOIP phone suppliers in the UK?
1:53AM 0 asterisk h323 backtrace
1:34AM 0 API asterisk
12:30AM 0 E100 card and channel bank
 
Tuesday May 27 2003
TimeRepliesSubject
11:50PM 0 DB limts ?
10:27PM 1 SIP Conferencing
10:25PM 1 Asterisk for call logging.......?
8:52PM 2 Caller-ID questions and suggestions
8:06PM 0 Bridging two outbound calls with iconnecthere
6:35PM 0 bindaddr and multiple ethernet connections
4:51PM 1 asterisk & PostgreSQL
4:39PM 0 VoIP Traversal of NAT and Firewall
3:53PM 0 Kernel Version for CAPI AVM Fritz PCI V2 /chan_capi /chan_alsa update to latest version
3:44PM 1 Kernel Version for CAPI AVM Fritz PCI V2 / chan_capi / chan_alsa update to latest version..
3:07PM 2 Call Detail Record Analysis Packages?
2:33PM 1 chan_h323 + Ericsson Webswitch 100
2:15PM 3 Problem w/ Zaptel HDLC mode cisco Data Stability
2:05PM 31 2 4-port T1 cards
10:03AM 5 The Phantom Call.. T1 card too
9:11AM 3 Duplicate numbers with outbounding calls
8:01AM 18 TDM400P > BT
7:32AM 12 [OF] Cable Pinouts
7:05AM 1 Incoming calls using iconnecthere
7:03AM 3 Gastman windows build?
6:04AM 0 Message Wait for normal PSTN phones ?
5:53AM 25 SayDigits
5:52AM 1 please help (reposted) - re. * connecting to a commercial call service
5:49AM 51 Echo cancellation
5:28AM 0 transfert call by command line
5:13AM 1 Re: Asterisk-Users digest, Vol 1 #520 - 9 msgs
5:10AM 3 FAX and Data support in asterisk......?
3:53AM 1 Re AVM 1A Passive Card (Again)
3:36AM 1 How many T400P/E400P in one machine are possible?
3:23AM 5 Monitor with Mp3 format
 
Monday May 26 2003
TimeRepliesSubject
11:24PM 0 fax over ip problem.
11:21PM 13 The Phantom Call..
11:12PM 1 Intel 536EP Full Duplex support
9:06PM 6 Cisco 7960 SIP speed dial
1:18PM 3 [new user] VPN or NAT? (and a FAQ)
11:34AM 3 Newbie Big question
9:59AM 0 Some boards won't boot with X100/101P
9:43AM 0 TDM400P FXO port polarity on RJ45
9:31AM 5 asterisk chan_h323 problem
8:57AM 12 chan_h323 and extensions.conf
7:24AM 1 Quetsion about DISA...
6:46AM 4 Bug in PGSQL
4:48AM 2 Cisco 7960 with Asterisk & H.323
1:32AM 0 SIP phone choice?
1:30AM 3 FW: NEWBIE: HOWTO make asterisk only accept calls when extension is picked up
 
Sunday May 25 2003
TimeRepliesSubject
10:09PM 11 SIP & VB6 ??
4:38PM 1 VoIP to ordinary phone and then back
4:17PM 9 Rhino Channel Bank
1:29PM 0 Asterisk codec issue with sip / iax.
8:56AM 1 iconnecthere problem 481 "Call Leg/Transaction Does Not Exist"
5:59AM 4 Message Waiting and VoiceMail 2
2:56AM 1 SMS Service over SIP/IAX/h323/MCGP
2:34AM 0 Registering a FWD account in asterisk
 
Saturday May 24 2003
TimeRepliesSubject
10:57PM 2 Limiting number of channels or calls
10:57PM 5 For the Australian Asterisk users
5:42PM 13 Free World Dialup behind NAT
5:25PM 3 iconnect and digest authentication.
5:17PM 1 DECT and Asterisk...
1:55PM 0 busydetect difficulties
11:59AM 0 IAX/IAX2 protocol descriptions ?
9:45AM 10 more about Netmeeting and G.723
9:38AM 0 G.723 codec for H.323 channel
8:54AM 10 TCP/UDP Ports used by Asterisk
7:15AM 9 advantages of a sip phone over Wildcard TDM400P solution
6:31AM 0 Asterisk and VOIP conferencing system
 
Friday May 23 2003
TimeRepliesSubject
6:13PM 49 Unable to create channel of type 'Zap'
3:32PM 4 How to define an extension for chan_h323
2:49PM 0 Port Forward & asterisk on private IP
2:09PM 0 First demo between IAX2 and chan_h323 works !
12:49PM 1 Holiday Scheduling
12:42PM 1 Cisco ATA186/Asterisk registration - weird behavior
12:23PM 1 Beginner: Dial and Playback
11:50AM 4 Channel Status in AGI
11:35AM 2 cannot find expat
11:12AM 7 SIP and DTMF
9:33AM 22 Who would use Asterisk SS7?
9:24AM 1 Call transfering external calls to external lines
8:55AM 1 Gnophone no sound
8:11AM 0 H.323 Cease Fire
7:57AM 8 Asterisk crashes with segmentation fault on using many OH323 calls
7:21AM 3 iConnectHere - calls dropping out?
7:15AM 3 Codec problems
6:35AM 2 Softphones
6:13AM 0 asterisk and pbx integration (newbie)
5:50AM 0 h323 transfers
5:35AM 2 Please remove H.323 from Asterisk (was H.323 support is distrubuted with Asterisk (was Re: chan_oh323.so: Segmentation Fault))
5:22AM 18 LineJacks & Asterisk
 
Thursday May 22 2003
TimeRepliesSubject
7:44PM 1 Asterisk stops working for no apparent reason :-(
7:21PM 1 astman
6:43PM 0 Cisco 1760 config examples - SIP
2:42PM 3 MeetMe conference
2:11PM 3 new DTMF tones
1:13PM 2 Musiconhold and Parking crash
11:23AM 5 nfas on T400P?
11:22AM 0 About Hardware requirements
11:04AM 4 libpri and zap lib
10:48AM 0 Custom Caller-id
10:32AM 0 Call Parking Difficulties
9:52AM 0 R2/DTMF with Asterisk/E400P..how?
9:46AM 0 New user - two part question
9:35AM 4 Symbol NetVision phone with chan_h323 - Complete Success!
7:43AM 7 OT: BRI ISDN question
7:15AM 0 MGCP NOTICE message and WARNING messsage
6:31AM 0 iax show registry
4:51AM 1 Playback on H323 crunched
3:06AM 67 fxo cards
2:47AM 1 isdn issue
2:13AM 11 authentication h323
1:10AM 9 SIP UA Fax device
12:56AM 0 USB Modems & Type Approvals for the tdm400 ??
12:47AM 0 More help on oh323.conf file
12:15AM 0 Some warnings when starting Asterisk
 
Wednesday May 21 2003
TimeRepliesSubject
11:57PM 0 to jerjer or not to, i.e. not the question was ( chan_oh323.so: Segmentation Fault)
8:50PM 0 Relative Newbie with a SIP/NAT issue
8:29PM 3 gnophone conf question
6:14PM 6 gnophone/IAX problem
3:14PM 0 How many X100P's in a system.
1:19PM 3 Answer not detected?
1:05PM 0 gastman segmentation fault when pressing 'en ter' in a command win dow
1:04PM 2 ISDN and E400P
1:04PM 5 Dialogic Support
12:32PM 5 2 part question
12:03PM 5 Need basic help getting started
11:34AM 5 ISDN FXS for home use
11:22AM 1 gastman segmentation fault when pressing 'enter' in a command win dow
11:07AM 3 asterisk segments with h323 support
8:04AM 1 Segmentation fault on using SIP -> H323
7:39AM 1 Call transfer disconnect with pre paid cards
6:37AM 10 PCI Master Abort ???
6:24AM 1 serios problem with zaptel or asterisk
6:19AM 0 Windows GSM player recommendations?
3:52AM 1 Cvs from 20030521/1235CET exits on Alsa failed assertion
3:32AM 2 asterisk and gnugk
3:27AM 3 Symbol NetVision - anyone?
1:22AM 1 Asterisk ACD
12:44AM 51 chan_oh323.so: Segmentation Fault
12:38AM 0 pbx_wilcalu.so undefined symbol
 
Tuesday May 20 2003
TimeRepliesSubject
11:03PM 0 Voice Mailbox problem
10:08PM 14 Using Arrays
3:39PM 18 IRC
3:15PM 0 ICH codec 19
2:07PM 5 Startup problem
11:28AM 0 Statup problem
11:09AM 0 Registering soft phones.
10:50AM 0 (no subject)
10:13AM 0 Fw: H323
9:51AM 2 (no subject)
9:21AM 0 WARNING[65545]: ... I don't know how to authenticate methods
8:55AM 0 busydetect=yes shows answered call on originating caller hangup
8:50AM 9 ATA186 through NAT, over Dialup, success story
8:36AM 15 IAX2
8:23AM 16 How many X100P's in a system..
7:24AM 0 Optimization
7:10AM 13 Wildcard X100P availability in EUrope
7:07AM 4 Need help with zapata.conf
3:39AM 5 chan_h323 core dumps
3:34AM 3 interesting questions ?
12:32AM 0 FW: SIP-firewall problem?
12:00AM 1 Error Connecting using libiax
 
Monday May 19 2003
TimeRepliesSubject
8:02PM 1 Wildcard E100P and E400P
7:34PM 3 Illegal instruction on a new asterisk build.
5:02PM 0 ATA type IAX unit ?
3:44PM 1 MGCP and Cisco ubr924
2:57PM 0 yet another snom issue
2:24PM 1 RE: AVM A1 (OLD) CARD
1:44PM 0 G729 snom cont.
1:43PM 8 G729 and snom
1:21PM 0 Call Recording static on in.wav
1:17PM 6 Questions about AGI and Wildcard boards
1:09PM 1 Asterisk Threads
12:47PM 1 Equipment Selection
11:37AM 1 Outgoing SIP Groups
11:24AM 0 CAC Adit 600 FXO-8 Cards ????
10:47AM 6 Call Group
8:57AM 1 gatekeeper problem
8:54AM 1 CAC Adit 600 FXO-8 Cards
7:20AM 3 CDR-Event on AstManager
6:54AM 0 Thanks!
6:44AM 1 app_capiCD.so
5:28AM 5 Number not allocated
4:27AM 1 A doc with all available...
4:07AM 2 Call between G.711 and GSM
3:42AM 0 netmeeting
2:43AM 8 transfer problems
2:41AM 2 G.729 warning
1:37AM 2 How to declare specific ip address for H323
 
Sunday May 18 2003
TimeRepliesSubject
4:28PM 0 Dial application ?
2:43PM 0 Problems with "r" modifier in Dial - does not work in SIP channels?
11:44AM 4 G.729: Typical usage scenarios
8:50AM 1 micky mouse sound with chan_capi solved?
8:16AM 1 Using Keypad for app_directory
6:49AM 3 Music conferences ?
6:03AM 9 CTRL+D exits Asterisk immediately
5:28AM 3 SNOM100 GSM again
4:03AM 9 DECT to Voip gateway
3:14AM 1 Inter-Asterisk Calls
 
Saturday May 17 2003
TimeRepliesSubject
11:56PM 2 Discriminating between two incoming lines?
11:54PM 7 little ADSI problem
10:18PM 5 E400P and 2 X100P working, but not together
3:51PM 3 Stupid question about recording prompts
1:59PM 4 FXO "Starting simple switch" after hanging up
1:02PM 0 Check for a phone off hook
11:14AM 1 XTEN Lite TROUBLE
5:48AM 0 error to load chan_iax.so
1:06AM 0 Debug for SIP and reINVITES (ATA-186)
 
Friday May 16 2003
TimeRepliesSubject
6:24PM 6 How to handle call waiting?
2:13PM 0 Info - Adit 600 console password reset
1:29PM 12 TDMoE
9:51AM 13 Extensions.conf sugestion?
8:32AM 6 SIP/H323 based channel bank?
7:59AM 11 Snom100 GSM
7:43AM 1 DynExtenDB DNID problem
6:51AM 0 asterisk, sip and SRV record
5:25AM 0 OpenH323 channel driver, Q931 Calling party number
3:07AM 3 kphone fails to register with asterisk (sip)
 
Thursday May 15 2003
TimeRepliesSubject
9:57PM 0 Call handoffs ?
9:28PM 14 Soft SIP phones (with RING !!)
8:57PM 0 Responder: Re: chan_capi version 0.2.1
1:31PM 1 4 line phone w cid recommendation?
1:18PM 18 SIP behind NAT (*sigh*)
12:51PM 0 "Nice" hang up
12:44PM 0 Best encoding values for moh?
11:42AM 5 Echo heard by person receiving call
10:59AM 8 Problems compiling zaptel
8:39AM 0 CallerID through iconnecthere not working
8:13AM 0 asterisk will not load zapata channel driver
7:59AM 1 problems with X101P and s400P cards
7:06AM 0 Current SIP channel features supported
6:55AM 3 Linux SIP/IX clients
5:33AM 31 Cisco 7960 SIP Firmware
2:46AM 0 OT: MGCP
1:29AM 4 mp3 playback (mpg123)
 
Wednesday May 14 2003
TimeRepliesSubject
11:16PM 9 Trouble with *80, *82
12:56PM 26 Channel banks
12:43PM 39 Call forwarding
8:02AM 5 AVM Fritz/Capi without suse ?
6:53AM 0 asterisk recarrying problem
5:57AM 0 (g)astman autodetecting extensions?
5:54AM 2 core dumped...
5:46AM 10 Asterisk - Connction to analog PBX?
4:34AM 0 (no subject)
4:00AM 0 IAX over 1394
2:45AM 5 G.729 Codec on Dialup
2:42AM 15 asterisk problem
1:34AM 1 CLI save dialplan problem
12:13AM 0 CallerID in h323
 
Tuesday May 13 2003
TimeRepliesSubject
10:49PM 5 app_transfer
10:19PM 5 is iaxtel down?
5:16PM 2 Voicemail2 and MWI
4:52PM 8 invalid argument 22 when modprobe wcfxs and wcfxo
2:01PM 11 Semi-ot: voip provider with 800-service?
1:19PM 23 Music on hold, Call Parking, etc
12:29PM 0 Music on hold sounds really bad
8:07AM 4 Cisco 12 SP+ IP phones
7:14AM 5 beginner's question!
2:31AM 3 set variables
12:00AM 3 Nortel i2004 IP Phones
 
Monday May 12 2003
TimeRepliesSubject
11:33PM 1 Newbie: Getting demo to work via ATA-186
2:05PM 1 Gastman compile errors
11:50AM 1 ISDN channels
11:13AM 1 chan_h323, asterisk, core dumped
10:06AM 7 AW: Asterisk-Redhat 9 install guide.
8:50AM 5 Sound Quality - Part 2 (mp3)
6:00AM 2 Processor
4:34AM 0 Using variables
3:48AM 0 call queues - one beep
3:44AM 6 (no subject)
 
Sunday May 11 2003
TimeRepliesSubject
7:05PM 0 Ha, just found a way around australia approv als......
3:55PM 0 error playing music on hold
8:04AM 3 RTP stream path : please help!
5:25AM 2 sip reply sent to wrong port
4:15AM 5 Sound Quality
 
Saturday May 10 2003
TimeRepliesSubject
4:53PM 1 quick questions for the gurus out there
4:42PM 6 Wireless IP phone for *PBX
10:28AM 1 vonage and asterisk
9:17AM 56 Voicemail2
8:41AM 1 Call forwarding questions
 
Friday May 9 2003
TimeRepliesSubject
10:22PM 7 test message
9:01PM 1 gastman....
4:51PM 0 how to config oh323.conf and extensions.conf
4:08PM 2 All station page and operator console....
11:45AM 0 Pingtel softphones, SIP proxies: experiences/summary
9:53AM 3 asterisk-oh323, new version 0.5.2
9:51AM 4 SIP Confusion
7:14AM 0 GSM warning
6:11AM 3 OH323 Channel Driver buffer sizes
1:46AM 0 relocation error
12:10AM 2 Configuration for ATA186 behind a NAT?
 
Thursday May 8 2003
TimeRepliesSubject
7:23PM 5 DBget and DBput in extensions.conf
7:15PM 11 Ha, just found a way around australia approvals......
4:05PM 0 Configuration/Management
4:02PM 0 Ring detection of X100P really bad
12:40PM 0 ATA 186 AudioMode setting for a call to PSTN with X100P
8:57AM 2 Asterik Hardware support
7:40AM 4 Send CallerID in netmeeting
6:35AM 7 Problems with H323
2:23AM 1 SIP client registration
 
Wednesday May 7 2003
TimeRepliesSubject
10:35PM 1 SIP-firewall problem?
8:03PM 10 vmail.cgi cannot read/delete messages
5:20PM 0 SIP phone audio quality in conference bridge
4:45PM 0 Re: rfc3389 and ATA config: Fixed
4:05PM 0 rfc3389 and ATA config
3:09PM 12 SIPPROXD for SIP thru NAT
1:42PM 0 forward a call
12:09PM 2 Does anyone know of an SIP java applet?
8:47AM 3 Question about STREAM FILE.
8:31AM 4 Mailing list delays
7:42AM 0 Error messages from asterisk
7:11AM 5 Music not on hold
6:48AM 1 Asterisk problem, - unable to load chan_oh323
4:54AM 2 MGCP broken
2:49AM 1 Are includes cumulative?
12:30AM 11 ISDN2e card
12:20AM 1 H323 ocnferences
 
Tuesday May 6 2003
TimeRepliesSubject
5:32PM 0 BEHIND A NAT inbound/outbound
1:14PM 14 SIP configuration question
12:56PM 1 SIP NOTIFY Message
12:38PM 3 Backup/Failover * Server
10:21AM 1 Using ICH for outbound when * is behind NAT
10:18AM 0 chan_capi version 0.2.0 released
9:45AM 0 T1 PRI with groups
8:07AM 0 openh323gk installation
7:35AM 3 oh323gk
5:54AM 4 AVM C4
5:25AM 3 capi + bri ?
 
Monday May 5 2003
TimeRepliesSubject
11:39PM 17 IAXTEL toll-free gateway
8:52PM 22 On-Hook ADSI
8:38PM 3 bandwith issues, ISP hosting services, etc
8:10PM 0 affordable softphone with g729 support
3:19PM 3 G723 - Has anyone gotten SIP_CODEC= to work?
12:50PM 0 ADIT 600 FXO Ports Remote DisconnectrSupervision
12:28PM 0 TDMoE implementation suggestions
10:26AM 1 FW: HDLC & T100P
10:04AM 0 Host requirement?
9:26AM 3 AVM ISDN B1 V4 card
8:27AM 7 oh323 problem
8:02AM 0 HDLC & T100P
7:05AM 0 hardware for 2 external and 12 internal lines (all isdn)
4:47AM 2 (no subject)
4:25AM 10 Exclude prefixes
4:10AM 0 Asterisk Hardware question
3:43AM 0 G.729 between Asterisk and SNOM??
 
Sunday May 4 2003
TimeRepliesSubject
8:59PM 1 safe_asterisk
11:56AM 1 Comments? Supermicro P4SGE P4 FCPGA MB and Digium cards
 
Saturday May 3 2003
TimeRepliesSubject
10:21PM 0 * as a SoftSwitch/Router solution
8:31PM 1 V 1.3 Guidlines for Custom Rules
8:16PM 8 Monitor Asterisk
6:06PM 7 Failed calls from SNOM100 to *
2:43PM 2 default voicemail password
9:44AM 3 Error working with X101P and S400P cards (fwd)
9:02AM 3 Execute command after hangup / MWI
8:01AM 0 unsubscribe practice@webjogger.net
3:15AM 1 SIP & Caller ID & outgoing line
2:08AM 1 Cannot find Context internal
 
Friday May 2 2003
TimeRepliesSubject
11:17PM 3 IAX tollfree extension conf
11:03PM 1 WARNING (Sipsock_read) Recv error: Resource temporaily unavailable
9:22PM 0 delta three account to Transfer to outside p hone number.
9:04PM 4 Wildcard X100P Choppy Sound and Chipmonk Recordings
4:19PM 0 Digium Announcements - Asterisk G.729 Availability & thevoice.digium.com
11:24AM 1 X100P and BT Caller ID
9:41AM 11 SIP Peers unreachable
6:07AM 1 Alchemy Cybergear
5:49AM 1 Cisco ATA186 CLIP/CallerID
12:14AM 9 Termcap support not found
 
Thursday May 1 2003
TimeRepliesSubject
11:22PM 3 Asterisk and unknown codecs and GSM
11:10PM 8 Echo Cancelaltion in Zaptel Changes
5:35PM 1 TDM cards and Asterisk
4:55PM 0 Picking up parked calls
3:27PM 3 Command line completion causing crash?
2:46PM 2 Routing calls by DID
12:58PM 7 Reduce echo?
12:49PM 0 Strange SIP registry behavior
12:45PM 5 Max number of connection in IAX ?
10:52AM 0 MSN registered, but make phone call command is grayed,fixed
9:34AM 0 AVM Fritz-PCI v2.0 and Asterisk
7:54AM 7 No Dialtone
7:11AM 0 Re: [Asterisk] MSN Shenanigans
6:49AM 2 Youch! Painfully loud beep...
6:19AM 0 DTMF issue with SIP