asterisk users - Feb 2006

Tuesday February 28 2006
8:35PM 7 incoming calls dropout on PRI over TE110p
8:04PM 0 Compiling Intel G729 error
7:57PM 4 Problem with two cards Digium
6:55PM 0 unicall channel on asteriskathome 2.5
6:15PM 8 How hard to create Asterisk for Compact Flash?
5:58PM 8 Asterisk with T1 card on laptop
5:36PM 0 Re: Polycom Default Ring Volume [OT]
5:10PM 4 How to determine the power draw on TDM2400P?
4:44PM 3 Capturing DIALSTATUS on a PARTICULAR channel if multiple-dialling?
3:29PM 20 A room full of Cisco 7960s behind NAT
3:16PM 0 SER ,Asterisk and MWI
2:59PM 1 Auto login via Remote User
12:42PM 3 Re: Polycom Default Ring Volume [OT]
12:21PM 1 How to check if transcoding is setup to work properly
12:17PM 11 Sipura SPA-3000 and PSTN dtmf
11:34AM 1 H.323 ( HW PBX to *)
11:19AM 6 Comfort noise support incomplete in Asterisk (RFC 3389)
11:00AM 0 Replicating functionality from our prior PBX
10:45AM 1 Sound quality issue in one direction and wctdm problem with APIC enabled kernel
9:51AM 0 callthru and CDR
9:45AM 2 GSM phone reception range extendor
9:37AM 5 Re: Echo and other reasons to migrate to BRI
9:26AM 1 why incoming DATA CALLS are answered as VOICE by asterisk IVR?
8:53AM 0 How to determine duration call when is used Attended Transfer
8:52AM 0 Re: Echo and other reasons to migrate to BRI from POTS? Was (Echo on PRI/BRI?)
8:05AM 4 Cannot boot machine up after working on zapt el....
8:00AM 0 changing source email address of pager notifications
7:53AM 2 Conference bridge dimensioning
7:44AM 2 Cannot boot machine up after working on zaptel....
7:40AM 3 Austria isdn p2p empty DID
7:29AM 0 Asterisk hangs up - h323
6:58AM 0 playing hold time announcement without queue position announcement
6:54AM 1 Set CallerIDNum on a PRI
6:50AM 0 variables internas
6:46AM 1 Problem with incoming call, Please help
6:45AM 0 FW: 7960-tones.xml (Schochet, Wes)
6:30AM 0 newbie debugger needs a little guidance
5:55AM 1 Re: Polycom Default Ring Volume [OT]
5:47AM 3 monitor outgoing calls in queue / campaings
4:53AM 3 FW: Re: Delay on Phone ringing
4:11AM 0 VAD, CNG, for Zap
3:43AM 1 Q: Status of feature Call Deflection / Partial Rerouing in chan-capi and zaphfc
2:46AM 1 Problem calling out
2:46AM 0 My or provider error?
2:36AM 3 run with incorrect E1/T1 jumper settings
2:22AM 0 R: Re: courtesy message calling mobile phones
12:28AM 2 chan_capi and Eicon Diva
Monday February 27 2006
10:38PM 0 HST Saphir III ML PCI and Linux/Asterisk
8:03PM 1 asterisk -rx 'commands'
6:04PM 0 7960-tones.xml
3:01PM 0 voipstunt can't get call in asterisk
2:43PM 0 Cisco upgrade to SIP was: Covad anyone ...
2:17PM 0 RE: Cisco 7960 upgrade to SIP
2:03PM 12 AGI Scripts Terminate too Soon
2:01PM 2 Echo on PRI/BRI?
1:14PM 3 Covad anyone ...
1:03PM 0 Polycom 501 issues
12:54PM 3 Asterisk with HT 488 FXO
12:01PM 3 Matching '*'
12:00PM 0 RES: RTP and Signalling
11:41AM 12 TDM400P digium card
10:41AM 1 billing - different tarif per phone
9:02AM 0 TE411P problem-- probably stupid.
8:16AM 0 chan iax2 auto congest
7:54AM 4 courtesy message calling mobile phones
7:31AM 1 Asttapi - what's wrong?
7:09AM 3 jitterbuffer and DTMF conflict?
6:59AM 4 Problem with chan-capi: outgoing calls on two lines
6:32AM 0 Newbie h323 question
6:27AM 0 how to configure my asterisk@home 1.0.9 to do call forwarding ?
6:16AM 0 Polycom bootrom and SIP software
4:44AM 1 Asterisk and Hipath interconnections
4:17AM 6 res_features pickupexten
4:17AM 1 Problems dialing to another Asterisk server
3:55AM 3 IAX provider recommendation wanted
3:20AM 0 Zap tuning for echo/gain
Sunday February 26 2006
9:43PM 0 FW: BLF not working after reload
9:31PM 0 advanced options access problem
9:03PM 2 Music on hold and conferencing on OS X
7:22PM 0 another nat question
7:04PM 23 Asterisk question
6:53PM 3 nat=yes and qualify=yes viable NAT solutions?
6:48PM 0 HT-1000 chipset experience
4:17PM 0 Anyone using LG LIP-100 ip phone
3:03PM 5 Prepaid / postpaid solution
1:42PM 1 Limiting Sip Calls ?
12:38PM 8 BLF not working after reload
10:35AM 6 Voice Over WiFi
10:26AM 6 Skype vs. an Xlite registered to Asterisk
8:01AM 8 Internal Server Error
6:37AM 2 authenticate problem
1:57AM 7 Asterisk Web-Based Voicemail?
12:50AM 5 Newbie config help? Wellgate 3701a
Saturday February 25 2006
10:03PM 0 Choosing a GSM gateway for personal use.
6:50PM 0 Problems with certain Global Variables not passing correctly.
3:56PM 2 question
11:42AM 1 OT- Rwanda DSL growth
10:56AM 5 Anyone using the GSM gateway from CyberTelecom ?
9:46AM 1 Asterisk as a dedicated Analog PSTN gateway
5:25AM 2 Asterisk, SIP phone , NAT
5:19AM 2 always on agents and queues
4:56AM 4 Polycom Default Ring Volume
4:49AM 2 metermaid patch
12:49AM 2 Unknown RTP codec 100 received
12:20AM 0 Problem calling a ZAP channel with svn 10842
Friday February 24 2006
8:10PM 0 Queus seem to work different between IAX and ZAP channels
3:06PM 0 Reading sound in eagi script and recognizing DTMF sounds at thesame time ?
3:00PM 0 no sound in NAT configuration scenrios
2:56PM 0 7940 and Subscriptions
2:35PM 5 ParkAndAnnounce2 Feature Request
2:26PM 3 Asterisk Topology
2:22PM 0 Generating Polarity Reversal on FXS line
1:21PM 2 Is Asterisk a PBX?
1:10PM 3 chanspy instability
12:13PM 0 RE: [Asterisk-Users ] RE: Monitor a call in progress. (Steve Totaro)
12:11PM 1 ImportVar Syntax
11:41AM 0 disallow, allow codes
11:38AM 2 Missing 31 DTMF tones over ZAP
11:35AM 0 problems with dialing
10:34AM 0 incoming peer register problem
10:19AM 6 Call quality problems
9:42AM 0 Trouble Chan Spy
7:34AM 7 Possible Bug in SIP Stack.
7:12AM 3 S100U and TigerJet
6:39AM 6 Problem with T1 installation
6:31AM 0 lspci don't have Tiger Jet Network Inc
5:37AM 0 Asterisk configuration for h323 calls
4:35AM 0 What's with Indications/SetLanguage/Zaptel/RingBack ?
4:27AM 5 How can I debug spandsp?
3:11AM 4 a2billing without IVR
2:45AM 1 Polycom IP 601 Buddy Watch doesn't work after Asterisk reload
2:02AM 0 can't dial some particular numbers (providers ?) with asterisk sip / zap channels
Thursday February 23 2006
11:30PM 1 Which Quad Port FXO is Best?
11:28PM 2 spandsp debug or logging
11:10PM 52 fax receive using TDM400P
8:26PM 1 anyway to a2billing without IVR
7:36PM 7 GPS-enabled cell phone/PDA
7:06PM 2 digium TE405P and intel motherboard
6:44PM 1 mysql problems
6:40PM 0 maxmessages and maxgreet per mailbox
4:16PM 6 Analyzer for Milliwatt
3:53PM 2 Incoming/Outgoing call question
3:42PM 0 How to reset Digium card while asterisk is running?
3:33PM 0 Choice One PRI?
3:33PM 4 How to query a table from the keypad?
2:55PM 0 How to install Zaptel?
2:35PM 1 How can I force Asterisk t not override my codec order?
1:43PM 1 Explain Yellow Alarm in a Legacy Integration
1:08PM 30 Linksys WIP300 WiFi Phone
12:57PM 0 isdn problem
12:55PM 0 Okay can somebody explain this...
12:04PM 5 Is setting the variable _TRANSFER_CONTEXT required in features.conf?
11:48AM 1 not consistent log from asterisk
11:17AM 0 Zaptel CiscoHDLC / Fedore FC4
11:13AM 7 Cisco 79xx and SIP 7.5 Problems
11:13AM 5 sip registration fails with 404
11:09AM 6 Voicemail problems
11:03AM 1 Streaming Music On Hold - Reality Check
10:30AM 12 OT: VoIP over bonded link
10:28AM 0 Detect answer and hangup
10:18AM 4 UK X100P installation help
10:12AM 4 Polycom IP601 Question
9:17AM 5 IAXModem/Hylafax problem
9:08AM 1 Pickup call on Hold
9:07AM 4 Monitor a call in progress.
8:07AM 9 Keep getting message in logs that pbx.c cannot find extension context 'default'
8:02AM 0 Features set in the features.conf stopped working after upgrade.
8:01AM 4 Codec order sent wrong from Asterisk
7:33AM 2 Calls not going through
7:14AM 0 problems while dailing outside
6:55AM 4 Configure DID
6:27AM 0 Is anyone using hinting?
6:23AM 5 chan_capi-cm 0.6.4 random outgoing MSN problem
6:15AM 21 mpg123 alternative?
6:09AM 14 auto provision of IP501 polycom
5:53AM 3 SV: Polycom 501 ACDlogin
5:22AM 1 What SW/HW phones support sendtext feature (trying to send speech recognition results back to user)?
5:07AM 0 broken CDR (Master.csv) reports with HFC cards in Asterixk 1.2.x?
5:04AM 8 sipura 841 mass provisioning
4:11AM 3 Polycom 501 ACDlogin
2:53AM 3 register => 2345:password@sip_proxy doesn't care about port
1:46AM 6 username as extension
12:57AM 21 chan_capi-cm-0.6.4
Wednesday February 22 2006
9:22PM 1 TDM 400P in Malaysia
9:22PM 2 mysql phone number pattern match query
8:27PM 0 problem playing back voicemail
8:16PM 0 Clipcomm product feedback required
8:10PM 6 context being ignored by inbound sip call
8:06PM 3 IAX2 through Shorewall rpoblem
5:18PM 0 Queues and On Hold
3:02PM 2 Important: Application DIALPLAN STANDARD/GUIDELINES needs to be established.
2:44PM 0 Problem transferring call to a meetme conference
2:00PM 2 voicemail files in Asterisk have rights 600 , I need 644
1:57PM 0 Some Hardware & Asterisk Applications Questions
1:55PM 2 snom 360 problem - only one call works after reboot
1:44PM 0 DATA calls answered by IVR, but I don't want that
1:41PM 0 What are these error messages in my logs?
12:56PM 0 Outbound problem sip chanel
12:31PM 2 Problema calling from elesign h.323 to iax device
12:26PM 0 ISDN interface cards with pass-through
11:34AM 0 Is SIP "canreinvite" working ok?
11:21AM 0 Cisco 7960 dialing trouble
10:56AM 4 Hints between servers?
10:33AM 1 FC4 and yum install; how to configure questions
10:03AM 1 Fromuser required but overrides SetCallerID
9:42AM 0 R: queue behaviour
9:28AM 6 Streaming Music On Hold
8:48AM 0 debugging asterisk configuration
8:28AM 0 problem with SU100
7:55AM 27 Best ATA for general residential deployment??
7:13AM 1 "Proxy Authentication Required" issue
6:25AM 3 Voice conferencing server capacity
6:17AM 1 SV: Re: SV: Re: SV: Re: Fromstring when sending e-mailonrecievedvoicemail
5:38AM 1 SV: Re: SV: Re: Fromstring when sending e-mail onrecievedvoicemail
5:20AM 0 Problem with receiving faxes with spandsp - full log included (long)
5:14AM 0 Realtime queues with Firebird SQL through unixodbc
4:56AM 0 Cisco 79xx <=> Asterisk - SIP or SCCP?
4:10AM 3 DTMF Mode supported by VoiceMail Application
3:37AM 3 TFTP server for GrandStream BT phones / need testing
3:18AM 6 Polycom IP 601 Buddy Watch problems
3:16AM 1 Detecting disconnect on TDM400P with 3 FXO ports and 1 FXS port
2:34AM 3 Cisco 79xx firmware
2:31AM 3 did from sip trunk
1:46AM 2 Cannot see the caller id , When calls made from one server to another
1:29AM 7 Asterisk hints
1:19AM 1 SV: Re: Fromstring when sending e-mail on recievedvoicemail
Tuesday February 21 2006
11:07PM 0 Call AGI when agent answers call in queue... ?
9:05PM 0 meetme feature request (or maybe its there already?)
6:13PM 3 Asterisk and T38 Fax
5:03PM 6 TDMoIP and Asterisk
4:16PM 0 chan_bluetooth jabra 200 / 250
4:00PM 1 Matching variables in extensions.conf
2:11PM 0 Catching _ALL_ characters
1:56PM 1 Test my test-branch!
11:47AM 0 commercial package for vertical services
11:35AM 0 how to tape letters in xlite
11:24AM 3 Call queue design issues and suggestions
11:23AM 0 Looking for programer...
11:16AM 2 Outbound Routing does not use Multiple Trunks
10:57AM 73 What business IP phone to use
10:46AM 1 Sangoma A200D analog card with fxo's
9:47AM 0 realtime sip_buddies does not store ip address
9:19AM 5 Voicemail 0 for operator call routing
8:24AM 3 Send flash through zap channel
7:48AM 0 Application pppd
7:44AM 2 SV: Re: Fromstring when sending e-mail on recievedvoicemail
7:33AM 1 SV: Re: Fromstring when sending e-mail on recievedvoicemail
7:03AM 0 asterisk 1.2.4 doesn't detect the PSTN hang up
6:57AM 0 asterisk related job offer in Florida
6:36AM 0 Set CallerIDNum for outgoing calls on a PRI+DDI line
6:34AM 7 pickup problem on Asterisk 1.2.4
6:26AM 2 Fromstring when sending e-mail on recieved voicemail
5:41AM 8 Recommended rack-mountable server anyone?
5:28AM 0 polycom and its minibrowser
5:22AM 0 API or Call command
4:51AM 1 Sirrix BRI errors
2:16AM 2 immediate pick up in "s"
1:53AM 5 sniffing sip password/uri/host info
1:42AM 2 Setting up an EICON CARD with CAPI
1:17AM 1 DTMF Tones in RTP Payload as Well as in Events = Duplicate Tones
12:14AM 0 Session Media 183 and Ringing Tone 180 Passing To SIP At the Same Time
12:09AM 2 PSTN connection via IP/ethernet
Monday February 20 2006
11:58PM 3 Asterisk behind Centrex
10:19PM 0 Need to Hire PHP Programmer(s)
10:03PM 1 Dial timeouts and SIP 302 redirects
9:38PM 1 realtime, iax, trunk
9:30PM 1 SIP registration on Sipura 841
8:00PM 0 White Noise on TDM22B FXO Channels (newbie)
7:46PM 0 Issue w/ Polycom 501 phones in a queue...
6:10PM 1 Grandstream BT-101 POS Error
5:49PM 1 Incoming ISDN DATA calls answered by asterisk IVR! - How to stop that?
5:22PM 4 Download "Asterisk: The Future Of Telephony" [More Info]
4:55PM 3 Download "Asterisk: The Future Of Telephony"
4:45PM 0 chan_sccp .conf changes
4:24PM 6 Fwd: Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)
4:24PM 14 Multiple TDM400P's in a single machine
3:46PM 6 Incoming Calls Getting Crossed - Weird
2:35PM 0 Trunk calls ring internal analog phone
2:23PM 0 Asterisk & Broadvoice Incoming Calls Problems
2:11PM 6 Dell PowerEdge 2850
2:08PM 2 Dial from AGI = no ring back ??
1:37PM 14 good voip
1:03PM 2 Linear Queues Strategies for 3rd Party Application
12:42PM 2 g729 quality at GSM bitrates
11:40AM 18 Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
11:31AM 0 Zap channels Deactivated with Bristuff-0.3.x after upgrade from 0.2.0
11:31AM 1 problem with outgoingcallsUnabletocreatechannelof type 'ZAP' (cause 34 -Circuit/channelcongestion)
9:56AM 3 asterisk error
8:39AM 0 Landmark digital key systems and Asterisk
8:35AM 6 calling from SIP to a h.323 device with oh323
8:16AM 0 A good SIP VB6.0 component to use?
8:15AM 1 q931 85
7:37AM 0 Strange SIP registration situation
7:03AM 0 SIP ATA gives no ring tone on IAX2 route
7:02AM 1 queue behaviour
3:54AM 2 Where can I get the tar.gz sources of libnewt?
3:54AM 1 call parking "hint"
2:01AM 0 automatically start application from thecommandprompt
1:44AM 7 spa3000
1:27AM 1 ~1 sec delay from callee answering to call established on dialout
Sunday February 19 2006
11:17PM 0 Live Communication Server and Asterisk
9:45PM 5 Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)
9:34PM 4 Queue Messages now playing when caller is inside queue
8:53PM 0 Call forward on unavailable timer issues
8:33PM 3 Asterisk start errors with TDM2413E
8:32PM 3 chan_capi setting ${DNIS}
6:30PM 12 Loops and Variables
5:47PM 0 Viking CPC-Disconnect
3:45PM 2 Line Dropouts on E405P
12:17PM 0 salesforce
10:00AM 4 spandsp 0.0.2pre25
9:46AM 4 [slackware 10.2 and TE205P] Unknown signalling method 'pri_cpe'
8:04AM 1 any doc/example for ?
7:24AM 5 Cisco 7905 can't register
7:17AM 4 Wildfire messsaging server
4:59AM 1 Cisco 7960 Register Problem
Saturday February 18 2006
10:35PM 1 snom 360 incorrect US indications
10:03PM 4 co-location providers in Ottawa, Canada
6:24PM 0 COMMPARTNERS Resellers
3:47PM 4 Asterisk as MGCP User Agent
2:51PM 1 An array of extensions in my lab
12:35PM 18 Application Faxing using SIP
12:29AM 0 new jitter implementation for sip
12:08AM 0 Asterisk-Netsec Ranch Networks
Friday February 17 2006
11:05PM 13 Bridged line appearance
9:07PM 1 Intro and first questions
8:53PM 0 FaxToEmail for diferent Channels and different Mail accounts?
8:09PM 1 Outbound ZAP Dialing
5:43PM 0 Supported protocols in pri
5:34PM 0 Festival and Asterisk - different voices? => SOLVED!
4:29PM 0 Softphones and other VOIP PBX's
3:59PM 1 Hold and Call Waiting - Budgetone 100
2:46PM 0 I must be missing something zimple...
2:34PM 3 MixMonitor and command
2:20PM 0 uniden 1 touch dial
1:13PM 0 Polycom 301 line key display
12:14PM 1 A unique 'click to call' project - Could use some advice <--one thing I forgot
10:48AM 2 indications issues in Singapore?
10:45AM 8 g.729 woes
10:05AM 0 Quintum Tenor AX 24 Port SIP FXS "UnsupportedMedia Type"
9:55AM 1 A unique 'click to call' project - Could usesome advice
9:46AM 1 simple iaxmoden configuration
9:33AM 3 problem with outgoing callsUnabletocreatechannelof type 'ZAP' (cause 34 - Circuit/channelcongestion)
9:30AM 0 [Fwd: using AMP custom extensions]
9:27AM 1 A unique 'click to call' project - Could usesomeadvice
9:25AM 0 using AMP custom extensions
8:47AM 0 A unique 'click to call' project - Could use someadvice
8:39AM 0 Intrado / VoIP E911
8:26AM 0 vISDN with Asterisk and HFC passive cards.
8:14AM 2 Cheap BRI card
8:06AM 6 A unique 'click to call' project - Could use some advice
8:05AM 5 SPA-941 & hint
7:36AM 2 [OT] List messages and end user outages
7:26AM 1 Quintum Tenor AX 24 Port SIP FXS "Unsupported Media Type"
6:42AM 15 MOH from RCA jack?
4:59AM 0 codec negotiation with SPA-3K
4:51AM 3 free tollfree termination
3:17AM 1 FW: AGI onAnswer function: does it exist?
2:41AM 2 aastra v1.3.1 firmware
2:02AM 4 one way / irratic voice over iax and g729
1:14AM 12 how to add stun functionality in asterisk
12:18AM 1 SIP Problem Fedora Core 4 and Asterisk 1.2.4
Thursday February 16 2006
11:22PM 0 ztdummy configuration issues
10:33PM 1 Playing sound File using GotoifTime function
7:26PM 0 Ottawa Asterisk Users Group
7:13PM 1 SOLVED - Channel bank woes - no outbound calls
6:45PM 0 Big problems with Voicemails ODBC Storage
5:56PM 1 Festival and Asterisk - different voices?
5:38PM 0 No Ringing Sound & No periodic-announce
5:31PM 2 Cisco 7960 won't register
4:40PM 2 zoom FXS/FXO gateways
4:35PM 3 ARI 0.06
4:22PM 2 79xx's and call queues
4:20PM 1 Update to the latest zaptel driver - Congestion gone, but scary write errors replaced it
4:09PM 0 Sorry for the multiple-posts... I had a mailserver-hickup
4:02PM 2 Install instructions for FOP Flash Operator Panel do not make sense...
3:44PM 3 Safely editing voicemail.conf
3:39PM 22 Anyone using the GSMgateway from CyberTelecom ?
3:35PM 3 Sangoma analog cards?
3:20PM 2 CISCO 1760 with 1 BRI
2:50PM 3 "No D-channels available!"
1:59PM 2 Random Hangups/Disconnects
1:55PM 0 automatically detecting failed registration
11:58AM 11 How do I install speex for asterisk?
9:24AM 4 AGI Flakyness *sigh*
9:17AM 0 Lots of lost interrupts when running HFC ISDN card in NT1 mode
8:57AM 3 Problem making outbound calls on TE210P using NFAS
8:39AM 1 Non sensical AGI Error
7:23AM 1 Firmware version 1.3.1 released for AastraIPphones
6:54AM 0 error on AMP route
6:46AM 2 show calls
6:43AM 0 AGI onAnswer function: does it exist?
6:40AM 0 Status UNKNOWN
6:28AM 4 Firmware version 1.3.1 released for Aastra IPphones
5:55AM 2 asterisk-1.2.4 + asterisk-addons-1.2.1 for mysql realtime
5:43AM 1 BT102 and ringtones
4:56AM 12 asterisk h323
4:30AM 0 Call Detail Records for Inbound Calls
4:06AM 10 iax2 trunking known problems?
3:35AM 6 FXO port on TDM400P hangs!!
2:49AM 0 Asterisk 1.2.4 (behind NAT) IAX registration "Refresh 0" problem
Wednesday February 15 2006
11:03PM 2 Dialing multiple phones with Macro-exten-vm
9:33PM 0 L option of Dial does not work properly
7:57PM 0 Speex echo cancellation
4:23PM 0 Asterisk - Vega 50 Disconnect Issues
3:45PM 0 [asterisk-dev] Zaptel 1.2.4 Released!
3:21PM 1 Zaptel 1.2.4 Released!
3:09PM 2 Increment Variable
2:51PM 3 Anyway to pass CIC in sip header
2:46PM 0 is there a web interface to this mailing lis t?
2:43PM 4 Channel bank woes - no outbound calls
2:35PM 12 Random Disconnects - or ARE they?
2:29PM 11 is there a web interface to this mailing list?
1:45PM 3 Bridge Calls with G()
12:50PM 0 [CAVPdiscussion] OT: RFC: Canadian Association o f Voice over IP Users (CAVU)
12:05PM 3 Alarmreceiver
11:32AM 5 Automated wake up call
11:28AM 2 PIKA Technologies Inc. Announces Support for Open Source Asterisk PBX
11:05AM 5 SPA-941 stutter tone
10:59AM 4 Hint priority
10:49AM 3 Channel bleedover?
10:44AM 0 arris e-mta
10:15AM 1 problem with outgoing callsUnabletocreatechannel of type 'ZAP' (cause 34 - Circuit/channelcongestion)
10:05AM 0 Channel SS7
9:43AM 1 Next Montreal meeting - the 21st - featuring a conference call with Mark Spencer
9:21AM 7 asterisk silence suppression?
8:26AM 4 Software E.C. Along with Tellabs
8:12AM 0 queue_log analysis
8:08AM 4 Fwd: Which ATA device do you recommend?
7:42AM 3 CDR for Inbound Calls
7:27AM 0 forward to gateway
7:14AM 0 VOIP provider iristel, setup account
7:01AM 0 which ATA SIP is better with asterisk
6:58AM 4 G723 error
6:29AM 1 interface to dpnss
5:50AM 1 Asterisk large-scale deployment w/analog phones
5:32AM 0 Zaptel problem on 4 Processor Opteron SMP system
5:14AM 11 Aasterisk large-scale deployment w/analog phones
4:47AM 0 Switch statement
4:42AM 4 SIP and firewalls?
4:35AM 2 Asterisk running on DMZ (no NAT) PROBLEMS- OPTION message is out of State
3:21AM 0 Brief pauses during calls
3:11AM 0 inbound DID trunked
Tuesday February 14 2006
8:22PM 5 asterisk t.38 pass
7:00PM 1 Firmware version 1.3.1 released for Aastra IP phones
5:25PM 0 Adjusting frequency asterisk sends NOTIFY's to ATA's at for MWI.
3:09PM 14 Good VoIP providers that support Asterisk PBX's
2:43PM 0 can't dial zap extensions?
1:35PM 0 Changes to sip.conf in 1.2.x ?
1:28PM 12 Asterisk and Snom 360
1:28PM 0 Not passing CALLER id on in follow me script
1:12PM 3 ZAP extension, DTMF?
1:05PM 3 Grandstream hold one way audio -URGENT
1:03PM 0 asterisk and S.E.R.
12:53PM 9 Fax to Email with Asterisk and Lucent TNT
12:38PM 0 How to create latency on purpose
12:09PM 9 Multiple AGI Issues
11:56AM 8 BRI Newbie - What Hardware, PCI, in the US?
11:56AM 2 Softphone and 911
11:40AM 0 Skilled API consultant required - preferablywith intergration
11:21AM 0 Dial command to connect two channelsand bypassasterisk server
10:15AM 2 Instant Messaging: with SIP or XMPP
10:11AM 1 [help] warning 4246
10:07AM 0 Bristuff-0.3.0-PRE-1l and TDM400 with fxo ports
10:07AM 5 Podget or Similar
8:56AM 2 Rough Two Days
8:51AM 2 Use one sip account for multiple sipura
8:38AM 3 Nat, SIP, Realtime problem
8:30AM 1 SPA-941/2 Monitoring
8:26AM 3 Dial command to connect two channels and bypassasterisk server
8:25AM 0 Guidance need for trunking using SIP
8:17AM 7 ChanIsAvail
8:14AM 16 Solution for 1 time blast of 200, 000 recorded calls
8:04AM 0 Planet VoIP Phones
8:03AM 6 consult about Digium Card
8:02AM 1 Can Asterisk send RTP to a specific port number?
7:26AM 0 Lucent Avaya Partner ACS T1 module
7:02AM 0 Help Asterisk with Phoneserve
6:39AM 2 about g729 license
6:13AM 0 SIP Header VIA when behind NAT
6:10AM 0 Asterisk and MOH for Queues
5:40AM 2 Telmex PRI line configuration problem
5:21AM 5 Developing a call centre app. Communication with asterisk?
5:21AM 0 uniden uip200 loosing registeration
4:18AM 1 voicemail recording format
4:04AM 7 audio cuts out
2:29AM 1 fax pass-through
1:37AM 10 Call centre - * hang's up
Monday February 13 2006
11:20PM 2 Different Voice Prompts at Different Times
11:08PM 3 Dial command to connect two channels and bypass asterisk server
9:26PM 2 Terminating AGI Scripts
8:02PM 0 Problem Faxing
7:08PM 0 RE: Asterisk-Users Digest, Vol 19, Issue 90
7:00PM 1 Asterisk: Agent logs into queue, and there are calls in the queue, but calls don't go to agent
6:22PM 1 Asterisk: Agent logs into queue, and there are calls in the queue, but calls don't go to agent.
4:53PM 2 Skilled API consultant required - preferably with intergration
4:52PM 1 problem with outgoing calls Unabletocreatechannel of type 'ZAP' (cause 34 - Circuit/channel congestion)
4:46PM 0 Send HookFlash after answering a ZAP(analog) channel
4:43PM 0 AGI Scripts Staying in Memory
4:37PM 2 Asterisk Televantage integration
4:36PM 0 HELP, SPA-2002 - SPA-2002 singleside sound
4:23PM 6 Traffic prioritization and 'class of service' for SIP
3:50PM 1 sip expire 60
2:21PM 1 iLBC issue: An ilbc frame that isn't a multiple of 50 bytes long from RTP (38)
1:49PM 0 Manager cmd: originate without picking up thefone?!
1:41PM 1 Sagoma w/EC x TE411
1:15PM 1 Manager cmd: originate without picking up the fone?!
1:08PM 0 Detecting Agents and Chanspy
12:26PM 1 RE: Asterisk-Users Digest, Vol 19, Issue 89
11:43AM 2 TDM04B/TDM2401E Card
11:16AM 1 Send HookFlash after answering a ZAP (analog) channel
9:45AM 0 AAH 2.5 pone paging broken
9:28AM 0 Asterisk 1.2.4 Quality Issues
9:23AM 6 TAPI Recommendations
9:14AM 1 Bug in AMP 1.10.010 in sip outbound callerid
9:03AM 0 Call over SIP channel becomes a zombie
9:02AM 0 Asterisk register ip phone
8:28AM 2 Why is asterisk ignoring my context?
8:24AM 0 FXO port on TDM400P hangs
8:18AM 0 trunk 2 IAX server :- getting error ' Unable to support trunking on user 'ho' without zaptel timing'
8:16AM 1 automatically start application from the commandprompt
8:09AM 0 automatically start application from the command prompt
7:53AM 1 asterisk still tries native bridging
7:50AM 1 PrivacyManager Broken?
7:44AM 2 problem with outgoing calls Unable tocreatechannel of type 'ZAP' (cause 34 - Circuit/channel congestion)
6:37AM 1 problem with outgoing calls Unable to createchannel of type 'ZAP' (cause 34 - Circuit/channel congestion)
6:17AM 4 Voicemail - direct call
4:48AM 0 Limiting SIP bandwidth
4:25AM 4 Waiting for your help...
4:24AM 0 problem with outgoing calls Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion)
3:39AM 1 How to Get SIP Header : To Field ?
3:13AM 2 Alcatel 4200 series pbx
2:08AM 0 Hooking up with Ser
Sunday February 12 2006
11:36PM 9 Best quad-port fxo solution with EC?
2:53PM 0 Asterisk with
2:49PM 6 Aastra phones and common directory?
1:10PM 0 SIP massive deregistration
1:06PM 0 asterisk call start detection
12:34PM 1 Softphone --> Bluetooth Smartphone
11:51AM 2 IP phone with many speed dial buttons
11:04AM 0 Voice Drop Due to Low RAM
9:16AM 3 dtmfmode=auto, but doesn't work
8:06AM 4 To connect between more than 2 asterisk server [ links needed ]
8:05AM 1 help on dial plan
7:38AM 11 asterisk + door opener
2:25AM 3 Zap, Caller ID problem
2:05AM 5 Problem with Playback sound in 64 bit machine
2:04AM 2 dual TE410, both span 3 is broken
1:22AM 0 strange problem with asterisk in media proxy mode
Saturday February 11 2006
8:13PM 0 bad sound frequency
7:24PM 7 Problem with Wait() and chan_capi-cm?
6:40PM 0 Dell server
2:45PM 0 Busy signalling for mobile callers ?
2:15PM 1 MOH broke with 1.2.4 .. ?
1:21PM 0 RE: ex-girlfriend (ex-boyfriend)
12:47PM 3 Codec issue with my iaxy
11:30AM 1 Asterisk 1.2.4 and IAX MOH
7:48AM 3 TE411P Really Bad Echo ORION
7:48AM 0 Can I configure the console to ring on one sound card and the headset on another sound card?
5:09AM 6 configure TE205P on asterisk@home
4:18AM 5 Dialing part of the extension
4:15AM 1 Help with dialplan
4:12AM 0 Problem with CLI output on Asterisk@Home
4:08AM 0 Qwest disconnect supervision?
3:38AM 1 FYI: new firmware for 7905/12 - RPID support
3:09AM 2 No Voice when canreinvite=no
1:44AM 0 Chan capi failing post build 8015, possible causes?
Friday February 10 2006
11:51PM 0 ruby-agi-1.1.1 released !
10:38PM 2 OH323 Peer
10:17PM 1 Error running iaxcomm
10:03PM 0 RE: ex-girlfriend (ex-boyfriend)
8:29PM 0 Vegastream clockslip problems
5:36PM 2 Working SPA 841s now return 404 Not Found for INVITES and OPTION packets from *
4:54PM 0 IAX Extension Dialing Question
3:45PM 0 Asterisk with Cepstral for TTS
3:05PM 0 TDD
3:00PM 2 Asterisk 1.2.x + oh323 on Debian Sarge.
2:47PM 1 Virtual Extensions
2:42PM 8 Sendmail with exchange
1:58PM 5 Setting up Polycom 501 with 2 Different Extensions
1:24PM 0 Agent supervisor configuration
12:59PM 0 Repeating Zap Message
12:30PM 0 Multiple Asterisk Server Question
11:15AM 6 More Polycom IP501 questions
11:09AM 0 I: ZapRas
11:08AM 0 Re: Asterisk-Users Digest, Vol 19, Issue 72
11:06AM 1 SIP Aliases
10:52AM 0 SIP compact headers
10:49AM 2 MixMonitor & ControlPlayback of g729 files
10:15AM 1 Cisco 79XX firmware 7.5
10:02AM 1 Some articles
9:51AM 1 T1 Channel splitting PRI/data not working
9:40AM 0 TDM - Analog Trunk - CallerID question
9:37AM 0 calling to sip provider
9:16AM 0 Forwarding any number issue
8:56AM 0 Yuck! Asterisk Crash...
8:16AM 0 Make Meetme start only when somebody puts in the admin PIN
8:08AM 2 RE: ex-girlfriend (ex-boyfriend)
8:04AM 0 Half Solved - Fail over to Pri on VoIP connection failure
8:02AM 1 Problems with Cepstral and Asterisk
7:16AM 8 Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...
6:18AM 0 cdr (again) and deadlocks
4:12AM 0 Sip + Cisco 7940/7960 + Panel + DND + queues
4:07AM 2 [ Re: [ RE: Corrupt CDR records in Asterisk 1.2.x]]
3:41AM 2 QSIG error -- can somebody explain?
3:19AM 2 Obtaining billsecs in the dialplan after a call?
2:13AM 2 2wav2mp3, monitor, mixmonitor, mpg123, queues
2:00AM 0 Any way to grep through fast moving consolemessages?
1:51AM 1 Expression GotoIf - bug or personal misunderstanding?
1:22AM 1 STUPID question? Tellabs echo can cards and PSTN?
Thursday February 9 2006
10:56PM 3 IP Authorization
10:19PM 2 Asterisk - Brooktrout
8:57PM 5 Problem win Unicall
7:55PM 1 4 TE411P in one server installation
7:39PM 1 TE210P + MicroITX as E1 to TDMoE appliance?
7:36PM 2 How come I don't have the MeetMe applicationregistered?
7:22PM 2 Any way to grep through fast moving console messages?
5:37PM 3 Unistim Packet Decoder
4:18PM 0 Possible for Asterisk to output CLID to invoke3rdparty app?
4:13PM 0 Possible for Asterisk to output CLID to invoke 3rdparty app?
4:10PM 1 Possible for Asterisk to output CLID to invo ke 3rd party app?
3:48PM 3 Asterisk and Xen
3:45PM 1 Possible for Asterisk to output CLID to invoke 3rd party app?
3:31PM 0 RE: Is my math on traffic/bandwidth correct?
3:09PM 2 TDM2400P FXS Only vs. T1/E1 to FXS Channel Banks
3:00PM 0 re: voipjet -- Workaround if needed
2:54PM 1 Polycom remapping SpeedDials
1:27PM 1 Problems with gnugk, asterisk, and ooh323
1:13PM 1 Static problems with Asterisk + Polycom phones
12:15PM 0 Manager API 'Redirect' is not working for both end of a call.
12:03PM 0 SOLVED: Re: Polycom IP501 with Asterisk -distinctive ring?
12:02PM 0 SOLVED: Re: Polycom IP501 with Asterisk -distinctive ring?
11:49AM 3 Optimizing Linux to run Asterisk
11:32AM 0 Dumb question... block 00
10:48AM 0 FXS ATA and Pots wiring
10:42AM 1 Re: Help on Vicidial
10:39AM 1 How come I don't have the MeetMe application registered?
10:12AM 0 re: Polycom IP501 with Asterisk - distinctive ring
10:10AM 1 Re: Polycom IP501 with Asterisk - distinctive
10:08AM 0 Caller stuck in MoH after being answered by a phone that was forwarded to.
9:54AM 0 ztmonitor output weirdness
9:31AM 1 Issues in Australia? Ringing, iaxy etc
9:24AM 0 I need help on VICIDIAL and auto dial
9:04AM 0 tdm400p setup in china question
8:55AM 3 Meetme echo cancellation
8:55AM 0 Asterisk Native Sounds re-release
8:49AM 2 Polycom IP501 with Asterisk - distinctive ring?
8:25AM 2 stable ISDN BRI card for asterisk
8:10AM 0 Sip One way audio
7:14AM 15 Asterisk vs. Traditional PBX
7:06AM 1 Asterisk with Billing
6:30AM 0 Question on SIP authentication with users from OpenSER
6:10AM 5 Dell PowerEdge 1800 and TE410P
5:56AM 27 asterisk logger - urgent!!!
5:47AM 0 Queue transfer
4:32AM 0 Asterisk 1.2.x + ooh323 from addons - incoming call goes always to default context.
4:32AM 1 clid and src fields wrong in cdr
4:29AM 0 Busy problem
4:14AM 1 SPA-3000 VOIP-PSTN gateway - longdelaybetweenanswering and ringing
3:05AM 6 Corrupt CDR records in Asterisk 1.2.x
2:22AM 1 Leading 0 on caller ID with internal S0 (HFC)
2:16AM 4 Queue - check agent
2:15AM 9 sip to oh323 converter converts sip uri to h.323 number and not h.323 url
2:15AM 0 Firefly & iaxLite dont stop ringing when answering incoming call
1:34AM 1 How can I send DTMF from the console?
1:10AM 1 TDM400p
12:43AM 1 Voicemailmain() refusing connection problem
12:25AM 0 Fax transmission interrupt on ISDN network
12:18AM 0 Queue - joinempty
12:17AM 9 What ATA should I buy?
12:15AM 0 NSLU2 Asterisk
Wednesday February 8 2006
11:34PM 0 Re:
11:31PM 0 Re: Asterisk-Users Digest, Vol 19, Issue 58
11:20PM 0 Re: Asterisk-Users Digest, Vol 19, Issue 58
10:43PM 0 OOH323 Configuration
10:16PM 3 Polycom dialplan restriction
9:54PM 0 SIP-H323 Help and Multiple Listening Port
8:55PM 0 Asterisk returning 403 Forbidden response
7:40PM 0 Faint background noise/crackle on FXS porton TDM400P
7:21PM 1 incoming call release after 1 ring
6:36PM 3 Bandwidth: to seperate or not to seperate
6:26PM 5 Two Lines, Two Businesses
6:15PM 0 PRI Group behavior - CHANUNAVAIL
6:10PM 9 ztdummy on gentoo 2005.1
6:00PM 3 sip channel status - how?
5:59PM 5 Remapping Polycom IP501 buttons
3:49PM 0 Zap Auto disconnect after xx seconds of silence
3:46PM 1 Re: Need to retrieve Call-ID from dialed number
3:43PM 7 lists problem, Gmail????????
3:33PM 1 Digium TDM04B Outbound routing
3:05PM 0 MINNESOTA: TwinCities Asterisk Users Group - Saturday 02/11/2006
2:57PM 6 Connecting to live calls
2:47PM 1 Polycom IP501 MWI goes off periodically
1:47PM 0 "Say YES to continue" prompts
1:21PM 3 sipura 3000 and other probs
1:05PM 3 SPA-3000 VOIP-PSTN gateway - long delay between answering and ringing
12:34PM 2 Problem with Incoming Caller ID on Nufone Since Upgrade
12:20PM 10 sipdiscount
12:03PM 16 more cpu intensive echo cancellers ?
11:54AM 7 odd 'digital' sound artifacts
11:43AM 0 Need to retrieve Call-ID from dialed SIP channel(w/o CDRs)
11:39AM 1 Handset phone to replace Flash Operator Pane l
10:19AM 3 fax <-> misdn <-> zap <-> fax // fax <-> misdn <-> ATA <-> fax
10:04AM 3 Need to retrieve Call-ID from dialed SIP channel (w/o CDRs)
9:21AM 2 Chan_BT question WAS: Asterisk with USB
9:14AM 1 SIP on IP aliases
8:51AM 0 ARI - Voicemail not showing - Problem solved!
8:30AM 0 SIP to H.323 Native bridging ...
8:22AM 4 PRI to PRI not passing callerid
7:50AM 2 SV: GotoIf number exists in file. How can i do this?
7:17AM 0 Cisco 7920 wi-phone firmware
7:11AM 3 Performance differences 64-bit vs 32-bit
6:37AM 4 GotoIf number exists in file. How can i do this?
6:22AM 2 RE: X100P help required
6:01AM 2 channel.c: Avoided deadlock for '0x91a8b20', 10 retries!
5:40AM 0 Asterisk and Cisco AS5350
4:37AM 0 agents.conf
4:30AM 2 PRI Bridging and Recording
4:28AM 1 Possible AGI Bug in Asterisk?
3:24AM 2 Faint background noise/crackle on FXS port on TDM400P
2:04AM 16 Fedora Core 3 or Fedora Core 4? yum update ornot?
Tuesday February 7 2006
11:26PM 1 MeetMe - Party's are not exchanging Audio - Is this BUG?
10:45PM 2 Handset phone to replace Flash Operator Panel
9:41PM 0 Fedora Core 3 or Fedora Core 4? yum update o r not? also: SpanDSP -pre25 for 1.0.9 is out w00t!
8:28PM 0 FXO Line not Hanged up
8:26PM 3 Mitel 5220 IP phones
8:22PM 0 RE: Asterisk-Users Digest, Vol 19, Issue 47
8:21PM 0 Re: Opinions needed on call quality vs
7:19PM 1 orphaned sip channels channels?
6:28PM 5 Sipura SPA 3000 logic
4:12PM 4 alternative to realtime?
3:13PM 0 Coppercom SIP experience?
3:07PM 4 touch tones too fast ?
3:06PM 0 Help on queues
3:06PM 0 Secure voicemail passwords?
2:58PM 0 moh about twice as fast
2:51PM 0 xlite and letters
1:27PM 1 Opinions needed on call quality vs network latency
1:16PM 2 Re: two tellabs 2572 echo board in a 253c mounting
1:13PM 0 Multiple call groups
1:07PM 2 IVR Menu
12:27PM 1 SetCallerID and CDR
12:26PM 1 AMP 1.10.010 Config Problem
11:10AM 10 911 and ISDN PRI
10:22AM 0 Not receving anything from the list
9:20AM 0 extension h and DeadAGI
8:43AM 1 MFC/R2 in Brazil
7:55AM 10 Asterisk with USB
7:48AM 0 TDM Cross-connection
7:26AM 14 virtual extension per user ?
6:40AM 0 Broken faxes when other call disconnects
6:30AM 3 Better i18n for Asterisk?
6:27AM 0 problem with Zaptel
6:23AM 3 asterisk and week-ends
4:53AM 4 ATA's and faxing
3:03AM 4 No sound on 10% of incoming calls
2:09AM 0 Modifying dialplan for DUNDi compatibility
1:46AM 0 transferred calls: not 2 but only 1 recorded by cdr
1:41AM 3 asterisk to FWD
1:38AM 3 Problem with ZAPHFC: internal S0 hangs when hanging up
1:05AM 4 chan_bluetooth - concurrent calls?
12:30AM 0 cosmetic bug on CLI ?
12:07AM 8 Welltech USA? and Wellgate Products?
Monday February 6 2006
8:56PM 2 dummy Technology/resource for Dial
5:15PM 0 Polycom 501 netboot not working
4:59PM 0 New issue tracker for handling licensing issues for Asterisk, Zaptel and related projects
4:31PM 5 Free IAX login
4:20PM 4 FXS with v.90 modem support?
4:18PM 3 TDM04B FXO Asterisk@Home
4:00PM 16 Cisco 2620 as PRI gateway
3:58PM 3 bug in bristuff?
3:38PM 3 seg fault 1.2.4
3:11PM 4 two tellabs 2572 echo board in a 253c mounting assembly?
3:04PM 0 Re: Will not authenticate incoming VOIP provider
2:56PM 0 Can Asterisk and new ShoreTel 6 talk to each other?
2:07PM 0 Cannot Dial Out From *
1:52PM 0 Cannot Dial out.
1:15PM 0 PGSQL asterisk command
1:05PM 3 One way audio - it doesn't make sense
12:36PM 2 New GSM 1-8 ports Gateway / Terminal for sale (with SMS Feature and Many more) £99 per unit
11:59AM 3 echo cancel from telco
11:54AM 0 wrong dell
11:48AM 2 thomson speedtouch ST2030
11:44AM 1 Will not authenticate incoming VOIP provider calls
11:40AM 0 TDM421p: Noisy FXS problem
11:19AM 2 asterisk 1.2.4 seg faulting today had been working fine since update
11:18AM 0 Oh323 channel problem
10:51AM 4 SV: Help on queues
10:48AM 35 Asterisk native sounds now available!
10:33AM 0 Called party number
10:24AM 1 Deploying VoIP on a WAN
10:20AM 12 change languages from an IVR
10:11AM 0 Asterisk + Avaya DTMF problem
9:44AM 3 php agi configuration issue
9:42AM 1 TE210P mother board
8:52AM 0 PRI in spain with ONO
8:45AM 3 Uniden UIP200 and Asterisk v1.2.4: problem not registering
7:03AM 7 Rtp packets being dropped
6:44AM 1 Problem with ARI and seeing voicemail...
6:33AM 0 DTMF level
6:32AM 0 Re: Asterisk-Users Digest, Vol 19, Issue 34
5:27AM 0 codecs choice
5:19AM 0 Some feedback and issues on using chan_bluetooth
5:10AM 1 .version in zaptel
4:34AM 2 IAX registration expiration
4:04AM 2 SV: BAD/GOOD Echo Cancel
3:47AM 0 Channel juggling, what is it good for?
3:19AM 29 BAD/GOOD Echo Cancel
3:18AM 4 SV: callback script?
1:27AM 1 intel 536 ep as fxo -> possible?
12:46AM 1 French and German translations?
Sunday February 5 2006
8:36PM 45 TE411P Really Bad Echo
7:06PM 1 AVAYA H.323 IP phone account and Asterisk
4:08PM 0 Strata DK280 + Asterisk@Home
3:57PM 2 1 ISDN BRI to IAX2/SIP... (*) best tool or?...
3:30PM 9 Billing inbound calls per minute
1:38PM 4 (newby) Asterisk on the open internet & security
5:59AM 0 Sirrix PC140 Quad card
5:55AM 4 re: questions about sip requests to asterisk 1.2
1:29AM 13 IP PAX gateway to PSTN
1:00AM 0 ??: Search for Links for "Communicating PC to PC inthe same lan through Asterisk "
12:51AM 4 Search for Links for "Communicating PC to PC in the same lan through Asterisk "
12:23AM 0 early media
Saturday February 4 2006
7:43PM 0 Difference between VoiceMail and VoiceMail2?
3:05PM 0 Maximum retries exceeded on call/phantom calls?
12:08PM 0 Visio-type symbol for an Asterisk/VoIP server?
10:59AM 0 How can I configure to call from the consolebymeans of a sip phone,
7:28AM 8 Routing Calls via chan_capi with AVM FritzCard
6:57AM 0 No audio for outgoing calls
12:10AM 0 How can I configure to call from the console bymeans of a sip phone,
Friday February 3 2006
10:59PM 0 Asterisk SIP phones to Cisco Unity viaCCM4.0SIPTrunk
10:43PM 1 64bit processor and 32 bit digium card
9:02PM 5 g729 license question
8:56PM 2 User web portal for Asterisk
7:59PM 3 can asterisk to say chinese like say english
6:29PM 1 Cisco AS5350
5:44PM 1 Zaptel 1.2.3 with Asterisk 1.0.9
4:19PM 1 Fast AGI performance question
3:40PM 3 MWI on Polycom 501.
3:11PM 1 No path to translate from Zap to SIP
1:59PM 3 Calls fading in and out
1:46PM 0 error cdr mysql addon
1:36PM 1 RE: 5, 000 concurrent calls system rolloutquestion
1:00PM 1 Re: delaying "answer" for a number of ring or an amount of time
12:38PM 0 Re: Sipura SPA-2002 rings randomly
11:29AM 0 php+agi
11:23AM 0 Events when the target of the call
10:59AM 2 chan_sccp availability?
10:03AM 3 click to talk
9:53AM 0 FW: Web Interface
9:11AM 0 varion card
8:14AM 2 Pattern Match - 0 or 1 digit
7:37AM 16 Re: delaying "answer" for a number of rings or an amount
7:15AM 2 Events when the target of the call answer
6:16AM 0 Re: [Serusers] high-availibility setup using f5 bigip
6:08AM 4 hardware and network requirements
6:07AM 4 inform the agent about the queue he is answering
5:05AM 0 Musiconhold in zapata.conf
4:54AM 0 How can I configure to call from the console by means of a sip phone,
4:49AM 7 cmd set with multiple values
3:18AM 16 CallerID popup
3:11AM 3 international calling via POTS in Russia
1:33AM 0 Pound to Hangup an ongoing call
12:49AM 10 SV: SV: delaying "answer" for a number of ringsor anamount of time
12:21AM 0 TDM 400 FXO FXS Test
Thursday February 2 2006
10:51PM 3 Configuring Meeting Room from Asterisk Manager API
10:11PM 2 RE: 5, 000 concurrent calls system rolloutquestion
8:27PM 2 Zhone channel Banks
7:32PM 6 PRI Presentation Restricted bit honored?
7:22PM 3 Re: delaying "answer" for a number of ringsor an amount of time
5:31PM 2 Any Digium Supplier/reseller accepts Paypal ?
4:31PM 2 routing question: multipath routing for SIP
2:40PM 0 Re: 5, 000 concurrent calls system rollout question
2:08PM 1 SV: delaying "answer" for a number of rings or anamount of time
2:05PM 0 stream file 16k sample and 16 bit data
2:00PM 1 RE: 5, 000 concurrent calls system rollout question
1:51PM 3 Slightly OT: and Freeswitch
1:00PM 0 Fw: Agents, queues and zombies
12:51PM 0 Agents, queues and zombies
12:14PM 2 delaying "answer" for a number of rings or an amount of time
12:12PM 9 How to handle "provider UNREACHABLE" in the dialplan?
11:44AM 0 Events when the target answer
11:32AM 0 Sip - no peer or user found on incoming call
11:25AM 5 ISDN Eicon Diva Server V-BRI
11:17AM 0 Asterisk at SCALE 4x
10:58AM 0 POTS lines vs. using T1 to connectphoneservices?? HELP
9:35AM 0 Re: Euro-ISDN
9:33AM 0 Re: Euro-ISDN
9:28AM 1 return code from AGI
9:26AM 1 Callerid Name
9:21AM 8 limit sip sessions
8:45AM 0 POTS lines vs. using T1 to connect phoneservices?? HELP
8:41AM 0 Anyone know a good ITSP in Canada that suppo rts *?
8:27AM 19 Rewind MusicOnHold?
7:47AM 1 Re: Contents of Asterisk-Users digest...
7:41AM 1 POTS lines vs. using T1 to connect phone services?? HELP
6:20AM 2 Regarding cdr_manager.conf
6:20AM 13 Asterisk on laptop connected to POTS line
6:17AM 4 Call completes but then drops?
5:41AM 0 [Fwd: Re: Asterisk for Call Center (missing reference)]
5:39AM 3 Anyone know a good ITSP in Canada that supports *?
5:27AM 2 Outbound Call & SIP Results
5:13AM 2 DeadAGI variables confusion
5:12AM 0 SV: Outbound Caller ID number on E1
4:36AM 4 Outbound Caller ID number on E1
2:50AM 1 Pri Hang up outgoing calls
2:40AM 2 callback script?
1:40AM 11 OT O'Reilly Asterisk TFOT
1:25AM 1 Setting MSN for outgoing ISDN calls
1:04AM 0 realtime queue not working realtime in asterisk versions above 1.2.0
1:04AM 0 agi/cagi call limit using group_count
Wednesday February 1 2006
9:46PM 1 Anyone in or around Redmond, WA?
9:15PM 3 TE411P or TE406P
8:19PM 2 supermicro server model
8:13PM 0 determining if a call to a SIP extensions isfrom a queue
7:46PM 4 Anyway to do this?
6:03PM 4 winnipeg canada
5:19PM 1 Asterisk SIP phones to Cisco Unity via CCM4.0SIPTrunk
4:46PM 1 RE: Asterisk-Users Digest, Vol 19, Issue 10
4:13PM 2 fax possibilities
4:05PM 0 Work in Ukraine
3:53PM 0 Re: Polycomm IP600 continues to ring
2:46PM 1 SV: Re: CallerID Problem
2:32PM 0 SV: Re: CallerID Problem
2:16PM 2 No Audio on Local Machine, Remote works fine
2:12PM 0 how to log agents into a queue
1:29PM 7 Blocked Callerid
1:11PM 3 Dumb Dialout Question
12:46PM 0 RESOLUTION: SetCDRUserField not working in A@H?
12:01PM 7 Receiving faxes with spandsp - strange problem
11:58AM 1 Caller ID patches - updated
11:03AM 7 DTMF Sporadicaly Being Generated
10:56AM 1 New version of snom soft phone
10:48AM 2 Dundi key Problem
10:34AM 0 Direct pickup
10:22AM 2 RE: Euro-ISDN
10:09AM 1 Digit timeouts vs includes in diaplan
9:26AM 0 asterisk 1.2.1: everything ok but strange messages appear on linux console
9:19AM 0 RE: Asterisk-Users Digest, Vol 19, Issue 6
8:41AM 5 changing cisco 7940/7960 standard menus ?
8:40AM 0 iax2 native transfer question.
8:02AM 4 XLite dtmf issue?
7:32AM 1 determining if a call to a SIP extensions is from a queue
7:22AM 4 Cisco Gateway and Context Issues
7:14AM 1 query about Three way calling
7:07AM 0 Dial command exits non-zero
7:05AM 0 RE: Asterisk-Users Digest, Vol 18, Issue 206
6:49AM 0 test of FXO and FXS in TDM400P
6:09AM 0 Please help - access out side number after waiting few seconds
5:59AM 1 Swapping lines using dtmf
5:21AM 0 asttapi 0.08 - the memory could not be written
5:12AM 16 (newby) Is PING a good indicator of latency?
5:12AM 3 (newby) EURO-ISDN line question
5:11AM 12 (newby) IAX Trunk on low bandwidth connection
4:59AM 0 Help with Grandstream Handytone 386 together with Asterisk and a connected modem
4:19AM 0 can't hear 'service messages' when iax is in the middle
2:08AM 0 a recipe for compiling asterisk 1.2.4 with h.323 support
2:08AM 1 RE: Euro-ISDN
2:03AM 0 SRV mapped to host
1:54AM 1 SetCDRUserField not working in A@H?
1:19AM 1 Unable to Register to Asterisk through Proxy
12:43AM 1 ISDN busy line