asterisk users - Jan 2007

Wednesday January 31 2007
TimeRepliesSubject
11:56PM 1 Softphone for Palm
10:48PM 1 Fax from PAP2 through a zap channel to PSTN
8:40PM 4 no lights on TE405P, but shows up in lspci, modules loaded
7:57PM 8 kewlstart disconnect threshold
7:52PM 4 FreePBX/Debian Aborts Call While Connecting
7:14PM 5 Which Java FastAGI implementation has the most "market share"?
6:28PM 0 How would you compare feature set to a Metaswitch?
4:44PM 5 how to get the status of failed call files
12:18PM 0 jastAGI
11:34AM 1 Polycom IP 501+India
11:09AM 6 Help with semaphores
10:45AM 0 E911 Bill Announced
10:04AM 0 Compiling NVFaxDetect and other Newman apps on Asterisk 1.4
9:59AM 5 Queue Status
9:57AM 0 Storing recordings
9:11AM 0 (no subject)
8:50AM 0 Line drops strange problem(got event On hook)
8:23AM 8 Testing IVR / Callcenter applications
6:36AM 3 Hi Honies! I'm home!
6:21AM 2 Regarding Call Queue
4:14AM 0 Asterisk 1.2.14 bristuff app_pickup.so
1:35AM 0 ELMEG IP290 and voicemail
 
Tuesday January 30 2007
TimeRepliesSubject
2:44PM 1 OT: Asterisk 1.2.X, IAXModem 0.2.0 + HylaFAX+ 5.0.3 interop probl em
1:26PM 2 Should I use sip gateway of PCI card?
1:00PM 7 Toll-free dialing via PRI problem
11:48AM 1 Record file name Agent
11:44AM 0 Signaling OK but no voice through X100P
11:06AM 8 Queue Dial Plan
10:44AM 2 Cisco SmartSwitch
10:22AM 11 Give "Busy" to the 3rd call on a BRI using chan_capi
10:18AM 1 One-way audio after several minutes 1.4.0
9:55AM 2 web-meetme cbmysql not registered
9:52AM 11 musiconhold restarts for every extension
9:30AM 0 Diva PCI 2.01 + isdn2linux + asterisk: Dropped a signal frame
9:05AM 20 Re: [asterisk-dev] Dynamically Adding A Context
8:17AM 6 Dynamically Adding A Context
8:12AM 11 Asterisk dual contexts stupidity
7:48AM 0 Looking for Sugar CRM installer for an Asterisk
7:24AM 2 Comments on Billing reconcillation with providers
7:02AM 1 No intercom splash tone?
5:58AM 1 Strange problem
4:45AM 6 Problem with Voipjet ...
3:16AM 1 snmp Monitor for asterisk boxes
 
Monday January 29 2007
TimeRepliesSubject
10:38PM 2 Timeout in IAX vs SIP
9:14PM 2 detecting avaya busy tone
9:08PM 0 Cisco PRI gateway with MGCP control
7:28PM 0 "disconnect clear time" -- calling party control and TDM-400
6:35PM 1 TDM Cards or PSTN>VOIP Gateways?
3:45PM 0 Dropped call issue with IAX Trunking
3:16PM 2 Asterisk, VoIP and Linux Blog.
10:39AM 25 Installed TDM02B - Problem when other end hangs up
10:00AM 1 internal and external interfaces
8:31AM 1 SIP + short numbers + name of customer
8:02AM 1 LookupCIDName / LookupBlacklist syntax
7:55AM 4 put Agi script in queue
6:40AM 0 Rxfax and txfax
5:57AM 4 Pickup() ringing extension and call waiting
3:58AM 2 licence quick question
3:29AM 1 parsing extensions
3:08AM 0 SIP SDP keep original codec selection?
12:26AM 5 Rxfax and Txfax on Asterisk 1.4
 
Sunday January 28 2007
TimeRepliesSubject
8:09PM 1 Queue Manager
7:52PM 6 Cordless SIP Phones
7:06PM 0 Trouble outgoing VOIP Provider Calls
6:13PM 0 Test Hardware
6:06PM 2 Trouble with incoming calls
5:42PM 6 Heartbeat on Digium T1 PCI cards?
4:16PM 8 Voicemail from sip phones
4:03PM 0 Automating the setting/clearing of a flag
3:09PM 0 Add current extension dynamically to template?
2:51PM 0 Re: Migration to Asterisk 1.4
1:39PM 0 PHP sip client
12:54PM 6 T1 Wire Level Tapping
9:41AM 0 Channels Banks that support neon MWI
8:15AM 11 NAT: RTP Path Optimization
8:04AM 4 Mabe OT? What managed switch is best for VoIP application?
7:07AM 3 Enterprise quality SIP provider
4:16AM 0 AsteriskNow - H323 support for trunks
2:03AM 2 Transfer on RTP timeout?
 
Saturday January 27 2007
TimeRepliesSubject
10:08PM 2 Response on dialin - no extension
3:32PM 1 HFC-card and TDM400 with bristuff
1:31PM 1 FXS - Init Indirect Registers UNSUCCESSFULLY.
12:18PM 0 BarCampUSA Tickets go on sale this Thursday the 1st of February 2007
12:09PM 1 How to fix error when paging
10:55AM 2 max tnt pri voice channels 56k or 64k, does it matter, selection parameter?
8:20AM 10 H.264 *Not Patented*
5:50AM 3 Simple question
4:17AM 6 Via EPIA channel_find_locked: Avoided initial deadlock
 
Friday January 26 2007
TimeRepliesSubject
11:33PM 18 Does X100P decode caller ID?
11:08PM 0 IP-to-IP dial: no answer or no listener?
7:41PM 0 realtime sipusers and rtcachefriends... bigheadache!!
3:25PM 1 How do I change the rtp packet size in a Cisco 7490 from 10ms to 20ms
2:55PM 2 Sample Config.
2:42PM 3 Show call coming back from Call Parking
2:36PM 1 Nobody there, continuing...
1:42PM 5 X100P - zttools says red status
1:10PM 7 Polycom Provistioning Issue
12:31PM 7 Only secretary can call the boss, all others only reach the secretary when dial the boss extension
12:29PM 0 Asterisk dropping audio
11:38AM 4 ATCOM AT 468 manuals and firmware anyone?
10:53AM 0 Asterisk on IBM NEBS compliant Blade Server
10:52AM 2 PHP AGI script callerid question
10:31AM 1 h323 compile error
10:27AM 0 TDM400P with FXS module problem
10:20AM 1 Analog FXO status checking
9:31AM 0 TDM2401 (FXO) Hangup
8:54AM 4 International Carriers
7:27AM 0 Recompiled app_xyz.so and Asterisk Dynamic Loader
7:02AM 0 Problem solved
6:56AM 0 wireless sip phone with auto answer - are there any
6:55AM 1 Ringing oddity/stupidity
6:54AM 0 Dialplan - play sample, interrupt on * and return value?
5:22AM 1 strange msg
5:22AM 1 Asterisk Recording & Volume
3:55AM 1 asterisk.conf
3:43AM 2 Hello Everybody, my problem with voicemail.conf
3:27AM 1 WellTech 380x Gateway
3:04AM 2 pickup internal and external calls
2:48AM 12 Sangoma card dying after 1hour
12:21AM 0 barge calls and record them at the same time
12:17AM 3 Zap channels staying offhook - restart required
 
Thursday January 25 2007
TimeRepliesSubject
11:36PM 0 Re: Realtime - one database driver, multiple databases
6:56PM 0 TC400B Transcoder Card Shipping
6:48PM 0 dacs support on Digium T1 equipment.
5:17PM 1 Failing to compile chan_capi
1:27PM 11 IMAP Voicemail Storage
1:27PM 9 Adding 4 more POTS lines
12:33PM 0 low audio (sometimes)
12:07PM 4 Cannot xfer parked callers
11:39AM 0 SVN trunk synchro failure
11:37AM 0 On-hold calls dropped when new call comes in
11:31AM 1 unable to create channel, in strange state, exited non-zero, etc.
10:40AM 2 Asterisk 1.4 problem with ztdummy and MeetMe()
10:16AM 1 background() with "m" option
10:11AM 1 dialplan and "*"
9:49AM 1 IAX softphone fails through PRI trunks with Hangup
8:16AM 4 TE110P and HDLC problems
7:50AM 7 NTL Hangup
7:42AM 0 Planning 48 Station Install, Need advice on several topics
7:32AM 15 1.4 - SLA
5:18AM 0 Initial DTMFs arriving too quickly?
5:05AM 5 Do I need a CH1 licence for Cisco Phones ?
4:55AM 4 Asterisk very slow when internet down
4:33AM 0 asterisk 1.4: gui registration differs from non-gui
2:29AM 4 Starting Asterisk in vvvvvvvvvvverbose mode
12:03AM 2 issue with ivtv & wctdm zaptel drivers (TDM PCI Master abort)
 
Wednesday January 24 2007
TimeRepliesSubject
11:31PM 2 SPA3K to SPA3K DTMF issue
10:38PM 1 Polycom Firmware -- Was: Asterisk 1.4 & Polycom buddy status
9:59PM 5 Multiple parking lot
6:40PM 1 Best way to connect analog modem
6:24PM 0 realtimeinsert and realtimedelete functions
2:28PM 2 Disconnected Calls
1:20PM 1 Call parking causes Asterisk to crash
1:05PM 1 Dell Server Question
12:55PM 12 setting up AMD
12:26PM 2 channel name
11:47AM 3 Semi OT - Point to Point FXO/FXS Gateway Communication
11:01AM 2 convert URI string to lowercase
10:48AM 2 OT - Cisco 7960 functionality
9:20AM 2 Getting confused on signalling mode Vs framing and encoding: T1 CAS
9:17AM 0 Agent Pre Acknowledgement Message
9:08AM 0 iax.conf setvar= like sip.conf setvar=?
9:05AM 1 iax2 prun realtime peer only can't prune user
7:38AM 5 Panasonic Hybrid Integration Advice Needed
5:55AM 1 Grandstream GXP2000 and Interception of call ?
5:38AM 1 vxml support
5:27AM 1 AOC on misdn?
3:43AM 0 NAT
3:31AM 0 beronet DTMF detection problem with some Telecom Italy lines
3:00AM 2 Digium Forums
2:42AM 0 Asterisk IAX and Shorewall QoS ?
2:38AM 1 Query Failed because: Incorrect information in file: './asterisk/sip.frm'
2:11AM 1 ChanIsAvail kills dialplan processing when no Zap available on 1.2.14.
12:44AM 0 NewTopic - Asterisk and Cisco AS5300 via E1/PRI
 
Tuesday January 23 2007
TimeRepliesSubject
7:56PM 7 No D-channels available! Using Primary channel 16 as D-channel anyway!
7:08PM 0 cmd Backgound problem with option m
5:27PM 1 Echo on IP phones...
5:03PM 1 DeStar 0.2.2 released!
5:00PM 0 Problem connecting PAP2 over wifi bridge
4:08PM 0 * 1.0.9 Voicemail record name does not playb ack in Directory() <--solved
3:56PM 1 DB_DELETE Function in 1.4
3:07PM 0 automon and MONITOR_EXEC
2:30PM 0 AW: Snom 320 echo
2:10PM 5 Snom 320 echo
1:14PM 1 OT: High Quality Wireless Headset for Cisco IPPhones and *
1:01PM 1 OT: High Quality Wireless Headset for Cisco IP Phones and *
12:08PM 3 [OT] Mark Spencer Presents AsteriskNOW on Youtube
11:29AM 0 * 1.0.9 Voicemail record name does not playback in Directory()
9:34AM 0 IGNORE: AEL parse failure on 1.2.14
9:12AM 0 AEL parse failure on 1.2.14
9:08AM 1 Rhino cards lock up system -- anyone else ever seen this?
8:07AM 12 Asterisk 1.4 & Polycom buddy status
7:47AM 4 stress-test realtime voicemail with sipp
7:36AM 1 "bad gateway" error on snom display
6:54AM 0 problems with dtmf
5:37AM 0 PRI/Q.sig between Cisco & Nortel
5:01AM 1 "No Mailbox" Prompt
4:36AM 0 beronet BRI card sometimes not detecting tones
4:36AM 24 How to exit from console?
3:28AM 6 Dial plan constructions suggestions?
3:27AM 6 Can't find asterisk.ctl under CentOS installation
2:42AM 1 Operate on registrations
12:42AM 9 weird undocumented extensions such as s-BUSY
 
Monday January 22 2007
TimeRepliesSubject
9:09PM 0 2-way MS-GSM support in Asterisk?
7:47PM 3 OT: Optimum voice problems.
7:42PM 0 Aastra 480i freezes
7:16PM 0 Why app_rx and app_tx when we have IAXModem and Hylafax and hy-email2fax? Should we reinvent the wheel?
7:13PM 0 Weird names vs. correct agent's ext.
6:56PM 1 Music on Hold on IP Phones with FreePBX 2.2.0
6:35PM 1 Fwd: Hater
5:41PM 0 7 points of comparison Polycom 430/501 and A astra 480i. Which one to choose ?
5:37PM 2 LDAP get and Asterisk 1.4
5:35PM 0 how to make a video phone call
5:01PM 2 Streaming audio file while working in background ?
3:27PM 0 CentOS and 1.4
1:55PM 1 2 ring delay before asterisk answer
1:08PM 4 X100P how do i recieve incomming calls?
12:15PM 11 7 points of comparison Polycom 430/501 and Aastra 480i. Which one to choose ?
11:32AM 1 Load Balancing
11:23AM 2 agi script as member in queue
11:07AM 0 Videopodcast about Asterisk
11:02AM 6 tdm400p not working with brazilian lines
10:56AM 0 Asterisk and VoIP @ Southern California Linux Expo (SCALE 5x)
10:53AM 0 IP of SIP server changing
10:32AM 4 Detecting Disconnected Numbers - PRI
10:28AM 1 Requirements for faxes to work properly
10:19AM 1 STUN and SNMP
10:16AM 2 No Audio for Extension to Extension
10:09AM 1 QueueMemberStatus/Status Field
10:06AM 6 Problems with rxfax
 
Sunday January 21 2007
TimeRepliesSubject
11:38AM 0 VoIP-GSM gateway problem
11:34AM 1 ISDN30 and TDM400P + FAXing ...
10:01AM 2 Backports to 1.2.14 of 1.4.0 app_queue features.
 
Saturday January 20 2007
TimeRepliesSubject
11:59PM 1 Connect a Skype adapter to TDM400P
7:44PM 0 Attention all Aastra IP phone users...
7:17PM 0 1.4 svn voicemail broken?
6:46PM 0 extra sounds description file?
4:21PM 1 func_odbc still working in trunk?
10:10AM 2 SIP registration problem w/ SBC
8:33AM 4 On what distribution is www.asterisknow.com based on ?
8:10AM 1 error message
6:01AM 6 Cisco 7970 Unprovisioned
2:27AM 1 Connecting 2 asterisk servers
12:06AM 0 CAS on Sangoma boards
 
Friday January 19 2007
TimeRepliesSubject
8:28PM 8 chanskype
7:48PM 2 Asterisk 1.4 and g723
7:01PM 2 Anyone know what this warning is about? Nothing in list history about it either..
4:03PM 1 Re: asterisk-users Digest, Vol 30, Issue 79
3:57PM 1 Incoming SIP line does not display CallerID correctly
1:53PM 1 using the Manager to connect caller to conference
1:28PM 0 Set(X=10|g) vs Set(GLOBAL(X)=10)
1:07PM 9 Ebay Unwired Buyer, Using Asterisk?
12:46PM 1 how can PRI, BRI and analog cards achieve a synchronous clock / timing
11:59AM 0 MIT Using Asterisk - VM Server
10:51AM 11 mISDN
10:06AM 0 Open Source Hosted PBX
9:49AM 10 Disconnect Supervision UK / BT solution?
8:59AM 3 Red: Sip Phone CID
8:41AM 1 Set Parameter of Call Files
8:19AM 1 Integrating asterisk with Toshiba Astrata DK380
7:38AM 2 Announce option for meetme - is it used?
7:21AM 0 CPU Bandwidth Consumption
6:10AM 0 direct transfer in features
6:01AM 0 pickup call out of menu
5:40AM 1 meetme ${DATETIME} variable update
4:15AM 2 Voice Recognition
3:33AM 0 mysterious SIP packets to Cogent
12:43AM 1 IAX2/SIP gateway for Belgium and western Europe
 
Thursday January 18 2007
TimeRepliesSubject
11:18PM 3 How to limit IAX calls
10:58PM 1 Need help with if command
10:19PM 1 Detecting open SIP channels in the dial plan
10:09PM 0 queue stats - outgoing calls
9:10PM 26 NAT solutions
7:59PM 2 Dialplan - busy and unavailable without priority jumping
3:03PM 1 COMPLETEAGENT vs. COMPLETECALLER
2:13PM 0 meetme list (unmonitored)?
1:48PM 4 Queues Question
1:37PM 2 Simplifying similiar sip trunks
11:48AM 0 Re: [cisco-voip] voice router with free gatekeeper !!!
11:15AM 2 Snom has dialtone after putting a person on hold
10:35AM 3 RE: Polycom buddies question
10:00AM 17 About BRI / ISDN hardware. What to buy?
10:00AM 0 (OT) Madge LMC 10.0
9:46AM 1 Passing video calls / bearer capability thru PRI
9:45AM 3 connecting a FXS-to-sip 4 port device to an avaya system
8:53AM 0 Re: Realtime Voicemail Password Change WORKING NOW
8:21AM 4 Sip Phone CID
7:31AM 0 Thoughts on CPE server...
7:28AM 5 1 phone 2 voicemail accounts
4:50AM 1 TDM 400P in the UK - doesn't see ringing calls hanging up before answer
4:32AM 3 Asterisk not hanging up
3:45AM 3 Bristuff with 2.6.19
3:37AM 0 changing VoiceMailMain functionality
3:35AM 6 IAX call limit
3:35AM 0 re: putting 2 SIP channels together - hangup issues
3:19AM 1 Linksys (PAP2) Registration problem
1:29AM 2 Problems with Digium TE410
12:46AM 1 sangoma a102d + Asterisk 1.2.14 ... bridging together 2 call legs on same PRI?
12:43AM 3 function call out of AGI script
 
Wednesday January 17 2007
TimeRepliesSubject
10:48PM 1 help. newbie asterisk installation problem.
4:22PM 0 Hospitals using Asterisk?
3:26PM 0 Asterisk Legacy PBX integration and fail-over question,
2:47PM 0 STUN in Asterisk 1.4
2:02PM 0 Unknown warning messages
1:35PM 3 I need to connect Asterisk to a Nortel Meridian phone plant
1:01PM 3 Network\Snom phone oddity
11:29AM 4 One way choppy sound
11:04AM 2 AbsoluteTimeout with canreinvite=yes
9:34AM 4 FW: Realtime Voicemail Password Change Not Working
9:08AM 7 Erratic Snom MWI lights
8:59AM 9 Asterisk + Unicall + Telmex E1 MFC/R2 Argentina + Meridian
8:43AM 3 Asterisk 1.4 and CDR
8:41AM 1 Dtmf tones and SIP
8:39AM 1 transfer problem
8:32AM 0 Monitor or log peer performance
8:08AM 4 Callback/ringback
7:38AM 1 dtmf problem -- second part
7:13AM 2 Asterisk registration
7:08AM 2 2 Questions: Answer with music don't work and Voicemail direct access ?
3:30AM 1 Question about FXO/FXS device.
3:27AM 0 Re: [asterisk-dev] Question about FXO/FXS device.
1:55AM 0 TDM2400 Hardware Echo Cancel (Adam Sharples)
1:36AM 0 disable external-outgoing calls per extension
12:43AM 2 Using the SIPAddHeader Application
12:15AM 0 newbie asterisk 1.4 installation problem
12:09AM 6 windows mobile 5 softphone for square screen devices
 
Tuesday January 16 2007
TimeRepliesSubject
11:08PM 0 Dell 860
10:39PM 2 TDM404B VS TDM2401B
8:14PM 4 Realtime Voicemail Password Change Not Working
7:51PM 2 Refreshing DNS lookups
6:32PM 2 Really Big Queues
6:17PM 2 prompt for "send a message" not played in VM main, HOWTO resolve
5:39PM 0 Absolute Timeout or Dial Limit option???
5:25PM 0 ERROR[2453]: chan_zap.c:8142 zt_pri_error: !! Unexpected Channel selection 3
3:45PM 2 Polycom IP601 - some hints working, not others?
2:08PM 6 Audiocodes GPL
1:41PM 0 Help with DISA
1:35PM 0 asterisk startup is slow
1:33PM 1 Ring tone too loud on IAX channel
1:32PM 0 Asterisk 1.2.14 and Audiocodes Mediant 1000
1:01PM 2 Outbound IVR for Asterisk
1:01PM 0 Asterisk Bootcamp in Pacific Northwest (Vancouver, BC)
12:56PM 6 force ulaw passthrough if call from modem extension?
10:53AM 0 MP3player distortion with Asterisk 1.4
10:27AM 1 Asterisk, SpanDSP and RXFax
10:08AM 7 How to detect long calls
8:00AM 3 IAX2 softphones can't (won't?) use PRI trunks....
7:45AM 4 IAX Trunk timing
7:07AM 0 spa942 and asterisk 1.2
6:59AM 12 TDM2400 Hardware Echo Cancel
6:42AM 0 Polycom phone locks up, send sip busy messages
5:13AM 0 Disallowing unauthorized calls to Cisco & Polycom phones
5:08AM 0 IAX Channels language
5:02AM 2 command like break ore exit in the dialpan
4:10AM 3 J1/INS1500 and the Redirect Number
3:52AM 0 ENUMLOOKUP debug
2:34AM 0 zaptel hardware detection with genzaptelconf
2:01AM 1 Didn't get a frame from channel
1:24AM 0 input request: progzone and zaptel hangup
 
Monday January 15 2007
TimeRepliesSubject
10:18PM 5 two-level administration tool for Asterisk (reposting)
6:22PM 5 Practical limit on dial prefixes for a route
5:17PM 0 S110M (FXS) Modules no longer seen on TDM400P
5:02PM 0 1-way audio
3:48PM 0 S400M (FXS) Modules no longer seen
2:45PM 4 Recording queue calls after an xfer?
2:31PM 0 Addpac 2620 don't relay DTMF to PSTN
2:28PM 9 connecting 2 asterisk servers through OpenVPN
2:00PM 3 Audiocodes Mediant 1000, Polycom, and no ringback on transfer
12:56PM 4 Queue cmd option 'i'
12:38PM 1 Asterisk PBX '&' '||' Grandstream GXP-2000 problem
12:34PM 0 help create asterisk cookbook
12:23PM 4 Nufone
12:22PM 6 Delay in Call Distribution using the Queue Application
11:36AM 3 Software callcenter
11:08AM 15 Queue and Interface time out
10:55AM 1 I have to register asterisk/sip with a sipproxy that does not support authentication?
10:51AM 1 TDM400P, fxotune and ADSL filters - Just a FYI, FWIW
9:58AM 0 .call files - no hangup
9:53AM 7 php agi - first phrase truncated, all others fine
9:25AM 2 ANY ADVICE ON THIS????
7:47AM 0 SIP transfer issue
7:43AM 1 what happened to "sip list peers"
7:23AM 0 Parked calls with Asterisk 1.4.0
5:37AM 1 Wanpipe 2.3.4-2 + kernel 2.6.19 = problems
3:52AM 2 Installing Asterisk 1.4 Documentation
3:50AM 4 Rt db lookup
3:47AM 0 OT: Quad-band cellphones with wifi & stablesipsupport
3:38AM 1 phpagi transfer example
3:22AM 0 Asterisk Realtime and MD5 authentication
 
Sunday January 14 2007
TimeRepliesSubject
11:01PM 4 OT: Quad-band cellphones with wifi & stable sip support
5:40PM 1 E&M ?
4:01PM 6 To 1.4 or not
3:54PM 1 Asterisk not hanging up calls
12:08PM 0 realtime mysql db performance difference with matching extensions
11:58AM 1 Problems with mISDN TE line
9:57AM 1 RE polycom fails registration
8:41AM 4 RE : TDM2400p bad sound quality
4:25AM 2 functions - fork
1:23AM 2 Particular DialPlan
1:08AM 0 External resource timeout
1:06AM 0 Ring PC speaker
12:54AM 5 Polycom registration fails
 
Saturday January 13 2007
TimeRepliesSubject
5:48PM 1 Caching Caller Name
1:51PM 3 fxotune Error
12:23PM 0 Hints on FXO channels
9:30AM 1 Possibleto use zaptel 1.4 with asterisk 1.2?
7:26AM 1 1.4 and "sip list peers"
1:13AM 0 AMAFlags always"Documentation" (or 3 inastcdrmysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill)
 
Friday January 12 2007
TimeRepliesSubject
7:23PM 2 Dropped calls
5:51PM 1 SIP phones at multiple locations
2:47PM 2 cepstral voice still nags after registration
2:21PM 0 GXP-2000 Firmware
1:53PM 1 phones can make outgoing calls but no incoming
1:52PM 26 Nat Question
1:43PM 0 Musiconhold and dial
11:38AM 4 5v capable motherboards
11:19AM 4 two level administration tool for Asterisk
11:15AM 1 TDM2400p bad sound quality
11:11AM 0 R: asterisk-users Digest, Vol 30, Issue 50
9:39AM 0 realtime queues
9:31AM 4 Voxbone Question
9:27AM 5 One of my incomming lines is busy yet there is no indication in FOP.
9:23AM 1 FW: Redundancy
9:23AM 0 Provisioning
9:23AM 1 External Ringers for Cisco Phones
8:24AM 1 How to detect which end hung up the call
8:04AM 1 Identifying Queue on Cisco 7960
8:03AM 11 SPA 3000 won't relay DTMF to doorphone
7:24AM 2 SLA
7:01AM 2 realtime extensions, labels
3:12AM 5 FW: Get dialed numbers in AGI
1:29AM 1 Not Registering Port with VSP.
12:47AM 1 boot up problem
12:06AM 4 Snom Record / Voice Recorder Button
 
Thursday January 11 2007
TimeRepliesSubject
6:28PM 2 Question Regarding Visual Park Functionality - Hardware/Software
6:23PM 20 Echo...
5:00PM 5 Problem with Zyxel P-2000W v2 not receiving calls
4:20PM 10 DND - message
3:40PM 0 Trying to understand mISDN kernel buffer messages...
3:33PM 3 Queues without music on hold ?
3:08PM 2 Asterisk Compilation and Installation
3:06PM 3 Symbolic Link
2:52PM 4 Sipgate displayes on web interface status Offline
2:36PM 0 Re: Digium TE407P vs. Sangoma A104d
2:08PM 6 Voicemail IMAP
2:05PM 0 Getting two Asterisk machines to talk to each other
1:46PM 3 Read Voicmail Boxes
1:41PM 0 "Dropping Incompatible Voice Frame"
1:28PM 10 Parked calls and the # key
11:57AM 2 Restrict International Calls
11:07AM 1 Queues Service Level
10:17AM 2 authentication issue!
10:12AM 0 Queue log analyser / report
9:55AM 2 Native music on hold not playing on incoming calls
9:41AM 9 Suggestion for a new asterisk setup.
9:23AM 0 What would make Asterisk Ignore INVITES?
9:12AM 1 realtime sipusers and rtcachefriends... big headache!!
8:53AM 6 Asterisk Manager Interface: Auto-answer of 'Originate' command
7:54AM 4 play music while continue executing dial plan
7:46AM 0 Need help getting anterisk application to load
6:52AM 9 "real life" example of SLA definition
4:37AM 2 Has been working for 9 Months - Very Very Strange I cannot dial specific extensions from my dialplan - NOT A CONTEXT PROBLEM!!
4:10AM 1 DTMF signaling after Dial
3:15AM 1 Stuck somewhere - INVITEs ignored?!
3:12AM 2 Problems with agent dynamic login
2:20AM 2 calls to SPA942 disconnect after 15 seconds (chan_sip.c set_destination: can't find address)
 
Wednesday January 10 2007
TimeRepliesSubject
6:06PM 0 Extensions in a macro
5:52PM 1 Priority + 101 exit
5:20PM 0 Festival Problems
2:28PM 1 Round Robin Queue
2:22PM 0 generating SIP errors
12:53PM 8 Possibility to catch DTMF when 2 users are in a conversation
12:17PM 0 SIP invite and sip.conf relationship?
12:15PM 4 Random dropped calls...
10:41AM 1 dundi ENCREJ
10:34AM 1 VIA EPIA DeadLock Issues
10:05AM 5 Proper use of the Local channel
9:53AM 0 app_system.c:105 system_exec_helper: Unable to execute '/sbin/zapscan.bin'
9:39AM 0 1.4 and zap bugs
9:18AM 1 Service Level Compliance
9:10AM 18 Directory too difficult?
8:59AM 23 Get dialed numbers in AGI
8:57AM 2 RTP directly
8:46AM 4 Send email notification
8:37AM 5 how to realize chief - secretary (or Manager - Assistant) setup with Asterisk?
8:36AM 2 Asterisk HA
8:30AM 0 SPA-3000 and Asterisk 1.4.0
7:01AM 1 Zap 1.4 error line 0: Unable to open
6:34AM 0 Calls die when the answering party transfers
6:02AM 0 libpri Calling Line ID
3:33AM 1 Sip dynamic host question
3:33AM 0 DTMF on Snom
3:10AM 0 one way audio when forwarding from ser to asterisk
2:52AM 2 Which H323 module for asterisk
2:06AM 3 caller id not transferred to SIP device
1:25AM 1 Redundancy
12:25AM 0 cannot call out
 
Tuesday January 9 2007
TimeRepliesSubject
11:41PM 1 ztmonitor output while idle
6:01PM 8 Problem with zaptel drivers or card
5:26PM 2 Attatching VM via email for more than one user
4:51PM 4 Zap 1.4 error line 0: Unable to open master device '/dev/zap/ctl'
3:28PM 1 MINNESOTA: TwinCities Asterisk Users Group - Saturday January 13th 2007 - 11:30am
3:12PM 11 Is there a low cost cell phone base station for asterisk ?
2:55PM 2 Asterisk 1.2.11 - ResponseTimeout being ignored
12:22PM 1 How to test VOIP quality?
12:17PM 0 Console\DSP
11:57AM 0 VOIP provider reliability
11:27AM 1 getting tones during conversation
11:16AM 3 Caller Id problem
10:44AM 2 ooh323c calls
10:30AM 0 Is there any Asterisk controllable thermostat?
9:41AM 4 Asterisk and Avaya IP Office
9:04AM 16 Asterisk build for Suse 10.1
8:52AM 3 Fax through Sangoma A102
8:29AM 0 Strange queue behaviour
7:50AM 8 Snom side car annoyance
6:34AM 0 Asterisk + 7910 + Skinny Reset
2:21AM 4 Asterisk and 3PCC
2:04AM 1 Problem with polycom video conference
1:59AM 0 Bad FCS & hangup
12:27AM 1 Record of all calls
 
Monday January 8 2007
TimeRepliesSubject
9:34PM 0 Allowing inbound VoIP Calls from VSP
9:08PM 1 No CDR from Outbound Call
4:17PM 3 Call Sound Volume Low : between extensions and over ZAP.
2:44PM 0 SIP rt load from db
2:19PM 0 snom 190 (etc.?) dialscript for * debugging and kaddressbook
2:17PM 5 delete=yes is not working
11:32AM 0 MixMonitor write issue
11:22AM 4 OT:spa942 provisioning
11:13AM 0 Strange error
9:58AM 3 G729 license counting
9:47AM 1 Realtime Voicemail Table Column Name Question
9:17AM 2 ARA extensions ordering
8:59AM 11 Adding 4000 Lines to asteriskdb via asterisk -rx ?
8:09AM 1 Block some number outgoing from joust oneextention
6:21AM 0 IAX call path optimization with more than 3 legs
5:24AM 5 SV: Manage 'full' log file
5:02AM 2 Manage 'full' log file
4:07AM 6 MFC/R2 problems
12:58AM 0 Goto not jumping to current context
12:27AM 3 jitterbuffer on sip.conf
 
Sunday January 7 2007
TimeRepliesSubject
11:28PM 1 Interrupt rates and voip traffic
10:07PM 3 Problems with park
10:06PM 2 AMAFlags always"Documentation" (or 3 in astcdr mysql) even after Set(CDR(amaflags)=bill) or SetAMAFlags(bill)
9:29PM 2 snom 360 auto answer
8:27PM 53 Some queries on g729 license.
8:03PM 0 How to transfer Voicemail messages between 2 Asterisk servers
6:32PM 0 Re: asterisk-users Digest, Vol 30, Issue 7
5:12PM 3 Hanging up a 3-way conference when middle user hangs up
6:14AM 0 LUSYN patches
12:38AM 1 Scalable IVR with asterisk
12:19AM 5 How to get dial tone back
12:02AM 3 "Reserved" extensions?
 
Saturday January 6 2007
TimeRepliesSubject
9:46PM 2 Question about AGI and variable storage
8:19PM 1 Parking a call a second time using #700..
7:54PM 0 Shared Line Appearances in 1.4
4:08PM 5 SIP/RTP Nat problem, can't solute it.
2:05AM 0 SIP Reinvites
1:50AM 0 Hint and call-limit issue
1:36AM 1 SIP trunk to a Boscom/Claro/IP Gear Robocom
1:13AM 5 Handling SIP 482 condition
12:44AM 2 Secure a Asterisk Server ?
 
Friday January 5 2007
TimeRepliesSubject
11:04PM 0 Asterisk is used in U.S. prisons?
10:13PM 0 asterisk 1.4.0 didn't compile chan_zap.so
4:16PM 1 .call files no longer generating CDR files
4:13PM 3 Call waiting notification
3:42PM 1 Voicemail personalised greetings using DB/IMAPbackend?
3:28PM 1 DiD for less then $4
3:23PM 0 Random "unknown" codec format IAX calls
3:17PM 9 Voicemail personalised greetings using DB/IMAP backend?
2:05PM 1 Multiple users and a single extension
1:54PM 0 Has anybody voipstunt working?
1:15PM 1 asterisk 1.4 debian packages
12:22PM 7 SIP/TCP?
11:13AM 1 asterisk (FreePBX) and queues
10:35AM 1 ASterisk OOH323c
10:31AM 3 how to register nokia with Asterisk
10:02AM 7 how to transfer calls when analog phone has no transfer button
9:44AM 1 addons 1.4 and cdr_addon_mysql not installed !
9:06AM 0 idle SIP channels problem
7:19AM 1 radius
5:57AM 1 integrating with Asterisk and OpenSER for Voicemail
5:47AM 5 Asterisk and IM
5:37AM 6 fax transmission
5:25AM 0 Invalid DivertingLegInformation2 component received 0x38
4:40AM 0 How to build 1.4 with res_crypto.so
1:48AM 1 Which g729 module for HP DL 360 G3 (Xeon CPU's)?
1:17AM 2 chan_zap.c: Failed to read gains: Invalidargument
12:47AM 3 chan_zap.c: Failed to read gains: Invalid argument
 
Thursday January 4 2007
TimeRepliesSubject
11:51PM 6 Which is GUI to edit Asterisk IVR logic
10:33PM 5 POE draw on Aastra 480i
8:33PM 0 3-way calling MGCP capture
7:23PM 7 IAX vs SIP trunks between Asterisk boxes
7:01PM 4 MusicOnHold Files
6:59PM 0 DISA Ring Back
6:48PM 6 HowTO configure voice T1
4:53PM 8 Dimensioning a 50 sip phone installation
3:30PM 0 How to routing call to Quintum.
1:00PM 0 proxy howto
12:18PM 0 Re: Alert: Steering Committee Reminder and Agenda
12:08PM 3 How big a pipe can IAX2 go?
11:42AM 2 Trouble compiling asterisk 1.2.14
11:13AM 1 asterisk sip peer/user matching methodsforauthentication backwards?
10:57AM 31 Best inexpensive home office router for VoIP (QoS with maybe PoE)
10:43AM 0 SIP peer lookup problems
10:26AM 1 #include not working in 1.4
10:22AM 0 asterisk sip peer/user matching methods forauthentication backwards?
10:17AM 0 Asterisk 1.4.0 segfault
10:08AM 1 Convert a file from WAV to WAV49 or GSM for Asterisk
10:04AM 0 Sangoma A102 w/ EC module gets intermittent echo/audio artifacts <--followup and resolution
9:34AM 3 [Fwd: PRI Problems]
9:27AM 0 System() and Trysystem() in extensions.conf => get the result ?
9:26AM 0 PRI Problems
9:06AM 0 mISDN crypto?
8:46AM 1 Realtime voicemail passwords
8:26AM 0 Create a group of SIP acoount for outgoing calls ?
6:26AM 6 postgres and asterisk
6:18AM 4 Digium Wildcard B410P
5:02AM 4 Maybe a NAT problem
3:23AM 2 Hi reg. asterisk Compilation
3:14AM 6 bypass menu for certain numbers?
2:12AM 4 Cisco AS5300
1:58AM 0 Required freelancer for installing hylafax on Asterisk Box
 
Wednesday January 3 2007
TimeRepliesSubject
10:50PM 0 asterisk sip peer/user matching methods for authentication backwards?
9:15PM 4 caller id ring tones for Asterisk Phone
6:38PM 0 Re: asterisk-users Digest, Vol 30, Issue 4
6:07PM 4 ztdummy on 1.6
5:41PM 0 1.4 segfaulting when manager client is connected
5:17PM 0 [Announce] Web-MeetMe 3.0.0 RE-released
5:13PM 0 v140 ./configure not finding installed ssl
3:51PM 14 Any quiet 24 port POE switches out there?
3:26PM 4 Asterisk Core Dump in app_queue - Anyone seen?
3:22PM 7 Detect IP path before calling
2:29PM 2 Gentoo ebuild for 1.4?
2:27PM 0 ARI help
1:31PM 0 have a phone number from stanaphone and a workingtrixbox, h
1:22PM 16 over 200 queues, anyone?
12:39PM 1 have a phone number from stanaphone and a working trixbox, how do I connect them?
12:24PM 3 Error on answer a SIP 401 message
12:23PM 0 Cisco 79x1 Auto-Answer
11:35AM 0 Park and Page
11:32AM 15 [Announce] Web-MeetMe 3.0.0 released
11:16AM 2 Is chan_zap.so loaded?
11:05AM 8 API: how to bridge originated call?
10:48AM 7 Polycom Power Specs
10:40AM 2 SIP Dial out timeout
10:26AM 5 answer machine detection
10:11AM 0 Sangoma A102 w/ EC module gets intermittent echo/audio artifacts
9:29AM 0 [BULK] Fonebridge2
9:25AM 0 Sangoma A102 w/ EC module gets intermittent echo /audio artifacts <---More information
9:05AM 3 Voicemail to email
8:33AM 1 Fonebridge2
8:32AM 2 Sangoma A102 w/ EC module gets intermittent echo /audio artifacts
8:04AM 0 MeetMe() not recording calls
6:43AM 8 Sangoma Remora A202
6:40AM 0 native music on hold distortion between files
6:11AM 5 voice fax modem and asterisk
5:20AM 0 ISA server Issue (Maybe off topic)
4:17AM 0 Dubai Caller ID
4:00AM 8 SNOM loses server registration
 
Tuesday January 2 2007
TimeRepliesSubject
8:15PM 6 connecting asterisk (trixbox) to traditional phone lines?
5:22PM 2 Double quotes in CDRUserField?
5:12PM 1 extension problems
5:01PM 29 OnHook Call Announcement...
3:17PM 1 OT: Admin manual for Linksys Sipura SPA-2102
2:57PM 3 queues - limiting ringing calls to queue members
1:17PM 6 Call connected, cannot hear or speak - $20 for fix
12:45PM 6 yet another faxing issue (outbound only, via ATA)
12:18PM 0 SpanDSP and Asterisk 1.4
10:20AM 2 How to show a debugging remark in a sip or extensions context?
9:00AM 6 802.1x support in wired sip hardphones ?
8:55AM 0 [asterisk-biz] Slightly updated UK English voice prompts
6:48AM 0 Save SIP DEBUG output to a file
6:20AM 14 Best Hardware for Asterisk Server?
6:20AM 0 Avoiding deadlock-line drop problem
2:57AM 4 asterisk and mysql
2:52AM 1 chan_oh323 early media
 
Monday January 1 2007
TimeRepliesSubject
11:56PM 3 Problem with centos 4.4 and jabber/gtalk (really iksemel)
11:54AM 1 Help needed with Polycom dialplan pattern matching
7:36AM 0 Thomson ST2020 and voicemail