asterisk users - Feb 2007

Wednesday February 28 2007
9:17PM 1 voicemail advanced options problem with mysql datbase
6:26PM 0 Using ooh323 with Gatekeeper controlled dialling
5:14PM 4 Help Needed: Can't make "local" calls on a brand new PRI
4:45PM 0 ooh323 patch: fix for Cisco IOS Gatekeeper re-registration problem
4:06PM 1 AEL & Blacklist question
3:45PM 1 Paid support offered
3:43PM 3 read write or only read fields in cdr?
3:33PM 1 OT: Patch to OSS app for CDP without a Cisco switch - TESTERS WANTED
3:32PM 1 extensions.conf & sccp.conf howto call external number
3:18PM 2 Newbie extensions.conf question
3:03PM 2 No Caller ID Name PRI NI2
2:15PM 0 Occasional SMS problem
1:57PM 0 Send DTMF's before the call is answered
1:56PM 1 1.4 lost internet internal phones loose registration
1:45PM 3 Newbie Planning Help
1:33PM 4 Help: CallerID Name not being sent on outbound PRI trunk
1:23PM 2 this i a test
11:56AM 1 Run-away Asterisk
10:17AM 2 Changing from email address for vociemail.conf
9:55AM 0 seeing DTMF passed to Voicemail
9:54AM 3 Registrations, how many is too many?
9:31AM 5 about bluetooth channel
9:18AM 3 multiple phones registered for the same user
9:08AM 1 h323 how to set it up?
9:00AM 1 Timing, use analog card, ZT Dummy etc.
5:05AM 1 groups
3:22AM 7 Problem with TE212P
3:12AM 0 Asterisk 1.4 does not load
Tuesday February 27 2007
7:46PM 0 Limiting call volume
7:15PM 1 Help understanding SIP SHOW CHANNELS
7:05PM 2 No sound with Playback() or Background()
6:34PM 2 Voice mail is not giving unavailable or busy prompts
6:08PM 1 Not registering Port with VSP
6:02PM 5 TE110P: Error ==> Asterisk died with code 1.
4:12PM 2 jittery audio in voiceprompts
3:58PM 1 Quintum configuration ASM200 Analog 2 tenor port
3:18PM 2 Saving Dialplan in CLI
2:13PM 1 SER / IAX solution
1:26PM 2 TE212P on FC6 - stack overflow?
1:23PM 0 asterisk CDR and mysql
11:01AM 1 Net-talk
10:50AM 1 H323-to-SIP proxy
10:37AM 0 rtc: lost some interrupts at 1024Hz
9:59AM 2 Polycom Firmware
9:54AM 2 running asterisk through cellphone
9:40AM 0 sip.conf "limitonpeers=yes" in asterisk 1.4
8:54AM 0 Grandstream SYSLOG error codes
8:44AM 1 Billing Telephone Number (BTN)
8:37AM 0 call-limit in 1.2 HEAD
8:06AM 1 Do I understand GROUPs correctly?
7:23AM 2 RES: asterisk-users Digest, Vol 31, Issue 115
6:33AM 1 NetFilter (IPTables)
6:23AM 1 VLAN vs RealLan
6:00AM 2 Authentication Command
5:31AM 0 OutBound Proxy calls failing
5:12AM 1 FW: Cisco 7960
5:07AM 0 Forbidden - wrong password on authentication for INVITE
5:02AM 0 Autentication
3:16AM 0 mgcp codec problem about ulaw
3:03AM 1 Cisco 7960
2:52AM 1 Error Message.
2:33AM 2 a BUG?
12:36AM 1 chan_sip.c:10173 handle_response: Dont know how to handle a 202 Accepted respons
Monday February 26 2007
10:11PM 1 To use asterisk or proprietary hardware, that is the questio
9:06PM 7 How to get values of local channels context
8:01PM 2 XM Radio Stream to Asterisk
7:23PM 3 Asterisk -> Streaming Audio Bridge
6:20PM 3 Playback uses channel's language, background doesn't
5:20PM 1 Asterisk to Asterisk SIP Trunk and CallerID
5:03PM 1 Digium S101I echo - how to control it
2:31PM 1 Caller ID not getting to analog extensions
2:06PM 1 deprecated - CLI help vs. source code
11:48AM 3 Yellow or Red alarm on TE110P ????
11:45AM 2 Ex-Girlfriend syntax and RealTime Extensions
8:31AM 0 AstriCon Europe 2007
7:42AM 3 How set CallerID via Macro or something
7:15AM 0 Out Proxy Call
4:24AM 0 IAX/SIP Inter Asterisk Transfer
3:58AM 2 SetCIDNum is not available on 1.4svn
2:02AM 0 Asterisk TE110P Hipath 3750
12:47AM 1 Newbie would like some planning advice.
Sunday February 25 2007
7:20PM 7 Sending Email From the dialplan
1:32PM 1 Marks SNMP HowTo
1:08PM 2 Dialling ZAP channel from analogue
10:28AM 0 Looking for automatic sound announce device
10:09AM 2 - has anybody tried it?
6:52AM 0 VoiceMailMain plays oldest message first
Saturday February 24 2007
8:22PM 1 ERROR: relation "cc_ui_authen" does not exist
8:04PM 0 1.4.0 spews garbage on CLI, crashes
7:35PM 5 Sending SMS
5:29PM 0 Wildcard Testing
5:12PM 0 Voicemeup @ 0.008 per minute USA /CAN
10:53AM 8 To use asterisk or proprietary hardware, that is the question
8:05AM 6 dial a pager and enter DTMF
7:22AM 0 Analogue Phone Problems on a TDM11B
3:31AM 1 Somebody can help me?
1:44AM 0 Call was hangup when LIMIT_WARNING_FILE was playing
Friday February 23 2007
11:53PM 1 Accessible documentation vor blind users
10:16PM 2 Any way to get rid of AEL created contexts?
7:14PM 1 ReceiveText()?
5:20PM 0 Asterisk callshops
4:23PM 1 H extension don't work with parked calls
4:17PM 0 New Community Blogs
3:14PM 0 MusiconHold
3:14PM 2 Voice mail server
2:48PM 2 GSM cleanup (pops, clicks and static)
1:44PM 1 asterisk
1:11PM 1 Queue Macro Problem
1:00PM 0 SOLVED: Call forwarding and 1.2.x
12:57PM 3 cisco sip firmware update for cisco 7970
12:05PM 2 SIP Test
9:55AM 1 Polycom SIP 501 Transfer Question
9:21AM 1 SLA more than 100% ?
9:07AM 1 ooh323 hang up after the call is answered
6:06AM 2 Dial() command h and H options for SIP channel
4:42AM 1 CWI, call-limit and incominglimit
3:39AM 1 Asterisk and DTMF
3:29AM 0 Job offer near Los Angeles
2:44AM 0 Have an AGI script as a queue member
1:52AM 1 peer-to-peer RTP trouble in SIP
1:13AM 3 Sellvoip configuration....Please Help!!!!
12:41AM 1 default "insecure" setting
Thursday February 22 2007
10:52PM 1 "Trunk" version of Asterisk?
10:10PM 0 choppy playback
6:27PM 0 Application RealTime
4:24PM 0 Strange Noise
3:13PM 3 Argentine Asterisk Wiki
2:32PM 1 asterisk with TCP transport
2:23PM 4 Possible to light up a LED on Snom phones?
2:20PM 3 upgrading from A101 to....A102
2:08PM 0 Passing call status/progress between protocols
12:57PM 2 AG-188
9:30AM 3 New tutorial: DTMF tone detection
8:46AM 0 Asterisk - VoiceGenie IVR
8:38AM 1 Lastest SVN (1.4) and realtime call limit
7:47AM 1 Asternic Flash Panel
7:39AM 2 Configuring Asterisk.
6:58AM 0 SIP RE-INVITE after an Answer()
6:48AM 0 Polycom IP 601 help needed
6:40AM 0 RE: Asterisk to Cisco's Rescue...again...AuthenticateLD Calls
5:58AM 6 Asterisk and Cisco PRI gateway config
5:22AM 2 What means: Request to schedule in the past?!?!
5:15AM 0 Newbie: registration failure (fwd)
5:02AM 1 GotoIf DURATION
4:49AM 1 VoIP Internet Server
4:28AM 3 An ISDN ISPBX to Voip Gateway??
3:36AM 2 b410p + fax (echo cancellation)
1:32AM 2 fax support
1:23AM 1 Answer() command?
1:22AM 0 cannot get whole DNID with ISDN line
1:20AM 3 queue information into db
1:00AM 0 Destroy a zombie sip channel
Wednesday February 21 2007
8:43PM 2 SIP response 603 driving me nuts
6:01PM 1 Asterisk to Cisco's Rescue...again...Authenticate LD Calls
3:52PM 3 Snom 320 password
3:05PM 3 SIP 406 error - cause?
2:27PM 0 monitoring cluster-based call-centers
2:07PM 1 How to separate outgoing extens from the contexts from sip.conf?
1:58PM 2 How does Asterisk use SIP info command
1:19PM 0 Problem on Asterisk to Register lines for out/in calls
11:45AM 1 Problem Installing Zaptel
11:26AM 3 Trixbox -- ACPI and IO-APIC?
11:16AM 1 Monitoring which users are online in realtime
10:17AM 0 Zaptel 1.2.14 Released
9:16AM 0 Trixbox ;TE110P ;DELL OPTIPLEX GX240
9:01AM 1 HELP!! Dropping calls on Bridge
8:52AM 3 Zaptel 1.4.0
8:23AM 0 Using asterisk with vpb driver (OpenLine4)
8:04AM 0 jingle + asterisk 1.4
6:44AM 0 Trunk - strange behavior
6:26AM 0 IAX Realtime - show peers works?
5:50AM 0 how to detect who starts one touch recording
4:54AM 1 Channels hanging when SIP phone gets reset during call
4:39AM 0 Dialout option problem in voicemail.conf
4:12AM 1 AGI DTMF Problem
2:12AM 0 How to read channel occupation from PRI INTENSE DEBUG ?
1:48AM 0 Hint a sip account
12:05AM 0 How to read "pri intense debug span" data ?
Tuesday February 20 2007
11:56PM 1 How to repeat pri show span and zap show channel commands
11:49PM 0 Open Source VOIP at Toronto Conference
11:40PM 2 Help! How to get ANSWEREDTIME after DIAL a ZAP channel?
9:33PM 0 Asterisk behind OpenSER - Getting SIP reinvites to work with an ITSP
8:43PM 1 trixbox not sending ring back to caller
8:22PM 2 Asterisk CDR MySQL
6:05PM 6 They ignore my DTMF!
3:10PM 4 Passing a variable from one Asterisk box to another
2:25PM 0 Can't get ANSWEREDTIME after hangup using ZAP
2:02PM 0 Tipping Point IPS blocking Asterisk SIP quaility messages
1:35PM 1 CDR reports short call length
1:18PM 2 Rules about congestion
11:13AM 0 B410P - Please an advise
10:50AM 3 analog channels calling out not detect DTMF
10:22AM 3 Asterisk / ACT CRM Integration
7:04AM 0 unwanted chanspy: strange behaviour
5:15AM 1 Asterisk-1.2.10 not releasing SIP sessions
4:01AM 6 FW: zaptel 1.4.0 on Fedora Core 6 x86_64
3:37AM 1 Agents busy in queue
3:08AM 2 Mask the caller-ID
2:39AM 0 problem with dialout option in voicemail.conf
1:24AM 0 Communication between servers
Monday February 19 2007
9:47PM 2 sip to sip ?
8:50PM 1 zaptel 1.4 svn fails to compile (xbus-core.c)
6:11PM 1 SIP interface status and calllimit
5:05PM 2 Auto load of zap drivers
2:37PM 1 Kernel and zaptel versions
1:00PM 1 Setting Caller-ID / Point Codes
12:45PM 1 Good 4 Port PSTN Gateway
12:39PM 6 Open CallerID Database?
12:32PM 2 UTStarcom F1000 - WLAN connection unreliable
12:23PM 0 Cisco 7941/7961 w/Firmware 8.2.1 and NAT
10:26AM 1 Attended Transfer with snom phones
9:11AM 0 Re: asterisk-users Digest, Vol 31, Issue 80
8:41AM 0 SIP resigtrations and OpenSer
7:55AM 2 Transfer Caller ID
6:53AM 1 Asterisk with Radius users authentication
5:37AM 1 Asterisk Inbound Problem
4:18AM 1 Asterisk PPPD with analog lines
4:03AM 0 Asterisk and a modem pool.
3:14AM 1 Problems with CentOS ztdummy kernel 2.6
Sunday February 18 2007
1:55PM 1 HT488 doesn't disconnect FXO
12:08PM 5 Looking for starting point?
9:37AM 0 Asterisk consultant needed in Paris
5:09AM 3 chan_sip.c:1968 create_addr: No such host:
Saturday February 17 2007
7:09PM 1 Unable to start Asterisk 1.4 on CentOS 4.4 (installed from ATrpms)
3:01PM 1 Confederated SIP service.
1:15PM 0 dialing out with TDM2402E card other system not grabbing DTMF digits
9:22AM 1 Weird problem in wctdm24xxp driver
9:15AM 0 Fwd: musiconhold.conf in realtime
8:17AM 0 musiconhold.conf in realtime
12:46AM 3 Problem with busydetect and cell phones
Friday February 16 2007
10:41PM 0 manager command queue...
8:28PM 0 IAX vs SIP - Getting Asterisk out of the media path
6:59PM 1 X100P ring detection failure
4:11PM 0 F1000 web configure
4:04PM 0 How to configure Asterisk queue with Vonage account?
3:02PM 3 Does Asterisk support DNIS?
2:36PM 0 Help needed to server code on Vxworks
2:06PM 1 DNIS on T1 channels
1:31PM 1 iaxmodem - fax tone?
1:20PM 1 Open Source VoIP at FOSDEM
12:23PM 0 How can I use 'Asterisk Manager API' to hold and retrive an active call?
12:06PM 2 Experiences with FoneBridge2 / TDMoE?
12:01PM 1 Sangoma A101 install problem
11:26AM 0 sangoma 102 and CAB-E1-RJ45BNC
10:56AM 1 Distinct call permissions for each user
10:31AM 1 MixMonitor & RingBack Tone Issue
9:50AM 5 FW: Problem Transferring Direct to Voicemail
7:13AM 7 Summary of "Trixbox vs. custom install"
5:55AM 2 Jabber/Asterisk Integration
5:28AM 2 Asterisk callerID
4:23AM 2 freepbx with ASTERISK 1.4
3:37AM 1 64 bit HPEC modules available?
1:23AM 2 Pickup application
12:43AM 0 AstLinux + RT PREEMPT
Thursday February 15 2007
10:48PM 0 SIP/2.0 404 Not Found
10:39PM 6 PSTN Calls from SI.P: buzzing and pops
9:14PM 0 SIP Redirect from Asterisk behind a NAT
3:34PM 0 New AstLinux Branch: RT PREEMPT ("realtime" Linux) - Looking for testers
2:42PM 1 Meetme - is this statement from the Wiki still true?
2:19PM 2 Native format prompts
2:13PM 0 Guest registration in SIP
2:08PM 2 Asterisk Queues Problem
12:45PM 4 Long call setup times on SIP to zaptel calls
12:28PM 0 Asterisk guru wanted, SoCal (LA/OC/San Bernardino County)
12:19PM 0 h323 - SIP conversion
11:57AM 2 7912 phones loosing registration
10:25AM 6 Connect PBX CO Port to TDM FXS Port
9:49AM 0 addons 1.4 and cdr_addon_mysql not installed !
9:44AM 3 Maximum Number of Calls Asterisk Can Handle
9:19AM 0 Pause a Audio File Problem
8:48AM 0 No Ringback, only on 1 SIP provider
8:12AM 3 OT - IP Network Call Recording
8:11AM 1 Hint and CallerID
7:52AM 1 Feeding digit input to PauseQueueMember
7:01AM 1 Multi-calendar Overlay Layers?
6:57AM 3 asterisk freeze due to "too many open file" error
6:04AM 7 Call forwarding
5:29AM 1 Interruptible announcements in queue application
2:40AM 1 Symbian IAX client
1:50AM 0 directed call pickup with PICKUPMARK
Wednesday February 14 2007
11:12PM 4 Best FXO Gateway
7:13PM 1 [Fxo] Digium TDM01B vs. OpenVox A400P01?
5:58PM 0 Requested contexts didn't get merged
5:30PM 1 Strange behaviour with Dial cmd
5:10PM 0 custom sip header
4:40PM 0 Useragent List
4:18PM 1 Need info for creating * as a gateway for other * servers.
3:55PM 2 moving WiFi phone
1:07PM 3 S101I (IAX) limitation
1:00PM 2 Asterisk vendors in Houston, TX
12:36PM 2 Macro Usage
11:45AM 1 Limit on SIP phones on one server
11:36AM 0 libunicall + hashtable.c + asterisk crash
11:31AM 1 zaptel 1.4 svn doesn't compile
11:28AM 0 Connect a legacy PBX to an Asterisk Server
10:43AM 0 Zoiper softphone version 1.03 now available
10:25AM 4 Guide to better performance using * ?
10:09AM 2 Fanless solution
9:49AM 0 Realtime via ODBC breaks for Voicemail
9:42AM 2 "Unable to launch Sendmail" warning
8:21AM 2 Compiling Zaptel-1.2.13 on FC3
8:05AM 1 CAS signalling on span 2 conflicts with HDLC with FCS check on channel 20
7:58AM 0 Please some advice on setting hook-flash timing on Linksys PAP2
7:28AM 2 Problem Transferring Direct to Voicemail
7:23AM 2 MRTG with 4 graphs
7:19AM 5 Bandwidth shapping device
5:46AM 1 To jitter buffer or not to jitter buffer?
5:18AM 2 SIP response 482 "Loop Detected"
3:42AM 0 asterisk 1.4 is stable?
3:40AM 0 Asterisk & CME integration using h323
3:38AM 6 Fax with T.38
3:27AM 6 genzaptool from "xorcom"
2:10AM 2 Can anyone help me out with Polycom 2.1 firmware please?
1:46AM 1 Following call forwards
1:16AM 1 Asterisk 1.4 and chan_misdn
12:33AM 3 GSM Gateway promotion from £69GBP
Tuesday February 13 2007
8:45PM 2 E911 SIP or IAX providers?
6:04PM 3 Sending SMS from Asterisk
4:18PM 3 SMS via VoIP and web
2:41PM 1 FRITZ!Box Fon ata
2:22PM 1 PRI Call Start
2:13PM 3 AgentCallBackLogin vs AddQueueMember
1:30PM 0 Compiling Asterisk With ZapTel?
1:27PM 0 Your favorite switchboard application software ?
1:00PM 0 Asterisk 1.4.0 and callwaiting eventually drops call
12:45PM 1 How can I use "Asterisk Manager API" to hold and retrive an active call?
12:06PM 0 Blocking collect calls in Brazil
10:18AM 0 problems with trunks IAX2 and queues
9:16AM 4 Nortel 81C MSDL Trunking to Asterisk TE110P, Nortel Resetting PRI Channels
9:14AM 1 AGI "GET DATA" and "WAIT FOR DIGIT" don't work
8:38AM 1 Paging Followup
7:31AM 1 Using Dynamic Groups instead of AgentCallbackLogin - how to log which agent took the call?
6:24AM 1 Using Asterisk/callerid with "pay as you go"
6:07AM 1 Zaptel Won't Compile In Slackware 11.0 Kernel 2.4.33
5:56AM 1 Originating calls: Astmanproxy vs Direct Connection vs Call files
5:39AM 2 problem with safe_asterisk
5:30AM 7 error when compiling asterisk-1.4
4:24AM 6 Recomended POE Phones
3:36AM 1 Can Asterisk handle 7000 SIP users?
3:25AM 2 Customisable In-band ringing?
3:19AM 0 Error when compiling zaptel 1.2
2:43AM 1 question about regex
2:29AM 1 asterisk-sounds doesn't exist in the sources, how can i get it?
1:12AM 0 Local channel characteristics
Monday February 12 2007
10:24PM 2 Digium Card ?
7:24PM 1 End Wrap-up Time?
2:23PM 1 Small CDR Billing Program
1:51PM 2 colors in the console
1:33PM 0 Using Asterisk/callerid with "pay as you go" VOIP providers
11:43AM 2 i m looking for a document that allow me to install well an asterisk server
11:17AM 2 AsteriskNOW Migration
10:54AM 4 Zaptel install...
10:14AM 1 FW: After upgrade to 1.4 transfers don't workproperly
10:13AM 0 sendmail problem
9:42AM 3 Trixbox vs. Custom install
9:24AM 0 Parking via ## still broken
9:07AM 1 AGI question
8:48AM 2 Witch kernel version may i use to run well asterisk
8:19AM 2 T1 card recommendation
7:42AM 1 phpagi - Event "On Hangup"
6:56AM 1 fxotune on TDM24XXE card
6:43AM 0 Quintum gateways
6:41AM 3 Bad audio quality on SIP
6:06AM 2 Problems Asterisk with Digium TDM400 card => he don't see the disconnect
5:18AM 0 Resque Calls from someone who is already speaking
3:12AM 0 Asterisk-Java 0.3 Milestone 2
2:45AM 0 Using Asterisk's manager interface to recieve calls
2:36AM 3 Disable root shell from CLI
12:20AM 1 Cisco Router for supply a connection from PABX to Asterisk
12:11AM 0 Cisco Router for supply a connection from PABX to Asterisk ?
Sunday February 11 2007
9:49PM 2 Extensions in macro
7:58PM 0 chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1
7:05PM 0 realtime and save ip server in database
12:57PM 5 pmp_l1_check=no with zaphfc (Bristuff)
12:21PM 2 AsterikNow vs Trixbox
12:11PM 1 Debugging a SIP / AudioCodes Problem
8:49AM 2 TDM02B not working
7:15AM 2 Can not compile latest zaptel -1.2.13
5:25AM 0 TE110P working hardware configurations
5:12AM 1 TDM2400 and 3.3v pci
Saturday February 10 2007
11:33PM 1 Digium S101I as a traveling companion
9:15PM 3 Dial out from AGI
7:59PM 1 recording issues
4:47PM 9 Mini-ITX board + FXO PCI card?
10:55AM 3 Dell PowerEdge 2950 Sharing NIC IRQ with Digium Card
9:56AM 1 canreinvite problems
4:46AM 0 Unable to lookup host in c= line
2:21AM 0 problem with installing tdm2400
12:09AM 1 SIP retry time too low
Friday February 9 2007
7:18PM 3 changing callerid to ring groups callerid
6:30PM 6 The High Performance Echo Canceller (HPEC)
6:09PM 2 asterisk and multiple cpus/cores
6:06PM 1 Outbound Call Transfer Problem
4:37PM 7 Dialplan checkup
2:17PM 0 ring requested on channel
1:31PM 1 Detect hang-up
1:13PM 4 asterisk 1.4 FC5 and Gtalk
12:48PM 1 RFC2833 SIP trunks and DTMF
12:15PM 1 CallerID on Dish 301 Receiver
10:59AM 0 Re: asterisk-users Digest, Vol 31, Issue 37
10:42AM 0 Asterisk 1.2.15 released!
10:42AM 0 Zaptel 1.2.13 released!
10:25AM 0 Dependencies on DB?
10:06AM 2 Chan_Cellphone
9:22AM 0 Conference & Page question
9:09AM 0 TDM2400: some FXS module fail
9:05AM 1 anyone remembers where to check this list threads on a web site?
8:45AM 0 *****SPAMZ***** Conference & Page question
8:16AM 3 receiving fax with junghanns quadbri bristuff
7:51AM 1 Conferencing Phones ...
7:49AM 0 Asterisk 1.2.14 - Chanspy, sound issues.
7:10AM 1 Problems with GXP2000 and Asterisk => Call pickupand Voicemail
2:38AM 0 Misdn instability with asterisk 1.4
Thursday February 8 2007
11:23PM 1 call park and call transfer example
9:56PM 1 Problems with GXP2000 and Asterisk => Call pickup and Voicemail
9:30PM 2 requesting real world meetme capacity numbers
8:46PM 1 Queue extension issues
7:12PM 0 SIP Re-Invite behind a NAT
6:43PM 1 TDM400 with 1 FXO
5:02PM 1 Recording and MWI
2:50PM 1 Auto Answer (Paging)
2:42PM 0 tickets went on sale this week
2:08PM 1 Any Way to Get # Functionality in DISA
12:35PM 2 Asterisk outbound calling does not wait for answer before playback
11:47AM 2 Suppliers in Canada
10:58AM 4 error when compiling zaptel-1.4
10:36AM 3 Automatic Dial, Play message
9:57AM 3 Skutch AS-66 and an X100P
9:36AM 0 Transfer -> announce -> ring
8:57AM 0 Transfer
8:27AM 1 After upgrade to 1.4 transfers don't workproperly
8:21AM 0 Re: asterisk-users Digest, Vol 31, Issue 29
7:14AM 3 Digium cards on Vmware
6:00AM 0 mysql error
5:01AM 2 dCAP
4:10AM 0 Realtime asterisk queues only reload queue members when a new call joins the queue
3:54AM 0 T.38 FAx
3:17AM 0 dial application timeout
3:10AM 2 problem with asterisk AGI
2:28AM 0 Activate/Deactivate zap channels in realtime
1:45AM 11 Best phone for easy provisioning
1:36AM 3 Asterisk and 802.11g
Wednesday February 7 2007
10:57PM 1 Spliting video and audio
7:45PM 1 Large number of prefixes in a route to a trunk
5:53PM 3 Linux Kernel Timer Frequency and Asterisk
4:48PM 3 Red alarms
4:05PM 1 does any one knows of a Softphone that works under terminal services?
1:12PM 1 After upgrade to 1.4 transfers don't work properly
1:04PM 2 Softphone +Realtime
12:36PM 0 RE: Linksys auto provision
12:30PM 2 List problem handling HTML E-mails?
12:13PM 0 Zaptel bug
12:04PM 2 CPU & motherboard for 100+ simultaneouse calls on Digium Quad E1 TE411p
11:50AM 0 semi-private call
11:44AM 0 Can't get asterisk to compile chan_zap (was "NewIssue")
11:26AM 3 Diagnosing poor call quality
10:02AM 2 Can't get asterisk to compile chan_zap (was "New Issue")
9:27AM 2 AMI Originate and release channels
8:55AM 3 Trying to register an G.729 codec boght from Digium and the "register" command does aboslutely nothing
8:04AM 4 Billing pulses
7:24AM 0 prob with not recognizing hangup, pickup - python
7:19AM 0 Connection problem w/ Attended Transfer
6:46AM 1 Chanspy severe sound problems
6:41AM 1 H323 to SIP - One way voice
6:30AM 1 Asterisk Cmd to ID Mobile from Phone#?
6:08AM 0 SIP/Console -> ISDN ticks
5:42AM 0 Glitches in voicemail prompts
5:32AM 0 dnsmgr seems to have died
5:25AM 0 one touch recording problem in asterisk 1.4
3:26AM 1 registration not timing out?
2:27AM 0 Pickup
1:53AM 2 Type of wake-up Call
12:57AM 4 s-${DIALSTATUS} extensions
12:29AM 9 Digium TE110P
Tuesday February 6 2007
9:21PM 0 International validation
9:10PM 2 Disconnection supervision: what about PBX
7:30PM 1 Are there any IP phone in the market have such features?
7:12PM 2 Buddy list order
6:16PM 3 Help - Poor Voice Quality
5:16PM 0 ooh323 drops registration with Cisco IOS GateKeeper - bug or config issue?
1:41PM 1 yellow alarm after weeks without trouble
1:33PM 2 Mysterious tables starting with "stats_"
12:46PM 2 CD needed: no way to burn
9:59AM 4 Something wrong with the list?
7:33AM 0 Call Connections Dropped
7:10AM 0 troubles gsm-gateway no free channels
6:55AM 0 dtmf not recognized with misdn-install: help for alternatives
5:35AM 0 Hardware perfect for TE412P runnning * 1.4
4:40AM 1 asterisk server as a voicemail server forlegacyPBX -- FXO or FXS???
4:38AM 0 asterisk server as a voicemail server forlegacy PBX -- FXO or FXS???
4:06AM 2 BindPort
4:04AM 1 pridialplan/prilocaldialplan
3:33AM 0 Asterisk H.323, Cisco IOS Gatekeeper(s) intra-zone call routing and TETRA
2:22AM 0 ExtensionStatusEvent
Monday February 5 2007
11:42PM 1 Inserting a pause with Sipura in between
10:27PM 4 Having Trouble With Wait Command in Callback Context
9:34PM 1 How to access environment variable?
6:03PM 2 asterisk server as a voicemail server for legacy PBX -- FXO or FXS???
5:40PM 0 Help - Received response: "Forbidden" from'"Unknown"
5:21PM 0 Callfiles to Meetme Fails (was: RE: Using Local Channels with Originate)
4:46PM 2 Howto use PRI lines (E1 or T1) for "data calls"?
3:09PM 1 Question on G.729
2:53PM 1 Sending sound to an open channel....
2:44PM 1 OpenSuSE Firewall2 - Traffic Shaping
2:10PM 2 how to install Zaptel on Fedora linux 5
10:41AM 0 Packek2Packet Bridging vs. Native Bridging
10:22AM 0 TDM2402E routing incoming calls out to cell phone low volume
9:20AM 0 Reliably detecting hangup
9:09AM 1 Trixbox 1.2.3 - TDM400 FXOs - Outgoing Calls - Transfer # Not Wor king
7:57AM 1 Test to Speech
7:30AM 1 'h' extension and which one applies?
7:29AM 2 Use Digium TE110P Single T1 / E1 PCI Interface Card for connect a old PABX ?
6:46AM 2 playing wav49/gsm files on linux?
6:04AM 0 *****SPAMZ***** Asterisk cluster - keep up connection?
5:52AM 1 format_wav.c:247 update_header: Unable to find our position
3:40AM 0 translation error
2:52AM 0 Anybody using "VoiceRD" or "Thinkbright" service?
2:13AM 2 voip office pbx
2:11AM 0 SNOM phones stay "in use" after transfer
1:57AM 5 Asterisk Faxing Support
Sunday February 4 2007
11:42PM 1 Local hangup after Dial()?
9:49PM 0 Does TE212P card work on HP DL380 G5?
9:08PM 1 Help - Received response: "Forbidden" from '"Unknown"
8:01PM 0 WARNING[27249]: chan_zap.c:6299 ss_thread: CallerID returned with error on channel 'Zap/4-1'
7:33PM 0 Tampa Asterisk User Group Meeting Monday
6:57PM 1 FreeBSD Compile Errors
6:32PM 0 How do I debug?
5:03PM 5 Unicall/R2 for Asterisk 1.4 Available for TESTING
4:42PM 2 SIP privacy headers
3:43PM 1 Continue line in config files?
3:27PM 1 Problem loading AstDB into variable on restart
12:54PM 0 dnsmgr died?
12:29PM 9 Zap FXS slow to reset?
12:04PM 0 VM - User Only able to set unavail message
8:37AM 1 bristuff mailinglist
8:17AM 1 Detecting answer with an analogue card
5:26AM 0 SMDI support on trixbox
4:08AM 0 Help sought: Asterisk H.323, Cisco IOS Gatekeeper(s) intra-zone call routing and TETRA
3:28AM 0 Asterisk and multicore processors
2:23AM 1 Asterisk 1.2.14 and bristuff 0.2.0-RC8s
2:21AM 2 TDM400 noHangup
1:39AM 1 TDM400 stopped bridging outgoing FXO call
12:58AM 1 Interact with IVR
Saturday February 3 2007
10:47PM 2 Command to disconnect a call
6:24PM 1 Single BLF for ALL trunks in use
2:06PM 2 Google Talk without gmail accout?
11:35AM 2 asterisk 1.2 branch revision 53132 failed to compile
10:50AM 3 error dialing a SIP user. chan_sip.c:1994 create_addr: No such host
5:22AM 1 misdn and prostgres_cdr on asterisk 1.4
Friday February 2 2007
9:40PM 0 Call Waiting broken on ZAP
2:13PM 0 7912 issues half audio
12:38PM 1 asterisk server RFC conformance
11:48AM 1 1.4 res_snmp dependencies (Debian)
10:35AM 1 queues and LOCAL for members
10:17AM 0 Local channel with /n doesn't hangup after transfer. Why?
10:06AM 0 No RTP packets received by Asterisk when calling SIP to SIP
9:51AM 0 FOP (or equivalent) and timers
8:53AM 1 CallerID Name not available.
4:31AM 2 Asterisk logging everything?
4:12AM 0 install-misdn compile problem with debian
4:09AM 0 Line drops
3:05AM 1 volume control in VoIP
2:13AM 1 WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels ( when I use asyncgoto)
Thursday February 1 2007
11:19PM 1 Asterisk cann't redirect the calling party to anothere Exten.
10:28PM 3 How to Clone Asterisk
7:45PM 3 windows SIP Softphones ?
6:43PM 0 Re: why there havn't""fileaboutasterisk1.4.0?
6:28PM 1 API Originate Action - distinguishing between No Answer and Invalid phone number
6:20PM 2 make: expand.c:489: allocated_variable_append: Assertion `current_variable_set_list->next != 0' failed.
6:18PM 0 Asterisk Scaling/Load Balancing for iax soft clients
3:25PM 1 Using Local Channels with Originate
3:15PM 1 Please help parse this GotoIf line
3:03PM 0 server hardware choice,
1:30PM 1 Dial option G - Passing parameters?
12:57PM 8 Dell Servers
9:15AM 1 dialplan logic based on caller ID
8:05AM 1 3 PCI slot with exclusive IRQ ? please advice!
7:47AM 2 SendText() question
7:27AM 1 Re: why there havn't ""fileaboutasterisk1.4.0?
7:00AM 0 Re: why there havn't "" fileaboutasterisk1.4.0?
6:37AM 0 asterisk 1.4 and r2mfc or unicall
5:45AM 0 Dialplan programming vs. AGI vs. ???
4:15AM 1 CDR - uniqueid
3:19AM 0 Enhanced PickupChan
3:15AM 0 extensions.conf gotoif and label
2:30AM 1 why there havn't "" file about asterisk1.4.0?
12:21AM 2 strange caller display