asterisk users - May 2006

Wednesday May 31 2006
TimeRepliesSubject
10:32PM 5 Openion on Sipura SPA-2100
10:00PM 5 how to decrease answer time !
9:51PM 0 app_ices.c broken pipe error : bug ??
9:15PM 2 : Re: Upgrade ONLY asterisk from an AAH install
6:54PM 1 Problems with ZAP dial timeout
5:25PM 1 clicking and popping with capi, okay with mISDN
4:19PM 0 Libmfcr2 won't compile
4:15PM 0 Ringing to Outside Line
3:56PM 4 MFC/R2 for Voice and Data
3:30PM 8 Connect 2 Asterisk Servers via PRI
3:14PM 8 Alternative to FWD
3:13PM 1 Brief silence followed by DTMF tone on T1 line
12:55PM 1 Can you dial with different CID's?
12:21PM 7 AEL2 and CID
12:17PM 9 Forcing Marker bit
11:56AM 0 NEW => Asterisk Event Monitor
11:35AM 0 TDM no dialtone on connected phone
11:34AM 6 Upgrade ONLY asterisk from an AAH install
11:23AM 3 PAP2-NA Authentication Issues
11:11AM 0 No system sound with Asterisk@Home
11:07AM 1 Looking for a VoIP solution...
10:48AM 1 Zap Flash()
10:33AM 6 Converting .wav to .WAV
10:23AM 0 AEL #include ( Now Labels & Goto() )
9:13AM 0 Hold Status
9:09AM 6 Asterisk crashes at startup
9:05AM 7 Explicit Dialplan Exit
8:15AM 7 SIP Presence
7:58AM 3 Labels and Goto()
7:53AM 0 DTMF Again
6:39AM 0 Incoming IAX going to wrong context
6:11AM 0 Bristuff PickUp and call transfers - can it be done?
5:39AM 3 Global variables - collision?
5:08AM 2 Zap Channels , for round-robin search and call
4:59AM 0 Asterisk Bootcamp in Europe :: June 12-16 and the Asterisk SIP Masterclass in Chicago, July 2006
4:58AM 6 Zap channels ringing too loudly
4:41AM 3 Centos cause Asterisk crash
4:41AM 0 extra parameter for DB read function
3:52AM 5 Need help with Junghanns Quadbri
3:31AM 1 *****SPAM***** Upgrading
3:00AM 0 Fax to Email issue with Spandsp tif not correctly sized
2:23AM 0 Asterisk receiving call from Panasonic TDA extension issue
2:01AM 1 INFO: TFOT book- n priorities and labels
1:52AM 0 AGI MySql
12:51AM 2 Nokia E60 , experience as SIP client
 
Tuesday May 30 2006
TimeRepliesSubject
11:11PM 1 Questions from a working doctors' office installation
9:51PM 0 Linksys spa 942 handsfree SIP->PSTN/GSM
9:25PM 1 Sip gateway don´t hangs up
8:13PM 15 Handset recommendations
6:33PM 0 Register Today For AstriCon Europe
6:22PM 1 Shared Call / Bridged Line Appearances (SIP-B)
6:19PM 0 zt hook failed
6:15PM 5 Got SIP response 405 "Method not acceptable" back from xxx.xxx.xxx.xxx
5:29PM 4 Polycom replacement handset
4:19PM 1 BEST PRICES ON NMS DIALOGIC DIGIUM VOIP WWW.VOICEINTERNATIONAL.COM
3:46PM 6 Asterisk 1.2.8, Zaptel 1.2.6 and libpri 1.2.3 released!
3:26PM 0 Problem with tor2 driver and Zapata Tormenta 2 Quad T1/PRI Card
2:31PM 3 Still can't get asterisk to play voicemail files occasionally
2:08PM 2 Callerid and trunk
1:55PM 2 Dropped SIP connections never being closed?
1:42PM 15 Compiling Asterisk-addons
1:29PM 2 No sound?? HELP
1:17PM 3 instalacion
1:00PM 9 How to strip a digit
1:00PM 1 Is Asterisk svn link down ?
12:57PM 40 AEL #include
12:19PM 2 Asterisk::AGI and DIALEDTIME
12:03PM 1 CallerID outbound
11:53AM 0 RE: Asterisk-Users Digest, Vol 22, Issue 169
11:40AM 4 Unicall Protocol Failure
10:15AM 0 app_conference sources?
9:50AM 3 Zaptel and 2.6.9-34.0.1.EL Kernel on CentOS
8:54AM 1 patch application
8:50AM 0 Dumping outbound audio on hold
8:43AM 2 Automon
8:33AM 0 IAX softphone with RSA support?
8:32AM 0 LDAP directory app?
8:22AM 1 Hardware requirements for Asterisk
6:34AM 0 no extension from ISDN phone with bristuff
6:05AM 0 Extensions, devices and dialplan
5:09AM 4 Panasonic PBX
3:10AM 2 problem about asterisk realtime.
2:26AM 2 Asterisk restarting in a minute
2:19AM 1 sIp port numbers
1:56AM 10 I guess my server capacity is ok
 
Monday May 29 2006
TimeRepliesSubject
10:42PM 25 E1 hardware for asterisk
10:31PM 3 sip interopability problem
4:30PM 0 Melbourne Asterisk Group meeting Thursday
4:23PM 9 Simple windows / web Asterisk user software?
4:05PM 0 Sipura 941 missing blind transfer soft button?
3:56PM 8 Problem with IAX2 dialin with portunity
3:32PM 7 Recent debian packages?
11:33AM 16 app_conference DTMFs?
11:07AM 1 Re: Nufone Echo Test
10:46AM 5 Asterisk Internal sip calls I can´t send/recive
8:53AM 0 Brother 8360P fax cannot connect to TDM400
7:03AM 8 How to enable call waiting on Sip Phones
6:27AM 15 registration at Voipbuster times out
6:14AM 4 I can't call PSTN numbers
6:07AM 11 Memory-leak 1.2.7.1
5:14AM 0 Define call-groups
4:25AM 0 pedantic on sip.conf
3:56AM 0 Asotel Dynamix DW-04/S with asterisk?
2:56AM 1 Ring-Answer with Polycom 501 and Asterisk
2:15AM 0 New Zealand Voice prompts announcement
12:44AM 0 doubts about asteriskconfigurationfromdatabase
12:33AM 5 TDM2400P with echo canceller not working
 
Sunday May 28 2006
TimeRepliesSubject
11:04PM 7 doubts about asterisk configurationfromdatabase
8:17PM 4 IVR sounds not on certain inbound route
7:36PM 0 Go2call Configuration
6:43PM 4 doubts about asterisk configuration from database
4:18PM 4 Asterisk registers but won't complete calls.
2:09PM 3 Asterisk Radius Module
1:36PM 5 hook into authentication
12:40PM 2 Analogue phone w/ TDM400
12:13PM 0 SIP and sound breaking
10:36AM 13 Calls connected, but no audio
4:55AM 0 SER qualify
4:26AM 0 My Call drop after 60 to 63 Seconds!!
3:46AM 1 FreeBSD Digium g.729 codec seg faults on rev 30652
 
Saturday May 27 2006
TimeRepliesSubject
2:06PM 10 TDM
1:43PM 1 Fw: features
1:26PM 2 Polycom 600 presence indication on *LED*?
10:26AM 1 Dcap Test
9:54AM 2 Web based interface
8:51AM 1 Compiling chan_bluetooth
6:30AM 4 Calling a person over Internet
6:17AM 0 JabberStatus
3:03AM 4 amportal doesn't start with brestuff(ISDN)HFC-PCI
 
Friday May 26 2006
TimeRepliesSubject
9:53PM 1 asterisk with centos 4.3 sources compilation
8:54PM 18 mpg123 or asterisk
8:32PM 0 RV: DELL PowerEdge 2850 and TE4110P and TE110P
7:11PM 0 Polycom 601
3:13PM 0 AMP and version numbers.
2:29PM 1 External Custom Extension Timeout
2:10PM 0 Sip Notify cisco-check-cfg - Does it still workwith 8.2?
12:33PM 5 Sangoma A200 4 port FXO card suddenly stopped answer on channels 2, 3, 4
10:35AM 3 Busy Signals
8:58AM 6 UK experts only. BT Outgoing caller ID not showing
8:51AM 1 End of migration: adding support for some an alog phones
8:36AM 1 OT: American Telecom Approved by FCC to Certify DECT Phones in US
8:33AM 2 IAX2 + port translation
7:51AM 2 large duration calls
7:38AM 4 Two questions about Asterisk@home and backups.
7:35AM 1 hints/subscriptions accross IAX
6:21AM 1 VoIP provider for Turkey from India with Asterisk
6:15AM 2 Need a recomendations and config samples. FXS<->SIP terminal with 4 ports.
6:07AM 3 hint priority and realtime
6:01AM 8 using a billing system
5:51AM 0 Getting stuck right at the beginning
5:37AM 6 End of migration: adding support for some analog phones
3:56AM 1 my kernel not detect my TDM400P card
2:49AM 3 Polycom 301's drop last two digits of dialed number
2:22AM 2 Asterisk.NET authentication problem
1:56AM 0 SIP call problem
1:46AM 1 Not able to make any calls
12:15AM 0 No sound when the call is diverted
 
Thursday May 25 2006
TimeRepliesSubject
11:26PM 2 Modules for X100P
11:00PM 5 Agent Callback, how to "see" wath queue is calling the agent?
10:25PM 2 PAP-2 Conferencing Problems
10:10PM 0 IAX registrations fail over time in SVN-trunk
8:10PM 6 pap2 bridging problems
6:32PM 7 X100P fails to initialize
6:19PM 0 Citel Handset Gateways and BLF (subscribe) buttons?
4:57PM 3 RRMEMORY / Queues Not Working Right
4:11PM 2 Way to disable codec in dialingplan
3:42PM 27 Snom firmwares suck <--additional datapoint to consider
2:49PM 0 Re: [asterisk-biz] Selling Bulgarian (+3592) DIDs at 1.5 USD
2:31PM 0 problems with TXfax
1:38PM 0 Anyone going to cluecon?
1:26PM 5 No rings before auto attendant
1:12PM 0 RE: Asterisk-Users Digest, Vol 22, Issue 147
12:54PM 2 Paging Phones stay off the hook if you dont wait long enough.
12:43PM 4 jitterbuffer causes flaky IAX2 incoming connections?
11:37AM 0 FW: [isp-clec] Treasury disconnects tax on long-distance calls - with refunds
11:21AM 0 PRI Moving channels?
10:41AM 8 FreePBX virtualization
10:22AM 9 Compilation issues with s390
9:53AM 7 Asterisk codec negotiation patch
9:30AM 1 Asterisk Manuals
9:17AM 0 Asterisk and sysmask - anyone?
8:51AM 18 PCI Problems
8:49AM 0 RE: Asterisk-Users Digest, Vol 22, Issue 132
8:34AM 4 Failover Problem
7:53AM 0 Glueing apps and phones together
7:47AM 2 Volume configuration on Polycom Soundpoint 501phone
7:32AM 0 Re: Implementing Paging on the Linksys SPA9XX phones (working)
7:16AM 0 CallerID from cell phone not being rewritten
7:03AM 2 "Error" on Polycom 501 & 601.
6:56AM 2 IVR & transcoding & g729 license
6:49AM 2 [asterisk-biz] RE: OT: AudioCodes MP124-C/FSX/AC/SIP
5:47AM 0 Anyone got a used T1 card I can have?
3:43AM 4 VLAN info
3:14AM 5 Voice Mail Audio Progression
1:10AM 2 connecting asterisk to hylafax via t38modem: is it possible?
1:00AM 0 TDM2400P Problem
12:42AM 1 playback windows recorded sound
 
Wednesday May 24 2006
TimeRepliesSubject
4:57PM 2 PCI-X PRI hardware
4:44PM 0 SPA-941 called number distinctive ring with Personal Directory
4:40PM 2 Realtime Asterisk Problem
3:47PM 0 uClibc and g729
2:20PM 0 Dual Line SIP config to the same provider
2:09PM 1 database lookup
2:00PM 11 Is NuFone Really Dead?
1:42PM 3 latest @Home questions
1:11PM 8 Spoofing a BLF Signal?
1:11PM 9 What and When is the next version of Asterisk?
1:00PM 5 OT: AudioCodes MP124-C/FSX/AC/SIP
12:55PM 2 Lighting up a light on an aastra phone
12:25PM 2 DHCP configuration for Cisco 7960?
12:02PM 1 Problem after upgrade to 1.2.7.1
11:15AM 7 TE406P - MFC/R2
10:49AM 1 Misdn 0.2.1 BUSY tone
9:00AM 8 macro-dial
8:56AM 1 DUNDi in 1.2.7.1
8:48AM 2 Placing call files in/var/spool/asterisk/outgoing/ does not work
8:43AM 0 Placing call files in
7:35AM 2 Generate two calls from Asterisk and bridge them
6:48AM 1 How to add H.323 channels on Asterisk 1.2.7.1
6:06AM 2 Placing call files in /var/spool/asterisk/outgoing/ does not work
6:02AM 1 Configuration for different Asterisk branches
5:38AM 0 [Fwd: IVR and operator]
5:16AM 2 Video SIP Softset
4:59AM 0 SIP Video software
3:30AM 13 GXP2k and BLF problem
3:26AM 2 asterisk amportal start/stopped/start/stopped for all the time
2:45AM 0 spanDSP & app_rxfax.so
1:52AM 3 How to prevent more than one agent to login to the same extension??
1:26AM 6 SV: USB headsets?
1:01AM 14 USB headsets?
 
Tuesday May 23 2006
TimeRepliesSubject
11:48PM 0 [asterisk BUG]hangup
11:46PM 3 chan_zap.so error, asterisk stopped
10:52PM 2 Configure Voipjet.com content in Asterisk
10:51PM 0 FAX with PRI
10:13PM 3 Packetization configuration of IAX channels
9:06PM 2 Quintum Tenor DX 3020 problem to register on Asterisk
7:07PM 1 multiple registrations with Polycom IP600
4:30PM 1 SPA 3102 Caller ID in Bellsouth/NA
3:55PM 1 They are? Re: Now that Nufone is dead...
2:36PM 0 IVR and operator
2:14PM 1 Getting the Server IP
2:04PM 0 Wacky Failover Situation w/SIP - Bug?
1:36PM 3 More Alison Keenan British English files
1:02PM 1 Problem with options to "Dial" application
11:12AM 2 Queue Count
10:46AM 5 PSTN -> CCM3.2 -> Asterisk CLID
10:22AM 0 CVS servers being taken out of service
10:09AM 8 AGI ?
10:08AM 3 Database Integration
10:05AM 0 Virtual VOIP numbers going to separate Asterisk mailboxes?
9:08AM 0 Zaptel Module.symvers missing
8:40AM 3 Transfer extensions processing control to Manager
8:21AM 0 Sip.conf: domain=huh?
8:10AM 1 Monitoring queues
7:48AM 35 Now that Nufone is dead...
7:48AM 7 What about T400 T1 cards?
7:39AM 1 Can Asterisk work in a proxy setting- a challenge
7:23AM 0 [asterisk BUG]
7:19AM 2 res_snmp
7:02AM 0 SIP Softphone or API which supports QoS (DiffServ/DSCP) needed
6:50AM 2 Asterisk connecting to a proprietry PBX
6:46AM 5 Queues - Can I PAUSE an agent instead of LOGGING OUT?
6:14AM 0 Problem in php-asmanager.php
5:56AM 7 Best VoIP provider for Asterisk
5:26AM 21 Are my expectations too high?
5:17AM 2 Outband call from php script
4:21AM 1 Im a Beginner
3:01AM 2 config files for Eicon Diva
1:38AM 1 AW: Free/Open pci telco card
1:22AM 1 Status: Provisioned, Down, Active - Long
1:21AM 0 A call from a call file always does a redial?
1:11AM 1 Free/Open pci telco card
1:10AM 5 Logger rotate & master.csv
12:58AM 0 SIP session number
12:40AM 0 [Fwd: Faxing - machines stop talking, line stays up]
12:35AM 2 TDM400P , "ztcfg ?vv error ", "Does it have to do with my PC hardware ?"
12:23AM 0 Faxing - machines stop talking, line stays up
 
Monday May 22 2006
TimeRepliesSubject
10:18PM 14 US telco lingo
5:25PM 1 Timeframe for QueueStatus values
3:11PM 4 I've broken voicemail
2:31PM 2 How to detect call forwarding to voicemail
2:27PM 1 Initial second lost on SIP phones
1:52PM 0 Voicemail: cannot use serveremail as variable
1:28PM 0 PRI bi-directional early media
1:19PM 0 SIPCHANINFO and 1.2.7.1
11:15AM 2 A few queue questions
10:22AM 1 FXS Caller ID revisted
10:21AM 0 UUI field
9:34AM 0 Persistennt Data of Queue with Dynamic Agents
9:16AM 5 Centos 4.3 Issues
8:50AM 0 Asterisk Nortel Legacy Integration
8:28AM 1 TLS from a Sponsored Google Summer of Coding?
8:21AM 1 Script AGI on C
8:11AM 10 Office to Office via IAX2 problems
7:20AM 4 Option to reach someone in voicemail?
7:16AM 2 exten => *0. not possible
5:35AM 6 I get MOH when the caller hangs up
4:54AM 6 Problems with Park and MOH
4:17AM 0 Got reject for frame 0, but we only have others!
4:13AM 2 how to customize voicemail
4:04AM 2 Recommended SIP phones?
2:55AM 1 Asterisk on Proxy
2:04AM 0 Please help on chan_h323.
1:57AM 4 SIP to IAX - forcing codec pass thru
1:53AM 16 FW: WiFi / GSM VoIP Handsets..
1:18AM 0 string parsing in extensions.conf
12:35AM 2 Not able to configure TDM400P with asterisk@home
12:08AM 1 behaviour depending on count of used lines
 
Sunday May 21 2006
TimeRepliesSubject
11:06PM 3 Snom 320 Shared line + speed dial
9:17AM 1 Limit outgoing calls
8:00AM 0 update or add DID's to directory Assistance
6:47AM 1 Skill-based routing
6:04AM 1 transfer outside of a call?
5:28AM 2 Upgrade 7960 from SCCP 3.0 to SIP 7.5
2:27AM 1 Events offered by
2:15AM 1 no ringtone
 
Saturday May 20 2006
TimeRepliesSubject
5:33PM 4 Configuring a TDM400P with one FXS port
6:43AM 0 "Slash Tone" at pstn cut-though?
6:31AM 1 h323 to sip ringing indication
4:52AM 1 $1000USD for fix of Asterisk g726-32 codec
4:30AM 5 Any IP phones with pro-audio connections?
2:52AM 8 How to unlock old SCCP Cisco 7960 ?
2:45AM 1 Cisco 7940/60 SIP firmware 8.3
 
Friday May 19 2006
TimeRepliesSubject
10:23PM 0 DID Provider via Asterisk
8:23PM 2 hardware help ?
1:30PM 1 Dell PowerEdge 1600 Compatibility Issues with Digium Card
10:25AM 0 Setup up Intellitouch ITC-3002 Sip phones with Asterisk
10:06AM 8 PRI dialing IVR with inband DTMF
9:05AM 6 British English voice files are ready for download
8:33AM 4 Non automated call parking
8:28AM 1 IAX Trunk
7:27AM 1 RTP Packetization
6:57AM 0 SpanDSP issues (oh fun!)
6:32AM 2 Not joining queue when empty
6:18AM 0 Forwarded Calls crash the system on 64 bit
6:02AM 16 Snom firmwares suck
5:27AM 1 Call detail records for Digital Receptionist
4:51AM 0 call recording - contrlo of Ast in 'h' extension
3:43AM 3 AsteriskOUT
3:02AM 2 voicemail access on the Thomson ST2030 ?
2:54AM 0 Faxing with Asterisk using both ISDN and FXS
2:40AM 0 help about modem
2:10AM 1 Watchguard Firebox 1000 woes
2:10AM 1 Development news :: Smarter medialess calls!
1:55AM 0 Call Transfer does not work
1:30AM 5 X100P not recognised on FreeBSD system
1:20AM 0 Show queues statictis
1:15AM 6 Experience with IBM X346 machines and Sangoma
1:06AM 2 Max Number of Extensions
12:38AM 2 SIP useragent?
 
Thursday May 18 2006
TimeRepliesSubject
10:43PM 0 Error building Oh323
9:14PM 1 Digium card firmware
8:35PM 0 Fwd: [Announcement] Asterisk-IL mailing list
5:08PM 0 <SOLVED> Need help with Dial M option and destinationcontext
3:05PM 3 SIP Header Info
1:59PM 2 VoiceMail Groups
1:22PM 0 Pulling the mISDN number from an incoming call
1:03PM 0 E&M and Dial tone
11:19AM 3 R2/MFC Configuration.
11:14AM 0 Powertouch 350 CallID display continued
9:38AM 2 Auto Dial Out Madness
8:56AM 0 OT: Aastra Powertouch 350 caller id
8:36AM 5 Polycom - missed calls dial back
7:58AM 4 Polycom 601 -- programming buttons.
7:19AM 0 Applet to test VoIP quality
6:59AM 1 SIP re-invite and billing
6:52AM 6 Home asterisk system with single PSTN Line
6:33AM 0 Asterisk - SPA-3000, 407 error
6:26AM 4 Default dialplan??
6:06AM 0 multiple calls using IAX
6:03AM 2 ACD Light on Phone?
4:18AM 6 just softphone
4:07AM 0 tdm21B in china
4:00AM 3 Unable to register channel
3:29AM 1 SNOM, g722 and 16 kHz audio
3:29AM 0 Failing SIP registration brings * down
1:57AM 1 DM/V1200-4E1 with asterisk
1:31AM 0 Unable to set channel to linear mode
12:16AM 0 Trunk Si without autetification
 
Wednesday May 17 2006
TimeRepliesSubject
11:32PM 0 Asterisk@home default password doesn't
10:36PM 2 Meetme conf
8:53PM 0 [Fwd: Calls being hung up]
7:20PM 3 [OT] Disconnect Tone in US
4:28PM 1 Is there a dialplan emulator available?
3:33PM 6 SPA-1001 behind NAT -> Internet Asterisk box -- BOUNTY!
3:27PM 0 Need technical info about dialers
3:00PM 5 Slackware 10.2
2:48PM 4 Asterisk & Meridian Tie Line
2:25PM 2 AAH not getting IP address, likely to be network card?
2:19PM 3 New To Asterisk - Advice needed
2:18PM 0 AutoDialer Software
1:59PM 22 Quad BRI card
1:10PM 0 Audio problems 50% of the time. (kurt x)
1:00PM 0 RES: GET DATA and STREAM FILE commands, don´t work
12:59PM 7 Ringing indication not working as expected
12:54PM 5 Variable Inheritance - Set in Child, Read by Parent
12:08PM 0 DM/V1200-4E1 (Intel PCI 4xE1 ports)
11:37AM 5 Audio problems 50% of the time.
11:19AM 0 Upgrade issues
9:55AM 3 Providers using Embedded Devices
9:53AM 1 Weird Error When upgrading 7960G to 8.2
9:51AM 0 Asterisk Using Multiple Databases with ODBC?
9:51AM 0 Weird Error Upgrading 7960's to 8.2SIP
9:43AM 1 Asus P5GD1... anyone using with Asterisk ??
9:38AM 0 Can two asterisk servers share the same dialplan by using FreePBX?
9:29AM 3 Listening on Multiple Interfaces
9:19AM 0 RE: Asterisk-Users Digest, Vol 22, Issue 97
9:11AM 0 Asterisk SIP Gateway behind NATS - SIP/2.0 404 Not Found
8:29AM 2 SIP redirect
8:26AM 0 fax & asterisk 1.2
8:08AM 0 Overwriting SIP headers
7:19AM 12 soekris hadware
6:50AM 2 Diverse servers
6:27AM 2 IAX crackilng
6:25AM 0 (no subject)
6:00AM 0 Reading queue_logs
4:54AM 1 TDM does not disconnect
4:25AM 0 Re: Reasons for a SIP channel to hang ? - partially resolved
4:21AM 0 Asterisk Manager and Events Problem
4:05AM 1 Deadlocks in 1.2.7.1
3:30AM 12 Plan to free myself from AAH
3:30AM 0 A CDR issue of agent.conf <createlink feature>
3:29AM 1 (newbie) Zaptel/ztdummy compiling on debian
3:13AM 2 SIP Min-Expires
2:55AM 1 NO ringing tone while dialing
2:36AM 4 no SUBSCRIBE request sent
1:17AM 2 Asterisk@home default password doesn't match
 
Tuesday May 16 2006
TimeRepliesSubject
8:26PM 9 mISDN & FAX
8:20PM 1 Asterisk as a proxy
5:50PM 1 GXP-2000 w/ 1.1.0.11 firmware
5:37PM 18 DELL PowerEdge 2850 and TE4110P and TE110P
5:25PM 0 Need help with Dial M option and destination context
2:48PM 0 News from France: Free, SIP and Asterisk
1:55PM 0 AstriCon Europe Update - 6 Weeks To Go
1:47PM 3 Having a Blonde moment.
12:42PM 3 Multiple Registers
12:41PM 3 Polycom 501 logo onscreen
12:33PM 0 Asttapi for Asterisk 1.2 Testers Needed (was RE:Asterisk TAPI - Outlook click2dial)
12:28PM 6 Delay when ringing internal extensions on incoming zap call
12:27PM 7 chan_capi-cm and dialing without number
12:16PM 1 change dchannel number
11:57AM 1 Asttapi for Asterisk 1.2 Testers Needed (was RE: Asterisk TAPI - Outlook click2dial)
11:52AM 0 Asterisk Broadvoice outbound calling loop, now it goes to voicemail
11:11AM 2 Using REGEX function
10:13AM 1 error leaving voicemail in multiple VM boxes
10:05AM 3 chan_capi-cm and type of number problem (ToN)
9:38AM 18 Netherlands zaptel.conf
9:28AM 4 EICON, chan_capi-cm and averlap receiving
8:00AM 9 WARNING[4033]: Avoided initial deadlock for 'Zap/63-1', 10 retries! ... + Kernel Panic!
7:55AM 0 Re: [Astlinux-users] British English Female files ready for download
7:40AM 3 Meetme and authentication
7:28AM 1 crackling on IAX between asterisks
6:51AM 0 Reasons for a SIP channel to hang ?
6:15AM 0 Paging, Aastra 9133i, and Being on the phone!
4:21AM 2 call monitoring and indications / beeps
3:16AM 0 test -please ignore
3:00AM 3 regexp
2:36AM 0 call waiting announcement on agent phone
2:35AM 0 Join the Asterisk Video Task Force if you're into video telephony development!
1:52AM 4 asterisk and ODBC
1:33AM 0 iax2 disconnect problem
1:15AM 34 WiFi VoIP Handsets..
1:07AM 1 tdm2400p: fax detection not working
12:24AM 0 problem with sip registration with database
 
Monday May 15 2006
TimeRepliesSubject
11:24PM 1 Tr: Re: The OpenNMS Group, Inc.: opennms and asterisk pbx
10:47PM 5 unicall dialing problem
10:27PM 2 Career Opportunities
7:17PM 4 Asterisk as a bridge between voip clients and POTS confrence bridge
6:53PM 8 Asterisk on a WRT54G?
5:17PM 9 Multiple announcements in a queue ??
5:13PM 2 Outgoing Calls Not Working all the time
4:50PM 1 TDM400P static on call
4:32PM 5 Voicemail volume wav vs. wav49
4:09PM 2 Is it possible to delete global variables
3:54PM 3 Asterisk X100P - Interrupt a call?
3:43PM 3 queue help
2:21PM 1 Asterisk didn't start with app_swift.so
2:00PM 0 Asterisk didn't start with
1:58PM 1 Please help.. I need a h323 user for tests
1:57PM 1 Encrypted IAX termination
1:55PM 1 Realtime Postgres via ODBC
12:55PM 0 SNOM autoanswer question
12:30PM 1 RE: [PROBLEM] Still exist --> DTMF Tones, occures in Asterisk - Channelwide
12:16PM 2 Asterisk with SIPconnect
12:00PM 0 Vancouver Asterisk Users Group
11:53AM 16 Which is the best fax-modem for testing ?
11:13AM 1 Please..... need some help
10:52AM 3 How to tell if RTP stream is has been reinvited?
9:49AM 10 Turning AAAH into a call-center
9:47AM 28 Eicon Diva - problems building new v3 melware driver
8:17AM 6 VOIP adapters to connect PSTN lines to SIP phones
7:18AM 1 GET DATA and STREAM FILE commands, don´t work
6:55AM 1 Broadvoice does it again
6:51AM 0 Ottawa Asterisk User Group Kickoff - Wed -- May 17 -- 5:00
6:50AM 0 Echo cancel voip channel?
6:36AM 0 fax possible with standard modem
6:26AM 0 problem with sip registration ramdomly
5:23AM 0 Voicemail indication on Mitel 52xx phones
2:33AM 0 A sugestion for asterisk
1:44AM 0 agent deadlock
1:40AM 1 View Agent Status on the Web
1:14AM 1 VoIP Adapter
 
Sunday May 14 2006
TimeRepliesSubject
11:25PM 1 E1 + sangoma + soekris
10:01PM 1 Getting Realtime running (1.2.7.1)
5:01PM 1 Asterisk Manager interface
4:01PM 2 911 @ Zap Channel Breakin
2:05PM 0 VoipBuster issues?
6:39AM 0 [patch] fix for redirect manager action with BRIstuffed Asterisk
6:01AM 0 Re: h323.conf and realtime
4:41AM 0 IAX/SIP to germany with own callerid?
 
Saturday May 13 2006
TimeRepliesSubject
11:39PM 3 plainvoip - IAX2 call rejected
11:36PM 0 Contract Work : On-site NYC
6:03PM 1 Looking for Level 3 DID's, USA termination, USA 800 termination/Orig
5:29PM 0 Spam? Re: Cisco 7970 problems
7:08AM 10 Confused !
6:00AM 0 Re: [asterisk-dev] SNMP support for Digium Cards
5:46AM 0 Re: [asterisk-dev] SNMP support for Digium Cards
5:15AM 0 Re: [asterisk-dev] SNMP support for Digium Cards
4:59AM 0 Re: [asterisk-dev] SNMP support for Digium Cards
3:38AM 0 parking a call /put on hold
2:56AM 0 RE: snmp and asterisk
 
Friday May 12 2006
TimeRepliesSubject
11:59PM 1 Sipura 1001
3:18PM 3 VoiceMail application: "j" option not working as I supposed
1:30PM 13 fc5 and link to sources?
12:37PM 1 Cell phone dialed digits too short to be recognized by asterisk
11:57AM 7 DUNDi and Voicemail
11:21AM 6 voicemailmain()
11:21AM 1 Plain Text Passwords for IAX and SIP
11:10AM 3 Dial Command Reference for SIP channel
10:32AM 0 RE: snmp and asterisk
10:30AM 1 Having Rinback tone generation issues with 1.2.7.1
9:39AM 2 Cisco 7970 problems
9:11AM 2 Help Avaya 4606
9:02AM 7 Music on Hold restart at beginning for each call
9:02AM 3 Voicemail WAV to PDA Problems
7:01AM 2 URGENT please call parked / MOH
6:06AM 1 call parked / MOH
5:19AM 1 Speex fans?
5:09AM 9 S100-FX v2 audio quality
5:05AM 4 Sangoma A200D problem
4:57AM 2 email -> fax gateway with billing possibilities?
4:41AM 0 SCCP audio problems
2:52AM 0 RE: [PROBLEM] Still exist --> DTMF Tones, occures in Asterisk - Channelwide
2:18AM 0 extension.conf for overlap
1:45AM 8 Echo cancel: chan_misdn vs bristuff? HFC card vs expensive card?
1:07AM 0 Alarmreciver finally found ATA
1:05AM 0 SIP/NAT disconnection issue
12:59AM 0 Sip domains, contexts and CHECKSIPDOMAIN
12:39AM 3 monitoring sangoma cards via snmp
12:39AM 0 Asterisk & BRI in the USA - Episode 2 "The Phantom Sales Rep"
12:34AM 1 TE110P on E1
 
Thursday May 11 2006
TimeRepliesSubject
11:27PM 0 issue has arisen
10:18PM 4 Please Help Me...Urgent
9:50PM 1 Linksys IP Device Bulk Provisioning Guide
9:03PM 0 Asterisk + G.729 on Sun T1000/T2000
8:38PM 1 Issue for RE-INVITE with G.729
8:08PM 0 ast_dsp_call_progress
7:40PM 1 How many SER and asterisk servers does FWD users.
6:33PM 1 Canada Termination
5:19PM 0 Delete global variable
4:00PM 0 British Voice talent records Asterisk prompts
3:56PM 10 MeetME Conferencing
3:55PM 2 Problem setting locale for voicemail
3:33PM 0 Zap DTMF detection
2:19PM 2 Asterisk TAPI - Outlook click2dial
1:52PM 0 FW: Voicemail problem, not playing back
1:36PM 1 Re: Voicemail problem, not playing back
1:31PM 8 Paging and Auto Answer on Grandstream GXP2000
12:22PM 5 sangoma A102 installation question
10:44AM 0 TE410P <=> Dialogic D/240SC-T1
10:03AM 5 Asterisk and Brooktrout TR1000
9:24AM 1 anyone doing voice audio detect VAD on analog lines
9:19AM 6 'extensions reload' clears Regextens
9:18AM 1 Voicemail problem, not playing back audio
8:44AM 2 budget tone 100
8:32AM 16 Dialling a DUNDi Route
6:45AM 0 onsite tech for N Carolina and Boston
6:43AM 0 Directory by name access inside of voicemail
5:50AM 8 Call parking from legacy PBX over PRI??
5:16AM 1 TigerNetwork IPH202A/B are OK ?
5:01AM 0 tdm400p card for sell (4xFXS)
2:51AM 3 Supervised Transfer how to do?
1:03AM 2 Eicon Diva Server - Fax and data modem support
12:02AM 1 mISDN trouble with a HFC Cologne card, Asterisk Asterisk 1.2.4 on Linux 2.6.16.11 - incoming DTMF detection
 
Wednesday May 10 2006
TimeRepliesSubject
10:53PM 0 Sip jitter buffer patch + Asterisk CallingCard
9:44PM 1 difference betwen a TE411P and TE410P
9:25PM 0 Sharing an outside line between a modem and a TDM400
8:53PM 5 CentOS 4.x and ooh323
6:54PM 2 REPOST: features.conf *1 Call Recording
5:42PM 1 mg3000-r fxo gateway provides more feature to work with asterisk
5:28PM 1 asterisk -rx 'sip show peers'
5:06PM 5 ISDN, TE205P, I'm goind crazy :>
3:01PM 1 ISDN Bridging with Bristuff
2:16PM 2 Headsets
11:22AM 0 QSIG suopprt in Asterisk
10:04AM 3 Web Admin
9:49AM 2 Is there a way to not propagate a context included inside other context?
9:47AM 5 One sided call
8:00AM 8 ethernet interface shares interrupts with tdm card
7:39AM 0 OH323 vs Panasonic IP Hybrid
6:52AM 1 Dropping Number on Dial Out
6:18AM 19 features.conf *1 Call Recording
6:05AM 0 Hints and busy lamps for phones registered on SER
5:32AM 2 No zap/sip/etc options?
4:59AM 3 ISDN and Asterisk
4:49AM 0 No audio in either direction on Zap -> SIP or SIP -> Zap calls
3:05AM 0 Realtime extension
1:39AM 3 asterisk monitoring / res_snmp
 
Tuesday May 9 2006
TimeRepliesSubject
11:53PM 3 How do I monitor the whole conversation on a Zap channel ...
10:52PM 0 MCC 1.4 released
10:16PM 0 How to make calls to US using Asterisk?
9:12PM 4 exten statement execution order
7:46PM 1 FW: Solid-PBX
6:36PM 0 asterisk and NEC SV7000S playing together?
6:34PM 9 Announcement: FOP 0.26 released
5:57PM 0 DID -> SER -> Asterisk call transfer
3:45PM 0 problem with hang up with TDM31B
3:29PM 7 PRI in Shanghai China
2:58PM 8 Caller ID forwarding
2:57PM 0 Cisco 2851 as T1 Gateway and Asterisk
1:25PM 0 Intellitouch ITC-3002 2line phones are ok?
12:39PM 20 voipjet down?
12:36PM 0 soft phone code
12:31PM 1 Sip and dbsecret
12:23PM 4 Configuring utstarcom1000 on asterisk
11:42AM 3 Incoming SIP or IAX2 via NAT
11:38AM 4 PSTN Incoming call on real line disrupts VoIPcall over DSL circuit - EXPLAINED
11:36AM 2 TE411P or TE410P
10:28AM 1 Call recordings management
10:13AM 0 How to distinguish between UNEXISTENT channels v/s UNAVAILABLE channels
10:02AM 1 Asterisk 1.2.7.1 and SIP registration
9:51AM 0 Best CPU (of expansion hardware?) for g.729 enc/dec ?
9:33AM 2 Many music on hold files
8:36AM 3 H323 calls will not stay connected
7:58AM 5 Asterisk on EM64T
7:37AM 1 Asterisk settings Net2Phone
7:32AM 0 Re: poor state of IAX2 code? (was: why a per fectlyfine iax2 host becomes UNREACHABLE?)
7:25AM 4 Transferring calls between two Asterisk Servers
6:29AM 0 SciTel Brix-QE card
6:17AM 3 Shared call recordings with ARI!
5:46AM 8 grandstream GXV-3000
5:31AM 19 Bristuffed Asterisk: Hangup problems
4:56AM 0 Using ChanIsAvail and SIP
4:29AM 3 regarding freepbx
4:14AM 1 A@H Memory Limits
2:24AM 1 Asterisk Realtime with Oracle
2:24AM 5 EICON DIVA - which linux kernel
1:49AM 4 [SOLUTION] DTMF Tones occures in Asterisk
1:39AM 6 Problems with TDM400P and FXO modules
1:17AM 4 Best way to intercept an incoming call on asterisk 1.2 ?
12:48AM 0 Billing when forwarding incomming calls from SIP phone
 
Monday May 8 2006
TimeRepliesSubject
9:16PM 4 Asterisk documentation..
6:58PM 0 MINNESOTA: TwinCities Asterisk Users Group - Saturday 05/13/2006 11:30am
5:08PM 0 Local Los Angeles VOIP equipment retailers?
3:17PM 1 MeetMe, async password requirements...
1:12PM 2 *.conf utilities for Asterisk
11:42AM 28 PSTN Incoming call on real line disrupts VoIP call over DSL circuit
11:40AM 4 transfer variables
11:19AM 1 Non-supervised pass-through
11:03AM 5 Asterisk/Zaptel 64-bit?
10:31AM 0 Looking for New Service Provider
10:27AM 2 Message on Hold
10:06AM 3 Most comprehensive management?
9:36AM 1 Running down an echo problem on outgoing calls
9:26AM 0 I: Dialstatus results
8:22AM 3 Expansion module
8:05AM 1 Dialing status detection
7:43AM 0 duration / billsec problem
7:29AM 0 AstLinux 0.4 Released - with build system
7:17AM 1 UpState NY SIP provider
7:16AM 2 How do I monitor a Zap channel ...
6:47AM 10 MySQL replication for voicemail
6:12AM 2 Dialstatus results
5:58AM 2 Quad ISDN card
5:50AM 0 (no subject)
4:50AM 4 Voicemail bomb
4:47AM 9 app_wakeme.c (Wake-up Call Manager) v0.1.0 released
4:20AM 4 [nativecode=Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (111)]
2:08AM 5 Junghanns GSM card
2:03AM 0 Asterisk 1.2.x with app_rxfax
12:29AM 0 gxp-2000 Asterisk PSTN
12:05AM 0 iax2: dropping too many packets
 
Sunday May 7 2006
TimeRepliesSubject
11:14PM 0 Session Border Controller (SBC)
11:03PM 10 CallerID retain on internal transfer
5:05PM 0 Chanspy Specifying Agent not Working
3:40PM 2 SSH from System() ?...
1:46PM 2 Voicemail indication for analog phones
12:46PM 2 Need a Service that allows me to call Toll Free Outbound numbers
8:56AM 1 Announcement Haiku
8:39AM 0 canreinvite=no and codecs.
3:23AM 0 app_rxfax problem on 1.2.6
2:41AM 0 more one asterisk hardware
1:58AM 2 another question about hardware for using with asterisk
1:54AM 1 Assterisk prompts
1:09AM 0 [Fwd: Re: asterisk hardware]
 
Saturday May 6 2006
TimeRepliesSubject
3:54PM 1 Upgrade SVN failed !!!
1:26PM 10 Voicemail error
12:03PM 3 www.SavaJe.com
11:07AM 3 Register Asterisk to FWD via SIP
8:32AM 0 Sipura register with FWD every 60sec
4:42AM 19 TDM4xxP
2:56AM 0 Gigabit Ethernet with multiple VLAN's or Fast Ehternet and with two separate cards?
 
Friday May 5 2006
TimeRepliesSubject
6:47PM 1 Multiple periodic announcements in queues? Possible?
6:07PM 0 CW options not changing
5:35PM 4 How to determine if a device is in a call
4:44PM 3 Info
4:29PM 5 Silent Attendant
4:06PM 5 ASTERISK DISA FOR INCOMING DID CALL
4:02PM 0 REGISTER that isn't a register
3:34PM 0 Passing Callerid
3:24PM 0 ODBC Voicemail storage and app_directory
3:03PM 0 asterisk behind load-balancing switch
2:48PM 1 Bandwidth via my Asterisk PBX
2:38PM 1 Asterisk <--> NAT <--> Internet <--> NAT <--> Sipura-3K (No Asterisk)
2:31PM 0 300 DID's required in Alpine Texas Area code 432
2:06PM 8 Code parsing error?
12:46PM 0 Passing SIP Subscriptions???
12:40PM 45 Call Center Phone with Auto Answer
11:19AM 0 AASTRA 9133i and PIX Firewall
10:43AM 1 Spam? Re: Cisco 7970 running SIP question
8:55AM 0 Spam? Re: Cisco 7970 running SIP question
8:18AM 2 Cisco 7970 running SIP question
8:16AM 0 Problem on Zap Channel with IVR
8:07AM 0 Access to sip.conf username field from dialplan
7:47AM 0 asterisk 1.2 & hisax teles 16.3 isa
7:34AM 1 Realtime, 2 server setup problem?
7:24AM 0 CARD.XML for MGCP cisco phone
7:21AM 0 Call Transfer Disconnect (CT-5)
6:58AM 1 problem g729
6:57AM 6 Dumping queue_log to MySQL
6:31AM 3 Registering Remote Sipura to Asterisk (both behind firewall)
6:17AM 0 Repost: External voicemail and MWI on internal phone
6:16AM 0 Call Hold and Retrieve
4:56AM 0 Re: Asterisk-Users Digest, Vol 22, Issue 26
4:23AM 2 AW: AW: DTMF detection when outgoing call tomobilephones
1:36AM 0 DTMF Tones within my Asterisk on all type of Channels
 
Thursday May 4 2006
TimeRepliesSubject
11:17PM 2 Asterisk on amd SERVER
9:53PM 0 Is FWD down ???
7:26PM 0 SPA941 et al LED indications
3:41PM 0 asterisk can't find address host. Problem in chan_sip.c
3:13PM 1 Fwd: meetme conference latency degrades...
2:33PM 6 why a perfectly fine iax2 host becomes UNREA CHABLE?
2:21PM 0 asterisk <-> SIP provider, two way connection
1:48PM 0 Voicemail records funny - Asterisk 1.2.7.1
1:14PM 2 Help with IRQ conflict between wct2xxp and eth0
1:12PM 2 Unable to get TDM400p working
12:51PM 3 Volume configuration on Polycom Soundpoint 501 phone
12:03PM 1 Switchboard solutions, interactions with handset
12:02PM 0 Realtime rtignoreexpire bugged ??
11:45AM 8 Tool for Polycom configurations
11:23AM 0 TE410P & T400P together in a server
11:14AM 0 Soonr
11:11AM 0 OT: D-link DI-102
9:19AM 0 remapping sof-keys on Polcyom 301
8:31AM 23 why a perfectly fine iax2 host becomes UNREACHABLE?
8:11AM 0 disa and caller id
8:06AM 4 AW: DTMF detection when outgoing call to mobilephones
7:10AM 2 DTMF detection when outgoing call to mobile phones
6:46AM 6 SV: Polycom 501 - Disable DND feature?
5:50AM 0 SpeedDial on GXP-2000
4:41AM 3 PCI voltage
4:35AM 2 Unwanted conference with snom320 and asterisk 1.07bristuffed
3:45AM 1 Meetme from MySQL
3:39AM 0 Internet exposed asterisk server.
3:34AM 6 number that starts with star on PAP2
3:27AM 1 Pattern matching DISA
3:19AM 5 ISAC support?
2:16AM 0 AW: SIP Phones behind dynamic IPs
1:59AM 0 SetGroup and CheckGroup. Need some help on the dialplan
1:53AM 5 SPA941 SPA942 BUG. auto answer does not work.
1:09AM 0 Using console channel with specific codec only
1:02AM 0 Unwanted conference with snom320 and asterisk 1.07 bristuffed
12:40AM 3 TDM400P and monoBRI auto-dial call difference: caller phone does not ring
12:22AM 1 Polycom 501 - Disable DND feature?
 
Wednesday May 3 2006
TimeRepliesSubject
10:55PM 0 Extension '' in context 'whatever' from '123456789' does not exist.
6:57PM 0 Running applications when a queued callisanswered
6:48PM 4 meetme conference latency degrades...
5:08PM 1 Running applications when a queued call isanswered
4:28PM 0 Vodini & *
4:17PM 4 hyperthreading and zaptel
2:30PM 3 Setting QUEUE_PRIO
2:07PM 2 dialing FXO gives wrong billsec
1:55PM 2 How would you go about calling a list of numbers and 'speaking' a message?
12:14PM 0 SIP w/NAT on Grandstream 496 and Call-Waiting
11:50AM 1 my asterisk crashed
11:43AM 1 echo in Snom 360 phones
10:58AM 0 RE: [asterisk-biz] Colocation Denmark
10:46AM 0 Colocation Denmark
9:46AM 0 Forwarded Numbers and Timeouts
9:41AM 0 Selecting the outbound port from FXO device
9:22AM 5 Voipjet Problem?
9:15AM 4 SIP Phones behind dynamic IPs
9:08AM 0 G.722 Softphone?
8:45AM 7 QSIG support in Asterisk
8:04AM 2 Running applications when a queued call is answered
7:58AM 0 Listening a conversation
7:36AM 1 LDAPget
6:32AM 22 Can I recreate a Fax from a recorded file?
5:45AM 0 Limit on number of SIP channels?
5:10AM 2 Simple Dell Computers
4:43AM 0 Phone UNREACHABLE: Plays "agent-incorrect" to Queue-caller ??
3:19AM 2 asterisk intergration in third party web application
3:03AM 0 mysql failures handling
2:16AM 0 Future pickup feature
2:12AM 0 Which distro for Intel D915GAG-L ?
2:02AM 0 Asterisk SRPMs and patches
2:00AM 2 brittle IAX connections ?
1:08AM 3 Huawei EP201S
1:07AM 0 Asterisk auto-dial out: behaviour difference between analog and ISDN channel
12:58AM 0 GXP2000 provisioning: what is cfg.txt file?
12:52AM 0 Can't compile ael_lex.c on HEAD
 
Tuesday May 2 2006
TimeRepliesSubject
11:24PM 1 SV: How does asterisk behave when multiple phonesare logged in on a single SIP/account?
10:32PM 1 Unicall MFC/R2 B3,B4 and clear back
8:54PM 0 OT - but relevant
8:35PM 0 Asterisk Imposter binary
8:17PM 5 Queue reporting seems broken.
8:17PM 0 asterisk hung again
7:42PM 0 Grandstream GXP-2000 call end
6:12PM 2 PAP2/Sipura XML Provisioning File
5:47PM 0 Insights on SIP channel usage in * 1.2.7.1 are welcome!
2:40PM 0 Half hangup issue
1:40PM 0 PRI Transfer Disconnect
1:30PM 8 Sangoma Card Question
1:28PM 0 Help with multiple company setup
1:14PM 4 Asterisk technician needed in Buenos Aires Argentina
1:13PM 0 Ringing extensions in a call group.
12:00PM 0 The CAVP is now accepting memberships applications
11:11AM 0 Using qualify=yes guarantees failure on iax2 behind NAT (was: RE: Using frequent keepalives to eliminate needforNAT port forwarding?)
9:26AM 0 Telasip config problem/question
9:02AM 0 Need help configuring TE100P and 3 X100P clonewith MD3200 chipset
8:36AM 1 Need help configuring TE100P and 3 X100Pclonewith MD3200 chipset
8:35AM 0 Commands possible in the h extension, message delivery with confirmation
8:35AM 2 Speeding up UK BT incoming call detection
8:17AM 5 Sip show inuse
7:47AM 2 dnd error message in the log
7:38AM 3 Need help configuring TE100P and 3 X100P clone with MD3200 chipset
6:54AM 2 Need help in asterisk fax
6:32AM 1 SIP trunk ring tone
5:17AM 29 Zapata Telephony interface and torisa module error
3:22AM 0 MeetAsterisk London and Brussels
2:19AM 2 Questions on ANI
2:08AM 3 asterisk with Dialogic BRI /2VFD
12:44AM 4 Under which project , auto-dial feature comes
12:09AM 1 Meetme volume increase/decrease
12:00AM 0 2 process running concurrent in dialplan
 
Monday May 1 2006
TimeRepliesSubject
11:57PM 0 Re: 482 Loop Detected on sip calls
11:56PM 2 How does asterisk behave when multiple phones are logged in on a single SIP/account?
11:40PM 1 /var/spool/asterisk/outgoing/ prematurely hangingup
10:43PM 5 /var/spool/asterisk/outgoing/ prematurely hanging up
10:37PM 1 unable to set outgoing callerid
5:35PM 1 Using frequent keepalives to eliminate need forNAT port forwarding?
5:25PM 2 Using frequent keepalives to eliminate need for NAT port forwarding?
4:14PM 0 Cisco 2621 router for voice and data?
4:12PM 0 wellgate 38XX with VAD and call files
4:00PM 4 SPA-1001 behind NAT -- mucho hair pulling
2:49PM 3 Music on Hold from Soundcard
2:24PM 1 Polycom SoundPoint 501 + Asterisk
2:19PM 0 Spam? Re: CallerID Name problem
2:06PM 0 app_icd
1:45PM 0 Asterisk-Users Digest, Vol 22, Issue 1
1:16PM 6 Digium TDM400P vs Sangoma A200 for 2 x FXO
10:51AM 1 Listening on one IP and binding to other IP - is this possible ?
10:41AM 7 Problems with zaptel and TE210P
10:12AM 0 Can i use same group with 2 or more hfc-cards ?
9:45AM 0 7941G - Any success stories?
9:36AM 14 CallerID Name problem
9:24AM 11 voicemail dialout
9:17AM 3 auto-dail for ZAP channel, the application gets executed before the call attended
9:00AM 6 Softphone ready to go installed on USB flash drive
6:38AM 1 GXP-2000 Message Waiting Light
6:32AM 0 Sangoma A200 preventing Zap channels
6:05AM 6 Cant get voicemail
5:54AM 1 Is there a way to monitor DTMF tones in a channel?
5:48AM 0 anyone have solution to dtmf problem in console driver?
5:30AM 1 Frappr mapper
5:24AM 0 Asterisk Bugs?
4:07AM 1 Cepstral , options to read the contents of a file
3:48AM 1 Auto-Dial , problem in calling Application , Guidance requested
3:40AM 1 Anyone willing to share an Australian dialplan.xml file for Cisco phones?