asterisk users - Sep 2007

Sunday September 30 2007
2:15PM 0 Asterisk Dropping Calls (Richard Young)
1:19PM 0 Asterisk 1.4, h.323, OpenCom 1010
9:34AM 1 Selecting a specific line from Zap/g
Saturday September 29 2007
9:06PM 0 Big problems with TDM2400 :(
12:56PM 3 FAX detection not working
9:52AM 1 VoIP, Asterisk, What Do you really need ?
1:22AM 3 meetme conference using g729?
Friday September 28 2007
6:53PM 1 odd audio problem
6:36PM 0 Nano syntax highlighting.
6:02PM 0 Proximity detection versus GSM receiver
5:57PM 1 Proximity Detection: Motorola Q + Bluetooth + Asterisk
4:28PM 2 Changing contexts "on the fly"
2:41PM 0 VoIP, What Do you really need ?
2:39PM 1 Recommend Digium Hardware?
1:40PM 1 Non-USASCII chars in sip.conf?
1:13PM 1 Multiple Meet me conferences
12:55PM 4 . (period): Wildcard match; matches one or more characters
12:52PM 1 How can I know if I wrote the configuration like correctly
11:57AM 0 Proper trunk to connect two systems.
8:57AM 1 Ringing Groups, SIP Forward and looping problem
7:16AM 0 Conference call today at 12:30 PM EDT
3:39AM 1 call relation in call transfer
12:58AM 0 Asterisk Appliance with VoIPStreet
Thursday September 27 2007
10:30PM 1 Polycom 501 won't take new bootrom.ld or sip.ld...
10:27PM 0 Problems Connecting Two Asterisk Installs ViaISDN PRI Cards
7:56PM 0 Timeout issues
7:07PM 1 Zap channel stuck in conference
5:57PM 0 SIP interface status
4:48PM 0 TE120p and music on hold
4:03PM 15 Cisco 7940G licensing with asterisk
3:48PM 5 Which Asterisk version to use?
3:25PM 3 Digium acquires Switchvox
2:23PM 2 IAX configuration
12:45PM 0 astcc sometimes doesnt write on mysql
12:26PM 0 ADIT TDM T1 <> Asterisk MGCP
11:24AM 3 Music On Hold - How to increase volume ?
10:34AM 4 3-way calling
3:56AM 1 help with channelbank audiocodes MP-124
3:35AM 0 Doesn't seem to want to transcode.
3:05AM 0 voip hacking article
12:35AM 0 h.323 out of media path
Wednesday September 26 2007
11:39PM 1 asterisk audits
11:38PM 4 Asterisk realtime error
8:48PM 2 ChanSpy issue
8:05PM 0 Networking Question
7:54PM 3 How to "busy out" zap channels
7:49PM 2 Ast_log
4:10PM 1 IAX gsm bandwith calls
3:58PM 1 Routing issue
3:45PM 1 Asterisk - Spandsp Fax not working?
3:31PM 1 faster timeout in ENUMLOOKUP() function
3:04PM 1 Manager Originate Action and Cancel
1:36PM 1 Supermicro PDSME+ and TE110P [ ref:00D36mPe.50033qy57:ref ] NEW CASE 22828
1:31PM 1 chan_sip falls over with undefined symbol ast_pickup_ext
1:24PM 2 My G729 problem re-visited
12:48PM 0 Grandstream HT 502 ATA stops receiving calls
12:08PM 0 configuration of wanpipe for asterisk.
10:42AM 1 Busy problem
9:08AM 1 DTMF signalling, SIP, and Background()
7:07AM 4 Music On Hold
4:53AM 2 SIP Panel?
4:17AM 0 Grandstream GXW-4008
2:25AM 0 asterisk-users Digest, Vol 38, Issue 83
Tuesday September 25 2007
10:37PM 1 Configure one call per line on Cisco 7941/7961
9:19PM 1 Help with Sip Registration
8:40PM 1 Dutch Number for Inbound
6:34PM 2 Point-to-Point SIP link without registration
2:57PM 2 swift.conf - cepstral voice quality adjustment options
2:41PM 1 Multiple Home system with SIP
2:32PM 2 show queue (queue name)
2:22PM 3 Zaptel- Compile Error
12:55PM 5 Do I need to run #modprobe zaptel for Digium
9:27AM 9 Asterisk Redundancy
9:14AM 4 Anyone else having problems with the list
9:09AM 1 Backuping VoIP provider with PRI
8:59AM 4 Grandstream GXP2020 / 2000
8:35AM 0 [EVENT] Asterisk and VoIP in enterprise
7:54AM 2 HOWTO/FAQ question (Location: Sweden)
6:44AM 2 running twice
5:37AM 1 Completing my Configuration
3:52AM 0 Analog Telephone Adapter
2:36AM 1 ExternNotify Voicemail
12:49AM 2 Yikes! Polycom 501 chokes on BootRom 4.0.0?
Monday September 24 2007
11:38PM 0 CID spill after second ring
10:36PM 0 Extensions Configuration
9:16PM 0 Asterisk with multi-line appearence? How?
9:14PM 0 Asterisk as Media Server
8:57PM 0 TDM2400 answer detection
8:56PM 1 DTMF dropping digits
6:36PM 0 Spur error with Siemens Hi Path
6:31PM 3 CallerID problem Asterisk 1.4.2
3:22PM 0 Adding allowed codecs to Asterisk
2:57PM 0 Anyone use the Linksys phones? (Zeeshan Zakaria)
2:18PM 2 Asterisk 1.4.12 Release?
11:30AM 0 asterisk crash
11:29AM 0 Asterisk Dropping Calls
11:28AM 3 Asterisk and OCS integration
9:38AM 2 Virtual server Solution
9:07AM 2 Sangoma or digium ?
7:21AM 1 # to transfer calls
6:12AM 0 asterisk canreinvite option questions
5:59AM 0 Call hangup after 60seconds
1:04AM 2 asterisk cli - vi keybindings ?
12:25AM 1 Help with log entries.
Sunday September 23 2007
5:13PM 5 Anyone use the Linksys phones?
Saturday September 22 2007
8:40PM 0 Looking for a reliable source for DIDs from Mexico, Colombia and Venezuela
5:48PM 1 Looking for recommendations on Nufone and Gamachi
3:51PM 2 error messages related to mysql in asterisk CLI
2:44PM 2 errors messages in asterisk CLI
10:47AM 0 HITBSecConf2007 - Malaysia Materials & Photos are up !
4:34AM 1 prepaid application recommendation
3:07AM 2 Realtime table columns
Friday September 21 2007
9:36PM 4 Polycom 501 Phones Rebooting
8:44PM 0 Confused about Asterisk 1.4 RTPQOS...
7:06PM 1 SIP and Firewall
7:03PM 1 UNICALL MFC/R2 + Asterisk 1.4
4:07PM 1 Authenticate() application and CDR
3:18PM 3 SIP over TCP
2:18PM 3 Asterisk 1.2.13 and presence
1:21PM 0 "HiarPinning" via TDM400 in the UK ...
1:08PM 1 call limit
12:55PM 3 Asterisk and MS Exchange 2007
12:49PM 0 Problems bringing up ZAP trunks via PRI
6:24AM 0 Voxalot User and Peer details.
3:22AM 1 Dialing an external number and then passing it to an extension...
Thursday September 20 2007
10:31PM 4 Asterisk 1.2.24 simultaneous call limits.
10:18PM 1 GROUP() issues for me
7:31PM 1 Polycom 330 + Asterisk, phone locks up. * key will do it
7:23PM 0 Astricon Ride From Airport to Conf Hotel
5:48PM 10 IAX Java Softphone?
5:20PM 9 Problems Connecting Two Asterisk Installs Via ISDN PRI Cards
4:52PM 1 Queue Question
3:14PM 2 The device state is still 'Not in Use' ... check UPGRADE.txt
2:30PM 2 Outgoing SIP packets out of order?
12:33PM 0 Ghost calls from phones
10:49AM 5 Horrible problem - calls losing sound
8:48AM 0 loop detected
8:34AM 1 asterisk crash and core dump:
6:30AM 4 Newcomer Question
3:32AM 0 Video doesn't work for outgoing call?
1:57AM 1 OT: Samsung Sprint CDMAoIP
Wednesday September 19 2007
9:54PM 1 Building an RPM from Asterisk 1.4
9:43PM 2 AMI extension states
9:33PM 2 Asterisk on Fedora Core 4
8:48PM 1 crash after callbackagent ackcall
6:11PM 1 Short Audio Drop Out During Calls
5:17PM 3 Dial() Command Parameter L Overflow?
3:56PM 1 Freeswitch Vs Asterisk
3:54PM 0 How many SIP phone connect with asterisk
3:41PM 1 How to cancel the password check in VoicemailMain()
3:20PM 2 Hfcmulti and B410P Digium Card
3:02PM 2 what is softswitch
1:01PM 2 Supermicro PDSME+ and TE110P
12:37PM 1 dtmf issues on PRI and 1.4.11
12:25PM 0 Configuring Loose routing method
11:24AM 0 Vedio confrancing with asterisk
10:31AM 2 Multi-sip rings
10:20AM 1 off-topic: Avaya 46xx, release 032207 ... help
10:03AM 1 Queue serializes call delivery ?
9:51AM 0 Howto pickup call from queue?
8:50AM 0 RTP Read too short with T.38
8:43AM 1 AgentCalbackLogin not loging in race condition ?
8:40AM 0 openser/ser/Asterisk user meeting (beer drinking in Vienna)
5:43AM 0 asterisk directory dialing
5:32AM 3 VoIP Provider for business
3:11AM 18 sip.conf best practices?
Tuesday September 18 2007
8:26PM 6 Limiting Simultaneous calls
8:19PM 3 Comfort noice sample (gsm/mp3)
8:03PM 4 Linux limits
4:41PM 2 ISDN PRI debug in Asterisk
4:27PM 1 Queue agents w/ DUNDi
4:07PM 2 ISDN data packets
3:55PM 1 Dell Power Edge 1900
3:53PM 3 Interesting Conference Request - Can this be done ?
1:27PM 0 T1/PRI pricing
1:01PM 0 Bug labs
12:15PM 2 asterisk crash and core dump
10:44AM 1 stanaphone issues. can someone verify my config?
8:20AM 2 Randomly half-voice at sip/zap
6:59AM 0 Issue with Asterisk realtime
12:56AM 1 Asterisk 1.4 and Cepstral
12:27AM 1 Chan_SCCP vs. Chan_Skinny
Monday September 17 2007
10:34PM 0 Asterisk and HPC Cluster
10:19PM 3 Enabling MySQL UNIQUE from cdr.conf
7:50PM 1 Softphone RTP Session Start-up Delay
6:58PM 7 Why does everyone seem to dislike *now?
5:06PM 2 Call Center SoftPhone with Auto Answer
4:05PM 0 analog topex and Digium cards (maybe OT??)
2:08PM 1 RTP Call Disconnect
12:09PM 2 Filesharing + video + voice supported Soft phone
10:32AM 1 Problem with asterisk-perl-0.08 and Asterisk >= 1.2.20
7:50AM 0 [SOLVED] fax machine detection for outgoingcall on DIVAcard
6:29AM 1 extensions for conference call
6:09AM 0 Asterisk 1.4.11 over NAT
5:02AM 1 Wondering why I can't post
3:33AM 3 Voicemail.conf
Sunday September 16 2007
2:24PM 0 Problem with asterisk 1.4.11 and playing files to meetme conference
9:08AM 0 Replacing an SPA 3000
Saturday September 15 2007
6:18PM 2 Astribank and caller ID from PSTN
4:21PM 2 AGI/PHP: missing arguments
1:55PM 1 External FXO port.
Friday September 14 2007
11:04PM 0 Zaptel ztdummy module causes playback to fail
9:28PM 0 3 way Calling
9:00PM 2 AGI script fails on IAX channels (from call file).
8:38PM 6 Force a new user to configure Comedian mail?
8:13PM 1 g729 on
7:05PM 3 ztdummy kills audio
5:58PM 1 MOH Files Volume
5:23PM 4 how to route outgoing calls on IP-level
4:53PM 0 AstLinux 0.4.8 Released
4:49PM 2 Prompt for extension with standard dial-tone.
4:39PM 4 Can Asterisk match a literal "*" in extensions.conf
3:52PM 1 Help Drop Calls
2:10PM 2 DISA and DTMF detection problem w/ FXO port on a TDM400
1:55PM 0 AsteriskNOW + legacy PBX integration
1:02PM 2 outgoing call restriction in extention.conf
12:45PM 1 Mutipoint Conferencing?
12:15PM 1 [Serusers] user meeting (beer drinking in Vienna)
5:56AM 0 VOIP Users Conference Friday September 14 @ 12:30PM EDT
5:47AM 1 Asterisk voice quality tuning
1:07AM 1 Include zaptel in kernel...
12:05AM 6 DECT SIP phones
Thursday September 13 2007
9:10PM 3 Voicemail in 1.4?
8:56PM 1 FreePBX (2.3) - Good? Bad? Ugly?
8:55PM 2 TDM400P
7:34PM 1 TCP connection to AMI broken after 15 minutes
7:32PM 5 CallWithUs Service?
6:31PM 0 ZAP to invalid SIP device call looping
6:30PM 0 Very fast playback
6:26PM 0 problem
5:32PM 2 Paging to external speaker like in airports etc...
5:18PM 1 Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX
3:32PM 1 how to disable wctdm auto modprobe during boot
2:16PM 2 DTMF error on asterisk
1:51PM 1 SMS in France - allways get "NAK"
1:45PM 2 Fwd: Bad FCS error
1:30PM 1 how to determine if a SIP extension has DND on or off
11:11AM 0 [phpAGI] generate a call from a SIP device to Asterisk
10:19AM 3 Asterisk cli
10:03AM 2 ztdummy problem in fedora7, kernel
9:08AM 1 fax machine detection for outgoing call on DIVA card
7:13AM 0 Support of simple E1 CAS signaling (MFCR-like)
6:37AM 0 asterisk call back dail plan
6:24AM 0 Licensing and provisionning
4:33AM 1 call transfer detection in dial plan
3:38AM 2 FW: Problems with two trunks
3:18AM 1 Trunk & Outbound Route for a Cisco VOIP router?
1:08AM 1 Zap channels: no sound with certain call paths
12:44AM 1 Problems with two trunks
Wednesday September 12 2007
10:26PM 0 AsteriskNOW
8:45PM 3 Agent Callback Login in 1.4
8:44PM 2 Conference bridge.
7:45PM 1 Looking for Asterisk Consultant in San Franicsco
7:43PM 0 (no subject)
5:44PM 2 Callback for unanswered transfers...
4:52PM 2 Digium Appliance
3:57PM 2 Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?
3:51PM 1 res_snmp
3:49PM 0 Solution: Sysmaster and Asterisk
2:59PM 1 Astribank 32 and Far End Disconnection
2:55PM 1 Direct dialing to correct extension from analog lines
1:57PM 0 Wanted: VoIP Engineer for Warsaw !
1:23PM 2 Problems with Asterisk behind a firewall
12:21PM 0 fax and answer machine detection for outgoing call on DIVA card
10:46AM 1 TDM2400P: Power alarm error on boot
10:42AM 1 TE405P intermittent yellow alarm
10:01AM 0 AAI2UUI - how?
7:57AM 2 Generating an old-fashioned dialtone
Tuesday September 11 2007
10:06PM 1 IAX2 NAT issues
8:50PM 1 Chan_sip Entry
8:26PM 1 TDM400P periodic sound clicks on FXS
7:17PM 2 bug in 1.2.24
6:31PM 1 TDM400P not answering or making calls
5:02PM 0 SIPAddHeader cmd from Realtime MySQL, not getting all the 'appdata' field
4:50PM 1 Mark Spencer: Digium is Growing Up (VONMAG)
4:18PM 0 dtmfmode rfc2833 and info
3:38PM 1 exit ChanSpy with DTMF
2:32PM 1 Linux-HA and Asterisk
12:56PM 2 Another State Of The Punctuation Mark question - Vonage
10:30AM 3 Prevent multiple sip registrations
9:38AM 4 Installing Asterisk on to CentOS 4
9:04AM 5 Flash IDE
7:22AM 2 Asterisk 1.4.11,, SegFault
6:39AM 1 Asterisk on NGINX Server?
Monday September 10 2007
11:26PM 0 rtptimeout on Asterisk 1.4.x
9:33PM 0 DTMF
9:01PM 5 Asterisk Manager API - Originate command
7:25PM 1 Cisco UC 500
6:27PM 4 Partitioning DSL input
5:28PM 2 Siemans SIP/PSTN phone S450
3:12PM 2 Failover SIP logic
10:18AM 1 56k modem configuration
9:09AM 1 USA Termination
8:04AM 1 New Project: AskoziaPBX
6:37AM 5 online active call watching
5:07AM 5 Connecting Legacy Pbx With Asterisk With FXS.
Sunday September 9 2007
10:35PM 1 SIP endpoint that does not register
10:31PM 3 canreinvite
10:28PM 3 nat=yes
10:26PM 2 What is the difference between increasing the verbose level and the debug level?
10:14PM 1 Maximum retries exceeded on transmission
7:35PM 1 DTMF bug in dsp.c and 1.4.11
6:16PM 4 Strange Behaviour
5:28PM 1 Which cause less CPU usage: GSM or wav??
3:45PM 1 Softkeys wrong with chan_skinny
1:55PM 0 what is the usable feature in DUNDi?
1:40PM 0 [mythtv-users] Real Time Clock Alarm Broken with 2.6.22+ kernel
2:50AM 1 Difference in show channels
12:32AM 2 Asterisk on Ubuntu Feisty
Saturday September 8 2007
3:22PM 0 Forgotten SIP session
8:52AM 1 redendent asterisk server for backup
7:58AM 3 Configure extension by software
12:02AM 1 Musiconhold instead ringing
Friday September 7 2007
11:41PM 0 SIP INFO request in asterisk
9:03PM 0 voice to text
8:52PM 0 chan_misdn
6:31PM 3 T1 to SIP conversion, standalone device
5:44PM 1 Asterisk + Realtime + Manager reload = crash
5:21PM 2 Meridian S1 to Asterisk via T1
4:03PM 1 Channels in use?
3:52PM 3 Show Callee name on Display
3:45PM 1 New Installed X100p
3:21PM 1 Manager connection timeout
2:29PM 0 Errors: Too many SIP headers and Unknown SDP media type in offer: video 10702 RTP/AVP 34 31
1:15PM 0 Sysmaster and Asterisk
1:08PM 1 queue static agents
12:56PM 1 Broken UDP streams
3:39AM 1 how to DUNDi branch office with area code?
1:42AM 0 MINNESOTA: TwinCities Asterisk Users Group Meeting - This Saturday Sep 8th, 2007 (Only hours away)
1:08AM 0 Connecting Asterisk to Alcatel OmniPCX
Thursday September 6 2007
8:15PM 1 Random Double Digits
7:44PM 0 Inbound SIP issues
7:05PM 6 Build your own "appliance" concept
6:44PM 1 Cascading queues & calls not joining unavailable queues.
5:43PM 2 Register Extension
5:36PM 1 Dead SIP channels
5:16PM 0 DTMF Problem with International Calls
4:57PM 3 Skype + Asterisk
3:49PM 0 Help needed - ISDN is "redialling"
3:38PM 0 Digium Innovation Awards
3:35PM 1 Asterisk Users Conference Friday @ 12:30PM EDT
3:05PM 2 Different Networks
2:46PM 0 (txfax+spandsp) fax is successfully sent, but Asterisk keeps sending DelayedRetries.
1:58PM 7 SIP Debugging to separate log file
1:55PM 0 31 seconds because it is directly bridged to another RTP stream
11:58AM 3 Multitenant or Multiple virtual machines
11:30AM 0 Asterisk on UML (User Mode Linux)
9:00AM 2 bridge on DIVA card and how to see it
8:07AM 2 asterisk voicemail to email and relaying
5:58AM 2 FAX machine connect with audiocode SIP device
5:28AM 1 Choppy sound while converting alaw to ulaw
4:46AM 2 alphabetical extension patterns
1:46AM 0 Asterisk 1.4 Ignoring SIP ACK's on 487 Responses
12:18AM 1 14. Re: ztcfg error : TE110p error with " CAS signalling on span1 conflicts with HDLC with ... (Carlos Chavez)
Wednesday September 5 2007
7:04PM 0 Benchmark
5:32PM 7 Can asterisk give half-ring periodically for MWI?
4:44PM 4 special kind of billing
4:34PM 1 Asterisk + LDAP or RADIUS
4:32PM 2 DTMF Relay Problems
4:20PM 0 ANNOUNCEMENT: Asterisk-Java 0.3.1 released
3:09PM 1 Dialplan regexp
3:05PM 0 Presentation and mISDN
2:56PM 4 ztcfg error : TE110p error with " CAS signalling on span 1 conflicts with HDLC with ...
2:09PM 2 No Dial tone came from fxs modules
1:53PM 1 rxfax() problem - fax signal seems to be ignored
1:31PM 1 Overhead paging over IP
1:13PM 1 Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0
12:23PM 8 Ping
10:23AM 2 TDM400P (TDM22P) and aux power.
6:53AM 1 Issue with calling queues
6:37AM 0 FAX with asterisk
6:35AM 0 outgoing call restriction
4:36AM 3 E1 Line Tapping
3:04AM 0 CID debugging help needed
3:01AM 0 about ChanSpy,thank you!!!
2:58AM 5 How to make call from asterisk?
1:48AM 0 Tone detection while Dialing
Tuesday September 4 2007
11:07PM 6 Overhead paging over IP...
8:39PM 6 Udev issue on zaptel install
4:49PM 2 Asterisk w/MS SQL Server 2005
4:38PM 1 Cisco 79xx XML Apps (was: Re: Cisco Directory Format)
4:23PM 4 Asterisk Died message
1:46PM 1 SIPBroker vs SIPgate
10:30AM 0 NAT-troubles with RTP
6:47AM 11 stop log/debug messages into /var/log/messages
6:41AM 1 Asterisk Manager Interface, reliably monitor NewCall for an extension
4:24AM 1 unsuscribe
3:50AM 1 VSP authentication to incorrect context
2:13AM 1 FW: Account Registration Failed
Monday September 3 2007
6:03PM 3 Manager Originate without phone off hook?
3:23PM 0 Append Extension number sounds to Voice Mail Message?
3:21PM 1 ADIT 600 & CMG <=> Asterisk question
2:49PM 0 Digium 4FXO PCI card or Grandstream 4FXO Gateway?
2:08PM 1 Asterisk with app_RPT question
12:52PM 1 Dificult macro, please advise
12:10PM 1 Setting Callerid with chan_misdn
8:18AM 0 Wanted: VoIP Engineer for Warsaw!
8:14AM 0 Account Registration Failed
6:41AM 0 enter menu
6:25AM 1 unnumbered priorities
12:48AM 0 Grandstream GXW-4104 ???
12:45AM 1 Wireless VOIP Keysets? Recommendations?
Sunday September 2 2007
10:15PM 1 off-hook warning tone
2:03PM 1 How can i send my sip channel 3 to mailbox 2? Please Help!
1:32PM 2 Cisco 7960 or 7960G
Saturday September 1 2007
1:16PM 0 phone as control interface (was 99 bottles of beer)
9:28AM 2 Escape characer for Digit Timeout
7:55AM 0 Asterisk MESSAGE Method for SMS
7:30AM 3 Zaptel modules are being installed in different directory
6:38AM 1 asterisk 1.2 or 1.4 for conference call service
6:35AM 1 A102d sangoma's card and ztdummy
5:59AM 0 OT - How to script softphone installation
1:15AM 0 Problems with Asterisk 1.2.23 and Polycom 601