asterisk users - Jul 2007

Tuesday July 31 2007
TimeRepliesSubject
10:45PM 0 Number disappears when picking up a call
10:35PM 8 Dropouts and echo
10:34PM 1 DTMF integration pana d500
9:24PM 1 not hearing dtmf tones
8:13PM 1 Problems using TE412P and TDM400B in a IBM x3650
7:55PM 6 .call file problem
5:43PM 0 AsteriskNOW and Custom VoIP
5:41PM 0 Repost: Aastra BLF directed pickup: anyone have this pickup-mgernoth file?
5:23PM 4 Connecting GSM Phone to Asterisk Box
5:21PM 0 AstriCon -- Last chance to speak!
5:20PM 2 Wrinkled faxes or missing lines with Hylafax + IAXModem + Asterisk
5:14PM 0 FW: NYC Special Event - Join Us
4:35PM 0 OT: Polycom Directory XML via PHP
3:42PM 2 Zaptel update trouble
3:08PM 1 g729 setup help
2:16PM 3 Queue Time to Speak to Agent Algorithm?
1:48PM 0 ExternalIVR() broken in 1.4.9?
1:26PM 2 Turn off SIP 183 Session Progress in Asterisk 1.4.8
1:21PM 3 Welcome to the "asterisk-users" mailing list (Digest mode)
12:39PM 3 1and1 dedicated servers have been down for a few hours .
12:37PM 0 SIP Refer ... rejected?
9:15AM 3 asterisk on 64-bit?
5:36AM 30 Royalty for On Hold Music ?
5:06AM 2 asterisk or asterisknow
2:53AM 0 Zaptel compiling broken: error: conflicting types for '__kernel_dev_t'
 
Monday July 30 2007
TimeRepliesSubject
10:53PM 0 Asterisk with Speechphone
9:40PM 5 Manager - QueueAdd
9:29PM 11 software bloat - is this really useful to anyone?
8:46PM 0 RTP Session Streaming
8:05PM 0 String truncation problems on FreeBSD Sparc64
6:59PM 0 Questions about SPA3102.
5:30PM 1 MeetMe through DeadAGI has changed to return -1 on Hangup
4:23PM 0 AGI Que "Say Time"
4:10PM 0 Zombie (Masqueraded) Channel CDR Problem
4:00PM 5 Silly MeetMe() question.
3:10PM 0 Trouble getting sound from a call
3:09PM 1 Dial plan question: PSTN via Linksys SPA3102 then IAX if busy?
2:46PM 4 Strange ISDN Troubles
2:21PM 1 iax2 trunk registration with auth rsa
1:55PM 1 AGI and exec Playback
1:54PM 2 Creating an SIP softphone
1:45PM 7 Description for each sound files
1:32PM 0 announcement server
1:32PM 0 What is SIP conntrack status ?
1:00PM 4 G729 licenses installed - voicemail has no audio...
12:38PM 1 How to use 1 channel from TE110P for data transmission
12:24PM 9 outbound caller ID
12:01PM 9 Zaptel channel reservation
11:19AM 4 Lightweight IAX balancer
9:40AM 3 TE212 or TE220
8:14AM 1 how to configure zaptel for incoming call
6:34AM 3 Next Friday at 12:30 PM EDT: Asterisk Users Conference TDM inside and outside the box
6:10AM 4 Huntgroup with asterisk feature
6:09AM 0 codian with asterisk voice confrance
5:13AM 0 asterisk 1.4.8 and google talk - no audio
4:19AM 2 Programming with libiax2
4:03AM 1 Queues with logged in agents that are not reachable
 
Sunday July 29 2007
TimeRepliesSubject
2:05PM 2 Dial from Phonebook of Evolution or Thunderbird
3:27AM 0 Asterisk 1.4.X support for Solaris 10?
 
Saturday July 28 2007
TimeRepliesSubject
10:35PM 5 Calling to users in other asterisk servers
3:07PM 0 New York Asterisk Meetup Aug 9th
9:18AM 5 global variables and updates
6:41AM 1 Query2
6:33AM 4 Query1
2:34AM 0 SVP support
 
Friday July 27 2007
TimeRepliesSubject
8:31PM 0 Astricon 2007
6:02PM 7 Nufone problems
5:49PM 2 SIP "Max Channels" Setup
4:09PM 7 Unicall/Dont know how to handle Accepted
3:57PM 5 Locking a device to a codec
3:15PM 2 Will the Sangoma A40003X fit in a 2950?
3:08PM 1 Avaya SIP phones (4610SW) and MWI
1:58PM 0 Asterisk Users Conference Friday at 12:30 PM, EDT
1:46PM 4 CONSOLE=Phone/phone0 and CONSOLE=Console/dsp and Zap/g2
1:33PM 0 Telco Testing locks up asterisk
1:27PM 1 chan_mISDN module does not load
1:26PM 12 Asterisk Wiki
12:26PM 1 Asterisk advanced concepts
12:21PM 1 asterisk meetme confrance problem
10:09AM 2 Queues strategy "leastrecent"
8:40AM 1 ISDN: Problems starting off [another attempt]
8:26AM 0 Keep playing Background while dialling invalid dtmf extensions
8:16AM 2 Asterisk Users Conference Friday at 12:30 PM EDT
7:57AM 0 auto dialout call status
7:06AM 1 Can someone Stop this autoreply stuff?????
6:53AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.
6:50AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?
6:50AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails
6:48AM 0 Please unsubscribe or moderate rp@raha.cc?!?!...
6:47AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE
6:44AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.
6:43AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?
6:43AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails
6:42AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE
6:40AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.
6:40AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?
6:39AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails
6:37AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE
6:37AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.
6:36AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?
6:36AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails
6:32AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE
6:31AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.
6:30AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?
6:29AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails
6:26AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.
6:26AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Re: Display IE
6:26AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?
6:25AM 0 Autoreply: Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails
6:23AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.
6:23AM 0 Autoreply: Autoreply: Autoreply: Re: Display IE
6:22AM 0 Autoreply: Autoreply: Re: Attaching VoiceMails on E-Mails
6:22AM 0 Autoreply: Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?
6:19AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.
6:19AM 0 Autoreply: Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?
6:19AM 0 Autoreply: Autoreply: Re: Display IE
6:14AM 0 Autoreply: Re: Attaching VoiceMails on E-Mails
6:11AM 0 Autoreply: Is it possible transcode (ilbc <-> g.729) in software ?
6:09AM 0 Autoreply: Re: Display IE
6:06AM 0 Autoreply: Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.
6:01AM 0 Is it possible transcode (ilbc <-> g.729) in software ?
5:59AM 0 Autoreply: Autoreply: Autoreply: Re: Dialtone when automatically picking up.
5:56AM 0 Autoreply: Autoreply: Re: Dialtone when automatically picking up.
5:53AM 3 Sangoma on Fedora 7 x86_64
5:52AM 0 Autoreply: Re: Dialtone when automatically picking up.
5:39AM 0 Autoreply: Re: Newbie Advice on Asterisk and Linux
5:33AM 1 Autoreply: Queue stats
5:25AM 0 Autoreply: Re: SunRocket / ALLO / etc special offer
4:36AM 14 polycom custom ring tones (slightly OT)
4:18AM 0 Autoreply: Re: Queue stats
2:31AM 7 Need help with inbound IAX
2:24AM 3 Default Asterisk Numbers
2:23AM 4 Problems with new logic being 'n' option to Queue in 1.4.9
12:55AM 7 Attaching VoiceMails on E-Mails
 
Thursday July 26 2007
TimeRepliesSubject
11:08PM 3 ISDN: Problems starting off
10:17PM 0 (Update) Digium FTP server will be replaced with HTTP server
8:57PM 2 SetCallerPres and Cisco PRI
8:51PM 1 Ring forever
8:20PM 1 vm-duration announcement missing?
8:08PM 3 Newbie Advice on Asterisk and Linux
6:29PM 0 missing digits on PRI
6:23PM 6 Grandstream RTP keepalive packets causing Asterisk warning
5:10PM 3 Digium FTP server will be replaced with HTTP server
2:37PM 11 Queue stats
1:43PM 11 IAX connections broken
1:28PM 3 Asterisk Conference Call
12:37PM 4 Asterisk 1.2.23 and Sangoma a102 no incoming calldetected
12:01PM 0 Queue stats from the dial plan
11:55AM 11 Query
10:13AM 1 tdm400p fxs module busy
6:06AM 0 strange problem in asterisk + mediant2k setup
4:09AM 0 Asterisk 1.4.9 reproducibly dumps core on Solaris 10
4:03AM 1 Asterisk 1.2.23 and Sangoma a102 no incoming call detected
 
Wednesday July 25 2007
TimeRepliesSubject
9:25PM 1 Call report by pinset
9:07PM 0 Problem with asterisk-addons - checking for mysql_init in -lmysqlclient... no
9:05PM 5 Dialtone when automatically picking up.
7:53PM 1 Asterisk-1.2 and Centos 5
7:26PM 0 Asterisk with RFC 3313 support
7:11PM 7 WAV49 output in sox
5:48PM 1 Asterisk Supported Harware Architecture
5:35PM 10 IAX2 INBAND DTMF?
5:20PM 7 Asterisk Vm functionality question
4:50PM 5 X100P pass through questions
4:10PM 3 Asterisk 1.4.9.tar.gz download fails
3:27PM 0 Asterisk Freeze Problem
2:46PM 3 international calls with national rates
2:32PM 1 help with mfcr2 and pri
2:22PM 0 Parking calls via cli / manager / dialplan
1:52PM 2 Add prefix digits in dialplan extention
12:55PM 5 Display IE
12:13PM 5 SunRocket / ALLO / etc special offer
9:37AM 0 Help with SIP to PBX
4:36AM 2 Post voicemail processing.
 
Tuesday July 24 2007
TimeRepliesSubject
11:36PM 0 Asterisk 1.2.23 and 1.4.9 released
11:26PM 0 ASA-2007-018: Resource Exhaustion vulnerability in IAX2 channel driver
9:23PM 0 How I can configure asterisk to register as gatekeeper server with another gatekeeper
9:19PM 1 Testers needed for VoIP router solution
9:09PM 17 What is the best softphone work with Asterisk
8:27PM 2 SIP IP Trunk, between Asterisk and Softswitch
8:02PM 3 rxFAX core dumps
6:32PM 0 mISDN & Asterisk 1.4: HFC-S card not responsive
4:51PM 0 Problem Hangup Help
4:48PM 3 Digium adaper S101I - IAXy Losing connection
4:45PM 0 VoiceTronix 12 + Far-End Hangup Detection
4:25PM 3 SIP jitter buffer and asterisk native bridge
3:44PM 11 [beginner] Problem of detecting call
12:17PM 6 MySQL components in asterisk-addons not being built
9:43AM 0 Diva Server BRI hangs up after about 25 seconds
9:27AM 8 Dial out through multiple Zap groups
3:47AM 5 CallerID from POTS to SIP
2:21AM 3 Astribank-8BRI
 
Monday July 23 2007
TimeRepliesSubject
10:54PM 1 G729 with SIP and H.323
8:19PM 3 TDM04B & FIOS No Hangups Often
8:09PM 1 is there a tool that gives DTMF information on T1
7:31PM 0 Can Asterisk hear on two IP addresses?
7:30PM 0 Problem w/ MySQL update from perl AGI script
6:56PM 0 chan_mobile - no Audio (Asterisk + bluetooth + chan_mobile)
6:33PM 8 Dialplan
5:40PM 0 Fwd: Asterisk and COS bits
4:41PM 2 Voicemail .lock- files voicemail box not accessible
3:49PM 0 app_changrab, replacement for meetme and conference: returning to dialplan
3:36PM 0 Asterisknow recompile kernel module
2:55PM 3 Problem Hangup
2:53PM 4 IAX Encryption
1:44PM 9 phone directory with asterisk
1:26PM 7 Polycom IP 4000 Soundstation SIP Conference Phone Question
9:54AM 0 users in sip.conf or in users.conf? what is the difference?
8:42AM 0 1.4.8 jabber integration
8:40AM 3 ODBC Connection failed
6:59AM 1 VPN on Asterisk
6:38AM 3 Can Asterisk hear on two IP addresses? And can I do
4:16AM 4 extension.conf doesn't reload?
4:15AM 2 Upgrade and keep the configuration
4:11AM 0 IP Trunk between Asterisk and another IP PBX
2:28AM 0 CAS signalling and FAX solution
12:33AM 1 Viable Alternatives to TDM400P
 
Sunday July 22 2007
TimeRepliesSubject
5:58PM 1 DTMF recognition problem with PSTN
4:45PM 11 Debian etch and web voice mail - how to configure it?
3:54PM 2 Asterisk CTI interface to control legacy PBX
2:29PM 5 Wake-Up Call didn't work
1:07PM 0 te110p stays in red alarm
11:32AM 5 Music on Hold and Announcements
10:37AM 5 IMAP and ODBC voicemail storage
9:53AM 1 Asterisk-1.2.22 DeadAGI Hangup
7:58AM 1 Problem X100P - clone asterisk 1.4.8 no hangup
1:34AM 0 AstLinux 0.4.7
12:24AM 1 Asterisk and COS bits
 
Saturday July 21 2007
TimeRepliesSubject
5:11PM 0 700Mhz Spectrum
1:38PM 1 Call Initiation with Asterisk
12:51PM 0 New tutorial: compiling Asterisk 1.4 with zaptel and H323 support
10:52AM 16 Best and easiest soft phone for my Dad..
3:22AM 0 asterisk-users Digest, Vol 36, Issue 61
1:03AM 3 Configuring Sangoma A101D with Asterisk 1.2.18 & zaptel-1.2.17.1
12:44AM 12 Has anybody used fanless computers of logic supply with asterisk?
 
Friday July 20 2007
TimeRepliesSubject
11:55PM 4 POE injector
9:34PM 1 Aastra phones loosing service...
9:01PM 2 Announcing Digium/Asterisk World's Conference Program
8:33PM 4 Problem
7:16PM 2 ulaw to g726 conversion
4:13PM 2 Asterisk IVR Performance
2:30PM 1 asterisk novice needs help.
12:28PM 2 asterisk hang up the Caller after recording voicemail
12:15PM 0 sip softphone for PDA window mobile 2003 / 5.0 ?
12:10PM 1 Which IP Phones will work with non-Asterisk PBX systems too?
11:32AM 0 Asterisk Channel and VLC
10:31AM 0 Caller is hanged up after recording voicemail
9:45AM 3 priorityjumping not working, Dial goes to n+1 not n+101
8:38AM 7 Asterisk Freeze
6:41AM 5 pattern base call routing
3:32AM 6 * core file not recognized
1:15AM 4 Any plans for proper faxing support
 
Thursday July 19 2007
TimeRepliesSubject
10:34PM 2 Polycom 650 freezing on Transfer
9:16PM 0 job opportunity
9:03PM 0 Any open source OSS system for asterisk?
6:37PM 4 Problem after upgrading from 1.2.21.1 to 1.2.22
6:01PM 0 Blank Voicemails/Vonage Problem
5:24PM 1 Idefisk softphone - official 2.0 release - Zoiper
4:55PM 5 Why using usecallerid=no?
4:05PM 1 iaxtel.com down?
3:28PM 7 PRI Card
2:41PM 15 Blank Voicemails
2:05PM 1 1.4.X howto disable able xpp with ./configure
1:55PM 3 open up firewall ports for Asterisk - safe?
1:29PM 0 Does anyone have the file: pickup-mgernoth-2006-07-28.patch.txt
1:20PM 27 G729 copy protection
1:16PM 2 Parsing IAXPeers from Asterisk Manager (PHP API)
12:52PM 16 Upgrade Procedure
10:58AM 1 asterisk libraries dependecies
9:30AM 0 Asterisk with 2 Public IP-Is it possible?
7:44AM 1 Force asterisk to re-resolve dns names?
5:30AM 3 New book "Asterisk Cookbook" any good?
4:07AM 4 Gtalk/Jabber connect issues in 1.4.8
 
Wednesday July 18 2007
TimeRepliesSubject
11:41PM 1 cdr_addon_mysql.so is not created
11:37PM 6 In Vancouver is it a local to call from 778 to 604, and vice versa?
9:23PM 0 Ncurses dependencies
9:14PM 3 Flash(), Centrex Lines, and 3 way calling
9:01PM 5 Redundancy / Failover
8:47PM 0 Does Asterisk support STUN or TURN? How to configure asterisk NAT traversal?
7:39PM 2 Force SIP hang up.
5:07PM 3 Error Configuring Asterisk (FREEPBX)
4:40PM 3 Remote vm system message pickup
3:32PM 8 Problem building Asterisk 1.2.22
2:12PM 3 what codecs for LAN
2:11PM 0 calls dropped with te110p E1
2:09PM 1 large setup for asterisk
2:09PM 3 Issue in insatlling addons-1.4.2
2:06PM 16 how to use call transfer
12:18PM 0 PRI Change Channel Identification from Exclusive to Preferred or Negotiation?
12:00PM 1 Asterisk Voicemail Imap Storage with MS Exchange
11:25AM 0 Queue to outgoing Zap channels when congested
10:31AM 2 E1 Virtual Callcenter
9:51AM 1 blind transfer on hook-flash from SIP phone
7:47AM 0 How to change Zap channel negotiation/exclusive etc..?
4:00AM 3 AudioCodec MP114
1:04AM 1 Any way to determine remote Asterisk version
 
Tuesday July 17 2007
TimeRepliesSubject
10:44PM 4 bristuff for hfc card on Xscale 80219
10:30PM 0 ASA-2007-017: Remote crash vulnerability in STUN implementation
10:28PM 0 ASA-2007-016: Remote crash vulnerability in Skinny channel driver
10:27PM 0 ASA-2007-015: Remote Crash Vulnerability in IAX2 channel driver
10:25PM 0 ASA-2007-014: Stack buffer overflow in IAX2 channel driver
10:22PM 0 Critical Updates: Asterisk 1.2.22 and 1.4.8 released
8:46PM 2 No sound from Festival, but *something* is happening
7:39PM 19 Asterisk 1.4, Unicall and Nextel...
7:22PM 0 Can Asterisk hear on two IP addresses? And can Ido routing for calls from private to public or public toprivate IP addresses
5:54PM 1 Asterisk 1.4.6 crash using queue app
4:26PM 3 media not accpetable with outgoing call on cisco
2:09PM 3 2 PRI on asterisk
2:09PM 2 Asterisk and ATA-186 question-- calling one port from the other port..
1:43PM 5 Zap channels unavailable?
12:59PM 2 chan_isdn with HFC-compatible
12:24PM 1 Problem with H option of Dial()
12:06PM 1 Not hearing the caller after 2 x Flash
12:04PM 3 asterisk web interface
12:02PM 0 help with sip configuration for sipgate.de on asterisk 1.4
11:57AM 0 No music on hold on ISDN line
11:04AM 0 Suggestion for installation
9:26AM 0 Digitized audio at the beginning of a call
9:25AM 2 Music on hold problem
8:56AM 9 Asterisk PRI Busy Problem
5:15AM 2 Asterisk Hosting (Dedicated Servers)
2:23AM 4 MultiParking
1:32AM 1 DTMF regeneration on PRI
 
Monday July 16 2007
TimeRepliesSubject
11:43PM 2 CID on Polycom Phones
10:38PM 1 OT: How to compile Zaptel for different kernel version...
10:27PM 0 No audio after updated Asterisk 1.4.7.1
6:51PM 4 Zaptel 1.2.19 and 1.4.4 released
6:38PM 0 Pri problems
5:58PM 5 OT - Cisco Callmanager System Prompts
4:57PM 0 Dial and option G
4:14PM 0 Dual dtmfmode?
3:30PM 0 Registration Interval
3:13PM 4 asterisk 1.4 and gnugk with ooh323
2:44PM 5 USB Cordless
2:31PM 4 Crontab script to check health of Asterisk server?
12:48PM 1 Cisco 7940 log on/off
10:01AM 1 [Asterisk]Asterisk's behavior of a simple call
8:36AM 4 I want to record each phone call
 
Sunday July 15 2007
TimeRepliesSubject
7:38PM 0 choppy sound when transcoding (after os update)
6:49PM 4 1.4.7 chan_alsa : snd_pcm_open failed
2:50PM 4 Asterisk with iax2 over satellite
11:14AM 0 asterisk ncurses dependencies
3:31AM 2 TimeStamp a Recording
2:17AM 4 surge protector?
 
Saturday July 14 2007
TimeRepliesSubject
8:51PM 0 1.4 Crashes
8:20PM 4 tT in callparking
7:56PM 3 's' extension Asterisk 1.2.18
11:23AM 5 Zaptel/mISDN and call transfer
9:58AM 2 gui for conferencing
3:54AM 3 open source screen pop software for asterisk
2:27AM 1 calling from ACT
1:30AM 1 Info about Providers
 
Friday July 13 2007
TimeRepliesSubject
7:57PM 1 Media Proxy Mode in Asterik: SIP and
7:52PM 0 b410p and DTMF: dtmfthreshold in 1.2.18 zaptel drivers please?
7:15PM 0 Selling a Digium TDM400P w/ 4 FXO cards
4:52PM 1 Can Asterisk hear on two IP addresses? And can I do routing for calls from private to public or public to private IP addresses
4:19PM 1 Media Proxy Mode in Asterik: SIP and H.323
4:16PM 4 Macro: s-NOANSWER, _s-.
2:06PM 3 asterisk-addons compilation "error: dereferencing pointer to incomplete type"
1:58PM 2 limit simultaneous calls
1:41PM 0 no ringback from SIP server when originating call
1:25PM 2 Transfer Question
1:20PM 2 Distribution lists for voicemail
12:33PM 0 Problems with RNDIS
10:06AM 0 asterisk snmp
8:24AM 1 QUEUE_WAITING_COUNT
 
Thursday July 12 2007
TimeRepliesSubject
10:12PM 0 Queues monitoring software - OrderlyStats now FREE
9:27PM 0 AstLinux 0.4.6.1 Released
9:08PM 0 Outpulse with Asterisk
8:58PM 0 PRI and Local numbers
5:50PM 3 Different SIP From and Auth?
5:01PM 10 Trials with 1.4
4:40PM 0 analog call progress - simplified I hope
4:22PM 1 Test
2:25PM 2 how to load phone registration information
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1:23PM 0 Additional Wildcard TDM2400P Setup
12:43PM 1 exit ChanSpy with DTMF
9:12AM 2 Queues monitoring software
7:55AM 7 Lines Not being Hung UP Major
3:35AM 3 USB Modem with asterisk
 
Wednesday July 11 2007
TimeRepliesSubject
10:25PM 1 Asterisk as outbound proxy
10:24PM 1 Access specific port of Mediatrix 1204 from Asterisk
10:08PM 5 Codec Negotiation
7:39PM 1 Voicemail messages not deleting
7:32PM 7 Queue property
6:59PM 3 Pass Dialed number to a script
5:57PM 0 analog call progress - simplified ( I hope)
3:04PM 3 Call Waiting
1:55PM 4 MOH stop and resume when i hold
1:30PM 0 VoIP + IM unified client
12:10PM 2 iax2 peer become UNREACHABLE
11:00AM 0 how to force asterisk to read registration information from DB
8:32AM 1 Asterisk and Hardware Requirements
7:51AM 3 Music on hold stops on blind transfer
4:37AM 2 Looking for a USB phone handset or headset
1:20AM 1 extensions reload -- what impact?
 
Tuesday July 10 2007
TimeRepliesSubject
10:11PM 0 mixing video and non-video clients
10:08PM 3 video calls - Windows / Linux interoperability ?
7:52PM 0 Macro Goofiness
7:48PM 0 Odd AGI Issue - STREAM FILE, GET DATA not playing file
6:15PM 4 Asterisk 1.2.21.1 and 1.4.7.1 released
5:05PM 0 G722 and Polycom 550
4:17PM 0 Asterisk, AudioCodes, Caller ID
4:13PM 8 PlayDTMF and Asterisk Manager
2:24PM 7 Edit ulaw file
12:28PM 2 video phones on 1.4.7
12:09PM 0 VAD/CNG
11:01AM 0 sharing phone registration information between asterisk servers
9:47AM 2 DUNDI behind NAT?
6:16AM 1 asterisk-users Digest, Vol 36, Issue 25
5:42AM 3 ZAP TDM and DTMF issue
4:45AM 1 Asterisk 1.4.7 and MOH
4:25AM 0 how to register several clients with different number but using single authentication account ?
 
Monday July 9 2007
TimeRepliesSubject
11:40PM 0 Meetme delay?
10:45PM 0 Asterisk 1.2.21, 1.4.7 and Libpri 1.2.5, 1.4.1 released
8:35PM 0 Digium cards for sale in Pakistan
7:15PM 3 Basic asterisk Autodialer?
6:41PM 0 allow third party registration/invitaion
6:40PM 6 Setting Appearance on Outbound Calls?
3:21PM 4 Problems sending more than 2 SMS with asterisk / smsq
1:06PM 5 Very bad TDMF tone !
10:39AM 3 Background transfers with callback
10:22AM 12 Monitor events?
8:28AM 0 CTI application controling Asteridk
 
Sunday July 8 2007
TimeRepliesSubject
11:43PM 2 Asterisk and Mitel 3300 ICP
11:41PM 0 Sip trunk between Asterisk and Mitel 3300 ICP
1:22PM 3 Zapata, Junghanns Card and a leading 0 on inbound calls
12:58PM 1 Asterisk Help
8:25AM 1 Early Media Handling
5:56AM 4 asterisk is not sip proxy
5:33AM 2 Auto Fall Through when kicking users in MeetMe
4:15AM 0 SIT tone detection on Zap channel (PRI)
 
Saturday July 7 2007
TimeRepliesSubject
10:50PM 1 Call Waiting curiosity...
8:07PM 2 Polycom multiple registrations
6:29PM 5 Fax and Asterisk
4:20PM 0 Help doing one modification to libmfcr2.c of Unicall
4:11PM 20 Sip Providers
4:02PM 11 installing * from source
3:04PM 4 Corporate Feedback to OSS (was: Re: Mushtaq Ahmed is out of the office.)
2:06PM 4 Which features are lost when canreinvite is turned on ?
8:55AM 0 Asterisk TV is about to go live
6:16AM 2 Unable to install Asterisk Now Beta 6
12:15AM 3 Channel name in queue log replaced by a manager event?
 
Friday July 6 2007
TimeRepliesSubject
8:30PM 0 LinkSys ATAs with GSM Support?
6:14PM 0 PRI CallerID and Apple iPhone
4:21PM 9 OT: Blackberry and Asterisk voicemail files.
3:27PM 0 Blind transfer from Queue in AGI script failuire
2:32PM 0 SIP peer unrechable when using an aliased interface
11:58AM 2 Asterisk Manager
8:05AM 4 Mushtaq Ahmed is out of the office.
6:10AM 2 Issue using zaptel's dynamic spans.
 
Thursday July 5 2007
TimeRepliesSubject
10:20PM 1 Missing TRANSFER event in queue log when using Local Channels
10:13PM 1 sounds
7:32PM 4 Call Queues
7:13PM 1 sometimes half audio on 7960
6:17PM 3 Call Screening Not Working
5:10PM 0 IAX-Voicemail
4:50PM 1 IAX additional call-data
4:12PM 0 Visually impaired employees
3:51PM 1 Simple CDRs w/Asterisk/OpenSER.
3:18PM 5 REGEX expression for NXXNXXXXXX?
3:16PM 0 exits in NJ
2:50PM 0 connecting 1.2 and 1.4 using SIP
1:16PM 1 SIP / STUN / Network - Help!!
12:37PM 2 G729 on Solaris SPARC/x86/x64 Codec
11:08AM 4 sometimes calls drop during attended transfer
9:42AM 2 Asterisk E1 card support Q.SIG
9:02AM 1 AgentCallBackLogin vsAddQueueMember
4:09AM 3 Asterisk console filtering and logging
12:04AM 1 Need Help in Asterisk BLF/Presence/Hints
 
Wednesday July 4 2007
TimeRepliesSubject
11:17PM 0 FW: Openmoko ads now on youtube
11:14PM 3 Xorcom Bri and asterisk crashes
10:56PM 0 Problems with SIP Registration on VPN Link
9:14PM 0 ANNOUNCEMENT : A2Billing (Asterisk2Billing) - V1.3.0 STABLE (Yellowjacket)
7:55PM 4 reliaclear.com
4:59PM 5 Upgrade Asterisk
4:38PM 2 Call still in queue after Reject Signal
3:31PM 0 H263-2000 video format
2:46PM 1 Problems with misdn and ChanIsAvail
12:43PM 36 List delays
10:32AM 1 Dialout Macro and transfer call in progress
9:26AM 1 Asterisk TV will go live this Friday
6:35AM 0 asterisk hardware E1 pri card
5:47AM 0 single digit dial extension
5:39AM 1 call transfer not working
5:30AM 0 system recording problem using wav file
4:58AM 2 Need advice to get wcte11xp and wcfxo to load
2:05AM 2 Asterisk Support Question
1:11AM 1 QueueMemberStatus
 
Tuesday July 3 2007
TimeRepliesSubject
10:25PM 1 Distinctive ring detection not detecting ring cadences
10:23PM 1 Digit Convesion and Digit Insertion
10:12PM 2 Putting a password on the international call
10:06PM 5 Determining the used codec for the IP Trunk (SIP Trunk)
10:00PM 0 TDM800P cards with one way voice
6:11PM 1 res_config_mysql.c: MySQL RealTime: Failed to connect database server ..
5:44PM 6 Asterisk and Panasonic TDA200
5:29PM 2 got 404 when route calls through Asterisk to another proxy
4:14PM 5 Session Border Controller time...
2:44PM 1 lookup a anonymous internal caller
2:36PM 1 help with internal extensions
1:42PM 8 garbled calls
12:17PM 9 Google acquires Grand Central
11:43AM 1 Configuring BLF or Asterisk presence/Hints feature
11:20AM 17 Suing Dell||Dull Computers for CID abuse
7:00AM 6 Need Advice/Suggestion
6:51AM 0 Configuring BLF or Asterisk presence feature
 
Monday July 2 2007
TimeRepliesSubject
9:02PM 3 DID providers in Toronto
8:54PM 2 Sip phones using the wrong context for an outbound call
8:45PM 0 trying to get vpb to compile
7:27PM 7 Asterisk 1.2 TDM24xx and B410P
5:37PM 2 Question about dnsmgr
5:13PM 0 FXO interface modules in Bombay, India
2:45PM 5 Help. Cannot compile version 1.4.6 with the following error
12:34PM 6 softphone with g729 codec
10:45AM 4 Call transfer in asterisk
10:18AM 5 Gigaset 450IP loses registration
9:56AM 1 "Random" all circuits busy now message
7:40AM 0 Authenticaion on incoming calls
1:40AM 0 Single ringer phone for incoming calls, that every one can answer
 
Sunday July 1 2007
TimeRepliesSubject
11:47PM 1 Asterisk strange behaviour
10:39PM 0 Rockwell ACD - "Take back and transfer"
10:22PM 4 Installing AJAM
10:00PM 0 ANNOUNCEMENT: Asterisk-Java 0.3 released
7:00PM 1 How can we block the calls for specific code
6:09PM 5 Not able to find the file zaptel.conf after compiling asterisk and zaptel
5:52PM 5 Creating a Voicemail ...
5:00PM 4 Suppress MusicOnHold in Queue
3:14PM 2 problems with dtmf using asterisk-1.4 rev r 6745
1:21PM 0 Transfer outgoing call - macro
12:58PM 0 asterisk / MSN & DID
10:04AM 3 the-asterisk-book.com online (unstable version)
9:37AM 0 Solved: fail tp load modules
1:11AM 4 G729 , upgrade asterisk