asterisk users - Aug 2006

Thursday August 31 2006
10:37PM 1 MOH help needed with fresh install
9:33PM 3 Can not hear the telco System Announcement
9:31PM 2 Help in dailplan in asterisk
8:40PM 4 Asterisk server crashes after two years
7:59PM 1 Using Thunderbird (mail client) to call Contacts from Address Book
7:43PM 0 Am I looking for automon?
7:12PM 2 Adit 3104 randomly reboot
5:53PM 2 quadbri & TDM400P on same pbx ?
5:47PM 0 How to use *411 using either last or first name?
5:44PM 11 Sipura SPA3000
3:34PM 12 Polycoms, Attended Transfer and Canreinvite = yes
2:37PM 2 does OOH323 channel support Early Media?
1:52PM 9 File structure question
1:48PM 1 Off Topic: Hardware Required
1:38PM 5 US Toll-Free DID Providers with Caller ID NAME?
1:18PM 1 SPA-942 Sound Quality
1:14PM 7 0005162: RTP Packetization : Few questions
12:58PM 1 Snom Function keys
11:48AM 0 weird sound with IAX
11:19AM 0 Compatibility INTEL E7520
9:53AM 0 Question about 7940s and call forwarding
9:46AM 1 Per DID Codec Negotiation
9:38AM 1 Polycom HD Voice
9:29AM 0 Problems using Queues with Autofill option
9:05AM 1 How is GXP2000 with latest firmware
8:58AM 5 Problems compil 1.2.11
8:57AM 2 Asterisk Sending Data to a Web Page
8:38AM 0 app_rxfax and T.38
8:33AM 3 Help Preventing Click to Call fraud on Asterisk Servers!
8:27AM 12 help me!!Problem on incoming calls
8:20AM 4 Missing Agent Function
7:42AM 0 Parked call, park-dail context
6:47AM 4 Sangoma A104 2 ports as E1 and 2 ports as T1 configuration
6:38AM 3 "best" BRI card ?
6:35AM 1 Got error when compiling asterisk 1.2.11
6:30AM 12 Cisco 7970 8.0.4 SIP firmware
6:16AM 0 editing configs thru web/ apps
5:57AM 0 Junk at beginning of frame
5:30AM 1 RTP Proxy
5:00AM 7 Fax with asterisk?
4:50AM 0 How to use a Half E1 with Asterisk?
3:55AM 2 Problems with recording
3:50AM 5 voicemail as email and attachment
1:45AM 0 CallerID and call progress pri
12:54AM 0 Wellgate 3804a: Got SIP response 486 "Busy Here"
12:52AM 0 GIZMO and Asterisk, "Failed to authenticate"
12:11AM 2 Toll-Free numbers
Wednesday August 30 2006
11:01PM 1 SER+iptables+Asterisk
8:39PM 1 * during voicemail greeting to access mailbox
8:07PM 3 How to run a batch file on the asterisk CLI
7:38PM 3 iax vs. sip?
7:36PM 22 911 versus 9.911
6:53PM 0 Static vs dynamic meetme rooms
6:19PM 3 question of CLI
3:39PM 0 w as pause dialing issue
1:50PM 1 visual indication of temp. closed mode
12:58PM 8 Polycom 501 config questions
12:53PM 0 Ascom Eurit 133 cordless ISDN phone
10:14AM 1 Speex Problemz
10:02AM 0 Intertex IX68 GW2 AIR 802.11G ADSL2+ ?
9:50AM 2 Asterisk speaks Russian!
9:29AM 1 Asterisk => Master and Slave ?
9:06AM 4 upgrade problem on IP phone 9133i
9:03AM 0 OT: Any thoughts on the new Xserve?
8:55AM 1 Agent solution w/o id/password
8:32AM 5 asterisk presence (from manager API)
8:15AM 0 Help please ==> Wrong password
7:14AM 0 New to Asterisk...
6:58AM 2 Sangoma Problems - A104d not detected
6:09AM 1 IAX call drops, recent instability
6:08AM 0 PrivacyManager
4:51AM 0 Prompts playback changing tempo in SMP kernel
3:58AM 0 personal address progress pri
3:51AM 6 Cisco 7960G SIP firmware 8.4
3:34AM 0 caller display problem
2:47AM 0 Voicemail, how to localize date in email notifications?
2:40AM 1 Snom 360 Function Keys
2:20AM 0 Line detection with TDM400P
1:38AM 0 GXP-2000 update to betafirmware?
Tuesday August 29 2006
10:43PM 2 MixMonitor and g729 licenses
7:11PM 13 does anyone offer truly unlimited voip in the US
6:49PM 13 SER Dispatcher Load Balance How-To?
6:43PM 0 zap fxo to sip fxs intermitently not connecting to each other
6:21PM 2 Unknown CLI output
3:54PM 1 New Parrot application, repeats what you say and more!
3:48PM 1 Digium makes the list!
2:28PM 0 Administrator Forum Email
1:00PM 0 CPU configuration for 250 calls SIP to SIP to IAX and fonebridge and two asterisk servers
12:59PM 0 Asterisk 1.2.11 and ${SIPDOMAIN} variable
12:03PM 0 OT: Bandwidth calculations and PCI/PCIX/PCIE
11:12AM 1 SIP T1 timer and qualify=yes
11:01AM 0 GXP-2000 auf Betafirmware updaten?
10:35AM 1 Re: [asterisk-biz] Asterisk Tools
10:26AM 2 Copying a recording to a voice mail box
10:21AM 0 [Fwd: Re: Asterisk t38passthrough]
10:08AM 3 DTMF between cisco and sipura going through asterisk
9:04AM 1 Advice needed - asterisk & Mitel 200SX
8:29AM 3 IP interface "box" for Meridian type digital phone
8:27AM 0 Which BRI Card ?
8:15AM 3 Connecting two asterisk servers
7:48AM 4 Detect if cell phone or users
6:57AM 5 Analyze core file prodeced after safe_asterisk crashh
6:17AM 0 playback() breaks audio in zap->iax->iax->zap channel
6:12AM 4 Asterisk codec strangeness
4:30AM 0 working chan_bluetooth enviroment
4:28AM 2 Handytone 286 T.38 SDP parameters
3:24AM 3 Mix Monitor call quality
3:19AM 1 Asterisk - Comfort Noise
3:02AM 2 Asterisk 1.2.4 I hear other party's voice only when I speack need help
2:59AM 4 transform bridged call into a conference
2:55AM 7 does misdn-mqueue work if compiled with gcc 4?
2:48AM 2 sip giving problems, please help.
2:16AM 0 SIP Error message
Monday August 28 2006
11:57PM 0 Providers that offer contract
9:21PM 0 IAX2 Bandwidth setting
9:18PM 2 Selecting outbound trunk
8:01PM 1 Debian and Asterisk IAX2 channel driver
7:28PM 1 Asterisk Manager Interface Question
6:49PM 2 Can I increase DTMF sensitivity?
4:59PM 1 ISDN BRI, and Trixbox
4:23PM 0 Is there a Blue tooth wireless headset that willwork with asterisk?
3:46PM 0 Asterisk queues and dynamic members
3:24PM 1 Is there a Blue tooth wireless headset that will work with asterisk?
3:23PM 0 Changes in handling anonymous calls entering asterisk
2:03PM 6 Voicemail/Email Integration
1:31PM 0 Multiple Queue Problem
1:23PM 7 Can anyone recommend a large button sip phone for the elderley.
11:53AM 6 manual mods with GUI in place
10:49AM 4 Problem with a TDM400P
10:15AM 1 Call parking with Polycom's - works but MOH stops in one scenario
10:12AM 0 Re: asterisk-users Digest, Vol 25, Issue 139
10:10AM 16 Asterisk with PABX
9:31AM 0 Timeout Registration IAX2
9:08AM 0 Changes in handling anonymous calls entering ast erisk
9:06AM 0 Queue problem - autofill option
8:32AM 4 Missing number 2 in "advanced options" of VM
8:11AM 2 REGISTER attempt
7:20AM 2 Question about context for incoming calls
7:14AM 1 Make Asterisk server initiate a Call
6:56AM 1 Grabbing authenticated mailbox value from VoicemailMain()
6:54AM 0 AEL2 patch issues
6:52AM 3 H264
6:47AM 6 How to set MWI
5:11AM 0 GROUP() and queues
4:58AM 3 lost packets when bridging zap and iax
2:49AM 4 Remote CAPI - ISDN over TCP/IP
2:27AM 0 newbie request
12:07AM 7 Tracing audio problems
Sunday August 27 2006
11:01PM 1 TrixBox install
10:03PM 2 how to enable REACHABLE/UNREACHABLE messages in logs
8:32PM 0 Trixbox – Called party can't hangup
5:15PM 10 Max number of SIP devices registered to an extension
4:49PM 1 SEXY WOMAN wants to know about =>Callback in within voicemail broken
2:31PM 2 Shared NFS or Shared MySQL for redundant secondary server?
9:19AM 2 detecting a users number using the dialplan or AGI
8:50AM 4 Cannot dial out through SIP provider
8:33AM 4 CDR Function - Asterisk-1.2.10
8:01AM 0 [RESOLVED] One way audion on Sangoma
5:40AM 0 asterisk registering as extension to another asterisk server problem
4:56AM 5 Dial C option
3:33AM 0 Doubled digits on vm pasword
1:41AM 0 about MusicOnHold / Playback
1:29AM 0 Voicemail's mail formate
Saturday August 26 2006
10:58PM 2 "hint" for Hold
10:20PM 6 Call Max Time
4:21PM 0 ticks in the pstn side audio
4:21PM 0 determining meetme user number
2:48PM 0 ANNOUNCEMENT: Asterisk-Java 0.3-m1 released
9:33AM 2 getting SIP to listen on multiple ports
8:52AM 3 Nobody is responding. Why? (Implement music on transfer)
8:31AM 6 Problem with Tycho Voicemail
8:26AM 5 Asterisk Performance without RTP?
7:24AM 3 can not get ${LEN(VAR)} and greater than ">" to work for me
5:24AM 2 Uptime Record?
12:41AM 0 app_txfax / app_rxfax
Friday August 25 2006
8:39PM 1 Asterisk Real Time Engine - Fails to Connect to MySQL
5:35PM 5 Help compiling asterisk-addons on Debian?
5:03PM 1 CentOS4.3 or Debian 3.1r2?
4:26PM 2 Linksys PAP2 Ring Settings
3:10PM 0 Re: asterisk-users Digest, Vol 25, Issue 119
2:31PM 0 Using asterisk to simulate ISDN BRI line
12:54PM 2 What are my logs telling me here?
12:51PM 0 read more than 2 digits on festival
11:50AM 6 Will Asterisk work with Exchange 2007 UM?
11:27AM 2 [RESOLUTION] Polycom microbrowser issue Error HTTP 406 withIIS
11:18AM 5 DNS
9:42AM 1 Standard for transfer via IAX provider?
9:22AM 6 7970 'LoadID incorrect' problem
8:36AM 5 misdn-init.conf card parameter for a monoBRI
8:23AM 2 New Asterisk Voice Changer 0.4
6:47AM 1 Singapore
6:31AM 0 Can i use the FXO of a addpack in Asterisk
5:51AM 0 Polycom microbrowser issue Error HTTP 406 withIIS
4:40AM 3 How do you simultaniously dial multiple MSNs on one ISDN BRI b-channel?
2:58AM 2 Does anyone use T.38?
1:55AM 0 Multiple Vulnerabilities in Asterisk 1.2.10 (Fixed in 1.2.11)
1:04AM 1 quadBRI beronet card: how to specify which ISDN channel to use to make calls
12:39AM 14 IP phone with 2 ethernet jacks
Thursday August 24 2006
11:32PM 0 Re: [Users] Mysql problem
7:02PM 2 Can the codec/format for name/greeting in voicemail be changed?
4:36PM 1 hint status not updating on inbound
3:04PM 3 Attempt to setup paging and intercom
2:57PM 13 Idiot questions
2:19PM 3 Snom phones locking up
2:06PM 1 Polycom microbrowser issue Error HTTP 406 with IIS
11:29AM 0 hotel teledex integration anyone?
11:19AM 5 RE: [asterisk-dev] Phone status
11:17AM 3 Phone status
10:55AM 2 Modems dialing over sangoma a104d
10:53AM 3 Call Parking Ring Back (Snoms)
10:53AM 7 Asterisk t38passthrough
9:31AM 1 No outbound with A2Billing
8:43AM 3 Wellgate 3804a
8:37AM 0 originate from group + dialplan
8:33AM 2 SendText Queue Notification
7:19AM 0 need help with error code
6:49AM 0 Monitoring/Listening In (Scott Pinhorne)
6:49AM 0 About IVR and Oracle (Tim Panton)
6:32AM 6 SV: E61
6:19AM 0 Quiet on the list today?
6:07AM 7 E61
5:14AM 2 Active Directory Listing Feauture
5:12AM 3 quintum Calling Card
3:54AM 0 Multiple lines in body of UserEvent
2:42AM 1 Monitoring/Listening In
Wednesday August 23 2006
11:03PM 0 AstLinux 0.4.3 Released!
6:02PM 17 Nokia E60/61/70 and SIP
4:39PM 0 SJPhone and Asterisk over H323
4:25PM 0 Getting strange behavior on SIP channels after upgrade to 1.2.11
4:06PM 6 One way audion on Sangoma
3:43PM 0 howto install asterisk on freebsd release 4.11
3:07PM 4 About IVR and Oracle
2:42PM 0 isdn30 uk setup problem
1:54PM 0 NoCDR()
1:45PM 0 MySQL undefined symbol: __pure_virtual
1:34PM 0 Unable to start special tone
1:00PM 0 client socket to asterisk manager gets disconnected
12:35PM 10 Annoying Bristuff
11:40AM 3 NAT problems
11:13AM 1 3COM NBX and Digium Cards...
10:43AM 1 USB GSM gateway for Asterisk?
10:33AM 1 IAX2 extn not registering on 4569
10:32AM 1 Choppy calls on IAX trunk but no problems on internal calls
9:55AM 0 Silent Calls (Ghost Calls) When Picking Up Queue Calls
8:54AM 0 Connecting Asterisk to Avaya Definity over H.323
8:53AM 6 Cisco PIX firewall and nat=yes
8:44AM 1 Registering IP Phone To Asterisk
8:11AM 0 Weird compile problem
6:53AM 6 Cisco Router QOS and IAX2
6:48AM 1 VM - advanced options?
6:48AM 0 SV: Hint extension issue - bug?
5:58AM 3 Slightly off-topic: Opinions of Comcast and Bellsouth?
5:39AM 2 Direct to Voicemail
4:51AM 0 Call Handoff
3:33AM 0 dtmf during a call
2:06AM 0 Auto Congestion
2:00AM 2 Adding/Removing Prefixes
1:57AM 1 Dialling from extension to extension with Manager
12:56AM 2 column width in CLI
Tuesday August 22 2006
9:21PM 0 Missing Extension
8:45PM 0 Multiple site multi server setup
8:29PM 2 problems with wevbmail
7:43PM 10 Calls over VPN
6:40PM 2 Working Sipura 3000 or Linksys 3102 configuration?
6:33PM 11 How to set externip in sip.conf automatically?
6:30PM 0 Asterisk 1.2.11, Asterisk-Addons 1.2.4 and Zaptel 1.2.8 Released
6:01PM 5 SSH connection hangs on logout?
5:48PM 1 problem with asterisk billing time...
5:36PM 2 Simple CDR parser to print to webpage
5:24PM 0 Non-zaptel hardware based timing sources
5:20PM 0 Speech Recognition Apps
5:09PM 1 Setting the contact header on outbound INVITE
5:09PM 4 Prompts recording for Asterisk
3:53PM 1 No CLID from PSTN using X100P FXO Card
3:49PM 6 Strange SIP response
3:22PM 12 Hint extension issue - bug?
3:19PM 0 Asterisk, two eth and two providers
1:46PM 1 Anybody using Eicon SoftIP with Asterisk
1:32PM 0 PRI Ethernet Bridge
11:26AM 12 How can I implement Music on Call Transfer?
9:07AM 4 Unable to match on CallerID in an include block
7:55AM 1 AMI initiate call probs
7:44AM 2 Polycom 501 vs 601 provisioning
7:20AM 21 Realtime Extensions -- Comments?
6:31AM 2 R: Snom360 with 6.2.2 firmware
6:30AM 0 No audio from IVR of Cisco Call Manager
6:28AM 3 placing a call with the Manager interface
6:27AM 3 Asterisk and spandsp
5:55AM 1 Setting RPID privacy?
5:02AM 1 re-writing the dial plan - some hints please
4:45AM 2 using asterisk + sangoma a102 to simulate telco PRI: is possible?
4:43AM 1 if command for or missing callerid?
4:24AM 2 Sipura 3000 dialplan strings.
4:03AM 0 R: Text to Speech
4:00AM 3 Compilation
2:48AM 2 Asterisk forum -
1:59AM 15 GSM gateway and FXO ATA
1:19AM 0 lbProxy
1:17AM 0 H323 can not register to remote openh323gk?
1:04AM 0 NuFone chan_h323
12:44AM 3 Snom360 with 6.2.2 firmware
12:36AM 2 How to modify incoming DNIS?
12:23AM 1 1.2.10 and
Monday August 21 2006
11:28PM 8 No retry after DNS failure
8:20PM 0 Meetme bug or feature?
8:15PM 2 Re: SIP Debug to file - Is it possible?
8:02PM 0 Quick, hopefully easy, question
6:59PM 0 Indonesian MFC-R2
6:58PM 2 Re: SIP Debug to file - Is it possible?
5:50PM 0 SIP Encryption in China
4:54PM 2 SLA.conf
2:36PM 2 Call file do 2 outbound call
2:15PM 4 Manager API: matching an Originate to the Newchannel event
2:03PM 2 Voicemail and languages other than english doesn't seem to work well
1:34PM 0 Double dial dtmf sounds
1:12PM 8 Text to Speech
12:37PM 11 Realtime and hints
12:05PM 1 Realtime and labels
10:01AM 1 Asterisk in Xen 3.0
9:39AM 1 Portuguese sound files available?
9:10AM 0 Status of Monitor
8:41AM 0 Size of realtime appdata field under MySQL
8:26AM 0 failed calls
8:14AM 0 Cancelling outbound call: is Asterisk behaving correctly
7:50AM 0 DTMF + voipjet
7:26AM 0 SV: Choppy sound zap-to-sip, but not sip-to-sip?
7:08AM 0 Is it possible to call System dialplan application via AMI?
7:07AM 0 IAX2 TRUNK CPU consumption
6:52AM 1 Configure mailserver to deliver voicemail
6:22AM 7 Joining calls via manager.api or AGI
6:15AM 0 Choppy sound zap-to-sip, but not sip-to-sip?
4:43AM 0 polycom_acd_functions branch and outboundproxy
4:39AM 0 how to set 'transfercapability'
4:20AM 1 running agi application in the background
3:52AM 12 Zaptel install - Fedora Core 5
2:39AM 4 zap channel media volume
1:41AM 0 Re: no audio issue (
1:15AM 2 SIP ActiveX?
1:03AM 0 IAX2 Auto fallthrough
12:04AM 0 queuememberstatus overwhelms manager socket connection to asterisk
Sunday August 20 2006
11:53PM 1 no audio issue
11:02PM 1 sox &gsm
7:42PM 3 Asterisk installations in Germany
6:28PM 1 Asterisk not parking calls - causes? how to fix?
4:54PM 1 Call to a queue killing Asterisk?
4:53PM 2 Announce caller-id
3:13PM 2 Linksys SPA-941 Message Waiting Indicator
2:20PM 2 Sending signals to asterisk
1:55PM 2 Analog-to-VoIP: blade?
11:20AM 3 Polycom IP430 won't finish boot
10:54AM 0 No-audio problem
9:47AM 2 Metermaid - Parking Slot
9:38AM 9 Connecting an cellphone to asterisk
8:06AM 1 Indonesian MFC/R2
7:55AM 6 Ignoring PRI call?
1:58AM 4 Asterisk Jobs Update
1:51AM 4 How to find which queue member answered a call?
Saturday August 19 2006
9:10PM 2 recommended hardware specs
9:06PM 0 asterisk and hipath 3750
8:36PM 0 Mailcall question
1:43PM 0 Zap channel-modem/fax conn. problems
5:15AM 6 New Voicemail Client for Win32, Linux x86, Mac OS X released
4:55AM 0 TAMPA BAY Asterisk Users Meeting Monday
1:19AM 0 Asterisk + DTMF + g729
12:21AM 0 [Linksys 3102] Couple of issues
Friday August 18 2006
10:04PM 2 Asterisk - SIP client latency
7:08PM 3 loading the prompt files in memory on Asterisk startup
4:17PM 2 SLA Doc
2:42PM 3 Iaxy and SendDTMF??
1:55PM 1 How To NOT Generate A CDR For A Call?
12:55PM 2 chan_skinny - in trunk r40360 - error "unsupported format '0'"
12:39PM 0 Realtime Extensions and 'include =>'
11:28AM 15 Apache for FastAGI
11:04AM 1 Realtime Peers Disappearing
10:50AM 2 MaxRetries:1 - Problems Dialout Call files
9:57AM 2 Static in Monitor recordings
9:44AM 1 Recent additions to the Digium Asteriskdevelopment team
9:43AM 5 Dialplan "or" matching
9:32AM 2 TE207P
9:20AM 0 video call monitor
8:29AM 1 call barge
8:15AM 2 Ringtone/gentone/busy and g729
7:04AM 1 Presence SUBSCRIBE/NOTIFY behaviour
4:57AM 5 PRI gurus - Does any one know the meaningn of span 1 received AOC-E charging 149502040 units
4:35AM 0 Maximum length of CID using SET CALLERID in AGI?
1:22AM 1 Extension presedence.
Thursday August 17 2006
11:12PM 3 Equivalent of channel switching?
10:48PM 4 SIP_HEADER function; what names are available?
10:21PM 0 Always On Top 100 private companies
10:13PM 5 Polycom 601 Issues
8:46PM 0 dynamic queue problem
6:54PM 1 Frustration cubed
3:49PM 0 t.38 asterisk-trunk
3:11PM 6 Sending Email From A Dial Plan
2:55PM 0 Cepstral and Asterisk again...
2:24PM 1 Where can i get a telephone number of Brasilia or Rio de Janeiro in Brazil
2:13PM 1 Asterisk and T1 Extensions.
2:12PM 7 Return data from Fast AGI
2:10PM 1 Turn Off chan_sip Debug Messages
1:08PM 0 Realtime Extension Lookups
1:03PM 2 Accessing SIP URI (not ${SIPURI})
12:33PM 2 Realtime include
11:30AM 1 VoiceMail and Fax on same extension
11:09AM 3 astbill white screen!!
10:16AM 0 Dial statement problem
9:47AM 0 I can´t set to work two tdm2400p and one TE205p on same machine, please help
9:28AM 1 Changing CID
8:55AM 0 Call Parking initiator cannot retrieve parked calls
8:01AM 0 sangoma a102: "Rx Error: 'Retry' exceeds maximum (64k): pci fatal error"
7:48AM 0 pri rdnis found as Facility but not set
7:44AM 0 480 "Temporarily Unavailable" message
7:39AM 0 Assigning specific RTP ports to SIP clients
7:35AM 1 strange behaviour of a zaptel device
6:44AM 0 BRI<->PRI switching and synchronization (data/fax calls)
6:06AM 0 spanDSP + rxfax
6:01AM 12 PRI problems - no D channel
5:37AM 2 valgrind + Asterisk
5:32AM 0 AGI transfer question
4:42AM 11 Challenging problem regarding CallerID in TDM 04B (Trying to solve since 8 days)
4:20AM 1 Sip suppression
3:24AM 0 Dial out based on SIP invite
3:00AM 0 SendDTMF - how to use
2:30AM 0 Dialing out using SIP terminal
2:29AM 0 Conflict between S & L option in Dial?
2:21AM 0 CDR inaccuracies
2:15AM 0 ExtensionState always returns 1
2:07AM 0 wanpipe1:w1g1: Rx Error: 'Retry' exceeds maximum (64k): pci fatal error! (0x0007C03C)
1:09AM 4 Festival through AGI can't handle strings longer than 15 chars
12:07AM 0 Need help: RTP Stream not set up correct only when calling out
Wednesday August 16 2006
11:42PM 4 Re: what is the real use of AEL?
10:04PM 0 How to transfer call without getting dropped
9:46PM 1 how to add prefix 0 (zero) when coming from ISDN trunk
7:23PM 1 Not Working
5:25PM 6 What to use beyond T1's?
4:55PM 3 DMTF issues on voicemail on Zap
4:19PM 8 Set DID
4:01PM 4 Cepstral and Asterisk
3:12PM 10 Recent additions to the Digium Asterisk development team
2:54PM 1 SIP-NAT failure on dynamic IP
2:29PM 1 No "zap" command?
1:16PM 0 Strange CLI Output
1:15PM 0 Trixbox Fax to PDF
1:02PM 1 Force immediate re-registration on sip reload
12:33PM 0 calling in-out
11:03AM 8 Asterisk Real Time and sip.conf file used at the same time
10:12AM 2 Restricting Incoming SIP Calls Without "call-limit"
9:49AM 53 Asterisk 'Hosting'
9:48AM 0 Attended Transfer call return with asterisk + sipura spa2002
8:46AM 0 MFCR2 and Unicall PDF
8:40AM 7 Digium TDM400P Vs Sangoma A200
8:35AM 3 >>Trolltech woos developers with "open" Linux phone<< Who'll be the first with * on a mobile?
8:14AM 2 polycom config error 0x4020: possibly related to RE:Polycom upgrade issue?
7:03AM 0 Asterisk@von - Von Fall, Boston Sept 11-14
6:53AM 0 Asterisk Training - Boston, US and Malaga, Spain
5:10AM 1 Extension for Incoming Call through Zap Channel
4:27AM 0 Support a malformed SIP INVITE
4:18AM 0 REQ: BATM gw-232 sip firmware
1:58AM 1 Problems with outgoing calls on a TE410P
12:40AM 0 RTP Stream not set up correct at outgoing call
12:29AM 0 capi (divas4linux) bearer setting
Tuesday August 15 2006
9:12PM 1 postfix and asterisk
7:07PM 4 Polycom upgrade issue
6:43PM 1 Question about queue
6:32PM 11 modprobe wctdm fails in /etc/rc.local on FC5
5:44PM 0 anothet tes
5:39PM 0 sip host and registering
4:33PM 0 AsteriskSpeaksGoogleTalk - User is always disconnected - Problems
4:29PM 1 New asterisk jukebox needs testing
4:23PM 6 How to reject a call without picking it up, (E1-T1-ISDN)
4:04PM 0 macro-dialout without specifying trunk
3:35PM 3 Multiple registrations to the same asterisk server
3:18PM 1 STRFTIME dialplan function not picking up system timezone
1:57PM 1 Asterisk & Gizmo?
1:41PM 1 7970 SIP image
1:10PM 8 Cisco 7960 password reset
1:10PM 15 New Device
12:28PM 8 Softphone for Windows Mobile 5?
12:05PM 1 IAX unstable with large number of calls?
10:28AM 11 1.2.10 - g726 Issues
10:23AM 6 SIP asterisk over Linksys VPN
9:28AM 0 Intel D945G chipset
9:16AM 4 Page Groups
8:50AM 7 Can budgetone 101 display name part of cid?
8:05AM 32 Manager Interface API's
8:02AM 0 ARI
7:06AM 0 Hangup Problem with PSTN and ISDN
6:09AM 0 extensions.ael - calling an exten from a macro
2:12AM 0 PRI Clock Signal Problem
Monday August 14 2006
11:18PM 4 Sending SIP 183 Session Progressing
9:38PM 2 Problems with incoming authentication
9:10PM 0 Sorry! My Bad!
9:09PM 2 Asterfax and Gentoo
8:36PM 5 1 way audio. Dual NIC's.
8:20PM 2 Ringing after answered on zaptel
6:44PM 3 Run As User Asterisk
6:41PM 3 Config quesiton: all inbound on PRI
6:10PM 0 Reason to hit failed extension
5:25PM 2 PRI Dropouts (Solved)
5:23PM 1 Is anybody moderating this list?
3:53PM 4 Zap difficulties
3:28PM 1 Sending INVITE to an unavailable phone - Bug?
3:24PM 4 Asterisk load testing
3:15PM 4 SPA-942 TFTP Provisioning
1:54PM 5 Asterisk and PHP?
1:53PM 0 Cisco 7961 SIP & Presence / BLF
1:52PM 1 pyAst
1:46PM 2 Asterisk And Java?
1:45PM 2 Asterisk time not the same as unit time ?
1:23PM 2 reloading agents and queues
1:09PM 3 More SNOM, Message Indicator/Retrieval issues
1:02PM 2 Dapper Drake, Asterisk, and Faxing
12:57PM 1 g.711 Codec Question
12:47PM 1 Anyone know a DID provider in Panama (country code 507)?
12:35PM 1 Cron Job to Drop a Call File When the Hard Drive Gets over 50% Full
12:34PM 0 ESCAUX net.PBX, new template with autoconfig of all major IP Phones
12:06PM 1 channel.c: Avoided initial deadlock for '0x8de2dc0', 10 retries!
11:52AM 0 Linksys and Call Park
10:09AM 1 OT: Changing Cisco tftp root directory
7:31AM 2 prob with star input agi-bin
7:25AM 1 Queue Management
5:25AM 5 Problems with Hangup
3:53AM 0 Re: ESCAUX net.PBX registration and boot sequence (was Re: [asterisk-biz]ESCAUX releases net.PBX Free Edition)
2:14AM 9 CallerID is not displaying for my incoming calls
1:44AM 3 Associating an Originate Request to a Channel before the call is answered
1:25AM 1 queue announcements when using ringback
Sunday August 13 2006
9:20PM 1 Queue Monitoring Broken?
5:08PM 0 ANNOUNCEMENT : Asterisk2Billing V1.2.3 (BrainCoral)
4:09PM 0 Astribank
9:36AM 2 Callback in within voicemail broken
7:58AM 0 abhishek invites you to join Zorpia
4:23AM 1 CDR Variable
1:56AM 15 Macro inside macro
1:06AM 6 911 Testing
Saturday August 12 2006
9:02PM 0 Re: asterisk-users Digest, Vol 25, Issue 35
1:59PM 0 problem with mfcr2 protocol
12:07PM 0 OT: Call For Papers -- 2007 Southern California Linux Expo
10:48AM 3 Issues compiling addons on Fedora Core 3
9:38AM 0 Declined to talk, Call rejected: 603 Declined
7:27AM 1 SPA3000 dialplan coding...
Friday August 11 2006
9:31PM 4 SIP header challenge
7:30PM 2 safe_asterisk to start latest version from SVN - trying asterisk with googletalk
7:09PM 0 SuSE 10.1 zaptel init script
4:14PM 3 AgentcallbackLogin()
3:45PM 6 Abstraction for a newbie
2:14PM 1 multiple offices / hard phones / service provider
11:51AM 1 jitterbuffer SIP-IAX possible?
11:34AM 17 Inbound Calls & SIP/2.0 404 Not Found
11:23AM 0 GXP-2000 Call Transfer Problem
10:52AM 12 Auto retry on Busy
10:38AM 2 Fast busy signals... Satisfying my curiousity
10:13AM 1 Call transfer issues
10:09AM 0 Callback feature in voicemail broke?
9:46AM 1 Digit timeout on Asterisk Assisted Transfers
9:43AM 0 Bind Ounbound SIP Trunk to second virtual IP on server
9:17AM 0 Connecting to another server
9:10AM 0 Agent Transfer Locking up Queue() Application
9:01AM 0 Odd Busy tone on Aastra phones
8:54AM 2 MailboxExists not branching to n+101
8:52AM 1 DTMF-CallerID on POTS
6:28AM 0 Has anybody a usefull example for the DIAL-option G(context|exten|prio)
6:13AM 3 USA Toll Free
6:03AM 17 Unable to receive Incoming calls to my DID. Please tell me the solution
5:41AM 10 Asterisk IAXmodem HylaFax?
5:04AM 2 Polycom just disconnects
5:03AM 1 Asterisk GUI tool needed
3:08AM 2 In CDR record not what I want
2:53AM 3 High Availability with PRI failover
2:48AM 5 Port Forwarding SIP rtp
2:24AM 0 where/when to set__TRANSFER_CONTEXT ?
1:38AM 0 question about oh323 and ring tone
Thursday August 10 2006
11:26PM 3 No path to translate from SIP/30-09df8928(4) to SIP/3470075XXXX-09dfb518(256)
9:44PM 0 Phone number lookup public database
7:56PM 5 Quick One - PHP Script to restart Asterisk
2:26PM 13 Correct syntax for Set(CALLERID(all)...
2:21PM 2 A good price for FXS 48 ports?
12:13PM 0 MINNESOTA: TwinCities Asterisk Users Group - Saturday August 12th - 11:30am
11:01AM 1 Snakes On A Plane using Asterisk?
10:59AM 0 IAXy can't connect to analog phone
10:40AM 1 Asterisk Voicemail Setup
10:23AM 0 Odd IAX stats
10:20AM 4 sms callback?
8:58AM 7 Set DID?
8:27AM 1 Samsung Prostar DCS
7:59AM 6 can i detect a voice with asterisk ?
6:50AM 5 Fwd: Dropping incompatible frame killing Asterisk
6:16AM 0 Question on iax2 show netstats
5:09AM 0 Help with newbie: D-link admin setup
4:52AM 1 Realtime SIP Authentication
4:49AM 0 Clipped audio at beginning of SIP calls.
3:18AM 1 Sangoma A101 problem
1:43AM 2 transfer call von D-channel
1:17AM 1 Iptables ,rtp
Wednesday August 9 2006
9:48PM 7 SIP trunks: order or type
7:48PM 1 Callback and Asterisks
7:44PM 6 Warning - doesn't deliver - stuck in a hole
6:08PM 1 How many digits are collected
1:51PM 1 Mac Address Authentication Methods
12:27PM 0 wildcard always busy
12:26PM 2 Ignoring the # key on a call
12:13PM 0 DTMF codes in feature.conf not comming through
12:00PM 5 can Digium FXS channels support been half mile to 1 mile length away from phone?
11:50AM 1 BriStuff | HFC-S | Progress() | Early B3 on incoming calls from PSTN
11:10AM 0 High Availability
10:50AM 2 RE: Ever donate Software to Digium? If you did your afool.
10:38AM 6 Integrating Asterisk with a Panasonic D500 using MFC/R2
10:23AM 0 Autoreply: Re: Autoreply: Tri-Link Technologies?
10:18AM 4 Autoreply: Snom MWI
10:11AM 0 Autoreply: How to, astcc and virtuemart
9:56AM 3 Autoreply: Tri-Link Technologies?
9:55AM 2 Snom MWI
9:54AM 0 How to, astcc and virtuemart
9:54AM 0 Autoreply: e&m wink, TE110P, * answers too soon
9:51AM 0 Autoreply: Deployment for less than 10 phones
9:50AM 0 Tri-Link Technologies?
9:49AM 2 e&m wink, TE110P, * answers too soon
9:46AM 0 Deployment for less than 10 phones
9:32AM 2 ESCAUX releases net.PBX Free Edition
8:48AM 0 Prague PTT?
6:39AM 0 Jabber: Difference between client and component
6:20AM 10 Sipura SPA-3000 vs Sangoma A200
5:11AM 2 Phone Newbie Questions
4:44AM 0 Ever donate Software to Digium? If you didyoura fool.
3:05AM 3 Two card NT-TE mode
12:29AM 0 UK mobile "reject" codes
12:17AM 0 FW: problem with queues
Tuesday August 8 2006
11:59PM 4 realtime+mysql
10:11PM 13 Ever donate Software to Digium? If you did your a fool.
8:13PM 0 hints causing hang in reload
6:52PM 1 Handling inbound and outbound calls passed from a proxy
6:09PM 2 Sangoma A200D and DTMF Detection
5:10PM 0 polycom config script
2:47PM 0 ARA & Regseconds
12:34PM 7 ${BLINDTRANSFER}->accountcode ?
12:21PM 0 MP-124
11:44AM 1 Possible to To Have Different Outgoing VM Messages, but One Mailbox?
11:24AM 5 HFC-S Cards in the UK
10:33AM 1 Host failed to authenticate as teliax
10:20AM 0 IAX Trunking Only ONE-WAY
10:10AM 6 polycom headset question
9:47AM 3 PRI Connection in Lima, Peru
9:37AM 16 Polycom 1.6.7 firmware?
9:26AM 3 A question about AGI and RECORD FILE
8:47AM 1 Asterisk and failover
8:37AM 0 Zaptel trunk failed to compile - Still but another error
8:21AM 0 Cisco Phone Configuration Tool cannot find Files
7:58AM 0 Probelm with IAX peers
7:53AM 0 Stopping Queue after nobody picked up the call . .
6:47AM 0 V: IAX trunk behing NAT with dynamic IP
6:45AM 3 Jitterbuffer on SIP
6:38AM 0 Asterisk with BT's broadband voice service.
6:07AM 0 help with app_sms and chan_capi
5:16AM 2 SV: IAX trunk behing NAT with dynamic IP
5:09AM 0 IAX trunk behing NAT with dynamic IP
4:08AM 2 coredump in SVN trunk
3:46AM 4 RE VoipNow 1.2.0 Beta
3:07AM 7 Problems with Codecs in Asterisk
2:55AM 7 AGI doesn't execute PHP5 script
2:30AM 3 set minimum iax jitterbuffer
1:06AM 0 ISDN Y cable
12:50AM 3 Bluetooth phone as FXS/FXO with asterisk?
Monday August 7 2006
11:25PM 1 problem- 0:10 long message
7:33PM 3 Polycom 301 and Linksys SRW224P PoE Switch
6:49PM 6 agi script runs even if no answer
6:49PM 0 (no subject)
3:16PM 1 Music On Hold Class Not Makin' Sense
2:53PM 0 NVFaxDetect and 1.2.10
2:45PM 0 SIP musicclass
2:25PM 1 Re: Meetme chat room with many users, and only 4 can talk, is there a max amount of users?
1:51PM 0 Voicemail Platform
1:44PM 1 voicemail in mp3 format
1:08PM 0 FXS gateway/Channel Bank
12:02PM 3 sip incoming stop working, what to look for in logs?
11:56AM 1 MOH Silence
11:32AM 0 "Off-circuits are busy now. Please try your call again later"
10:15AM 1 res_sqlite problems
9:50AM 5 E1 for Voice and Data with MFC/R2
9:15AM 1 looking to pay a consultant to help with my asterisk installation
9:14AM 11 By week extension dialing
8:35AM 3 Fwd: * and GTalk testing
8:28AM 10 Hotels...
7:42AM 0 Re: [asterisk-dev] Tuning Software Echo Cancellers
7:16AM 2 Conditional branching
6:47AM 2 Ragi without rails possible ?
6:43AM 5 DTMF problems
5:37AM 2 Re: [asterisk-dev] Questions regarding g.729 and g.711 in Asterisk
5:07AM 0 Caller ID problem on TDM400 FXO
5:01AM 0 Inbound problems, no audio
4:25AM 0 G729, IAX, polycom - trying to using 2 codecs
3:51AM 1 Ztdummy - No audio in BackGround function
3:35AM 1 SER + Asterisk PSTN calls don't hung up
1:27AM 3 Video Conferencing over Asterisk
12:47AM 1 New people in this world and his problem with ISDN
Sunday August 6 2006
8:55PM 2 HP ProLiant and Digium 24xxp
7:31PM 3 How to "emulate" Music on Hold in a PHP AGI script?
5:28PM 4 Variables sip redirects and call forward
3:57PM 1 Anyone use for a click to call app? - Doesn't seem to want to work for me
12:10PM 0 AG-168V not registering.
11:36AM 4 Ring Groups
10:21AM 0 previous reload of asterisk did not finish
6:03AM 2 Using a DB for Configurations
Saturday August 5 2006
11:59PM 2 for some of my users, VoiceMail is being cutoff when leaving message
8:17PM 1 Linksys SPA-3000 Administration Guide
6:16PM 3 g729 and trafic
5:43PM 3 Help with perl AGI script
2:18PM 4 how to check the status of a channel
10:26AM 0 load average with MOH
8:45AM 1 Japanese Sound Files
8:26AM 0 [Solution] Call Asterisk from GoogleTalk and have it tell you the status of your IAX2 links.
6:31AM 12 Fax tone detected, but no fax extension for CAPI
4:13AM 4 cisco 2600
1:19AM 2 Help - call recording being cut short if transferred
12:09AM 3 autocreatepeer in iax
Friday August 4 2006
7:07PM 2 Check call duration on active call in CLI?
3:59PM 8 Setting CALLERID on a residential telco line
3:55PM 0 Aastra VLAN issues
3:29PM 0 How to play music on hold from within PHP AGI scripts?
3:17PM 1 Simple config question
2:20PM 14 Asterisk and Siemens Legacy PBX
1:56PM 0 DISA + Voicemail + DTMF
1:31PM 0 Mediatrix 1204 and Asterisk 1.2.10
1:19PM 1 Running AGI in background
12:56PM 1 Steve Totaro I am trying to reach you.
11:45AM 1 Dialplan routing based on CallerID
10:59AM 1 Problems with monitor / mixmonitor stopping if using Local channels
10:51AM 1 Is the manager good for high traffic?? but only with one connection to it
10:49AM 1 Festival Not Working
10:11AM 0 Jabber questions
9:38AM 1 AgentCallBackLogin+Queue
9:15AM 8 (OT) DS3 Barrel/T-connector/Adtran MX2800 Problems
8:23AM 1 Sangoma A200 and Disconnected Cables
7:57AM 18 SIP/Qualify
6:01AM 0 sendtext() to another machine
4:03AM 0 SV: Help debugging strange asterisk behaviour (update)
3:58AM 2 speech gaps with iax2
3:23AM 5 Load balancing of IAX2
3:17AM 1 How to connect Snom softphone from my home?
3:16AM 1 Configuring meetme recording quality (8kHz to 32kHz or higher)
2:10AM 0 ANI agi
1:33AM 1 Asterisk with AVM B1 and HFC
1:01AM 2 asterisk dosenot compile
12:19AM 0 Asterisk@Home, call reporting and performance
Thursday August 3 2006
6:56PM 0 Opinions on Rhino PCI FXO cards
4:03PM 0 New UK prompts
3:48PM 1 Problem dialing out with a TDB400P
3:43PM 0 Encoding recorded queue calls to mp3
3:23PM 0 trinary expression
3:09PM 6 Prevent a Polycom contact list to be overwritten
3:01PM 1 Detecting voicemail from CO on FXO
1:51PM 1 Echo cancell
1:33PM 7 Run a script at certain CLI writes
1:32PM 3 Using Flite in a call file.
12:16PM 3 Detecting voicemail from CO on FXO port
12:01PM 1 Detecting voicemail from CO on FXO port andpassing to H.323 phone. Possible?
11:48AM 9 How to forward a call to an outside line
10:49AM 0 Queue bug: When 2 callers call in, only one is processed until the first is answered
10:22AM 0 Forbidden - wrong password on authentication for INVITE
9:29AM 2 IAX Variables
9:19AM 2 VoiceMail being cutoff when leaving message
8:58AM 0 Reboot Mediatrix
8:39AM 1 MoH native volume
8:21AM 1 wip 300 opensource code - changes to support SIP MESSAGE
7:20AM 2 Ringing all extensions
6:19AM 0 volume adjustment?
3:47AM 24 What I can use with ASTERISK to call clients to remind them about their appointments
3:29AM 2 How to check if channel varaible have been set/not empty?
3:18AM 2 queue in realtime
3:16AM 2 Garbled initial voicemail prompt
1:14AM 1 IAX2 Trunking CPU usage
Wednesday August 2 2006
9:48PM 2 [OT] FYI: Polycom phone intermittent disconnects
6:38PM 4 About Digium cards and HP DL servers
6:13PM 1 Asterisk 1.0.1 in SuSE 10.0
5:58PM 1 Subject: Slow dialing from PBX via E1
5:01PM 0 IP JitterBuffer for 1.2.5
3:00PM 6 need dialout help in python script
2:28PM 0 RE: Asterisk with VoIP phone (shadowym)
2:10PM 0 chan_zap.c: Failed to read gains: Invalid argument
1:10PM 0 Ateus Easy gate call progress
11:45AM 6 creidt card processing sripts for asterisk
10:56AM 1 IAX Trunking
10:46AM 4 Rookie question, trying to learn
10:44AM 0 GSM analogue router
10:36AM 0 Strange behavior with SIP registration/connectivity
10:10AM 1 asterisk optimizing
10:00AM 0 DTMF intermittent on menu.
9:54AM 3 canreinvite=yes and RTP dropping in and out
9:50AM 6 Limitations of IAX
9:27AM 6 SIP_HEADER() read-only
8:25AM 0 - Coder Needed for Patch
8:22AM 1 Issue with IAX2 and Real Time configuration
8:09AM 3 Arrays ???
7:52AM 4 Asterisk, Linksys SPA-3000 echo
7:07AM 0 Playback() does not work
7:06AM 2 Call Routing based on Caller-Id
7:01AM 0 Dell Poweredge 1950 / 2950
6:50AM 7 polycom soundstation 501 crash
5:52AM 4 FXO module burn out !?
5:42AM 0 Follow ON calling on DISA
5:03AM 0 newbie - suggestions on installing Asterisk for SOHO
1:08AM 0 Slow dialing from PBX via E1
12:29AM 1 Asterisk config with Analouge Audio Codec model number MP108FXS
12:27AM 2 Problem with Cisco7970 SIP load / call transfer
12:08AM 0 SV: VOIP phone for Receptionist use
Tuesday August 1 2006
10:51PM 1 SER local as an Asterisk Trunk
7:35PM 0 softhangup() problem
7:05PM 0 A2Billing - destination
5:44PM 0 ANNOUNCE: libss7
5:23PM 0 RE: asterisk-users Digest, Vol 25, Issue 2
5:20PM 10 VOIP phone for Receptionist use
5:03PM 2 Asterisk with VoIP phone
3:52PM 1 Polycom IP600 HTTP Provisioning problem
2:36PM 8 Unicall stack, right versions?
2:05PM 4 Dundi and Dial Arguments
1:28PM 1 rx_fax problem
1:13PM 1 ISDN incoming call - inband info and announcements BEFORE ANSWER
12:52PM 0 Line drops
12:28PM 0 MySQL 5.0+ and the MySQL addon - Can use stored procedures?
11:14AM 6 IAX and Accountcode
9:59AM 0 Codec selection / IAX tunnels
9:54AM 0 Controllable hold music
9:45AM 1 Extend analog phone via SIP (OT)
8:52AM 0 Park / ParkAndAnnounce
8:46AM 1 Problem with distortion of initial voicemail prompt
8:27AM 3 SV: Help debugging strange asterisk behaviour
8:16AM 0 Media direct from IAX Phone to IAX Phone
8:11AM 1 Help debugging strange asterisk behaviour
8:01AM 2 Missing Fast AGI calling 'h' exten without hanging up
7:22AM 1 AddQueueMember and Local channel
7:16AM 4 *****SPAM***** Is there a smarter way to ban expensive calls in dial plan?
6:24AM 2 nat and qualify questions
6:02AM 0 SoftHangup with Polycom_acd_functions release of asterisk
4:40AM 0 Permission for files generated by voicemail
2:48AM 0 SRTP help