asterisk users - Aug 2008

Sunday August 31 2008
11:09AM 4 Asterisk IVR Scalability
2:56AM 0 Intermittent "rejected because extension not found" On Incoming DID
12:15AM 2 security on localhost connections
Saturday August 30 2008
8:55PM 0 Re: Reliable wireless SIP phones (Tzafrir Cohen)
5:32PM 3 beta9: how to set callerid on incoming iax?
5:17PM 6 Congestion in Outgoing call through PRI
4:51PM 5 Wi-SIP & 802.11f - Inter Access Point Protocol HANDOFF
4:31PM 2 Heist of MagicJack SIP credentials?
7:05AM 0 Incoming Calls via SIP Trunks
1:49AM 0 Zap channel DTMF regeneration
Friday August 29 2008
9:46PM 1 Audio data between concurrent SIP and PSTN
7:57PM 0 CDR userfield recording name
6:14PM 4 Call monitor/barge/train
4:44PM 1 music on hold is not working
4:39PM 1 Issue when dialing multiple extensions using & ------Please Help
3:52PM 0 chan_mobile
3:35PM 2 Connecting two asterisks via IAX
3:17PM 17 Faxing through Zap cards
3:03PM 0 track 1.6 progress
2:58PM 12 Wi-SIP vs. SIP-DECT
9:35AM 0 Asterisk cdr_mysql inexact values
4:56AM 3 Asterisk CDR Problem
Thursday August 28 2008
11:50PM 1 Is including a linefeed in the JabberSend message possible?
9:53PM 10 Transfers on AgentLogin()
5:44PM 1 troubleshooting mISDN...
5:40PM 4 GSM recordings
5:24PM 0 Caller ID in IAX trunk, SIP trunk, between extensions and from FXO
5:12PM 6 sip conversations overlapping!!!!
5:08PM 0 meetme + jitter buffer
4:07PM 0 VoicePulse Time out?
2:32PM 1 asterisk linkedin group
2:11PM 0 OT: SEP<mac addr>.cnf.xml file for 7911 with SIP 8.3.5 firmware
1:06PM 30 Reliable wireless SIP phones
11:26AM 11 Asterisk Queue's
11:18AM 0 Weird asterisk error: ztscan command not found
10:59AM 4 Problems with DTMF on IVRs
9:16AM 7 Console softphone
8:32AM 5 H323 protocol
7:22AM 1 execute command after sip register
7:08AM 5 Asterisk 1.4 -> 1.6
6:10AM 1 Asterisk CLI Show Error :- ("**Unknown**") instead of ("Zap/22-1", )
4:01AM 3 can not load from asterisk!
2:02AM 8 remove queue call
1:06AM 8 Pri to sip interfaces
Wednesday August 27 2008
10:43PM 0 Asterisk and Linksys One (PHB1100)
9:05PM 1 OT Polycom URI and IP address dialing. Not.
6:07PM 5 Off-Hook (type II) CID passing to Asterisk via Linsys/Sipura
5:08PM 7 Problem with Call Forward
3:20PM 1 Fax issue over cisco gateway
2:50PM 1 Callback voice Quality
2:46PM 3 VUC Friday: asterisk 1.6 fax, Drawing for Free Astricon Pass
2:16PM 1 problem making outgoing calls
2:13PM 0 sip show peers from shell or from CLI [SOLVED]
1:55PM 1 Call Files
1:27PM 28 PRI Splitter
12:00PM 6 sip show peers from shell or from CLI
8:53AM 0 asterisk-1.6, Remote-Party-ID Header not sent
6:09AM 1 compile Dahdi !
4:44AM 1 Digium Coffee anyone? PCI Expresso? WTF?
4:41AM 0 Asterisk for calling no of users
12:28AM 6 Need application, CID number match list to call cell phone
Tuesday August 26 2008
10:55PM 1 FreeTDS Versions?
9:56PM 2 app_jack and calling with pc only
4:24PM 6 X100P Card in OFFHOOK state
3:49PM 0 te410p remains in red-alarm
2:49PM 3 Asterisk connected to the PSTN vs. a commercial solution
11:49AM 9 Limit to the length of string ?
8:51AM 0 Asterisk/Other PBX interconnection
5:55AM 0 about iax2 prune realtime
1:55AM 11 implementing an intercom with asterisk
Monday August 25 2008
10:33PM 10 is shared_lastcall available in 1.4
5:29PM 0 sip.conf templates and realtime
5:12PM 2 How to query a remote MySQL DB from dialplan
4:32PM 2 Get call status and hangup
3:26PM 3 Call transfer over IAX trunk
1:46PM 0 Problem with dtmf in voicemailmain
1:06PM 18 sip peering between 2 asterisk
12:08PM 2 asterisk realtime
11:23AM 1 Static IP for SIP?
10:51AM 4 TDM2400P Voice Quality Problem
10:26AM 5 Really WEIRD: can register but can not call!
9:41AM 0 Problem using blind transfer
8:50AM 0 A question about the ${CHANNEL}
6:47AM 0 Consequent Dial commands not ringing
6:31AM 0 ASM / AMI Assisted Live Transfer
6:10AM 3 Read Command
2:13AM 0 wct4xxp alarmdebounce
Sunday August 24 2008
11:25PM 0 RemoveQueueMember race condition
4:34PM 0 [Xen-users] Xen 3.2.1 and PCI passthrough
4:29PM 6 MWI working perfectly. Shouldn't it be broken??
4:13PM 0 Asterisk - forcing PSTN Line "ON" Hook
3:26PM 6 entering a password to have access to a sip account?!
9:50AM 0 Realtime SIP
6:51AM 0 RTP timestamp modification during SIP video call
4:36AM 6 Question about Dialing DTMF
Saturday August 23 2008
4:19PM 1 Anything to convert from JSON into Asterisk dialplan variables?
3:16PM 1 Asterisk, 2-way radio systems, app_rpt and chan_rtpdir
1:43PM 7 Semi-OT Satellite?
11:34AM 0 Free Today !
10:30AM 16 OT - Which rackable case for mini-ITX boards ?
9:49AM 4 Problems with D-channel (PRI)
8:56AM 5 Global VoIP Calls?
6:53AM 0 Blind Transfer is not working in incoming calls
6:39AM 1 Voicemail has issues with DTMF
Friday August 22 2008
11:28PM 1 ztd-ethmf
11:00PM 1 frequent channel reset problem
10:25PM 1 Fw: [asterisk-dev] frequent channel reset problem
6:40PM 1 em wink
6:26PM 3 queue timeout
3:39PM 0 RES: DSS1 vs SS7
3:06PM 0 Friday's conference meeting - Astricon is in the air
2:55PM 2 Diamondware spatial conferencing
12:08PM 0 Asterisk, Xen and a TDM400P
7:37AM 1 interesting RDNIS question
7:28AM 6 set callerid with plus sign
6:34AM 2 About the CALLIDNUMBER of the fxs
3:10AM 9 Linksys - Sipura VMWI splash ring
12:34AM 0 The problem of the fxs
Thursday August 21 2008
11:45PM 3 How to block incoming calls on PRI
10:03PM 8 Suddenly the voice become like robot (cutting), like sick man
5:04PM 1 ultramonkey and asterisk
4:50PM 5 Siemens Gigaset IP in USA (S685 IP in particular)
4:50PM 9 5 min limitation on phone calls! how to!
4:40PM 2 Question: Soft phone for ACD agents?
4:02PM 2 Asterisk and Huawei SoftX3000
3:39PM 4 Automatic call to voicemail on login?
3:35PM 7 After Dial execution, using DIALEDTIME, ANSWEREDTIME
3:19PM 2 callfiles/manager api originate call fails
3:08PM 10 Asterisk Realtime pounds MySQL
3:03PM 0 1st call after some time has one way speech, but calls after that are fine..
2:47PM 7 A Suggestion To Asterisk Appliance Developers
2:29PM 3 Anyone using asterisk on centos 4.X without hardware cards and using console/dsp
1:26PM 2 OT - Asterisk-Stats - Billsec instead of Duration
12:16PM 2 How can I determine if IAX trunking is being used and how many calls are being trunked?
12:11PM 6 Changing callerID in a context
11:57AM 3 IVR question
9:52AM 0 Any chance this is related to fastagi: received mini-frame before full voice frame
6:36AM 8 DSS1 vs SS7
2:06AM 1 The problem of the ${CALLERID(num)} for the fxo
12:06AM 1 Problem with Qualify sip peers...
Wednesday August 20 2008
11:29PM 3 Asterisk build-environment in Xen-DomU
7:53PM 0 2BCT from the Asterisk side
7:09PM 4 Two peers, same IP and port
5:21PM 1 3-way conference call
5:00PM 17 Is there a way to encrypt passwords stored in the realtime database?
1:04PM 1 Reproduce DeadAGI behavior with AGI
12:21PM 0 Why does a perfectly fine iax2 host becomes UNREACHABLE?
12:06PM 14 FTC Bans Prerecorded Telemarketing Drivel
10:37AM 2 vicidial mysql problem
9:55AM 0 [Question]Setting CallerIDName using Asterisk Manager Interface API
9:33AM 0 how to cross compile asterisk for cirrus edb9302a board
9:12AM 0 IAX2 and transfer=mediaonly, Error unable to transfer but there is sound.
2:14AM 5 Linksys SPA3102-NA firmware upgrade on Linux
Tuesday August 19 2008
3:59PM 8 Missing 'full' log
3:09PM 0 Free US Based Echo Test
11:57AM 6 Help with Asterisk to Huawei SoftX3000 registry problem
11:10AM 7 Inefficient Codec Translation
8:06AM 0 Allow asterisk to receive calls
6:09AM 2 GUI: User account for SIP Provider results to "Punctuation and Special Characters are not allowed in this field."
5:54AM 0 asterisk-users Digest, Vol 49, Issue 44
2:45AM 3 Gnudialer runninig
Monday August 18 2008
7:59PM 5 opening Doors with Asterisk!?
5:06PM 0 asterisk-users Digest, Vol 49, Issue 43
4:41PM 7 US-based echo test servers?
4:30PM 2 Asterisk Stops...where to look?
4:03PM 5 Strange putconfig bahviour
7:23AM 3 Voicemail
1:34AM 1 ZTDUMMY Running but IAX2 message:Unable to support trunking on peer 'XXXXXXXX' without zaptel timing
Sunday August 17 2008
10:26PM 0 asterisk -n switch
5:04PM 0 asterisk-users Digest, Vol 49, Issue 42
4:56PM 0 1.6 call-limit
4:07PM 3 pollmailboxes
8:51AM 2 Running asterisk as non root user
Saturday August 16 2008
6:47PM 1 disable auth between two asterisk
5:08PM 0 How to create a user that uses mISDN using Asterisk GUI
5:04PM 0 asterisk-users Digest, Vol 49, Issue 41
3:53PM 4 Which is the correct GUI vor Asterisk 1.4 ?
1:57PM 1 dialplan reload and dropped calls
8:36AM 0 Getting cdr(billsec) 0 -- please help
5:02AM 0 asterisk-users Digest, Vol 49, Issue 40
4:57AM 1 Maybe a crazy idea, but are there Asterisk hoster outside there?
1:50AM 0 Basic outbound calling issue : a lot closer
Friday August 15 2008
9:22PM 2 Asterisk AGI and php problem....
8:17PM 2 dahdi link broken
7:48PM 0 asterisk-users Digest, Vol 49, Issue 39
7:06PM 5 Problems assigning mISDN Trunk using the DIGIUM Asterisk GUI
5:52PM 22 PRI TBCT - Practical Experience, Anybody?
5:37PM 0 Incoming Bogota DID
5:27PM 1 Cisco 7960 audible hold reminder?
5:04PM 0 asterisk-users Digest, Vol 49, Issue 38
4:52PM 3 zaptel timing
4:43PM 5 dahdi and ztdummy
2:49PM 6 DID's needed for Reston Virginia - + hosted asterisk
2:28PM 2 Problem with Aastra 480ci and qualify=yes
12:18PM 0 asterisk-users Digest, Vol 49, Issue 37
11:59AM 2 noise cancelling headset vs handset
5:56AM 4 AstDB/Berkely DB - Hash function? Balanced-Tree? b-Tree? Linked List?
5:31AM 11 Asterisk stress call test
3:16AM 0 asterisk-users Digest, Vol 49, Issue 36
3:03AM 0 integrating SipTheeSkype in asterisk
12:53AM 6 asterisk realtime and creating "new" contexts
Thursday August 14 2008
11:17PM 0 Comdial IP thru Asterisk to cell
8:22PM 0 Astricon approaches and more tomorrow at 12 Noon EDT on VUC
8:09PM 7 USA Lata AreaCode Database
7:49PM 4 Asterisk vs c-client issues
6:35PM 8 ANSI terminal colors
6:22PM 0 asterisk-users Digest, Vol 49, Issue 35
5:58PM 0 [asterisk-dev] Asterisk + OpenIMSCore
5:19PM 2 Still have a callno...
5:15PM 0 asterisk-users Digest, Vol 49, Issue 34
4:06PM 5 AMI and extensions.conf
3:55PM 0 asterisk-users Digest, Vol 49, Issue 33
8:25AM 1 Accept anonymous connections from Unknown Peer
5:30AM 0 asterisk-users Digest, Vol 49, Issue 32
4:51AM 27 Unable to create ZAP Channel
1:46AM 1 Asterisk 1.4 T38 UDPTL Pass Through MAX TNT and Linksys 2102
Wednesday August 13 2008
11:48PM 0 Callwaiting and CallerID on ZAP PSTN Line
11:44PM 1 Cisco 7960
11:40PM 18 Asterisk might be dropping RTP packets before reaching eth int?
10:28PM 1 ztdummy on centos 4.6 i386
10:20PM 3 Open door automatically...
9:24PM 0 Asterisk stops sending RTP packets to ethernet interface
9:06PM 6 New GUI for Realtime Asterisk - RAGUI
8:03PM 2 rtc issue
7:09PM 4 seeking hardware recommendation PCI versus PCI Express E1 card (te407p vs te420bf)
5:17PM 0 call forward spa 841 and asterisk 1.4.21
5:05PM 0 asterisk-users Digest, Vol 49, Issue 31
3:41PM 6 ENUM lookup
2:56PM 1 Asterisk and Radius
2:32PM 0 asterisk-users Digest, Vol 49, Issue 30
2:31PM 0 Decline message
2:30PM 1 Decline issue
1:56PM 11 Sending Set Asynchronous Balanced Mode Extended
9:50AM 8 cmdRecord issue related to "iax2 received mini frame before first full voice frame"?
5:22AM 1 Off Topic: Disable Polycom Soundpoint DoNotDisturb Feature
1:35AM 0 asterisk-users Digest, Vol 49, Issue 29
Tuesday August 12 2008
10:45PM 7 BLF functionality
9:32PM 1 Error after svn co of lastest zaptel 1.4
8:42PM 3 LNP Problems
6:45PM 0 Problems with queue member status
6:27PM 8 OT: Asterisk on fitPC
5:30PM 7 I used to use an Asterisk server, but now it is overkill, ...
5:14PM 0 asterisk-users Digest, Vol 49, Issue 28
2:48PM 1 intermediate accounting records
2:44PM 1 distinctive ring on sipura
2:39PM 4 CDR accuracy
1:29PM 2 Asterisk issue
12:48PM 0 DTMF is Not working in VOICEMAIL
12:28PM 3 Unable to compile asterisk-addons from trunk
9:26AM 0 InBound call Barging
3:17AM 0 park calls - cannot hear digits being played
2:14AM 0 asterisk-users Digest, Vol 49, Issue 27
Monday August 11 2008
10:59PM 1 Intermittent T.38 pass through
8:38PM 4 phone rings "once" before playing message
6:34PM 1 Phone system layout suggestions
5:07PM 1 Asterisk Realtime CLI command
5:04PM 0 asterisk-users Digest, Vol 49, Issue 26
4:21PM 2 Asterisk broadcast to web
4:19PM 0 asterisk-users Digest, Vol 49, Issue 25
4:10PM 2 Originate Status Monitoring
2:50PM 0 out going call files and correct dial status
1:46PM 4 Asterisk Realtime Unregister
11:08AM 2 1.4 SVN / dahdi / meetme / -> unable to open pseudo device
10:15AM 0 Found unknown media description format
9:20AM 1 DCAP on Linuxtag - Berlin 2009
8:01AM 3 deadalocks in asterisk
7:03AM 4 The problem DIAL with option T,t
Sunday August 10 2008
6:31PM 2 H323 Issue
5:07PM 0 asterisk-users Digest, Vol 49, Issue 24
Saturday August 9 2008
10:30PM 0 Explain t38 and how it relates to Asterisk
6:30PM 1 asterisk - iaxmodem - create_addr: No such host: ttyIAX
5:08PM 0 asterisk-users Digest, Vol 49, Issue 23
10:16AM 2 New FXO/FXS interface needed..
9:23AM 0 Building Asterisk- on Ubuntu 8.04.1 LTS x86
6:53AM 3 cdr-custom rotate?
6:01AM 1 how to know what codec is being used
5:58AM 0 asterisk-users Digest, Vol 49, Issue 22
5:47AM 1 channel_find_locked: Avoided initial deadlock for
Friday August 8 2008
10:02PM 2 VICIDial error
8:46PM 1 can I get a little criticism?
8:31PM 1 SIP TLS error: ast_make_file_from_fd: FILE * open failed
8:27PM 1 Auto Dialer proof of concept
8:23PM 9 IAX2 encryption - LAN. no, INET: yes???
7:19PM 2 System call never returns
7:17PM 1 chan_mobile: scrambled audio, no MOH, no call signalization
5:31PM 5 Semi-OT: ServerBeach for VoIP
5:07PM 1 asterisk-users Digest, Vol 49, Issue 21
4:30PM 5 channel variables not kept
3:16PM 2 Zap channels stuck...
2:58PM 0 Asterisk configuration
1:08PM 7 manager/originate
12:17PM 0 asterisk-users Digest, Vol 49, Issue 20
9:48AM 6 asterisk-addons-1.6.0-beta4 compile error
6:11AM 1 h323 channel compile error
Thursday August 7 2008
10:47PM 0 asterisk-users Digest, Vol 49, Issue 19
9:55PM 4 AGI and Call Center to do CRM integration
8:04PM 11 Asterisk end-user GUI?
7:31PM 13 FAX t.38 on Asterisk 1.6?
5:12PM 0 asterisk-users Digest, Vol 49, Issue 18
5:04PM 6 FWD $30 membership-fee
4:46PM 4 BRI AND DATA connection
3:49PM 0 asterisk-users Digest, Vol 49, Issue 17
3:37PM 3 Improving the speed of chan_sip
1:32PM 4 outgoing call file and agi detect busy
1:24PM 1 Voicemail on PRI
12:39PM 0 I canĀ“t hear the warning sound in Dial command
10:45AM 2 problem controlling dialplan order
9:34AM 0 asterisk-users Digest, Vol 49, Issue 16
9:20AM 0 [HELP] Regarding stripping of fmtp parameters for Video.
5:28AM 1 Randulo: An open suggestion for the VOIP users Conference
3:35AM 1 Capture digits, set as variable..., use for caller id?
1:11AM 0 Trying to understand Messages from chan_zap.c
Wednesday August 6 2008
9:04PM 6 intercom/paging with grandstream gxp2000
8:08PM 3 Digium B410P: problematic Bri connection between * and a legacy Philips PBX
5:45PM 4 Strange beep during calls
5:40PM 6 Max amount of concurrent calls on and iax trunk
5:04PM 0 asterisk-users Digest, Vol 49, Issue 15
4:47PM 0 Regarding fmtp parameters.
3:40PM 0 asterisk-users Digest, Vol 49, Issue 14
3:24PM 1 Action on login
1:05PM 3 problem with iaxmodem!
1:02PM 7 Transcoding
12:47PM 3 asterisk realtime user deletion
11:34AM 2 OT: Cisco 7961 SIP downgrade from 8.3.3 -> 8.0.4SRS2 failing
9:50AM 0 About the features.conf of it's transfer
9:12AM 2 shared mysql connection in dialplan
9:09AM 2 does astcanary really work?
7:42AM 0 asterisk-users Digest, Vol 49, Issue 13
6:27AM 0 Asterisk takes incoming call before extension was submitted
12:38AM 1 G722 capable soft phone?
Tuesday August 5 2008
11:26PM 0 libpri versions 1.2.8 and 1.4.7, and libss7 version 1.0.1 released
10:10PM 7 email notification to external email address
8:03PM 0 When shall SIP phone reply "480 Temporarily Unavailable"
6:36PM 4 Asterisk to Avaya
4:12PM 0 ZRTP in Asterisk
4:01PM 1 Codec g729 issues
3:47PM 12 vars in Macros called by DIAL with option M
2:49PM 3 Getting Asterisk out of the RTP media path
2:30PM 3 Grandstream RS-232 config (slightly off-topic)
12:40PM 9 Queue Penalties not working properly
8:16AM 3 "Asterisk dead but subsys locked"
5:55AM 2 a simple Asterisk AMI interface with Delphi (or Lazarus+FreePascal)
12:35AM 1 Off Topic: 8x FXO Gateway
Monday August 4 2008
9:28PM 0 Buffer re-sync with Openvox card...
5:36PM 2 in-call start monitoring
3:30PM 2 multiple asterisk approach
3:13PM 2 Building Asterisk-
1:34PM 5 OT: TechShop
12:57PM 6 Asterisk Realtime with MySQL Registration Failed
9:49AM 14 Autoanswer in Nokia SIP clients?
9:04AM 11 Customized Queuing Strategy
8:32AM 0 gsm files instead mp3 files in a conference room!
7:58AM 2 skype and Asterisk opensource integration
4:02AM 3 FC2 and Zaptel
Sunday August 3 2008
10:59PM 0 asterisk-users Digest, Vol 49, Issue 6
10:15PM 2 Bad recorded audio quality (upgrade).
8:06PM 0 Random reboots on IP-601 after changing network topology
5:54PM 4 TDMoE with Telco
5:13PM 0 No MOH on SIP hold nor on park
4:20PM 1 Least Cost Routing
10:10AM 0 SIP Registration
7:05AM 6 PRI device is down
3:30AM 1 lookup for ''
Saturday August 2 2008
9:07PM 0 Asterisk Now - rPath virtual appliance
8:59PM 3 How do I issue a Flash to Zap (PSTN) from SIP?
7:03PM 1 Voxeo
1:53PM 13 2000+ user Asterisk PBX
Friday August 1 2008
8:02PM 2 Cisco 7970, CTLSEP<mac>.tlv
7:16PM 0 sip show peer [load] says not a realtime peer
6:34PM 2 Call Logs
6:29PM 0 app_flite 0.6 released
6:20PM 7 how many quad T1 cards
4:34PM 1 [Dundi] Looking for new peers/limited time only
3:36PM 6 auto provisioning phones
1:43PM 6 HI ~ good friend,
1:29PM 7 Asterisk Queues problem
1:13PM 3 asterisk appliance A50 vs asterisk open source + fxo cards
9:27AM 0 Authentication on an LDAP server
7:57AM 1 Comparing origination from CLI and from AMI
7:16AM 0 Russian Calling card Voice prompt
4:27AM 2 invite from rahul
2:33AM 1 XMPP developers
12:28AM 4 AMI able to call from known endpoint to unknown endpoint?