asterisk users - Nov 2009

Monday November 30 2009
10:25PM 0 Asterisk and XMPP Jingle : testers needed
9:58PM 0 AST-2009-010: RTP Remote Crash Vulnerability
9:50PM 3 Asterisk 1.2.37,,, and Now Available
9:41PM 1 DAHDI - BRI - Astribank
4:42PM 1 Polycom 500 format file system on every reboot
1:18PM 0 UniMRCP Integrated Asterisk Deployment
9:45AM 0 Warning: __ast_register_translator: plc_samples 160 format f/__ast_string_field_init: trying to reset empty pool
8:32AM 2 No application 'ReceiveFAX'
5:32AM 0 Gtalk Asterisk integration
1:29AM 2 AGI stuff
Sunday November 29 2009
5:30PM 1 Asterisk H323 channel and the UDP/TCP rage ports (Q931, H245, T120, RTP)
12:34PM 3 Parsing custom SIP headers
1:22AM 2 VoiceMail greetings
Saturday November 28 2009
10:17PM 1 DAHDI/1-2 v. DAHDI/2-1 ??
1:49PM 2 can't hear anything at incoming calls
12:35PM 0 NvFaxdetect and Asterisk 1.4.27 - Someone get it work?
11:52AM 2 Max how many users in sip.conf
7:17AM 2 Free Polycom Provisioning Tool
Friday November 27 2009
10:27PM 1 Asterisk + Teliax = First Part Of Audio File Playback Cut Off
7:01PM 0 queue hangup
2:50PM 1 Which IP Phone and the codecs
2:33PM 0 Good quality replacement for Linksys SPA-3102 recommendation.
1:50PM 0 Need help with this conf
12:54PM 2 1800 DID Provider - Suggestion
12:11PM 1 Realtime SIP Register
12:08PM 1 Virtual Phone for CDR Logging
9:33AM 1 ISDN30 Timing Sources (Jon Morgan)
Thursday November 26 2009
7:33PM 0 AGI and Music on hold
7:05PM 0 TE420B - CPU usage increase
6:57PM 2 Problem with Portech MV-372
2:13PM 1 Polycom retrieve call from hold
1:41PM 2 TE412P with zaptel
11:47AM 1 app_read does not seem to work with SIP early media (it answers the channel)
11:38AM 1 CDR & Queue
9:32AM 0 GUI for Asterisk+LDAP - testers needed
12:45AM 1 Unable to open sound file error
Wednesday November 25 2009
11:41PM 1 Agent with External Number as Extension
11:01PM 2 Restricting transfers between SIP phones
8:07PM 7 Questions about static
7:57PM 1 Channel Variable
4:58PM 1 office / homeuser
12:42PM 0 asterisk + res_config_ldap = asterisk.core
10:18AM 6 How many lines do you use.
10:09AM 0 DGP 301hard phone incomming problem.
9:59AM 4 ChanIsAvail querry
8:38AM 0 FW: Change the FROM filed username and From
12:08AM 0 Where are documented channel-dependant Dial options ?
Tuesday November 24 2009
11:03PM 3 1950's UK rotary dial phone
9:12PM 1 Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = ' ')
8:24PM 1 Route Non-Call Data to Agent Through Queue
7:05PM 2 Crosstalk - Is there a debug option for logging this?
6:19PM 1 snapgear/mcafee sg560 rebooting
5:21PM 0 Change the FROM filed username and From Calling id in asterisk
4:47PM 2 audio cuts out during IVR
4:18PM 4 Ring group issue
1:21PM 7 keep asterisk in RAM
1:05PM 1 Cianet channel bank with noise and echo
12:12PM 2 IVR for asterisk
12:07PM 3 distribute free call minutes over different channels
8:49AM 3 Experience with LLDP
3:30AM 1 asterisk trunk CURL hangs in the dialplan
2:43AM 1 DIDs > PBX > Multi-channel balanced audio output?
1:25AM 2 can't get pap2 to register from outside the LAN.
Monday November 23 2009
11:22PM 0 Got SIP response 420 "Bad Extension" back from
9:05PM 2 SIP over TCP/TLS for 1.4 branch
8:18PM 0 TDM400P alarm state
6:11PM 2 GotoIfTime problem - possible bug
5:59PM 0 ADSI...
5:41PM 0 best channel driver for 1.4.x and beronet/junghanns 4BRI?
5:40PM 0 Asterisk 1.4 and kernel panic and IRQ interrupts
3:37PM 2 Questions about Voicemail
3:07PM 1 Music On Hold
2:15PM 3 Please some enlightment on ENUM !!
12:49PM 4 Connect Two Asterisk's using isdn Cards
10:21AM 7 Get the extension dailed
10:07AM 1 Is Answer really needed
7:17AM 1 Meetme 'o' - what actually it does..??
12:06AM 2 Yealink SIP-T22P Auto Provisioning via HTTP ?
Sunday November 22 2009
10:54PM 1 End to End delay calculation
7:15PM 1 Portec - feedback wanted
7:06PM 0 Sending call information to handset
5:25PM 1 Wierd problem
4:09PM 1 transferring SIP call: no voice
1:15PM 1 Prevent Dial if any extension is busy
10:46AM 1 Development on top of freePbx Gui and AsteriskNow
5:58AM 0 How do I take out one office out of the call stream?
Saturday November 21 2009
10:34PM 1 Verification number / code
3:18PM 4 DIDs
1:34PM 0 PCI analog cards on * vs. Quintum
7:15AM 3 Connect two Asterisk Server in IAX ?
Friday November 20 2009
11:23PM 1 Cisco 7961 - can't place calls
11:03PM 1 How to change outgoing DTMF frequencies on zaptel?
10:40PM 1 Trasnfer to a different VM box after leaving a VM
9:41PM 1 server unresponsive
8:39PM 1 Problem with blind transfers
8:39PM 1 2.6.31+ XFS - All I/O locks up to D-state after 24-48 hours (sysrq-t+w available) - root cause found = asterisk
5:11PM 1 PHP AGI : handle Event /AGI session
2:21PM 2 Mix of Swedish and English voice prompts
12:36PM 1 Dial Plan Application(main-menu)
12:07PM 1 I don't know how to authenticate
2:04AM 2 Setting up Nokia e71: registration problem
12:05AM 0 Sip phones on localnet AND outside localnet problem
Thursday November 19 2009
10:05PM 0 Dahdi channels interference
9:48PM 3 Newbie
9:18PM 1 Type Of Number setting (pridialplan) is not effective
7:32PM 7 AXVoice Server Hacked.. accounts info leaked
5:36PM 0 Can asterisk PRI/BRI support redirect calls
4:50PM 1 make sounds - doesn't pull all audio tarballs.
2:50PM 1 Meetme
2:46PM 1 Dahdi_genconf replies Empty configuration -- no spans
2:37PM 2 Asterisk 1.4.27,, and Now Available
7:49AM 2 Send the same message to list of users
6:19AM 1 Asterisk crashes : Failed to start PBX
6:01AM 2 Dahdi and Junghanns QuadBRI
5:34AM 1 SIP Calls on Asterisk fails after 25000 calls
3:29AM 2 Gain
Wednesday November 18 2009
8:00PM 0 Off Topic
7:39PM 0 Problem install wctdm24xxxp [resolved]
6:08PM 2 Problem install wctdm24xxxp
5:04PM 0 Asterisk 1.2.18 and meetme causing Audio bleeds
5:03PM 0 AGI and paging
4:47PM 0 Bug CDR report - dst "s" ?
3:21PM 2 Queues without agent login
3:06PM 1 clever ways to "share" an extension between sip and fxs
10:34AM 3 asterisk makes kernel panic
10:29AM 2 Saving CDR on Different Databases
9:37AM 0 question about call transfer
Tuesday November 17 2009
5:27PM 2 New Open Source CTI client for Asterisk
5:20PM 2 asterisk-users Digest, Vol 64, Issue 52
4:33PM 3 newbie question
2:50PM 3 softphone/debug panel with BLF
2:46PM 0 *1.4 Received SIP subscribe for unknown event package: call-info
1:35PM 0 Cisco 7960 md5secret password problem
8:55AM 0 help vxml and asterisk support
8:53AM 3 vxml and asterisk support
4:30AM 1 Cisco 7971 behind NAT
3:35AM 2 max call duration
1:42AM 1 Understanding Congestion to incoming caller
1:22AM 1 Question about OSLEC or HPEC with AsteriskNow
Monday November 16 2009
11:19PM 0 Asterisk VoIP Security Webinar - Video Now Available
9:47PM 1 Pbx-cards
9:24PM 3 Queues
8:59PM 0 SIP Change canreinvite=yes/no from dialplan?
8:55PM 0 Limit IAX calls on a peer, in and out
8:40PM 1 can't call through voip provider
8:27PM 1 Problem with sounds DTMF's phone keys
2:50PM 1 asterisk cdr - remote ip address
2:40PM 1 MixMonitor and Call Latency during conversation
1:24PM 0 ENUM and Asterisk 1.6
12:14PM 2 Security Against brute force attack
12:08PM 1 Problems with dahdi on asterisk with TE122
10:16AM 1 Kamailio and asterisk Integration
10:01AM 2 Odd Local Channel and 0 billsec issue
7:32AM 0 ZAP/DAHDI outgoing faxdetect
6:20AM 1 How to write the incoming stream to pipe/socket instead of .gsm file
12:15AM 1 SendFAX causes restart
12:13AM 0 IAX2 ring cadence / time
Sunday November 15 2009
9:51PM 4 Changing labels on Phones
9:03PM 1 ip source aware Authentication
8:00PM 1 thx fred
7:27PM 4 Hardware Requirement for asterisk
6:05PM 2 Sip incoming call issue with Asterisk 1.6
4:53PM 0 Asterisk cmd Dial, disconnection party is source or destination?
2:33PM 1 Call IAX2 => "Call rejected, CallToken Support required"
1:28PM 1 VeriFone Omni VX-510 Credit Card Machine
11:39AM 1 call log, call detail
6:52AM 3 Database postgresql not able to start
5:31AM 6 Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server
Saturday November 14 2009
7:19PM 0 hi friend
6:47PM 1 Queue application in Asterisk 1.6
5:46PM 1 Brandable SIP SoftPhone (Windows) ?
5:36PM 5 music on hold
5:29PM 0 OT AG-188N
3:20PM 1 Asterisk with T38 Fax
3:09PM 0 Asterisk with H323 channel and Gnugk: no voice
9:50AM 1 Multi-Site GUI
8:20AM 2 Error Dialplan ?
7:59AM 1 Inquiry:Where to download Asterisk 1.4.13 for Debian server?
6:39AM 3 Inquiry:How to stop Asterisk?
Friday November 13 2009
11:55PM 1 Xorcom Astribank udev issue in Ubuntu 9.10
7:55PM 2 Multi Tenant Asterisk Server ?
7:47PM 0 asterisk SIP hangup
7:11PM 2 openSuse 11.2 and dahdi-linux
6:36PM 3 No dahdi_zttools in AsteriskNow?
6:16PM 0 Dear 78% 0FF on Pfizer.
4:32PM 1 destroy zombie session
4:20PM 0 VUC Today@12 ET: Allison Smith
3:22PM 1 FW: hi Dan
11:28AM 0 asterisk systems hang with "hfcmulti_rx no memory for rx_skb"
10:44AM 1 RTP traffic through Asterisk??
10:31AM 1 little boy on asterisk and Debian
7:21AM 1 Multimedia PBX Solution
7:17AM 1 Health IVR Recordings
3:18AM 2 Will Digium iaxy stop working with asterisk 1.6; as it is discontinued?
12:13AM 0 TDM400p , asteriskNow and may other woes.....
Thursday November 12 2009
11:53PM 1 Home line noise problem
10:19PM 3 Request for Review: Building Queues with Asterisk
5:30PM 1 How to send DTMF on Zaptel with 50ms tone duration and 50ms gap between the digits?
3:44PM 1 solution for NAT issues?
3:38PM 1 Asterisk with FreePBX
2:53PM 3 "POTS 4K linear codec"
2:31PM 5 my kernel is dazed and confused
2:28PM 1 Dell Poweredge T105
2:22PM 1 Codec interface
1:47PM 3 allowguest defaults to yes for SIP
1:39PM 0 AST_CONFIG, MEETME_INFO and meetme.conf
1:29PM 1 BLF with SPA941?
1:13PM 1 Termination Question
12:24PM 3 Incoming Call Ring
11:36AM 0 Scheduling destruction of SIP dialog
9:34AM 0 Cisco 7970 SIP endless ringing...?
9:33AM 2 Need Adapter/Gateway with PSTN-interface
8:16AM 0 [Asterisk 0013405]: [patch] T38 gateway (fwd)
6:31AM 2 soft phone (X-lite) not able to register with asterisk
6:10AM 2 softphones (x_lite) not able to register with asterisk server
2:29AM 1 Can't connect to voip provider over NAT
Wednesday November 11 2009
11:34PM 2 Asterisk keeps sending invite to sip phone "No response to critical packet"
10:50PM 1 What happened to netxusa?
9:34PM 0 AstriCon Videos and Presentations: First batch is on-line!
9:17PM 1 How to control DTMF tone duration on Zap channels?
8:23PM 2 Bug or feature: SIP chanvars not overriden
8:08PM 1 TE121 - Idle system load at ~0.3 - Bad DAHDI behaviour ?!
8:04PM 2 SIP source address error
4:19PM 1 Issue calling from WAN to LAN extension
4:13PM 2 Best practice to set up 4 line phones
4:00PM 0 DAHDIScan() only returns dead air
2:01PM 4 Bad quality of call
12:20PM 1 Voicemail after hangup
11:45AM 1 Unable to execute
10:57AM 1 hosted / virtual IPBX platform
5:14AM 1 SIP response code 603
Tuesday November 10 2009
9:30PM 1 user extension in asterisk GUI
9:19PM 1 Silent Dialing
6:16PM 2 how to configure softphones in asterisk
5:06PM 1 Questions about Dahdi's /etc/dahdi/genconf_parameters
2:31PM 2 Setting outgoing callerid on when using a PRI
1:35PM 2 Hangup
1:04PM 2 looking for an Asterisk supervision (status viewer) tool
11:36AM 2 CDR Import
9:45AM 1 Call audio leaking between calls
2:10AM 0 Extension in use
1:28AM 1 Is voicemail to text possible?
12:12AM 2 Gradstream Budge Tone-201
Monday November 9 2009
11:58PM 1 SendText
10:03PM 1 Call declined
9:25PM 3 is an extension is use
8:07PM 0 chan_mobile Voice setting
7:14PM 0 FreeBSD, ztdummy & OHCI
5:19PM 1 Allow Header
5:06PM 0 got SIP response 482 "Loop Detected" back from
4:32PM 2 how to configure softphones in asterisk server
3:41PM 4 local channels
10:52AM 3 E1 Extensions.conf
10:14AM 1 How to know AMI status
9:14AM 0 fromuser & fromdomain
12:20AM 0 CDR userfield -
Sunday November 8 2009
10:40PM 0 E1 connectivity problem (HDB3, CRC4MF, ISUP, V3)
7:13PM 2 outbound routing
5:33PM 0 Modem card
2:23PM 2 CDR userfield not written into DB
12:22PM 1 Failure of user registration with XLITE
6:20AM 3 how to check version of asterisk
3:38AM 3 Text messaging
12:42AM 0 Set DESTINATION CID for outbound calls
Saturday November 7 2009
7:45PM 4 Help with concurrent VoIP calls
7:18PM 1 Asterisk 1.6.1 + Cisco AS5300 + Fax T38 ?
6:58PM 0 [DAHDI] "failed on channel 1: No such device or address"
5:29PM 1 help in installing asterisk
12:12PM 0 Nov 7 TODAY & Nov 22 - Join Global FreeSW GNU(Linux) HW Culture meeting via VOIP - BerkeleyTIP GlobalTIP - For Forwarding
10:08AM 1 Difference between 'core show channels' and 'sip show channels' ??
7:51AM 0 AMI is not loaded
5:23AM 1 Trouble registering Cisco 7942
12:02AM 6 Location
Friday November 6 2009
8:52PM 1 Question on peering two Asterisk servers
8:22PM 1 Best dahdi switchtype to emulate (network side)?
7:02PM 1 AMI Originate and Variable header
5:08PM 4 problem while compiling asterisk tar file
4:13PM 1 Need opinion about GSM codec for Internet
3:48PM 2 Routing incoming call based on caller id
2:09PM 1 sip set debug
11:30AM 0 Syncing phone numbers DB with cellphone?
11:29AM 0 Setting up an automatic Fax Call Back service
9:11AM 1 which asterisk,libpri,dahdi tar file to compile
8:19AM 0 [VUC] Friday Nov 6 @ 12 Noon EST: Village Telco
6:47AM 1 app read accept # sign
6:43AM 2 odbc to ms-sql server
4:57AM 2 Question about callerid?
4:33AM 1 asterisk,libpri,zaptel
Thursday November 5 2009
11:17PM 1 Asterisk 1.4 DISA is jumoing after one digit in the DISA context
9:36PM 0 MeetMe thinks DAHDI is missing
9:15PM 0 SIP 503 instead of SIP 480 in asterisk debug mode
9:06PM 0 RTP Proxy
8:45PM 2 Prevent cell phone voice mail capturing call
7:42PM 0 Chan_mobile instability
5:33PM 1 Cisco 7912 Phones + Asterisk
2:54PM 3 programming phones
2:42PM 2 7777 & *65
2:18PM 2 Asterisk 1.4 remote pickup
1:15PM 1 Cisco 7912 SIP Firmware
12:47PM 0 SIP brute force attack from Korea
12:06PM 1 Playing Sound during dial
11:27AM 2 faxes received on mISDN
10:34AM 0 fax standard extension and Playback
8:14AM 1 SendJabber question sending Links
3:07AM 1 dialplan pattern matching
Wednesday November 4 2009
8:21PM 0 Asterisk 1.2.36,,, and Now Available
8:20PM 2 Cisco SPA3102 Thoughts & Other Recommendations
8:12PM 0 AST-2009-009: Cross-site AJAX request vulnerability
8:12PM 0 AST-2009-008: SIP responses expose valid usernames
7:43PM 3 How to resell my trunk/provider to others?
7:32PM 0 Social Networking Event * Berlin Nov 12
6:57PM 0 Fwd: Asterisk conferences
4:51PM 2 Minimum hardware requirements for 10 concurrent calls?
4:44PM 2 Asterisk on a MiniITX board+Atom1.6 2gb+Sangoma USB?
4:02PM 1 ExternalIVR testing
3:12PM 0 memory leak with static users
3:11PM 0 channel destruction after a transfer call
1:23PM 0 Call Transfer Problem
12:24PM 0 UK Vodafone messaging, ISDN, Wrong CallerID being used.
11:48AM 3 Asterisk crashing
11:01AM 2 Help in Perl AGI
10:23AM 1 segfault wall
9:16AM 2 Asterisk SS7 Sigtran Protocol
6:00AM 1 Personal invitation from daminda edirisinghe
1:48AM 0 Fwd: Seminarios Tecnológicos en Fundación Proydesa
Tuesday November 3 2009
7:02PM 1 ring groups with different caller id
5:28PM 1 MusicOnHold works Externally, but not internally
4:09PM 2 Extra CDR fields
1:56PM 5 Asterisk and Software Data Modem
1:26PM 1 Asterisk Realtime Extensions => for all context ?
12:14PM 3 Problem with ChanIsAvail
11:39AM 0 Redirecting Calls and MeetMe Rooms
11:26AM 0 Popping sounds on voice prompts
11:25AM 0 routes and trunks
10:22AM 1 dahdi channel not showing up
5:25AM 0 Exchange 2007 UM issues with Asterisk 1.6
4:18AM 1 Core Dump - Asterisk 1.4.24 - Elastix
2:56AM 1 turn the ring tone OFF during dialing
Monday November 2 2009
10:39PM 2 Asterisk as Outbound Proxy ?
9:04PM 0 DTMF Timing and Fujitsu F9600 Switch
7:11PM 1 Unexpected control subclass '-1'
6:21PM 0 Execute Macro AFTER connecting to a channel
5:22PM 7 Asterisk 1.4 and Fax
4:33PM 0 Nagios check_asterisk_peers needs rights to question the Asterisk-server
3:37PM 2 Remote IP Phone's
3:18PM 1 Remote Party ID
2:52PM 0 change L(x[:y][:z]) parameter of DIAL command after call is bridged
12:38PM 1 supermicro hardware + sangoma
12:11PM 4 GSM and Wav format
11:59AM 1 MySQL CDR
10:14AM 1 Asterisk Fax Module
9:37AM 2 hardware requirements for asterisk
8:36AM 0 PSTN Line Parameters Checking
8:33AM 3 Xorcom device not showing up in /proc
6:49AM 5 Forward DID to another server
2:40AM 4 include statements in IVR
Sunday November 1 2009
10:23PM 2 Dialstatus
9:13PM 1 asterisk 1.6.0 seems to have improper dial status when dialing dahdi extension
8:46PM 1 pattern matching DID
8:21PM 0 PSTN Lines and AEX808B
6:42PM 0 Execute the specified macro for the called channel AFTER connecting to the calling channel.
6:27PM 2 Tutorial for SIP user
4:44PM 0 Originate with Local channel to any app-only extension hangs up immediately?
4:24PM 0 Exchange 2007 and Voicemail with Imap Storage
12:31PM 1 Skyp SIP? - what is free for a home *
9:53AM 1 usage of manager events to create custom reports
8:42AM 1 Error in MeetMe modules ?
5:10AM 0 need help debug asterisk-1.6 sip connection