asterisk users - Oct 2009

Saturday October 31 2009
9:22PM 1 Determining extension's sip.conf default mailbox
6:56PM 1 Long pause during dialing to IVR
6:44PM 2 Asterisk, Realtime and specify MySQL Table Name ?
6:36PM 0 PRI line resetting on incoming call
4:04PM 1 !<command> from Manager
3:41PM 2 Calls disconnects after short time
3:20PM 1 Disconnecting during the call, analog lines
1:33PM 3 OT - Number Portability
6:57AM 0 Local channel that runs a custom app... why immediate hangup?
12:46AM 0 strange dialing HELP !
Friday October 30 2009
11:28PM 1 Asterisk- DTMF with SIP is not working
9:31PM 3 AEX800P on HP Prolaint ML115 kernel panic
7:23PM 3 voicmail: no entry in voicemail config
6:05PM 1 asterisk 1.6 - doing dnsmgr lookup for... / call fails
6:01PM 1 Cannot make calls
5:22PM 1 SNOM 870
3:57PM 2 Real replacement for AgentCallBackLogin() on Asterisk 1.6
3:05PM 7 Voicemail file
1:24PM 1 Queue device state problem
11:55AM 4 [IAX] Recommended soft- and hardphones?
11:54AM 2 DAHDI/ZAP overlap dialing
11:08AM 1 inbound routes
6:05AM 2 asterisk 1.6 enable cdr_mysql
Thursday October 29 2009
11:36PM 3 Unable to set TOS to 184?
9:49PM 1 Astreicon presentations
9:16PM 0 AsteriskForge Now Open
8:50PM 1 Booting Error for /dev/kmem
8:26PM 0 !! Unknown IE 50 (cs5, Unknown Information Element) on console.
10:49AM 1 Zap inbound hangup problem
7:48AM 1 Async Agi problem
1:05AM 5 Dynamic DNS trunk
12:29AM 2 GUI for hunt groups?
Wednesday October 28 2009
10:57PM 3 Mysql CDR in Addons does not record CLID
10:46PM 1 Asterisk 302 Moved Temporarily
9:26PM 0 Asterisk/Cisco AS5300 => Two problems in
8:24PM 1 MOH
6:16PM 1 SIP 18x Messages
2:59PM 5 need a local tech
2:44PM 1 Clear pending SIP channels
2:07PM 2 deploying asterisk
11:41AM 0 sip fullcontact and port values
11:34AM 1 CDR(billsec)
11:27AM 1 SIP Peers still ping with SIP OPTIONS on a reload
10:53AM 1 The SIP in the Mobile Phones are not able to register on asterisk
8:50AM 1 Asterisk Server with Panasonic PBX
6:29AM 1 SIP client MAC address.
5:50AM 1 Dialing out a T1
5:09AM 2 Asterisk/Cisco AS5300 => Two problems in incoming (extension not found)
3:03AM 2 Confusion on caller-ID with SIP provider
Tuesday October 27 2009
11:29PM 2 need to find firmware for cisco ata-188
9:13PM 5 Software for PC-PC voice comunication
8:58PM 1 The Mobile devices are not able to register on my asterisk
7:26PM 0 Codecs with MixMonitor (,a) option
2:18PM 10 How to dial multiple extensions at once like in a ring group and put them in conference?
1:59PM 1 pri intense debug span 1
1:10PM 0 Fax with a AEx410P and Beronet BN4S0 => Sending Problem
12:56PM 1 RTP timestamps
12:35PM 0 [OT] Snom M9
9:49AM 3 installing
9:34AM 3 Installing Asterisk
Monday October 26 2009
11:36PM 0 Asterisk Now Available
8:26PM 0 AST-2009-007: ACL not respected on SIP INVITE
8:05PM 2 What is the best way to configure this?
7:59PM 1 DAHDI not detecting RINGING Status on the Channel
7:52PM 1 Answer call from another device
6:43PM 1 state_interface backport issue
4:59PM 1 Cancel attended transfer
1:07PM 1 IAX jitterbufer oddity
12:49PM 0 Call Record.
12:02PM 3 No tone, one way communcation.
10:10AM 1 Common Community for exchange the routes via Asterisk boxes
Sunday October 25 2009
11:39PM 1 some issue with libpri cant go past 1.4.1
9:10PM 1 chan_echolink
8:58PM 2 Asterisk as the recording server for Avaya Definity
4:17PM 0 FW: Queue Transfers
4:16PM 0 Queue Transfers
2:47PM 1 Asterisk-stat! - help needed (once again due to mailserver problem)
2:19PM 2 SIP interconnection problem
1:51PM 2 help sip show on CLI : no such command
12:33AM 2 test
Saturday October 24 2009
4:26PM 3 OT - mISDN and B410P questions
2:57PM 2 SVN newbie - No trunk/ in
2:40PM 0 Tim Panton Astricon Presentation "recreated"
10:24AM 0 Manage users & administrator for the asterisk-gui
10:05AM 0 AMI script..
Friday October 23 2009
9:24PM 1 IVR reports?
8:44PM 1 how to announce the agent answering in a queue to the caller
8:26PM 3 SIREN14 call setup and record/playback
7:23PM 0 Crash with app_mixmonitor
6:21PM 0 Asterisk SIP to Cisco IAD2430 Series?
4:20PM 1 Strange IAX2 / Iaxmodem problem
3:19PM 1 GUI for asterix management
1:33PM 1 Recording management for IVR
11:36AM 2 How to generate 183 Session Progress
11:01AM 0 asterisk crashes when calling gtalk user
6:48AM 2 interfacing asterisk with a legacy PBX
6:33AM 1 AstriCon videos: a question of method (Robin)
Thursday October 22 2009
11:54PM 1 Asterisk MOH playing old audio for first 30 to 60 seconds
11:19PM 1 OSLEC with DAHDI and Linksys/Sipura
11:04PM 1 Poor VoIP voice quality in one direction from three providers
10:34PM 4 AstriCon videos: a question of method
5:12PM 2 hangup from which side
4:42PM 1 GSM 6.10 codec for Asterisk
4:33PM 2 IAX Hardphones.
3:48PM 1 Can't configure Cisco 7942 avec factory reset
3:07PM 2 ChanSpy in Asterisk 1.2.24
2:50PM 4 OT - How to organize TFTP root directory ?
2:30PM 2 carefulwrite: write() returned error: Broken pipe
11:32AM 1 queues autopause
9:45AM 1 Audio issue in skype for asterisk
6:43AM 2 AGI STREAM FILE and not blocking execution
4:07AM 2 ivr menu not hanging up call
2:48AM 0 Asterisk crashing -- multiple phone entries
Wednesday October 21 2009
8:19PM 1 Incorrect voice mail format on transfer
6:50PM 5 Asterisk and Nuance Vocalizer TTS Engine
6:30PM 4 Concurrent calls including mysql taking lot of time for execution
4:54PM 1 OT - Gigaset Chagall - How to download firmware without Internet access ?
4:10PM 0 DAHDI: TCM PCI Master abort
3:08PM 0 Intermittent Low volume
2:57PM 1 polarity on some channels
12:42PM 0 ringing... or lack thereof
11:39AM 0 TxFax works only with one of 2 PRI
11:05AM 3 Need Help
10:46AM 1 RAMDisk vs Extarnal server for recording
9:47AM 3 Searching on how to keep local calls... local
8:04AM 1 error - sources for the 2.6.18-92.1.22.el5xen kernel
2:06AM 1 ChannelStateDesc: Ring ?
Tuesday October 20 2009
11:34PM 4 Linksys 962
10:36PM 1 OutCALL
10:26PM 1 Is there a way to force a codec on an incoming sip uri call?
10:25PM 2 Kernel panic w/ DAHDI 2.x/Digium TE220B
9:50PM 1 Libpri- Released
8:40PM 3 High Volume Call Center SIP versus IAX2
6:32PM 3 troubleshooting NAT
2:08PM 0 srtp crypto lifetime
12:24PM 6 Syncronizing files on different Asterisk servers
11:59AM 0 cascaded pickup
11:01AM 2 AMI 1.0 -> 1.1 with originate.
9:43AM 0 need help with card for IVR
8:55AM 3 Cisco 7921
8:03AM 2 (no subject)
6:26AM 0 Voip interconnection with ACME SBC-asterisk-users Digest, Vol 63, Issue 52
12:30AM 2 all our circuits are busy now
Monday October 19 2009
10:42PM 1 Cisco 1751 setup with asterisk
9:51PM 0 ANN: Asterisk-Java 1.0.0.M3 Released
9:37PM 3 Dial a external number with extension
9:29PM 3 IMAP voicemail using subfolders fails.
7:08PM 1 Looking for the asterisk 'off' sound file
6:06PM 3 delay in processing dtmf
4:17PM 3 asterisk services not starting up
3:58PM 2 Astricon talk on wideband codecs
3:47PM 3 update CDRs in mysql during a call
2:18PM 2 Problems replaying G.726 - only noise
1:55PM 2 VoIP interconnection with Acme packet SBC
1:42PM 0 Missing digits from CallerID on TDM400P?
11:20AM 0 question about getting instance ringing member in queue
10:12AM 0 announcement tone to callees of app_page
Sunday October 18 2009
11:34PM 2 BTS
11:20PM 4 Customising Firmware
10:18PM 1 SIP Headers
8:27PM 1 SIP debugging enabled : written to log
6:43PM 1 Asterisk Expert Freelancer in Karachi
6:05PM 1 Call from 'sip-id' to extension 'sip-id' rejected because extension not found ?
2:24PM 1 Asterisk+Sphinx4 for simple mobile phone <-> server speech recognition
2:06PM 0 Kernel Timer
10:32AM 4 Snom870 sidecar
8:46AM 0 Sunday 18th 12N-3P PDST Global Asterisk Mtg - BerkeleyTIP - for forwarding
8:24AM 1 SIP debugging enabled : written to log ??
2:29AM 4 Astricon
12:30AM 7 Asterisk Monitoring
12:01AM 0 Best practices for reliable carrier grade telephony
Saturday October 17 2009
6:26PM 3 Possible bug in app_meetme.c
3:18PM 1 sched_settime: Request to schedule in the past?!?!
2:02PM 3 OT - DECT SIP Phones
1:45PM 4 how to limit the calls leaving a queue?
1:02PM 0 OT - KVM and PCI telephony cards
Friday October 16 2009
11:46PM 0 Nehalem & Digium Wildcard issues?
11:37PM 5 IVR
11:06PM 1 Whither asterisk-addons?
8:04PM 2 SIP to IAX to SIP
6:28PM 1 can i use Asterisk to send sms to my database users?
4:33PM 0 OT Old Sipura OK - Linksys (junk)
4:20PM 0 Origin of "Exceptionally long voice queue length queuing to IAX2/blahblah" messages
2:40PM 1 inquire if SIP connections are active or not
1:41PM 0 app_swift issue
10:56AM 2 Soft phone not registering
10:07AM 2 Invite after bye?
10:04AM 1 Check if a variable is set
6:16AM 3 Linux/Asterisk on game consoles?
5:13AM 1 The City of Amsterdam has been deploying asterisk throughout the city!
2:37AM 1 Mixing SIP/TDM in MeetMe
Thursday October 15 2009
10:19PM 2 OT wanted old Sipura firmware 2.0.13
9:52PM 0 Asterisk and FreePBX Amazon EC2 instances are now available in Europe
9:21PM 2 question on SIP and call manager
7:58PM 2 2 IPs for an Asterisk server.
7:50PM 2 A little OT but need an opinion on Aastra 57i CT
7:24PM 1 OT - Can't upgrade Cisco 7942 to SIP
7:20PM 3 DS3 capacity calls using asterisk
6:57PM 0 Best way to detect fax in Asterisk 1.6??
6:57PM 1 Testing the Timing Device
4:42PM 1 best way to make 5-10 simultaneous calls to the same did at a set time of day
4:41PM 1 sporadic one-way audio
4:34PM 4 PSTN to SIP line ratio
3:16PM 0 Does ADA 1.1 or ADA Pro exists ?
1:28PM 1 Where to find IMAP storage doc ?
12:50PM 2 hi
10:27AM 2 Asterisk with a Cisco AS5300 gateway
8:05AM 4 Calls hang up after 20 seconds
2:56AM 1 Callpickup works for outside calls but not inside calls
2:41AM 2 MWI for multiple voice mail boxes
Wednesday October 14 2009
9:24PM 3 Extension Paging
8:52PM 1 Door Phones
7:53PM 0 WaitForSilence doesn't work unless Background is called?
6:44PM 5 multiple call
6:00PM 1 Cisco router
4:33PM 1 PostgreSQL problems
4:04PM 2 ACD & ASR
2:39PM 2 Config Files
1:24PM 1 no outbound calls
1:01PM 1 Asterisk 1.4 vs 1.6
11:58AM 2 Queues with unavailable members
11:19AM 0 SIP RealTime defaultuser Field Cleared
9:36AM 1 DTMF failing in some calls
7:27AM 2 FXS to SIP gateway
6:10AM 2 DAHDI Dummy for Linux VServers
3:19AM 1 ChanSpy on asterisk 1.6
12:42AM 2 T38 negotiations in RTP
12:22AM 8 Asterisk in the Cloud
Tuesday October 13 2009
11:56PM 4 AMI input streams limit?
6:08PM 0 Bridge command in 1.6
4:04PM 11 Best Firewall Suggestions?
3:07PM 2 Dial Delay
9:31AM 0 missing CDR records in cdr while kick from meetme
8:19AM 3 strange transcoding values
Monday October 12 2009
9:53PM 0 asterisk dialplan to share fax line
8:36PM 0 video support with voicemail question
6:19PM 0 How to send a digit to a channel??
5:36PM 5 G729 in asterisk upgrade issue
4:05PM 0 Asterisk 1.6 with TDM2400 and 2 FXS modules
3:56PM 0 live audio streaming using monitor, mixmonitor or chanspy
1:21PM 0 libss7 problem with dialing a non numeric string
12:24PM 1 tealtime static
12:21PM 0 meetme and confbridge
10:51AM 2 SPRINTF option : format %1$s not supported
3:25AM 1 How to do a 3 party Warm Transfer in Asteriks 1.4
Sunday October 11 2009
11:42PM 0 Grandstream 2010
11:15PM 5 Call Recording and Posting
Saturday October 10 2009
10:25PM 3 Method to use SOX inside a Dialplan
9:18PM 1 Asterisk to Asterisk access voicemail - not working
8:46PM 2 outgoing sip calls work; incoming calls fail
8:39PM 2 Mp3 for IVR prompts
4:13PM 1 Grandstream GXP 2010 : multiple accounts not working
2:32PM 0 paging/intercom
1:34PM 0 Slightly OT: Astricon and Google Wave
1:09PM 1 delay to dial
1:15AM 2 Wifi GSM handover
Friday October 9 2009
11:20PM 0 lawnmower man "attack" ??
11:12PM 1 lawnmower man "attack" sip tag=Zerogij34 some one else notice this in 20th september or recently?
9:48PM 2 Incoming extension not working.
8:45PM 1 choppy sound
8:27PM 1 ${REASON} not getting set.
5:44PM 3 Chanspy
3:52PM 0 calls ansowered for 1 second or less
3:03PM 1 Billing applications
2:07PM 1 VoiceMail and IMAP
12:54PM 1 wrond DTMF detection on Zap channel
10:16AM 0 Trunk and Pstn line
9:09AM 1 G.729 and Voicemail
9:03AM 0 Asterisk Queue & Agent
8:00AM 1 Today's problem: Inbound call routing
7:41AM 1 Digium G729 licence unattended install
3:53AM 2 SIP Hard Phone with SMS
Thursday October 8 2009
9:37PM 1 Realtime static does not work in 1.6.1 or 1.6.2
8:40PM 1 Help setting up IMAP_STORAGE on CentOS 5
8:03PM 2 Best QoS for Linux
7:43PM 4 No sound on voicemail from analog line
5:35PM 0 asterisk 2bct/rlt calling
5:10PM 0 Fuori ufficio
3:31PM 0 Suggestions for low level RTP stream generator?
3:22PM 1 g729 free codec any idea
2:55PM 0 Friday Noon VUC with guest Alex Robar
2:52PM 2 Best afordable router with QOS for *
2:29PM 2 Server-side scripting when SIP phones register
2:25PM 1 Having trouble with "IF" and blanks
1:43PM 4 Dialplan problem
1:20PM 2 How to keep difference between 2 SIP-accounts/trunks from same server ??
12:40PM 2 Asterisk and Sheeva "wall wart".
12:22PM 1 MeetMe option question
8:57AM 2 limiting number of channels to be accessed
7:03AM 0 MeetMe + SLA
12:05AM 1 Drop Call on ICMP Port Unreachable?
Wednesday October 7 2009
8:57PM 2 Can dial long distance but not local?
7:20PM 1 VPS Server
4:25PM 1 DTMF Issues
3:22PM 1 Need provider recommendations for the UK
10:21AM 2 system cmd + fax line
10:06AM 0 Asterisk Integration with RDP Property Management Software?
Tuesday October 6 2009
10:40PM 0 Problem sending a DTMF remotely. Please need help!!!
8:22PM 0 Asterisk 1.4.27-rc2,,, and Now Available
8:15PM 1 Asterisk suddently restarts ...
8:01PM 7 MPG123 Dying
7:49PM 2 Transfers from Queue Calls
4:18PM 2 adding modules
3:55PM 1 Is anyone doing real time updates to where asterisk registers?
3:40PM 4 Cent OS 5.3 All Updated Asterisk Installation Giving Error
2:35PM 1 Asterisk + Monitor() Poor quality
2:05PM 0 video support over iax
1:51PM 0 parse transparent ISUP parameters
1:33PM 0 What happened to MACRO_EXTEN in AEL macros since 1.6?
12:07PM 0 Lancom 1722 and Asterisk (i need HELP)
9:24AM 0 fsk callerid with DTAS start?
5:02AM 2 T38 REINVITe issue
Monday October 5 2009
11:14PM 5 Networking Concept
6:34PM 6 Receptionist GUI?
4:53PM 2 Method to downgrade asterisk
4:48PM 0 web module for video calls
3:45PM 2 dahdi dies with "No more room in scheduler"
3:26PM 0 Asterisk and QSIG
3:03PM 1 DTMF problem during read()
2:55PM 3 OriginateResponse Event
2:47PM 0 What dahdi_dynamic and dahdi_transcode modules are for?
2:17PM 3 Questions about app_jack.c
11:27AM 1 Problem sending a DTMF remotely. Please need help!!
9:14AM 1 Grandstream GXW4024 experience
7:47AM 1 AEL problem: bug or feature?
4:04AM 1 Drop calls when using Flash Operator Panel
1:05AM 1 Peculiar error message when using Q-SIG
Sunday October 4 2009
2:41PM 0 Hacking the network
10:28AM 9 Zaptel problems on SUSE 9.3
9:45AM 1 x100p card
12:05AM 3 After call into console/dsp hangup hear ringing
Saturday October 3 2009
5:19PM 0 Ideasip
4:02PM 0 ERROR[1499]: rtp.c:2482 ast_rtcp_write_sr: RTCP SR transmission error
1:19PM 1 Asterisk on NSLU2 : Grandstream not registering
1:17PM 1 Calls being dropped - Cisco 7940 with SIP 8.12 image
9:36AM 1 Asterisk and Jack
12:40AM 0 Problem sending a DTMF remotely. Please need help...
Friday October 2 2009
10:38PM 1 Asterisk,click2talk, webphone
6:42PM 3 app_hackblock to prevent SIP/IAX reg trolling
6:22PM 3 Extra Sounds Missing on install
5:58PM 2 Asterisk HA Current Thoughts (Centos 5.3 Platform)
5:48PM 0 Asterisk + Skype deployment
2:51PM 0 Sending a DTMF remotely with PlayDTMF problem.
2:42PM 1 One side SIP goes dead on length conversation
1:20PM 2 Followme
12:52PM 1 How to call extensions and add them to a conference room
12:34PM 1 Creating a clear channel on zaptel
12:30PM 1 Problem with inbound calls - asterisk
9:10AM 0 srtp issue
8:48AM 0 Free version of softswitch with billing and routing released
6:50AM 0 Busy() returns immediately on IAX trunk
12:14AM 1 IAX2 Call rejected, CallToken Support required
Thursday October 1 2009
6:27PM 1 QOS/DSCP for IAX?
5:02PM 1 DTMF problems during a message play
2:10PM 3 What are the reasons for VoIP echo?
1:04PM 1 Is there a way to get info who disconnected the call into CDR?
11:13AM 1 Busy app timeout
11:09AM 1 portech MV-378 SIP GSM Gateway
10:23AM 1 RTP Delayed during RTCP
10:18AM 0 Friday Oct 2: Digium's new Speech Recognition for Asterisk
9:43AM 2 INVITE Sending Local IP
8:30AM 2 help on ${RTPAUDIOQOS}
7:32AM 2 Softphone in Web
4:56AM 3 "got stuck at 150 calls, above that not working in stress test"
2:56AM 0 Issue with SIP & QSIG phones in MeetMe conf room