asterisk users - Nov 2010

Tuesday November 30 2010
10:25PM 2 Error loading module '': cannot open shared object file: No such file or directory
6:25PM 1 Rhino Channelbank...
1:14PM 2 Correct operation of timout parameter for dial application
12:11PM 0 Default From and Contact header domain
9:47AM 2 Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented)
9:28AM 10 TCP port, VPN and resolving the cutting voice problem
8:46AM 0 Transfered Number is local extension of PSTN
4:13AM 1 OT: for those wondering on the stability
Monday November 29 2010
5:11PM 1 Trouble with TE122 on HP DL120G6 - can't disable USB
5:07PM 3 How to initiate a two-party call from within Asterisk
5:01PM 1 ID'ing failed auth IPs
10:23AM 0 resending cause codes
10:08AM 4 Asterisk on smartphone?
Sunday November 28 2010
10:21PM 2 Stability..
7:14PM 0 Problem in receiving calls from E1
5:03PM 4 Firewalling and Asterisk
1:42PM 0 Asterisk loopback calls form one extension that playbacks to another that records for performance measuring
4:28AM 0 AstLinux 0.7.4 Release now available
Saturday November 27 2010
8:03PM 0 Strange Logfile Entries.
7:40PM 2 Preserve CallerID on transfers
5:03PM 1 DAHDI 2.4.0 produces HDLC errors with echo canceler
3:51PM 3 sip echo server
10:52AM 1 change date
9:25AM 3 How to hangup all channels
Friday November 26 2010
6:39PM 1 echo calls
4:58PM 0 asterisk-users Digest, Vol 76, Issue 58
4:20PM 1 Meetme Realtime in 1.6
1:29PM 0 IAX trunk two Asterisk
11:43AM 3 New implementation asterisk
11:05AM 1 HA8 + B400M not configured with genconf_parameters
11:00AM 4 Asterisk 1.8 crashing
Thursday November 25 2010
9:57PM 0 d-link dvg-3032 guidance
5:05PM 0 IAX inbound failing
4:54PM 1 Unit of measurement dahdi_monitor
4:23PM 2 Timing cable usage necessity
12:51PM 0 Siemens HiPath 1120
2:57AM 2 Spam
1:30AM 4 Incoming calls through SS7 for data modem transmissions - possible??
Wednesday November 24 2010
5:45PM 1 Disable connected line updates for dahdi PRI channel
5:07PM 1 Why doesn't Asterisk project document certain important features of Asterisk officially?
5:01PM 0 IPv6: What You Need to Know Now
4:43PM 3 kernel: dahdi: Detected time shift.
1:44PM 0 Originate Response.
12:48PM 2 Avoided deadlock Error
12:11PM 0 DTMF CallerID
11:19AM 1 TDM calls fall after some minutes
11:16AM 2 SPA942 on speaker phone does not hang up?
11:09AM 2 asterisk-1.8.0 compilation error
10:30AM 1 Asterisk 1.6 and Music on Hold
10:12AM 1 Contradiction in GROUP() function
8:02AM 2 astcanary ?
7:20AM 1 action at registering or de-registering
Tuesday November 23 2010
11:57PM 2 Function SIP_Header not registered
8:33PM 1 asterisk 1.8.1-rc1 + sip transfer fix
4:30PM 0 Asterisk 1.8.1-rc1 Now Available
1:25PM 2 Asterisk Log viewer
1:24PM 1 wideband recording in Asterisk 1.8
12:31PM 2 Asterisk 1.8 Release Schedule
Monday November 22 2010
10:46PM 3 Asterisk pass a call to status answer while still ringing
9:27PM 0 libpri Now Available
6:38PM 1 asterisk and cisco 7970 - multiple lines
5:28PM 0 Using AMI to harvest / record HOLD time - Using FreePBX
4:24PM 0 Polycom dial w/o "Dial", while on-hook?
4:22PM 5 Someone has hacked into our system
1:47PM 2 Call recording format
9:31AM 1 res_musiconhold.c Bug - Patch to solve?
9:09AM 2 URGENT Help needed
5:20AM 3 Is existing CDR in Asterisk is enough for complete billing
5:06AM 1 Quintum AFT800 on Asterisk 1.4.29
2:41AM 1 Early audio (long distance codes) not working after upgrading to 1.8?
2:37AM 0 cisco 7970 multiple lines with asterisk
1:14AM 2 SIP Extensions and loss of Internet connection
Sunday November 21 2010
11:13PM 2 DAHDI phantom pickup when ringing
10:01PM 0 How to configure a Linksys PAP2T ATA to connect an analog fax machine to Asterisk
11:58AM 2 Please help me in configuring asterisk for the scenario
11:26AM 2 Asterisk behind D-Link ADSL router with private IP
Saturday November 20 2010
6:36PM 1 ConfBridge
6:17PM 0 AGI CDR Update (with set variable) problem.
4:55PM 0 sip attended transfer beep
Friday November 19 2010
9:56PM 2 Installing Asterisk to it's own directory
6:55PM 1 Make call in AMI.
4:13PM 0 help with annoying warning message: Asked to transmit frame type ulaw, while native formats is 0x2 (gsm) read/write = 0x2 (gsm)/0x2 (gsm)
3:33PM 1 callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk
2:42PM 3 FFA (Fax For Asterisk) tif file (size) problem
1:59PM 0 Asterisk 1.8 and Dial(SIP/peer_name) to undefined peer
12:28PM 2 Ekiga can register but not my IP phone
11:59AM 0 Using Local Asterisk Server with Siphon - Can't hear voice issue
6:34AM 1 call forward problem
Thursday November 18 2010
9:54PM 2 ISDN-FAX with Asterisk
8:01PM 2 IAX2 and INVAL packets
6:37PM 2 Meetme and MOH
5:34PM 1 Asterisk parking question
5:04PM 1 Asterisk 1.8 VM_DUR problems
1:39PM 2 Asterisk Load Balance and Failover
10:58AM 0 Announcement Transfer with call-limit = 1
10:36AM 3 How to register SIP phone on Asterisk on Centos 5.5 64bit
6:21AM 2 exceeds the maximum size of ast_fdset error on Asterisk-1.8.0
3:54AM 3 usage of account code in CDR
2:16AM 1 Register SIP on Asterisk on Centos 5.5 64bit
12:31AM 0 Supported: ms-early-media
Wednesday November 17 2010
8:30PM 2 GSM and SS7 Questions
6:54PM 1 Asterisk and IPv6
4:40PM 6 How many Asterisk PBX operating in the World?
1:09PM 0 One way audio problem
10:58AM 0 Newbie question on GSM adapter
5:46AM 1 Asterisk runs at 100% CPU
Tuesday November 16 2010
8:37PM 0 billsec and duration issue
7:44PM 0 Fwd: Delivery Delayed: Re: Ring back tone with asterisk
7:03PM 2 T1 with Robbed Bit Signaling
6:38PM 5 HA - asterisk service is not starting
6:17PM 1 S110M not working
5:35PM 0 OT - Call Waiting features with Kirk 600v3
2:28PM 0 SPA941 WMI not lighting up when natted
2:28PM 3 Recommended *WRT router to run Asterisk?
2:08PM 2 Avoiding deadlock
12:57PM 1 Issues with Local Channel
9:09AM 1 DAHDI / dial in / overlap digits / timeout
2:00AM 2 How to construct a call center on asterisk
Monday November 15 2010
11:04PM 4 Best way to connect to a MySQL Database
8:11PM 2 SIP calls destroyed after 1:20
6:35PM 7 Door Contacts via Asterisk?
4:22PM 0 Maybe a little OT??--- Obtaining DIDs in Hyderabad, India
1:30PM 2 Volume on meetme recording
12:46PM 0 Asterisk Maintenance Checklist
11:05AM 2 Problem When Using Polycom with 2 Lines
10:47AM 2 friend, peer confusion in sip.conf
8:03AM 0 (no subject)
2:48AM 0 Asterisk disconnected suddenly
Sunday November 14 2010
3:00AM 1 upgrade
2:11AM 1 A few questions regarding Asterisk 1.8.0
1:38AM 8 dial plan and sip
1:18AM 0 EXTENDED: Scheduled maintenance for various Asterisk community services
Saturday November 13 2010
7:06PM 0 problem registering to
6:43PM 1 Nat Issue - I think
5:10PM 0 eSXI and Asterisk?
10:15AM 1 CallerID from Samsung PBX line on FXO
9:36AM 2 asterisk 1.8 fax woes
6:14AM 1 asterisk-stat v.2 and mysql 5.1.51
Friday November 12 2010
10:01PM 0 Asterisk and Tandberg Gatekeeper
9:36PM 1 Scheduled maintenance for various Asterisk community services
7:56PM 1 Call failed becaus of SIP tanslate
6:17PM 3 Sending calls to a particular T1 port.
5:08PM 1 Context issue
4:47PM 0 Asterisk Sip trunking routing problem
11:07AM 3 Official Documentation for Asterisk 1.6 Realtime ODBC Tables
9:23AM 1 Asterisk 1.6.20 / CDR issue with app-dial / bug or feature?
1:22AM 1 TTS in Asterisk on Solaris
Thursday November 11 2010
10:21PM 3 T38 re-invites issue
8:36PM 0 Asterisk Released
8:32PM 0 Asterisk 1.4.37 Released
4:28PM 1 VoiceMail customizing
3:35PM 2 Asterisk Playback sound dropping on linphone
9:31AM 3 Limit Call Duration with L-option of Dial : announcement
8:23AM 0 Asterisk and ENUM LOOKUP? E.164
Wednesday November 10 2010
11:51PM 0 1.4.36 - Warning Dropping incompatible voice frame on Local/ on multiple atxfer a->b->c...->d...
9:17PM 0 Reboot any(?) SIP Polycom -- provisioned or no.
7:52PM 0 Problem with AMI
7:11PM 1 Selecting 'ODBC_STORAGE' from outside of 'menuselect'
7:09PM 2 Asterisk 1.8 -- queue not recognizing that agent is busy
6:51PM 1 Random call drops on IAX2
5:27PM 1 multiple devices wants to call through single peer (trunking)
3:08PM 0 Asterisk IAX2 Realtime issue
2:52PM 1 Phones don't stop ringing
2:42PM 1 MFC/R2 detected as ISDN PRI
2:40PM 0 Friday @12 Noon EST: PhonoSDK from Voxeo Labs
12:55PM 1 dahdi module disappears from AsteriskNow after kernel update
12:05PM 0 CDR Billing issues
5:39AM 0 Asterisk ConfBridge application – Delay in voice path
Tuesday November 9 2010
11:56PM 0 hello xC
7:38PM 0 zaptel debugging
5:57PM 1 Asterisk 1.2
5:07PM 1 Asterisk 1.6 and Username in Dial
3:57PM 1 Asterisk 1.8 and Zimbra
3:35PM 1 SMS Gateway
10:30AM 0 Asterisk Voicemail Realtime and 'VirtualBoxing'
2:31AM 1 Store CDR (call detail record) to Oracle database
2:17AM 1 Is this a DDoS to reach Asterisk?
Monday November 8 2010
11:05PM 0 Asterisk 1.6 and Kamailio 3.1 realtime integration tutorial
9:43PM 3 Addons for Asterisk 1.8?
8:59PM 1 Asterisk 1.8 Multiple Parking Lots
6:12PM 3 Get the Uniqueid of Action Originate in the AMI
10:55AM 4 Integrating With Asterisk
7:35AM 0 MWI SUBSCRIBE Settings
7:31AM 0 Asterisk re-started automatically
6:16AM 0 VAD in asterisk
6:08AM 0 Trixbox/Asterisk integration With SugarCRM
1:02AM 1 Asterisk with HUD Lite
Sunday November 7 2010
11:08PM 2 Asterisk 1.6 (realtime) - Overwritten CallerID(num) Problem
8:59PM 2 install
6:29PM 7 Big practical systems
4:23PM 2 Any good guides for installing Asterisk on Embedded systems like Alix boards?
3:26PM 2 "scratchy" sound on TE410P
2:11PM 3 Why are the hackers scanning for these?
Saturday November 6 2010
9:48PM 1 Abandoned queue calls do not produce a CDR?
4:54PM 2 One way voice with Asterisk
4:22PM 2 Asterisk spontaneous reboot
2:30PM 1 OT: certificate for softphone
1:20PM 1 sip and iax2 audio volume gain
5:17AM 0 gigasets A580IP Recall Button
3:57AM 0 Any way to stop Playtones(dial) when the user presses a key, emulating a CO's behavior?
12:05AM 1 Call using password
Friday November 5 2010
9:35PM 1 Soundpoint IP 430 -- discontinued.
8:44PM 1 res_ais Error
8:04PM 2 Alternative to Proxmox
7:32PM 1 Unable to place 2 or more calls to a DID
7:30PM 0 Polycom WEB UI configuration - What needs to be put in for basic SIP registration?
4:50PM 0 No audio with gtalk client behind http proxy
3:21PM 1 Asterisk default sound files
3:21PM 1 Asterisk 1.8 Installation Problem
3:11PM 3 Elementary question - accessing feature codes from cell phone
3:04PM 2 Funky IAX behavior between 1.4 and 1.8
3:01PM 0 Using Dial() but no CDR is generated for this outcall
2:58PM 2 How to append custom option to Contact: header on outgoing SIP INVITE msgs?
2:52PM 1 Asterisk in the third world - Astricon 2010 keynote follow-up
2:27PM 1 GROUP_COUNT not counting correctly
3:10AM 0 How to check PRI status from dialplan
1:22AM 5 MixMonitor
12:30AM 2 Determine channels in use from CLI
12:12AM 3 Short rings for extensions when part of the Queue
Thursday November 4 2010
11:24PM 0 [backport] Allow app_dial to play 'indication tone while ringing' back ported to 1.6.2.X
10:31PM 1 Phones slow to ring
9:40PM 0 mISDN issues again
8:51PM 2 useless mpg123 processes hanging around
7:24PM 1 upgrade 1.6 -> 1.8: wrong password!
5:56PM 1 Is queue Members priority supposed to show in the "queue show" command
4:07PM 2 Multiple extensions - same context
1:22PM 0 Asterisk + Mediatrix
12:14PM 1 UNREACHABLE/Lagged happening on "bulk" register/subscribe
9:38AM 3 Urgent Help Required
9:36AM 1 (no subject)
9:15AM 0 Help Required (How to acheive packetization time of 60ms over SIP/IAX2 trunk)
6:37AM 0 ring delay and DTMF related problem in asterisk
Wednesday November 3 2010
10:30PM 1 doh! chan_dahdi.conf
9:34PM 0 Asterisk/Asterisk SCF Project Wiki
8:05PM 1 Gotoif changed in 1.8?
2:36PM 1 Ring back problem on SIP calls. Order of 183 Session Progress and 180 Ringing
2:14PM 6 Migration from 1.2 to 1.8 in production
1:18PM 3 How to make the sum of a ${VARIABLE} + 1 ??
11:48AM 1 Asterisk linphone call dropping by itself
8:08AM 1 inbound call issue...
1:10AM 1 Asterisk and SIP a Provider in Brazil
12:29AM 5 ADSL Load Balancing
Tuesday November 2 2010
7:20PM 3 IAX or SIP - connecting two Asterisk servers together
7:06PM 0 Asterisk community services powered by Atlassian tools
4:48PM 0 Need testing: chan_unistim improvements
4:19PM 1 Feature Request for 1.10 - ISDN power-save mode
4:13PM 3 Asterisk, VoIP and Samsung iDCS100
1:13PM 0 Sip, Qualify=200 that doesn't qualify. How to signal this state to the Peer
1:02PM 0 Noise while passing channel using tde205p card
9:27AM 2 Ring Freq
Monday November 1 2010
9:11PM 4 Issue with asterisk
8:24PM 1 DISA problem in 1.8.0
8:21PM 0 Queue Group not forwaring calls to agents
7:47PM 0 Ringback problem. Order of "183 Session Progress" and "180 Ringing"
3:55PM 4 FW: Under heavy attack
11:53AM 0 2nd network interface for RTP/media
11:52AM 1 MoH and stuch channels
10:40AM 0 Force direct RTP
1:59AM 1 Music On Hold Help