asterisk users - Jan 2010

Sunday January 31 2010
TimeRepliesSubject
10:51PM 1 SIP Registration Failure Logging
9:08PM 0 request for testing: MixMonitor Mute
8:54PM 3 sip to dahdi and billsec
8:47PM 0 asterisk-users Digest, Vol 66, Issue 75
 
Saturday January 30 2010
TimeRepliesSubject
8:53PM 2 FAX over ISDN PRI
1:48PM 22 MATH
12:29PM 0 Aastra RFP-32 and CLID
8:57AM 1 forward call back up same trunk to external cell phone problem
3:57AM 8 Astribank problem
2:27AM 0 Asterisk status "488 Not acceptable here" on receiving fax
 
Friday January 29 2010
TimeRepliesSubject
11:20PM 0 Caller ID not working properly on some phones...
11:07PM 4 callerid not working over sip
10:25PM 6 Help for MOH - sounding scratchy/static on hold
8:09PM 1 Digium fax - sending fax call file vs manager originate
8:07PM 0 Broker lines on a T1 : Signaling convention?
7:03PM 0 New feature: Asterisk Manager Interface commands for DeviceState
6:07PM 4 microphone on Polycom 550/650
5:54PM 2 Questions about asterisk and spa2102
5:12PM 0 smsq command
3:29PM 1 Problem with ringing (or absence of...) with Polycom forwarding
3:13PM 3 Cell phone redialer?
2:09PM 3 1 Asterisk server, multiple registrations to ITSP
1:25PM 8 disable comfort noise
1:10PM 0 Address family not supported by protocol
11:39AM 0 VUC Today at 1 PM EST: Counterpath/Bria
11:30AM 0 chan_mobile problem with audio (distorted)
12:15AM 7 Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
 
Thursday January 28 2010
TimeRepliesSubject
11:08PM 2 beroNet BN4S0e PCI Express ISDN Card with chan_dahdi
10:57PM 2 Cell Phone dialing
9:31PM 0 How to set sip client idle or busy in Asterisk ?
9:29PM 0 AsyncGoto/DAHDI ?
9:17PM 14 Use of "603 Declined"
8:28PM 4 Dial cellphone from one PBX1 to PBX2? is it possible?
7:25PM 5 TDM2400 card FXS problems
5:09PM 14 911, location
4:37PM 5 rtp.c:883 ast_rtcp_read: RTCP Read too short
4:25PM 0 New feature in app_queue: Give members a penalty time for not answering (help testing)
2:16PM 0 Polycom Soundpoint 300IP
12:21PM 4 Fw: OfficeSIP Softphone
11:23AM 1 Inserting white noise / music / sound file into mixmonitor
10:02AM 0 Asterisk Database
8:14AM 1 iax client for symbian s60
8:12AM 6 iax softphones - not reconnecting
7:14AM 0 Database Configration
2:32AM 4 Linux-based hard phones?
1:23AM 0 yum install asterisk16 for Fedora Core 8
 
Wednesday January 27 2010
TimeRepliesSubject
9:18PM 4 Data transfer
4:39PM 1 Asterisk Database Configuration
3:48PM 4 Mitel integration
3:28PM 0 Need recommendation for ISDN-BRI cards for use with Asterisk
1:59PM 7 CDR problems with Queue
1:23PM 1 Asterisk, NAT, and RTP?
11:07AM 2 astdb
10:47AM 7 Unregistred users can pass calls, peer being static
10:00AM 3 Connecting to an External EPBX without an SIP provider
9:25AM 4 CDR messed up when using queue
2:27AM 4 Realtime Queue not work in 1.6.2.1
 
Tuesday January 26 2010
TimeRepliesSubject
3:54PM 0 Problem with Digium card, not transfering outgoing calls [Solved]
3:48PM 3 Attended Transfer with REFER
3:39PM 0 Anyone going to HD Communications Summit - Europe Feb 12th?
3:10PM 2 [inter-pbx commnication] trying to make PBX1 talk to PBX2
12:22PM 0 settings for soft phones
11:11AM 2 Error and call drops
9:03AM 3 Sip Trunk takes incomming calls for 2 minutes and then stops
4:21AM 1 Asterisk 1.2.37 + BLF + ParkedCalls + SPA962
2:16AM 0 StopPlayTones() after first digit?
 
Monday January 25 2010
TimeRepliesSubject
10:58PM 1 Disa not fully bridging outbound call
8:57PM 8 How to make SpeechBackground keep playing if utterance doesn't match our grammar
8:50PM 0 [OT] spa3000 (Regional & Line1) NL settings required
3:20PM 0 Problem with Digium card, not transfering outgoing calls
2:52PM 4 ASTSBINDIR not being picked up by safe_asterisk
2:26PM 0 Call tagging
2:22PM 5 Call recordings and sensitive information
1:27PM 4 sip.conf with versatel and two NICs very strange problem
12:10PM 3 queue
10:24AM 1 Web-Meetme 4.0 and Asterisk 1.6.2
9:51AM 9 Detected digit 'f'
9:12AM 2 MYSQL grammar diff in 1.6.2.1?
7:46AM 3 MySQL RealTime Error
5:44AM 0 Adminpin for conference room
 
Sunday January 24 2010
TimeRepliesSubject
10:05PM 1 [OT] Snom M3s
9:19PM 2 two stage dialing in a SIP dial plan
7:51AM 2 ReceiveFAX and SendFAX questions
7:25AM 0 AOC advise of charge
2:08AM 4 odd issue with the with SIP over VPN
 
Saturday January 23 2010
TimeRepliesSubject
5:47PM 0 Jabber Server
2:11PM 2 fax over IP - http/ftp-provisioning - intercom
1:14PM 2 Xorcom problem after update from zaptel to dahdi-2.2.1
12:12AM 2 IAX ans SS7
 
Friday January 22 2010
TimeRepliesSubject
11:58PM 5 Siemens Gigaset + Asterisk Query?
11:43PM 0 Handling SIP error codes/ISDN codes
3:50PM 9 Polycom phone DND state
2:22PM 11 Set CDR userfield for Queues
12:26PM 21 Snom vs Polycom
12:14PM 0 Asterisk 1.6 mysql 'NO ANSWER' disposition problem
11:53AM 0 Meetme conferencing - large deployment SIP or ZAP?
11:49AM 0 OT - SPA3102 not detecting CID - Which settings to tune ?
9:35AM 0 FW: Call Xfer issue between DataCenter and User Site
9:13AM 4 OfficeSIP Softphone
8:51AM 7 GoToIfTime issue
8:06AM 11 MYSQL problem
2:08AM 5 Trouble getting feature codes to work
12:56AM 3 Popular Gigabit Phones
 
Thursday January 21 2010
TimeRepliesSubject
8:40PM 4 pri CLI command not available
5:42PM 0 chan_ss7 or libss7, which is more stable?
5:35PM 1 Echo cancellation in a sip channel
3:51PM 10 odbc question
1:48PM 0 Feature codes not detected
1:46PM 3 Caller hang up not detected
10:59AM 1 DTMF reception during WaitForSilence
10:50AM 1 Asterisk & LDAP authentification
4:27AM 1 Pass-through Call Recording Transfer Information
3:17AM 1 Asterisk 403 Forbidden message with port translation
 
Wednesday January 20 2010
TimeRepliesSubject
11:20PM 2 Setting MixMonitor options from Queue
10:28PM 16 Virtual Asterisk Installation
10:13PM 2 queue groups in asterisk 1.4
10:06PM 0 DAHDI-Linux 2.2.1 and DAHDI-Tools 2.2.1 Released
8:57PM 1 Using SIPPEER status with CUT function? SOLVED
8:42PM 2 Using SIPPEER status with CUT function?
7:16PM 3 DTMF Issue?
6:24PM 0 More than a line with same extension + Polycom + Provision Tool
5:57PM 3 Call Xfer issue between DataCenter and User Site
4:51PM 2 More than a line with same extension + Polycom 320 + Provision Tool
3:40PM 1 AstLinux 0.7.0 Released
2:39PM 5 Polycom Soundstation Conferencing Unit
2:19PM 2 Odd message: "correct auth, but ..."
2:17PM 0 Selecting IP address for RTP
12:16PM 0 sendtext() SIP MESSAGE to Bria or Eyebeam
8:56AM 0 Which to choose? Realtime extension OR Static extension with MYSQL command
 
Tuesday January 19 2010
TimeRepliesSubject
9:17PM 2 How to enable a range of IP addresses in realtime sip_buddies
6:56PM 0 Initialize mailbox greeting message
6:48PM 0 Call drop-out on second incoming call.
5:33PM 3 ast_queue_log to mysql asterisk < 1.4 ?
4:46PM 0 Detecting incoming faxes and forwarding to phone fax machine
2:59PM 3 test case with queues and system()
7:08AM 0 B2bua
6:46AM 3 wav to gsm can't play
2:07AM 0 Asterisk answers with wrong sip entry
 
Monday January 18 2010
TimeRepliesSubject
8:27PM 0 TDM2400P "Unable to set SW Companding on channel .."
5:13PM 29 Dahdi/callerid issue
7:23AM 1 How to play the voicemail recorded?
7:12AM 0 What's customer_id mean?
2:11AM 0 Will SIP connection stop automaticlly when detect no voice between the channel after a period of time?
 
Sunday January 17 2010
TimeRepliesSubject
8:25PM 10 help with picking out a digium card.
5:39PM 2 Dial String command after audio background
2:34PM 2 receive text
11:09AM 9 How to escape characters in Dialplan
11:05AM 0 How to escape the Pound-Char in Callfiles
 
Saturday January 16 2010
TimeRepliesSubject
10:55PM 0 Do any providers support speex codec?
5:47PM 0 Anyone have provisioning documentation for LeadTek devices?
3:54PM 9 Cross compiling Asterisk, Dahdi..
12:04PM 5 Hint for realtime peers
10:17AM 0 Asterisk 1.4 or 1.6 automated install
10:02AM 4 Howto regret blind transfer?
3:49AM 0 Realtime cached values
12:02AM 1 Echo on Polycom phones
 
Friday January 15 2010
TimeRepliesSubject
6:11PM 0 TE410P generates only 1 interrupt
5:27PM 8 Asterisk 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
5:23PM 0 info for Busy for incoming internal call but not for exterrnal
5:17PM 1 Digium Asterisk World at ITEXPO - Yahoo keynote update
4:52PM 2 Changing ring cadence on FXS lines
4:06PM 3 DAHDI and Analogue lines (UK)
3:29PM 0 Asterisk 1.6.2.1 Now Available
3:29PM 0 Asterisk 1.6.1.13 Now Available
3:28PM 0 Asterisk 1.6.0.21 Now Available
3:27PM 0 Asterisk 1.4.29 Now Available
2:50PM 0 Getting Answered Stations instead of Group in cdr?
2:25PM 0 : Asterik with out registration.
1:43PM 0 Logs problem of queue_log-mysql
12:32PM 0 OT: Inbound South America numbers
11:56AM 5 jitterbuffer and PLC
10:48AM 6 Realtime queue not work
7:23AM 1 Question about Presence and IM feature
6:54AM 18 10/100 voip phones and gigabit connection
 
Thursday January 14 2010
TimeRepliesSubject
6:41PM 0 Friday Jan 15 @12 Noon EST: Hacking VoIP
6:38PM 3 GXV3140 and Xlite video
5:35PM 6 Fax Detection on SIP
4:20PM 1 Dahdi and FreePBX
4:11PM 0 Ringing for incoming call
4:03PM 1 Can not play WAV-files attached to mail from my own Asterisk
3:33PM 2 Languages
3:22PM 3 Dahdi issues
2:53PM 8 iaxmodem / hylafax receive problem
2:15PM 3 Followme Options
1:48PM 0 different between ring groups and queue?
10:53AM 2 Ringing issue
10:08AM 1 Lagged Extension
9:55AM 4 how to strip + from the caller-ID
9:10AM 0 What about the performance visit MYSQL in DialPlan code?
9:07AM 0 Attend CampKDE Jan 15-18 via Voice over Internet (VOIP), BerkeleyTIP
4:45AM 0 ISDN Cause codes for unanswered calls
3:29AM 2 is there some Chinese version of sounds available?
 
Wednesday January 13 2010
TimeRepliesSubject
11:48PM 0 asterisk / NEC2400 / PRI
10:07PM 0 Asterisk 1.4.28 intermittent one way audio?
4:49PM 0 FW: [mythtv-users] VMWare on the backend. Viable solution?
5:56AM 8 Major benefits when upgrade from Asterisk 1.4 to Asterisk 1.6?
3:56AM 4 Polycom Mute Problem
2:59AM 1 Odd Voicemail Issue
 
Tuesday January 12 2010
TimeRepliesSubject
11:05PM 4 Xorcom 32 channel FXS gateway
10:48PM 1 AudioCodes MP-114 - SAS (Stand Alone Survivability) configuration
7:07PM 1 Problem logs queue_log-mysql
5:57PM 0 VMs & IMAP Storage
5:56PM 4 Minimal Asterisk Web Interface?
5:43PM 3 SIP Security
5:38PM 15 Inserting a wait in a sip dial
5:16PM 3 Question about SIP registration
4:28PM 3 Send 503 or 603 error after answer()
11:57AM 0 Virtual ISDN device /dev/XYZ
10:55AM 24 Beginners Guide to setting up a Call Centre
5:57AM 5 Multi-Tenant Parking
5:56AM 0 Why agent log out automaticly?
3:26AM 10 is roundrobin and rrmemory the same meaning?
2:44AM 0 Interfacing to NEC Xen Master PBX
 
Monday January 11 2010
TimeRepliesSubject
9:49PM 0 Problem with call transfer and Polycom 430
7:05PM 5 SIP over VPN -- no audio to other remote/VPN connected phones
5:21PM 0 ChanSpy doesn't hangs up
5:05PM 1 MeetMe Conferencing - Announce your own join/leave to yourself and other conference members
4:58PM 0 Custom date formats with new mode say.conf?
4:26PM 15 Sipgate > DTMF not detected
3:23PM 5 Asterisk core dumps when using PrivacyManager
12:27PM 0 TONIGHT Join 5-6P Mon 11th - 1st Evening Meeting test IRC & VOIP online Asterisk at BerkeleyTIP-Global - for forwarding
11:25AM 2 Extension Status
10:57AM 0 Temporary loss of audio on all SIP channels
10:45AM 5 Attempted break in ?
7:43AM 0 asterisk-users archive
7:16AM 0 PHP-Script (AGI) doesn't finish after upgrading to 1.6.0.15
2:37AM 0 How to test if a telephone is busy now?
2:19AM 2 How to use AGI php script function $agi -> exec_dial
2:15AM 0 Zhang Shukun ?????
 
Sunday January 10 2010
TimeRepliesSubject
11:00PM 1 Weird Polycom SP 650
10:33PM 2 app_swift 1.6.2 DTMF issue
9:31PM 2 Problem with my dialplan
9:17PM 1 Grandstream GXW-4024
6:58PM 0 Directory and Voicemail Problems after upgrading from 1.4 to 1.6
3:29PM 0 Off-line subscribed phone amber on SPA942?
10:24AM 10 You won't help me anymore?
7:33AM 8 No dial-tone with X101P FXO card
1:00AM 0 Music / Background
12:06AM 0 Queue - Update CDR
 
Saturday January 9 2010
TimeRepliesSubject
9:22PM 6 Using HASH() and REALTIME_HASH()
6:03PM 1 UK dialing tone
3:22PM 1 Quick Installing Asterisk-1.4 on Ubuntu
1:00PM 5 Choppy MOH
9:57AM 0 Asterisk CallerId problem?
 
Friday January 8 2010
TimeRepliesSubject
11:11PM 1 How can I get codec info on active calls
10:37PM 1 Multicast RTP Paging
8:57PM 0 Queue_log file and mysql logs together!
4:47PM 0 Semi-OT: Configuring SIP trunks with Cisco UCM 7.0.
4:07PM 0 [VUC] Today at 12 Noon EST (6PM CEST, 9AM PST) iNum with Voxbone
3:33PM 0 Cheap femtocell's ahead
9:14AM 1 How to recieve number returned by $AGI->wait_for_digit($timeout)
 
Thursday January 7 2010
TimeRepliesSubject
11:27PM 11 AGI perl script set timeout within script?
10:20PM 1 voicemail /odbc problem
8:28PM 1 Crash in Asterisk
6:19PM 0 dns messages on console
4:15PM 3 Sip REFER failes w/603 Decline (Policy), Polycom Phone
3:49PM 27 Please remove me from the mailing list.
2:06PM 0 Dialing OutBound SIP trunk using Dial() command
1:38PM 3 How to dial a number make two phone Ring at the same time?
11:11AM 0 queue and linear strategy
11:00AM 2 Explain what asterisk.conf's "internal timing" option is
8:19AM 1 error compile dahdi with latest kernels.
6:08AM 1 compile one additional module without recompiling all asterisk
4:16AM 0 Question about PLC of Asterisk
1:53AM 3 How to see STDERR message?
1:07AM 0 video with x-lite
 
Wednesday January 6 2010
TimeRepliesSubject
11:23PM 6 iaxmodem to ReceiveFAX crashes Asterisk equipped with B410P
10:52PM 0 Urgent: Which spandsp version is recommended for 1.6.1 ?
9:18PM 0 DEVICE STATE "In use"
7:45PM 3 question on makefile
7:15PM 0 Asterisk 1.6.1.x SMDI MWI w/Fujitsu F9600 Problem
11:43AM 2 Zaptel compilation problems: Data Mode!!
10:44AM 2 Inquiry:How to define incoming route for sip?
9:36AM 0 MITEL
9:16AM 2 Fastagi-mapping problem
3:09AM 0 Originate from the Dialplan
2:40AM 2 Merlin Legend integration not routing calls back to PSTN.
 
Tuesday January 5 2010
TimeRepliesSubject
10:24PM 36 Faxing: Anyone have a compiled executable?
9:41PM 0 send faxes as "3,1 kHz Audio"
7:53PM 40 Really Silly Question From Total Newbie
7:25PM 3 Canadian call quality issue
2:38PM 1 DTMF detection on dahdi with b4xxp (again, some more details)
2:16PM 0 Newbie: MITEL and Asterisk
1:24PM 15 CallerID on Indian PSTN is not working.
12:33PM 2 Realtime LDAP Queues crashes
9:43AM 0 (no subject)
9:38AM 0 Get Queue Info
9:30AM 0 {Spam?} MeetMe/Dahdi for FreeBSD
6:00AM 0 automatic dial from database
4:53AM 0 Inquiry:Asterisk sending dialed digits in one-by-one digit format?
1:41AM 4 AGI and embargeability
 
Monday January 4 2010
TimeRepliesSubject
11:44PM 2 T.38 ITSP?
9:03PM 0 lpc10
8:09PM 0 Dialout from Meetme conference
6:48PM 0 Register sip FXO per gateway
5:57PM 0 H323 Disconnects after 15+ minutes
4:46PM 2 Script to show asterisk stuff
3:21PM 3 caller getting cut off intermittently
1:42PM 5 ZapRAS priviledge error
1:16PM 13 Dahdi and oslec
1:08PM 1 Realtime Queue Members Not Ringing
8:51AM 3 Some minor configuration issues with queues
8:34AM 3 DNS reload on trunks for outgoing calls
6:17AM 3 MYSQL queries from dial plan
6:14AM 0 Asterisk realtime chat
6:10AM 0 differences between asterisk 1.6.1.x and 1.6.0.x
5:24AM 2 Free FaxForAsterisk ReceiveFAX not working
3:13AM 0 Dahdi causes panic on server restart
2:14AM 4 Outgoing Calls Only -- Firewall Rules
 
Sunday January 3 2010
TimeRepliesSubject
9:08PM 0 asterisk-users Digest, Vol 66, Issue 4
12:04PM 0 Inquiry:How to join Asterisk real time chat?
 
Saturday January 2 2010
TimeRepliesSubject
2:01PM 7 Help getting info from caller
7:06AM 9 verifying correct loading of VPMADT032
6:22AM 0 Inquiry:Asterisk sip ?
 
Friday January 1 2010
TimeRepliesSubject
9:57PM 0 AudioCodes MP-114 2xFXS/2xFXO - FXO not working correctly
4:34PM 23 SIP Listen Multiple Ports
1:50PM 0 Happy New year 2010
12:01PM 6 PBX Extension Help
2:17AM 0 AudioCodes MWI