asterisk users - Jun 2010

Wednesday June 30 2010
TimeRepliesSubject
11:32PM 7 Pbx_lua vs. calling lua thru AGI?
10:56PM 2 Warning spamming for any unsynchronized ISDN port with dahdi-2.3.0.1
6:39PM 5 Return agi script.
3:28PM 8 Problem with extensions in IVR and queues
3:24PM 3 Problem in establish call from a2billing users.
2:26PM 2 How to work Asterisk with Video Conference
2:20PM 8 queue command in asterisk 1.4 with macro-argument
12:33PM 4 Minimum modules required to run VoIP and CDR
10:32AM 4 Adding Congestion to CDR logs
8:28AM 19 Echo problem in VoIP-calls
2:58AM 2 RE How to break pri DID to multiple SIP Trunks
12:23AM 0 Long shot... Order Logix
 
Tuesday June 29 2010
TimeRepliesSubject
11:54PM 4 Anyone can share their config file for Cisco phone please?
9:50PM 6 Dial options not working
9:23PM 1 Can't call my extension
9:14PM 1 Carrier needs more call examples
8:28PM 2 Asterisk 1.6 (and 1.4) DTMF problems using RFC2833
8:24PM 0 libpri 1.4.11.3 Now Available
7:50PM 0 Migrating from key system to asterisk
6:36PM 1 Strange Asterisk/SIP call forwarding behavior
4:51PM 4 Voiceprompts i.e. voicemail and conferencing in multiple codecs
2:39PM 7 Find a way to block brute force attacks.
2:06PM 3 SIP Delay with remote stations?
1:53PM 22 What TERMINAL software do you use for MS Windows platform and WHY?
1:10PM 0 OrderlyStats SE 1.6.2l now available.
12:32PM 6 cmd Authenticate
12:12PM 1 Problem with GoToIfTime
12:02PM 12 peer IP address in CDR
9:19AM 3 Is Centos 64 bit or 32 bit better?
8:11AM 1 Hot to configure trunk in asterisk with a2billing.
7:23AM 1 How to Add IP address to SIP Domain
6:41AM 1 transfering active call to user's voicemail
6:04AM 0 T.38 Peer Negotiation Fails
5:15AM 0 How to configure key sequence in features.conf.
2:20AM 1 What are the guts of AsteriskNOW and how it compares to other popular flavors available?
2:04AM 7 What‘s the best operating system suggest for Asterisk 1.6.2.9
12:40AM 22 Update the LCD with the callee's name after dialing
 
Monday June 28 2010
TimeRepliesSubject
11:58PM 4 restricting sip users to a certain useragent
8:27PM 1 Problem with TE411P and DAHDI
7:39PM 6 Asterisk 1.6 and multiple parking
6:49PM 3 sip server
4:00PM 3 Handling DTMF for number 4
3:45PM 1 Never seen Problem !!!
2:00PM 3 Pickup a ringing Queue member
12:08PM 2 sip add header
9:15AM 2 Problem attended transfer with ilbc
7:16AM 1 Use one group for ISN truncs
7:07AM 0 Use one ring-group for ISN truncs
 
Sunday June 27 2010
TimeRepliesSubject
10:23PM 0 CID
4:58AM 0 Vestec Tech Support
2:09AM 2 append CID label
 
Saturday June 26 2010
TimeRepliesSubject
11:25PM 1 Support from Vestec
9:08PM 7 Codec negotiation
4:57PM 0 Truth in advertising
1:21PM 4 [voice mail] Estimating file size?
12:25PM 2 Up-to-date list of Asterisk appliances?
11:33AM 2 Detecting hook flash in asterisk
9:28AM 4 Error - Failed to extend from xxx to xxx
 
Friday June 25 2010
TimeRepliesSubject
11:07PM 2 Non-native codecs - MELPe?
7:29PM 1 sip_xmit: sip_xmit returned -1: Operation not permitted
5:03PM 0 Meetme delay - normal?
4:57PM 2 Call drops on group paging asterisk - 1.4.22.1
3:23PM 1 Configure WAN Phone
11:25AM 11 Is there a default dial plan that is not in extention.conf?
7:48AM 3 G729 license key registration
6:59AM 18 [CRON] Right way to restart Asterisk and Zaptel?
3:24AM 8 Big time system
 
Thursday June 24 2010
TimeRepliesSubject
7:46PM 5 OT: Bandwidth calculations
7:32PM 2 T.38 on a MAX/Lucent/Ascend TNT
6:54PM 2 SPA8000 outbound CID problem
5:55PM 4 Dialplan for conference
5:53PM 0 A lot of : doing dnsmgr_lookup for - Asterisk installed from YUM
3:49PM 2 dialplan reload 1.4.33
11:36AM 2 Friday at 1PM: SIPVicious has a new tool: svcrash
10:26AM 4 Astersik can not detect DTMF key
7:34AM 6 Very strange registration problem
5:52AM 0 87.230.80.186 - Trying to register
4:10AM 0 parking on ast 1.6.2.8
 
Wednesday June 23 2010
TimeRepliesSubject
11:21PM 0 50 mantis issues marked 'Ready for Testing'
8:45PM 4 "Hidden" memory leak
4:08PM 19 one for your filters
3:05PM 3 I look ARI (Asterisk Recording Interface)
12:44PM 2 help with sip 401 unauthorized
12:04PM 0 Hangup Detection Problem In Turkey
11:20AM 6 Need USA DIDs
7:21AM 0 CallWaiting
 
Tuesday June 22 2010
TimeRepliesSubject
11:06PM 0 Asterisk 1.4.33.1 Released
9:55PM 0 SMS in landline
8:38PM 2 Xorocom Missing files...where to get it? astribank_upgrade
7:31PM 4 joining 2 conferences together
6:21PM 0 Endless loop with asterisk directory
6:21PM 11 Asterisk distribution for a Call Center
6:04PM 9 Workaround for bug in Linksys Firmware 6.1.3(a) (or greater)
5:34PM 0 SkypeKit
5:27PM 1 Running SIP on non-standard ports
5:24PM 1 Internal timing bad for Fax?
5:02PM 6 Call file structure and syntax
4:41PM 2 Sangoma - how to show channels in use?
4:03PM 0 Video not working with PortSIP SDK
1:30PM 9 PRI span problem - no D channel
12:01PM 1 Unregister and register SIP phones by using num pad on phones?
11:26AM 9 Anybody using TE410P on BT ISDN with DAHDI?
10:36AM 9 UDPTL T38 via NAT
9:07AM 0 asterisk-users Digest, Vol 71, Issue 36
8:51AM 1 storing DTMF inputs
8:13AM 6 Local channel usage
8:09AM 3 NO ANSWER before playback or background function?
7:47AM 0 Unable to set callerid for incoming skype calls
6:11AM 2 Generate cdr on Hangup
4:58AM 2 Can't make calls out of Astribank - show regdump doesn't show voltage - Getting warning due to old driver
1:41AM 0 Update to chan_ooh323 wrapper
 
Monday June 21 2010
TimeRepliesSubject
9:38PM 2 AMD
9:09PM 4 How do I access the Dialstatus numeric code received?
7:04PM 10 when to use e1/t1 card?
5:39PM 1 How to tell if a dropped call is my fault
5:10PM 4 Polycom firmware: split vs. combined
5:04PM 1 How to find a single call in logs
4:25PM 3 Create Conference and exit myself
4:08PM 3 DAHDI: Inbound BRI call, DDI not presented
3:45PM 2 What is the voicemail "u option"
3:39PM 0 Switchboad like application
3:08PM 5 Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?
12:54PM 1 using call file
12:48PM 9 [AGI] What scripting language for embedded hardware?
12:32PM 3 ISP down internal phones become unavailable
7:27AM 5 features.conf - parkedcalls - transfer
7:25AM 0 call dialing
4:06AM 1 Asterisk 1.6 + Jabber crashes
 
Sunday June 20 2010
TimeRepliesSubject
11:45PM 3 Compiling H323
4:21PM 0 Deleting some of the CDR data - How to do it safely?
3:36PM 0 load balance meetme
 
Saturday June 19 2010
TimeRepliesSubject
3:47PM 3 dahdi span
2:58PM 4 Using SetVar with System() is it possible?
2:34PM 0 OT - Explain RDNIS
12:38PM 0 Asterisk ODBC
9:21AM 3 Muti Asterisk
9:05AM 0 playing file when using call file in /var/spool/asterisk/outgoing in asterisk
8:29AM 9 Voicemail ODBC
7:09AM 3 asterisk appache issue
5:19AM 1 Can sip clients connect with each other directly (RTP session) ?
1:58AM 4 Linksys SPA94x keep-alive reply replies to wrong address (1.4.32)
1:43AM 2 dahdi modules installed wrong location
 
Friday June 18 2010
TimeRepliesSubject
9:03PM 1 Asterisk 1.6.2.9 Now Available
9:03PM 0 Asterisk 1.4.33 Now Available
7:22PM 1 How to get asterisk to playback personal greetings using grandstream gxp-2000
4:40PM 22 Why asterisk down when inet server down?
3:09PM 0 Relaunch of the Kansas City Asterisk User Group
1:54PM 1 question on nortel sip connection
12:30PM 5 Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'
12:24PM 1 OT: Physical SIP phone with inbuilt VPN support
10:38AM 0 Friday June 18th at 12 Noon EDT: Session Border Controllers, 1PM Bria iPhone SIP app
10:12AM 9 asterisk issue
8:51AM 4 Automatic attendant - Error in CLI.
8:36AM 3 CDRs not getting generated on Free PBX
8:21AM 2 What's diff between ${CDR(start)} , ${CDR(answer)} , ${CDR(calldate)}
7:20AM 0 device or sound card busy
6:54AM 0 Weaknesses of Asterisk still there?
 
Thursday June 17 2010
TimeRepliesSubject
8:26PM 0 calls dropped after 20 seconds in a non NAT situation
7:23PM 9 Music on Hold problema
5:21PM 3 Asterisk SIP/IAX peers can't connect after Firewall change?
4:59PM 3 applicationmap and ChannelRedirect
3:00PM 0 app_swift v2.0 released
2:58PM 2 Slightly OT: Cisco SPA525G and network errors
2:54PM 9 IVR extension dialing error
2:21PM 1 Asterisk no audio on calls problem.
1:37PM 6 Check if variable contains strings
12:24PM 1 VAD and cRTP, any thing else?
11:53AM 1 DTMF detection issues
9:06AM 1 calling machine over sip
8:46AM 0 error message in CLI regarding SET Timeout
7:18AM 0 writing echo in inbound file
1:38AM 4 Asterisk + Dahdi does not work with BRI NT mode
 
Wednesday June 16 2010
TimeRepliesSubject
9:54PM 3 DAHDI PRI error message
7:40PM 0 Call hangs up after exactly 1 minute
3:58PM 4 read data from file system and put in a variable
3:38PM 0 H323 Trunk Problem calling from Asterisk to Avaya PBX
3:31PM 3 TDD/TTY Support
3:25PM 1 Blind transfer feature
1:21PM 2 ring no answer / RONA versus HangUp
12:21PM 7 Asterisk + E1 card
8:28AM 1 Problem with dahdi and with freepbx
8:15AM 0 Asterisk +Dahdi does not work with BRI NT
6:21AM 0 asterisk sip trunk configure
2:07AM 0 Fwd: [INSTALL #RKZ-745226]: Digium Support Survey, Partial Faxes
 
Tuesday June 15 2010
TimeRepliesSubject
10:00PM 1 numbers
9:21PM 0 Extract user part from SIP URI
8:36PM 2 Voicemail vm-intro played even when temp greeting is setup
6:29PM 2 Asterisk hangs up for some calls
4:09PM 15 Unable to pickup an extension, tryi
2:50PM 9 Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'
2:24PM 0 Asterisk reject SIP INTITE from different
1:57PM 1 Corba interface
1:22PM 5 Cutting the CallerID(RDNIS)
11:43AM 7 Asterisk reject SIP INTITE from different source ports
8:07AM 6 can't seem to register, status unmonitored
5:22AM 3 Skype for SIP
5:17AM 1 Skype for Asterisk - what processors/platforms does it run on?
1:31AM 14 a2billing for residential voip usage
 
Monday June 14 2010
TimeRepliesSubject
10:02PM 3 Configure Voicemail for Large Systems
9:59PM 0 Small PC for Asterisk appliance to support 2 x Sangoma A200 (2 x PCIe standard cards)
4:00PM 17 How to pass variable back and forth from dialplan to php file?
3:47PM 3 How to disable day light saving on Snom 360 phones?
3:45PM 7 Unable to pickup an extension, trying everything
3:22PM 0 Hint priority in RealTime
2:50PM 0 Multiple parking lots - 1.6
12:19PM 2 calling peer from server
11:41AM 2 Call queues - issues, can't make it work.
10:04AM 2 logging stopped suddenly
7:47AM 0 debug message: Internal timing is disabled
7:26AM 23 Small PC to build and run Asterisk
7:10AM 1 Issues running Asterisk + Iaxmodem + Hylafax on same machine
6:35AM 1 PSTN call hunting
1:19AM 0 No 2nd invite from asterisk after challenging original invite
 
Sunday June 13 2010
TimeRepliesSubject
6:59PM 1 AGI library for C/C++
6:23PM 0 Asterisk AMI
10:35AM 2 bug with Moh on MeetMe ?
 
Saturday June 12 2010
TimeRepliesSubject
3:30PM 2 MeetMe problem
7:17AM 18 Qwest PRIs
 
Friday June 11 2010
TimeRepliesSubject
10:15PM 0 Using 5th gen TE420 with Asterisk 1.2?
9:55PM 31 How to stop intruder from registering sip?
8:32PM 3 OT: Free DID/SIP accounts
8:11PM 3 Call ended after 31 seconds
6:35PM 5 WARNING message when play
4:43PM 3 contacting
4:22PM 0 Asterisk SIP realtime and realtime DB tools
4:08PM 3 asterisk log problem
2:39PM 3 MeetMe
2:31PM 6 no ring back 180 with SDP
1:49PM 0 ZA16E and FXO-200 modules with asterisk
10:40AM 5 HDLC Bad FCS (8) on Primary D-channel
10:10AM 0 Pri show span and PtMP mode
8:22AM 1 chan_dahdi compilation with embedded
12:19AM 7 Dual Atom mobo - call capacity
 
Thursday June 10 2010
TimeRepliesSubject
9:05PM 0 Eyebeam hangs when you dial an unavailable number
7:24PM 3 Priority between calls in different queues
6:59PM 6 ISDN -> SIP
4:49PM 7 tuning software echo cancellation
3:10PM 1 Am I having problems with codecs? or am I not receiving an invite at all from my DID provider?
1:46PM 1 understand which asterisk thread is consuming CPU
1:02PM 4 Ring + Music on Hold in the same call
11:52AM 1 warning : sip_xmit
10:56AM 1 Group call limit
10:32AM 0 Loud Noise when trying to call through PSTN.
8:38AM 1 asterisk registration
7:05AM 0 Dial with MOH
6:49AM 0 How to kick/mute using ConfBridge application
5:15AM 1 CDR in case of CallForwarding
 
Wednesday June 9 2010
TimeRepliesSubject
9:28PM 1 [compat] section in asterisk.conf : compatibility with pipe delimiter
6:41PM 0 1.6 how to use groupcount and counteronpeer in queues to avoid ringinuse
1:49PM 2 SIP Witch
1:30PM 1 OT - Astmanproxy download broken ?
12:19PM 1 get Asterisk version from within dialplan
12:19PM 2 PSTN-IVR call
9:19AM 0 AMI Queue information about incoming call's channel before link
1:50AM 0 CID name in Facility message for Q.SIG
 
Tuesday June 8 2010
TimeRepliesSubject
11:11PM 1 early media issue from phone co.
9:06PM 0 NMI received for unknown reason
8:25PM 0 (no subject)
7:33PM 5 own Caller ID
7:30PM 0 memory leak
6:20PM 1 Asterisk-Addons 1.6.0.6 and 1.6.1.4 Now Available
5:48PM 0 libpri 1.4.11.2 Now Available
3:41PM 1 LumenVox *.gram reload
3:40PM 7 reloading realtime sip peers
1:50PM 6 Deleting extension makes it usable?
1:24PM 0 Problem with iax2/rsa registration
1:09PM 14 Out of Office
12:52PM 5 Limit total length of calls to a specifig SIP peer
10:38AM 0 Unavailable issue with SIP realtime and app_queue (*-1.4)
3:44AM 4 Issues with Vestec ASR
 
Monday June 7 2010
TimeRepliesSubject
9:10PM 0 Announcement before absolute timeout / how to terminate a meetme conf?
5:27PM 1 IAXmodem in dialplan
3:22PM 0 Still no(isy) app_jack in the box
8:33AM 4 How to play Floating point numbers?
 
Sunday June 6 2010
TimeRepliesSubject
6:48PM 11 Error of FreePBX after installing from Yum Repository of Asterisk
6:46PM 3 problem with port 5090 registration
4:27PM 2 Assign dhadi channel to several groups
1:33AM 0 Strange problem with zap channel.
 
Saturday June 5 2010
TimeRepliesSubject
11:41PM 17 Controlling calls
8:16PM 13 Still sipping frustration - only getting state ACK
4:41PM 0 Queue with PopUP screen for customer
4:26PM 5 Can one adjust the voicemail-menu when using VoiceMailMain() ?
4:23PM 0 dsp.c: digit_state.current_len
2:13PM 4 Problem with GROUP()
 
Friday June 4 2010
TimeRepliesSubject
11:54PM 1 originating a sip call from the CLI
9:58PM 4 Press twice *
8:38PM 3 1.6 issues
4:09PM 1 Using Local in queues a good idea? (or at least not a very bad idea?)
12:40PM 4 Asterisk on Ubuntu
10:58AM 0 BerkeleyTIP Join June Global Free SW HW Culture Mtgs via VOIP or in Berkeley
8:52AM 2 Create dialplan restrictions based on the IP Address of the SIP Client?
2:28AM 1 Wierd error when compiling 1.6.2 branch from SVN
 
Thursday June 3 2010
TimeRepliesSubject
11:00PM 0 Small VoIP company looking for Asterisk Scalability and Maintenance Engineer
9:04PM 1 other codecs
7:56PM 6 problem with inserting records into cdr
7:52PM 8 11.6.2 segfaults after dtmf on dahdi channel
6:40PM 0 SIP: match_auth_username=yes doesn't seem to work
6:07PM 8 <UsingWaitorPlaybackinhextension@gmail.com>,
5:32PM 0 OT: Cisco ATA 186
4:09PM 2 Codec G.129 A vs A/B
1:24PM 7 how to get call duration
1:16PM 7 Is this failed Asterisk setup typical?
11:12AM 0 how to run deadagi script after "status: expired"
 
Wednesday June 2 2010
TimeRepliesSubject
11:37PM 0 SIP message problems - retransmit and lost messages
9:35PM 0 libpri 1.4.11.1 Now Available
9:35PM 7 DAHDI volume
6:57PM 7 Persuing the gtalk issue - not only jack-related
5:03PM 1 timeout problem with basic conf
3:52PM 0 sipconnect 1.0
11:03AM 6 How do you hangup a call without terminating your session?
5:57AM 11 HElP me I am a beginner
 
Tuesday June 1 2010
TimeRepliesSubject
9:00PM 4 Definite app_jack trouble - unsolvable
8:51PM 4 Voicemail bug(?) with Asterisk 1.6.2.8-rc1
8:38PM 0 Asterisk 1.4.32 Now Available
8:36PM 2 Asterisk 1.6.2.8 Now Available
7:19PM 11 no sound between extensions
5:51PM 0 Caller id, sip header from problem
4:21PM 1 Asterisk and gtalk part 2
2:27PM 8 Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward)
1:42PM 0 Silence suppression and internal timing
10:37AM 0 Getting ANI on UK BT ISDN - Is SS& required?