asterisk users - Jun 2011

Thursday June 30 2011
TimeRepliesSubject
8:35PM 0 patch openr2 asterisk 1.4.42
8:32PM 0 Skall vi gå på Smaklösa?
3:09PM 11 Cannot figure out pound key in qwerty keyboard
1:24PM 0 cisco sip
11:38AM 0 SendFax: not setting the fax header
8:22AM 2 asterisk recording problem
12:46AM 0 Problem callerid ignored by using callfiles
 
Wednesday June 29 2011
TimeRepliesSubject
9:57PM 4 No audio format found to offer.
5:57PM 0 atxfer fails to read data
5:46PM 1 Asterisk/SIP Issue - Long Shot
5:35PM 0 Asterisk 1.6.2.19 Now Available (Final Maintenance Release)
5:34PM 0 Asterisk 1.4.42 Now Available (Final Maintenance Release)
5:23PM 1 Asterisk 1.4 func_odbc frustrations
4:06PM 3 Polycom ip320 dtmf issues
1:49PM 10 Load Balance Trunks
9:58AM 5 OT - Polycom - Which provisioning protocol to choose ?
9:13AM 2 dialplan execution stops after ReceiveFax
 
Tuesday June 28 2011
TimeRepliesSubject
11:45PM 0 for suport server
8:54PM 0 Asterisk 1.4.41.2, 1.6.2.18.2, and 1.8.4.4 Now Available (Security Releases)
8:53PM 1 Clarification of the terms shown on CLI
8:31PM 0 AST-2011-011: Possible enumeration of SIP users due to differing authentication responses
7:34PM 2 IVR
7:10PM 0 Set a specific BLF key on Polycom 650 [SOLVED]
6:21PM 2 Add # at the end of dialled number
5:30PM 4 Asterisk 1.6 Dahdi on Centos 5.2
5:00PM 0 Using Dial() on SIP and DAHDI connections simultaneously
4:58PM 1 Set a specific BLF key on Polycom 650
4:32PM 2 Outgoing calls get dropped on high-latency connections.
3:30PM 6 MixMonitor - garbled/corrupted WAV files
10:59AM 2 No audio after a reinvite changing codec ----> with SIP DEBUG!!
9:59AM 2 Asked to transmit frame type slin, while native formats is 0x8 (alaw)
 
Monday June 27 2011
TimeRepliesSubject
7:53PM 0 Fax with Asterisk and T38Modem
5:45PM 0 Question regarding progressinband
5:21PM 2 Asterisk 1.8 Paging without DAHDI or MOH Shoutcast with DAHDI
5:02PM 0 rtptimeout on 1.8.4
3:33PM 0 FW: Asterisk 1.8.4 - Google iCal not working
3:06PM 0 interface to gstreamer
12:04PM 0 Festival VOICE MAIL TO FEMAIL
11:36AM 0 Asterisk changing SIP INFO dtmf duration
11:26AM 5 not find files in asterisk 1.8
11:11AM 6 ReceiveFax to G.711
9:30AM 2 Agi script for working hours PBX
1:25AM 9 Conference feature
 
Sunday June 26 2011
TimeRepliesSubject
7:55PM 3 Asterisk 1.8.4 - Google iCal not working
7:06PM 3 Cisco IP Phones 7942 and Skinny/SIP in asterisk
 
Saturday June 25 2011
TimeRepliesSubject
4:24PM 0 HDLC Overrun with Chan SS7
12:39PM 3 Cisco IP Phones and Skinny in asterisk 1.8.4.2 "tooooooooooooooooo"
 
Friday June 24 2011
TimeRepliesSubject
8:55PM 5 t.38 virtual fax software?
3:28PM 0 failed to authenticate user message
9:40AM 0 eComm - Are you going to San Francisco next week?
7:37AM 3 Vm on a System running Asterisk.
6:41AM 3 Monitor Asterisk and Ast-gui
4:54AM 0 issues/jira
 
Thursday June 23 2011
TimeRepliesSubject
8:14PM 0 Asterisk 1.4.41.1, 1.6.2.18.1, and 1.8.4.3 Now Available (Security Release)
2:12PM 10 Problem with detecting fax on PRI/DAHDI channels
1:47PM 0 menu asterisk
11:45AM 2 VMX Locator
9:01AM 0 How to set BRI-to-BRI trunk using 2 HA8+B400M cards [SOLVED]
 
Wednesday June 22 2011
TimeRepliesSubject
11:01PM 0 Fwd: FW: loadstar and ok one
8:00PM 0 Asterisk/Kamailio dinner in Madrid thursday next week - June 30th
4:51PM 0 Setting the Source IP on asterisk 1.8
4:09PM 4 Aastra phone # key in dialplan
3:28PM 4 How to set BRI-to-BRI trunk using 2 HA8+B400M cards
1:32PM 2 iLBC re-licence
1:22PM 0 Unable to include switch 'Realtime/@' in context
11:32AM 2 Digium HA8. Does settings order matters (in system.conf) ?
5:40AM 4 Office timings only work asterisk after that voicemail
2:10AM 3 Question on pause in dialing
 
Tuesday June 21 2011
TimeRepliesSubject
11:21PM 0 calleridname presentation Asterisk ==> Siemens
8:56PM 1 how to get on hold events with AMI
6:41PM 22 call paging interrupts call when using Mitel 5224
5:07PM 1 DTMF begin ignored
1:06PM 0 asteriks and ntt Japan
12:37PM 1 Call to *2*999... : IP-phone configration
12:23PM 0 Voice recognition recommendations?
10:17AM 0 AMI Suddenly not giving full response to 'Command'
9:57AM 0 dropped calls on android voip connection
8:48AM 0 xorcom asterisk patch for sending rtp stream to remote oreka server
7:28AM 0 Easier to remember transfer and pick-up key sequences
4:56AM 1 Looking for Sipura-2000 Latest Firmware
1:38AM 1 : Re: ITSP failover for PRI
12:43AM 6 gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko
 
Monday June 20 2011
TimeRepliesSubject
11:33PM 2 Inbound CallerID displays asterisk
11:09PM 3 "pickupsound = beep" kills call pickup in Asterisk 1.8.4.2
7:09PM 3 Get second cipher in an extension
6:09PM 8 Asterisk call limitation
6:00PM 1 Problems with pickupgroup/callgroup with Asterisk 1.8.4.2
5:17PM 6 menu issue
2:40PM 2 Integration of OpenVXI
11:04AM 2 different format in asterisk
5:56AM 0 Realtime Failover - Multiple DSN 's in extconfig.conf
 
Sunday June 19 2011
TimeRepliesSubject
10:44PM 0 ooh323 errors while compiling asterisk 1.8.3 and 1.8.4
5:42PM 4 ITSP failover for PRI
4:24PM 10 Problem with ReceiveFAX app from FFA
9:13AM 12 SMS with Asterisk
 
Saturday June 18 2011
TimeRepliesSubject
6:16PM 14 Direct RTP with Asterisk
1:33AM 3 ooh323 errors while compiling: asterisk 1.8.3 and 1.8.4
 
Friday June 17 2011
TimeRepliesSubject
7:24PM 1 Next Asterisk 1.8 Release
5:54PM 4 asterisk voicemail distribution groups
5:25PM 1 background audio for inbound leg
5:20PM 1 click to call
2:51PM 0 RTP Streaming
11:30AM 1 Missed calls and groups
 
Thursday June 16 2011
TimeRepliesSubject
10:12PM 2 Queue Log in Mysql
8:11PM 1 CDRs in 1.8
8:08PM 5 Bridged Digital call
6:09PM 0 show channels does not show hold status
1:21PM 1 fast AGI memory leaks
11:36AM 7 MixMonitor
10:49AM 0 Have your suggestions on Hardware configuration for Asterisk.
9:59AM 3 Cisco IP Phones 7942G (skinny): TFTP and required files
9:50AM 0 sipp application/dtmf-relay not work properly in Asterisk!
8:11AM 0 Channel variables not available during xfer?
8:11AM 2 Sending SMS on Friday at 12 Noon EDT
7:18AM 5 Inbound call not dialing exten
6:54AM 1 #include filename
5:52AM 6 How to secure our Asterisk server from hacker's ?
3:26AM 2 Web based call back
3:23AM 10 Goggle voice incoming dialplan
 
Wednesday June 15 2011
TimeRepliesSubject
5:27PM 1 Re connecting to SIP Provider with virtual IP, from pacemaker cluster
3:28PM 0 connecting to SIP Provider with virtual IP from pacemaker cluster
1:33PM 2 change destination on digit
11:06AM 4 DIGIUM PRI CARDS REQUIRE
11:00AM 0 CONFERENCE CONFIGURATION REQUIRE
10:53AM 2 sig_pri.c:985 pri_find_dchan: Span 1 No D-channels available! Using Primary channel as D-channel anyway!
10:20AM 2 VOICEMAIL CONFIGURATION
8:40AM 0 Asterisk - dialog-info+xml - NAT
8:17AM 0 How to configure unused BRI ports from a HA8 board ? [SOLVED]
7:30AM 1 How to configure unused BRI ports from a HA8 board ?
6:45AM 2 call file challenge...
5:34AM 3 Dial out conference
5:15AM 0 asterisk + stun
 
Tuesday June 14 2011
TimeRepliesSubject
11:27PM 3 sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway!
9:48PM 0 SPA504G Unable to Transfer Established Call
9:41PM 7 Voicemail issue
9:36PM 2 dahdi_genconf and BRI NT spans in system.conf
9:20PM 0 How to set a HA8 board + B400M in NT mode ? [SOLVED]
9:19PM 0 Possible timing issue?
8:48PM 0 sysmon on Centos Asterisk system using 100 perc CPU..How to kill it?????
8:11PM 1 How to set a HA8 board + B400M in NT mode ?
6:47PM 1 sig_pri.c:985 pri_find_dchan: Span 1: D-chanannel anyway!
6:19PM 11 Ground Start ATA / VOIP Gateway
3:37PM 1 Dahdi 2.4.0 and Squeeze [SOLVED]
3:36PM 3 Polycom BLF
2:26PM 3 Page() bumps user out of a call
1:44PM 2 Dahdi 2.4.0 and Squeeze
1:18PM 1 sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel
9:44AM 3 sig_pri.c:985 pri_find_dchan: Span 1: No D-chanannel anyway!
 
Monday June 13 2011
TimeRepliesSubject
6:44PM 6 PAP2T provisioning via SRV record?
6:19PM 0 multiple asterisk on 1 machine or other idea for using multiple network connection
5:57PM 0 skinny and 7961
4:55PM 8 No audio after a reinvite changing codec
4:10PM 1 call an external number for other server
2:33PM 1 AMI on hold events
10:44AM 4 asterisk queue 'ringall' stratagy
10:04AM 23 Cisco IP Phones and Skinny in asterisk
9:34AM 0 announce-frequency not respected
8:39AM 3 Communciation delay betwwn speakers
 
Sunday June 12 2011
TimeRepliesSubject
9:16PM 3 Configuring Cisco Phones to register on Asterisk: The configuration files
4:42PM 4 A question about Caller ID
 
Saturday June 11 2011
TimeRepliesSubject
3:42PM 0 vTiger component
3:37PM 11 Siemens gigaset as180 as a internal mobile extension
3:07PM 0 chan-mobile bug: bluetooth connection is disconnected immediately when the call hangup
2:29PM 8 TFTP to be installed in Linux same asterisk machine to be used with Cisco
7:42AM 2 Full SIP dial string
 
Friday June 10 2011
TimeRepliesSubject
9:26PM 6 Queue not sending call to Agent
8:49PM 1 Incoming Call Recording
6:31PM 1 Request: please test modification to EWS calendar functionality
6:27PM 2 asterisk 1.8 PRI random call drop issue
2:16PM 0 (no subject)
2:15PM 3 AMI question
11:00AM 12 Connected Line ID
9:32AM 2 Asterisk issue or VoIP provider issue ?
9:26AM 4 How to remove asterisk ?
6:07AM 1 [FreePBX] Digium addons
4:57AM 0 Friday: Sipgate, Yate and Astricon
 
Thursday June 9 2011
TimeRepliesSubject
10:57PM 1 Access Voicemail Asterisk 1.8 FreeBSD 8.2
9:32PM 7 Permanent restart after upgrade
8:10PM 0 Insert name in SIP registry
7:47PM 0 Polycom 501 Settings/subscription expiry
7:05PM 18 Question about voip.ms service.
6:53PM 2 Fwd: Re: ControlPlayback's options
5:40PM 3 SIP/IAX guest access?
3:10PM 0 IRC canal
12:27PM 0 Asterisk, attended transfers and DTMF mode
11:41AM 0 how to tell if call is on hold
2:00AM 0 Change to pickups in Asterisk 1.8 - not working on local channels?
 
Wednesday June 8 2011
TimeRepliesSubject
10:48PM 0 Question on how many phones
10:13PM 7 How asterisk use pri channel
9:32PM 6 LXC and Dahdi
6:43PM 3 No IVR listen at device end......SIP phone is working fine
6:40PM 13 CallerID issue
6:26PM 3 how asterisk work with VoIP trunk?
6:16PM 11 issues.asterisk.org/jira not working
6:09PM 1 Asterisk and Audiocodes PRI card
5:44PM 2 Interesting PRI issue
4:00PM 1 Update problem | CLI commands missing
3:26PM 0 call transfer back to a sourcing switch
1:57PM 16 Asterisk 1.8 broken MWI
9:55AM 0 Call queues on load-balanced asterisks
7:00AM 0 Hints problem - NAT problem?
7:00AM 3 Queue log in MySQL DB
6:49AM 1 Asterisk: BYE is received late
6:09AM 3 Looking for Email to Fax Solutions
2:20AM 3 After wiki.asterisk.org was upgraded my user no loger exists.
1:17AM 5 PRI hangup request, cause 18
 
Tuesday June 7 2011
TimeRepliesSubject
11:44PM 2 What is wrong in m
9:12PM 3 reload chan_dahdi.conf without disconnect active calls
6:17PM 3 why doesn't "s" accept incoming call
5:04PM 1 Asterisk 1.8 minimum modules/configuration
4:52PM 2 trunk between 2 servers
12:30PM 0 Call Conference and Call transfer and Voice mail settings
11:44AM 0 Refactor of CDR - Comments please.
11:13AM 0 Asterisk 1.6 - subscriptions.
8:59AM 5 Different callerid for different extensions
8:47AM 4 Connect intercom to Asterisk?
8:24AM 0 IPv6 and IPv4 NAT not working
6:34AM 0 chan_mobile one way comunication problem
6:31AM 4 how to know length of file in seconds
5:53AM 0 Is this feature or Bug of all Asterisk versions ?
5:31AM 2 How to get DTMF in Konference module in Asterisk
3:02AM 0 sccp problem
2:12AM 4 Pops & clicks at the end of sound files
2:11AM 2 tls/srtp: sip_xmit error: returned -2
1:19AM 7 PRI issue its BUSY
 
Monday June 6 2011
TimeRepliesSubject
9:59PM 0 half sip registration at 1.8.3
8:43PM 2 asterisk 1.8 issue with polycom dialplan
7:31PM 0 Subject Change: Playback/background App over Network (was: AGI STREAM FILE not working?)
4:59PM 0 unsubscribe
4:36PM 0 DAHDI - skipping sound on voicemail
3:55PM 1 Asterisk GUI - the one from Diguim/Asterisk - issues on Asterisk 1.6x
3:09PM 0 Bridged Call
2:07PM 3 Asterisk Online Training
10:57AM 0 About Asterisk SIP NAT Config
5:08AM 2 issues.asterisk.org
4:12AM 10 AGI STREAM FILE not working?
 
Sunday June 5 2011
TimeRepliesSubject
6:30PM 3 Asterisk users Calculation
4:25PM 0 DTMF issue in app_konference using with asterisk 1.8.3.2
4:18PM 3 broken SVN asterisk 1.8 ?
2:56PM 1 asterisk 1.6 - 511 Command not permitted causing high CPU usage
11:50AM 0 Blind transfer issue on Asterisk 1.8.4.2
 
Saturday June 4 2011
TimeRepliesSubject
6:05PM 1 example sip.conf for csipsimple?
6:30AM 0 Obtain SIP From and To Tag for CDR
 
Friday June 3 2011
TimeRepliesSubject
1:55PM 0 Queue base polycom custom ringtype
9:25AM 6 Voxbone numbers
7:08AM 3 MOH uploading is not working with 1.4
1:31AM 0 chan_dahdi.c, dtmfmute, rtp.c
 
Thursday June 2 2011
TimeRepliesSubject
9:46PM 0 ChannelRedirect
7:37PM 0 asterisk logger permission
7:27PM 0 asterisk-users Digest, Vol 83, Issue 3
7:07PM 2 Asterisk 1.8.4.2 Now Available (Security Release)
6:49PM 0 Help launch the stackexchange (stackoverflow) telophony site
6:46PM 3 How to continue processing a context after a Hangup
5:57PM 3 Can I use phone line to recive faxes?
3:03PM 4 RealTime Queue Logging in 1.8
1:01PM 24 benefits of asterisk 1.8
7:39AM 1 Does anyone know about asterisk 1.10
5:55AM 1 Three-way conference in Asterisk
 
Wednesday June 1 2011
TimeRepliesSubject
11:51PM 2 Migration from Mantis to JIRA
9:50PM 3 Question about "null routing" calls to DIDs we don't handle
6:31PM 0 Moh on transfer call
5:00PM 0 Immediate 180 Response on Invite
3:11PM 1 Dahdi_genconfig - "Empty configuration -- no spans"
9:11AM 0 DBdeltree: Error deleting key from database
6:06AM 15 busy hangup HDLC Bad FCS (8) on Primary D-channel