asterisk users - Dec 2010

Friday December 31 2010
TimeRepliesSubject
2:59AM 12 Base memory usage
 
Thursday December 30 2010
TimeRepliesSubject
9:58PM 0 Users of CEL Please comment on Bug
8:31PM 1 Find media and sip endpoints IP address durring "h" extension
6:29PM 1 VUC; Friday December 31st - 2010: The Year in VoIP
9:10AM 1 Force different codecs on call base
7:36AM 2 Usage Reports
4:55AM 6 call is not going to Voicemail with "1,n"
4:36AM 0 Music Jukebox and IVR voice chat
 
Wednesday December 29 2010
TimeRepliesSubject
11:15PM 7 GotoIf CALLERID(num)
11:16AM 17 Log and forward calls to cellphone?
8:16AM 0 Trixbox CE for Multi Tenant
 
Tuesday December 28 2010
TimeRepliesSubject
8:01PM 2 Replacing digital pri card
4:33PM 9 Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?
1:32PM 1 Queue announce parameter problem
12:10PM 3 How to reload queue on the fly?
6:26AM 2 OutCall for Outlook only shows Name from CLID and not Number - hence not pulling contact
1:42AM 0 How to use google voice for voicemail transcription
 
Monday December 27 2010
TimeRepliesSubject
9:40PM 4 Panasonic KX-TGP500 w/Asterisk
6:37PM 9 Asterisk 1.4.38 - unknown signalling bri_cpe
6:08PM 13 Using SIP stack within Asterisk to reboot phones - Possible?
5:10PM 0 CEL and custom values.
1:02PM 2 malformed SIP / routing issue
12:05PM 1 G729a and G729 interoperability
11:05AM 1 Queue Member relationship and AstDB
7:25AM 7 anyone who has experience with chinese clone cards like zycoo, ctvon, chinaroby, etross, iit, realtone
 
Sunday December 26 2010
TimeRepliesSubject
11:01PM 1 Asterisk 1.8 Realtime Queue not working
1:08AM 0 Which GSM-Modem to build a Gateway?
 
Saturday December 25 2010
TimeRepliesSubject
11:58PM 3 sip.conf, realtime, and LDAP
11:04PM 9 sip attack.. fail2ban not stopping attack
11:01PM 3 load balance with 2 wan connections
5:28PM 3 asterisk realtime & calling sip users
2:49PM 2 Remote VOIP/SIP Phones through two routers
1:31PM 4 Agents login
 
Friday December 24 2010
TimeRepliesSubject
2:40PM 2 One way crappy audio in iax call - Asterisk 1.6.2.15
1:36PM 4 live audio stream in asterisk
12:30PM 7 Moving asterisk from one network to another.
12:22PM 0 Prepaid Billing for Asterisk and Gnugk
12:20PM 0 Cisco IP Phones and AVAYA IP Phones: Provisioning the profile
10:12AM 9 SRTP unprotect: authentication failure
8:27AM 0 Today at 12 Noon EST
 
Thursday December 23 2010
TimeRepliesSubject
9:07PM 2 Zombie DAHDI FXO channels
6:52PM 4 Asterisk 1.8 and Realtime
4:33PM 0 Asterisk 1.6 iax auth rsa failed with policie not found
3:28PM 0 Panasonic trunk asterisk over h323
1:33PM 0 MOH RBT problem
10:51AM 1 OT - Alcatel OXE IP trunking licence price
7:41AM 2 Forward voicemail to group of people
5:41AM 0 Asterisk handling multiple simultaneous calls for IVR
3:06AM 0 Maximum retries exceeded
 
Wednesday December 22 2010
TimeRepliesSubject
9:23PM 0 Asterisk 1.8.1.1 Multiple Parking Lots
6:31PM 1 Siemens OpenStage phones and Asterisk
5:51PM 0 CDR on MySQL
5:42PM 9 Possible Bug (Include ${HANGUPCAUSE} in CDR)
3:14PM 2 How to list used extensions + assign extension to a roaming phone
3:02PM 0 setting up callerid
2:58PM 12 * 1.8: cannot load g729 free codec (on 1.4 it worked!)
2:49PM 2 dahdi-channels.conf for Digium TDM2400
2:23PM 17 Vacancy - Asterisk MySQL Support Engineer 45K South London
2:22PM 0 Include ${HANGUPCAUSE} in CDR
12:44PM 8 Asterisk hangs up call after 20s
11:50AM 4 Maximum E1 Ports on Asterisk ?
10:21AM 3 Wise selecting of outgoing channel
10:10AM 2 callerid and user on voicemail
1:51AM 1 Asterisk as a caller ID
12:58AM 12 Simplifying dial-plan
 
Tuesday December 21 2010
TimeRepliesSubject
8:20PM 4 What is equivalent function to "mv" command in php for Asterisk Spool directory usage?
12:51PM 0 Friend/user/peer in plain English?
11:39AM 4 MeetMe -> ConfBridge: hint not working
10:49AM 1 SOLVED: Re: Setting `userfield` from within a callfile
10:47AM 1 app_voicemail.c how to enable debugging?
 
Monday December 20 2010
TimeRepliesSubject
10:16PM 0 Re: What's up?
10:02PM 4 cdr_mysql stopped working
9:19PM 0 Upgrading DAHDI and Asterisk
8:39PM 10 Asterisk 1.6 produces *many* zombie processes on Debian.
5:46PM 4 SIP 420
5:35PM 3 Unexpected dialplan match
4:33PM 2 Setting `userfield` from within a callfile
12:50PM 4 dahdi issue on digium AEX800
10:48AM 5 start services automatically
9:41AM 6 Asterisk 1.8 - Dial problem if SIP friend is UNREACHABLE.
9:19AM 0 Asterisk Clustering and DUNDi J Richardson whitepaper
8:42AM 16 DIALSTATUS on CANCEL
7:16AM 0 deadagi on v1.4.xx
1:54AM 1 (no subject)
 
Sunday December 19 2010
TimeRepliesSubject
2:02PM 2 Problem with AASTRA phone of NO SERVICE
11:23AM 1 In which version is eventfilter working?
12:03AM 19 Specifying DID for outbound calls
 
Saturday December 18 2010
TimeRepliesSubject
7:07PM 3 How to install the new cdr-stats?
11:58AM 4 Asterisk and Alcatel digital phone's
 
Friday December 17 2010
TimeRepliesSubject
5:17PM 1 How to block everyone outside of our lan
3:49PM 2 transfer from sip to dahdi, connects caller to MOH stream and not target
3:40PM 11 Wireless Desktop VoIP Phone?
3:08PM 2 Asterisk Freeze In 1.4 realtime
2:45PM 5 Attack problem
11:22AM 2 Contradiction between 2 AMI actions QueueSummary and Queuestatus
11:17AM 1 HA: what is missing to keep ongoing calls during failover ?
10:57AM 4 Voicemail Forwarding
10:21AM 8 Ported Asterisk in Android
9:14AM 3 dahdi show channels / how to get the call duration for active calls?
9:07AM 0 Asterisk and Tandberg VCS
7:14AM 3 Pass DTMF to IVR gateway through SIP phone conferencing.
6:00AM 0 start music on hold coredump
4:33AM 6 How to find , internal, external inbound or outbound
 
Thursday December 16 2010
TimeRepliesSubject
8:16PM 4 PTMP BRI Unable to receive TEI from network in state 2(Assign awaiting TEI) - Asterisk 1.6.2, Latest DAHDI, LibPRI
4:33PM 0 app_voicemail: "3" for advanced options does not have an effect _while_ the vm-message is played
4:17PM 0 Dialplan not found
1:50PM 0 chan_iax2.c handle_call_token: Call rejected, CallToken Support required
10:54AM 17 Call sip:user@domain.com?
9:52AM 0 Junghanns OctoBRI - Invalid sync priority
7:37AM 0 No subject
 
Wednesday December 15 2010
TimeRepliesSubject
11:07PM 2 Echo Cancellation Problem - Invalid Argument?!?
9:27PM 0 Asterisk 1.8.1.1 Now Available
9:23PM 3 Recommendation for a Linux based SCADA
7:53PM 2 res_odbc dependeny issue
5:46PM 1 Asterisk 1.8 with web-meetme crash
2:46PM 5 Which version to use: 1.4 or 1.6 or 1.8
2:33PM 2 Transferring problem within Queues
2:19PM 0 Asterisk Community Mailing Lists Service Disruption
1:38PM 2 Two asterisk servers, two different service providers
 
Tuesday December 14 2010
TimeRepliesSubject
6:56PM 0 503 Server error response for REGISTER request (Asterisk v1.6.0.5)
3:57PM 3 Anyone know how to receive partial (interrupted) faxes with app_fax?
3:56PM 5 Asterisk + VOSP account working configuration?
3:49PM 1 Announce Remaining Call Time
2:19PM 5 Converting asterisk h264 recordings
12:58PM 0 Debug messages.
12:09PM 0 Asterisk dynamic span error
9:20AM 1 Asterisk on smartphones ?
6:58AM 7 Asterisk and Dahdi ON Amazon EC2
 
Monday December 13 2010
TimeRepliesSubject
9:08PM 0 What to check for when there are sound interference using SIP channels only? standard debug methods?
8:49PM 0 asterisk-users Digest, Vol 77, Issue 27
6:28PM 1 Configuring server to call SIP numbers on the Net?
6:00PM 9 Voice mail distribution - missing messages
3:44PM 2 Application to test STUN + broadband?
1:06PM 0 Problems after upgrading libpri from 1.4.11.2 to 1.4.11.5
11:48AM 7 Mail Integration
10:47AM 0 Asterisk for Testing PC-1.5 MTA
8:16AM 2 Asterisk 1.6.2.10 & video
2:04AM 2 1.8.1: playing imaginary sound files
 
Sunday December 12 2010
TimeRepliesSubject
2:24PM 12 Atcom IP-4B ISDN IP PBX?
4:28AM 0 Transfer (sip -> dahdi) results in moh for dahdi
3:26AM 0 pickup problem
 
Saturday December 11 2010
TimeRepliesSubject
1:40PM 1 No more room in scheduler
8:06AM 17 Why does "sip show peers" show my router/gateway address as the client IP address?
 
Friday December 10 2010
TimeRepliesSubject
6:47PM 8 1.6.2.14 > 1.6.2.15: blind transfer works but not Xfer on aastra
5:28PM 0 HP DL360G7 with T1 card(s)
4:45PM 5 UDP buffer overflows?
12:06PM 0 Multiple digits as takecall in followme application
6:32AM 1 Audio ports
 
Thursday December 9 2010
TimeRepliesSubject
9:31PM 7 (Fwd) Re: Configuring Softphone
1:57PM 7 Asterisk SIP attacks and sshguard
5:46AM 1 Load testing SFA
 
Wednesday December 8 2010
TimeRepliesSubject
9:45PM 0 DAHDI channels not hanging up FXO
8:23PM 0 Asterisk 1.8.1 Now Available
8:23PM 0 Asterisk 1.6.2.15 Now Available
8:22PM 0 Asterisk 1.4.38 Now Available
7:59PM 2 filtering AMI Event: RTCPSent
6:56PM 0 Asterisk 1.8 debian packages?
6:48PM 2 [headset/mic] Volume too low + echo in * (Gilles)
4:45PM 14 How to quickly move on to Dahdi channels when SIP provider fails?
3:52PM 1 QUEUE_PRIO
2:31PM 3 Asterisk segfault in dmesg
2:06PM 8 [POTS/BRI] Neutral comparisons of PCI vs. box?
11:36AM 3 Video codecs: H263 & H264
9:43AM 10 Error building network library on OpenSolaris and 1.8.1-rc1
2:31AM 6 Configuring Softphone
12:15AM 2 debug audio or channel
 
Tuesday December 7 2010
TimeRepliesSubject
5:38PM 4 Dahdi issue with Asterisk 1.8.0
3:55PM 2 [headset/mic] Volume too low + echo in *
2:43PM 12 Snom (vs Polycom) - provisioning
12:53PM 6 No MOH with parked call
9:42AM 2 'Bookmarking' a place in a sound file
9:30AM 0 DUNDi and Lua dialplan
2:38AM 1 no audio on end-point when call is connected/bridged via PBX
 
Monday December 6 2010
TimeRepliesSubject
11:48PM 1 Execute DialPlan Context without Answer App
10:49PM 0 Mac OS X 10.4 support
3:41PM 5 [3102] How to rewrite CID name + number?
2:23PM 1 Asterisk 1.6.2.10 video call
1:10PM 0 Fw: Sip Hangup after critical packet SIP DEBUG attached
12:08PM 0 Sip Hangup after critical packet
11:37AM 1 Callee side blind transfer is failing in 1.8
10:45AM 1 Linkedid member in Channel structure on 1.8
12:19AM 3 no audio
 
Sunday December 5 2010
TimeRepliesSubject
6:36PM 2 HA8 cards and RED alarm
 
Saturday December 4 2010
TimeRepliesSubject
6:37PM 0 ISDN TE/NT Mode
3:15PM 2 Error messages with chan_dahdi
1:02AM 3 Polycom Park by EFK
 
Friday December 3 2010
TimeRepliesSubject
8:58PM 4 Issue with MOH - Asterisk 1.4.17
6:36PM 1 What are the minimal permissions required to read the PeerStatus and Registry events?
4:19PM 5 Asterisk 1.6 (Web-meetme)
12:57PM 0 Caller id is not proper when I do call forward
9:32AM 0 OT - Gigaset and PoE
5:34AM 1 Sharing Fail2ban data
12:56AM 5 Asterisk error - 1.6.2 SVN - voicemail files "corrupted"
12:41AM 0 Astmanproxy on FreeBSD 8.1
12:09AM 6 Version compatibility question...
 
Thursday December 2 2010
TimeRepliesSubject
9:32PM 2 MP3s not decoding properly for MusicOnHold.
9:06PM 13 Asterisk ports
8:03PM 13 DAHDI on VMWARE
5:13PM 8 + on Caller-ID
4:58PM 10 alarm POTS lines
2:19PM 14 Push central phone book to phones
8:44AM 1 rotate of logfiles
8:28AM 0 [OT] Any comments on Comcast and Level 3 story this week?
8:02AM 0 Dahdi 2.4.0 and unplugged spans [SOLVED]
 
Wednesday December 1 2010
TimeRepliesSubject
7:05PM 3 codec_g729a implicated in file descriptor buildup
6:33PM 3 Abandon events in cdr
5:44PM 6 Issues with 1.8 and BlindTransfer
5:34PM 0 Problem with Queue_log and CDR.
3:56PM 2 Dahdi 2.4.0 and unplugged spans
3:30PM 0 MixMonitor not recording in version 1.8
12:22PM 1 Reasons of OriginateResponse
12:15PM 3 Dahdi on Realtime.
12:10PM 0 <solved!> Asteris 1.8 and mISDN - 'mISDN' (cause 66 - Channel not implemented)
10:46AM 0 Broadsoft-like BLF List URI ?
2:04AM 2 Trying to configure a SIP software phone
12:55AM 10 Zaptel / Asterisk on Solaris
12:34AM 9 Asterisk with MySQL Cluster