asterisk users - Apr 2012

Monday April 30 2012
TimeRepliesSubject
6:26PM 0 chan_mobile with Nokia 6021 - incoming SMS causes call to drop
9:10AM 2 Cannot send mail from System command
9:09AM 3 Calendar Integration Problem
 
Saturday April 28 2012
TimeRepliesSubject
6:46PM 1 Music as ringtone
 
Friday April 27 2012
TimeRepliesSubject
8:16PM 0 medooze MCU versus confbridge
7:28PM 1 Master Registrations?
7:05PM 0 Scheduled Maintenance for Asterisk Project community services
6:45PM 0 Asterisk as sip client Unable to create channel of type 'Console' (cause 0 - Unknown)
4:39PM 5 Flashphoner
2:01PM 6 Playing Multiple File ( simultaneously ) on Channel
1:20PM 1 ReceiveFax and multiple pages
8:28AM 1 Fwd: Flashphoner
7:42AM 2 No UDPTL ports remaining
 
Thursday April 26 2012
TimeRepliesSubject
11:45PM 2 POTS(FXO) line getting Red alarm after first ring(5 or 6 seconds)
8:57PM 0 Peer SIP authentication with Taqua switch
4:54PM 10 Asterisk + Phones behind different Nat Firewalls
3:37PM 0 OpenVPN design w/ Yealink
2:29PM 1 Call recovery feature
9:41AM 5 Mac OS X sip client with Video support
 
Wednesday April 25 2012
TimeRepliesSubject
9:49PM 0 Restart single dahdi span
6:29PM 0 Open source replacement for AudioCodes nCite 1000 SBC
12:21PM 4 CONNECTEDLINE() updated during SIP events?
12:08PM 5 Hangup Cause and SIP Response Code
7:35AM 4 chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 returned -1: Operation not permitted ??
 
Tuesday April 24 2012
TimeRepliesSubject
8:57PM 5 Looking for IAX trunk/DID to replace Junction Networks
3:39PM 6 Asterisk - Nortel transfer problem
2:06PM 6 Strange problem on ougoing call
1:32PM 0 Digium D40 Direction map 'X' key not functioning
11:06AM 0 Nicaragua PSTN Frequency Parameters
10:59AM 9 Set SIP peer state busy
7:34AM 0 Asterisk don't use "context="
7:27AM 3 Question for a Jira bug marshal
 
Monday April 23 2012
TimeRepliesSubject
8:35PM 0 Problem with blank/empty voicemails
8:32PM 0 Grandstream 1.0.3.30 BETA Firmware
6:25PM 0 AST-2012-006: Remote Crash Vulnerability in SIP Channel Driver
6:25PM 0 AST-2012-005: Heap Buffer Overflow in Skinny Channel Driver
6:25PM 0 AST-2012-004: Asterisk Manager User Unauthorized Shell Access
1:10PM 0 Asterisk 1.6.2.24, 1.8.11.1, 10.3.1 Now Available (Security Release)
12:59PM 8 HELP!! Caller ID "unknown" for all inbound call (Satria Anamarta)
 
Sunday April 22 2012
TimeRepliesSubject
6:07PM 1 meetme identify user number
7:49AM 7 HELP!! Caller ID "unknown" for all inbound call
 
Saturday April 21 2012
TimeRepliesSubject
4:51PM 1 DAHDI 2.6.1 - What does " Build OSLEC EC if in the tree" feature means ?
1:41PM 1 Open source speech recognition engine?
6:19AM 8 No extension found ?
 
Friday April 20 2012
TimeRepliesSubject
6:05PM 3 DAHDI-Linux 2.6.1, 2.5.1 and DAHDI-Tools 2.6.1, 2.5.1 Now Available
5:20PM 5 Advice on Asterisk Conference
3:30PM 1 E & M signalling and Dahdi
2:11PM 1 Experience with virtual servers?
 
Thursday April 19 2012
TimeRepliesSubject
10:59PM 1 asterisk 1.4.39 and dahdi 2.6 on Ubuntu
9:29PM 0 Auto answer Asterisk ; Unable to create channel of type
8:11PM 8 Company info
6:19PM 1 dahdi cannot make simaltaneous calls
11:20AM 3 Call recording and transfer issue (asterisk 1.8)
10:17AM 0 how do I catch the audio stream in real-time for another application?
5:34AM 0 upgrading from asterisk 1.4 to 1.6
 
Wednesday April 18 2012
TimeRepliesSubject
3:42PM 6 Delete "Session timer" ?
1:46PM 0 Far end nat traversal not working
1:12PM 11 asterisk 1.4.39 and dahdi 2.6
10:57AM 7 FXO -> GSM Gateway Problem
8:02AM 11 g729 freezes 1.8
6:27AM 3 hints and server-side DND (do not disturb)
 
Tuesday April 17 2012
TimeRepliesSubject
10:04PM 1 Asterisk 1.8.10 getaddrinfo
3:18PM 10 Account code script needed.
1:10PM 1 Process a variable in a string.
12:38PM 5 Incoming SIP call is rejected always.
10:28AM 1 ExtensionStatus event
 
Monday April 16 2012
TimeRepliesSubject
2:26PM 2 10.3 : sip loses registration ?
1:37PM 0 Dial Local doesn't honore the channel language setting
1:36PM 7 Custom Application recording problem
12:53PM 0 When CALL-ID were same , I could hijack another session
12:20PM 0 Far end nat traversal for media is not working always
12:12PM 5 Invite + decreasing sequence number => 500 Error?
8:56AM 1 SNOM phones? Please test this patch (broken hints with notifycid=yes)
6:06AM 1 OT - Which Sugarcm 6.2 plugin to add click-to-dial (with asterisk) ?
1:11AM 2 Caller id issues
 
Sunday April 15 2012
TimeRepliesSubject
8:48PM 1 asterisk 1.4.39 and dahdi 2.6: chan_dahdi? dahdi?
8:34PM 0 cdr documentation - new fields
7:30PM 0 Call status register
3:03PM 5 Caller ID problem
1:48PM 2 Disable services on startup
11:15AM 3 Transcoding degradation G711<->iLBC
9:49AM 0 Can't make Asterisk send authentication to remote peer on INVITE [SOLVED]
8:44AM 13 Set variables from one asterisk ta a second.
 
Saturday April 14 2012
TimeRepliesSubject
11:09PM 9 Unable to create channel of type 'IAX2' (cause 20 - Unknown)
8:36PM 1 dahdi versions before 2.5 compilation error and ubuntu
3:19PM 1 Dahdi QSIG with Tadiran Coral - not working
9:30AM 2 Can't make Asterisk send authentication to remote peer on INVITE
2:08AM 5 deleting: res_ldap.conf
12:33AM 2 BUSY vs. CONGESTION
 
Friday April 13 2012
TimeRepliesSubject
6:56PM 1 Question about asterisk to Cisco
4:53PM 8 priorityjumping - asterisk 1.8
2:43PM 0 Release Announcement: Adhearsion 2.0 for Asterisk 1.8+
8:36AM 1 Personal queue with one agent: add calls to extension
7:25AM 1 T.30 Fax and Echo Cancelation
7:07AM 1 Simple Gateway to setup calls bewteen two or 3 asterisk systems .
 
Thursday April 12 2012
TimeRepliesSubject
8:36PM 6 GXP1400
8:08PM 0 Ringing detection ?
7:09PM 2 Asterisk 10 & app_swift problem
6:04PM 4 Asterisk 1.8.12.0-rc1
4:31PM 1 DAHDI FXO Call Issues / Indication Types
9:19AM 0 T38 gateway issue
9:07AM 1 chan_sip.so module not loading
 
Wednesday April 11 2012
TimeRepliesSubject
10:25PM 2 DAHDI inter-digit timeout = 0
11:47AM 0 red5sip SIP ua can't register
11:39AM 7 Dahdi-2.4.0+2.4.0 means ??
11:01AM 0 AMI Originate double call
6:40AM 0 OT - How to localize Freepbx 2.10 or 2.9 ?
 
Tuesday April 10 2012
TimeRepliesSubject
8:15PM 11 Run AGI while agent ringing instead of only when connected
4:07PM 0 Recent FreePBX vulnerability attacks
2:05PM 3 MessageSend, SIP, and call files
12:14PM 0 Strange Asterisk port behavior
11:11AM 0 How to disable CDR adaptative logging on asterisk 1.8 ?
7:19AM 0 Experiences with Polycom-Kirk 6000 and DECT/GAP handsets
12:34AM 7 MYSQL INSERT QUESTION IN DIALPLAN
 
Monday April 9 2012
TimeRepliesSubject
10:21PM 2 syntax error from digium fax manual ??
2:44PM 2 Combining multiple SIP providers
2:17PM 1 VMWI DAHDI
2:09PM 0 ChannelRedirect with callee channel
12:08PM 2 Call Deflection with DAHDISendCallreroutingFacility
11:42AM 6 Monitoring voice-quality with sip/rtp/rtcp
8:22AM 6 Video Conference in Asterisk1.4 (using asterisk gui)
 
Sunday April 8 2012
TimeRepliesSubject
1:44AM 1 another non-root problem: unable to set utime ??
 
Saturday April 7 2012
TimeRepliesSubject
8:09PM 4 Unable to access the running directory (Permission denied).
2:09PM 0 Cannot resume call on hold
1:25PM 0 ISDN incoming call disconnected after picking up phone
5:51AM 2 Google TTS - Asterisk
12:05AM 3 Which file is loading these lines?
 
Friday April 6 2012
TimeRepliesSubject
3:30PM 4 Call Transfer not working
 
Thursday April 5 2012
TimeRepliesSubject
5:35PM 20 Dial Plan - Routing via Caller ID
11:58AM 1 Does Cisco 79XX with SIP firmware support asterisk's BLF ?
8:21AM 4 sip pregi net account registration
 
Wednesday April 4 2012
TimeRepliesSubject
11:41PM 0 FollowMe and billsec field of the CDRs
8:50PM 2 Change extension for international ?
8:48PM 6 issue with Digium TDM410P
1:35PM 4 Asterisk 1.8 and DeadAGI
9:31AM 3 cross ivr is comming in my ivr system
 
Tuesday April 3 2012
TimeRepliesSubject
9:05AM 10 process_sdp: Multiple audio streams are not supported
8:31AM 1 Max number of PCIe cards
12:05AM 5 extending fallback numbers
 
Monday April 2 2012
TimeRepliesSubject
2:46PM 4 Limit Call ?
12:44PM 5 Asterisk ACL
10:18AM 1 fax tone testing
6:53AM 0 dial rule problems( on e1 interface) after upgrading 1.8
 
Sunday April 1 2012
TimeRepliesSubject
4:42PM 1 404 Response to Invite - Should be 401
3:22PM 0 10.3.0: gtalk_request: No XMPP client to talk to, us (partial JID)
1:04PM 4 Telephony Card: GSM slots + Analoge