asterisk users - Apr 2010

Friday April 30 2010
5:23PM 1 Embedded IAX
3:18PM 2 B400P card crashes conncection
2:02PM 1 Fwd: Re: SpiderMux?
1:16PM 0 get hold event
12:26PM 1 GXW4024
11:20AM 1 HDLC Receiver overrun on Wildcard TE410P
10:51AM 0 IAX trunks and audio codecs
9:28AM 0 Problems with t38modem and bitrate sent to t38-termination service
8:57AM 1 Call-Waiting, implementation ideas
8:01AM 5 Asterisk and Patton
5:30AM 0 Caller ID on Asterisk and Astribank
5:10AM 0 Friday 12 Noon EDT: Media5fone Mobile SIP Client Symbian S60 & iPhone
3:55AM 2 Continuing after a TIMEOUT(absolute)
Thursday April 29 2010
8:33PM 3 Calls Dropping
7:59PM 2 Asterisk stopping for no reason
7:36PM 4 ATA shootout: PAP2T versus Grandstream Handytone 286
7:02PM 2 Code in extensions.conf to leave a voice mail in another PBX ?!
5:45PM 0 Odd Issue With Polycom Phones]
3:25PM 1 Strange Invite issue
1:51PM 1 Dropping incompatible voice frame
12:56PM 1 incoming call should ring on several dahdi channels
12:06PM 0 Polycom 330 not connecting
10:01AM 1 Starting call recording using a dynamic feature to call a macro
9:53AM 0 mysql realtime schema
9:02AM 8 AGI <==> DeadAGI
8:47AM 1 Asterisk Query
7:20AM 1 SpiderMux?
2:35AM 1 Issue with (pattern) matching extension
2:12AM 2 No change in payload. (SDP)
1:57AM 1 Duplicated DTMF with bridged IAX channels maybe?
Wednesday April 28 2010
7:29PM 1 Strange Error -- ASterisk 1.6
5:20PM 0 sip jitter buffer
5:14PM 1 simple dialplan question
3:59PM 0 asterisk core dumps after cdr database writes using odbc
2:37PM 2 Gateway E1 <=> Asterisk ?
1:44PM 0 command-line dialplan "compiler"
1:10PM 0 Execute Macro when queue is answered
12:49PM 2 dialplan
8:50AM 6 Asterisk 1.4.30 is slow sending STDIN to AGI script
5:12AM 2 BN8S0, dahdi, wcb4xxp
1:48AM 6 Dial plan question.
12:15AM 2 Broadvoice inbound fails on Asterisk 1.6.1
Tuesday April 27 2010
4:06PM 0 callprogress issue
2:53PM 2 Record call without caller interference
2:41PM 5 E3 Card on Asterisk ?
1:54PM 2 Problems for Skype for Asterisk
12:27PM 2 Connect 2 asterisks servers
11:31AM 4 dialplan question
10:00AM 0 Message notification without MWI
2:12AM 0 Redone setup, bizare problems
Monday April 26 2010
9:40PM 1 Building Asterisk-RPM for
8:23PM 3 Inbound route question
8:11PM 0 DTMF from SIP phone to FXS/FXO
6:21PM 1 Dahdi will not compile on Unbuntu Studio Linux 9.10 (Karmic) 32bit
5:32PM 0 Taqua users out there?
4:45PM 0 problem of registration with Asterisk using exosip2
4:35PM 0 SIP authentication
3:51PM 0 How to disable dialog-info based call pickups (Was: Re: 1.6.2 - Pickup and SIP Replaces header)
3:16PM 1 1.6.2 - Pickup and SIP Replaces header
2:37PM 0 misdn accountcode?
2:21PM 2 [PATCH] Make Queue announcements more consistent (
12:22PM 1 : app_voicemail is calling sendmail without any argument
10:21AM 1 play a sound from the callee before putting it in connection.
Sunday April 25 2010
7:19PM 0 CONNECTEDLINE(), progressinband=no and 183 before 180 (with latest trunk)
6:24PM 0 SIP gain
6:21PM 2 hardware clock drift and CDR
5:37PM 1 DAHDI Congestion cause 34
3:59PM 1 How to debug the problem of Asterisk using so much of CPU percentage...?
2:15PM 0 Asterisk 1.6, dialplans, and IVR
2:06AM 3 VoIP monitoring tools
1:23AM 1 VOIP Monitoring tools........
Saturday April 24 2010
8:51PM 0 PrivacyManager
6:19PM 2 Asterisk and Archlinux
9:30AM 2 Manager events & safety
3:19AM 0 automatic call with call files
2:59AM 0 Multiple Parking Lots in Asterisk 1.6.x
1:36AM 2 Asterisk not recognizing ACK from an OK message? Help debuging SIP retransmit problem
Friday April 23 2010
9:52PM 1 What is needed to test issue 0013573: [Patch] Allow realtime_multi_ldap to behave like other realtime_multi functions
7:21PM 6 RTP over TCP
5:11PM 3 Playback all the sound files
2:25PM 0 VUC Friday: Bill Miller, former VP of Product Management
2:36AM 0 Hans Rauser
1:55AM 1 asterisk running @ 100% load doing nothing
Thursday April 22 2010
9:02PM 4 More efficient dial plan for a list of selective inbound numbers
7:18PM 2 Swaping out phones.
6:37PM 0 Avaya UUI
5:53PM 1 Hangup after n seconds using originate ?
5:31PM 2 Follow-me to my answering machine :-(
4:14PM 3 How to do analog e&m on asterisk?
8:41AM 0 DAHDI User-User information "Message longer than it should be??"
7:24AM 1 Need to patch Asterisk for problem with FreePBX Call Confirmation
1:25AM 2 Security tests
12:40AM 5 ${HANGUPCAUSE} is always 0 in the h extension
Wednesday April 21 2010
10:29PM 1 Time difference in CSV CDR's and MySQL CDR's
5:25PM 3 Asterisk choking on voice messages announcements
5:11PM 1 Improving audio bitrate for all callers in a conference room for a podcast
11:02AM 1 Why there is no in asterisk16-
1:22AM 2 Unable to load
Tuesday April 20 2010
9:57PM 0 IBM X3650 with Asterisk???
9:31PM 1 Manipulating audio in asterisk
6:22PM 0 SIP one-way audio
6:19PM 0 Initial audio dropping
5:11PM 1 Improving CLI Help - was [Re: 1.6.2 No "soft hangup"?]
5:02PM 4 How to record a call in a single file when transfered...
4:18PM 0 I figured it out!!
4:05PM 2 1.6.2 No "soft hangup"?
2:59PM 0 Feature Request - SoftHangup with delayed playback option
2:46PM 0 How to tell if a channel is on hold or not from diaplan ?
1:34PM 4 Voice mail "maxmessage " setting per mail box
1:11PM 2 Read Timeout
9:49AM 1 Portech MV-374 does not register
8:55AM 0 Dozens of SIP NOTIFY messages with unique call ID's, and the same mailbox repeated multiple times on
8:12AM 0 Yesterday EC2, today Netnation Europe V.O.F.
6:48AM 1 Put a call on hold with Manager
1:07AM 6 Calls drop after 20 seconds
Monday April 19 2010
10:30PM 0 new in Bridge(), How does it work?
6:52PM 1 Zap PRI failed with Cause 34 - Where to check for problems?
5:46PM 3 A matter of context
5:13PM 0 Xorcom Experience
4:00PM 1 Help with FastAGI server in Windows
3:49PM 0 Hangup after 1 second of ringing ?
3:47PM 0 RTP Timeouts not clearing calls
3:41PM 1 zapg723toslin did not update samples
3:14PM 1 G729 exhaustion conditions
1:06PM 5 Evaluating Asterisk
9:22AM 3 Detect if a Number is up or not
8:48AM 1 B400P and A1200P changes card order
6:14AM 2 OpenSIPS with Asterisk Backend
2:38AM 3 Extensions Reload | Asterisk Freezes ? 1.4
Sunday April 18 2010
10:44PM 2 kamailio
5:59PM 0 The best way to stop an ongoing call
5:53PM 1 meetme / upgrade to
5:43PM 1 problems originating an outgoing IAX2 call
12:25PM 0 Asterisk-stat - Bugs
12:18PM 1 Slightly OT: OMA DM Solution
10:10AM 2 Amazon EC2 SIP floods - you can help
9:10AM 1 Bug or feature: cdr_odbc.conf.sample
5:18AM 0 VOIP at BerkeleyTIP-Global meeting on Sunday April 18 12N-3P, & April 27
Saturday April 17 2010
9:20PM 0 B410P and DTMF
9:14PM 2 Changing storm-prevention behaviour in logger.conf
4:31PM 1 QUICKLY - What is the command to change .wmv to Asterisk compatible format using ffmpeg? or any other tool?
3:25PM 1 X-lite direct sip call - Is it possible?
1:38PM 1 DIALSTATUS variable and qualify=no
9:25AM 1 Realtime changes not reflected realtime
Friday April 16 2010
9:39PM 2 Testing a sip call through Asterisk?
8:59PM 7 AGI, FASTAGI or Windows Voice Server
4:10PM 1 On CLI SIP don't appear
4:02PM 1 incoming ghost call
2:00PM 2 Recording music in Queue
1:19PM 1 IAX connection slow between 1.4 and 1.6 dists
10:15AM 2 SS7 over an FXO interface
10:09AM 0 Friday April 16 @12 Noon EDT - Tim's Excellent Island Telephony Adventure, AstriEurop, and more EC2 rant
8:12AM 0 Spam and that recent 'attack' ...
5:16AM 2 How to set up Fax on Asterisk - Using analog Fax machines and HT502 (or FXS of a Digium TDM410P)
5:04AM 3 Delay the HungUp
Thursday April 15 2010
11:45PM 0 moh files not playing in sort order
7:25PM 0 Asterisk-Addons 1.4.11,,, and Now Available
5:11PM 1 SIP devide call-forward behaviour and CDRs
3:45PM 1 Avaya 9640 Convert to SIP (slightly off topic)
2:11PM 1 'o' option on Dial application
1:09PM 1 Asterisk/Polycom Dialed Party Name
12:59PM 1 Transfer_CONTEXT behaviour
11:18AM 1 shared lines (sla) with Asterisk 1.4.26, any hints?
8:31AM 0 Set CDR amaflags not work
7:31AM 0 say.conf implementation of Indian Languages to play numbers and dates
1:55AM 1 How can I record the conversations in a conference call?
12:36AM 0 Regarding remote registration of SIP user on zoiper
12:26AM 1 Queue call to specific queuemember
Wednesday April 14 2010
8:55PM 2 [Conference] Audio/Video
8:50PM 1 Interpbx connection
8:05PM 0 Vestec vs Lumenvox
5:13PM 0 Sending SMS problems.
5:04PM 0 two FreePBX servers with load balancing
4:37PM 4 FastAGiin Windows Server
3:44PM 2 Conference Meetme
2:52PM 3 Converting GSM calls to SIP
1:12PM 0 can't upgrade from
6:47AM 0 Shorewall rate limiting rules?
1:06AM 1 Ring Two Extensions Simultaneously with different caller ID values?
Tuesday April 13 2010
10:26PM 1 Interesting One Way Audio
10:25PM 2 1.6.0 verses 1.6.2
9:02PM 0 What's are the possible return values of AMI Originate when Async is set to 0?
8:20PM 0 SIP registration failure stops all SIP activity
7:51PM 1 DAHDI-Linux and DAHDI-Tools 2.3.0 Released
7:19PM 2 Possible AGI bug?
6:51PM 1 Do AMI Events have timestamps?
6:17PM 2 iptables miss up phone calls if not used properly
6:07PM 2 Time variables in system application
4:56PM 1 Merge .csv files
4:48PM 1 Using chan_lcr (and mISDN v2) ?
4:46PM 2 Merge master.csv files
4:14PM 0 Problem with Callfiles
2:37PM 1 SIP equivalent of zap "c" option
2:22PM 3 Is down ?
1:57PM 1 Is restart of span a concern on PRI?
1:50PM 2 Full transfer details on inbound calls
1:50PM 0 [asterisk users] asterisk realtime - database driven dialplan
12:53PM 2 cat /proc/zaptel/*
12:19PM 1 dahdi_scan and OctoBRI. Bug or feature ?
11:53AM 0 Dial an extension with follow me
10:46AM 3 protocol used to connect Asterisk and GSM core network (MSC)
9:42AM 1 problem of "when memory become 50% or more then sound become noisy?"
9:11AM 2 SNOM M9 base station A to base station B
5:54AM 2 All incoming calls landing in [customers] context
4:33AM 0 ATA status intermittent
2:49AM 0 PRI Gurus ONLY - Too complex of an issue - SOLVED
1:14AM 1 PRI TBCT - Practical Experience, Anybody?
Monday April 12 2010
9:58PM 2 Being attacked by an Amazon EC2
8:50PM 1 Flood of REGISTERs - attack?
7:22PM 2 PRI Gurus ONLY - Too complex of an issue
6:40PM 1 Monitoring calls via sound card
6:17PM 1 cause 66 - Channel not implemented
6:02PM 0 Subscribe to a MWI when acting as a SIP client?
5:19PM 2 Asterisk room monitor
4:16PM 2 Dahdi, junghanns and qozap
3:50PM 1 Change in menuselect handling of sound files (in 1.6.1.X)
9:55AM 0 Outgoing routes with two PRI
Sunday April 11 2010
8:57PM 2 mISDN installation via yum
8:55PM 0 Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCode
5:45PM 1 Asterisk in Debian/Lenny without support?
3:24AM 1 over running my did's
Saturday April 10 2010
10:54PM 1 Remote registering fails
10:24PM 0 How Cisco ATA 186 through SCCP with skinny.conf ?!
9:34PM 10 Being attacked by an Amazon EC2 ...
8:14PM 1 Asterisk script to repeat dial of a number
7:46PM 2 PRI - Native ZAP bridge fails - Is this my patch?
4:50PM 1 Asterisk + DRBD Performance
3:50PM 2 Sending RTP media to a different server than SIP Signaling
1:35PM 1 Repeated: Got SIP response 489 "Bad event" back from
10:32AM 2 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED
12:17AM 1 Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCodes
Friday April 9 2010
11:52PM 3 Problems with Fax over TDM410P
9:40PM 2 res fax help
7:27PM 1 Callerid over IAX Trunks
7:26PM 2 Asterisk & Timezones
12:34PM 5 run script after completed
11:10AM 3 scratchy sound
8:03AM 0 Friday Apr 9th 2010 @ 12 Noon EDT: More Cloud Telephony
6:21AM 0 SNOM M9 : expand range
2:33AM 1 asterisk-users Digest, Vol 69, Issue 16
1:54AM 2 tones detection
Thursday April 8 2010
11:48PM 2 IVR menu sound processing for AMR and GSM + live test available
8:30PM 3 long return times from System() calls with
3:28PM 3 jitterbuffer
2:09PM 1 Linksys/Sipura SPA-3201 FXO/FSA with Asterisk
1:38PM 0 MeetMe Options with S(10)L(100)
12:37PM 0 realtime jitter/latency measurements
11:25AM 1 Asterisk died after 80 days uptime
11:00AM 3 dial extension and play sound file from shell on asterisk server?
9:28AM 0 OT - S450ip and R-key transfer
8:28AM 0 Opportunity to author Asterisk books- Packt Publishing.
6:35AM 2 Split E1 ISDN service for another device.
2:44AM 2 Need help with a pika warp asterisk appliance problem.
Wednesday April 7 2010
9:23PM 1 Rebooting Polycom's - Could not create address for 'XXXX'
6:30PM 3 URGENT - How to exclude one ZAP channel for outgoin and incoming calls
5:07PM 1 Agent Callback methods?
3:26PM 3 D-Channel Span Up without Down
3:07PM 3 PSTN issues
1:30PM 2 AGI + Dial + stream file ?
Tuesday April 6 2010
8:50PM 0 Busy(20) returns non-zero and exits immediately on IAX channel
8:34PM 0 dialplan checker
3:56PM 1 testexpr2
2:02PM 1 IAX Problem
1:07PM 2 Limit Number Of Simultaneous Outbound SIP Calls
12:37PM 1 SIP Dialplan Failover Solution
12:02PM 2 polarity reverse
8:26AM 2 Anyone coming to Paris next week for AstriEurope?
8:16AM 1 Which rule for Asterisk to Asterisk-addons compatibility ?
4:36AM 2 Cache sound files for faster processing
2:37AM 1 OT: Wireless headset / phone combination
Monday April 5 2010
10:07PM 1 Debug help
9:51PM 2 Access denied for user 'a2billinguser
8:54PM 1 Please sign Petition - Stop Child Labour
7:30PM 0 SIP Outdial Not Detecting Ringing Line
6:55PM 2 spool directories and filename
5:45PM 1 External Extension with Extension
4:14PM 2 call files in 1.6
2:28PM 5 Continuous bothering message -- Remote UNIX connection disconnected
1:21PM 1 trying app_fax.c
Sunday April 4 2010
12:06PM 1 [OT] phpagi help
Saturday April 3 2010
12:14PM 2 asterisk 1.6.1/1.6.2 binary packages
Friday April 2 2010
3:53PM 0 change IP address of AA 50?
3:41PM 2 asterisk start with php
2:11PM 2 How set debug file for RxFax application
1:50PM 1 Asterisk and spandsp fax problem
1:02PM 3 Asterisk send calls to SIP Trunks with Round Robin Call Distribution
10:23AM 1 RTCP How to stop
7:26AM 1 Gosub replacement within AEL2 dialplans
6:50AM 1 Strange Centos Problem with Dahdi installation
5:20AM 0 there have any one run asterisk on ubuntu enterprise cloud ?
Thursday April 1 2010
10:50PM 1 SIP Connection Question
9:05PM 2 canary_thread
9:01PM 0 Question about MaxRetries in the Asterisk Outgoing folder
7:18PM 1 asterisk-gplonly dependency in asterisk-addons RPM
6:36PM 2 problem compiling asterisk with cdr_odbc
5:50PM 3 Exceptionally long voice queue length errors...
2:36PM 3 RPID on called party
1:15PM 2 Problem with Sangoma A104 and euroisdn pri
12:28PM 1 Asterisk Load Balancing with Redfone/TDMoE driver
9:46AM 0 OfficeSIP Communications Makes Its VoIP SIP Products Open Source
5:53AM 7 Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8