asterisk users - May 2010

Monday May 31 2010
8:08PM 10 Voicemail : mail attachment to multiple mail-addresses
7:17PM 1 Definie gtalk troubles over here
6:59PM 4 Reloading queue members (realtime DB)
3:13PM 0 testing my asterisk with gtalk (and JACK) - please help
1:32PM 4 Read and set the UUI in asterisk
12:46PM 1 Suddenly "HDLC Bad FCS (8)" errors on ISDN-PRI, changed nothing
11:58AM 0 Asterisk 1.4.31 IAX2/RSA BUG
9:33AM 2 Queue ringall problem.
8:43AM 1 Why Manager account log on and log off alternatively all the time?
8:18AM 0 IAX2 Load test
4:31AM 0 Dahdi PRI T1 Setup for TE210P
Sunday May 30 2010
3:19PM 3 Wierd behavior of illegal extension
2:06PM 15 How to use one single IP as origination
1:12AM 1 DID's for Chatham, ON
Saturday May 29 2010
9:02PM 13 Best way to limit outgoing calls per trunk
12:20PM 1 asterisk-users Digest, Vol 70, Issue 63
5:56AM 2 Switchvox vs Asterisk codebase
Friday May 28 2010
9:05PM 12 Normalizing called numbers
6:24PM 3 Asterisk + app_fax + OpenBSD 4.7 minor issue
4:40PM 9 DAHDI Help (made a cardinal sin :()
8:43AM 3 [Dahdi/system.conf] "fxsks = 1" deprecated?
8:15AM 0 Dead air between answer and packet2packet bridge (Bug 12708?)
7:21AM 0 ADA 1.1 features
7:08AM 0 Can't specify connection port with ADA
6:44AM 0 Is AstManProxy still recommended with 1.6 and later ?
5:47AM 1 call-waiting
5:34AM 0 Asterisk-based "Incredible PBX"
12:00AM 3 "pri show version" still shows old version despite doing a make && make clean && make install for v1.4.11
Thursday May 27 2010
11:12PM 0 [X100P+Dahdi 2.3.0] Couple of questions
9:24PM 0 Aastra i740 and Asterisk
9:17PM 2 Pattern matching - how to ignore numbers after 10 digits
7:38PM 8 How to have Asterisk respond from the IP address used for registration
7:06PM 2 OpenVox B200P and D410P under Asterisk 1.6
6:15PM 0 GoogleTalk to Asterisk - choosing voice menu options
5:55PM 0 IAX2 Call Transfer
9:45AM 0 N900 video with Asterisk?
8:05AM 3 Meetmee user introduction disabled
3:12AM 2 BRI card(B800P) doesn's work with DAHDI(wcb4xxp) in NT mode
Wednesday May 26 2010
10:57PM 0 meetme changes between asterisk and
6:35PM 0 Extension state can get stuck in 'Ringing' state
6:34PM 5 Music on Hold
6:34PM 0 call droped if second caller enter meetme conference
5:18PM 4 Getting 'username' of sip peer
4:36PM 7 "ring splash"
4:27PM 1 Libpri 1.4.11 Released
4:20PM 3 VoIP over virtualized VPN
3:24PM 2 Better AMD module
3:17PM 12 [Dahdi] "DAHDI_CHANCONFIG failed on channel 1"?
2:59PM 6 OT: Windows TAPI command-line driver
12:55PM 8 Help with IP Routing
11:06AM 10 Jack in /usr/local/ means failure for asterisk
3:05AM 2 Getting "ghost" transfer or music on hold
2:48AM 2 Error compiling DAHDI...
Tuesday May 25 2010
10:59PM 0 Using Sangoma Call Progress Analysis behind NAT router
10:17PM 0 DAHDI Linux and DAHDI Linux Released
6:07PM 1 How to get ConfBridge user count
4:08PM 0 Converting video files into .h263
3:50PM 0 asterisk-users Digest, Vol 70, Issue 54
3:40PM 0 Is it possible to get video working on h323 calls?
3:14PM 1 SuSE Firewall2 - Port Forward Command
3:09PM 0 need sip reload to reach peers.
1:39PM 1 nortel meridian question
10:48AM 5 Little t38 bug?
5:05AM 0 was missing
Monday May 24 2010
9:00PM 1 mISDN compiling error
8:56PM 4 convert zaptel to dahdi?
5:56PM 2 sip and SSL
4:16PM 0 Agent Privacy - chan_local
3:24PM 5 TTS for asterisk
3:08PM 7 routing of calls
1:51PM 3 State of JACK support i9n Asterisk
1:09PM 8 Delay in IVR
11:31AM 2 [0017330] 1.6.1 and 1.6.2 + MySQL crases on ODBC Query (via func_odbc or sip realtime)
11:05AM 0 zap calls are getting dropped (unexpected disconnect message)
10:29AM 0 About testing Call transfer in asterisk
4:40AM 0 [Asterisk-User] Asterisk Video support
Sunday May 23 2010
8:54AM 5 Dahdi problems with kernel 2.6.32
Saturday May 22 2010
7:28PM 5 US "Truth in caller id act"... and it's impact on services
4:15PM 1 Manual Web-meetme
2:44PM 3 Caller ID questions
7:30AM 1 OT - How to query vcard-like data for CTI app
3:01AM 1 voice recognition suggestion
1:34AM 8 About Sangoma cards and Asterisk integration with other PBX
Friday May 21 2010
7:16PM 0 Question about 1.6: multiple IP on a single Asterisk box / multi ISP routing
5:29PM 0 FollowMe dials numbers but can't confirm the call or hear anything
3:40PM 0 Can't load ooh323 on Centos x86_64: capabilities failure
2:58PM 4 SPA-932 BLF Stays solid red for single extension
2:19PM 2 Connecting 1-2 GSM ports to asterisk?
12:41PM 1 Hanging up call - no reply to our critical packet
9:23AM 0 quality of the call at the hangup
7:48AM 5 Using unix socket to connect with database
7:34AM 4 CANCEL Reason
4:38AM 0 file command with alaw file
Thursday May 20 2010
7:51PM 0 Asterisk and Now Available
7:32PM 0 Asterisk transfer to a conference using feature code?
6:01PM 0 Attended Transfer using AMI
5:35PM 7 Softphones on thin clients...
4:43PM 3 Checking blank CallerID in Dialplan
3:41PM 19 Which issue is keeping you from updrading to 1.6.2 ?
12:58PM 0 Friday @12 Noon and 1PM
12:38PM 1 Asterisk T.38 Gateway code testing
11:47AM 2 run extensions after call moved to queue and answered by member
8:00AM 1 Sending fake auth rejection for user
7:12AM 0 Early injecting Jack between call parties
7:08AM 2 DAHDI and ESXi
Wednesday May 19 2010
8:00PM 5 Cause and cure for "Exceptionally long voice queue length queuing to Local"?
6:28PM 3 Asterisk and RFC 3261
4:42PM 3 DTMF Input from the User
8:37AM 0 Re-invite from Asterisk Server: Port number changes
4:38AM 10 Asterisk Cluster
3:47AM 2 a2billing DID and Queues
Tuesday May 18 2010
7:36PM 1 quick question on conf bridge
6:20PM 0 Location with PRI / Analog lines
5:13PM 2 NPA NXX Database
5:10PM 0 Peering with a Taqua T7000
3:26PM 0 Faxes from website works, but from regular don't: cause 16
2:15PM 3 About option U in Dial Ast version 1.6.2
10:14AM 3 Asterisk 1.4.30 & T38
8:17AM 1 automon filename does not follow the docs.
6:52AM 4 [ASTERISK-USER] Meetme Recording issue, recording is 2 times Faster then normal recording.
6:28AM 16 OT - Which SIP hardphone with 100 BLF ?
1:26AM 2 Callerid with DAHDI
Monday May 17 2010
10:14PM 0 new way to capture audio streams in calls
7:22PM 3 Commands to upgrade to latest Libpri? can I upgrade without touching Asterisk?
7:13PM 1 SIP SRV Registration problem
6:01PM 1 PRI down due to chan_zap.c: No more room in scheduler....Got SABME and Sending Unnumbered Acknowledgement...Any thoughts?
4:31PM 0 180 with SDP
1:01PM 0 ControlPlayback skip forward fails on mp3 file
12:08PM 1 R: new way of asterisk and kamailio(openser) realtime integration
10:28AM 8 Adding a context from the console
10:00AM 5 new way of asterisk and kamailio (openser) realtime integration
9:16AM 0 Hi
8:46AM 8 identify caller hangup or callee hangup?
4:36AM 9 Microsoft Response Point Voip server discontinued
Sunday May 16 2010
7:00PM 1 Digium TE121P + DAHDI
2:15PM 10 OK, I'm stumped
11:16AM 2 play a sound file directly to a caller channel
7:52AM 1 Aastra SIP phone regisration problems
5:22AM 2 Problems with Asterisk and two Linksys SPA941
12:28AM 9 ISDN config: LBO values
Saturday May 15 2010
8:32PM 3 Re-compiling q931.c
8:14PM 1 q931.c modifications for CLID Presentation
2:23AM 0 Problem with Music on hold
Friday May 14 2010
9:09PM 3 SIP realtime problem
4:56PM 5 realtime queues "membername" problem
4:31PM 0 SIP and codec negotiation
11:35AM 3 Agents
8:45AM 3 is my PHPAGI Soap code right?
4:53AM 1 Do you think my server is being attacked?
4:38AM 1 aastra pt 480e phone
4:05AM 0 Channel cannot be released
3:44AM 0 Delay on DTMF with SpeechBackground and Vestec
12:09AM 3 Are there AMI commands to manipulate a voice mailbox?
Thursday May 13 2010
11:24PM 0 Asterisk Call Recording *1 Status Indication
6:41PM 4 What does Asterisk give to reject a re-invite?
4:12PM 0 Sip session timers.
3:19PM 0 Sending SIP credentials in INVITE
3:10PM 0 Skype for Asterisk and instant messages
2:52PM 0 Asterisk Sip Proxies and SIP persistence
2:30PM 0 asterisk-users Digest, Vol 70, Issue 30
2:14PM 5 app_addon_sql_mysql.c:116 find_identifier
1:52PM 0 Asterisk Crashing with ERROR[1906] astobj2.c: refcount -1 on object 0xb1aab758 Ast Ver
8:17AM 12 LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality
3:45AM 2 Error at start of asterisk with cdr_addon_mysql.o
Wednesday May 12 2010
9:03PM 1 problems with unicall
8:28PM 0 Ringback
8:26PM 3 IAX2 - providers discontinuing support
8:04PM 1 Sangoma A101D PRI failing with ERROR - -- Got SABME from network peer. Sending Unnumbered Acknowledgement
6:04PM 1 pattern containing an asterisk
6:03PM 2 include sip configuration from another file in sip.conf
2:59PM 4 "bad magic number" log messages
2:14PM 3 asterisk-users Digest, Vol 70, Issue 25
1:59PM 2 Stress Test new system
1:12PM 4 Asterisk core dumping on SendFax with FFA
12:56PM 1 Help finding online training
11:38AM 0 Additional CDR values
10:36AM 0 Have a macro update a channel variable
10:07AM 0 One way audio problem, a=sendonly and a re-invite
9:51AM 23 SIP trunk between two Asterisk servers
9:29AM 2 Voicemail() app not available?
8:51AM 0 problem of "Cannot release Channel"
8:45AM 0 Could Asterisk PHP agi be a SOAP Client?
8:08AM 6 No ringtone when going from queue to dial-command
12:03AM 0 Customizing Asterisknow distribution
Tuesday May 11 2010
10:10PM 7 iax calls via
7:39PM 5 Digits and Vestec
7:30PM 14 Need fax solution for 1.4.xx
6:25PM 7 Lookup ${EXTEN} in database, update context/route if found... AGI?
5:36PM 0 E1 information
3:32PM 4 Problem with callerid(dnid) and queue
3:23PM 1 asterisk-users Digest, Vol 70, Issue 24
2:48PM 2 conf files vs astdb
2:31PM 0 queue member state in asterisk 1.4
12:43PM 0 AskoziaPBX 2.0 Released
10:51AM 3 bug in asterisk
9:21AM 3 Creating a HTTP Request on missed call?
9:09AM 1 Xorcom Astribank One call and dies
8:57AM 1 asterisk-users Digest, Vol 70, Issue 23
4:05AM 4 AGI and Severe Weather Alerts
2:31AM 1 AsteriskNow
Monday May 10 2010
11:19PM 10 Speech/DTMF mix?
10:30PM 0 Recording with extension and agent in queue
6:32PM 1 Using Asterisk? Get on the press list!
5:54PM 0 Pass MWI from analog line through DAHDI?
5:41PM 1 Manipulating the Blacklist database
5:38PM 1 Working with Blacklist database
3:54PM 2 shoretel pbx
2:35PM 5 DAHDI not detecting hangup
2:35PM 1 More clarification on outbound sip channels.
2:14PM 1 Dialing a SIP Peer without using register strin
12:46PM 0 Sometimes called party answers, but callee keep hear ringing, called party hears nothing!
12:36PM 3 Continue dialplan is source channel hangs up
12:29PM 0 Unable to hear voice when called from PSTN phone line
12:19PM 2 Records sets and ODBC
11:57AM 5 Simulating a commercial SIP provider
2:24AM 0 Problem of hearing transfer' s sound?
2:18AM 0 Problem of hearing attended transfer' s sound
Sunday May 9 2010
1:15PM 2 Re TrixBox
10:14AM 1 B410P and Patton smartnode : any success ?
9:45AM 0 Bug or feature: comments in chan_dahdi.conf.sample
Saturday May 8 2010
8:04PM 4 Conflict between jumper settings and dmesg with OctoBRI
7:41PM 0 OT - Newbie: Do I have a shared IRQ ?
1:41PM 3 JNET's qozap, dahdi and PCI-E Quad
12:05AM 4 text
Friday May 7 2010
11:01PM 1 DAHDI astribank Channel Unavailable
8:52PM 10
7:25PM 0 SIP REGISTER header not containing Allow-Events or Allow
6:17PM 1 Multiple SIP lines.
5:16PM 1 Execute AGI, Then Continue
4:37PM 4 Asterisk Bible?
3:35PM 7 "Contact header appears incorrect on this invite" Asterisk registering with another PBX
12:04PM 0 asterisk and gnokii on same server: scratchy sound
8:20AM 11 Getting presence working in 1.6.2
7:24AM 0 Issues with remote call setup
5:23AM 1 Video in Skype for Asterisk
4:33AM 4 Problem of "Playing 'pbx-transfer'"
Thursday May 6 2010
11:30PM 0 Contact header gets url decoded?
10:46PM 4 T.38 Fax With Flowroute SIP Provider
9:36PM 3 Possible bug in chan_sip:add_sdp
8:11PM 6 Questions About Fax for Asterisk
1:44PM 0 REALTIME in 1.2
1:00PM 2 problem with trustrpid
12:47PM 2 Make the call finish after executing Dial(G())
Wednesday May 5 2010
10:44PM 0 Still true: only first peer matched on incoming call?
9:03PM 0 T38 trunk configuration for relay appears to affect default trunks for voip
6:33PM 1 IAX2 Auto-congesting call due to slow response
6:29PM 2 Channels In Use
5:01PM 4 Registering a Cisco 7965 on 1.4.26
4:10PM 4 VoIP Termination in Japan
4:00PM 5 What is billsec in CDR?
2:17PM 0 res_config_mysql - maximum field length for appdata
1:39PM 17 OT: NAT in SPA922
1:22PM 14 Hash Dial Pattern Problems
12:30PM 4 SIP - SIP over PBX no audio when canreinvite=no
11:51AM 5 CDR to MS-SQL via ODBC issue
11:46AM 1 Confirm answering a call
6:35AM 2 Getting calee audio in Asterisk (real time)
6:19AM 0 Forwarding inbound mobiles
4:58AM 2 Transfer calls using ##
3:29AM 0 Problems with Asterisk working in Realtime with PostgreSQL
Tuesday May 4 2010
9:42PM 2 Productivity Suite on Polycom IP7000
7:54PM 2 problem with ringinuse=no, queue members receive randomly two calls
5:59PM 7 Asterisk Now Available
5:58PM 0 Asterisk Now Available
5:57PM 0 Asterisk Now Available
5:57PM 0 Asterisk 1.4.31 Now Available
5:56PM 0 Bridging old system (ESI IVX E) with new Asterisk server - it is robbery!
4:46PM 4 client-server encryption
4:23PM 0 queue members
1:41PM 0 Problem with AMI Originate
8:25AM 0 Check if extension loaded over AMI
8:15AM 0 DSCP QoS value in YeaLink phone settings
4:18AM 6 Interesting email project.
2:44AM 3 Channel failover
Monday May 3 2010
8:18PM 1 Run a script after Page application
4:59PM 6 sending T.38 fax negotiation problem
2:30PM 5 Bridging old system (ESI IVX E) with new Asterisk server
1:41PM 5 Reading the CDR
1:31PM 0 CallerID problem with astribank
1:22PM 0 Spy on Asterisk 1.2
11:15AM 4 RTP ports
10:04AM 0 Hangup Detection
9:00AM 0 Parking problem with outgoing calls
4:47AM 2 Calling a RESTful Web service from Dialplan????
Sunday May 2 2010
7:41PM 7 working example of t38 fax w/ 1.6.2?