asterisk users - Jan 2011

Monday January 31 2011
11:48PM 0 [OT] Streaming video on variable bandwidth connection?
10:22PM 1 Newbie Question...
9:12PM 1 Calling Directory app from AGI
5:51PM 2 save the calls with asterisk
4:49PM 0 Fwd: regarding error in SIPp
12:26PM 0 Issue with Asterisk not hanging up second leg when first leg hangs up
4:29AM 0 Regarding error in asterisk or SIPp
2:45AM 1 Error compiling Dahdi: invalid use of undefined type struct module
2:31AM 0 Losing registration - ast 1.4.39 and innomedia 6328-2Re
Sunday January 30 2011
9:21AM 3 faxter
5:45AM 0 Senddtmf inside a macro
Saturday January 29 2011
6:30PM 1 Determine When Call Is Picked Up In Queue
10:52AM 3 Reducing number of Asterisk processes?
2:59AM 1 console debugging
12:27AM 1 Can a duration limit be specified in spool call file?
Friday January 28 2011
6:43PM 0 asterisk-users Digest, Vol 78, Issue 66
6:25PM 0 Queue_log with Splunk
6:12PM 0 Asterisk Scenary
5:42PM 0 Asterisk 1.8.2 - TLS, user certificate
5:22PM 2 How to disable srtp in asterisk
12:37PM 3 Disabling Music On Hold
11:34AM 1 CDR issue - Problem logging CDR(userfield) in Master.csv
10:27AM 2 How to update sound files?
7:27AM 1 SendFAX dialplan example
1:05AM 1 RTP keepalive doesn't work
Thursday January 27 2011
10:52PM 1 chan_sip bug? (Asterisk 1.4)
9:22PM 3 A1200P comments?
8:01PM 0 OT: VoIP Users Conf Feb 4 with LifeSize
4:45PM 2 Queue - agent auto-answer
4:41PM 1 Multi-Tenant
4:00PM 1 Anybody ever see this before?
12:48PM 1 Callback when available
9:07AM 0 Bufferbloat! Friday on VUC @ 12 Noon EST
1:44AM 0 Asterisk 1.8 and Cisco 7920
Wednesday January 26 2011
11:29PM 0 SIP channel status - Why is it different when calling an internal extension rather than an outside line over SIP?
8:50PM 0 list of errorswhile registering client at asterisk with sipp
8:18PM 6 Asterisk Now Available
6:07PM 0 Really wacky problem with internal extensions.
4:28PM 5 Regarding error in Asterisk dail plan:
3:21PM 3 Pickup local/.... not working
2:26PM 1 Caching CALLERID(dnid)
1:52PM 9 Recommended Windows client to display CID?
11:29AM 0 Variable HANGUPCAUSE always empty with DAHDI
9:02AM 1 Return variables from func_odbc calls?
8:36AM 1 feedback mechanism
Tuesday January 25 2011
8:32PM 1 Help determining SpanDSP version
7:38PM 2 regarding quit, exit and stop now in asterisk
6:13PM 0 Problem registering two (and more) sip trunks
1:15PM 0 SIP RTP streams
7:31AM 0 Asterisk and Kamailio integration on cloud EC2 amazon no voice.
6:44AM 1 SIP, IAX2 and ISDN ISUP data
1:40AM 0 DNS A queries
12:31AM 2 Unknow "T" callerid
12:28AM 1 Lots of warnings: SUBSCRIBE failure: no Accept header: pvt
Monday January 24 2011
11:46PM 1 extconfig, realtime, and SIP
8:37PM 1 U-verse DTMF tuning for Zaptel
7:53PM 6 ReceiveFAX issue.
3:43PM 6 Unable to insert cdr-data into mysql-DB
3:39PM 0 Voicemail hangs up
3:01AM 3 Asterisk on Debian Lenny with timerfd
2:01AM 2 Outgoing FXO calls have no audio with callprogress=no
1:12AM 1 B410P: DAHDI BRI PTMP HDLC Abort (6) on Primary D-channel of span 1 (TEI Errors)
Sunday January 23 2011
7:56PM 1 Info on using LDAP with Asterisk?
7:41PM 2 Dialplan to bridge 2 legs?
7:04PM 6 end a call after a specific time period
10:10AM 1 RTCP packets when on hold
Saturday January 22 2011
6:16PM 0 Returned mail: see transcript for details
2:20PM 0 Hint for queues
12:49PM 4 Crossover cable for E1 ?
5:00AM 2 spandsp download
Friday January 21 2011
9:48PM 0 waitforsilence changed after upgrade to 1.6
7:03PM 0 Channel in an unkown state
6:53PM 1 Phone multi-registration
5:31PM 0 Queues with ringinuse=yes
5:22PM 0 Force Dahdi modules to load
1:52PM 2 MOH and parking
1:49PM 1 Inbound routes
12:11PM 4 Does Asterisk support NI-1 (DMS 100) and NI-2 for T1s?
10:54AM 1 Where are stored the CDR's?
10:24AM 1 Unable to receive calls (inbound)
8:42AM 0 Polycom SoundPoint IP 650 freezes on boot after adding just one custom ringtone
Thursday January 20 2011
9:29PM 1 Introducing easySysAdmin - automated security and telecom fraud protection
9:19PM 0 Asterisk Now Available (Security Release)
8:14PM 4 Asterisk to asterisk t.38
6:01PM 2 Asterisk 1.6 SSH Console Colors Debian Lenny
5:55PM 3 Polycom 500 / MWI
5:54PM 0 iNum at 12 Noon EST Friday
5:13PM 2 Mailing list question 2
5:01PM 7 Mailing list question
4:25PM 2 Accessing a 'user' variable via. dialplan.
3:34PM 1 OT - TTS in spanish
3:19PM 5 context problem
3:00PM 5 ReceiveFax
6:28AM 0 Using asterisk and icecast for live audio streaming.
4:08AM 1 Hi, agent intro-speech for outside caller
3:34AM 1 Internode weirdness
1:56AM 1 No more ISDN in Malaysia Telekom???
Wednesday January 19 2011
11:34PM 0 Cross Queue Priorities
8:05PM 15 res_fax
7:37PM 0 IAX between 1.6 and 1.8 has bad voice quality
6:24PM 1 intermittent problem on 1.4
3:57PM 1 agi dial termination cause ?
2:12PM 0 audiohook.c: Write factory 0x153cf678 was pretty quick last time, waiting for them
2:03PM 0 Asterisk fail over. From IP rewrite issues
10:41AM 1 Asterisk 1.8.2 and digium yum repositories
9:23AM 1 sip dos question
8:21AM 0 How to detect line tone?
4:47AM 0 No RTP Engine problem in 1.8.2
3:55AM 0 Make ConfBridge hang up on last participant
1:52AM 2 Asterisk extension not found problem...
1:34AM 0 progressinband, how much extra CPU load?
Tuesday January 18 2011
11:19PM 1 chan_sip.c: Failed to parse contact info
9:57PM 1 1.8.2: dahdi-2.4: calls dropping
6:09PM 3 Calling rules
6:04PM 1 Sendind e-mail with Hylafax
6:02PM 0 SIP Originate on
4:38PM 0 Asterisk Security Releases: AST-2011-001
4:35PM 3 AST-2011-001: Stack buffer overflow in SIP channel driver
3:58PM 0 Multiple Registrations
3:52PM 0 Asterisk SlackBuilds for Slackware Linux
10:41AM 1 Can I know if a call is transffered to asterisk
7:05AM 1 Ongoing problem with 1.8
6:43AM 0 mobile integration
Monday January 17 2011
11:16PM 1 Max call duration
11:01PM 2 Occasional robotic sound while call in progress
4:56PM 4 how to read mp3
11:29AM 1 Continuously core dumping of 1.8 on SLES
10:10AM 0 Sangoma A104d / overlapdial=yes / dial with audio one-way issue
9:56AM 1 'Bad authorization' error with Asterisk 1.8
9:50AM 1 app_calendar and SSL
Sunday January 16 2011
8:58PM 5 Basic Sip.conf and extensions.conf
8:37PM 2 crashing
12:29PM 1 Selecting the E1 cards for the call
8:42AM 1 T.38 Digium Fax Driver Success on Fail
6:27AM 0 chan_h323 and menuselect dependencies problem
1:46AM 0 app_fax watchdog timeout
Saturday January 15 2011
7:38PM 4 Sound quality issue
4:50PM 1 Problem with chan_dahdi and conferencing
7:01AM 11 Asterisk stops responding
1:31AM 2 Bruce B
12:42AM 14 Top Posting
Friday January 14 2011
11:48PM 0 Tools to Monitor Asterisk Servers and VMs
10:44PM 0 logging
8:46PM 1 Spectralink 8002
8:12PM 1 Asterisk 1.8.3 Now Available
8:11PM 0 Asterisk Now Available
8:11PM 0 Asterisk 1.4.39 Now Available
7:41PM 6 Why are 4 ports used for a single call?
5:58PM 1 Ghost ringing
5:42PM 1 Selecing the E1 cards for the call center
10:55AM 2 DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0'
8:55AM 0 Asterisk+h324m gateway issue
7:07AM 0 Friday Jan 14th @ 12 Noon EST: Humbug
3:05AM 1 5-7 second delay in connecting outgoing FXO calls
Thursday January 13 2011
5:37PM 5 CallerID and URL pop up for windows...
4:19PM 2 queue_log in MySQL database
12:31PM 4 Fax stopped working when upgrading to 1.8.2
11:51AM 1 WARNING T.30 ECM carrier not found
6:32AM 3 Polycom Blf / Directed Pickup
4:13AM 1 Call hung up?
2:00AM 1 SetVar Warning
Wednesday January 12 2011
7:14PM 1 Queue periodic announce...
6:23PM 2 Problems with ZAP Channels
5:47PM 0 Paid or Free software that would do pop-up from Outlook 2007 via Asterisk AMI
10:37AM 1 DTMF not being heard correctly by far end conference system
10:15AM 0 Fail2Ban & CSF
9:42AM 0 Why Local Channels are creating
Tuesday January 11 2011
10:28PM 1 Unable to get Fax t38 working with IrisTel trunk
8:33PM 1 Issue with Red Alarm with DAhDi
7:19PM 1 Using the Telco Call Transfer Features.
6:31PM 2 Asterisk timezone issue
5:16PM 3 Asterisk hardware server
3:52PM 2 Show voicemail in GUI
2:20PM 6 OpenVPN + SIP configuration?
8:14AM 2 asterisk fax problem
7:58AM 2 Do I need a sip proxy?
6:41AM 1 Fix Fake Answer Supervision In asterisk1.6
5:42AM 0 slow response to INVITE
Monday January 10 2011
9:03PM 1 New Dahdi error
6:04PM 1 Failed SIP registration kicks registered device off?
2:57PM 2 Call Back on Busy
2:47PM 2 How to reject an incoming call using AMI ?
12:58PM 3 sendrpid does not work!
8:50AM 0 Member penalty and Queue strategies
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5:06AM 0 OPTIONS Packet is retransmitting continuously
5:05AM 1 environment variable + res_mysql.conf
4:01AM 3 How to check a number online or offline
Sunday January 9 2011
7:34PM 0 Call parking question
1:27PM 3 Mail list Woes?
Saturday January 8 2011
8:01AM 0 Grandstream GXE2504A codec disable option
5:34AM 0 AstLinux 0.7.5 released
Friday January 7 2011
6:33PM 3 Definations of READ/WRITE parameters of manager.conf contexts?
5:06PM 1 AGI->Macro w/Agruments
1:41PM 0 Channel name changed in asterisk 1.8
11:38AM 3 Call queues on load-balanced asterisks
1:54AM 0 Anyone have Festival application working?
12:01AM 1 system lockup when going into conference
Thursday January 6 2011
8:42PM 0 SILK codec
2:06PM 2 Benefit of PRI vs SIP trunk calls
12:14PM 1 cannot answer incoming calls
4:26AM 0 using google for vm transcripts
3:54AM 0 TDM410 and DSL
2:12AM 1 Too Few Fax Detections
Wednesday January 5 2011
11:59PM 2 Weird phone behavior after recent CentOS 5 update
9:56PM 1 Asterisk replying to wrong port for NOTIFY messages
4:50PM 1 Polarity Reverseal....with analog line
4:45PM 0 TE420 issue: card 0 span N: isr2=XX isr3=Y
4:42PM 2 DTMF-troubles with Snom
3:47PM 1 Blind Transfer not working - 1.4.38
11:30AM 1 Add Privacy: id to SIP-invite
10:57AM 2 Calls Transfers
7:29AM 7 Are the Siren7 and Siren14 the G.722 HD voice codecs?
7:27AM 2 Asterisk Outlook integration
1:42AM 3 VoIP PoE phones for restaurant (kitchen)
Tuesday January 4 2011
11:13PM 4 Do not disturbe
11:09PM 2 problems inserting dahdi modules using Debian Leni
9:59PM 0 DAHDI and dialdebounce
7:44PM 0 Fwd: Announce: telepathy-ring 2.1.1
5:31PM 5 MOH problems (asterisk 1.4.38)
5:26PM 0 Queues, priorities and (miscalculated) holdtimes
5:14PM 0 OT - Contact center - How to Gmail-like label to incoming email
3:36PM 2 Call forwrading but call transfer back
10:14AM 1 Go from CALLINGout to just CALLING
5:00AM 1 Question About Conferencing Capabilities
4:04AM 1 1.8 MIBs
Monday January 3 2011
5:26PM 1 Clarification on DAHDI Fax Detection
3:30PM 1 VoIP PoE phones for restaurant
3:13PM 1 changed datadir
12:16PM 1 digim tdm2400p fxo fake answer supervision problem.
Sunday January 2 2011
11:26PM 1 Realtime SIP, multiple AX servers question
7:50PM 2 incoming
5:44PM 2 Forward voicemail not working
5:14AM 0 CDR Questions
12:04AM 2 Callback form to place on site for customers. Recomendation to achieve this.
Saturday January 1 2011
5:50PM 1 Cisco IP Phones and AVAYA IP Phones: How to configure in Asterisk
5:43PM 4 Saving the monitor file on new file always using Monitor(wav, Record1, m)