asterisk users - Apr 2011

Saturday April 30 2011
7:10PM 2 Using Asterisk + other FOSS projects to facilitate a call-in Internet Radio web-frontend
5:15PM 1 dial multiple extensions
2:29AM 26 HA Asterisk
Friday April 29 2011
2:51PM 0 Local channel scenario flushes CDR before dialplan end
2:10PM 4 SIP bad request
11:03AM 0 Hardware Server Configuration/8 or 4 port PRI Card
2:57AM 0 Friday on VUC: Jabber/XMPP
Thursday April 28 2011
8:30PM 1 anybody out there sucessfully using gnugk?
4:09PM 6 odbc error - server is gone
3:25PM 12 How to create distortion, echo, and chopping sound in a SIP trunk?
Wednesday April 27 2011
7:34PM 84 Discussion: Are we ready to leave 1.4 behind?
7:06PM 5 Echocancellation OSLEC vs MG2 ?
6:04PM 2 DHCP / DNS
5:34PM 2 asterisk practices
5:16PM 2 Asterisk, SIP & Firewalls
3:41PM 4 h323 with NAT
1:47PM 1 AGI WAIT FOR DIGIT - key press BEFORE command
12:26PM 2 Digium WCTDM24XXP DTMF CallerID
11:29AM 0 Konference module issue
11:22AM 10 how to know status of asterisk from php
6:55AM 0 Has anybody been able to install CDR-Stats all the way through?
5:18AM 0 Retaining original caller id
Tuesday April 26 2011
9:20PM 0 Seattle WA Asterisk Users' Group
5:04PM 4 Asterisk Now Available
5:04PM 0 Asterisk 1.4.41 Now Available
5:01PM 2 Asterisk Now Available
3:13PM 0 siren sound
2:32PM 0 play audio file to destination SIP channel on attended call transfer
2:13PM 7 Password to be ecrypted?
9:43AM 11 Orginate not working well with PSTN lines
8:14AM 3 How does wrandom strategy works with Queue?
Monday April 25 2011
10:02PM 4 PAP2T auto answer?
9:30PM 1 Transfer beep w/ Polycom phone
8:10PM 0 Registration problems - Vitelity
4:36PM 0 FILTER function and multiple ranges?
3:17PM 1 new confbridge
1:38PM 15 The new ConfBridge application is now in Asterisk Trunk!
1:51AM 2 (no subject)
Sunday April 24 2011
8:21PM 1 Realtime and priority labels
3:45PM 2 Best modem for chan_datacard
Saturday April 23 2011
5:52PM 1 ARA table definitions (1.8.*)
4:20PM 3 call files
3:48PM 5 DTMF not being sent ( RFC2833 )
12:56PM 2 Warm Transfer in Asterisk
Friday April 22 2011
6:49PM 0 nyc area pbx rfp 4000 extensions
5:48PM 0 Multi tenant Parking issue
5:17PM 0 WARNING T.30 ECM carrier not found
5:13PM 0 (no subject)
3:02PM 10 Cannot call to my server with SIP
1:55PM 0 question on register and dnsmgr_lookup
12:45PM 0 ZRTP SDK Source
10:13AM 0 T38 fax printer Windows client for asterisk 1.8
8:05AM 9 Flite issue
5:21AM 0 Help Required---Problem in Installation without dahdi
Thursday April 21 2011
9:48PM 0 Asterisk,,, and Now Available (Security Releases)
9:40PM 2 AST-2011-006: Asterisk Manager User Shell Access
9:40PM 0 AST-2011-005: File Descriptor Resource Exhaustion
6:52PM 0 Nationalprefix chan_dahdi option
5:26PM 17 missed call notification
4:30PM 1 IAX2 codec selection and video
1:53PM 2 Transcode ulaw/g722 problem
12:12PM 3 Asterisk Export Fax from Wave file
11:20AM 6 [asterisk-user] Can't get hostname on asterisk dialplan by ENV()
9:42AM 6 Nat=yes
9:34AM 0 Voicemail forward issue
Wednesday April 20 2011
9:32PM 2 py-Asterisk or pyst?
8:05PM 4 VoiceMail to text mail
7:40PM 3 allowguest=yes, how?
5:16PM 4 asterisk log - "=======" extension not found?
4:37PM 4 [IAX] Everyone is busy/congested at this time (1:0/0/1)
4:20PM 4 Call files or AMI originate for mass outbound call
4:07PM 4 issue with installtion asterisk
3:02PM 1 dtmf payload type problem during faxing..
2:34PM 0 1.8.x sip error returned -1: Invalid argument
9:50AM 2 No voice in MeetMe for SIP with
2:41AM 4 Configure IVR(Inbound and Outbound)
Tuesday April 19 2011
11:14PM 0 IP Address Management / Open Source / IPAM
5:09PM 5 How to know how many calls are into hold by asterisk command
4:23PM 0 sterisk+SS7 Error: chan_dahdi.c: Unable to start PBX on DAHDI/288-1
4:09PM 0 RTP and Signalling Dropping
3:51PM 0 chan_mobile: Dropping incompatible voice frame
2:50PM 1 chan_dahdi under bristuff 0.4.0-RC11 not building under Linux 10.04.2 LTS
8:43AM 1 ConfBridge and AGI
6:41AM 5 No voice in MeetMe for SIP with AGI_BACKGROUND
Monday April 18 2011
6:47PM 3 core show channels consise in asterisk 1.8.3
6:12PM 1 A101DE Sangoma Card in AsteriskNow 1.7.1
4:35PM 0 canreinvite yes or no for PBX
4:01PM 2 Meetme Time Limit?
1:46PM 5 Asterisk unresponsive
1:19PM 1 Softphone IAX
10:16AM 12 No Internet, no asterisk
10:06AM 6 Call Center Reporting
9:54AM 2 Registrations stops after 403 FORBIDDEN
7:16AM 1 Asterisk, virendra bhati has invited you to open a Gmail account
5:46AM 0 [OT, Job] Senior Software Engineer for exciting, high-growth startup
Sunday April 17 2011
6:46PM 1 Asterisk 1.8.3: Started but no SIP talking
Saturday April 16 2011
11:28PM 4 Jabber / GTalk / hints
11:13PM 6 Jabber / facebook chat?
9:58PM 1 CDR & ARI Question
2:17PM 0 PADLOCK asterisk 1.6
12:41PM 1 "chan_sip.c: No such host:" but I can resolve it from command line ?
7:56AM 15 Google Voice receiving call problem
5:24AM 6 any experience with cisco media gw with fax???
3:26AM 0 Duplicate cdr records with channel local
Friday April 15 2011
11:00PM 5 1.8.4-rc2: ReceiveFAX fails
7:29PM 1 Reach PSTN from another Asterisk
5:35PM 0 Hot to make call parking to Mult tenant
4:50PM 1 sangoma card rx/tx gain level
12:33PM 8 Good by asterisk 1.4? Please not.
12:13PM 2 If voice mail not found dialplan
12:02PM 6 Possible bug in Hangup() (Asterisk 1.4.x)
11:39AM 4 Friday April 15 at 12 Noon EDT
10:40AM 0 [OT] 802.11x roaming
10:10AM 0 How to get back park call
7:58AM 0 Would a job posting be ok for this list?
7:39AM 6 sip error logging
Thursday April 14 2011
10:03PM 0 Followme() and variables
8:46PM 1 Existing Asterisk 1.8 upgrade with new release
3:48PM 0 Asterisk modifies from header
3:22PM 1 Microsoft Lync server and Asterisk access
12:51PM 2 setting sip headers when using call files
5:22AM 1 Processing sip messages
12:46AM 1 Asterisk port 5000 open
Wednesday April 13 2011
8:20PM 0 Applet based softphone for Asterisk
8:00PM 8 Safe to upgrade to Centos 5.6 now ???
5:36PM 0 Fw: SIP Trunk send DID or DNIS information
4:16PM 1 Asterisk Tech Tips: Cookin' with Asterisk
2:50PM 3 Asterisk thread limit
2:14PM 10 T38 fax detection using g729
2:08PM 6 Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card
1:48PM 3 Problems With DAHDI on Ubuntu
1:08PM 4 AGI and forking
12:08PM 1 Fwd: Re: Asterisk as a Condo door opener/intercom
9:27AM 2 How to know extensions status ???
9:01AM 4 [OT] Yealink Phones
8:57AM 0 Poor call quality - line drop, chopping sound, like robotic voice, Both party could not hear caller voice
8:48AM 0 0018818: [patch] Crashing when using local channels and realtime on asterisk 1.8.3-rc2
8:15AM 19 Realtime SIP & peer status
5:13AM 1 Aastra 480i & Asterisk No musiconhold
Tuesday April 12 2011
11:30PM 0 Debugging DTMF Detection
11:08PM 0 Problem with Swift app and escape digits
8:31PM 4 Basic queue question
5:14PM 0 From CDR to CEL
2:42PM 1 CEL Logging to MySQL - Please Test
2:24PM 0 Authentication failure
1:37PM 1 Queue(): How to know Estimated wait time for caller in advance
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
6:01AM 0 No subject
3:37AM 3 Templates
1:42AM 1 Poor call quality – line drop, chopping sound, like robotic voice, Both party could not hear caller voice
12:32AM 0 Features.conf - Blind Transfer
Monday April 11 2011
10:38PM 0 DAHDI-Linux Released
8:11PM 2 Voicemail to email issue
5:56PM 0 "Wait for leader" allows crosstalk between participants
5:52PM 1 How to know the SIP status
2:44PM 7 Asterisk kernel CONFIG_HZ=1000
2:28PM 4 voicemail odbc "Length is ....."
2:26PM 3 Asterisk codec negotiation and canreinvite=no
2:04PM 1 Require dialplan
1:50PM 0 Problem with E1 (ISDN) + DTMF
1:43PM 2 Asterisk-Asterisk E1 connection
11:24AM 5 Asterisk MOH not working with Elastix asterisk
10:25AM 0 update CDR fields after Queue
9:44AM 1 Unable to negotiate codec with iax
5:07AM 10 Variable stripping/removing part of string
4:37AM 4 changing port 5060 to 5061
Sunday April 10 2011
4:37PM 5 Asterisk as a Condo door opener/intercom
1:32PM 6 AsteriskNow updated to Centos 5.6 and DAHDI doesn't work
Saturday April 9 2011
4:15PM 4 Asterisk FOP
2:52PM 4 Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension?
8:31AM 1 asterisk-users Digest, Vol 81, Issue 27
Friday April 8 2011
8:35PM 2 Call Recording using MixMonitor - close, but would like some more words of wisdom.
6:56PM 2 Documentation for Asterisk AMI Events?
6:13PM 30 send voicemail to multiple emails
3:07PM 0 Any PHP Ming + for Asterisk guides, tutorial, how-to anywhere?
2:48PM 21 IAX2/
1:56PM 2 Maniuplate callerID based off of callerID
1:43PM 0 User registration failure bug ?
9:57AM 14 Variable inheritance with dialplan command Originate
9:40AM 0 asterisk-users Digest, Vol 81, Issue 21
9:11AM 1 CRC Zaptel.conf
7:11AM 0 488 error in T38 Gatewaying in Asterisk 1.8 with patch 13405
6:51AM 2 MOH not working
Thursday April 7 2011
9:53PM 10 Occasional call from "asterisk"
9:26PM 1 Any way to temporarily disable a registered SIP PEER in Asterisk?
6:56PM 21 asterisk login to voicemail
4:18PM 3 MOH on DAHDI PRI Channels
4:02PM 6 No ringback even though progressinband=yes is set
2:58PM 0 Asterisk 1.8.x Skips DTMF Digits on a First DAHDI Initiated Call
1:34PM 0 AgentCallbackLogin slow in Asterisk 1.4
11:28AM 1 Compiling asterisk using NDK build
9:24AM 5 asterisk SIP MESSAGE method support
8:00AM 3 Asterisk Avaya SIP Trunking One Way Audio
Wednesday April 6 2011
8:59PM 4 asterisk meetme invalid extension
8:12PM 0 Options for DS3 to SIP
7:53PM 5 realtime mysql for 1.8
6:11PM 1 MWI not working on most ATAs in Asterisk
5:30PM 0 Problems with woomera (ISDN BRI) and playback app: Dropping incompatible voice frame
4:46PM 5 voicemail call back loop
2:49PM 1 Question About Codecs
1:45PM 7 BRI Configuration help me
10:54AM 13 Call recording - methodology
7:38AM 2 Call duration problem or maybe calls not hanging up problem
1:10AM 22 Asterisk 1.8.3
Tuesday April 5 2011
8:54PM 5 IAS trunk error AES encryption disabled. Install OpenSSL.
8:39PM 0 minmessage / maxsilence in voicemail.conf
7:45PM 1 asterisk-users Digest, Vol 81, Issue 12
6:52PM 14 dahdi and linux-2.6.38
6:31PM 9 Iptables configuration to handle brute, force registrations?
5:54PM 13 asterisk hints
5:03PM 2 agi create mailbox
4:50PM 12 agi voicemail callback
4:35PM 1 allpage issu on asterisk 1.8.3.x
2:36PM 2 Vestec for Asterisk
2:27PM 4 Iptables configuration to handle brute force registrations?
8:31AM 2 Asterisk 1.8 and new the command: exten => _X., 4, Wait, 2
7:15AM 1 Asterisk => Request a Code
2:08AM 0 dialplan is not finding my number asterisk1.8.3
1:00AM 2 Number Conversion
Monday April 4 2011
9:08PM 3 Read VoiceMail direct
7:51PM 9 WARNING chan_sip.c:3115 __sip_xmit
7:20PM 8 dialplan is not finding my number asterisk 1.8.3
5:08PM 1 MeetMe headache
4:39PM 0 DAHDI-Linux Released
4:29PM 2 call-limit bypass
3:13PM 1 SIP register and contact header
2:27PM 1 how to check if the call is using t38 except in the sip packets
2:09PM 3 Dialplan matching
2:00PM 2 call forwarding
1:36PM 2 Load Asterisk Module with parameters?
1:04PM 6 Asterisk crashes on high IO load
11:58AM 6 CDR fields not being written from "h" extension after "Dial" command completes.
10:34AM 0 Finding out asterisk settings from console
10:23AM 0 SIP channel able to add codecs once up and running?
Sunday April 3 2011
6:33PM 2 From 1.4 to 1.8: stdexten issue
4:01PM 4 Asterisk 1.6 => No sound/voice when i redirect the call
12:56PM 2 hello
10:28AM 0 [DIGIUM FAX] HANGUP problem
6:54AM 2 Top posting - there is no rule.
Saturday April 2 2011
6:03PM 5 Is it possible to dial an automated message when Asterisk receives an email?
2:09PM 0 tarnsfer automatically
8:46AM 3 Registration from '"000000" x 1000
8:36AM 0 automixmon output file location and exec command options
6:11AM 1 Problem getting TDM400P clone card to go off-hook and dial
Friday April 1 2011
9:25PM 2 Polycom 501 alternate
7:55PM 2 codec_dahdi find_transcoders: Failed to open /dev/dahdi/transcode
3:00PM 1 Android tablet voip?
2:52PM 1 call parking issues in asterisk
2:38PM 0 Digium launches flying phone-phone
2:33PM 2 Can gtalk.conf work with multiple GoogleVoice numbers?
12:36PM 2 BRI detection
11:57AM 17 Best Scripting Language
11:28AM 0 Double ## Feature to make another call during current call
11:04AM 3 Fax
10:52AM 0 Incoming SRTP call not working with Bria iPhone Edition
8:25AM 2 The SIP channel driver - I'm giving up.
7:39AM 4 Hold problem with Queue
7:19AM 0 OneAPI / ParlayX