asterisk users - Mar 2010

Wednesday March 31 2010
9:29PM 2 Necessary hardware
8:29PM 2 How to run Music while looking for the caller in Database
7:56PM 0 Upcoming Asterisk 1.6.0 and 1.6.1 Maintenance Changes
6:57PM 2 Multicast Paging
6:00PM 0 meetme() and dahdi_dummy on an embedded system
3:57PM 2 Reset personal voicemail settings
1:09PM 1 Unable to login to voicemail with Ekiga
12:06PM 1 Jitter Buffer and MeetMe.
1:05AM 2 Asterisk hangup all outging calls after 32 seconds
12:27AM 0 app_txfax.c
Tuesday March 30 2010
10:09PM 0 E1 card w/o echo cancellation
8:46PM 0 E-mails from Asterisk coming from root
8:16PM 2 convert from wav or mp3 to gsm
7:54PM 2 Dropped Calls
6:08PM 1 DAHDI 2.2.1, Asterisk - Channel unacceptable (6)
6:02PM 2 Priority based softhangup
5:38PM 5 Confusion on call forwarding
10:18AM 0 Asterisk realtime ldap:active directory
7:10AM 0 How can install and use Async AGI
6:17AM 0 Inbound configuration
2:08AM 1 a2billing wont pass the number
1:57AM 0 Diameter for Asterisk, Traffix Diameter stack ?
12:30AM 1 How are your PRI interrupts balanced? (+ Soft lockup BUG)
12:29AM 0 Asterisk and Call files
Monday March 29 2010
10:42PM 1 Trying to get reason for ending of AGI call recording
9:10PM 0 amr
8:46PM 3 Asterisk system for church call center
7:03PM 1 Asterisk, IAX, & Sub interfaces
5:32PM 0 No audio when calling via PSTN, before remote answers (with polarity reversal)
4:08PM 3 Foip solution
1:42PM 1 Realtime Issue
1:03PM 3 Slightly more advanced dialling..
10:18AM 0 queue autopause status
9:33AM 0 MixMonitor and StopMixMonitor
9:13AM 5 Continue a dialplan when the client hang up the call
7:34AM 1 is it possible to connect Digium TE420 and Cisco card?
12:55AM 9 24 FXS Port Voip Gateway and Asterisk
Sunday March 28 2010
3:19PM 1 Updating Asterisk and its use with MySQL
3:13PM 1 Libtonezone
Saturday March 27 2010
9:48PM 3 Trying to configure xorcom on Suse 11
8:33PM 1 migration
3:17PM 4 Cisco 7960 become UNREACHABLE behind pix firewall
Friday March 26 2010
9:41PM 2 dnd not working correctly
8:22PM 2 What does this error message mean
6:14PM 0 Re :Re: Sip module and dns (Alyed)
5:58PM 0 Re :Re: Sip module and dns (Alyed)
2:01PM 1 no voicemail on pstn line
1:18PM 1 problem with polarity reverse
10:29AM 2 Is there any Diguim distributor in Lahore
10:10AM 2 need help on setup rtp directly between 2 sip clients
9:13AM 0 Delay on sip channel
8:36AM 7 Asterisk load balancing and failover
6:39AM 1 [VUC] Voipathon 24-hour online party begins in 30 mintes
6:22AM 1 SIP/2.0 403 Forbidden
3:55AM 1 "Failed to play transfer sound! " during attended transfer
2:46AM 0 Time counting while playback
2:37AM 1 send a call from A to B use sip trunk prablem
Thursday March 25 2010
6:17PM 4 Background noise
4:48PM 0 call not routed
3:00PM 1 Static linking
1:07PM 0 intergration of Diameter
12:14PM 2 rtp.conf ports for inbound or outbound?
9:58AM 1 configure the sound for inbound calls
9:26AM 2 Attended transfer and callerID updates forSiemens Openstage phones
3:29AM 3 How to get Sip response codes in Dialplan?
3:26AM 0 Music class default requested but no musiconhold loaded
12:42AM 9 Maximum number of PRI calls on 1 asterisk box (no HW echo)
Wednesday March 24 2010
11:07PM 1 Aastra weirds IP 169.x.x.x
9:56PM 2 new server install errors starting asterisk
7:42PM 0 Dahdi-linux & Dahdi-tools Release Announcement
7:37PM 1 This is a test, hijack this
7:18PM 1 installing dahdi card
6:42PM 5 Asterisk 1.6 and OpenVPN RTP problem
5:51PM 0 AstLinux 0.7.1 released
4:26PM 1 software version (lets try it again)
4:26PM 0 chan_h323 and ToS
4:02PM 6 Restarting Asterisk using a script - Thanks to all -
3:28PM 2 software version
10:33AM 1 Firewall & audio : need a wide range to work !
9:32AM 0 Hook playback or ControlPlayBack cmd
8:48AM 1 G.729 Codec problem.
4:29AM 1 pstn calls not picked up
4:13AM 1 Mobile phone shut down, but Queue() Ring as usual
2:06AM 3 AMD reporting NOTSURE most of the time
1:33AM 0 Asterisk with Grandstream HT502 T38 Fax
Tuesday March 23 2010
11:16PM 4 Safe_asterisk doesn't exists???
8:40PM 1 permit/deny in sip.conf iax.conf
7:41PM 5 G.711a or G.711u ???
5:31PM 2 Sip module and dns
5:06PM 0 Strange Meetme disconnects
4:59PM 0 In Berlin this week? Kamailio/Asterisk community dinner on Thursday
4:45PM 0 Classic NO AUDIO problem - DD-WRT and NAT forwarding - HELP PLEASE!
4:21PM 1 Minimalize jitter in VoIP calls
3:07PM 0 distribuited ACD on many asterisk nodes
2:16PM 1 Asterisk crash - segmentation fault
12:57PM 5 Install dahdi on Xen virtual console
8:53AM 0 [asterisk-ss7]Chan_ss7 issue
8:50AM 1 chan_ss7 issue
3:14AM 0 (no subject)
2:43AM 3 How to make upgrades with Asterisk
1:38AM 3 Which folder for sounds?
Monday March 22 2010
11:40PM 3 Can I call myself on the same machine
8:57PM 1 Play music to caller after answer, before dial
5:33PM 2 Transcoding question
5:26PM 2 requirecalltoken & receiving IAX calls
2:31PM 0 DUNDi Confusion
1:56PM 2 Context vs. Custom Context
1:48PM 1 Call files : call multiple SIP-accounts
11:36AM 2 voicemail problem
11:25AM 1 PRI lines do not have CallerID activated yet it is
Sunday March 21 2010
8:57PM 1 Invalid Makefiles to install asterisk with ldap
8:27PM 1 Asterisk Manager Interface (AMI) proxy recommendation
6:23PM 1 Asterisk Died - Ver-
1:35PM 0 dahdi_monitor doesn't show data on RX & TX: broken card or cable?
1:06PM 1 test
12:52PM 1 How to get Asterisk to make batch calls?
12:30PM 6 Do i really need Dahdi and Libpri.
12:58AM 1 Early audio problem in chan_dahdi
Saturday March 20 2010
11:52PM 1 Voicemail, Asterisk and Grandstream BT200
9:35PM 1 -> T38 Fax: call drops
5:20PM 1 how to start callerid for india
2:34PM 1 SIP signal through one IP and media through different IPs
2:31PM 3 Asterisk general Timeout for digits
1:08PM 2 8Port Junghanns BRI card under Dahdi
10:24AM 1 basic pc to pc voip in lan
10:06AM 0 Elastix 1.6 continuos ring
Friday March 19 2010
8:11PM 4 Call Drops while doing assisted transfer from remote location
4:12PM 0 Setting Caller ID for attended transfer
3:37PM 2 register => 2345:password@sip_proxy/1234
2:50PM 0 SPA3102 + asterisk drop call and loop (was SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?) )
9:52AM 1 too much sockets open by asterisk
8:36AM 1 Strange initial RING
6:50AM 0 rtp connection remained when call busy using agi for call control
6:11AM 1 how to configure caller id
3:20AM 2 confbridge not working?
2:05AM 1 Define an array of sip number in sip.conf
12:19AM 6 (no subject)
Thursday March 18 2010
9:27PM 3 Free Daily Asterisk News iPhone and iPod Touch app
7:22PM 1 SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?)
4:49PM 0 Problem with forwarding: Now forwarding SIP/ XX to Local/
3:04PM 0 SIP Router Project
2:02PM 6 Asterisk DIES with no trace. PLEASE
1:39PM 0 Software for my laptop to send Fax via H.323 ?
12:18PM 1 Voicemail Remote Access
9:21AM 2 Live Audio Streaming- From Aux interface-Online resource
8:31AM 2 How to detect a PSTN telephone is busy or not?
4:29AM 0 Asterisk and OOo Smart Tags
12:49AM 2 Wanted: free DID number and provider feedback
Wednesday March 17 2010
11:39PM 2 DID number
11:09PM 0 Need help with auto-forwarding virtual extensions (Asterisk 1.4/GUI 2.0)
8:08PM 7 Asterisk DIES with no trace. PLEASE HELP!
5:41PM 1 BT ISDN-30 Call Failures
4:29PM 0 monitor SIP jitter buffer
4:08PM 1 Adding an external dial code
3:37PM 3 SIP codec negotiation / manipulation
3:35PM 2 Call Filtering
2:16PM 2 Asterisk running on a Xen Centos Server challenge!!!
1:13PM 2 Asterisk as a skinny/sccp "client"?
11:43AM 2 Asterisk and app_system FAILED using TRYSYSTEM
9:40AM 3 asterisk fax handeling
5:48AM 2 sip send image
3:13AM 0 [NEWBIE] Simple hunt group on SIP -- need confirmation
Tuesday March 16 2010
8:02PM 1 Outbound route prefixes
7:55PM 1 Asterisk hangup all incoming calls after 10 seconds
7:39PM 1 softhangup
2:02PM 0 Asterisk to be used with Ciscs media gateways
1:59PM 1 Asterisk + Sip Phone + BLF
1:29PM 1 different authentication requirement
11:59AM 2 DID/CID doesn't match "." (dot) in CID field
2:01AM 3 Asterisk 1.4.24 DUNDi CLI commands not found
Monday March 15 2010
9:53PM 1 Article - a method on how to evaluate an Asterisk server
9:15PM 0 How to find Asterisk compile time options for building app_swift module
7:40PM 1 Installing cdr_pgsql on asterisk
5:44PM 1 dnd
2:25PM 0 asterisk-users Digest, Vol 68, Issue 33
1:15PM 2 Android Phones ;-)
12:32PM 1 AEL in 1.6 and Gosub
8:25AM 1 CDR: Add Dialed Number Identifierfield (DNID) field into MySQL
5:27AM 2 High Availability Asterisk PBX
Sunday March 14 2010
10:56PM 0 Change SIP Release Code
10:23PM 1 queue MOH
10:10PM 2 Help with playing a recorded message in a conference.
5:38PM 0 Strange audio problem with Digium Wildcard B410
5:32PM 2 dahdi-linux-complete-2.2.1+2.2.1 failed to compile
2:30PM 1 Debugging log rotation problem
2:26PM 0 ooh323_indicate: Don't know how to indicate condition 20
1:38PM 0 DECT phone wont stop ringing
Saturday March 13 2010
9:40PM 2 DID forwarding ?
7:08PM 0 How to test my Dial(SIP/...) ?
6:40PM 0 func_devstate with latest 1.4...
4:33PM 1 IAX2 peer question
3:14PM 0 SIP debug on a per call base
2:40PM 2 Asterisk on MPLS VPN
1:03PM 1 adding agent with 2 phones to a queue
9:14AM 0 PBX_DUNDI question
6:00AM 0 DUNDILOOKUP doesn't return record
4:00AM 0 Skype for Asterisk and regular expressions
Friday March 12 2010
10:36PM 1 Setting up RTP to flow between endpoints directly bypassing Asterisk
8:30PM 0 Asterisk Now Available
8:29PM 0 Asterisk Now Available
8:24PM 0 Asterisk Now Available
8:23PM 0 Asterisk 1.4.30 Now Available
7:19PM 1 Asterisk x64 with Skype and DTMF on skype-out.
6:23PM 0 Installing chan_H323 by yum?
6:21PM 0 modem config & pots & documentation
6:08PM 2 Polycom not updating the directory list
4:00PM 1 t38 ATA
3:31PM 2 ExtenSpy Problem
2:54PM 0 Regarding - P-Asserted identity and Privacy - SOLVED
1:17PM 0 Asterisk 1.2 crash: gdb trace on core dump
12:17PM 1 1.2 to 1.6 and bristuff
9:19AM 4 Can not enable sip debug because CLI flooded
8:03AM 3 Time counting down and # detect
7:59AM 0 Fri March 12th @ 12 noon EST: SIP scanning, security and attacks + Hosted vs on-site voip
2:57AM 0 Running DEADAGI from h extension
12:07AM 2 Fwd: Switchvox SOHO 4.5 is Here
Thursday March 11 2010
7:54PM 1 Digium TE4xx T1 Bonding
5:09PM 2 Codec preference
1:44PM 1 Testers Need Issue #0016965: [patch] DBGet response does not end with a 'Complete' event
10:33AM 2 Is there a way for a peer to clear its registration from a server?
6:16AM 2 How to add custom CDR fields to MySQL
2:27AM 2 Phones won't stop ringing
2:19AM 0 Unable to forward voice or dtmf
2:07AM 2 press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license
Wednesday March 10 2010
10:58PM 1 Diaplan reload command not working
10:35PM 1 Phishing attempt posing as digium
8:37PM 0 Meetme Closes Conference After One Hour
8:32PM 2 PGSQL application
5:26PM 1 BLF and realtime SIP buddies
3:35PM 1 multiple RTP port ranges for SIP
3:34PM 1 00h323 cant get gatekeeper to connect
1:53PM 1 dtmf payload 100
12:58PM 2 Passing a parameter to voicemail
11:16AM 0 I loose incoming call after transfer
9:38AM 4 Extensions.conf changed but not take effect
9:29AM 1 callerid change name
8:09AM 1 func odbc and mult iquery
6:31AM 0 call features affected by native bridging between sip phones
6:13AM 0 CLI not working properly - Asterisk Freez
Tuesday March 9 2010
11:20PM 1 Which spandsp to use with 1.6.2?
10:27PM 0 Queue Member stuck in Ring+InUse?
9:50PM 1 Asterisk SMDI for Nortel Option 11
6:34PM 1 confbridge manager/cli
4:15PM 3 Snom Provisioning
11:31AM 0 Disable echo canceller Fonebridge
10:58AM 1 asterisk peer uses 5060 to send and 5061 to receive
10:54AM 0 Asterisk crash with chan_capi upon calling to PSTN
9:29AM 1 app_queue problem with Ringing state
8:51AM 0 DUNDI Sip authentication failure
1:08AM 1 Aastra, Asterisk 1.4 and Voicemail
Monday March 8 2010
11:57PM 1 SIP handset + SLA example
11:31PM 2 fax & spandsp
7:45PM 5 Dialplan behaviour
6:25PM 0 Voip Users Conference March 26th
4:53PM 1 Turning off DNIS on T1 set to FXO_LS protocol
1:49PM 0 Is it possible to configure Asterisk so that it does the Q.SIG “Path Replacement Feature” ?
10:10AM 3 Calculating R Factor and MOS metrics for VoIP
9:33AM 1 Play an audio file from a remote host
7:50AM 0 Dail of meetme options
6:18AM 3 dahdi not available in Asterisk
Sunday March 7 2010
8:18PM 1 Grandstream HT 503 Outoing 403 Forbidden
6:14PM 1 Attended transfer broken in
5:42PM 1 Caller Presentation Confusion
2:21PM 3 Callcenter open source program
9:52AM 4 press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.
3:18AM 5 dahdi-2.2.1 & kernel-2.6.32: working for anyone?
Saturday March 6 2010
11:50PM 2 Mail-2-Fax and Fax-2-Mail solution for Asterisk with T38
10:09PM 1 Custom App
8:14PM 2 saving pressed keys
7:50PM 0 SIP, internet calling, per-peer contexts, and multitenancy
5:46PM 1 MOH over IAX2 - NOT working
10:41AM 0 Audio problems ins conference zap->sip
9:45AM 0 SIPit 26 in Sweden - organized by Edvina
Friday March 5 2010
10:29PM 2 MOH Oddity
8:06PM 2 app_confbridge production ready?
5:40PM 1 Observation about DAHDI, FAX and Echo cancellation
5:18PM 0 Regarding - P-Asserted identity
5:00PM 1 State of 64 bits applications in Asterisk
4:17PM 3 Denial of Service Attack
4:10PM 1 AMI logs
4:08PM 3 Hardware requirements question.
3:33PM 2 FollowMe / Asterisk 1.4 Question
3:15PM 1 Asterisk 1.4 Followme Question
2:41PM 0 Follow-up to CALLERID(num) not working
2:18PM 1 Playback in h extension
12:38PM 0 MGCP FXO endpoint
11:20AM 4 Deadlock in Asterisk
10:38AM 3 Having problems with BLF
9:50AM 1 Asterisk Management API
9:27AM 1 iLBC installation problem
1:42AM 1 SIP / Echo Cancellation
12:22AM 0 PHPAGI and Asterisk 1.6
Thursday March 4 2010
11:35PM 1 Remote Agents
10:51PM 9 30 mins GSM file
10:02PM 1 InterPBX communication using SIP
3:46PM 1 time/date over POTS?
3:00PM 2 UK CallerID -v- Wildcard W100P
2:59PM 0 Asterisk & Sofaware & Polycom
2:19PM 0 Availstatus returns 20 ?
6:43AM 1 No Audio on pstn call
6:02AM 1 [asterisk-user] SIP / Echo Cancellation
3:37AM 1 how to create a dummy call
1:32AM 0 CallerID and distinctive ring detection
Wednesday March 3 2010
10:41PM 6 Identify scripts connecting to the asterisk manager
9:23PM 0 Looking for a configuration guru to collaborate with
9:23PM 1 Free 'Locked up' Channels
5:34PM 1 asterisk SIP, SIPAddHeader() and Cisco GED-125
5:22PM 2 Best practise for ISDN Video Conferencing..
5:21PM 1 911, channel full
4:30PM 1 forward problem!
3:27PM 0 Is this a bug?
2:22PM 0 CALLERID(num) not working
10:41AM 0 how can I release trunks after transferring 2 calls connected on trunks between the same machines.
10:13AM 2 Getting verbose or debug tracing in Asterisk
6:01AM 3 dahdi and oslec
4:41AM 0 how to play background music during record
4:32AM 0 asterisk-users] how to create a dummy call
1:23AM 0 Dial timeout problem with OpenVox A1200P Card / FXS module
Tuesday March 2 2010
11:27PM 1 Uverse, Asterisk and SIP
7:48PM 0 FW: ARI problem with monitor
7:37PM 1 Asterisk and cellphone/GSM voicemailbox
6:36PM 5 MWI and 1.6.1
5:04PM 0 asterisk-users Digest, Vol 68, Issue 4
4:54PM 0 ARI problem with monitor
4:28PM 1 realtime call peers status
3:05PM 1 Hide time consuming processed by prompt
12:56PM 1 dialplan reload: not working with large dialplans
11:44AM 2 cli_originate malfunction after upgrade from to
11:14AM 1 Sip module problem
10:48AM 0 1.4 chan_sip use internal IP for dialog-info+xml SUBSCRIBE, why?
10:14AM 6 Echo cancellation on DAHDI
7:50AM 3 SIP Trunk with "multiple" remote ip-addresses
6:26AM 1 Does Asterisk Support SIP TLS encryption
Monday March 1 2010
11:57PM 1 help with install
11:31PM 3 help!!! Internal extensions not connect
10:46PM 3 User on PC?
10:20PM 0 Solved:Re: OT:4 Line DECT Cordless phone without answering machine
10:08PM 1 OT:4 Line DECT Cordless phone without answering machine
5:04PM 0 Asterisk / Trixbox 2.6 Streaming MOH Problems
4:39PM 2 Unable to register a sip account with x-lite
4:00PM 1 Fwd: Erika DeBenedictis-Recommendation
3:39PM 1 AVM Fritz! mISDN with Kernel 2.6.32 - Any experiences? - Email found in subject
2:34PM 0 Attended transfer: transferring a call as soon as the destination starts ringing
2:31PM 0 SPA3102 Firmware Upgrade via TFTP fails
2:25PM 3 Asterisk and Cisco DTMF
2:06PM 1 rtcachefriends & qualify
1:05PM 1 Swift from eagi, problems with prosody rate
11:42AM 2 MeetMe and usernum
10:22AM 2 Is answer() necessary ?