asterisk users - Jan 2009

Saturday January 31 2009
10:40PM 9 Avaya and Asterisk sound one-way
7:10PM 1 asterisk-users Digest, Vol 54, Issue 107
5:37PM 3 Zapata.conf
4:47PM 33 Quiet 24 port POE gig switch
4:04PM 1 iax clients were unregistered after 30sec
1:37PM 5 Is down???
10:24AM 1 where to find STUN Server howto
7:50AM 1 Call without Answer
3:24AM 5 Ideas on how to convert spoken name to text (or wav to text)..speech recognition software?
Friday January 30 2009
8:04PM 0 Can't hear audio when Playback(something, noanswer) on Zap
6:32PM 0 Duplicate Radius accounting in Asterisk.
5:17PM 2 Asterisk with Avaya
5:11PM 8 Asterisk VoiceMail: Is there a web interface for checking voicemail?
3:59PM 3 SIP.Conf - bindaddr per peer?
3:26PM 7 TAPI and Asterisk
2:08PM 9 How to elegantly set DEVSTATE values after restarting
1:43PM 0 Use of Re-INVITE and t,T (transfer) options
12:26PM 0 Friday Jan 30th at 12 Noon EST: Design the Phone of the Future
7:23AM 1 Where to find db1_dump185 in debian packages ?
1:56AM 4 looking for a link or pdf ot something about opensip/openser and load balancing
Thursday January 29 2009
11:58PM 0 Asterisk 1.6.1 Release Candidate 1 Now Available
11:55PM 2 manager API with no login?
7:29PM 2 Can I use an interact and visa terminal through VoIP?
6:15PM 3 early dial: asterisk and ATA
6:15PM 9 32 bit server is ok?
5:23PM 2 Eyebeam or Xlite
4:19PM 1 Managing codecs
4:00PM 8 howto configure an asterisk to send credentials in a REGISTER message to another asterisk
3:30PM 2 RTP/NAT Traffic to private IP
11:54AM 1 blind transfer on hook-flash from SIP phone
11:01AM 2 Don't get asterisk to run behind NAT router
10:47AM 7 Wanted information
10:32AM 0 Benqtelecom in cdr log
9:36AM 1 CDR when replaces
3:30AM 0 [asterisk-dev] DTMF queuing
2:47AM 5 GTalk Channel
1:15AM 13 Callback / Camp / Extention Free notify?
Wednesday January 28 2009
11:40PM 6 How to retrieve a phone number from call forwarding?
10:40PM 2 E1 conection to a Cisco2600
7:59PM 1 asterisk-users Digest, Vol 54, Issue 94
7:48PM 1 Scope of variable
7:01PM 2 SIP Registrations broken on
6:43PM 8 Call Recording Alias
4:45PM 0 problem joining a conference room
4:27PM 10 Inbound Call Disconnect in 3 seconds
3:35PM 15 Looking for SIP loud ringer
3:30PM 20 FAX
10:18AM 2 Zapatel early media issue
9:45AM 2 Record and then Read does not found file
9:19AM 4 route based from source
8:10AM 1 dahdi echocancel configuration for dahdi_dummy?
6:56AM 2 Improving asterisk documentation - sources and what the community can do
6:52AM 0 How many bounces does it take before you get unsubscribed?
Tuesday January 27 2009
11:46PM 1 Asterisk & Twitter - Release/Announce only 'channel' ?
10:53PM 4 Module res_odbc is not loading
10:21PM 3 dialstatus through a call file
6:38PM 2 Muted sound on a Linksys 962
5:49PM 57 USA BRI -- any hope at all?
4:55PM 0 Can't start Asterisk after installing Digium G729 licence [SOLVED]
4:46PM 6 RFC -- Improving the quality of the mailinglists
4:04PM 0 Queue time to answer/abandon + OrderlyStats Server Edition.
3:57PM 20 RFC -- Improving the quality of the mailing lists
3:47PM 2 T.38
3:45PM 19 Asterisk 1.6 dahdi only?
3:31PM 0 SPA-3102 in India - Problem dialing out PTSN
3:17PM 1 Asterisk - Nortel integration via SIP protocol
2:59PM 2 Webcall app needed
12:53PM 1 G726 Codec
11:25AM 1 Queue time to answer/abandon
10:30AM 2 server sizing for ~ 200 simultaneous call
9:24AM 0 asterisk-users Digest, Vol 54, Issue 83
7:48AM 1 Can't start Asterisk after installing Digium G729 licence
6:20AM 0 hangup problem(for spa400)
3:14AM 0 Help with cdr_odbc
Monday January 26 2009
10:28PM 1 Document with differences between 1.2, 1.4 and 1.6?
10:20PM 12 I need help
9:12PM 1 Dial weirdness
7:12PM 3 General Asterisk SIP/IAX provider question
4:22PM 0 goto iax problem
2:51PM 3 Strange Cisco/Asterisk anomaly
2:43PM 1 Suggestion for a new server for E1 line
1:57PM 1 * Queues with legacy pbx extensions ?
12:53PM 2 Network Card
12:08PM 3 Digium TE220 card partially detected
11:54AM 11 Auto Detect
11:28AM 8 German date format in voicemail emails
9:40AM 3 custom cdr userfiled
9:26AM 2 Voicemail
8:32AM 6 Start asterisk on boot
12:20AM 2 dialplan/config basics - analog hangup on keypress
Sunday January 25 2009
11:02PM 3 Zaptel transfer using any button or code, but not flash hook
7:02PM 3 simple dial plan - brain dead operator
6:55PM 8 soft phone
4:34PM 14 CentOS and BAT File
2:08PM 2 how to build a small asterisk pbx
9:23AM 8 Ntework Card
1:48AM 2 Choppy Sound On Bridging From SIP->IAX
1:23AM 3 asterisk help
12:12AM 2 monitoring SIP connection
Saturday January 24 2009
11:38PM 6 no dial tone tdm400p
10:26PM 3 interesting comment. New Physics?
9:08PM 6 Dahdi Init script for Suse?
7:30PM 2 Zaptel? Dahdi?
6:45PM 0 Having tone in my fxs, and loading the zaptel
6:18PM 0 idle-url for Cisco 7940 using Sip
5:32PM 7 NAT router for Linux
4:00PM 8 Reading/Writing the Astdb
2:00PM 16 Which policy for ISDN BRI support in NT/PtMP ?
11:50AM 3 Asterisk freezes with Fixup failed on channel SIP/...<MASQ>
6:46AM 1 local dialing
5:49AM 0 unistim - no dial tone frequecy, no number display when dialing
4:52AM 0 unistim only recognize "default" context
2:57AM 3 Nortel IP phone i2002 - DHCP server unreachable
1:13AM 5 Passing DTMF
12:22AM 3 Logging outgoing calls
Friday January 23 2009
9:55PM 13 Asterisk,,, and released
9:46PM 3 Long Delay after sip reload command
7:23PM 2 Cant Find
6:35PM 2 sip based fax
5:21PM 2 OSLEC, no echo, but a big noise in line
9:00AM 6 OT - Is Netgear ProSafe FS108P with PoE silent ?
6:58AM 1 Trying to do a transfer in agi
5:26AM 13 Packet8 hacked
Thursday January 22 2009
11:40PM 0 Psssst - hey buddy, wanna' get a job? (follow-up to asterisk-biz please)
10:16PM 3 Dumb question: retrieve values from OS-level commands?
9:22PM 3 Looking for Asterisk admin or related job
8:53PM 10 Newbie in Cisco Phone
8:11PM 2 random Linksys question
7:47PM 0 Asterisk Release Candidate 1 Now Available
6:57PM 10 Vicidialnow
6:39PM 0 Friday Jan 23 at 12 Noon EST: Open Source vs Commercial
5:22PM 6 (Fwd) New problem: "They disconnect your service for no reason
3:52PM 21 Root Password not taking
3:21PM 3 registration problem using asterisk 1.6
2:28PM 2 Zap connection problem
1:55PM 3 Incoming fax detection on mISDN hfcmulti B410P card
11:10AM 0 Fw: Re: mISDN BRI Asterisk 1.4
11:01AM 0 Query About Asterisk Dialog Event Package.
9:05AM 4 Help with Avaya integration
8:16AM 0 Few of my phones do not ring when in a queue?
2:31AM 0 DTMF queuing problems
12:35AM 5 oslec + dahdi
Wednesday January 21 2009
10:42PM 3 SIP realtime status...
10:30PM 1 recording failed
9:32PM 0 Polycom SoundPoint IP 500 + X100P + Sirrix PCI4S0 + Conrad HFC-S cards
9:19PM 8 soft ATA on linux with zaptel?
8:52PM 0 g729 with Cisco gateways?
8:14PM 0 Asterisk 1.4.23 Now Available!
6:29PM 0 Playfile to both legs of call
6:14PM 10 snap a number now digium?
4:01PM 4 Need Help
3:17PM 1 Fw: Re: mISDN BRI Asterisk 1.4
11:46AM 4 No Ring on Analog Phone using Rhino Channel Bank in China
10:48AM 7 integration with Microsoft CRM?
10:15AM 2 CDR 0.00 duration
9:33AM 1 Asterisk On Solaris Real Time
9:10AM 0 About Asterisk
9:00AM 0 Prob on DISA
6:37AM 0 Job description
4:58AM 1 Asterisk queues sending calls to members on the phone
Tuesday January 20 2009
11:04PM 11 PAP2T provisioning
10:02PM 3 extensions.conf -- what to do when command throws errors?
9:30PM 0 Timestamp on voice mail messages is based on wrong timezone
8:58PM 4 Using centos and kickstart to build a minimum installation
8:30PM 4 Problem with TDM808
7:07PM 0 Call Dropped in Voicemail / No Reply to Our Critical Packet w/ SIP Debug
6:49PM 3 dead sip channel
5:30PM 2 Setting up an outgoing trunk group
4:56PM 0 Stutter/chopoff first audio played
3:40PM 0 Outgoing CallerID w. DAHDI on ISDN BRI
3:08PM 4 Why does Asterisk not hangup?
2:46PM 0 channel var for Call on hold?
2:32PM 3 CallerID ANI issues
1:29PM 0 Hang up detection problems
1:04PM 0 X-Lite and Asterisk RTP cutting out
11:40AM 1 Called's channel
11:30AM 1 Siemens S685IP registration problems
11:16AM 3 Forwarding calls and trasfer calls
10:57AM 5 SIP DTMF problem with SNOM
10:18AM 9 the FXS ports of Digium and damaging if connected to Tel Line
9:42AM 1 asterisk-users Digest, Vol 54, Issue 53
9:09AM 3 Skype beta news ?
Monday January 19 2009
11:30PM 11 Problems With Playback of Audio On SIP Only System
9:03PM 4 looking for Asterisk experts
8:35PM 4 [somewhat OT] seeking ideas/input for my thesis
8:09PM 7 Suggestions on how to create a hunt or hunt like (rollover, multi-line) group or where to get one?
6:21PM 8 Interesting observation
4:56PM 7 Need help registering Cisco 7960 Phones on Asterisk
4:55PM 1 Server freeze & kernel panic
4:17PM 11 Fring and Asterisk
2:38PM 0 Asterisk On Solaris
1:49PM 1 Cisco 7941G-GE with Asterisk and CTPSEP odyssee
1:45PM 4 How to overwrite CDR(dst) value in h priority?
1:40PM 0 How to add SipAddHeader in outgoing call file.
1:11PM 3 adding numbers in dialplan
12:41PM 14 IAX IP Phone
12:03PM 7 G729 codec
11:51AM 1 indications.conf entry for Iceland
11:26AM 2 how to cancel new recorded message from voicemail menu?
11:08AM 3 followme order field
10:10AM 5 Description of Zaptel/DAHDI E1 alarms
6:15AM 0 Asterisk and PhoneControl
Sunday January 18 2009
7:43PM 6 Using a sidecar? Ideas?
6:27PM 0 Asterisk T.38 Passthrough + T38Modem/Hylafax - has anyone had luck with this?
4:28PM 2 Recordin call in asterisk
3:57PM 0 BRI on Solaris/SPARC
12:37PM 3 DAHDI trouble (again) Unable to open master device '/dev/zap/ctl'
12:19PM 0 is multiple contexts in alsa.conf possible
4:48AM 2 caller ID - handle_request_invite: Failed to authenticate user
2:07AM 0 ast_yyerror()
Saturday January 17 2009
10:06PM 1 canreinvite per route
9:39PM 1 Sip Trunk registration
7:40PM 0 asterisk support for multiply (two) console dsp devices
6:52PM 6 Call file in the future
4:51PM 3 Asterisk 1.6 T38 to G711 transcoding is this possible?
11:37AM 3 compare Linksys SPA8000 and Grandstream GXW4008
Friday January 16 2009
10:47PM 3 UpdateConfig : Appending line fails
8:26PM 0 Asterisk 1.4.23-rc4 Now Available
8:14PM 0 Crickets. Yes, crickets.
8:05PM 3 ATA gateway with 2 ethernet interfaces
7:58PM 0 about hardware
6:53PM 3 mini-PCI FXS card?
6:12PM 2 CDR problems -- two call legs create only one CDR. Using ForkCDR() not even working.
6:07PM 1 Voicemail message is dialtone
5:42PM 3 How to hangup a call manually...
5:21PM 1 Dialing from E1/T1
4:38PM 4 Remote RTP
3:52PM 4 Snom 300 vs Grandstream gxp
3:49PM 19 pstn hangs up: MWI no message waiting ??
12:27PM 0 Can not fetch SIP_HEADER incase of Transfer
12:17PM 0 dialing trunk-to-trunk
12:13PM 0 dialing trunk to trunk
11:13AM 2 want to add SipAddHeader in call out file
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3:56AM 2 CRTC and FCC Feeds
1:25AM 7 Asterisk Upgrade
1:00AM 8 Portech MV-378 with Asterisk
12:33AM 0 ISDN and routers...
12:29AM 0 gtalk and jingle again...
Thursday January 15 2009
11:56PM 1 multiple registration to sip trunking provider.
11:15PM 8 How to transfer a call from one Asterisk Server to another
10:29PM 1 Broadcast Phone system (for radio)
8:45PM 7 how to debug mime-construct with fax2mail?
7:52PM 0 Voicetronix Openswitch 12 + echo problem
7:42PM 0 Warning in CLI: Inringing for peer [PEER] < 0
7:40PM 0 Up To 20% OFF At Our Signature Style Event + Holiday Weekend Clearance
7:00PM 15 Asterisk - Trixbox
6:02PM 1 Patton SmartNode 4638 and ISDN2e
5:11PM 10 Digium TE220 supported protocol
3:34PM 2 problem with PlayDTMF: no error but no tone
2:44PM 4 R2
11:11AM 25 Call Stealing
4:09AM 1 call transfer in CDR
2:31AM 3 Has anyone used FaxGateway()
1:13AM 2 OT - Differences between modprobe and insmod
12:18AM 2 Dropping this SIP message, it's incomplete
Wednesday January 14 2009
10:55PM 5 Zap problems
10:30PM 2 1.6.1-b4: Can't get fax2mail work from System()
10:25PM 0 AMI API , Editing extensions.conf
10:20PM 0 Nortel files for bankruptcy protection
6:41PM 0 sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms?
6:32PM 0 Strange IAX2 registration issue
6:04PM 0 agi and set variable ( accountcode in aserisk 1.4)
5:11PM 4 G.729.1 - any interest?
1:02PM 32 evaluate SIP response codes in dialplan
9:33AM 3 gxp2000 and no sound asterisk 1.6
9:18AM 2 Any free video (or audio) softphone VOIP client under Linux with touchscreen friendly interface ?
8:24AM 4 Set caller ID to anonymous
Tuesday January 13 2009
11:58PM 2 FWD and IPCall
10:36PM 1 Asterisk Appliance
9:10PM 11 FWD and Asterisk
9:08PM 0 test
6:44PM 0 Problem with overhead paging with Alsa and OSS
6:03PM 8 mISDN BRI Asterisk 1.4
5:40PM 13 0800 UK number
4:12PM 0 [Re: CDR Rewrite -- Questions to the users]
2:04PM 6 404 not found from one ip-adress
1:31PM 6 Zaptel & multiple kernels
1:28PM 0 Realtime MOH
9:20AM 17 Dahdi caused Kernel to segfault
8:29AM 21 What are the various models of DID providers
4:55AM 1 cli reload error
12:05AM 4 Beware of DIDX & Super Technologies
Monday January 12 2009
10:28PM 2 FXS Help Needed...
9:15PM 1 u-law file header ?
7:26PM 4 Upgrade to v.1.2.31 ... weird change
6:51PM 3 CDR Rewrite -- Questions to the users (Steve Murphy)
5:15PM 0 WCTDM/Zaptel memory leak
5:03PM 5 a zaptel problem
4:53PM 4 bug 14153 and svn checkout.
3:51PM 9 CDR Rewrite -- Questions to the users
1:54PM 3 bug(?) bandwidth problem
1:05PM 0 Transfer in Asterisk 1.6
12:39PM 2 problem with dahdi and meetme
12:34PM 2 error messgae
7:45AM 0 no busy here
5:32AM 1 RTCP SR transmission error, rtcp halted
Sunday January 11 2009
7:17PM 4 sip peer permit/deny - Need some explanation
1:09PM 2 asterisk 1.4 with h323 for debian
12:06PM 8 chan_sip on non-standard port 5062 - contact has no port
5:35AM 3 Use ZAP/Dahdi channel for outbound only... no inbound?
12:33AM 2 Configuring Linksys spa8000 devices through xml
12:01AM 2 hdmi an console dsp
Saturday January 10 2009
7:44PM 8 How to monitor asterisk with SNMP?
6:13PM 3 Pay Phone Controller Project
12:43PM 0 line disconnected after 20 seconds no reply to our critical packet
9:53AM 1 Cisco VoIP QOS
9:36AM 6 Local channel Help required
2:36AM 3 Asterisk/GXW410x IP Analog Gateway
12:52AM 4 Lenny. Where to find zaptel patches
Friday January 9 2009
9:33PM 13 Spurious hangups on Sangoma A102d, Trixbox 2.6.1
9:05PM 9 Security communication dilemma: your help needed
8:57PM 3 fake ringback tone
8:08PM 1 Web Softphone
7:20PM 0 Fw: iax2 bindaddress: how to reload so iax2 can bind to an alias IP
4:36PM 7 lock SIP Account after too many failed logins
3:33PM 2 Queues, SIP channel and "In Use"
12:40PM 0 AmooCon - Call for Papers
12:27PM 1 slow ODBC reconnect
11:36AM 0 Asterisk does not reREGISTER in case of failure
8:22AM 6 iax2 bindaddress: how to reload so iax2 can bind to an alias IP
8:15AM 1 Friday Jan 9th at Noon ET: VoicePHP from TringMe
Thursday January 8 2009
11:07PM 15 Not Dialing 9
10:14PM 4 how many quad T1 cards
9:15PM 3 Playing MP3s...
7:28PM 0 AST-2009-001: Information leak in IAX2 authentication
7:06PM 0 Asterisk 1.2.31,, and released
6:24PM 5 AEL question: testing channel variables
5:44PM 8 Could you compile mISDN 1.1.8 on Lenny ?
5:41PM 1 Executive Assistant Guidance
5:35PM 0 Attended transfer problems
4:03PM 0 console/dsp with digital sound
3:56PM 2 Goto Question
3:17PM 0 SIP message routed back to mysql
2:21PM 2 SIP "peer" with different username/password for incoming and outgoing
2:06PM 8 Problem incomming from openser
1:28PM 3 AEL and };
12:12PM 1 is it possible to store vmsecrets outside of users.conf?
11:02AM 0 mISDN & Numeris Signaling (2 channels for 1 call)
10:00AM 2 Macro arguments seperator
2:50AM 0 dahdi_dummy only compile
Wednesday January 7 2009
9:53PM 1 rejected because extension not found
9:19PM 9 SLA and Polycom
9:09PM 2 How to listen in on a SIP channel?
5:19PM 5 mISDN compile problem
5:15PM 3 Are mISDN mailinglists active ?
4:38PM 0 Chan_alsa stops working on 1.4.22
4:30PM 7 recommendation for German sound files
2:59PM 2 1.6
2:59PM 3 How to use AMD "Answering Machine Detect" ?
1:52PM 3 CISCO 7940 United_States/7960-tones.xml
10:47AM 3 app_rxfax and app_txfax with Ubuntu?
8:25AM 9 [Asus Eee PC 900] as replacement for legacy BRI phone
6:36AM 2 \iaxclient-2.0.2 compile problem
Tuesday January 6 2009
10:16PM 0 If you use Realtime Extensions... READ THIS...
9:41PM 2 Asterisk 1.6 and LUA
7:59PM 6 Queue
6:33PM 12 any SIP client for BlackBerry?
3:53PM 24 Simple CDRs
3:35PM 1 .call file not updating MySQL CDR's
3:22PM 4 [Asus Eeebox] USB FXO adapter?
3:19PM 0 Fwd: A2billing Multiple Servers
3:06PM 1 Call transfer using agi
1:18PM 3 enabling silence suppression in asterisk
12:32PM 0 Asterisk Generating NetworkOOO (ISDN Cause Code 38)
12:21PM 1 "username mismatch, have <x>, digest has <y>"
10:26AM 6 Asterisk CLI got freezed!!
10:21AM 0 G.729 VAD issue
8:28AM 4 Problems getting 1.6 to run with user asterisk and group asterisk
8:02AM 1 R2D2 VOIP Kubuntu 8.4 Ekiga, voice conference
6:11AM 9 bridge 2 calls
4:11AM 0 chan_sccp and CISCO CP-7914 Module
12:24AM 11 Incoming side of SIP trunk does not work unless I add "insecure=very"
Monday January 5 2009
10:12PM 1 queue log parser
8:05PM 8 Agents, Queues and logon/logoff
7:27PM 4 CDR - What Changed?
5:40PM 1 cdr_addon_mysql 'Failed to insert into database' stops * call processing
11:36AM 0 G729 VAD issue
11:04AM 3 B410p, Ast1.4, France Télecom Numeris Double T0 problem
Sunday January 4 2009
8:42PM 4 queue log in mysql
8:12AM 7 Bad Audio : linux-2.6.24-etchnhalf.1-686 / asterisk-1.2.27 / zaptel-1.2.24 / libpri-1.2.5 / HP ML110 G5
4:12AM 2 Bring India together
Saturday January 3 2009
4:05PM 4 OSLEC
3:50PM 5 snom 320's creating inappropriate conference calls
11:18AM 0 BerkeleyTIP TODAY Jan 3 Sat- Party Time :) Video Talks: Asterisk, GPU
11:16AM 0 BerkeleyTIP - Hello, Introduction, Monthly Global GNU(Linux) BSD Free SW HW meeting
11:12AM 0 Hello Asterisk list. BerkeleyTIP & you
Friday January 2 2009
4:19PM 1 SIP URI: Allison Smith, Music-on-Hold Parody--outstanding.
3:37PM 2 Deprecated Realtime application, what's to be gained ???
1:21PM 13 Setting Periodic-Announce filename in the dialplan
9:26AM 6 2008 Post Count
4:41AM 0 Audiocodes MP-11X configuration to work with Asterisk
Thursday January 1 2009
11:07AM 0 DISA and the # key
7:28AM 2 New box, reload command takes 1 min.
12:07AM 10 Allison Smith, Music-on-Hold Parody--outstanding.