asterisk users - Jan 2008

Thursday January 31 2008
TimeRepliesSubject
11:37PM 0 Asterisk 1.4.18-rc4 Now Available
9:58PM 0 alcatel omnipcx
9:10PM 0 Problem picking up a call with PickUpChan or PickUp
7:25PM 0 FS: A20101D Sangoma Board 2 Port FXO 2 Port FXS w/ Echo Can
6:45PM 4 Bunch of set-up/usage questions (SLA, MWI, SMS proxy's, crypto, Fax, etc)
4:55PM 6 VoIP Users Conference Friday Feb 1st @ 12 Noon EST: Hosted IVR
4:46PM 1 Default delay time for Attended call
3:58PM 0 hint is hanging when remote party ends call on hold
3:09PM 3 Analog Adapters ?
11:45AM 1 Dropped calls
11:36AM 1 createlink with out agents in 1.4
10:24AM 1 Could not find ooh323.conf
10:09AM 0 OT - SIP phones supporting LLDP-Med
9:59AM 1 Incoming call from SIP proxy to asterisk
9:46AM 0 Realtime device update weirdness
7:35AM 2 Server Compatibility List for Asterisk
5:30AM 16 pulling my hair out over voicemail
5:08AM 2 How to get called number in featuremap
3:10AM 5 CallerID shows wrong values in manager interface
 
Wednesday January 30 2008
TimeRepliesSubject
9:51PM 1 Default delay time for Attended call transfer
8:55PM 2 G729 version to be downloaded for my machines
8:18PM 0 calls get stuck in the asterisk box
8:13PM 0 OT - Looking for used 2 FXS port pci card
7:52PM 0 Asterisk 1.4.18-rc3 Now Available
7:32PM 0 conf meetme all exited and still active
7:21PM 2 Unicall CRN 32769 - far disconnected cause=Switching equipment congestion [42]
6:13PM 1 Parking lot
4:38PM 4 func_odbc - trouble
4:23PM 11 Meetme voice quality problems
2:44PM 1 Using two SIP-Domains with asterisk
2:32PM 0 No audio one way
12:38PM 6 Can't read environment variable
12:09PM 1 Facing problem in installing asterisk-addons
7:21AM 11 Problem with DTMF dialing
3:34AM 2 sipsock_read: BAD! BAD! BAD!
2:34AM 4 asterisk gateway
1:21AM 0 Queue works with across server agent?
1:20AM 1 Queue - ${ANSWEREDTIME}
 
Tuesday January 29 2008
TimeRepliesSubject
10:51PM 6 chanspy does not pull the call back to asterisk after a reinvite
10:31PM 6 Source Based Call Routing
9:15PM 0 ShoreTel <-> Asterisk Integration
8:18PM 2 Queue member add
7:24PM 1 speex, ilbc and g729 codecs
7:11PM 2 When does Asterisk "REFER"?
6:59PM 2 POE draw on Aastra 480i
4:51PM 1 softmodems bank for ast.
4:29PM 2 Asterisk 1.4.18-rc2 Now Available
3:15PM 1 codec_g729a.so problem...
2:32PM 2 SET with pipe symbol
2:11PM 1 test please ignore
11:08AM 1 PRI Alarms, Comes Back, But Asterisk Won't Touch It!
10:03AM 2 Do Asterisk requires audio codec to be installed?
6:46AM 0 Installation of gatekeeper-H323plus
6:24AM 11 Asterisk mem leak behavior?
4:56AM 17 Asterisk's DANGEROUS Transfer CDR's
4:12AM 2 Dialogic card
2:59AM 0 Asterisk and MRTG, a little help please...WORKING
12:31AM 0 Asterisk 1.6.0-beta2 and 1.4.18-rc1 Now Available
12:03AM 2 MFC/R2
 
Monday January 28 2008
TimeRepliesSubject
11:38PM 3 IAX Calls - One Way Audio
11:05PM 4 SIP DTMF Troubleshoot
10:57PM 0 ISDN Internal Bus?
6:37PM 4 zaptel and oslec
6:07PM 3 Shut down one Zap line
12:36PM 3 Dial agent channel - busy
12:17PM 0 Monitoring audio on channel?
12:00PM 1 unable to hear voice with asterisk 1.4.15
10:51AM 0 mwi with sip
7:08AM 0 [AGI 1.4] Why doesn't Asterisk complain?
5:12AM 3 Using x-lite -Call failed 404 not found
 
Sunday January 27 2008
TimeRepliesSubject
11:28PM 1 Toll-Free setup on Asterisk Server
9:36PM 7 Asterisk and MRTG, a little help please...
4:27PM 13 Maybe a little OT---USB Handset
1:48PM 6 Best "Console" phone?
8:18AM 1 rxfax does not work (anymore)
6:57AM 4 [AGI 1.4] C sample?
3:41AM 2 Jabber and Asterisk?
 
Saturday January 26 2008
TimeRepliesSubject
9:28PM 2 Loosing IAX/SIP user's registration with asterisk as no-root
9:26PM 0 Loosing IAX/SIP user's registration
7:09PM 8 autoprovision 200+ linksys phones setup
4:29PM 3 GotoIf() on Auto-Attendant
3:43PM 0 Upgrade fails, need system upgrade advice
1:55PM 3 CHANUNAVAIL
11:20AM 4 Asterisk on Dell PowerEdge 2950
8:49AM 0 Extension Mobility with Asterisk and Cisco 79x1 phones
7:07AM 0 Avaya 9620 phone using Firmware 2.0.1.34 has working MWI lamp
5:00AM 0 Provide a proper link to download Libpri-1.4.3
2:03AM 1 Zaptel for 1.6-beta1
 
Friday January 25 2008
TimeRepliesSubject
8:37PM 1 Join me on Last.fm!
7:53PM 0 Script for seeding polycom phones with an extension directory
6:41PM 2 Problem with FollowMe
5:50PM 0 adding additional volume to console/dsp
4:07PM 3 Intercepting DTMF to initiate Voice Drop
2:40PM 1 Home use of asterisk
1:33PM 1 Disable IAX2 call path optimization
12:03PM 0 Adaptive jitterbuffer problem
11:35AM 4 Unprovisioned 7961
11:26AM 2 Asterisk Billing
11:04AM 0 Error in sip channel when asterisk created call (SIP invite request) is forked
10:48AM 2 Need sample configuration files for sipura/linksys ata
9:50AM 0 What kind of configuration do I need to run Asterisk ?
9:42AM 0 Share accounts several AOR
8:47AM 2 Maximum Paging Group Size?
3:54AM 11 Finding difficulty in installing Asterisk
3:06AM 0 Gentilini, Paul is out of the office.
12:35AM 2 SPA3000 -- PSTN to VoIP
 
Thursday January 24 2008
TimeRepliesSubject
10:38PM 0 dial extension number
9:28PM 1 Patton SmartNode Help
6:43PM 5 Help: dtmf mode
7:22AM 1 two zaptel card
4:57AM 12 Your "favorite" Asterisk application.
 
Wednesday January 23 2008
TimeRepliesSubject
11:38PM 4 Call Parking with multiple lots
11:07PM 0 nokia e51 (Christian Lox)
10:00PM 8 Replacement for Allison
9:04PM 5 Snom 320 Lost Settings
7:40PM 0 app_txfax
6:23PM 15 Peak number of calls?
5:55PM 1 LDAP support
5:23PM 20 Asterisk scalability
4:53PM 9 asterisk optimalization
4:31PM 0 No more audio with 99777 SVN version in certain case
9:23AM 5 Realtime problem host='dynamic' in 1.2.26.1
8:39AM 2 Modem bridging on Asterisk (no VoIP involved)
8:03AM 2 AsteriskIdeas.org :: Comment on submitted ideas
 
Tuesday January 22 2008
TimeRepliesSubject
10:36PM 3 Voicemail - is it possible to automatically use the extension being dialed from?
6:56PM 0 I am looking for an Asterisk subcontractor in New York City.
6:37PM 0 chan_sip deadlocks after some time
6:28PM 1 Echo in the outside call (E1)
6:04PM 2 AgentLogin by console
5:50PM 7 Difference between Asterisk and FreeSwitch
4:25PM 2 Followme
12:46PM 0 Caller id issue and Dial tone for sip phone on zero dialing
12:41PM 9 Free IAX / SIP Softphone with attended transfer
11:11AM 2 Custom Pickup and Transfer dial string
9:29AM 1 Discover Asterisk 1.4 :: Jitterbug, no, Jitterbuffers
9:02AM 0 So is anyone using 1.6 beta? What's the word?
8:29AM 0 Conference Hangup
8:20AM 5 TDM800P FXO problem incomming call
4:06AM 2 Asterisk crashed..
3:54AM 4 Polycom-SIP response 500
 
Monday January 21 2008
TimeRepliesSubject
10:56PM 2 FXS damaged at TDM22B
4:59PM 0 Aastra IP phone configuration generator
2:39PM 0 [asterisk-dev] Rgd Zaptel code for Asterisk
2:12PM 1 x100p & Asterisk 1.4.17 and Zaptel 1.4.7
2:04PM 1 Monitoring calls on demand
1:41PM 0 MGCP Thomson, "early" transmit problem
1:18PM 1 call on hold--hokk flash---i want to know if i can disable it
1:08PM 4 Loop Break
12:07PM 1 Astmanproxy
12:05PM 3 How to prevent logging of some entries in CDR
12:01PM 2 Qsig link
10:06AM 6 [Fwd: Re: Large issue - having trouble diagnosing.]
8:41AM 0 calls get stuck in asterisk
7:10AM 2 asterisk-addons-1.6.0-beta1---Error
4:04AM 5 Large issue - having trouble diagnosing.
3:39AM 1 blf and misdn
2:05AM 1 Polycom 320 Issue
1:14AM 19 I am having a problem connecting my X-Lite to my Asterix box
 
Sunday January 20 2008
TimeRepliesSubject
9:39PM 0 HT-488 tutorial
4:41PM 0 30 sec delay before voice is heard
4:10PM 2 Asterisk connect to Cisco As5400 gateway
3:40PM 11 SIP <> GSM
11:01AM 4 IAX and NAT Transparency
10:46AM 8 IAX softphone
9:33AM 0 Paging and conferences/chan_alsa.
8:57AM 5 IP Phone support SIP and IAX
2:32AM 23 Calls Being Randomly Bridged
12:28AM 2 SIPAddHeader in .call file
 
Saturday January 19 2008
TimeRepliesSubject
11:10PM 0 nokia e51
10:34AM 0 Call-out campaign variable problem
9:21AM 9 Nightly tarballs, would you use them?
4:06AM 7 New Polycom Provisioning Tool Released with BugFix
12:47AM 0 AMIProxyPal - AMI Proxy Project
 
Friday January 18 2008
TimeRepliesSubject
11:58PM 0 Asterisk 1.6.0-beta1 released
10:41PM 1 dtmf from Cell phones
10:26PM 0 asterisk chan_sip tuning
8:57PM 2 Looking for business-grade SIP Softphone
8:37PM 1 Probably a simple question. Dial a call.
8:13PM 4 SAY TIME + PHPAGI + Timezone
7:37PM 0 Maximum retries/no reply to our critical packet
4:00PM 3 OT: Call for beta testers (well... perhaps late Alpha).
3:41PM 3 R2-Unicall Asterisk as CPE and as CO
2:33PM 5 Accessing a MySQL database and using the same db for cdr
1:32PM 0 Asterisk and postgresql query
12:29PM 0 OT: To Admins: Missing DNS for list server
12:21PM 0 Advice on AMI and SIP (was: SIP)
12:20PM 0 SIP
12:14PM 3 Automatic call-out problem
11:38AM 0 Upgrading to Asterisk 1.4 :: Avoiding the hidden traps
10:39AM 0 VoIP Users Conference today at 1PM Friday EST
9:27AM 2 caller id issue for INDIA
3:28AM 0 Polycom Remotely Cancel Call Forward
1:18AM 4 Linksys PAP2 NA
1:16AM 0 Cisco 7910 Handsets: Skinny protocol?
 
Thursday January 17 2008
TimeRepliesSubject
9:32PM 0 IAX Trunk between two Asterisks
8:55PM 0 not understanding Cisco call manager connection for incoming calls
8:37PM 0 Paging Recording File
8:15PM 3 buffer-issue when piping live-streams into musiconhold
8:00PM 44 asterisk-1.2.26.tar.gz Thoughts?
7:51PM 0 PostgreSQL query results truncated 255 characters
7:28PM 2 SIP Proxy Issues
6:54PM 1 More voicemail cards needed...
5:47PM 0 Voicemail Callback
5:36PM 5 Device state of SIP doesn't change
4:46PM 0 Asterisk SVN mirror back up to date
4:41PM 3 Iax Encryption
4:38PM 3 modem through Zaptel/Digium?
4:25PM 0 sip channel - redirection - which context is used
3:33PM 0 Channels ID / Soft Hang Up
2:10PM 6 AEL includes?
1:09PM 4 Zaptel timing on TE405P
12:13PM 0 Asterisk Meetme & MeetMeAdmin cmd info-use
11:25AM 7 Asterisk desktop tools for OS X
10:40AM 0 callerid on atxfer
10:23AM 5 Single T1 with DIDs
6:26AM 0 Incoming calls on PSTN trunk not disconnected (bsnl, india)
6:19AM 0 FXO Module for the cPCI platform
5:33AM 1 asterisk-users Digest, Vol 42, Issue 51
4:27AM 8 Voicemail systems- flow charts, digit/key cards, etc
1:34AM 0 Asterisk on ClarkConnect
12:54AM 3 IMAP client in asterisk not trying to contact IMAP server
12:42AM 2 Problem with a channel
12:39AM 6 Anyone Using a Dell PowerEdge T105 in Production
12:11AM 1 AddQueueMember and Flash Operator Panel
 
Wednesday January 16 2008
TimeRepliesSubject
11:52PM 10 HDLC errors
8:24PM 1 Asterisk 1.4.17 and RXFAX via T38
7:11PM 2 asterisk to mysql database!
5:25PM 7 [IAX] Up-to-date list of soft- and hardphones?
4:02PM 3 Can DB() use SQLite instead of BerkeleyDB?
3:25PM 2 Zap Issues
2:42PM 2 Voicemail consultation problem
2:29PM 0 Problem with TDM400P
2:06PM 1 Asterisk Now Beta 6 and CISCO IP 7910
1:47PM 1 Does host accept dns or ddns?
1:46PM 1 IAX Trunk between two Asterisks: Authority, and Call Rejected
1:42PM 1 Backup Route
12:11PM 0 Dualphone "LAN" SIP/DECT phones
11:51AM 1 Unable to dial _99XXXXXXXX
10:55AM 7 Unable to open master device '/dev/zap/ctl'
10:39AM 4 Difference between TE121 and TE122
10:18AM 1 bad sound quality after Redirect
8:18AM 3 volume problem
8:00AM 1 SVN Server Issue?
6:44AM 0 help Unable to dial _99XXXXXXXX
 
Tuesday January 15 2008
TimeRepliesSubject
10:20PM 1 Channel fallback
9:54PM 7 WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101'
8:32PM 2 Attended transfers manager or phone
8:12PM 1 inbound Audio problems probably not NAT related?
7:00PM 1 asterisk 1.4 context
6:55PM 1 Record calls then send them to users voicemail
6:42PM 1 Heartbeat
5:32PM 4 Interrupt the swift text
3:41PM 14 Discover Asterisk 1.4 :: SIP Subscriptions
2:40PM 0 sip channel error - extension pattern matching problem
1:43PM 0 busy/congestion random
1:14PM 5 cisco ip phne 7911G with asterisk
1:06PM 1 Playing DTMF tones down a channel
1:01PM 6 SIP Reason
11:42AM 1 Fax machine detect
11:09AM 1 Console app
9:53AM 3 Meetme recording
7:17AM 0 pickupchan without bristuffed version?
5:02AM 2 Park() help, extension not heard
12:03AM 0 Zaptel 1.2.23 and 1.4.8 released
 
Monday January 14 2008
TimeRepliesSubject
11:59PM 0 SVN servers down for maintenance
11:23PM 3 Asterisk 1.4.17 crashing more
10:18PM 5 CID blocking ...
10:09PM 18 G.729 pre-compiled binaries and Asterisk 1.2.x.
10:04PM 6 app_voicemail for spanish
8:43PM 0 Transfer/Speed-Dial
5:52PM 6 Voicemail check
5:25PM 0 [asterisk-dev] Unstable releases lately
4:51PM 4 Verficar VoiceMail
4:09PM 8 Asterisk 1.4 Call Recording
3:47PM 0 Temporary Service - Dominican Republic DID
3:18PM 0 Meetme Record Format
3:08PM 0 Help needed for Fax2Email with Welltech FXO 3804
2:58PM 6 OT: reverse DNS error for lists.digium.com
2:15PM 1 Different ringing tones ...
12:44PM 1 Video Call and Asterisk
12:42PM 13 GSM SIM Cards and Digium, or GSM SIM Adaptor
12:04PM 2 g729 codec - simultaneous calls
11:55AM 3 AGISTATUS is SUCCESS even though my PHP script returned -1
8:47AM 2 State of the application chan_spy
8:38AM 1 [SOLVED + EXPLANATION]: Strange ISDN-problem with incoming calls out of the same city
5:42AM 0 Call parking
2:45AM 0 Aastra Venture
 
Sunday January 13 2008
TimeRepliesSubject
9:00PM 4 ProxyPal for AMI Proxy Development
5:33PM 4 problems with zaptel and Udev
5:17PM 2 Question about queues and the definition and agents
4:22PM 0 Adtran 750 and E&M Wink
2:20PM 0 Soundcard necessary on an asterisk server to get output of playback()?? -> Next step
12:11AM 2 Packet2Packet bridging occurring when not wanted
 
Saturday January 12 2008
TimeRepliesSubject
7:09PM 5 Asterisk RFC2833 to SIP INFO DTMF conversion erros.
12:49PM 1 ISDN channels not properly released after call
12:05PM 3 Perl-AGI process
10:50AM 2 My latest MFC/R2 update with asterisk-1.4.17
10:02AM 3 Discover Asterisk 1.4 :: Google Talk, XMPP and Jabber Integration!
5:11AM 3 MFC/R2 Signaling configuration
4:03AM 0 zaptel digit problem
 
Friday January 11 2008
TimeRepliesSubject
10:08PM 0 MONITOR CMD()
9:15PM 2 Fwd: Trying to build MFC/R2
5:54PM 0 Authenticate problems (Extensions)
5:50PM 0 Deadlock of asterisk on app_system
5:32PM 3 MRCP Asterisk Integration
4:48PM 0 Dialplan flow on device state change
4:38PM 3 Question about queues and the definition of agents
4:21PM 1 Soundcard necessary on an asterisk server toget output of playback()?? -> Next step
4:20PM 0 [OT] Call for speakers: BOB 2.0
1:41PM 1 Dealy while taking
1:04PM 0 Is rfc4662 (SIP Resource Lists notification) support planned ?
12:22PM 0 15% Off from New Cyber-Telecom.net Website
12:15PM 0 Lest we forget: Friday 12 Noon EST - VoIP Users Conference
11:57AM 1 Developing Help
10:32AM 0 OT - Where do most email2fax errors come from ?
9:56AM 1 interconnecting an asterisk server with an old alcatel PBX through a Digium B410P
5:55AM 3 PRI Down but zaptel lets calls through
5:27AM 5 Congestion/Forbidden issue with new carrier
 
Thursday January 10 2008
TimeRepliesSubject
11:46PM 1 Multiple fax extensions
10:52PM 1 Asterisk Realtime unixODBC timeout?
8:54PM 2 Sip calls drop one leg after about 2 minutes
3:48PM 4 WARNING[19164] chan_sip.c: Forbidden - wrong password on authentication for INVITE to '"unknown" <sip:unknown@xxx.xxx.xxx.xxx>
1:19PM 0 problem about TDM400P ringback detection
10:22AM 9 OT - Is handover included in DECT GAP ?
8:35AM 5 Using Asterisk as an Fax-Gateway for analog Fax devices
8:29AM 0 Kirk and asterisk
5:42AM 0 forward call intended for another domain
3:57AM 16 IEEE 802.1x capable sip phones
3:23AM 10 Asterisk 1.4 and ISDN-BRI support
2:33AM 11 OT: Traffic Shaping
12:37AM 3 Two Asterisk Boxes Playing Together
 
Wednesday January 9 2008
TimeRepliesSubject
11:28PM 0 FXOTUNE update
9:03PM 0 IAXy ringing
8:41PM 3 Polycom 550 IP SoundStation Fuzzy Voice Quality
8:06PM 0 Subscriptions, Firewalls, 489 "Bad Event" and Bug 7608
6:28PM 3 Intercom & Paging with Polycoms
6:09PM 7 WaitExten and Macros
5:32PM 0 Broken calls
3:38PM 3 Busy notification with call limiting by GROUP_COUNT()
3:15PM 3 Zaptel FXS Cards - Station Distance
3:11PM 0 [asterisk-dev] MixMonitor doesn't work right with SIP and Zap/Flash transfers
12:40PM 2 Newbie: confusion with the new FXO/FXS card
9:46AM 0 remote Snom 360: no voice passing thru
7:54AM 3 Set CDR userfield in a realtime dialplan
5:55AM 1 Register source port
5:01AM 0 conferencing help
1:46AM 1 Dialplan Recordings
1:31AM 2 Help! channel_find_deadlocked: Avoided initial deadlock for ...
1:19AM 0 txfax_exec: Transmission loop error
 
Tuesday January 8 2008
TimeRepliesSubject
10:26PM 10 Linksys SPA-9xx Audio Issues
10:19PM 0 get_data
9:45PM 0 debugging bluetooth communication using chan_mobile
9:37PM 3 CallerID Number incorrect in SIP packet
8:43PM 2 Simultaneous Callback?!
8:26PM 10 Is it possible to use spandsp and patton to do fax2mail ?
7:29PM 2 What's the best ztdummy?
6:06PM 11 [Zaptel] Checking that TDM card works?
1:52PM 6 Bugs??
1:11PM 0 Prevent Asterisk from rebuiling DTMF tones
12:46PM 0 Asterisk Nokia
12:23PM 2 Limiting number of simultaneous calls in E1 line
12:01PM 2 disable call waiting by default
12:00PM 0 communicating SMS messages in asterisk
11:31AM 0 chan_h323 and asterisk 1.2
9:11AM 0 no outoging calls with B410P
8:20AM 3 HPEC
6:41AM 1 Early media support for Asterisk behind NAT
6:08AM 6 Distorted audio over Eicon Diva Server BRI
5:19AM 3 app_rxfax.c and app_txxfax.c where?
4:41AM 2 help need
12:11AM 20 :POSSIBLE SPAM: conferencing help
 
Monday January 7 2008
TimeRepliesSubject
11:51PM 2 Background Noise Elimination
10:28PM 0 asterisk-users] Increase Volume - SIP
10:14PM 0 chan_mobile and W300i
8:04PM 1 GotoIf() help
7:21PM 5 asterisk CLI and no such command "stop"
7:02PM 0 [Asterisk 1.2 + TDM FXO] Incoming call not detected
6:59PM 2 Media gateways and video
4:06PM 0 service provider connection problem
3:48PM 2 pickup application failed
3:34PM 2 Multi-SPAN (4xE1) Zap Group (Outbound)
3:17PM 25 FWD and IPCall
2:26PM 21 How to check if a SIP phone is forwarded without ringing it ?
12:43PM 1 extension.conf with mysql
11:57AM 1 Strange migration problems from asterisk 1.2.13 to 1.4.10, dtmf related?
11:21AM 0 Presentation Restricted h.323-SIP issue
10:53AM 13 no outgoing calls with Digium B410P
8:34AM 2 Increase Volume - SIP
7:06AM 4 zaptel programming
2:14AM 5 Change Default Voicemail Message
 
Sunday January 6 2008
TimeRepliesSubject
8:02PM 0 Asterisk High Availability and Clustering
6:12PM 17 Which IP Phone is really the best?
10:12AM 0 New site for feature wish-list: Asteriskideas.org
9:20AM 1 [FreeBSD 6.2] Error compiling Zaptel from Ports?
 
Saturday January 5 2008
TimeRepliesSubject
11:20PM 0 Zap with SIP
10:55PM 0 Newbie Q: Good link to configuring NAT with Sipura ATA's & hardphones
7:50PM 8 asterisk on Hp servers
6:36PM 13 Detailed Instructions
10:40AM 13 iP0020 Phone busy signal all the time.
10:04AM 4 G729A Install Problems
9:45AM 3 ASTERISK cd-rom
5:45AM 5 how to block spammer calls
12:06AM 1 G723 Codec and Asterisk
12:03AM 1 GotoIf: OR, AND
 
Friday January 4 2008
TimeRepliesSubject
11:52PM 2 Conditional Dial
11:10PM 1 asterisk as sip server
10:45PM 5 b2bua
10:26PM 3 Remote hold on PRI
9:48PM 1 VOIP Provider wooes
9:41PM 5 Cisco 79xx XML services
6:52PM 1 Polycom IP4000 - Device does not match ACL
3:28PM 0 2 firewalls, different INVITES
3:00PM 3 x100p wcfxo hangup on outgoing calss
2:48PM 2 Asterisk content @ OSCON 2008?
10:48AM 7 Mark Spencer and guest(s) LIVE today at 12 Noon EST - 11 Central - 17:00 UTC
9:58AM 4 Unable to forward call on SIP channel after SIP response 302 Moved Temporarily
8:11AM 20 Cisco 7941G-GE with Asterisk and CTPSEP odyssee
7:35AM 4 Agents and AddQueueMember
5:45AM 1 Registration from sip failed for ACL error (permit/deny)
2:19AM 2 automatic call marking an extension
1:36AM 7 Using Asterisc for Taking Calls for Radio
 
Thursday January 3 2008
TimeRepliesSubject
9:08PM 9 OT - GEOPRIV and location based SIP services
8:39PM 4 1.4.17 - Breaks park announce?
8:15PM 3 Unable to retrieve my voice mail ... (password incorrect)
7:44PM 5 A thougt
5:31PM 1 Bad Link on Website...
4:52PM 3 Right timing for a queue call
3:08PM 18 HFC-S zap channels always busy
1:28PM 10 GSM Gateway behind SIP ATA?
 
Wednesday January 2 2008
TimeRepliesSubject
11:10PM 5 is Power fail transfer possible with asterisk?
9:57PM 0 AST-2008-001: Crash from transfer using BYE with Also header
9:39PM 0 Asterisk 1.4.17 Released
8:11PM 7 Missing "zap" command in Asterisk 1.4.16
8:00PM 1 How to stop the update of astdb?
7:41PM 6 1.4.?? or ZapTel 1.4.X DIGITAL Calls are Broken
7:14PM 0 Toroid: open source middleware
5:34PM 3 Polycom VLAN
4:49PM 3 AGI stream file
4:20PM 9 Incoming Calls
3:19PM 1 Asterisk E1/T1 Card configuration
1:38PM 13 Lamps on Snom phones
11:27AM 4 Invalid extensions
10:45AM 8 Asterisk dialplan date and time operations
10:29AM 2 auto dial and IVR
7:33AM 19 Two Asterisks behind NAT and need to link them using IAX trunk
6:23AM 3 Trixbox and mail2fax
2:06AM 1 Password protect a queue from callers?
 
Tuesday January 1 2008
TimeRepliesSubject
6:24PM 11 zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init
5:08PM 3 With rtcachefriends=yes, when do realtime changes take effect?
4:33PM 2 (no subject)
3:38PM 0 Asterisk + SIP + cisco phone confrance problem
10:06AM 16 [1.4 + FreeBSD 6.2] Playing WAV PCM file?