asterisk users - Apr 2008

Wednesday April 30 2008
10:55PM 0 Zaptel Channel Numbering
9:39PM 2 StatusComplete is getting me sick !!
7:59PM 0 Sobre retencion de liquidos... Interesante...
7:07PM 21 Discover connected Zap lines
6:00PM 11 Zaptel Compatibility
5:27PM 1 Connecting Analog to SIP gateway to asterisk server
5:08PM 4 One way audio...
5:06PM 0 AVAYA 8300 integration with asterisk 1.2.x
4:57PM 0 OT: anyone have a Jitterbuffer patch for Asterisk 1.2.28?
4:48PM 0 Queue setup and phones ringing
4:42PM 2 one way audio after call transfer
4:33PM 4 Customize Music On Hold
4:01PM 6 Asterisk - CRM Integration
3:56PM 6 Shared Line Appearance
2:34PM 0 Asterisk 1.4.17 and ExtensionStatus
2:08PM 0 asterisk-users Digest, Vol 45, Issue 85
1:36PM 0 Please Make Room on Your Hard Drive
12:22PM 3 Zap channels on TDM400P report to be unimplemented after upgrade from Asterisk 1.2 to 1.4 - (cause 66 - Channel not implemented error)
12:11PM 7 Asterisk on Xen or Dedicated
9:34AM 3 DTMF Issues
5:20AM 3 Portability in Asterisk
4:49AM 6 Sending caller name out PRI?
2:13AM 0 Jitter buffer not used in SIP -> chan_local -> ZAP path even with /nj for local channels
Tuesday April 29 2008
7:31PM 0 ooh323 asterisk 1.2.x
6:28PM 1 AJAM event subscription - Was: func_odbc creating records or best practice
3:51PM 0 Odd Zaptel Issue - Strange State 6?
2:49PM 4 Outbound international calls over BT ISDN30
2:49PM 1 Debugging DTMF
1:48PM 1 need a monitor for asterisk
12:49PM 0 Digium TE420B (Four Port E1 PRI PCI-E x1 with Echo Cancellation), URGENT
12:36PM 6 RAS with Asterisk and PRI
10:47AM 0 changing of ssrc between early-media and call media
10:30AM 2 Annoying Sipura problem?
9:13AM 0 PRI CallerID - leading zero added
8:25AM 0 Polycom 320 working in direct mode, characters dialing and timeout
7:02AM 0 AddQueueMember() and PersistentMembers
6:09AM 0 Asterisk V1.6.0 SVN debug WARNING(6830) a bug or deliberate?
12:49AM 3 Anyone have pricing on the Color Polycom Phone?
Monday April 28 2008
10:30PM 9 func_odbc creating records or best practice
10:03PM 7 OT: Polycom 3.0
6:48PM 0 Question on bridging
6:09PM 0 misdn, no free channels, similar to FAQ one
5:34PM 6 realtime queue callers
3:07PM 14 tftp issue
1:29PM 6 PRI hangup certain outgoing calls
1:21PM 8 Dell 1950
12:46PM 5 (no subject)
Sunday April 27 2008
3:41PM 15 zap not coming online on fedora 8
1:32PM 1 Sangoma A200 to A200D
12:17PM 12 Siemens Gigaset S685IP Review
11:55AM 0 Ploycom 320: characters and need to press DIAL if not registered
6:46AM 5 psql
Saturday April 26 2008
10:41PM 23 Manual Wardialer
6:08PM 1 [asterisk-dev] zap not coming online on fedora 8
4:31PM 13 yum install for specific kernel, how? And zaptel on fedora core 8
4:02PM 0 Channel variable settings
2:53PM 1 Outside call not coming through
1:32PM 5 Roaming callback?
1:01PM 0 how to retrieve sip tag from dialplan
12:11PM 5 Need comments on CRM development / Asterisk Customization
Friday April 25 2008
11:40PM 1 choopy audio when both side talk at the same time
9:55PM 3 Graphing Jitter Packet loss and Latency for SIP Calls
8:23PM 2 Asterisk using 100% of CPU
5:58PM 0 E-mail date is wrong
2:31PM 0 Friday Apr 25 @ 12 Noon EDT VoIP Users Conference
11:55AM 6 Upgrading to 1.4
11:31AM 3 Cisco 7960 odd behaviour ...
11:03AM 27 Asterisk for larg
9:33AM 3 Cisco to Asterisk migration
9:23AM 0 SIP response 400 on attended transfer
9:14AM 0 DNS Problems during zaptel upgrade
8:08AM 0 using "m" switch in dialplan
8:06AM 3 noisy analog lines
7:47AM 0 Play sounds to both caller and callee at the same time
6:23AM 13 Playback / Background / Read choppy, but musiconhold fine, even with ztdummy
1:57AM 1 followme scenarios
12:41AM 0 No DTMF on Sip Connection between two asterisk boxes?
Thursday April 24 2008
8:58PM 1 Full queue issues
8:48PM 3 ATA FXO / FXS - can forward to sip ?
7:43PM 3 GoToIfTime problem
7:37PM 3 No CallerID Transfer Problem
4:58PM 18 Digium B410P or Sangoma A502D?
4:51PM 7 IAX issues with
4:26PM 0 ring group question
4:02PM 8 No DTMF on Sip Trunk?
4:01PM 6 help...i cant do more...
3:50PM 3 Playing mp3-files – will it be OK?
2:47PM 1 Macro/Goto Help
1:30PM 1 T.38 VoIP providers
10:18AM 3 G723 pass thru
8:51AM 6 Forking in Dialplan
8:11AM 1 Newbie Polycom: Instant Messaging
7:28AM 9 MFC/R2 in chan_zap , Testers Wanted
6:43AM 0 Newbie Polycom: can Speed Dial display last name first?
Wednesday April 23 2008
2:54PM 0 cdr_pgsql and spool file
2:17PM 0 Asterisk SNOM and DTMF
2:11PM 0 ZapRAS, pppd plugin option
1:37PM 3 how to copy a variable without interpretation of the content
12:37PM 3 AEL: how to copy a variable without interpretation of the content
11:37AM 10 Next step in extensions.conf after answer the phone in Queue
8:08AM 0 Anyone using HKBN / 2b and have DTMF working?
5:59AM 9 prepaid on the trunks
3:08AM 0 question about asterisk setup...
Tuesday April 22 2008
11:05PM 2 Asterisk 1.2.28,, and 1.6.0-beta8 Released
10:58PM 8 AST-2008-006 - 3-way handshake in IAX2 incomplete
9:02PM 0 Can't transfer call
8:13PM 13 need examples of asterisk and mysql integration
7:43PM 0 OT - How to configure TAPI client-server ?
2:12PM 1 Cisco 7961 + 7914, speeddials, BLF & Asterisk 1.4?
1:47PM 0 Bip viol. in E1 cards
1:10PM 0 Cisco 79X1 speaker issue
12:41PM 3 Asterisk sends 486 Busy Here instead of 600 Busy Everywhere
11:56AM 3 WARNING: Remote host can't match request NOTIFY to call on Audiocodes MP-124 FXS
11:26AM 0 Check the answered channel in simultaneous sip call
11:10AM 44 Quality problems with ISDN PRI
10:28AM 2 lots of warnings from translate.c
10:22AM 2 features.conf Problem with DTMF_sequence
10:01AM 5 Parsing incoming extension till first @
9:11AM 2 OT: Linksys devices send incorrect REGISTER
8:55AM 8 Conditional include=> ?
7:58AM 0 Strange SIP packet
7:13AM 0 caller groups
3:17AM 12 Can I roll my own E911?
Monday April 21 2008
9:30PM 8 Disable transfer on all calls
7:29PM 8 Switch recommendation?
7:08PM 1 Monitor v/s MixMonitor
6:51PM 2 Phone notification?
6:47PM 2 Dual Interface config
3:41PM 5 Monitor not merging calls
1:34PM 0 Dialplan Visualization (Extensions.conf orDialplan Show)
1:31PM 4 Click-to-talk (Java application)
12:50PM 0 RTCP stats
12:14PM 1 UPDATED Asterisk Jingle Extensions.conf
11:33AM 0 Asterisk Jingle<->SIP GW Question
10:23AM 1 re-Invite post call establishment (for RTP bypass)
10:02AM 1 OT: UMA in UK, any use?
8:34AM 4 re-invite (bypass asterisk) post call establishment
8:20AM 0 API Originate - action on reject/busy/congestion
8:17AM 0 sip channel - detect ringing (nvlinedetect??)
7:58AM 4 Basic Possiblity Question.
6:53AM 1 Digium TDM410P Cards
2:38AM 0 echo on a B410p
Sunday April 20 2008
11:12PM 4 imaps - voicemail
8:23PM 2 Problems with Quality Voice in a Asterisk-E1-Unicall
8:11PM 1 Outbound PRI ISDN 30 problems
4:44PM 0 ISDN card freeze
11:20AM 8 chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9
Saturday April 19 2008
6:21PM 5 meetme with time condition
3:42PM 12 Error on load
Friday April 18 2008
10:56PM 1 Asterisk PBX using Outbound proxy
7:17PM 1 Polycom RTP port range
6:56PM 1 REGISTER Outboundproxy
6:53PM 4 Polycom LDAP Corporate Directory
6:48PM 5 Dialplan extension priorities
6:21PM 1 WAN and LAN connections
5:31PM 0 videoconferencing with asterisk issues
4:30PM 2 Question on groups
2:18PM 0 OT Nice IBM 1U Server Gets Along w/Old and New Digium Boards Cheap X305 $199
8:39AM 0 Friday @12Noon EDT: VoIP Users Conference on the Internet
8:28AM 32 Dialplan Visualization (Extensions.conf or Dialplan Show)
8:15AM 0 database connections question
8:13AM 6 Newbie Polycom: Subscription/Presence Problem
5:53AM 0 Guess I shoulda put a subject - sip diversionheader
4:25AM 6 SIP outboundproxy for asterisk
Thursday April 17 2008
10:19PM 0 (no subject)
8:47PM 2 multiple users collisions
6:15PM 2 End to end call monitoring?
5:14PM 2 users.conf and voicemail
4:14PM 14 G729 license count...
4:03PM 3 questions running 2 asterisk under the same LAN
4:01PM 0 status line header
3:08PM 2 sometimes UNREACHABLE!/REACHABLE doesn't appear in the log when it should
2:48PM 2 Differents routes for differents extensions
2:45PM 1 Sip or IAX device with professional balanced audio out
2:34PM 3 QOS for outgoing SIP ... Who needs QoS anyway!
12:31PM 0 FSX gateways
11:41AM 0 Constant ''CHANUNAVAIL' on PRI for Outgoing Only
11:03AM 10 buying cards from pakistan
10:31AM 2 call forking feature
8:59AM 3 Asterisk Warning 2512
8:59AM 2 imap voicemail
7:42AM 0 keep incoming codec same as outgoing on sip proxy
7:40AM 0 keep incoming codec same as outcoming on sip proxy
5:55AM 2 keep one line open
12:30AM 0 Asterisk and LVS
Wednesday April 16 2008
11:44PM 1 lightweight prepaid app using Dial and extentions.conf
10:49PM 16 QOS for outgoing SIP calls
8:25PM 2 extenspy and chanspy
7:51PM 2 chan_zap error 1.4.19 tone duration
6:27PM 16 Drag and Drop transfer application
6:20PM 1 Problem with hints (1.4.19)
5:51PM 11 Chanspy on Asterisk 1.4.19
5:33PM 3 Using Chanspy
5:25PM 4 PSTN to SIP
2:36PM 7 Best Click-to-call client
2:19PM 0 Callerid Error
1:41PM 10 DUNDi and SIP
1:40PM 3 Busy (congestion) signal and cell phones
1:10PM 1 Non-codec capabilities (dtmf): us - 0x1 (telephone-event),
12:57PM 0 Version FIOS MWI Detection - asterisk-1.6-beta7
12:56PM 2 Hangup conundrum with RxFAX
10:47AM 4 Simple queue announcements
7:53AM 2 Problem with B410P
4:19AM 0 [asterisk-announce] Zaptel 1.2.25 and 1.4.10
12:50AM 6 X-Lite and Presence?
12:30AM 3 wcfxo and X100P card won't play nice.
Tuesday April 15 2008
10:33PM 0 Good article about VoIP, etc.
9:46PM 1 Global call limit
8:51PM 6 CDR and transfers! :(
5:38PM 0 Problem with SIP, attended transfer and GROUP_COUNT
5:07PM 3 app_swift v1.6.1 released for Asterisk 1.6
3:54PM 4 PBX Console
3:52PM 0 asterisk online indicator
3:52PM 3 dialed number notify at invalid dial situation
3:06PM 0 [VOIP-Users-Conference] Re: Free FAX license from Pika
2:58PM 3 polycom 501 stopped working
2:43PM 0 Asterisk & Sys Admin in Chicago IL (full time)
2:19PM 3 gotoif syntax error
2:07PM 0 Asterisk on EC2
12:23PM 2 problem with Asterisk 1.4.19 - accountcode dissapearing
12:12PM 0 Lypp/37 Signals mashup contest
11:55AM 2 What kind of Specs for Conference server
11:08AM 0 Polycom phone reboots
10:06AM 5 SIP response 480 "Do Not Disturb"
8:22AM 1 voicemail odbc storage
7:10AM 0 Patch for call deflection with libpri
4:39AM 5 Conferencing..
Monday April 14 2008
11:20PM 3 Problem with SPA3000 -- dropping calls
10:54PM 0 CallerID in NZ
10:00PM 33 zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
9:39PM 15 Zap Codec
8:06PM 2 polycom auto answer
7:05PM 11 E911 Recommendations?
4:52PM 5 do cards just instantly go bad
2:19PM 1 sip.conf wont load completely
9:24AM 6 Recommend some good Click 2 Dial Application
9:02AM 6 Unable to load module
Sunday April 13 2008
5:40PM 4 compilation of asterisk 1.4.19 with ilbc already on system
3:46PM 2 Similar option as promiscredir to use in transfer (REFER)
4:11AM 0 Voicemail ODBC bug or feature?
3:34AM 1 cdr_custom outout to serial port
12:38AM 10 way to inquire status of T1 link
Saturday April 12 2008
8:27PM 0 How to automaticaly close calls when Asterisk didnt receive the bye request
8:00PM 4 problem TDM01B
3:43PM 0 ATA for Fax with BroadVoice?
1:02PM 0 PSTN gateway alternatives
2:43AM 3 X100M never goes on-hook state
12:32AM 0 No compatible codecs / static noise
Friday April 11 2008
11:25PM 11 NAT issue with Fortinet Firewall
10:32PM 8 Asterisk and the Mitel SX 200 integration
7:38PM 2 ZD Net article
5:57PM 1 Correlating queue_logs and cdr for abandoned calls
5:16PM 6 Asterisk temporary hangs when no internet connection
5:00PM 0 Need good voicemail documentation
4:59PM 0 Asterisk trunk/1.6 and nvfaxdetect
4:59PM 0 nvfaxdetect, nvvoicemail, and others
4:59PM 0 SIP KEEPALIVES, with QUALIFY and without making UNREACHABLE
3:05PM 0 Cisco 7905 / 7911G Reviews
2:56PM 0 problems in REFER request to a different machine
2:28PM 0 testing the list
2:23PM 1 manger hangup call
1:47PM 1 Friday April 11th @ 12 Noon EDT VoIP Users Conference
10:22AM 2 Strange CLI behaviour
6:40AM 5 OT - How to check HPET is on and working before installing Asterisk ?
6:03AM 8 bandwidth required for Asterisk running on T1
5:50AM 1 Loosing SIP registration.
5:48AM 6 tdm410p w/ echo - no full duplex
5:11AM 1 TDM400P Dialtone problem
4:39AM 0 Running asterisk + T1 + ztdummy on Debian vserver
3:36AM 0 Newbie Polycom: Will a big 0000-directory.xml crash the phone?
2:21AM 4 odd error compiling zaptel-1.4.10
Thursday April 10 2008
11:26PM 0 Any body tried MysqlPool-1.4
10:14PM 0 Excellent Paper on ECHO in VOIP Environment
8:48PM 5 Digium T1 Card Crashing Server (Dell 2950)
6:46PM 42 Is Asterisk really good??
6:41PM 1 Queue member state 'Not in use
5:14PM 2 Voicemail: afternoon audio file is missing
5:00PM 3 best way for call detail logging
3:24PM 7 Phantom Rings
1:09PM 2 [1.4.17] Disabled modules -> Zaptel stops picking up calls
11:13AM 0 A Simple Question
8:27AM 11 question about queue
8:25AM 0 followme than voicemail
8:24AM 0 Time out to disconnec: IAX trunk, SIP Trunk, Zaptel Channel
7:04AM 0 Festival in Italian does not work
6:55AM 6 Removing "Parsing /etc/asterisk/manager.conf" from CLI
4:50AM 0 Custom multilingual VM announcements & writing VM to 'default' context
3:45AM 0 Maximum Include level (10) exceeded- question
3:29AM 1 features on dial pad
Wednesday April 9 2008
10:18PM 2 multiple simultaneous access to single voice mail box
8:49PM 19 Jumped from 1.2.7 to 1.4.19, missing CLI colors
8:02PM 1 dial tree crawler?
7:12PM 9 setting dtmf mode for a particular peer
5:37PM 0 Magic Jack>>>> usable in Asterisk ?
4:39PM 2 zaptel 1.2.25 compilation error
4:05PM 2 Connecting Asterisk to Nortel Succession 4.0 sip...
3:56PM 3 [VOIP-Users-Conference] Potential subject for Friday - Does the Asterisk community need a 3rd party commercial software ecosystem?
3:50PM 0 PrivacyManager not working
2:17PM 1 Attrafax
1:50PM 1 *21*<number> # diverting
1:49PM 2 Queues +Exiting
12:41PM 2 Catch end of Eagi script when caller hung up...HELP ME PLEASE!!
12:16PM 2 MixMonitor fdiles
11:52AM 0 [VOIP-Users-Conference] Subject or Guest for Friday?
10:30AM 0 m switch in dialplan
9:33AM 0 Question about custom Asterisk billing engine
8:07AM 1 For Your Information - Our Experience with ATCom Phones...
8:01AM 5 queue logging
5:49AM 0 call forwarding
5:37AM 4 Message waiting indication(MWI) for voicemail - to H323 endpoints
5:35AM 0 MWI for voicemail - H323
Tuesday April 8 2008
11:54PM 4 RTCP not being sent when on hold
11:11PM 5 Zaptel 1.2.25 and 1.4.10 released
3:31PM 1 Digium HPEC license counting
11:23AM 1 testing please ignore
5:06AM 6 Newbie Polycom: Where is SoundPointIPWelcome.wav used?
2:01AM 1 Anyone have a method of keeping an incremental tally of calls?
12:05AM 0 Quiet recordings
Monday April 7 2008
8:47PM 3 new instalation os asterisk
6:16PM 0 Eagi
5:12PM 3 Astribank
3:47PM 4 DTMF between Asterisk servers.
2:31PM 2 Laying out things correctly
12:38PM 2 exited non-zero on 'Zap/1-1' in macro ...
12:36PM 0 Parked calls and callerid
12:25PM 5 Asterisk 1.4.19 crash with Realtime using SIP peers
7:33AM 0 Timing issue for Music On Hold - has *anyone* come across this?
7:12AM 0 callerid problem
5:27AM 3 One Touch Recording
3:31AM 0 Problems with Unicall and TE122B
Sunday April 6 2008
11:58PM 2 Newbie Polycom: Headset Suggestion for IP601
9:02PM 5 Need help with Cisco 7960
6:34PM 0 OT Good Price $.099 - Voip/ Skype PC Handset UP609 "Regular $29.99"
5:30PM 17 UK POTS - Is there a better card than TDM400P available ?
3:42PM 2 Help, problems with calls sent from nextone gateway
7:01AM 57 Where is the Digium DS3 card?
4:48AM 10 Half-duplex call on TDM2400p with VPMADT032
Saturday April 5 2008
11:11PM 9 Zaptel data mode not supported?
11:07PM 1 Question about Cisco IP phone + Asterisk + channels
8:20PM 15 Paging for analoge devices
4:45PM 0 TDM410 Callerid UK
12:16PM 5 IAX IP Phone
2:19AM 5 SellVOIP
12:55AM 4 iaxmodem + hylafax w/ DID routing
Friday April 4 2008
9:46PM 7 Is there a distro with hlyafax rolled in?
9:03PM 0 One-way audio after music on hold
8:23PM 3 howto debug bad iax voice quality?
8:06PM 0 Low Volume on Recorded Voicemail Messages
6:34PM 0 Transfer BACK to CallManager over SIP trunk?
6:24PM 7 rxfax issue
3:54PM 0 discrepancy between CDR clid and Polycom IP601 clid
3:02PM 0 Problem about calling from atrixbox to pbx extension
12:43PM 0 Sample configuration files for ATAs
12:18PM 0 Communication between two asterisk servers on two different subnets
11:51AM 1 tokbox - voice and video in the browser
11:14AM 2 Quick Help, Anyone? EMERGENCY
11:11AM 0 Problems with E400P Cards
10:35AM 11 Ring back when free?
10:09AM 1 Next Move - Hosting
8:26AM 0 Problems with Analog - SIP phone conversations
8:04AM 2 Friday April 4th @ 12 Noon EDT: VoIP Users Conference (Asterisk!)
7:28AM 1 rxfax crashes Asterisk (segmentation fault)
6:09AM 0 Need help install rxfax/txfax
5:46AM 0 Forking using Openser And Asterisk
5:19AM 10 Advice on best operator phone (with attendant console)
3:30AM 4 SJphone behind NAT/Firewall without sound
2:39AM 2 Click to call
Thursday April 3 2008
11:17PM 0 About outdail SIPCALLID
10:06PM 1 Hearing "transfer" during call
9:52PM 0 Vitelity and AsteriskNOW
9:28PM 8 C# SIP API to Comiunicate with Asterisk
6:53PM 0 IAX - FWD status
6:05PM 0 Problems with analog <-> SIP phone confif\gurations
5:40PM 0 NAT when outbound call leg is not a local subscriber?
5:34PM 2 Sending audio to a channel
5:10PM 0 Asterisk (or maybe Zaptel) goes to "sleep" after inactivity?
4:55PM 0 Config file for 'make menuselect' available?
4:38PM 8 ISPBX Announces COGOBLUE Interface and PBX Appliances
3:07PM 5 Wait for dialtone feature on FXO device
2:12PM 2 Send DTMF digit every 15 seconds during a call
2:11PM 19 Web page to show online extensions?
1:27PM 1 Listening on conversations for training?
11:56AM 22 ztdummy
11:10AM 0 transfer the call from zap/1 to zap/2 (FXS)
10:56AM 1 host unreachable
10:36AM 0 Asterisk i18n
9:43AM 0 call transfer issue
9:39AM 12 AsteriskNOW and IE
8:17AM 0 Strange problem with VoicemailMain
7:01AM 0 How does zaptel and Sangoma Media Gateway compare ?
6:46AM 1 Combined patch fixing queue-state and bug12127 for 1.4.x
12:10AM 0 problem with Kewlstart hangup detection
Wednesday April 2 2008
9:17PM 2 BRI hardware supported by 1.6 libpri ?
8:45PM 0 Asterisk parked calls and callerid
8:37PM 5 FW: [newtech-1] Skype 24 Hour Rolling Analytics
8:26PM 1 TE205P
8:09PM 0 Receiving 404 Not Found on all outbound calls when attempting SIP trunking between * 1.2.22 and Mitel 3300 CX
7:18PM 6 Analog modem as phone
5:51PM 0 OpenFrame
5:06PM 0 Asterisk 1.4.19 and Asterisk-addons 1.6.0-beta3 Released
4:20PM 0 zaptel_vldtmf
4:01PM 0 RTP no sound on asterisk
2:37PM 2 Problems with DELL 1600
1:28PM 9 zaptel alarm
12:56PM 0 misdn config warnings in Asterisk
11:12AM 8 CentPBX mirror?
8:40AM 2 show uptime and last reload
8:11AM 3 Howto connect to Cirpack softswitch with Asterisk ?
7:03AM 3 How to Hangup after DIAL is completed
3:49AM 3 How to wait before sending DTMF in DIAL command
1:20AM 1 FXS, Power and Sangoma
12:46AM 2 Virtual or Hardware SIP Modem
12:42AM 11 Asterisk on iPhone
12:12AM 1 call files
Tuesday April 1 2008
10:48PM 1 Asterisk and radius
10:11PM 5 TDM410E card, 1 FXO module - how to dial Out
8:29PM 3 g729 encoder/decoder
8:15PM 1 Calls randomly being placed on hold...
7:31PM 8 Voicemail- Recorded Mesage Low Volume
6:31PM 9 Digium B410P, bristuff and BRI support in 1.6
5:37PM 4 Zaptel support removed from Asterisk
5:24PM 9 help with no audio
2:29PM 4 breaking into asterisk channel
2:21PM 0 Hangup problem with meetme
1:25PM 0 ECT implimentation
10:14AM 9 interrupting MOH
9:31AM 2 Realtime MOH
12:38AM 2 Unicall + incomplete DNIS on international calls