asterisk users - Mar 2007

Saturday March 31 2007
TimeRepliesSubject
3:37PM 1 Setting a call to be recorded before Xfer?
2:14PM 0 Understanding the dial flags
12:22PM 1 Re: asterisk-users Digest, Vol 32, Issue 129
9:34AM 1 LUSYN patches
8:43AM 6 Sponsored development - Monodirectional audio handling
5:58AM 5 Meetme question
3:08AM 9 Question on Priorities
 
Friday March 30 2007
TimeRepliesSubject
9:44PM 1 Indicating agent status on the phone
8:13PM 0 Re: Lucent TNT - ring timer
7:30PM 0 Re: asterisk-users Digest, Vol 32, Issue 128
2:51PM 4 switchtype and signalling query
1:52PM 0 Redirect failed, channel not up.
1:49PM 2 Paging
11:45AM 2 Realtime call-limit
11:32AM 8 Multi-Level Queue
10:20AM 0 forwarding loop not detected
10:11AM 0 unconditionally redirecting incoming calls by 302 Moved Temporarily messages doing right accounting
9:48AM 10 SNOM 360
9:38AM 1 One way intermittent static to PSTN
9:35AM 2 call file vs. originate
9:07AM 0 SPA3102 PSTN fallback
6:56AM 1 xten web phone
6:49AM 0 pickupgroup / SIP / Cisc phones
5:57AM 2 Which IP Phones have buttons can be assigned to functions with Asterisk
5:32AM 1 Sipura SPA2000 Transfer Call
5:18AM 5 Speed Dial Application in *
3:33AM 2 bad case of buzzing
2:08AM 2 Asterisk 1.4 with Digium B410P - Timing problem
2:05AM 4 web based sip phone
1:42AM 0 Asterisk-Addon-1.4.0 MySQL
12:40AM 3 Security on long distance calls
 
Thursday March 29 2007
TimeRepliesSubject
10:39PM 3 Correct latency values in "sip show peers"
9:04PM 1 FW: REG : H.323 Configurations with Asterisk
8:53PM 2 Need help to strip variable
5:56PM 1 queue priority causes crash
5:35PM 3 Queue priority
5:22PM 5 SIP RTP Tunnel
3:42PM 2 Problem while using asterisk Realtime
3:26PM 8 Linksys SPA 3102 causing me problems
2:39PM 2 Setting rxgain per channel
2:01PM 5 CallerID + Name
1:43PM 3 Polycom Power
12:49PM 0 Re: [asterisk-dev] Find the name of queue
12:22PM 2 Call Waiting problems
12:04PM 2 DTMF Corruption Problem in 1.4.2 for SIP RFC2833 plz halp
11:18AM 4 UK PRI and outgoing CLI FYI
10:43AM 1 chan_misdn
10:36AM 0 Re: Problem converting a Cisco 7960 to SIP
10:29AM 1 bugetone 200's
10:24AM 4 Polycom 501 + Asterisk +Edit buttons
10:23AM 0 Hearing noise after 1min of calling
10:20AM 9 Asterisk hangs up SIP call after 6 200 retransmits
9:44AM 3 help - UNSUBSCRIBE
9:18AM 5 Re: Problem converting a Cisco 7960 to SIP
8:51AM 6 SIP & NAT
8:21AM 0 Scratchy Audio with Asterisk 1.2.4 over IAX on FreeBSD?
8:18AM 1 Is it possible to install CCM on a Linux platform ?
8:16AM 0 Asterisk does not reINVITE after 302Redirect & 401Unauthorized
8:05AM 7 maximum simultaneous calls
7:51AM 3 L options in Dial() dont seem to work....
7:49AM 0 Where are Spandsp changelogs or bugs available ?
7:33AM 6 Off Topic: Open Source USB Softphone
7:07AM 0 SV: Set(CALLERID(all) not working with 'unknown'call?
6:31AM 0 Asterisk Feature attended transfer
6:20AM 1 Interconnexion d'un serveur Asterisk à des PABX LG ( IP LDK)
5:46AM 0 DISCONNECT 41 hangup problem on PRI
5:38AM 0 txfax and result
5:04AM 1 cisco 7902
3:56AM 10 error in FreePBX
3:09AM 2 Set(CALLERID(all) not working with 'unknown' call?
2:32AM 1 Dell poweredge 860 acceptable forofficeenvironment ?
1:56AM 2 sip: failed the authenticate on INVITE
12:00AM 1 Voice mail
 
Wednesday March 28 2007
TimeRepliesSubject
11:59PM 0 vzaphfc installation?
11:30PM 1 Nice Transfer Feature
10:00PM 22 Call dies when I press *
8:51PM 1 Stepped deployment - T1 PRI passthru
6:16PM 1 How to place a call to a Google Talk user?
5:17PM 5 Polycom SoundPoint 501
3:52PM 0 asterisk-addons-1.4 write wrong uniqueid
2:45PM 2 Unsetting Global Vars
2:29PM 14 Polycom and Asterisk
2:06PM 1 SIP OPTIONS dialog not understood
1:57PM 1 App_RXFax Problem.
12:34PM 10 Multi-line phones - Asterisk uses wrong callerid
12:08PM 3 Transfering not working - how to debug?
10:56AM 0 Wanted: German to English translator for Asterisk documenation
9:45AM 0 BRI Cards
9:12AM 1 Asterisk: recommended installation
8:08AM 16 PoE - IEEE 802.3af
7:20AM 7 Development of new features in Asterisk Manager
6:30AM 1 h323
6:29AM 1 MOS Score
6:27AM 5 Meetme cant handle more than 5 users in a call?? hmmmm
5:41AM 0 * 1.4.1: connected to gtalk but no voice passing
5:19AM 23 wireless desktop phones
5:03AM 12 System from AMI
4:40AM 2 Friday asterisk users live conference/podcast at 12:30PM EDT
3:55AM 0 Voicemailmain not changing password?
2:21AM 0 REG : H.323 Configurations with Asterisk
1:53AM 2 Can I generate random SIP traffic?
1:24AM 2 Odd MeetMe bahaviour with MoH ...
12:59AM 2 just on my LAN
 
Tuesday March 27 2007
TimeRepliesSubject
6:24PM 0 Macro Dial - External DID
4:45PM 9 "Couldn't load variables.txt?aldope=xxxxx "
4:29PM 16 Inbound Voice Quality - Speed Change
3:13PM 0 ARI with * 1.4.2 won't display recordings
1:59PM 1 Erased log files
12:34PM 9 Park & No Announce?
10:29AM 1 P-Asserted-Identify or Remote-Party-ID, or both?
8:19AM 1 just call to user
8:18AM 9 TDM400p reliability
6:57AM 2 AOC billing
6:56AM 0 IAX Experiences [WAS: Question about DSP in Digium card]
6:54AM 1 Re: asterisk-users Digest, Vol 32, Issue 106
6:53AM 2 cisco 7905
5:10AM 1 UK BT PRI
5:00AM 0 Asterisk MSOutlook Dialer
3:50AM 0 AMI - delete voicemail
3:46AM 5 Using server side phonebook directory with SPA941
2:31AM 14 ztdummy and MOH
2:20AM 1 vzaphfc installation...
 
Monday March 26 2007
TimeRepliesSubject
11:28PM 6 how to define a pilot number
11:25PM 1 SIP Video Camera
7:43PM 0 rx_fax and Asterisk 1.4.2
7:35PM 1 Server Recomendation
4:36PM 0 SIP REFER
2:32PM 4 SRTP vs ZRTP in Asterisk
2:17PM 1 Emergency chan_sip issue
2:05PM 0 Device not registering after boot
2:00PM 3 SIP registration
1:20PM 12 Doorphone
11:51AM 0 Hang up detection time in FXS module
11:18AM 0 Getting an ASR Number
9:14AM 11 Polycom 601 loop
9:05AM 1 outbound call
8:33AM 16 Multi-registration ?
7:58AM 3 Asterisk and T38 ?
7:46AM 0 Registration timed out after a "sip reload"
7:40AM 1 1.4 - IAX2 - No registration for peer
7:16AM 4 How is this feature called ?
6:59AM 0 ARI with * 1.4.x
6:31AM 0 No Audio when integrating with openSER and Asterisk in the SAME LAN ,
5:23AM 4 Asterisk incoming caller id problem
3:51AM 2 Counting callers
3:17AM 2 cutting hash in dial app
2:12AM 4 Moving from Bristuff to mISDN
1:17AM 2 Failure acknowledgement time
12:17AM 0 Asterisk 1.4 Realtime problems
12:11AM 8 Two or More Bri Cards
 
Sunday March 25 2007
TimeRepliesSubject
7:31PM 2 Chan_cellphone and CentOS 4.x
7:15PM 1 how to check and set D-channel status
6:47PM 1 ztdummy install in the new zaptel 1.4.1
11:25AM 11 Anyone having trouble with claling US Domestic on Sellvoip?
11:06AM 5 voicemail is not playing messages
8:51AM 1 the age old telephone tree... why re-invent the wheel?
5:25AM 2 AOCD -> SendText()?
3:44AM 1 Answer Confirmation with SIP/AIX channels
 
Saturday March 24 2007
TimeRepliesSubject
11:43PM 7 Problem with ztdummy
6:11PM 1 asterisk: error while loading shared libraries: libiksemel.so.3: cannot open shared object file: No such file or directory
3:47PM 1 Timeout for conferences
3:39PM 7 Question about DSP in Digium card
12:33PM 1 Remote host can't match request NOTIFY to call
10:57AM 1 Asterisk with Dialplan or TrixBox for this case?
9:24AM 9 TDM analog cards, volume, echo, fxotune, ztmonitor and HPEC
8:57AM 1 Voicemail for an AT&T System 75
8:12AM 4 Need feedback on vitelity
7:34AM 3 Can be called on FreeWorldDialup/IAX channel, but can't make calls
7:16AM 1 Shoretel integration into Salesforce.com
6:21AM 1 Asterisk Viruses?
3:54AM 3 Issue with Hamlet ISDN PCI card(Cologne Chipset)
3:12AM 2 freepbx -> DB Error messages...
 
Friday March 23 2007
TimeRepliesSubject
3:45PM 4 SER vs Asterisk?
2:20PM 0 switches
1:12PM 2 Problem with busy and unavailable
12:13PM 0 IAX2 certificate based (RSA) user auth in 1.4.x
11:57AM 24 Doorphone vs. Grandstream BT101
10:25AM 3 Sendmail and exchange for voicemail integration
9:47AM 4 PC / Phone Combo
9:45AM 1 HUD Lite server on Debian
9:36AM 11 SIP/IAX peers UNREACHABLE and audio loss
9:27AM 1 Noob question regarding PCI 2.x & TDM400P Card
8:58AM 0 Debian Asterisk and MeetMe
8:49AM 3 Semi-OT: Use T.38 ATAs to Extend fax lines
8:40AM 3 cause 127
8:23AM 0 no incoming dad with mISDN 1.1.1 and asterisk?
8:12AM 4 SRTP testers needed
7:03AM 4 AsteriskNow Beta 4 with T1 Cards?
6:32AM 0 No Audio when integrating openSER and Asterisk , in NAT
5:10AM 0 minimal asterisk for iax2 bridge
 
Thursday March 22 2007
TimeRepliesSubject
11:23PM 1 Outbound SIP call from asterisk extension
9:50PM 0 Zaptel 1.4.1 Released
5:20PM 2 About FAX and PAP2
5:12PM 3 Problem in using Two BRi Cards in Asterisk
3:29PM 0 SIP REFER ans Dialplan
2:46PM 0 PASS CORRECT CALLER ID INFO TO OFFSITE TRANSFER
2:41PM 2 managers
12:49PM 8 Linksys/Sipura SPA-942 phones in larger deployments
11:52AM 1 chan_capi and only one B channel usable?
10:39AM 2 hardware spec
10:02AM 4 accepting a call, macros, and key presses.
9:49AM 0 Asterisk x Mera MVTS
8:43AM 1 using ael and extensions.conf togather?
7:51AM 1 Gizmo project answers every call - can I use it in hunt group?
7:29AM 3 ChanSpy and MeetMe
7:16AM 4 Asterisk 1.4.2
6:18AM 0 Bridged ZAP calls do not release
5:18AM 0 Digium b410p and 2.6.17 kernel bug?
4:40AM 6 302 Moved temporarely
4:28AM 0 Test Message
3:51AM 0 fax and mISDN: a chimera?
2:31AM 0 beronet BN8S0 and isdn phone
2:23AM 5 strange ring
 
Wednesday March 21 2007
TimeRepliesSubject
10:45PM 4 A request for your input.
10:30PM 1 CN=Diarmaid O'Loughlin/O=QAD1 is out of the office.
9:04PM 0 SIP peer disappearing
7:17PM 20 Cisco 30VIP Phone
5:23PM 2 Too Many Open Files, Hung SIP Sessions, Can I Increase File Count?
3:24PM 3 G729 'disappears' randomly
2:38PM 2 Looking for a terminations provider (carrier grade)
12:54PM 3 Voicemail mailbox number passed in connection?
12:49PM 3 Asterisk 1.4.2 chan_zap
11:17AM 0 install and setup app_mp4 application
10:25AM 1 How to resolve CallerID from AudioCodes FXO
10:03AM 0 Asterisk 1.4.2 Requires Zaptel from 1.4 svn branch for zap_chan?
9:48AM 11 Cisco 7970 with skinny on * 1.4.1
9:34AM 1 Dlink i2eye
8:55AM 4 PickUp a call with feature pickup (*8) from a IAX2 channel
8:25AM 0 res_musiconhold.c:1243 load_module: No music on hold classes configured
8:11AM 3 ses ActiveDirectory and also Ldap and Kerberos.
8:00AM 1 asterisk log analyzer
7:34AM 0 reducing the number of extensions for every user
7:23AM 0 Asterisk 1.4.2 Released
7:22AM 0 Asterisk 1.2.17 Released
7:09AM 7 How to get AEL2
7:07AM 0 Asterisk with AudioCodes Mediant 2000
6:47AM 6 wct4xxp problem
6:15AM 1 Metaswitch help needed
5:48AM 5 FWD outgoing problem
5:45AM 13 About Pickup Grandstream
5:23AM 5 automated dialout detect forward
4:55AM 7 Limit call duration
3:39AM 10 polycom random reboots
 
Tuesday March 20 2007
TimeRepliesSubject
9:07PM 7 wrong values in duration and billsec in CDR
8:37PM 1 SIP/Polycom Issue, Asterisk 1.2.16, calls dropped
12:00PM 1 codec_zap and Asterisk 1.4.1
10:26AM 3 Can't Compile w/HPEC
9:53AM 5 High Pitched Noise
9:31AM 7 Asterisk Automated Outbound Messaging
8:04AM 0 Activating Incoming Demo
7:25AM 1 Zaptel 1.2.16 Released
7:22AM 0 GXP-2000 Phones with Cisco 3560 PoE Switch
7:17AM 2 modem passthru
6:57AM 12 asterisk on debian
5:28AM 0 which spandsp for asterisk 1.2.16 (eom)
5:11AM 8 Problem with "AT&T Maintenance" protocol in PRI connection, no B+D channels available
5:07AM 4 error, install freePbx
4:33AM 0 PROGRESS code
3:32AM 1 Zapateller not playing audio via SIP Trunk?
1:49AM 2 Which parameters of a live Asterisk server would you monitor ?
1:40AM 0 how to interconnection asterisk(sip) with mera
 
Monday March 19 2007
TimeRepliesSubject
5:03PM 0 SIP provider did not send BYE if callee is an queue
4:57PM 25 Microsoft launches first PABX
4:06PM 1 Festival works extension to extension, not on trunk
3:52PM 0 1.4.1 - T38 Pass Through - Seeing some odd errors but the fax works.....
2:20PM 0 H.323 with g729
2:07PM 1 ExternalIVR() Dialplan function and Festival
1:52PM 4 Teliax problems, they say use SIP, more mature & better working than IAX
12:40PM 2 Zaptel Dummy Driver
12:10PM 1 Unicall Brazil
11:47AM 18 Configuring Faxs any help :)
11:16AM 0 One way dtmf tone on IAX
10:23AM 3 Cepstral and numbers
8:55AM 4 Dial(Local/${EXTEN}@longdistance)?
7:56AM 2 TDM400p, no CLI activity
7:40AM 6 Queue App - Free agent and waiting calls
5:23AM 1 no special context for sip peer
4:26AM 0 asterisk 1.4: choppy voicemail sound after upgrade from 1.2.9.1
3:06AM 0 make calls with different phone numbers
2:53AM 0 Conference server (or how to make a call withmorethan 3 u
1:32AM 2 Conference server (or how to make a call withmore than 3 u
1:14AM 5 Conference server (or how to make a call with more than 3 u
 
Sunday March 18 2007
TimeRepliesSubject
8:58PM 3 zttool always reports "OK" on TDM400P
4:41PM 3 camp on off-line phone
3:42PM 9 T1 cable for Digium T1/E1 Cards
11:30AM 1 Choppy sound with chan_capi + Fritz Card USB
11:07AM 1 Conference server (or how to make a call with more than 3 users)
 
Saturday March 17 2007
TimeRepliesSubject
12:41PM 0 1.4 sample postgresql configs
11:17AM 3 Queues
8:22AM 0 detecting missing end-of-queue records
4:21AM 2 Call counter for sip misbehaving
1:49AM 2 SMS Integration and SMS commands
1:33AM 0 Re: asterisk-users Digest, Vol 32, Issue 67
 
Friday March 16 2007
TimeRepliesSubject
6:15PM 0 Monitor queue calls with Local channel agents
5:09PM 0 Channel stuck problem..
5:08PM 15 Follow me on multiple numbers..
5:03PM 7 Jajah.com like script?
2:15PM 2 Pickup some else's call
12:47PM 2 Problems with MFCR2 and Meridian
11:33AM 0 DISA and repeating calls
11:32AM 13 Refund from SellVoip?
11:17AM 2 Error compiling zaptel 1.4.0
10:45AM 3 FW: Microsoft buys Tellme
10:43AM 4 Asterisk 1.2.13 Caller ID problem
9:35AM 8 proposal: a new mailing list for asterisk 1.4, why not?
8:43AM 0 MAX TNT Question
7:44AM 1 Cisco + Asterisk list anyone?
6:30AM 9 Cepstral voices
3:44AM 7 transfer=mediaonly : can't hear nothing
3:13AM 0 Transfer feature not working on asterisk 1.4.0
3:12AM 1 Warning LSP Low
2:57AM 3 SIP phone supporting more than 10 extension with a call transfer command
2:53AM 20 Dell poweredge 860 acceptable for office environment ?
2:04AM 1 Voicechanger update for asterisk 1.4
 
Thursday March 15 2007
TimeRepliesSubject
9:46PM 0 Re: busy/hangup/answer detection in PRI E1 channels (Vidura Senadeera)
8:34PM 2 Help! Echo problem even at T1 PRI?
6:32PM 2 voip-info.org is back!
3:38PM 1 Re: zapata with Tiger3XX compilation error
2:11PM 1 sip_nat.conf - Asterisk with two Ethernet Interfaces
1:52PM 3 Incoming Caller ID
1:39PM 4 A200 card problem
12:54PM 1 shutdown
12:12PM 1 Dropped calls in Asterisk - A general question
11:03AM 2 Freepbx Incoming call's configuration
9:13AM 1 asterisk n-way call problem
7:33AM 10 snom led not working with asterisk 1.4.1
7:31AM 3 qozap: t3 timer expired for span ...
3:08AM 2 busy/hangup/answer detection in PRI E1 channels
2:55AM 0 Meetme variables
2:43AM 0 MP3Player
1:41AM 0 Cost of Branded Equipment for Voip Provider Implementation
 
Wednesday March 14 2007
TimeRepliesSubject
10:15PM 3 DECT to SIP gateway experiences
10:08PM 0 AgentCallBackLogin Help!
8:59PM 4 DNIS/DNID
7:20PM 4 Voip-Wiki Site Information
6:16PM 0 Inbound PSTN CLID irratic with A200
5:15PM 7 Cisco 7912
4:59PM 36 voip-info.org status update
2:51PM 22 While the VoIP-Info.org site is down...
1:31PM 4 Packetization Rate
12:50PM 0 Sped up recordings with 1.4.1
12:26PM 3 Which SIP method/option to display a short text message ?
12:19PM 5 IAX2 - Congestion
11:05AM 0 Can I use an Intertel IPPhone Plus 7704500 with Asterisk somehow
10:36AM 8 Call center manager for Asterisk (Release 0.3)
9:56AM 0 ${EXTEN} is limited to 17 characters under IAX ?
9:09AM 0 IVR after hangup
9:02AM 1 [G.729] Input Gain
7:45AM 13 what happened to asterisk wiki???
7:18AM 8 Zaptel version for asterisk 1.2.16
6:41AM 2 Manager connection problems
6:18AM 0 Autoprovisioning ST2030S
5:54AM 4 What is the best phone to get when using a headset?
4:54AM 2 Earliest dial tone, after boot up.
4:49AM 11 What happend to voip-info?
2:32AM 0 ChanSpy with Record() : Doesnt seem to work for an unbridged SIP call
2:13AM 1 beronet BN4S0
1:16AM 3 strange things on call transfer
 
Tuesday March 13 2007
TimeRepliesSubject
11:35PM 1 T1 Integrator Birch
8:32PM 1 Digium S101i - Adapter DTMF works perfeclty
2:43PM 0 Press quotes needed from ENUM users on Asterisk
2:10PM 9 Asterisknow with video and X-Lite not quite working
1:10PM 1 RE: In Asterisk 1.4.x, Why Digium has two H323 channels?
12:54PM 0 asterisk 1.2.15 fax
12:41PM 0 Re: asterisk-users Digest, Vol 32, Issue 48
12:00PM 1 Getting 7970 to update
11:39AM 4 How to match wild card inside a GoToIf?
10:18AM 5 120 concurrent ZAP connections in asterisk open edition. Is that possible?
9:54AM 3 DST and VM timestamp
8:44AM 0 SIP hardphones with good jitter tolerance
8:38AM 1 IAX2 Question (Asterisk 1.4 tarball)
8:01AM 1 French PRI channel - exact signaling used
7:41AM 3 voicemail scenario
7:19AM 0 Queue - Call delivery problems - Asterisk 1.4 - autofill=yes
6:27AM 0 CDR and CallerID
5:06AM 0 MusicOnHold stops after upgrade from 1.4.0 to 1.4.1
1:53AM 1 Update Asterisk 1.2.12 to 1.4.1 ?
 
Monday March 12 2007
TimeRepliesSubject
9:33PM 5 great problem with sounds and ztdummy
5:38PM 0 RE: Playback 0.5% Too Fast?
5:32PM 2 Playback 5% Too Fast?
5:19PM 4 TDM-400, Polycom SIP phones, and echo problems
5:03PM 1 Voicemails with occasional speeded up portions
3:46PM 11 Call Back
12:00PM 2 Polycom: warble on registration?
11:12AM 1 deprecated ALERT_INFO var andAMI's Originate command
11:03AM 0 Incase anyone wanted it - SNOM USA DST settings
10:46AM 0 GXV3000 & Speakphone
10:46AM 0 LIDB/CNAM STORAGE DATABASE NEEDED
10:15AM 1 OT: Sipura DST Rules
10:14AM 9 FW: Seamless Multi Office Asterisk Deployment
9:55AM 1 GXP-2000 DST Change
9:52AM 2 Create meetme conference rooms on the flight.
9:05AM 4 SIP unicode support ?
8:50AM 2 ACM question
8:43AM 5 Filter IDENT(Port 113) on Linksys router puts remote extensions to one way audio
8:34AM 0 DST 2007 Config for Cisco 7970
7:50AM 7 Rebooting all Aastra phones
7:50AM 1 In Asterisk 1.4.x, Why Digium has two H323 Channels
6:24AM 3 New to Asterisk
5:33AM 6 Single sign on PC + phone?
5:32AM 0 Shoutcast music-on-hold
5:18AM 4 Rebooting ALL polycom phones
5:17AM 0 Citel Handset Gateway DST fix - FYI
5:16AM 2 AMI - DBPut
4:16AM 0 Problem with H323
3:58AM 0 Re: Help: CallerID Name not being sent
3:51AM 3 _ALERT_INFO replacement in 1.4?
2:49AM 0 Coming events in Europe
2:44AM 0 Pickup group
2:07AM 7 How many outgoing phone line/voip account do I need?
1:18AM 1 Problems with Voice conferencing
 
Sunday March 11 2007
TimeRepliesSubject
10:48PM 6 Problem configuring voice conference
10:15PM 1 Asterisk and Databases
9:12PM 3 Re-parking (or transfer) a parked call
6:53PM 6 DST changes for the US
4:40PM 2 g711 -> iLBC garbled voice in 1.4?
3:32PM 2 Complicated callback solution
8:37AM 5 Fix for TZ values updates for DST
3:37AM 0 How to best manage my dial plans as the continueto grow, and grow, and grow....
12:30AM 1 Follow Up on Cannot get back chan_zap.so module!??
 
Saturday March 10 2007
TimeRepliesSubject
5:56PM 5 IAX2 audio issues
3:41PM 0 how to determine the remote IP address when dialing in ooh323
12:34PM 0 Security and DTMF
11:54AM 0 Polycom call parking feature and Asteriskcallparking
11:10AM 20 asterisk on mini-itx
9:00AM 0 chan_misdn and Asterisk 1.4.1 - Early B3 not working any more
7:33AM 1 FastAGI Question
6:03AM 0 Connecting two asterisk server.
1:19AM 1 installation pb on debian etch
 
Friday March 9 2007
TimeRepliesSubject
10:00PM 0 DTMF issue with TDM404
6:29PM 1 How to best manage my dial plans as the continue to grow, and grow, and grow....
4:56PM 0 spandsp, app_rxfax: apps_Makefile.patch v1.2 > v1.4 = No Workie!
3:27PM 3 REALTIME and VIEW ENTIRE DIALPLAN
2:17PM 8 Polycom call parking feature and Asterisk call parking
2:16PM 8 Recorded file processing app wanted
12:53PM 5 play file and action only stop if one defined key has been pressed
12:17PM 0 OS X Frequent console disconnects 1.4.1
11:55AM 1 Cdr_mysql compile question
11:49AM 7 AEL #include file
10:49AM 1 Which hylafax client ?
10:27AM 1 RE: Coaching in asterisk
10:21AM 1 RE: Coaching in asterisk
9:22AM 0 Boot order of 2 TE110P and 1 TDM400P in the same
9:03AM 1 Another Faxing Question
8:42AM 0 YAACID and manager.conf security
8:24AM 1 sip tunnel
7:24AM 4 disable client side hangup after dialing 911
5:34AM 0 Re: Back to back E1 - asterisk <=> toshiba pbx - Call droping issue [ ref:00D36mPe.50032wycQ:ref ]
5:19AM 12 Is there any variable for Voicemail Password in Asterisk
12:52AM 5 How to enter bridge_native_loop???
12:41AM 1 Can't hear any sound (This time in plain text)
12:29AM 3 When to use Echo Cancellation cards?
12:13AM 6 Zaptel problem after upgrading to 1.2.16
12:10AM 0 Fwd: Can't hear any sound
12:04AM 9 Which VoIP router and switch to use for medium size business
 
Thursday March 8 2007
TimeRepliesSubject
11:03PM 5 Issues with a Linksys SPA 2102 and asterisk
6:01PM 3 Boot order of 2 TE110P and 1 TDM400P in the same machine
4:37PM 2 Is Allison going to be banned from foreign travel over polar bears?
4:16PM 7 Newbie Question
3:56PM 0 Re: Coaching in asterisk
3:37PM 3 1.4 compile issue
2:56PM 8 Call load balancing
2:43PM 2 Zap Channel Deadlocks
2:41PM 0 Asterisk SIP to MAX TNT Gateway, Sporadic Echo
2:37PM 1 No application 'Prefix' for extension in1.2x, what app I have to use instead?
12:35PM 1 outdial to phone for new VM notification
12:27PM 0 Number of groups?
11:47AM 0 transfers and CDR
11:30AM 4 Sender phone ringing while recipient talking
10:29AM 3 Packet2Packet Bridging Questions
10:19AM 5 Queue announcing hold sequence instead of hold time
10:00AM 4 Accessing Voicemail by dialing own number
9:03AM 5 Asterisk distributed deployment
7:58AM 0 cmd pickup Problem
7:10AM 0 Fwd: Back to back E1 - asterisk <=> toshiba pbx -Call droping
6:54AM 0 New Linksys SPA Daylight Saving Time Rule for US/Canada
6:50AM 1 Asterisk + Panasonic pbx
6:36AM 4 Hinting and Realtime
3:44AM 7 Empty Wildcard TDM400P as a MeetMe timer.
3:39AM 0 "pritimer" parameter in zapata.conf
3:27AM 2 Queue Announcements for Operators
3:21AM 1 How to handle SIP-Callerid?
2:31AM 0 Timeouts not working
1:20AM 3 Re: Pickup *8 with CallerID
12:06AM 1 Call recording and archiving
 
Wednesday March 7 2007
TimeRepliesSubject
10:56PM 1 Re: Back to back E1 - asterisk <=> toshiba pbx - Call droping
9:58PM 0 SIP remote crash bug
9:32PM 5 Number of SIP messages per minute
8:15PM 1 sip show channels
6:39PM 1 Problems with Zaptel Drivers
1:51PM 3 auto dialer
12:47PM 1 Asterisk Registering to other SIP servers.
11:46AM 0 gtalk2voip and Asteris
10:56AM 1 Problem HandyTone 488 does not call transfer
10:27AM 2 mobility with asterisk
9:07AM 1 AsteriskNow Beta 4 - zaptel.conf / zapata.conf problems
9:00AM 2 Asterisk Auto-dial out
8:42AM 11 Asterisk queue and agents
8:42AM 2 queue information in mySQL
7:53AM 4 OT Vonage V-Phone Adapter (Possible Hack)
7:38AM 3 asterisk and ssl
7:29AM 2 Realtime Extensions and "Include"
6:20AM 6 Background / Invalid Extension through cell phone
5:45AM 5 VoIP over Alvarion Wireless
3:31AM 1 capi installation problem
3:17AM 0 Asterisk 1.4.1 - Calling problem
2:59AM 0 Back to back E1 - asterisk <=> toshiba pbx - Calldroping issue
12:14AM 1 Back to back E1 - asterisk <=> toshiba pbx - Call droping issue
 
Tuesday March 6 2007
TimeRepliesSubject
11:03PM 0 Re: asterisk-users Digest, Vol 32, Issue 21
10:11PM 0 Anybody having problems using sellvoip?
9:22PM 0 Long term voicemail archival and synchronisationbetween multiple storage locations?
9:22PM 5 Nomination for Coolest App in 2007
9:03PM 3 GTalk/Jabber passing audio in 1.4.1!
3:21PM 1 Cancelling a digit in IVR
3:21PM 0 Add current caller to junk-callers-database
2:57PM 1 Compiling smsq in 1.2
1:44PM 3 Manager.conf '127.0.0.1 unable to authenticate'
1:11PM 1 How many gsm channels
12:02PM 1 Building a new voicemail system... Testers needed!
11:50AM 0 Long term voicemail archival and synchronisation between multiple storage locations?
7:21AM 1 Linksys PAP2 and Caller ID
7:11AM 0 chan_cellphone won't pair with phone
6:28AM 0 Pickup failover
6:22AM 3 zaptel 1.4 on fedora core 6 with dell pe 2850
6:00AM 0 Ringing does not terminate on mISDN after pickup
5:22AM 1 Asterisk 1.4.0 Installation error on Red Hat Linux 9.0-Urgent
5:16AM 5 Polycom 501 - Auto answer on one line appearance
3:11AM 1 visdn, misdn and the hell
2:15AM 0 [asterisk_voip] asterisk and ogg files
1:52AM 6 Micros-Fidelio - billing in hotel
1:21AM 0 web based sipphone
12:57AM 2 preventing voicemail pickup after SIP redirect ?
 
Monday March 5 2007
TimeRepliesSubject
10:39PM 0 app_queue not using exit context?
10:31PM 2 IAX2, DTMF and x86_64.
9:41PM 1 server generated outbound conference calls?
7:21PM 10 Polycom Questions
6:10PM 1 extra-sounds 1.4.5 timestapmed newer than 1.4.6 ???
5:42PM 11 [Announce] Web-MeetMe V3.0.1 released
5:04PM 2 Using Asterisk as Voicemail Server on a dinosaur Meridian System
4:37PM 2 Voicemail question
4:05PM 1 Setting Sip Headers From Dial App?
3:48PM 1 g.729 on solaris10/x86
3:15PM 10 A New Phone Service - www.virtualphoneline.com
2:02PM 2 Re: Asterisk Java w/ Threads
11:53AM 3 Rx+,Rx-,Tx+,Tx- of TE110P
10:54AM 3 TDM400P/FXS in a HP DL380 G5
10:11AM 5 How to disable MOH completely?
8:30AM 12 TC400B
4:59AM 1 SMS ON ASTERISK
4:44AM 1 HITBSecConf2007 - Malaysia: Call for Papers now Open
1:21AM 1 new kernel and zaptel
12:39AM 2 Read() status?
12:09AM 1 Is the 1.0.X branch vulnerable to the SIP issue?
 
Sunday March 4 2007
TimeRepliesSubject
4:41PM 8 Configurations Files of TE110P
2:20PM 2 running error, unable to load *.conf files: load_modules: No 'modules.conf' found - svn version 1.4.1
1:45PM 1 Real Time, sip.conf, [general]
12:45PM 0 x100p.com
11:12AM 3 When does local leg in call file start?
6:32AM 1 running error: load_modules: No 'modules.conf' found - vesrion 1.4.1 from svn
4:03AM 1 Voice quality issues
2:01AM 6 So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1
 
Saturday March 3 2007
TimeRepliesSubject
7:08PM 3 dial question
4:10PM 0 creating new asterisk application
1:17PM 3 hanging an asterisk box off of a PBX analog extension
3:43AM 1 gtalk2voip and Asterisk
12:29AM 1 Asterisk - e164 (enum) lookup confused
 
Friday March 2 2007
TimeRepliesSubject
10:28PM 2 How to fail an AGI
7:13PM 5 Need comparison between PBXtra, Trixbox, Thirdlane, Druid, Aheeva etc.
5:04PM 4 Asterisk 1.4.1 Released
5:04PM 0 Zaptel 1.2.15 Released
5:04PM 0 Asterisk 1.2.16 Released
4:48PM 1 How to log VERBOSE statement to a file?
2:21PM 1 IP addresses
2:20PM 3 svn 1.4 - mp3 support and changing the installation directory
1:46PM 4 REMOTE CRASH FIX
1:02PM 4 rtsavesysname not working in 1.4
12:37PM 2 PRI progress codes.
11:00AM 0 FW: Alec Saunders post about Mashable Telco's
10:36AM 3 DTMF from TDM400P and X100P
10:17AM 5 Double DTMF digits sent on IAX native bridge
10:15AM 1 Voicemail to SMS using asterisk
9:56AM 5 Help Voicemail to SMS using asterisk
9:21AM 0 WMI from Asterisk to Cisco Call Manager
8:56AM 2 DTMF detection problems on PRI channels?
8:35AM 7 Alec Saunders post about Mashable Telco's
8:29AM 8 Asterisk and Fax
7:38AM 1 T38
7:26AM 4 cmd page crashes Asterisk SVN-branch-1.4-r57207
4:15AM 1 Running Fax on E1 line
3:46AM 2 BLF not working with Asterisk 1.4.0
3:30AM 0 Rif: Re: Multiple simultaneous calls
 
Thursday March 1 2007
TimeRepliesSubject
8:28PM 2 Test
7:53PM 2 How can I use the "GET VARIABLE variablename" in AGI
5:07PM 1 2 Call locations
5:05PM 8 Polycom reject button
4:57PM 1 gtalktovoip and Asteirsk
4:46PM 2 Digium S101i - pickupexten doesn't work
4:05PM 9 build rpm fails
2:39PM 3 Asterisk 1.4.1
11:51AM 11 Tesco Internet Phone
10:12AM 9 Asterisk Realtime
10:11AM 0 About queues and multiple lines.
9:47AM 6 Multiple simultaneous calls
9:31AM 2 Extensions +International
9:24AM 0 Testing asterisk with sipp
8:54AM 2 blieve i my TE110P or My teleco provider ??
7:50AM 11 IAX best practices
6:41AM 4 Cannot hear ringback music from telco
5:46AM 0 3 way calling independent of phone hw.
4:57AM 7 UK SIP Gateway
4:27AM 0 Revolution Call Accounting Desktop
4:21AM 2 TDM400p Loaded only once
4:19AM 0 Issue with Calling Name ID in SIP: Asterisk sets Caller ID Number as Name if NO Name
1:40AM 1 transfer function
1:25AM 0 Siemens HiPATH 3700 with Asterisk
1:22AM 4 DTMF not being detected with 1 provider. Works with the other provider...