asterisk users - Jun 2007

Saturday June 30 2007
TimeRepliesSubject
8:43PM 3 Polycom echo problem
3:58PM 0 mISDN Fritz Passive Card
1:13PM 0 Asterisk 1.4.5 : Use I386 ou X86_64
11:04AM 1 Asterisk 1.4.6 Fedora 7 configure error
10:55AM 0 AEL + Realitme?
6:54AM 1 Cisco 7970G line buttons
4:08AM 3 Fwd: Problems with zap following 1.4.6 install
1:50AM 1 FW: fail to load modules
 
Friday June 29 2007
TimeRepliesSubject
6:50PM 8 awful list delays: 4 days!
6:30PM 0 Asterisk 1.2.20 and 1.4.6 released
5:31PM 0 Linking Asterisk with another SIP PBX
5:16PM 2 TE420 PCI Express Card
5:07PM 2 Problem getting a Perl script to run
4:08PM 11 Music on hold 1.2
3:41PM 3 Voice Mail not Receive
3:27PM 8 v1.4.x ready yet?
2:35PM 4 features.conf / DTMF / automon hell
2:19PM 1 Asterisk 1.4 Warnnings
2:18PM 0 DUNDi problem: offline peers still in request EID/EID_DIRECT field?
1:28PM 1 MOH question w/Cisco 79xx phones
11:28AM 1 Fwd: Call Wainting dysfunctions
10:51AM 0 nway call
10:06AM 4 asterisk call unique id in dialplan
9:32AM 2 How many number of parallel calls can make through asterisk
1:40AM 1 SPA-2100 Distinctive Ring
 
Thursday June 28 2007
TimeRepliesSubject
7:16PM 4 setup multiple phones for 1 extension
6:46PM 1 CDR Log analizer software
6:42PM 0 Anyone who can do live video feed to co-host asterisk "show" next week?
6:33PM 0 Robo Dialer
6:17PM 1 Shared Extension Appearance
6:00PM 6 network routing
5:59PM 9 Asterisk and IPv6
5:54PM 1 Avaya IP Office DTMF Issue
5:39PM 0 Avoided deadlock for '0x864e70', 10 retries!
4:40PM 3 Linking Asterisk with another SIP PBX (or SIP Softswitch)
3:41PM 2 RTCP NTP Clock skew
2:53PM 2 E1 not coming up
2:43PM 23 Caller ID Spoofing to be banned in the USA
1:00PM 2 FXS channel bank
11:55AM 2 registering Asterisk on SIP/Nortel MCS server
11:05AM 1 Asterisk 1.4.5 Inserting Random Digits in Dialed Number!
10:50AM 4 Fax passthrough howto codec upspeed
10:35AM 3 CDR and call transfer
10:30AM 3 error while compiling asterisk-1.2.19
9:20AM 3 fail to load modules
8:56AM 0 Work
7:37AM 0 registering Asterisk on SIP/Nortel MCS thing
7:13AM 0 Calls audio stops with latest Gigaset C450IP firmware + voicemail
5:50AM 4 Updated Manual for Asterisk 1.4.x
5:21AM 14 Call transfer feature
4:03AM 2 voicemail.conf serveremail
2:17AM 0 Any difference using * with Centos i386 and x86_64 ?
 
Wednesday June 27 2007
TimeRepliesSubject
11:21PM 3 Voicestick / i2telecom.com
10:15PM 0 Bypass local dialplan and redirect INVITE
8:32PM 74 North American voice BRI - Informal survey
7:24PM 4 Error While Calling AGI
7:06PM 6 .call file
4:50PM 0 QueueMetrics 1.4 released today
4:37PM 2 OpenSer/Asterisk PBX solution
4:25PM 2 Has anyone sucessful Asterisk to an Avaya IP phone
4:17PM 6 Customized Ring Tone
4:08PM 5 Using MSAccess to dial on a Zap line
3:13PM 0 Asterisk to Cisco 2600 GW DTMF Not Working, Working now
2:43PM 4 Asterisk+squid
12:53PM 5 Problems compiling Asterisk 1.4.5
12:41PM 1 Round Robin SIP peers?
12:27PM 2 Help with IAX Trunk
11:46AM 2 Wait to numbers
11:20AM 1 Self Calling test
11:15AM 0 ISDN data-call question
11:12AM 0 minibrowser for each snom phone
10:45AM 0 IAX trunking using a different port
10:01AM 4 Missing 'init keys' command
9:41AM 1 Zap dialling issues
8:29AM 0 Fwd: problem with one way audio
2:59AM 1 Module '***.so' did not register itself during load
 
Tuesday June 26 2007
TimeRepliesSubject
11:01PM 1 Asterisk to Cisco 2600 GW DTMF Not Working
8:57PM 1 No such host error from SIP for non-peer configuration.
7:45PM 12 Cisco 7941 localized menus with SIP firmware
6:27PM 0 TE412 / HPDL380G5 / * 1.4 / CentOS 4.5 Experience
6:22PM 14 kore dump
4:37PM 0 Slip Events
3:45PM 0 SpectraLink SVP protocol support in asterisk
3:11PM 8 More FAX over T1
2:21PM 0 rcf2833 DTMF broken in asterisk SIP channel?
2:18PM 1 Multi port IAX Gateway
12:56PM 0 No CID on Zaps - TDM400
11:14AM 1 call fail from audiocode to sip trunk
10:29AM 0 Test Message
9:38AM 2 Bridging two PSTN calls
9:00AM 0 VPN technology for snom 370
8:59AM 0 asterisk-users Digest, Vol 35, Issue 92
8:47AM 1 realtime_extensions
7:57AM 0 Asterisk + Legacy PBX
7:27AM 1 zaptel 1.2.18 and HPEC
5:07AM 9 Inexpensive Layer 3 Switch?
4:39AM 0 call transfer problem
4:36AM 0 CDR changes in 1.4.5 are confusing
4:10AM 5 Fax Throughput
3:31AM 4 Modification of Caller ID based on context
3:01AM 0 Spy a specific Channel
1:52AM 3 CDR Records "s" as dst
1:26AM 0 asterisk-users Digest, Vol 35, Issue 91
 
Monday June 25 2007
TimeRepliesSubject
11:42PM 6 AstPligg
10:33PM 2 Ring the second line when 1st line is busy
9:41PM 2 Problems with ChanIsAvail always return status 0
9:14PM 1 Dynamic DUNDi weight support in * - HELP!
8:51PM 0 four ringing and hangup with error
7:50PM 0 Help. Help. Help. How to make outbound proxy and host URI with different port?
7:49PM 14 Transfer Call to Cell Phone
6:33PM 0 Does outboundproxyport still work in 1.4.4
6:06PM 0 Outbound proxy setting with outbound proxy port in sip.conf
5:51PM 2 Asterisk 1.4.5, Cisco 7960, call dropped when sip client put on hold/transfer
4:30PM 2 Threading troubles 1.4.5 & IAX2-> SIP (FreeBSD specific??)
3:57PM 0 Set a global queue policy
3:37PM 3 callback and bridge problem
3:31PM 2 two channels, each dropping into the same context, different behavior.
2:41PM 2 asterisk-users Digest, Vol 35, Issue 89
1:26PM 5 Rining 180 and 183
10:19AM 7 Hi ability solution
9:53AM 2 outging select
9:13AM 2 g729 problem
8:38AM 3 Xorcom Bri 4 Port USB
8:27AM 0 asterisk not able to hear calling party ring sound
4:07AM 17 Best wifi IP phone for asterisk
3:52AM 1 Provisioning Linksys WIP330 phones
3:03AM 0 Faktortel in Sydney outage
 
Sunday June 24 2007
TimeRepliesSubject
3:56PM 3 Nokia N95 + Dial Plan
1:16AM 0 OT: Digg post
12:48AM 1 Call Path Optimization
 
Saturday June 23 2007
TimeRepliesSubject
3:48PM 1 Zaptel Compilation Error
3:16PM 8 Zaptel Compilation
1:32PM 2 TAS test equipment manuals
12:52PM 5 IAX client USB phone
11:08AM 1 Dynamic DUNDi weight
10:05AM 2 IVR question for asterisk
6:29AM 0 DTMF Problem with one sip trunk and two carriers
4:30AM 0 modules loading
 
Friday June 22 2007
TimeRepliesSubject
9:43PM 7 FAX over T1
8:39PM 1 Polycom 301 - Problem with AMI Originated Calls
8:22PM 0 asterisk-users Digest, Vol 35, Issue 81
7:34PM 4 1.4.5
6:58PM 1 Binding to multiple addresses
6:49PM 1 H.323 IP Phones and H.323 Traffic
6:28PM 1 Nuance Buys Tegic from AOL for $265m
6:20PM 14 inband DTMF for g729
5:48PM 8 Ring/Off-hook in strange state 6
5:37PM 1 Audio going one way for a few seconds during the call
5:08PM 1 Does Early Media have to be Ulaw?
4:57PM 0 Binding to multiple ports in sip.conf
4:53PM 2 access to asterisk server since internet
4:06PM 0 Hints
3:54PM 1 searching for compatible servers
2:24PM 0 chan_zap problems
12:59PM 6 got-name
12:11PM 1 Friday June 22@12:30PM EDT Asterisk Users Conference
11:41AM 1 POTS - Incoming Voice or Fax - How to tell?
11:00AM 5 problem with one way audio
10:35AM 1 Config for TEI parameter
9:50AM 4 asterisk 0 dial outgoing call
9:50AM 16 Query
9:10AM 4 international numbers...
8:00AM 6 chan_features.so / asterisk 1.4.5
7:20AM 0 RTCP NTP clock skew detected
7:13AM 2 qozap and zt_alarm_notify_no_master_change
5:53AM 1 Once Touch Recording
1:35AM 2 STDERR in AGI
12:50AM 1 hotline with Polycom
 
Thursday June 21 2007
TimeRepliesSubject
10:54PM 0 Looking to buy VoIP or Hosting Company
7:41PM 0 Console channels with two sound cards?
7:17PM 3 Use of ChanSpy
3:49PM 0 Bug in Ex-Girlfriend logic?
3:37PM 0 Forward to my phones the domain of the CALLERID in incoming URI calls
3:22PM 1 TDM800P - zaptel service startup problem
2:51PM 3 identifying what a user pressed to reach my phone
2:41PM 0 ENC: Action Originate (Asterisk Manager) X Monitor()
2:34PM 3 Zaptel wct2xxp driver causes LEDs to go black and RED alarm
2:33PM 2 CDR
1:57PM 0 retreiving callid of call from the dial application
1:35PM 7 AudioCodes Gateway and Asterisk
1:11PM 0 Using Queue - Zap problems (PRI)
1:07PM 2 ChanSkype
9:35AM 2 TDM400 one way calls
8:40AM 2 mediant 2000 with asterik configuration
8:34AM 0 mISDN problems
7:46AM 14 gtalk - no audio
7:10AM 1 Problem with Remote-Hold/MusicOnHold
7:05AM 8 Asterisk config files and #include
6:52AM 0 SIP/IAX2 Phones behind USR 9108 Router
6:24AM 8 asterisk 1.4.1 app_addon_sql_mysql
2:52AM 0 Asterisk 1.2.0 addon Radius
1:06AM 1 install Asterisk-addons 1.4.2
 
Wednesday June 20 2007
TimeRepliesSubject
7:02PM 0 Firewall on AsteriskNow
5:52PM 3 different codec for different extensions
5:29PM 3 Forcing Dial application to skip if called server is unreachable
4:27PM 2 How to Create Custom Context
4:11PM 1 ATT: Brian Fertig
4:00PM 2 X-Lite problems on basic asterisk setup
3:23PM 0 asterisk with mediant 2000 trunk
2:38PM 4 DTMF doesn't work between Asterisk and Cisco SIP Proxy
2:33PM 1 hanging up
2:18PM 1 Asterisk RealTime
12:37PM 1 Res: Record CDR in a Oracle database
12:36PM 0 Agent auto congesting
12:23PM 0 Query regarding connecting PABX with Application server
12:16PM 0 asterisk + mediant 2000
11:42AM 4 zlib1g
11:21AM 6 Single ringer phone for incoming calls, that anyone can answer
7:30AM 0 Error: Unable to allocate RTCP socket: Too manyopen files
6:38AM 0 WHAT happened to AgentMonitorOutgoing(c) in Asterisk 1.4.5 ??
12:05AM 1 ChanSpy SIP
 
Tuesday June 19 2007
TimeRepliesSubject
7:20PM 5 Ex-Girlfriend Logic in 1.4.4
7:02PM 1 Advice
6:41PM 7 Urgent. When the peer returned a 301 forwarded, asterisk thinks it's a local extension.
5:06PM 5 Inline record
1:01PM 18 PhpAgi call generation
12:00PM 4 Execute Chanspy
10:55AM 0 ENUMLOOKUP well succeeded but callee server unreached
10:34AM 1 problem with mISDN
9:58AM 5 Play dial tone withou answer
9:11AM 2 make config
8:36AM 0 peer timeouts and 489s
6:46AM 0 passing privacy information through asterisk
4:27AM 3 Need to increase call count
3:29AM 2 Invalid DTMF detection -- Invalid Extension Bug or issue
 
Monday June 18 2007
TimeRepliesSubject
11:55PM 4 How to config SIP blind transfer in extension.conf
11:49PM 1 AGI command
10:56PM 0 Asterisk-addons 1.2.7 and 1.4.2 released
10:03PM 0 sip <> zap calls choppy, where to setup the jbuffer?
9:57PM 6 180 Ringing with SDP
9:14PM 0 no sound with chan_mobile
6:10PM 2 MixMonitor Timestamp problem
5:37PM 2 SIP Termination with automatic debit
5:15PM 1 High availability Asterisk
4:28PM 0 Monitor recording losing sync
3:25PM 10 Phantom Calls
3:11PM 0 res_jabber over OpenSSL ready for testing
1:40PM 8 Blind xfer issue -- URGENT!
12:52PM 2 asterisk and SAP
12:05PM 3 atxfer attended transfer feature
12:03PM 0 Problem In Installing Asterisk on Solaris
10:50AM 20 chan problem
4:38AM 0 Regarding call transfer feature
 
Sunday June 17 2007
TimeRepliesSubject
9:07PM 5 CNAM.
1:49PM 3 SIP Peering--call terminated prematurely
12:45PM 3 Upgrade cisco SIP phone 7940
9:15AM 0 Mitel 5340 IP Phone
7:08AM 3 asterisk hang (Critical Response)
5:25AM 17 VPN on Asterisk
 
Saturday June 16 2007
TimeRepliesSubject
9:40PM 2 Convert or listen to .sln file
3:11PM 2 ipv6 on Asterisk
2:44PM 2 MixMonitor Problem
2:42PM 0 (no subject)
2:30PM 0 ivr testing script
1:29PM 1 Chanspy
3:47AM 0 Asterisk to Panasonic TDA200 with Unicall
12:44AM 0 Asterisk 1.4.anything on FreeBSD?
 
Friday June 15 2007
TimeRepliesSubject
11:18PM 0 Asterisk 1.2.19 and 1.4.5 released!
11:14PM 1 Accessing Voicemail(Asterisk) -- Remotely either from Cell or Landline
9:36PM 2 calling
9:10PM 0 scaling with SMP
9:05PM 3 Where an extension really is (DUNDi woes)
8:05PM 1 Community PBX?
6:37PM 0 FXS card with 3-way call, transfer and call waiting.
6:22PM 2 combining AGI with dialplans
2:06PM 1 can ENUMLOOKUP query multiple DNS servers without having to replicate the same code for each server?
1:12PM 1 g729 codec
12:45PM 2 Run as root?
10:17AM 0 hangup during voicemail announcement drops all calls
8:33AM 0 Error: Unable to allocate RTCP socket: Too many open files
5:28AM 6 app_rxfax vs (iaxmodem+hylafax)
4:57AM 0 Reinvite / one-way media.
1:07AM 0 No subject
1:07AM 0 No subject
1:07AM 0 No subject
1:07AM 0 No subject
1:07AM 0 No subject
1:07AM 0 No subject
1:07AM 0 No subject
1:07AM 0 No subject
1:07AM 0 No subject
1:07AM 0 No subject
1:07AM 0 No subject
1:07AM 0 No subject
1:07AM 0 No subject
1:07AM 0 No subject
 
Thursday June 14 2007
TimeRepliesSubject
3:02PM 5 My Kernel
2:50PM 4 Unicall + MFC/R2 line dropped immediately after connect
2:26PM 2 question on capacity
12:25PM 0 b410p
11:46AM 22 Que on A2Billing
11:40AM 28 Asterisk GUI
9:22AM 1 ODBC voicemail questions
8:56AM 11 Sending text to a phone that's no in-use ...
8:15AM 0 real-time HINTS
6:59AM 0 ESI Phone System Integration
6:58AM 2 Linksys SPA941
6:35AM 0 Adtran feature codes, extensions
5:35AM 5 Sugar Auto-Dial with Asterisk?
4:47AM 0 (no subject)
3:19AM 2 Qualify renders all SIP peers unreachable
1:45AM 3 TDM400p and te110p configuration.
12:05AM 0 Re: asterisk testing - thanx!
 
Wednesday June 13 2007
TimeRepliesSubject
8:12PM 0 Re: asterisk-users Digest, Vol 35, Issue 52
3:14PM 0 Disconnect tone detection.
2:48PM 1 What is the state of Asterisk Secure Remote Communications?
1:38PM 2 Addons
11:52AM 0 Digium mailing list server maintenance - Thursday, June 14, 5PM to 8PM CDT
10:14AM 0 second dial, force hangup for exit.
10:06AM 5 WAV file best sound quality
8:48AM 1 Weird sip registration problem
7:14AM 4 mISDN problem
6:33AM 2 Polycom + Voicemail + Display message envelope in LCD
5:48AM 9 Using Modems with Asterisk
4:19AM 9 problem starting asterisk, unable to load chan_zap
3:34AM 3 re:zaphfc problem (Josu Lazkano)
2:28AM 1 advanced asterisk logging
2:08AM 1 Voicemail prob
12:39AM 0 zaphfc problem
 
Tuesday June 12 2007
TimeRepliesSubject
3:19PM 0 anyway in meetme to mute all but one user?
2:16PM 1 Realtime Meetme in 1.4
1:38PM 14 Gigabit SIP Phones
12:05PM 4 CDR changes in Trunk -- Transfers, CDRs, Life, and Everything
11:50AM 1 Answering machine detection after Dial()
10:40AM 6 GotoIf Dialplan inquiry
10:35AM 5 Asterisk Faxing
9:53AM 5 Softphone behind NAT issues
8:55AM 3 HPEC and audioclipping
8:15AM 5 Transfer caller direct to voicemail
7:38AM 1 AsterFax
7:38AM 0 [asterisk-tech] ChanSkype
7:32AM 0 Zombie SIP channels
7:28AM 3 Bridge bug in 1.4?
6:59AM 3 Changing the Caller ID
6:56AM 0 Warning on CLI
6:54AM 4 call from ISDN
5:05AM 4 write some custom values to CDR table
4:24AM 5 SPA400 and asterisk
4:21AM 2 No audio after Dial with G option
3:59AM 0 SIP/NAT 1.2 1.4 questions
2:02AM 0 Possible mysql database corruption
1:39AM 0 On multiple dial phones continue ringing after picked up
1:16AM 0 config files to mysql convertion
 
Monday June 11 2007
TimeRepliesSubject
9:21PM 1 Crashes with Spandsp, app_rxfax.c, and asterisk 1.4.4
6:55PM 21 Slightly OT:CSU on Digium cards, and it's requirement
4:39PM 2 which Wifi SIP phones are the good ones
3:13PM 0 sip show registry shows nothing
1:55PM 0 AGI "RECORD FILE" for a video message
12:16PM 1 Introduction to AGI programming
12:14PM 1 CallerID issues
10:58AM 1 Multiple ENUM entries and Asterisk fails to dial
10:45AM 0 Grandstream 4104 - Asterisk (Incoming Calls problem)
7:28AM 0 Asterisk as an SCCP client
7:08AM 0 Different ECs in Asterisk
6:38AM 6 change moh during a call?
6:07AM 0 (no subject)
6:00AM 0 Help on text entry. using asterisk.
5:50AM 3 Searchable List Archives?
5:26AM 3 CDR on transfers of called ZAP channel
3:57AM 1 MOH Problems.
 
Sunday June 10 2007
TimeRepliesSubject
11:56PM 0 Going to VON Stockholm? Meet you at the Asterisk BOF!
11:45PM 3 basic asterisk knowledge
6:18PM 1 best format for audio via asterisk...
9:58AM 2 Blocking 900 calls
7:46AM 5 IAX Peers show command
 
Saturday June 9 2007
TimeRepliesSubject
9:10PM 1 ast_dynamic_str_thread_build_va() is defined with 6 args but only called with 5 args??
7:13PM 4 Polycom sip.cfg / voIpProt.SIP.requestValidation.x.request.y.event
5:21PM 2 Polycom 301 vs. 330
4:25PM 2 OT: CallManager ANI restamp.
1:00PM 2 How to tell what codec is used for each end of a call MD110->H323->SIP
12:57PM 0 H.323 trunk between MD110 and Asterisk
6:39AM 2 No sound, problem is not a NAT
1:08AM 0 Asterisk Users Conference Friday: New Asterisk Book and a visit from JerJer of Nufone
 
Friday June 8 2007
TimeRepliesSubject
7:47PM 0 Softphone for smartphone such as Nokia N90 / 93 / N95
7:44PM 0 CDR accuracy
4:38PM 3 Asterisk 1.4 with Unicall
4:33PM 0 FW: Delivery Status Notification(Failure)
4:00PM 32 Bad Echo between SIP calls
3:34PM 0 Zaptel 1.2.18 and 1.4.3 released!
1:07PM 0 Replacing SX-2000 Centigram Voicemail with Asterisk?
12:02PM 0 Log interpretation
11:47AM 2 Can Asterisk RAS?
9:45AM 9 Write to multiple databases as redundancy scheme
9:11AM 3 choppy sound with playback, background, etc... but not with musiconhold
8:52AM 0 No/unknown event '0' on timer
8:48AM 5 Hot GXP-2000
8:01AM 1 Not getting CID Name from PRI
6:52AM 5 call problem...
5:50AM 0 Asterisk, NAT and canreinvite=yes
2:37AM 0 tdd.c - How does one use this code?
1:26AM 0 Unexpected behaviour shown by "meetme kick confno usernumber"
1:03AM 3 Asterisk & MS RTC Library & Ethernet Capacity
 
Thursday June 7 2007
TimeRepliesSubject
4:38PM 8 IAX trunk with dynamic IPs
3:52PM 0 IAX-configuration
3:42PM 1 RFC-3389 problem
2:32PM 8 agi with java?
2:14PM 1 Q931 Error with H323
1:28PM 13 getting at ${CALLERIDNUM}
1:12PM 1 custom cdr fields and cdr_mysql, howto?
9:44AM 23 Provisioning Linksys PAP2T ATA's
9:35AM 11 Bridged PRI calls - processor involvement?
8:24AM 7 DUNDi and reinvites...
8:23AM 1 Meet Me video conferencing
7:01AM 5 Polycom phone registration problem
6:26AM 0 Need help on Text entry applicaon
6:11AM 0 Need help on Text entry for asterisk through touchpad
6:07AM 0 Need help on Text entry for asterisk through touch pad
5:29AM 0 Realtime Agents.conf
4:18AM 0 atxfer not working
2:20AM 2 AddQueueMember vs AgentCallbackLogin
2:08AM 0 MP3 as voicemail format
1:04AM 3 call Hold event asterisk
12:05AM 0 voice activated recording
 
Wednesday June 6 2007
TimeRepliesSubject
11:44PM 2 iax trunking on OpenBSD
10:43PM 0 SIP buddy watch
9:43PM 0 Solved: [SetAccount in extensions.conf]
7:59PM 6 Best Codec
5:44PM 3 1.4 Zaptel/Sangoma Issues on CentOS
5:17PM 0 SetAccount in extensions.conf
4:26PM 18 Slow list
4:15PM 2 Sending multiline SMS
4:08PM 3 Reload in 1.4 clears regexten
3:41PM 5 Console duplicate output problem
2:05PM 2 Polycoms lose registration and won't re-register
1:34PM 2 Phantom calls: Detecting hangup quicker
12:49PM 2 Voicemail marking messages as Old
12:38PM 1 Stanaphone/Asterisk issue: No Audio with SIP to only one provider when switching servers
11:50AM 2 Record CDR in a Oracle database
11:13AM 0 Queue Job
11:12AM 6 meetme realtime
10:09AM 4 zaptel make problem
9:51AM 3 Needed changes in Asterisk to change the SIP port to 5062
9:14AM 5 TCP<->UDP SIP proxy?
8:56AM 0 Voip-info.org
8:42AM 0 Polycom 320 messages
7:17AM 1 asterisk 1.2.18 problems...
6:39AM 15 PRI Partial Re-Rounting
5:22AM 21 blades?
5:07AM 0 Multiple Ethernet Interfaces
4:25AM 4 shorting flash time
3:48AM 1 CDR changes in 1.4.3?
2:33AM 9 Asterisk call quality detection
12:56AM 2 Zaptel and Libpri Compilation
12:56AM 2 Queue problem
 
Tuesday June 5 2007
TimeRepliesSubject
8:40PM 4 Outlook dialing
7:52PM 1 Recomender Server specs for 250 con-current calls
3:30PM 12 Verizon Interconnection
3:27PM 7 g729
3:23PM 3 Set caller ID based on SIP source.
11:20AM 3 Changing the From field in Asterisk email/voicemail
11:13AM 6 Where to find Polycom firmware with 330/320 support?
10:51AM 9 Hardware spec comparison
10:21AM 2 Cisco 7961G + 7914 Expansion Module
6:46AM 1 spa 3102 incoming call
6:35AM 1 addqueuemember recording and reporting
6:33AM 1 Asterisk on x64
6:16AM 1 Strange beeping sound during fax initiate session
6:12AM 8 NAT
5:57AM 7 X100P Clone
3:28AM 1 Meetme define context
3:26AM 1 Problem to park the call with #700
3:08AM 4 IAX2 Trunk No Sound
2:55AM 0 IS_REGISTERED from dialplan
2:12AM 3 cepstral TTS and app_swift
1:20AM 1 spa 3102 configuration
 
Monday June 4 2007
TimeRepliesSubject
9:55PM 0 chan_sip.c: That's odd... Got a response on a call we dont know about.
9:32PM 25 Noise on FXS ports (Sangoma)
6:45PM 0 Aastra 57i / 57i CT phones fail to re-register
4:56PM 1 Oddity
12:46PM 3 Get calling channel before pickup
12:35PM 5 Detecting card on the PCI Slot
12:28PM 3 answer a voip call, play info.
11:05AM 0 Asterisk 1.4.4 Segfaults with asterisk-ooh323 from addons-1.4.1
10:50AM 2 realtime ldap peer matching
10:33AM 2 cisco 7940 and auto-answer (aastra 480i vs 7940)
10:32AM 4 Calls being dropped
10:04AM 2 Delay in posting of messages to list
10:02AM 0 no ringing tone making attended transfer whith an IAX client
9:53AM 5 Debug meetme
9:20AM 1 AEL2 Includes in Macro...
8:33AM 1 no dtmf pcom 650 only outbound calls
7:41AM 8 background dialing
7:40AM 11 FX Dialing Odd
7:04AM 1 addqueuemember recording and reporting problems
6:04AM 4 debug logs
5:50AM 2 G729 License
5:25AM 0 IAX2 Trunk Problem
3:19AM 3 yum om centos
2:16AM 3 Wireless IP Phone with external Telephone Book
2:12AM 0 Issue with Grandstream ATA 496.
2:08AM 5 Digium Card
1:41AM 0 Mixing Vars into Voicemail WAVs
1:11AM 0 nvlinedetect for Asterisk 1.4
 
Sunday June 3 2007
TimeRepliesSubject
5:57PM 1 Can two card be configured on same machine.
10:29AM 19 wifi sip phone real-world experiences?
10:22AM 0 Strange problem with channel allocation
9:57AM 3 SIP Options Reply Ignored
9:32AM 2 Loud noise instead of MOH
9:27AM 2 Chan_mobile issue
7:03AM 2 FW: Centos kernel source
6:03AM 3 zaptel on CENTOS servercd
4:50AM 14 Centos kernel source
4:48AM 0 Telefonica in Czech Republic Blocking VOIP ?
2:02AM 3 Asterisk Queue
1:42AM 0 Asterisk Crash
 
Saturday June 2 2007
TimeRepliesSubject
4:48PM 2 linksys wip300 firmware
3:19PM 3 Dynamically adding Context in dialplan?
2:38PM 2 System Application, Fail/Timeout Issue
9:37AM 2 how to make busy sign
2:34AM 1 Asterisk registering problem
 
Friday June 1 2007
TimeRepliesSubject
6:53PM 0 WARNING[2160] chan_sip.c: Unknown SDP media type in offer: image 36458 udptl t38
5:57PM 1 Call Back Service
5:36PM 0 OutBound dial plan
3:00PM 0 Gateway VIP450FO and VIP 400FO
1:18PM 4 SugarCRM Integration
9:03AM 4 Cisco 7961G
8:59AM 0 chan_iax2.so issues
8:36AM 4 asterisk mysql support
8:12AM 0 Meetme problems
8:08AM 1 Asteris et winsip
7:51AM 1 G729 client and server Side
6:48AM 0 OT: "The Ignorance of Crowds" (was: OT Slightly: )
6:45AM 5 SIP & NAT ...
5:42AM 0 OT Slightly:
3:04AM 5 how can qualify=yes trigger some external event?
2:31AM 6 ZAP inbound/outbound connection taking too long