asterisk users - Sep 2008

Tuesday September 30 2008
TimeRepliesSubject
11:05PM 5 Asterisk in VM.
8:24PM 0 Transfer a call without announce : no sound
7:32PM 2 OT- NIU Framing
7:28PM 0 How to tell the underlying carrier for your ITSP.
6:08PM 0 Using AMI to View ZAP Channels
3:49PM 0 asterisk-users Digest, Vol 50, Issue 89
3:32PM 10 Asterisk Documentation now on voip-info.org Wiki
2:22PM 4 asterisk app store
2:14PM 2 Question about Asterisk and Java
10:54AM 0 Cisco 7911g
4:42AM 5 OT: real 2 line phone vs. 1 line and call waiting
2:50AM 1 problem with my softphone
12:47AM 6 Maybe OT - routing calls in PSTN
 
Monday September 29 2008
TimeRepliesSubject
7:59PM 2 Channel variables "materializing" ...
5:46PM 0 Cheap FXO Card?
5:03PM 6 How can Block a pri channel
3:33PM 1 Source of SIP "Remote host can't match request NOTIFY"
3:08PM 3 Zaptel Lines - How many are in use..
2:16PM 0 CLI and verbosity level [SOLVED]
1:34PM 1 CLI and verbosity level
12:03PM 0 SIP/IAX Interworking ip CANCEL behavior
11:26AM 5 OT - Avantages of ISDN PtP and PtmP
10:25AM 1 Disable CDR?
10:11AM 6 Knowing incoming call technology and channel [SOLVED]
9:17AM 2 Knowing incoming call technology and channel
8:24AM 0 identify/find a channel to pick it up
8:06AM 32 ATA for large networks
7:41AM 12 Creating Asterisk Binary Package
7:31AM 0 AGI defunct processes + GSM Playback - HELP!
6:00AM 9 uk tole-free dids?
 
Sunday September 28 2008
TimeRepliesSubject
6:11PM 0 Need help with Cisco 7960
8:21AM 4 Conferencing Hardware
2:46AM 1 Vividial issue
1:10AM 0 Eye P Media Soft Phone?
12:54AM 3 G.722 between Eyebeam and a Polycom IP650
 
Saturday September 27 2008
TimeRepliesSubject
10:52PM 7 credit card processing
10:16PM 0 Keeps Ringing After Answer
9:54PM 7 Troubleshooting one-way voice... how to peek into SIP RTP?
8:03PM 0 rtpkeepalive problem ?
7:11PM 14 FW: Google Alert - "dean collins"
4:11PM 0 Asterisk and VoIP educational resources
4:02PM 0 Philippines
11:15AM 3 Set A-Number in Sip Header
10:58AM 2 running out of disk space
9:08AM 5 Problem with pickup extension *8 from features.conf using IAX
7:41AM 19 test call generator
2:20AM 6 Split incoming call volume across queues on several asterisk servers
2:11AM 3 iPhone Sip App
 
Friday September 26 2008
TimeRepliesSubject
11:55PM 2 Audio Files
8:23PM 4 Extremely OT: I need someone who can parse a MS Word or PDF or RTF document
7:04PM 4 Bizarre international call problem.
7:01PM 2 Voicemail retention
6:54PM 2 Dial issue
5:25PM 2 server and 2 uniden phones no ringing
2:28PM 0 PRI TE110P Configuration (Solved)
1:38PM 0 Friday 2008-09-26 12:00:00 Asterisk + Skype on your box
1:28PM 0 Incoming URL handling Problem (Asterisk problem ?)
12:53PM 2 Get Call Length of Calls
12:27PM 1 setting DNID
11:27AM 3 Push presence from one asterisk to another
9:15AM 0 T38 fax gateway announcement
6:05AM 0 Skype channel beta
3:06AM 2 ZAP not answering call
3:04AM 1 Monitoring simul calls
2:59AM 1 Sip reload casuing issues
2:58AM 5 Music on hold for sub tenants
 
Thursday September 25 2008
TimeRepliesSubject
10:18PM 1 users.conf behavior
10:17PM 0 Mysql Command and number rows returned
10:08PM 0 Skype + Asterisk Interview at Astricon
8:29PM 1 Create virtual extension
6:29PM 25 sip forking needed for ekiga 3.0
5:38PM 1 PRI TE110P Configuration
4:58PM 0 Skype-asterisk connection announced (was Astricon people please post the announcement)
4:17PM 31 Astricon people please post the announcement
2:21PM 8 Dial Plan Issues
1:17PM 0 Monitoring trunk
1:01PM 7 OT: Do You Know What the Problem With CDMA is?
12:52PM 1 Ringing after console dsp hangup
12:05PM 0 SIP TLS
10:45AM 0 IMAP voicemail import
10:27AM 0 appconference low quality g729
9:10AM 0 Current available allarms in the Asterisk
7:31AM 8 Terrible Experience Net2phone A-Z termination
6:50AM 0 Problem making international calls
2:35AM 0 What happened to the register= setting in sip.conf?
1:21AM 4 g729 capacity
1:12AM 2 Asterisk 1.4 is asking me for Mailbox #
12:28AM 9 Asterisk on VMware Workstation 6
 
Wednesday September 24 2008
TimeRepliesSubject
10:23PM 2 Zaptel/DAHDI ztdummy only
8:18PM 0 Astricon 08 Videos & interviews & Voiceroute twitters on astricon
6:14PM 0 Timeout question
4:00PM 2 DID mode
2:20PM 1 Asterisk is covering the peers IP address in SIP and SDP messages
11:30AM 2 Voicemail cutting out after about 30 seconds
8:50AM 2 IAX Hangup floods link with repeated VNAK and HANGUP
6:47AM 1 asterisk console: "quit" is twice in history
3:26AM 2 Asterisk mysql CDR
 
Tuesday September 23 2008
TimeRepliesSubject
11:00PM 1 How to send indicating call privacy using P-Asserted-Identity?
9:25PM 0 A2Billing Callback Hangup after/about 20 sec!
9:02PM 6 "No route to destination" error
7:05PM 15 Asterisk 1.4 or 1.6
6:11PM 2 Short question: CPU hardware requirements for Asterisk
6:11PM 4 extension definition
4:36PM 0 Connecting TE212p to NEC XenMaster
3:37PM 0 Linksys 3102 with rfc2833 - NOT WORKING
3:16PM 5 Fwd: more on Free World Dialup groups and FWDLive
2:34PM 0 PRI incoming call forward / call redirect
12:28PM 12 Extension registration
9:57AM 5 chan_misdn troubles
9:52AM 8 AGI and prepaid billing
9:51AM 0 Registration by IP address
8:52AM 0 t38modem on OpenSuse
8:51AM 16 Fax with asterisk
8:36AM 3 Transcoding G.729 files
8:22AM 3 [1.4.21.2] Checking that already off-hook?
3:13AM 2 PSTN Simulator
1:52AM 0 Send us your suggestions on exhibits & tutorials to cover (video) at Voiceroute
12:58AM 4 How to hangup a channel immediately so that it doesn't get charged on cell phone
12:30AM 0 ast_func_write: Function not registered
 
Monday September 22 2008
TimeRepliesSubject
8:46PM 0 E&M wink/no audio
6:02PM 1 I can't call my remote users?
4:58PM 0 GotoIfTime and timezone specification
1:46PM 3 Problem using AJAM on asterisk 1.4.17
12:48PM 1 setvar for outgoing SIP channels?
10:30AM 2 Astricon news online?
4:56AM 9 Seemingly easy question: NPA/NXX
 
Sunday September 21 2008
TimeRepliesSubject
9:00PM 12 How to notify an event to every user
12:28PM 1 Asterisk weird behavior after upgrading
4:49AM 0 Astricon: Throw the dice, give a talk
 
Saturday September 20 2008
TimeRepliesSubject
10:21PM 3 how to add extensions and sip registrations dynamically
5:48PM 2 broadcast ability
3:17PM 0 callwaiting callerid
1:25PM 4 Cisco acquires Jabber
9:28AM 1 1.6.0-rc6 - SIP hold logic broken?
6:24AM 0 [CID] Unknown IE 18/21?
 
Friday September 19 2008
TimeRepliesSubject
11:27PM 2 Specific SIP answers on incoming calls?
9:52PM 1 SVN 1.6.0 / current does not compile
9:05PM 1 Loud noise on Zap port...
8:03PM 0 Last 2 days for early bird tickets to DruidCON 2008, 1-2 Oct in Atlanta GA
7:54PM 2 getting results messages from CLI commands via -rx
7:54PM 2 Dropping Phone Calls
4:40PM 0 Weird "permissions" issue when permissions check out...
2:47PM 2 TE110P or TE120P
2:47PM 0 VoIP Users Friday Conference @12 Noon EDT: Astricon run up meeting and more
2:05PM 0 dundi and zap devices
12:31PM 1 Preventing a call forward
9:29AM 10 SIP request send me 482 error
7:41AM 1 Dialing a 60anything number issue!
7:22AM 0 T38 FAX over a Broadsoft
6:26AM 4 PRI E1 Inbound calls hangup with busy after a few seconds
4:57AM 0 T100P detection.
2:49AM 4 what codec is sip using?
2:36AM 2 Follow Me app question
 
Thursday September 18 2008
TimeRepliesSubject
8:51PM 0 Polycom phones and DNS SRV
7:19PM 2 Old voicemail bounces users
5:09PM 7 device probe order question
4:20PM 7 OT: Cisco 1841 - Can it be made SIP aware?
3:01PM 7 OT - How to stream a A-Law/wav file to a browser ?
2:54PM 5 Custom Voicemail emails
11:42AM 6 Pre-paid Billing
11:41AM 2 How to make a Outgoing Call from Asterisk ?
11:34AM 1 rxfax and txfax
10:25AM 2 Verbosity best practice
10:16AM 4 BRI or PRI callerid
10:16AM 2 Get rid of "Really destroying SIP dialog"
9:46AM 0 482 Loop Detected
7:05AM 1 how to detect pickup...
1:03AM 0 Speech recognition on simultaneous SIP / PSTN calls
 
Wednesday September 17 2008
TimeRepliesSubject
11:49PM 1 strategy for measuring conference audio delay
10:14PM 0 Understanding of SIP Info Messages
8:02PM 3 app_confrence with loud voices
6:37PM 16 Digium training course
5:58PM 6 Restrict SIP registration to one ip address only?
5:25PM 0 Format ulaw|h ?
5:18PM 0 Part of some calls does not get recorded
4:25PM 3 DTMF detection problem on DISA
4:23PM 0 How to remove dialtone from DISA?
1:57PM 5 chan_iax2.c: No more space
1:28PM 4 pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use
1:12PM 2 codec of channels
9:50AM 4 dtmf passthru
9:45AM 2 SIP URI Forwarding
8:58AM 0 Asterisk 1.6.0-beta5 voicemail problem
8:57AM 1 realtime queue asterisk 1.6.0-beta5
8:04AM 1 Cellroute setup with asterisk
6:27AM 23 Help with MFC/R2
 
Tuesday September 16 2008
TimeRepliesSubject
6:39PM 0 iax reject using domain name
6:27PM 8 Parked Calls
6:04PM 7 Cisco + Asterisk
5:10PM 2 addons will not load when compiling Asterisk with DEBUG_THREADS
3:44PM 2 What is worng with that include in contrast to the example
2:21PM 1 Snom phones and P-asserted
2:20PM 0 One Week Until AstriCon 2008
12:52PM 1 Voicemail: Thunderbird extension to play wav file in attachment?
12:50PM 3 how to force Asterisk 1.4 to use soxmix
9:51AM 6 What is in practice the maximum no of simultaneous calls that Asterisk 1.4 can handle
9:49AM 0 Redirecting SIP RTP with one Asterisk behind a NAT. The other is on Public IP
2:44AM 2 dundi
2:11AM 1 Context always defaults to PSTN
 
Monday September 15 2008
TimeRepliesSubject
11:38PM 0 rc6: Dunno what to do with STUN message 0101 ??
10:37PM 5 RTCP-XR
8:13PM 0 Looking for an Asterisk consultant recommendation - LONDON, ON, CANADA
7:54PM 0 What characters can be present in SIP dial string passwords?
7:46PM 1 Need help with PHP script to authenticate user from database
5:22PM 1 FW: open source PBX survey
4:47PM 1 [FreeBSD] Right way to upgrade Zaptel from ports?
1:50PM 7 Callcenter monitoring tool
12:22PM 8 PBX appliances
11:06AM 7 UK call initiating party hangup control on analog home lines
11:00AM 2 Asterisk
9:25AM 1 call files hacking...
9:20AM 0 [OT] email netiquette (was: Re: Re: Asterisk realtime MySQL clients from same IP problem)
3:48AM 14 Setting up Asterisk to make calls using a VoIP provider and the regular phone line
1:43AM 0 asterisk-users Digest, Vol 50, Issue 38
1:28AM 1 CallerID Resolution
12:45AM 0 asterisk-users Digest, Vol 50, Issue 37
12:29AM 0 Degum Hardware Echo Cancellation
 
Sunday September 14 2008
TimeRepliesSubject
10:59PM 0 MGCP Configuration <=> ADIT 600 <=> T1 Port
8:34PM 1 MoH with an Aastra 9112i
11:46AM 2 Read dublicate dtmf
4:56AM 16 Streaming MoH on 1.4
12:40AM 14 Can someone give a plain english explanation of the HASH function?
 
Saturday September 13 2008
TimeRepliesSubject
10:19PM 2 What if some phone picks up
9:53PM 4 Queue Calls getting stuck in there
9:52PM 0 Getting realtime ASR and ACD from Asterisk
8:14PM 0 Help...Failed to initialize G.729 copy protection!
7:25PM 4 rc6: can't get a dialtone
2:36PM 0 how to monitor traffic against specific Calling ID
2:22PM 4 Newber
11:58AM 0 Which internet phone protocol best to, choose
10:43AM 0 app_conference
1:50AM 0 Can the outbound SIP leg Call-ID be set to match the inbound SIP leg Call-ID?
1:47AM 5 Is there a way to get the remote User-Agent info from an outbound leg?
1:39AM 2 Append String to CIDNAME
12:27AM 5 cdr_adaptive_odbc writing CDR before h extension is processed
12:13AM 2 Sip Info events
 
Friday September 12 2008
TimeRepliesSubject
10:44PM 1 [FreeBSD 6.3/Ports] Make does nothing
9:35PM 0 OpenStage20 Problem
9:00PM 1 SIp Signalling
7:55PM 0 Transfer via AMI
6:10PM 0 echo cancellation problem with dahdi
5:15PM 3 Setup speed dials on Cisco 7921
5:03PM 0 Encrypted IP phone compatible with Asterisk
4:51PM 1 SCCP - max lines per phone limit
3:28PM 15 Which internet phone protocol best to choose
3:19PM 2 SCCP port numbers used for audio stram?
2:41PM 1 Extension not found
12:29PM 0 Dial function, and no telephone line fixed in the fxo port
11:52AM 0 show g729 seems to no longer work in latest 1.4 version. What do I use please?
7:59AM 15 Amazing "show uptime"
6:59AM 0 VoIP Users Conference today at 12 Noon EDT
5:07AM 3 how to pass a variable in extensions.conf to AGI file
3:40AM 16 Asterisk and Fedora 9
 
Thursday September 11 2008
TimeRepliesSubject
9:57PM 0 Possible Packet loss but need an opinion
8:18PM 5 Unable to run make menuselect for asterisk-addons
7:29PM 1 Probably very simple... call a number and play a sound?
7:18PM 27 about application Jack and its runtime
3:56PM 0 Asterisk calleri id resolution
3:46PM 4 dahdi vs zap with latest version of asterisk -- having some problems
3:23PM 0 g729 passthrough
2:58PM 0 [Re: Asterisk CDR Problem for Export CSV (Asterisk-stat-v2)]
2:57PM 4 asterisk 1.6.0rc6 make menuselect failed.
2:00PM 1 IVR response of the pound key
1:31PM 4 Outside SIP Caller accessing voivemail
1:30PM 0 Sarfaraz has invited you to join iDeezire - Keeps Connected!
1:28PM 0 redirection of called
11:37AM 6 BLF call pickup on Linksys SPA932
9:58AM 4 meetme without zaptel
8:09AM 5 SHELL function strangeness
7:59AM 0 Language for app_queue, chan_local, chan_agent or whatever?
1:10AM 2 distinguish trunk from same host
 
Wednesday September 10 2008
TimeRepliesSubject
11:56PM 0 Zero time (nearly) call transfer.
10:54PM 0 Is there a way to get the Call-ID into the CDR?
6:22PM 8 Write Asterisk CDR MySQL records to multiple servers
5:21PM 2 Bell Canada (Nortel DMS100) PRI Outbound CNAM issue
1:02PM 2 Resilience using DNS or phone feature ?
11:55AM 1 How to make outgoing call from Pri ?
8:10AM 6 Newbie AEL2: Syntax for Hint
2:56AM 4 Asterisk and cloud computing (amazon EC2 + S3)
 
Tuesday September 9 2008
TimeRepliesSubject
11:30PM 0 nagios check_sip plugin
10:23PM 2 Asterisk phone conferencing performance
10:04PM 3 Pressing 0 to get an external line
9:40PM 2 Asterisk REFER
9:34PM 0 Asterisk 1.4.22-rc5, 1.6.0-rc6 and Zaptel 1.4.12.1 Released
7:19PM 7 SIP to IAX?
7:00PM 5 DruidCON 2008, 1-2 Oct in Atlanta GA, 2 free DruidCON conference passes to be given away!
6:15PM 0 Choppy Audio in One Direction
4:49PM 0 CLI and AGI question
4:30PM 3 Asterisk - Operator switch billing
3:30PM 0 AstriCon 2008 - Two Weeks To Go - Register Today
3:13PM 3 PRI auto-configure - continued from DEV list
1:42PM 1 Does X-Lite 'remember' Congestion state? (halfway OT)
12:50PM 13 Asterisk and Network Monitoring
11:25AM 0 Manager API -> call to agi
11:15AM 0 Call-Limit on Asterisk Cluster
10:50AM 3 Asterisk CDR Problem for Export CSV (Asterisk-stat-v2)
9:17AM 0 Session Progress
8:50AM 0 cli-originate and features
6:41AM 3 Dundi Help
4:34AM 1 Help needed creating gateway
4:25AM 0 Aastra Phone Parts
 
Monday September 8 2008
TimeRepliesSubject
11:50PM 1 Digium Hardware Echo Cancellation
11:12PM 8 OT: ARI
9:45PM 0 New Versions of Asterisk and DAHDI
8:25PM 0 Auto Attendant help
7:43PM 4 SIP Extension Config Issue
7:08PM 1 Asterisk T38 and Dialogic DMG 2000
6:11PM 2 Multihomed Server Issues
5:43PM 6 Pointers to replace astdb
5:07PM 0 Newbie questions: seting up extension for miSDN
4:32PM 1 [OT] Re: Asterisk realtime MySQL clients from same IP problem
4:00PM 0 How to read DTFMs from MEETME_AGI_BACKGROUND without blocking?
3:11PM 5 Originate
3:11PM 1 Help Astmanproxy - AutoFilter
2:15PM 6 Video on Hold?
1:51PM 6 Asterisk realtime MySQL clients from same IP problem
12:50PM 3 Help about the Rxfax on asterisk
11:06AM 2 mISDN or BRIstuff ...
9:35AM 1 Which kernel mISDN to choose
8:39AM 4 how to disallow the native bridge between the two channel
6:01AM 0 CPU Usage 100% when Voicemail Notification is sent
5:32AM 0 Streaming live music into a conference room
 
Sunday September 7 2008
TimeRepliesSubject
8:38PM 8 iLBC and G729 codecs
8:34PM 2 IAX vs SIP
1:22PM 2 error while trying to compile dahdi-tools-trunk
1:52AM 11 Problems with 2 Asterisk servers on same LAN
 
Saturday September 6 2008
TimeRepliesSubject
6:46PM 8 realtime queue reload
5:17PM 2 [FreeBSD 6.3] Right-way to recover Zaptel?
1:55PM 0 call-limit problem
 
Friday September 5 2008
TimeRepliesSubject
4:19PM 19 FAX over T1 Question
2:38PM 1 Dear asterisk-users@lists.digium.com 79% OFF on Pfizer
2:32PM 2 svn branches for dhadi and its tools
1:36PM 0 FW: Vivox SLim
12:20PM 1 Call-leg stays on MusicOnHold forever
11:37AM 0 Grandstream Video Phones & Asterisk..
11:27AM 5 Bridge 2 incoming calls
2:58AM 14 dahdi & tdm400p: no luck
2:40AM 0 Linksys 3102 - Call Waiting
1:43AM 0 libpri 1.4.5 priindication
1:43AM 1 The question about the M(X)option of Dial
12:09AM 9 Polycom BLF - multiple buddies
 
Thursday September 4 2008
TimeRepliesSubject
8:44PM 2 How to setup SIP so that RTP traffic flows from Source to destination
7:28PM 1 DAHDI FAQ not up. Anyplace else?
5:59PM 2 1.6rc4 chan_iax2 messages
5:45PM 3 extensions.conf programming?
3:16PM 7 ringback when the channel is answered
2:36PM 0 strange transfer problem
2:01PM 0 conf files for dahdi
1:22PM 0 Logs: messages, events, queue
1:03PM 0 New Install using DAHDI
12:10PM 0 MixMonitor + Originate
11:14AM 1 dial out via fxo gateway
10:55AM 2 iLBC codec
10:28AM 3 How to check mailbox exists (Received SIP subscribe for peer without mailbox)
10:02AM 0 Installing ValetParking?
9:25AM 1 #include changes in 1.4
9:12AM 1 Stability problems in Asterisk 1.4.18 (and other 1.4.xx versions)
8:15AM 1 Dial L( x [: y ][: z ]) option truncates colon (:) using AGI /_
5:42AM 2 Z-Wave or Zigbee for Office or Home automation using XML Browser enabled Screen Phone
3:58AM 9 ASTERISK supported Video phone
2:46AM 0 Voicemail "from an unknown caller"
12:12AM 2 All calls want to go out only on interface ZAP/g0
 
Wednesday September 3 2008
TimeRepliesSubject
9:31PM 0 Combine sip audio and video from different sources
9:18PM 7 New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI
8:37PM 1 Asterisk with E1 interface vs IP PBX
8:20PM 8 DID number
8:12PM 2 Ringing on Console after a page
7:41PM 0 Asterisk voicemail message order
6:55PM 17 G722 and Asterisk 1.6
5:57PM 0 sip to sip unplanned conference! help!!
4:53PM 0 SIP TLS / Nokia E51
3:24PM 0 res_cepstral.so
2:14PM 8 Asterisk Crash
1:18PM 2 Live operator as a service?
8:59AM 1 MixMonitor-Saving Recorded file with AgentId.
7:13AM 1 multiple passwords for one meetme!
6:50AM 1 Newbie Polycom: ACD AgentLogin display on phone
5:30AM 0 Offering FIFO service to receptionist with LIFO hardphone ...
 
Tuesday September 2 2008
TimeRepliesSubject
11:20PM 2 Mark Spencer on TWiT's FLOSS Weekly with Leo LaPorte & Randal Schwartz
11:16PM 2 Selectively disable echo cancellation?
6:58PM 3 Dial timeout to cell phones
4:41PM 0 SALE 71% OFF on Pfizer
3:07PM 5 AgentCallbackLogin AddQueueMember
12:56PM 7 Asterisk Trunk and normal
12:05PM 0 zaptel 1.2.27 ?
9:36AM 1 SetCallerPres
8:44AM 1 play remote file
5:19AM 0 Dialplan terminates when the caller hangs up
2:01AM 3 Redundant PSTN PRI Gateways using Asterisk
 
Monday September 1 2008
TimeRepliesSubject
11:08PM 0 Still badly in need for mISDN help!
5:11PM 1 Documentation of users.conf
2:15PM 4 Asterisk 1.6 beta
10:43AM 10 lists.digium.com monthly reminders
9:07AM 3 Problematic Trunk SIP: Got SIP response 405 "Method not allowed"
8:57AM 0 not able to make call to landline no...to mobile works fine
8:41AM 0 Penalties for agents
7:26AM 9 Gateway errors
3:28AM 0 PhoneControl integrations