asterisk users - Jun 2009

Tuesday June 30 2009
TimeRepliesSubject
10:00PM 1 Authentication Issue Between Servers
8:39PM 1 MeetMe not prompting for PIN
8:36PM 0 Asterisk & Adit 600 Configuration
8:17PM 10 Extension status as XML for an Aastra 57i
7:21PM 3 Intercepting a Call while ringing a device
5:42PM 1 Reception of vocal SMSs to landlines.
2:47PM 3 IAX2 help needed...
2:33PM 1 Setting CDR(userfield) from Macro called from feature doesn't work with cdr_mysql
1:52PM 0 Problem with DTMF detection in ast_app_getdata (*1.2)
1:21PM 1 Asterisk 1.6 WaitForSilence Problem
12:57PM 4 Asterisk module trouble
10:51AM 0 Redirect with ExtraChannel on Bridged call give AMI event with second channel name AsyncGoto/...<ZOMBIE>
10:45AM 24 Echo and static on PRI with errors
10:01AM 1 DUNDi Errors (ENCREJ)
9:22AM 1 Opensips+asterisk problem
9:13AM 4 Dial Chan_local Usage
9:02AM 3 Echo and static on PRI with errors.
9:00AM 0 Can I add one h323 endpoint to register at asterisk?
7:03AM 1 Question regarding SIP 183 "Session Progress" handling in Asterisk
6:38AM 1 Remote UNIX Connection Hanging Asterisk
2:42AM 2 cisco phone 7911 &
1:59AM 0 Resetting CDRs on inbound calls
1:22AM 0 Restricting domains with SIP Trunking
12:44AM 5 Queue Issue (1.4.21.1)
 
Monday June 29 2009
TimeRepliesSubject
11:24PM 7 Calling Number Verification Number? for BellSouth/AT&T
8:01PM 0 Asterisk ended with exit status 134... Asterisk exited on signal 6.
6:13PM 3 OT: Mobile voip - WCell
5:26PM 4 CRMy type app?
2:47PM 2 ISP< ->Asterisk <-> ATA <->DIALUP
2:33PM 2 Force Authentication
1:49PM 11 Calling non-extension numbers issue
1:32PM 0 Transfer dropping calls
12:15PM 0 asterisk 1.4.21.2 a caller waited in queue, after connect to agent hears silence
9:51AM 6 how to sniff RTP and SIP traffic only
9:07AM 5 unwanted locution
7:35AM 0 underlying sound during sip calls
6:50AM 1 about monitored calls storing
6:49AM 4 Callwithus.com is discontinuing IAX service
1:45AM 0 FW: re: Asterisk Outbound with Failover, alarm notification, dial status and hangupcause capturing to CDR from Dialplan
 
Sunday June 28 2009
TimeRepliesSubject
8:57PM 0 BUG in Asterisk 1.6.1.0 and issue in DAHDI 2.1.0.4
4:54PM 0 Recommendation / doubt about building of dialplan
4:36PM 0 Looking for real world uses of CallbackAgentLogin()
9:33AM 3 Sangoma A200
 
Saturday June 27 2009
TimeRepliesSubject
8:25PM 0 IPv6
6:48PM 0 1.6.1: unable to create channel IAX2 to Junction
6:19PM 0 Autoreply: Re: using http to provision a Grandstrea GXP2000phone
5:27PM 2 using http to provision a Grandstrea GXP2000 phone
9:33AM 1 2 problems I can't solve without any help
9:06AM 7 Skype for Asterisk. Any return of experience ?
5:50AM 1 Multiple parking lots use default park positions
3:01AM 4 Call Parking timeout fails
2:00AM 0 Audio distorted local side only
 
Friday June 26 2009
TimeRepliesSubject
11:48PM 8 IAX for internet file transfer?
7:17PM 1 G.729 licence in devices connected to Asterisk
7:12PM 2 Sounds format: GSM to G.729
6:56PM 0 Higher Ed/University users of Asterisk? Free gift!
4:51PM 0 format_mp3.so in 1.6.1
4:03PM 3 HW recommendations for small, cheap, reliable server
4:00PM 1 registration failed, not a local domain
3:23PM 0 Problem with RetryDial
3:10PM 9 T38 Fax Gateway for Asterisk 1.6
2:44PM 2 Normalize Voicemail Volume?
2:34PM 0 Background command not interrupted as desired
1:55PM 0 Friday at 12 Noon EDT: VICIDIAL
12:40PM 1 NOT chan_mobile
12:20PM 0 Problem loss 2 seconds audio when Packet2Packet bridging
8:42AM 5 Centrale FastAgi server down
1:55AM 2 Calls dropping
 
Thursday June 25 2009
TimeRepliesSubject
9:09PM 1 Persistent dynamic queue members
7:47PM 3 res_cepstral, register & existing Cepstral licenses.
5:56PM 3 video call doesn work
5:17PM 0 asterisk-users Digest, Vol 59, Issue 62 (Should be my PHP/AGI problem and odd behavior)
3:25PM 1 SIP registration fails
3:05PM 3 asterisk-users Digest, Vol 59, Issue 62
2:08PM 0 AMI Transfer?
11:43AM 0 iaxclient softphone: quality?
9:40AM 2 hotdesk and voicemail
 
Wednesday June 24 2009
TimeRepliesSubject
10:53PM 3 dahdi-linux-2.2.0 compile problem
10:37PM 0 How to set the II DIgits?
9:11PM 6 Removing line 2 from CISCO 7940g
7:43PM 13 PHP AGI Not Working and Odd Behavior
7:20PM 15 GUI for Asterisk
3:10PM 0 DAHDI Linux 2.2.0 and Tools 2.2.0 Release Announcement
2:10PM 0 Sangoma A200 and bristuff how install
11:53AM 0 Working chan_mobile/bluez anyone?
11:45AM 7 Announcement: Howler-optimised G.729A Solution for Asterisk
11:01AM 1 Outgoing CallerID for KPN in Belgium
10:21AM 1 Message Waiting Indication Astersk and kamailio
9:52AM 2 Asterisk + Jabber
8:08AM 0 problem with sangoma card a108d
5:12AM 0 Avaya 5610 SIP Firmware
5:10AM 0 Avaya 4620 SW SIP Config - not setting Proxy/Registrar
4:22AM 0 Are there any patches for chan_h323/chan_ooh323 to support video?
 
Tuesday June 23 2009
TimeRepliesSubject
9:10PM 5 1000Hz kernel
4:03PM 1 ADM v. homemade code
3:53PM 0 PRI cause code discrepancy
2:49PM 2 [extensions.conf] Any idea why not working as it should?
2:16PM 0 Asterisk/Digium Press Opportunities: Community Contacts
12:09PM 7 SIP 482 Loop detected
10:17AM 8 GSM mobile trunks
8:31AM 5 error in playback of voiceprompt????
7:31AM 7 music on hold file formats
1:00AM 2 How to force TDM410 to alaw
 
Monday June 22 2009
TimeRepliesSubject
9:12PM 0 Documenting configuration with Real Time
5:20PM 6 Learn Asterisk
3:57PM 1 RTP/SIP traffic prioritization and Linux issues
2:25PM 4 Different inbound routes for each interface on a TDM800P card.
1:27PM 5 Crash process Asterisk
6:51AM 7 Realtime extensions
5:19AM 2 [hylafax-users] "No Carrier detected" sendig fax with Hylafax-Iaxmodem-Asterisk
 
Sunday June 21 2009
TimeRepliesSubject
9:23PM 0 Asterisk and h323 gatekeeper functionality
11:33AM 0 Remote-Party-ID implementation on trunk
3:49AM 17 Nobody picked up in 20000 ms
2:15AM 3 Limit transfers
12:15AM 2 Meetme Talker Optimization
 
Saturday June 20 2009
TimeRepliesSubject
10:51PM 0 Configuring Deltacom pri in Florida
10:15PM 1 PRI cause codes
7:48AM 0 Join Asterisk Global Meeting Sunday June 21 using VOIP - BerkeleyTIP
6:22AM 1 Fw: RE:Nagios under *[solved]
3:24AM 3 newbie questions
 
Friday June 19 2009
TimeRepliesSubject
8:54PM 5 asterisk 1.6 and mISDN
7:08PM 2 IMAP voice mail storage
5:02PM 1 Switchvox HA options
3:25PM 2 Strange res_config_odbc error messages in 1.6.1.1
11:43AM 0 Asterisk and EC2 today at 12 Noon EDT
8:58AM 2 agent login status visual clue on Polycom?
8:34AM 12 Cisco 7941G & Auth
8:04AM 2 help setting tone zone
6:45AM 8 Dail in modem
5:59AM 0 Asterisk Flite Problem
4:28AM 2 Anonymous Connection form IP to use specific Context
4:24AM 2 Recompiling dahdi-linux after kernel update - To minimize downtime
3:18AM 0 Analogue card recommendation
 
Thursday June 18 2009
TimeRepliesSubject
9:47PM 5 asterisk-gui: http://id_address:8088/asterisk/static/config/cfgadvanced.html
9:18PM 1 Multiple Outgoing Lines: extensions.conf (Ioan Indreias)
6:27PM 4 asterisk-gui: read/write in the conf files or db
6:15PM 5 Nagios under *
5:49PM 4 Voicemail Password
5:40PM 2 dahdi and overlapdial problem
4:41PM 6 Speex problem installing on CentOS 5.3
3:08PM 9 Asterisk + mySQL
2:21PM 1 Configuring Asterisk behind a SIP Proxy
11:25AM 16 snom mass deploy help
10:19AM 2 Multiple Outgoing Lines: extensions.conf
10:08AM 10 Asterisk on AVR32
9:12AM 2 Asterisk 1.6.1 and dahdichanname = no
8:31AM 8 asterisk and openvpn and sip
6:35AM 0 failover trunk config.
6:16AM 3 how can I get Better natural Voice in Festival
3:06AM 3 Incoming SIP and the 's' extension
2:41AM 5 help setting up transfering
 
Wednesday June 17 2009
TimeRepliesSubject
11:10PM 2 Redundant Connectivity
11:07PM 3 Wideband (G722) MeetMe
10:48PM 6 What causes this error?
10:35PM 2 asterisk-gui: read/write in the conf files or db?
10:06PM 2 Function IMPORT and Local channels
9:48PM 0 File Permissions On Voicemails Left To Multiple Recipients
7:20PM 1 Incoming Call trouble with new *Now 1.5 setup
6:34PM 2 Nagios Asterisk
6:03PM 3 gap between Playback and Queue
4:14PM 11 Asterisks, Sip to Local PRI/PTSN issue
3:27PM 8 Polycom Stop Downloading Config
1:03PM 1 Debug: how to print a variable?
12:16PM 7 Scaling
10:13AM 1 MP3 File Play In Read application
7:22AM 4 modifying CID for different trunks
2:09AM 23 Installing LUA
 
Tuesday June 16 2009
TimeRepliesSubject
8:57PM 1 OT: Possible Fraud-Mike Low/Zigit/ZonFon/CallCheap
8:15PM 0 the correct way to setup a transfer with REFER in SIP
1:39PM 6 no sdp or contact replacement using externip
11:22AM 5 missing chan_dahdi.o in debian asterisk 1.4.x
11:02AM 7 Update Caller-ID after Dial()
7:29AM 4 Unable to use # as feature key prefix
7:23AM 6 feature keys no longer work after a call has been parked
3:46AM 2 No exten available after pass between servers
1:00AM 3 tdm loosing interrupts and latency
 
Monday June 15 2009
TimeRepliesSubject
5:28PM 3 Newbie, Question on making a PSTN call..
5:06PM 0 Asterisk 1.6.2.0-beta3 Now Available
4:31PM 0 Bug or feature : how to customize SIP REFER from dialplan
3:25PM 0 Open Source Call Statistics / Metrics Packages
1:46PM 1 Suggest Multi-tenant Predictive Dialer ?
1:44PM 2 asterisk and google talk
12:31PM 0 Suggest Multi-tenant Hosted PBX ?
11:55AM 2 Click-to-dial CTI for Windows
9:47AM 0 external RTP IP address
9:13AM 0 OT - Aastra - mapping transfer key
9:06AM 2 How to remove a GLOBAL variable from diaplan ?
9:03AM 4 Function IMPORT
8:37AM 0 setting codecs on the fly
7:57AM 3 Opinion on Attended transfer in features.conf
 
Sunday June 14 2009
TimeRepliesSubject
9:31PM 0 DNS queries based on channel name?
6:13PM 12 Open Source Soft Phone
1:15PM 0 No voice from the callee
2:13AM 2 FXS - TDM400 - No dial tone
 
Saturday June 13 2009
TimeRepliesSubject
11:31PM 7 Polycom registration errors
3:27PM 2 Dial with r option doesn't use 'ring' tone as defined in indications.conf
1:59PM 1 Is it possible to do this? (forward a call w/ 3-way calling)?
11:24AM 1 Preventing MOH from restarting the tune when a call is parked
4:23AM 2 1.6.0.10: core restart on ReceiveFax()
1:01AM 0 Fedora Core 10 and g729 codec
 
Friday June 12 2009
TimeRepliesSubject
7:48PM 4 AmooCon video recordings online
3:40PM 1 Asterisk + TC400B - Clock Trouble
3:15PM 2 sending sip info messages
1:58PM 27 Help building dahdi for debian
12:01PM 0 Friday 12th June @ 12 Noon EDT: VoIP Users Conference Skype to ZipDX
8:49AM 8 multiple PRI's in one group ..how??
8:06AM 2 Simple Queue Problem
7:38AM 0 PRI connection with ZTE exchange over T1 PRI
7:19AM 0 Dynamic DNS (was asterisk-users Digest, Vol 59, Issue 28)
6:51AM 1 asterisk-users Digest, Vol 59, Issue 28
5:03AM 0 FXO and fax-on-demand
4:11AM 0 Problems with ReceiveFAX (asterisk 1.6.0.3 and t38)
1:39AM 5 Current possible values for DIALSTATUS?
12:50AM 2 Writing for asterisk
 
Thursday June 11 2009
TimeRepliesSubject
9:17PM 0 Polycom Digitmap
7:30PM 3 SIP hacked connection?
5:47PM 4 Automatic Calling Feature?
11:45AM 0 Asterisk 1.4.26-rc2 Now Available
7:59AM 7 cant use h,1 at cancel!
7:21AM 4 OT - Aastra phones provisioning
6:03AM 1 cisco MC3810 weirdness with asterisk
12:02AM 2 In Dahdi: what we use instead of /sbin/ztcfg -vv
 
Wednesday June 10 2009
TimeRepliesSubject
11:46PM 1 PrivacyManager no longer working properly
10:00PM 5 Asterisk - SIP - TCP and Exchange 2007 Unified Messaging
8:44PM 2 problem with transfer application (REFER)
3:14PM 3 T38 support
3:04PM 5 Dialer program
2:42PM 0 Dial option limit call duration
2:24PM 0 sip calls not going through
2:00PM 2 Chameleon Mail
1:40PM 2 Resetting Marker Bits
1:19PM 4 External PRI Appliance
1:16PM 1 Rhino analog cards
11:17AM 8 Query about tdm410 cards
10:42AM 0 Problem with attended transfers
9:48AM 7 Problem with voicemail and NFS
8:27AM 0 optimising asterisk sounds for g722
6:38AM 0 DAHDI and ZAPTEL for automatically start (rc.local)
 
Tuesday June 9 2009
TimeRepliesSubject
9:55PM 0 zap not coming online on fedora 8
7:58PM 0 Help - create_addr_from_peer: 'UDP' is not a valid transport for 'exten1'. we only use 'TLS'! ending call.
6:58PM 5 IAX2 issue?
1:32PM 14 voicemail
12:32PM 0 FXO- no dial tone- no call progressing
10:51AM 8 hfcpci with 1.6 ?
8:08AM 2 Bitwise AND
2:42AM 0 alsa no input
 
Monday June 8 2009
TimeRepliesSubject
7:32PM 2 Help with asterisk core dump
5:48PM 0 SendText and sipsak
4:34PM 0 How to use Dial G option in AEL
4:29PM 9 OT: Grandstream, call pickup, ...
1:39PM 3 Asterisk VM and Android phone?
1:18PM 3 MeetMe: Mute All Lines Automatically?
12:32PM 1 SIP Strict Routing and canreinvite
12:18PM 2 Asterisk manager login with java not working
11:57AM 10 Best free text to speech..
11:32AM 8 Timeout when dialing dead peer
11:06AM 9 T.38 pass-through 488 handling problem
10:25AM 0 Push to Talk with Call Drop-Out?
8:03AM 3 Snom, Asterisk and Patton SN1400 - sending bye instead of hold
5:43AM 1 Achoring MEdia
1:10AM 0 remote queue members
 
Sunday June 7 2009
TimeRepliesSubject
4:51PM 5 Call recording in - out
4:21PM 0 BUSYDETECT_* flags
2:35PM 1 ANI
12:53PM 0 Callback with a2billing
10:19AM 4 chan_dahdi missing in * 1.6.1.1
8:20AM 2 Called party name with Cisco-2,811 gateway
 
Saturday June 6 2009
TimeRepliesSubject
7:05PM 5 Teliax: where's the space in CALLERID(num) from?
6:47PM 5 Sangoma A200 wt HW echo on 64bit Centos , 4 fxo and crosstalk!
3:14PM 1 What does it mean rc in the release version
1:08PM 1 FXO clock
9:33AM 6 DAHDI, and 64 bit machine
 
Friday June 5 2009
TimeRepliesSubject
10:31PM 0 Asterisk 1.2.33, Asterisk 1.4.25.1, Asterisk 1.6.0.10, and Asterisk 1.6.1.1 Now Available
9:25PM 13 How run AsyncAGI commands in background
10:55AM 1 Help with inbound dialplan
8:06AM 0 Friday June 5th @12 Noon EDT: Sipgate invades the USA, more HD Voice, Video chat
1:34AM 3 DTMF Problem w/ MeetMe
 
Thursday June 4 2009
TimeRepliesSubject
10:12PM 16 PHP/AGI/SetVar Issue
7:26PM 4 DECT USB dongle - an Asterisk channel?
6:24PM 9 broken pipe in perl agi
6:08PM 9 Phones dropping registration, but asterisk thinks phones are still registered
4:15PM 17 Asterisk AGI issues (at high load)
3:23PM 7 Question about core CDR system for multilpe servers
12:56PM 15 Digium Fax Driver
12:00PM 1 problem install Asterisk-FastAGI
9:27AM 1 how we can put anybody on hold using Asterisk with analog phone
7:40AM 1 CDR question
4:38AM 5 Asterisk eventually fails when connection dies
4:22AM 4 Meetme timeout
 
Wednesday June 3 2009
TimeRepliesSubject
6:38PM 4 Using DIALSTATUS question
5:01PM 0 RES: RES: SIP Response 181 - Is it possible in A steri sk?
3:34PM 2 IAX2 Channel Information
2:00PM 5 Can asterisk work here
11:51AM 0 Could not stop autoservice on calling channel
7:37AM 24 IP phone recommendation
 
Tuesday June 2 2009
TimeRepliesSubject
8:13PM 1 does transcoding take place when a SIP call (both ends using same codec) gets forwarded over T1?
7:15PM 0 PSTN Interface Card
4:14PM 0 Segfault on unload of chan_h323 in asterisk-1.4.25
3:51PM 0 Delivery reports about your e-mail
3:14PM 10 Realtime LDAP passwords
2:41PM 0 DAHDI Linux 2.2.0-rc5 and Tool 2.2.0-rc3 Release Announcement
2:20PM 11 RES: SIP Response 181 - Is it possible in Asteri sk?
1:38PM 2 SIP Response 181 - Is it possible in Asterisk?
1:31PM 1 Asterisk maximum user
1:14PM 14 Call quality - how to debug
10:07AM 0 problem with outgoing calls
8:52AM 1 zaptel to dahdi
8:06AM 3 error with dial timeout
6:10AM 0 Polycom IP321?
12:08AM 1 First ever Open Source Asterisk / Wave bounty
 
Monday June 1 2009
TimeRepliesSubject
11:07PM 0 Wave and Asterisk
7:47PM 4 extensions not being detected consistently
6:52PM 0 Suddenly the voice became karbage (like robot) using Asterisk
6:47PM 1 Asterisk 1.4.26-rc1 Now Available
6:31PM 0 Astlinux 0.6.6 Release
4:08PM 0 Playing sounds on local channels in Asterisk 1.6.1.0
4:02PM 6 SVN vs "Regular" Asterisk
2:23PM 3 CPU usage vs compiler flags
10:42AM 1 Digits timeout (ISDN)
9:52AM 4 Transfer call from analog telephone
9:43AM 12 MeetMe and setting conference timeout
9:27AM 10 [Atcom] Asterisk + LAMP on 128MB RAM?
5:36AM 2 IAX2 trunking with Older Asterisk, version ?
12:57AM 6 Suddenly the voice became garbage (like robot) using Asterisk 1.4.19.2
12:46AM 2 Asterisk 1.4.25 and zapata.conf