asterisk users - Oct 2008

Friday October 31 2008
TimeRepliesSubject
10:25PM 6 Asterisk installation
9:08PM 7 Call problems
6:13PM 1 CDR Posting Delay
3:55PM 0 VoIP phones supporting speex
3:49PM 0 Asterisk Legacy pBx help
3:30PM 0 No audio after transferring to voicemail
3:09PM 1 Monitor group calls (recording calls)
1:10PM 1 Copy protection issues with G.729 codec in Solaris
11:32AM 3 Asterisk/Machine Hang after calling in/out ISDN
11:20AM 7 Call terminates after 20 minutes
10:11AM 17 giving a user asterisk CLI access: how bad could it get
9:54AM 2 Friday Halloween Edition Oct 31 12 Noon EDT
9:34AM 0 MusicOnHold from a Sound card
9:10AM 0 dtmf have not callerid
7:18AM 9 twice normal beep before busy tone ??
5:02AM 1 Enter Value and continue dialplan
3:35AM 3 Asterisk with SC440 Dell(Big Problem)
 
Thursday October 30 2008
TimeRepliesSubject
11:15PM 3 1.4.22 vs 1.4.21.2 - IAX2 regression ?
9:27PM 0 Re: Message 245058
7:12PM 0 General development funding: discussion and survey
6:44PM 1 ISDN - BRI
6:41PM 0 Music On Hold (from a Sound card) Help
5:22PM 1 Asterisk Legacy PBX
5:22PM 0 Other lists
4:57PM 2 Old mantis e-mails
4:17PM 3 SIP # DTMF
4:03PM 0 Asterisk SVN bug segfaulting
3:59PM 0 Connection two asterisk via SIP (call forward)
3:31PM 1 Asterisk settings
1:30PM 1 Sangoma Question
11:51AM 0 Trouble with SIP/NAT
10:55AM 5 Linux Kernel >=2.6.25 Realtime issues
9:47AM 0 mp3player and shoutcast
9:17AM 3 SIP REGISTER
9:02AM 2 up to 3000 lines capacity asterisk Deployment
7:36AM 0 list-testing2
7:33AM 0 list-testing
12:10AM 1 'Asterisk is not thread safe' message
 
Wednesday October 29 2008
TimeRepliesSubject
10:31PM 9 Dimensioning a telephony system based on openser!
9:10PM 1 app_swift installation problems
6:27PM 0 CDP (was Re: network design philosophy and practice)
6:21PM 0 FW: Thecus N7700
6:17PM 4 Is anyone using * for 2 way video conferencing?
5:55PM 5 Current Open Source Billing Package
5:13PM 5 Blank Voicemail.Conf after Password Change
5:03PM 0 [OT] Flash player for call recordings - 8khz
4:08PM 3 Intergrating vicidial with trixbox
3:59PM 0 Headset Recommendation
3:21PM 0 What syntax to send user:pass in SIP Dial string?
2:19PM 23 network design philosophy and practice
1:55PM 2 codec not in channel variables
1:50PM 2 Complete OS/Asterisk disk
1:05PM 1 SIP ACCOUNT CODE not included in CDR when SIP Status is "Unknown"
11:10AM 2 Dial() - any way to limit waiting for a "RINGING" state?
4:10AM 1 any dialplan action on received jabber msgs?
2:55AM 1 Snom - we are puzzled
12:13AM 11 Decent Voip Phones for enterprise
 
Tuesday October 28 2008
TimeRepliesSubject
11:39PM 7 Sendmail for Voicemail
10:45PM 4 XML Cisco config file
9:54PM 6 Multiline Analog Setup
7:19PM 4 Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet
4:53PM 9 Dealing with progress codes
4:30PM 11 Anyone using an Intel Atom ?
3:33PM 1 AMI - Status Event.
11:41AM 0 Re Fring: Open VPN client to be installed on the mobile, which mobile?
7:51AM 0 OT - Choose PCI or mini-PCI for appliance ?
7:11AM 1 Best Sales 2008!
 
Monday October 27 2008
TimeRepliesSubject
11:41PM 2 gtalk/jingle full report
11:03PM 7 whisper time remaining
10:35PM 0 asterisk on freebsd
10:12PM 0 Sending a text-message to JABBER via CLI
4:16PM 0 make config update-rc.d on Debian
4:02PM 0 change codec mid-call
3:34PM 3 Forcing repacketization on SIP to SIP call
1:42PM 1 How to bind a SIP channel to an IP
12:36PM 3 Door phone
12:34PM 2 Secure WAN connection too * extention
12:31PM 2 KEY SYSTEM, Intercom
10:48AM 21 Fring: Open VPN client to be installed on the mobile, which mobile?
6:35AM 1 Asterisk 1.6 pbx_lua not creating any contexts
6:23AM 2 openser+asterisk
5:19AM 1 autodialed call forwarding via meetme or queue (was predictive dialer)
2:59AM 1 Asterisk 1.6 CDR no Clid information
1:53AM 4 Asterisk and voice recognition
12:45AM 1 CDR Records are not working
 
Sunday October 26 2008
TimeRepliesSubject
8:15PM 0 Asterisk on Freebsd 7.0 Release.
6:33PM 1 No incoming audio on Dahdi channels (TDM410P)
12:25PM 11 jingle/gtalk still very troubling
4:24AM 8 hammering imap vmail storage
2:30AM 0 Queue Warning messages
12:37AM 4 Strange ring tone: Long-Short-Short
 
Saturday October 25 2008
TimeRepliesSubject
7:25PM 2 gtalk dialstring?
6:02PM 0 someone to test gtalk with me?
5:54PM 11 Cheapest 4 port FXO
1:56PM 7 The skype channel...
 
Friday October 24 2008
TimeRepliesSubject
10:18PM 0 Asterisk iaxy adapter annoying beep during conversation
9:55PM 5 Sporadic One Way Audio
9:48PM 6 Fresh installed box
8:12PM 12 OT: Disable Polycom 650 Forward Softkey
6:33PM 1 Agents log in afterhours
4:00PM 8 Advice on ISDN and Asterisk in the UK
2:31PM 0 Freepbx or Trixbox Presentation
12:38PM 2 Problems with zaptel/ztdummy/asterisk.
2:00AM 5 Asterisk and Cisco Call Manager Express (CME)
1:31AM 5 Emerging dilema? DID forwarding meets SMS
 
Thursday October 23 2008
TimeRepliesSubject
9:20PM 2 Inbound DID + voice ports needed for vote monitoring project
8:36PM 1 Returning to Voicemail after returning call
8:10PM 2 is there a reference guide to "pri debug span" messages?
4:38PM 3 Channels are increasing without limit - Please Help!
4:36PM 3 Devstate and Voicemail
4:35PM 2 Hylafax asterisk iaxmodem problem
3:17PM 1 Atxfer Command
2:28PM 0 users.conf and sip call-limit
2:22PM 1 switching from 1.6.0-beta9 to 1.6.0.1 problems
12:30PM 3 problems with some incoming/outgoing calls
9:21AM 0 command - set sip_codec- does not work with asterisk-1.4.21
2:04AM 0 FW: [wwwac] Thursday 23 October 2008 NYLUG: Paul Charles Leddy on Asterisk, the Free Software Telephone System
 
Wednesday October 22 2008
TimeRepliesSubject
7:29PM 1 : Parking Issue
7:28PM 1 changing from default codec
6:35PM 13 Sonicwall potentially causing long ping times to SIP phones
5:27PM 7 asterisk video
5:25PM 44 fax / t38 gateway
3:59PM 0 Parking Issue
12:38PM 1 2 asterisk boxes
9:44AM 3 WebCall application
7:24AM 4 sip and nat
5:09AM 0 a question about linux/asterisk/commands
4:52AM 0 need asterisk tech to relocate to riyadh
1:11AM 0 Causes of auto-congestion on SIP?
 
Tuesday October 21 2008
TimeRepliesSubject
10:03PM 2 Does Asterisk support SIP Join Headers
6:56PM 2 hex b1 in CallerID sent by Asterisk On PRI
5:05PM 1 For Dial(), when calling party hangs up, redirect called party to another location in the dialplan?
4:43PM 6 Asterisk Console color
3:50PM 1 Generating 484 "Address Incomplete"
2:39PM 0 Problem with Portech
10:46AM 2 [help] Realtime Swich any context dinamically
10:30AM 0 Realtime : switch any context dynamically
9:34AM 0 Asterisk 1.4: ISDN congestion warnings
1:42AM 3 come back ring
12:40AM 1 prepaid approach
 
Monday October 20 2008
TimeRepliesSubject
11:50PM 0 TDM410P with EC doesn't detect DTMF after being on for ~1 hour
8:36PM 0 SERVICE CODES
7:01PM 15 How Secure Is Asterisk
6:10PM 7 a little regex help needed
5:58PM 6 OPENR2 in Thailand
5:19PM 0 Transferring Outbound Calls
4:17PM 0 I have probleme with asterisk
1:45PM 0 Problem in extensions.conf Configuration ${CALLINGPRES}
1:13PM 2 ISDN PRI Caller ID problem
11:11AM 0 B410P and asterisk 1.6
6:42AM 3 asterisk setup
6:39AM 2 QoS VoIP
3:28AM 2 Zaptel FXO offhook when connected to PSTN
12:46AM 0 Panama DID
 
Sunday October 19 2008
TimeRepliesSubject
8:50PM 0 Got SIP response 603 "Declined" back from 81.15.xx.xx
7:50PM 7 Asterisk Problem
7:30PM 21 adding a second extension
4:31AM 6 Latency woes, qos the fix?
2:58AM 0 app_confcall on Asterisk 1.6 update
2:31AM 5 IP Address on CDR
1:21AM 0 Does asterisk 1.6 support an authname with a domain?
12:13AM 1 Is there a way to specify the fromdomain from the dialplan?
 
Saturday October 18 2008
TimeRepliesSubject
7:16PM 3 OT: Polycom IP330 user problem
7:08PM 3 strange h323 delay issue
6:25PM 0 Looking to replicate OnSIP ........SER + Asterisk
7:30AM 4 Asterisk 1.4 and openLDAP
1:36AM 29 SER + Asterisk
 
Friday October 17 2008
TimeRepliesSubject
10:14PM 1 Phones lose contact
8:47PM 3 anoyingly answers already in use pstn line
8:28PM 3 Transfering Calls back on the same PRI
4:15PM 0 FW: Under the Radar Blog
4:10PM 3 Snom M3 firmware Update
3:10PM 1 Asterisk SIP and SRTP
2:52PM 3 Strip prefix
2:37PM 0 Whenever Asterisk restarts, what should happen to ongoing subscriptions ?
2:23PM 0 GET DATA Returning only a single digit
2:19PM 17 srv records not being honoured properly
11:48AM 2 How to launch batch whenever Asterisk (re)start ? [SOLVED]
8:17AM 3 asterisk +heartbeat (Wilton Helm)
6:12AM 8 How to add contexts in asterisk realtime?
3:00AM 1 Meetme "talker optimization" always on even when no "o" option present.
 
Thursday October 16 2008
TimeRepliesSubject
6:38PM 2 DAHDI and wait 'w'
6:38PM 1 International calls/pridialplan from a legacy PBX.
5:09PM 1 asterisk +heartbeat
4:39PM 0 Multiple Repeated tones with TDM02B
3:49PM 2 RELEASE message in q931.c
3:48PM 5 SIP: difference between Grandstream and Cisco when behind NAT
3:40PM 4 Triggering a call from bash
3:23PM 0 asterisk-users Digest, Vol 51, Issue 51
1:57PM 1 app_confcall build issues
1:03PM 0 Difference between followme and substitution ?
12:31PM 0 Sharing my Asterisk + SPA3102/PAP2 setup: What I've learned in 1 week.
12:23PM 0 asterisk cmd mysql and stored procedures
11:16AM 7 prective dialer
9:49AM 0 app transfer problem
6:57AM 8 How to launch batch whenever Asterisk (re)start ?
3:22AM 0 How to invoke an external C program and output an integer to the program?
3:19AM 0 Telrad Analog CID
12:16AM 0 Queue problem
 
Wednesday October 15 2008
TimeRepliesSubject
11:57PM 6 Cisco 7960 not always receiving incoming calls
11:19PM 1 Configuring SIP TLS
8:59PM 1 voicemail.conf
7:52PM 0 phoniceq e400p driver for DAHDI
7:38PM 2 Zaptel compile error after make update.
3:26PM 2 mismatched callerid on phone and CDR ?
1:28PM 0 LDAP authentication
10:57AM 2 Asterisk 1.4.21.2 - Issues with call parking
6:49AM 0 FW: asterisk-users Digest, Vol 51, Issue 48
 
Tuesday October 14 2008
TimeRepliesSubject
10:57PM 2 Help With AMI
10:24PM 2 SIP channels seem not to close after call is finished
8:26PM 3 Looking for a mentor
5:35PM 1 Speex Problem
4:46PM 8 asterisk+heartbeat
4:09PM 0 Asterisk 1.4.10.1 : PRI congestion warnings
3:05PM 5 Call files
9:42AM 7 Asterisk voicemail
1:15AM 27 Panasonic x Asterisk if I can emulate Panasonic fast!
 
Monday October 13 2008
TimeRepliesSubject
10:06PM 0 Asterisk help please
9:27PM 0 voicemail issues with 1.6.0
9:22PM 2 Tracking T1/PRI channel status - inbound vs. outbound
8:44PM 19 ISDN
6:21PM 1 ifbyphone/google analytics
5:13PM 4 Softphone Framework or Libraries
5:10PM 0 AGI Hangup
3:29PM 4 Need help for debuging
3:19PM 1 IP 650 Sidecar
2:50PM 0 MOH Bad
2:13PM 0 Unknown call every 30 minutes on the dot.
11:18AM 0 Installing CdrTool on free bsd
10:28AM 3 echo over digital line
9:41AM 0 ERROR:Failed to create H323 listener
 
Sunday October 12 2008
TimeRepliesSubject
9:17PM 6 setup for fax machine
6:56PM 0 Caller ID sip trunk
3:53PM 29 One Way Audio Problem
3:17AM 7 cli commands missing
 
Saturday October 11 2008
TimeRepliesSubject
6:41PM 3 Asterisk 1.6.1 + openais
6:02PM 4 1 second delay when connecting calls
4:20PM 9 Is there a way to test SIP credentials without making a call?
4:13PM 15 Asterisk For Windows ?
2:25AM 0 Mitel 5220 firmware
 
Friday October 10 2008
TimeRepliesSubject
9:49PM 0 Caller ID service and the ethernet stucking
9:17PM 13 Configuring Bandwidth.com SIP trunks to prevent one-way audio
8:43PM 10 is there a way
8:24PM 5 Budge Tones pick up wrong calls
4:57PM 12 Got event 17 (Polarity Reversal)...
4:52PM 3 Block Caller ID
4:04PM 14 Question about echo cancelation
2:09PM 9 Asterisk SIP calls stop working having more than 300 calls (more than 600 channels)
12:33PM 4 Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
10:52AM 18 How to enable inbound CLI for X-Lite/Asterisk phone.
9:09AM 3 Be aware of callcheap.com and Mike Low - It is scam
7:50AM 14 Compile logger-mysql.c with UNDEFINED REF to `mysql_error'
7:44AM 0 How to barge Inbound calls
3:18AM 1 1.6.0.1 ??
2:20AM 2 Asterisk CDR Analyser
1:37AM 4 Polycom 330 not dialing 4 digit extensions beginning with 11xx
 
Thursday October 9 2008
TimeRepliesSubject
11:31PM 15 DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((
10:00PM 4 Cisco 7960 sccp, Skinny and 1.4
9:59PM 8 Transfer/Park Question.
9:09PM 1 SIP problems?
9:09PM 2 Asterisk 1.6.0 CDR billsec and duration not working from h extension
5:38PM 1 asterisk-users Digest, Vol 51, Issue 26
5:22PM 0 ATA hangs up with fax detection on...
2:45PM 1 Asterisk-Panasonic TDA 600 error
1:25PM 5 Inbound Calls From AS5300 Rejected with 488 Code
11:06AM 0 Asterisk restarts on call parking
9:18AM 12 Hang up detection with TDM400P and Telewest/Virgin Media line
8:19AM 2 Ringtones for the console
7:52AM 0 Interrupt Asterisk's SayDigits()
7:34AM 1 H323
7:10AM 7 Howto analyze concurrent ISDN channel usage
6:57AM 0 Friday Oct 10 @12 Noon EDT VoIP Users Conference
4:51AM 1 Interpreting Asterisk Logs
3:41AM 4 Record name for conference...
3:14AM 3 Menu for call forwarding or voicemail
3:14AM 2 retransmitting NAT
2:44AM 1 conntrack_sip, iptables, and asterisk
 
Wednesday October 8 2008
TimeRepliesSubject
10:21PM 5 Update (IAX Trunking Help)
7:44PM 1 Sip Trunking
7:07PM 0 Help with IAX Trunking
5:51PM 2 Sample code fragement for subscribing to hints wanted.
5:33PM 5 make func_realtime work like app_realtime (1.6)
4:18PM 0 Mobile (cell) phone in a queue (but act like an agent
4:03PM 0 fastagi example
1:58PM 4 debugging hints in 1.6
11:44AM 0 SRV DNS failover - dial to proxy list
10:24AM 0 How is automatic redial/callback when available implemented ?
8:29AM 1 Call-limit bug in 1.2 ?
7:46AM 0 Zaptel -> DAHDI for dummies?
6:11AM 0 Can't find the path to Phoneprov directory
4:11AM 1 registration limit
12:30AM 5 changing passwords
12:26AM 0 "requested special control 20" ??
12:04AM 3 help no ring on caller side
 
Tuesday October 7 2008
TimeRepliesSubject
8:36PM 4 Question on screening calls / Question about the Dial g option
7:20PM 12 cisco phones getting SIP 401 unauthorized
3:45PM 1 Bad Destinations
2:58PM 0 Asterisk Callerid Help Needed
2:08PM 1 include in the DAHDI system.conf file and chan_dahdi.conf
1:19PM 1 Efax from Agi script
10:48AM 0 call center
10:25AM 23 Cisco 7906g & SIP
8:42AM 6 automatic call pickup
6:22AM 7 can't find mysqlclient : asterisk-addons-1.6.0
4:03AM 2 regcontext
2:24AM 0 Asterisk/AJAM Console
 
Monday October 6 2008
TimeRepliesSubject
8:30PM 4 ldap usage in 1.6.0
7:58PM 24 Matching *, + and # in the dialplan
7:02PM 6 Asterisk 1.4 or 1.6 ???
6:53PM 7 Help with remote users
6:00PM 2 Missing 'Queue' Application in 1.4.21.2
5:33PM 1 cdr,gsm file format
5:19PM 0 Nice recording interfaces
3:23PM 0 Semi OT: Global Crossing
3:03PM 27 PoE switch recommendations?
2:55PM 1 Hook Flash
2:21PM 8 AEL and swap from macros to contexts
2:08PM 7 Conneting Asterisk to Swyx pri
2:07PM 13 How to implement Ringing through a sound card for overhead paging
9:24AM 10 Tribox
7:46AM 6 R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)
7:26AM 5 Alarm events + asterisk dies
7:25AM 0 Fwd: Fonolo: Visually Navigate & Dial IVR Phone Menus in Web/Mobile Browser
4:39AM 9 asteriskt38.com
2:08AM 2 Dial out DAHDI Channel?
12:38AM 4 MS Exchange IMAP Voicemail
12:03AM 0 no per mailbox imapfolder override? wow.
 
Sunday October 5 2008
TimeRepliesSubject
8:30PM 5 OT: headsets
5:51PM 7 asterisk, phpagi and singleton
3:59AM 3 Asterisk Load Balancing
 
Saturday October 4 2008
TimeRepliesSubject
5:50PM 0 voicemail quota
5:30PM 8 Vitelity Asterisk configuration help
4:47PM 1 IAX denial of service
12:02PM 2 Mimic SIP Events framework in Asterisk without coding ...
11:18AM 0 2 stage dialing and 484 address incomplete [SOLVED]
9:11AM 2 Aastra phones and dns srv records
 
Friday October 3 2008
TimeRepliesSubject
11:53PM 0 DTMF Talk Off
10:43PM 1 DTMF issues...
10:07PM 0 Asterisk 1.6.1-beta1 Now Available
8:07PM 2 network monitoring - triggering a phone call in asterisk
7:06PM 0 catch the use of h option in dial
3:05PM 3 Ok message
3:03PM 5 MWI with Siemens Gigaset S450IP
9:21AM 17 How to add Callee's name into Dial command ?
8:08AM 32 sip clients for smart phones?
8:04AM 9 Improving the voice Quality,
7:59AM 0 2 stage dialing and 484 address incomplete
4:22AM 9 t1 cards
1:39AM 1 uninstalling zaptel
12:45AM 1 dahdi service start
 
Thursday October 2 2008
TimeRepliesSubject
8:42PM 2 DTMF
8:00PM 2 VOIP Provider
7:26PM 3 Channels crossing...
6:56PM 1 Asterisk 1.4.22 and 1.6.0 Released
6:09PM 0 dahdi-linux 2.0.0 and dahdi-tools 2.0.0 released
4:32PM 2 Asterisk Queue question
4:22PM 2 OT - Is sip.instance useful ?
2:43PM 2 Is SIPPEER curcalls working for you ? (was: Ongoing calls with SIPPEER, curcalls)
2:19PM 14 Zaptel-1.4.1 error cross compile
1:03PM 2 DTMF issue
12:28PM 0 IP address on mysql cdr
12:22PM 1 Ultramonkey LVS + asterisk
9:38AM 2 How to find the CDR call start time value
9:27AM 2 DTMF Problem
8:44AM 0 B410p question
4:58AM 1 Asterisk custom functions
1:21AM 1 cisco VAD and Asterisk recordings
1:19AM 2 Aheeva With Asterisk
12:57AM 2 rebooting snoms in 1.6
12:42AM 0 no audio, firewall problem?
 
Wednesday October 1 2008
TimeRepliesSubject
9:24PM 10 Asterisk - Failover System
4:42PM 0 ietf-sipping-config-framework
3:28PM 0 [Fwd: asterisk-users Digest, Vol 51, Issue 2]
3:19PM 0 SIP users limit
2:57PM 5 Ongoing calls with SIPPEER, curcalls
2:30PM 0 Sip Header Help
1:38PM 3 RTP sent before the INVITE ACK (for voicemail app)
1:01PM 0 Debugging
11:53AM 6 GSM / 3g channel bank
6:51AM 12 Software patents (was G723 on asterisk 1.4.1)
3:20AM 0 Polycom 3.1.0RevB
3:19AM 5 No reply to our critical packet
3:03AM 2 Cisco Dropping SIP support?
2:05AM 5 is DNS SRV enough for failover?
1:35AM 6 zap destroy