asterisk users - Jul 2010

Saturday July 31 2010
1:18PM 0 MeetMe transcode / format problem
10:04AM 0 Disconnect supervision tone detection working for india
Friday July 30 2010
7:40PM 0 Aastra ignore call button hangs up call instead of going to voicemail
6:00PM 1 Please test: STUN patch for Asterisk behind NAT
1:17PM 2 perform tasks outside a dial-plan (not during a call)
11:17AM 4 agi macro problem
10:01AM 3 How can i switch to samba server omitting sshfs
8:06AM 7 Asterisk and QoS
7:24AM 1 asterisk-users Digest, Vol 72, Issue 82
7:13AM 0 asterisk-users Digest, Vol 72, Issue 81
2:35AM 9 Aastra phones occasionally show "No Service" - Is there any network setting I can tamper to facilitate a quick DHCP renewal on the Aastra phones?
12:42AM 2 VUC Friday: Twilio OpenVBX
12:30AM 4 1.8.0 beta2: courtesy tone being played to callee
Thursday July 29 2010
10:49PM 3 Problem with Sangoma card...
10:31PM 5 COnfig File question
8:35PM 1 ignorant question about Digium cards and MeetMe
6:44PM 5 Disconnect supervision tone detection
3:28PM 9 How to extract channel-id of a user or peer
1:03PM 2 CDR and "custom name fields"
11:48AM 3 T.38 fax between ATA's and Asterisk and Cisco PGW 2200
11:23AM 0 dahdi (in use) problem with queues
11:22AM 1 How can conect Cisco Unified Communications Manager with Asterisk
8:15AM 1 SEPMAC.xml for Ciscp 7970 IP Phone
7:30AM 5 How to record and playback at the same time
6:19AM 4 Registering 2 phone numbers to same router
1:45AM 5 spam blacklist
1:06AM 6 Asterisk stopped after Internet connection dropped ?! Asterisk
Wednesday July 28 2010
10:22PM 5 app_swift.c:338 engine: Failed to set voice
9:33PM 6 Why do Zaptel calls drop all of a sudden? Could busy detect be the problem?
9:04PM 6 Asterisk unresponsive
7:06PM 0 3G-324M Open Source
4:30PM 0 what is rinstance parameter in sip header
4:23PM 2 Nat issue one way audio on IP dial
3:38PM 4 Subscribe Problem - Zombie Channel
1:38PM 2 [OT] fail2ban and pf
1:03PM 0 AMI Monitor - one file
12:04PM 1 Passing Variables From Dial Macro To Parent Ruby
11:49AM 4 IAX authentication oddity - Known issue? Fixed?
10:25AM 5 Answered call not bridged
9:29AM 1 Redirecting a call to another extension using asterisk java
1:12AM 5 Random DTMF Tones Only on heard on ATA
12:39AM 7 Recording interface (pause/PLAY/RERECORD)
12:38AM 3 Grab voicemail WAV file when done
Tuesday July 27 2010
7:50PM 6 Asterisk and Amazon Web Services
7:41PM 0 sip peer becomes unreachable in Asterisk 1.6
6:57PM 1 Peculiar Polycom IP6000 behavior
3:56PM 0 Asterisk 1.8.0-beta2 Now Available
2:21PM 23 Urgent help = RUBY & AGI
12:12PM 2 How to transfer a call to operator using FAGI asterisk
11:56AM 1 IAX bandwidth optimisation
11:40AM 2 CallerID disappear from CDR on transfer
7:47AM 3 1 second Audio Lag
5:38AM 2 urgent:how to transfer a call using asterisk FAGI
3:34AM 14 Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
Monday July 26 2010
10:48PM 1 VPMADT032 Failed! Unable to ping the DSP (2)!
8:20PM 8 MeetMe
5:36PM 6 Fail2ban - SuSEfirewall
4:39PM 3 Problem with Zap-Sip calls.
2:45PM 4 asterisk & distributed device state => res_jabber Versus res_ais
12:34PM 7 PBX Lua with Asterisk ODBC
12:15PM 5 Management interface
12:11PM 5 'dirty' upgrade of 1.4
10:41AM 27 URgent - capturing 'answered'
10:16AM 3 Optimize peers registration under jitter/delay.
9:28AM 0 URGENT - who picked up the call??
8:05AM 15 FreeTDS (Microsoft MsSQL 2008) and CDR
8:03AM 5 chan_skinny still maintained?
6:54AM 0 Adit 600 over MGCP.
5:20AM 2 No audio using xlite
1:54AM 1 SIP TOS Not being set
Sunday July 25 2010
10:57PM 2 Vicibox vs VicidialNow
9:11PM 2 Using Vertical IP2007 phones with Asterisk?
7:06PM 2 sounds Makefile error?
6:47PM 1 Proprietary add-ons for Asterisk 1.8
2:52PM 6 Cisco 7960 phone can't leave a queue
2:23PM 0 Audio Delay of 1-2 seconds, one way with Zoiper soft phones
4:52AM 6 undocumented change in expression handling in 1.8 beta
12:11AM 6 "Register Attacks" End of ENUM ?
Saturday July 24 2010
4:44PM 2 Integration with Toshiba Strata DK424
4:07PM 3 Exchange UM Play on Phone
4:14AM 11 getting some segmentation faults with 1.8
Friday July 23 2010
9:58PM 29 Asterisk 1.8.0-beta1 is Now Available!
8:53PM 1 voicemail
7:48PM 2 Attended Transfer question
5:11PM 0 Asterisk Now Available
5:10PM 0 Asterisk 1.4.34 Now Available
2:55PM 2 Dahdi dial plan
2:32PM 4 Poor-man's paging through multiple phones?
1:50PM 3 application call to Gosub affects flow of control, and needs to be re-written using AEL
1:35PM 0 calls don't hang up correctly on VM
11:26AM 1 488 Not Acceptable Here
11:23AM 17 Vocera Comm Badges
10:17AM 0 (no subject)
9:32AM 2 Channels not coming up
7:34AM 1 ringback tone after MOH, before queue member bridged
5:16AM 8 Why does a bridged channel stay open for 4 hours?
Thursday July 22 2010
9:47PM 5 Question regarding SMS(), SMSQ, SMSC
9:10PM 3 Does SIP limit to 3-way conference?
7:58PM 1 FW: hi friend
7:15PM 0 Receiving T1 Blue Alarm on asterisk 1.4.26, zaptel 1.4.12
7:14PM 6 Soft phones.
6:46PM 16 POE Splitters
5:11PM 15 [AsteriskNow] Errors with clean install (on main screen when making calls)
2:29PM 0 SIP URI Dial has one way audio
2:27PM 0 Good SIP provider Western US
2:13PM 1 Good provider that offers allmost free calling within Europe?
1:25PM 0 RTP delay
10:53AM 2 dialog module count
10:33AM 3 My Switch is being attacked using sip scanner tool (Service Abuse Attack)
6:25AM 3 Could Asterisk-addson- install in 64bit Cent-OS ?
3:06AM 9 asterisk app_fax, T.30, weird received faxes
Wednesday July 21 2010
11:05PM 5 Cisco Firmware
6:29PM 4 Redial dtmf tones randomly...asterisk
12:32PM 0 Musiconhold Problem
11:51AM 1 Cisco 7970 Not registering
11:38AM 2 Meetme Question
10:09AM 1 asterisk realtime SIP configuration
7:14AM 10 play alaw file with .wav extension
6:32AM 11 MOH distorted voice in Native and MP3 format
5:36AM 0 VoIP carrier g729
Tuesday July 20 2010
5:59PM 5 Problem with SIP
3:46PM 0 asterisk-users Digest, Vol 72, Issue 49
3:46PM 0 Got SIP response 603 decline, then the call hang up
3:35PM 1 Dahdi fails to compile in Xen DomU
1:42PM 4 Dahdi - Meetme problem on a VM
12:19PM 1 Different source IP address for each peer
12:14PM 4 OT - Gigaset and auto-configuration code
12:05PM 2 Preserving CDR(accountcode) in Local channels
9:02AM 11 Call not going through and failing because "never answered"
Monday July 19 2010
9:06PM 5 Problems with Dahdi trying to load OSLEC
6:16PM 8 Voice prompts
4:30PM 2 Swedish voiceprmpts
4:22PM 2 digium HW echocancellation - fax tone detection
3:47PM 3 Multiple sip.conf files?
3:11PM 3 Problem with E1
3:00PM 0 Pereserving the callerid value when presentation set to witheld over sip
12:41PM 3 Asterisk Queue + Caller ID issue
11:08AM 1 hi
10:01AM 3 Voip rates to Mali
3:26AM 5 T.30 fax receiving problem with app_fax
Sunday July 18 2010
8:21PM 0 Softphone's
6:58PM 0 FreeSide
2:28PM 1 Logging registration/unregistration of peers/extensions in database
11:48AM 7 Skype for Asterisk, Skype For SIP
Saturday July 17 2010
10:52PM 3 Audacity settings for Asterisk sound files
5:19PM 1 AGI execution after Dial
6:47AM 0 asterisk-users Digest, Vol 72, Issue 42
6:11AM 0 Queue Call Transfer Issue
4:35AM 1 AGI gosub return value
Friday July 16 2010
9:57PM 2 g729 codec loading
6:38PM 7 Video IVR Asterisk ?
5:59PM 6 RFCFS - reload specified file
5:42PM 0 Set Queue to ring in only one member
5:04PM 0 beeping during calls
4:56PM 1 (no subject)
4:43PM 0 1.6.2 ConfBridge suggestion
4:34PM 2 Busy Lamp Fields
1:19PM 0 asterisk-users Digest, Vol 72, Issue 39
10:44AM 7 chan_local - Asterisk
8:17AM 1 BLF - Realtime & Asterisk
7:48AM 0 Today on VUC: SIP-Aware Appliances to facilitate communications
5:49AM 1 Queue
4:55AM 4 IAX endpoints not Registering after upgrage from Asterisk ver
1:57AM 2 SKYPE - Authenticate incoming call
Thursday July 15 2010
9:06PM 0 Dahdi T1 CRC4 errors?
8:51PM 0 help with sip registration
8:13PM 4 Soft-phone on Black Berry
8:01PM 2 Does Flash Operator Panel allow for dragging a call into a parking lot?
7:24PM 1 Asterisk Manager Problem
5:13PM 1 QoS and Asterisk
5:02PM 0 WARNING[15867]: chan_sip.c:15766
4:40PM 7 Good script to make appointment?
3:17PM 0 Last call for AstriCon talks
2:38PM 8 One way audio when dialing multiple registrations
1:16PM 2 centos 5 rpm pacakges (add asterisk16-xmpp module)
9:50AM 0 MeetMe incorrectly reading key presses
8:36AM 2 Invalid host name
8:15AM 2 How to deal with voice SMS - Asterisk 1.4
4:30AM 0 Get channel name of originated channel
4:29AM 1 SKYPE - Authenticate incoming call automatically
Wednesday July 14 2010
10:29PM 3 sip message to ip 330 or 550 phones
9:36PM 1 DAHDI Outdial To Cell Phone Playing Music
9:27PM 2 Hosted PBX in the UK
6:53PM 8 realtime music on hold
6:44PM 7 Distinctive ring for INTERNAL calls only? How to do it?
6:23PM 1 Dahdi Echo canceller setup
4:27PM 3 beeping during call
4:04PM 3 Where should I look for MWI settings if Aastra phones don't do it?
11:17AM 3 BLF with Realtime
6:33AM 4 Asterisk core dumping on SendFax with FFA
5:47AM 1 Silence RTP
4:38AM 18 Can't compile DAHDI - wrong kernel source
3:53AM 2 How to pass through supported 100rel
Tuesday July 13 2010
5:35PM 0 asterisk un-registering from provider
4:30PM 10 STRFTIME function declared in globals context
2:40PM 0 How to install speex codec for Asterisk that is downloaded from Digium Yum Repository?
1:27PM 1 MyFuel Express FO - Shortcomings **PLEASE DELETE THREAD**
12:33PM 1 MyFuel Express FO - Shortcomings
8:52AM 17 OT: fail2ban, spam and mail servers
8:29AM 10 Recording from g729 to wav means transcoding ?
7:46AM 0 How to trace incoming AMI requests ?
3:14AM 0 OT: HUD3 and NON-Trixbox Asterisk?
Monday July 12 2010
10:09PM 1 Complex Dialplan Help Needed
7:58PM 1 browser pop-up on call ring
6:06PM 2 My own FreePBX FollowME module - Stuck at Reload - Anyone else had experience with this?
4:58PM 1 send Variable to remote system via AMI / Orginate
4:58PM 0 Inconsistent Behavior in SYSTEMSTATUS After System() Call
3:41PM 10 MAC Address prefixes of Voip equipment
3:36PM 5 Chanspy - Meetme
1:09PM 13 Remote-Party-ID party=called
1:08PM 5 Fax for Asterisk, capable of receiving from website but not from fax machine !!
8:59AM 0 DTFM Detection issues
8:17AM 0 ResetCDR not working after forced hangup
4:31AM 3 Use asterisk as a backend PBX
2:47AM 3 ztdummy IVR no voice
2:29AM 3 need information
Sunday July 11 2010
12:00PM 4 Dialplan question when using a round-robin
9:40AM 4 Is a device a member of a queue?
Saturday July 10 2010
6:16PM 8 How can get user inputs from called party after dial?
4:07AM 9 PHP can't insert - Can someone please help
1:57AM 1 False answer() being sent by cellphone providers
Friday July 9 2010
9:52PM 0 Pbx_för_Windows?_-_Email_f ound_ in_subject
8:54PM 1 Logging codec used in CDR
6:41PM 3 [Dahdi] Does it need all those modules?
5:27PM 4 chan_iax2: I should never be called!
4:57PM 12 <no subject>
4:48PM 2 Delay between answer and pickup ?
4:42PM 8 power outage
2:30PM 2 Call failed: 408 timeout
1:29PM 0 Pbx_för_Windows?_-_Email_found_ in_subject
12:41PM 8 Pbx för Windows?
12:36PM 0 Sip Proxy
11:38AM 2 Re : Re : Re : Communication IAX2 >SIP>IAX2
8:29AM 4 Click2call from an OpenOffice document
6:29AM 8 How to calculate number of speakers needed for PAGING and INTERCOM coverage area?
5:32AM 0 Rx/Tx fine tuning of analogue card to PRI card - Am I right with my theory?
Thursday July 8 2010
9:50PM 3 Not detecting hangup
8:26PM 0 How to know which party hangup() first when using analogue cards? - Dahdi and Asterisk 1.4.x
8:18PM 4 DTMF issues/redial tones with rfc2833
7:33PM 1 Level3 reseller needed
5:01PM 1 Incoming call doesn't finish when internal phone hangs up
4:22PM 0 connecting to EON millenium getting 405 message
4:04PM 5 Asterisk + Hylafax + Iiaxmodem - Outbound number.
3:48PM 3 Junghanns QuadBRi not really recognized in Dahdi
3:30PM 1 not sure what to change to point the timing to the at&t circuits?
1:36PM 0 How to integrate thirdparty RTP with Asterisk
1:17PM 0 call deflection support in chan_dahdi, libpri
12:54PM 0 Recordings in the bank.
10:49AM 1 AGI get full variable
7:51AM 1 Problem with call-limit
6:51AM 35 Asterisk Crashes - Segmentation Fault
Wednesday July 7 2010
11:19PM 7 GURUs - How to monitor all MySQL actions on an Asterisk/FreePBX server?
6:18PM 6 Conditional "includes" in iax.conf
5:45PM 0 Problems with dual FXO/FXS cards - noise
5:34PM 5 Communication IAX2 >SIP>IAX2
1:40PM 0 Caller ID on analog line
8:38AM 0 IAX calling presentation null
8:13AM 3 This may be a problem.. Answer not working on 1.4.32 over SIP trunk..
Tuesday July 6 2010
11:11PM 1 sip.conf User vs Username
8:10PM 7 Can't dial out through AMI
6:11PM 8 Y-cords - What are they ?
5:53PM 0 Problem with wct4xxp - cannot make calls
4:29PM 0 asterisk: call failed error 408 timeout
3:03PM 1 Dahdi - Which process to swap from Octo to QuadBRI ?
2:34PM 1 1.6.2: Using hints on multiple parking lots
1:10PM 1 IVR
11:40AM 18 How to secure Configuration files
10:37AM 2 Dahdi - alarm which clears itself - Should I care ?
9:34AM 5 ARA : Realtime or not ?
12:17AM 5 Externnotify on pollmailboxes=yes
12:09AM 0 97 issues marked 'Ready for Testing'
Monday July 5 2010
11:51PM 1 problem with voicemail contexts
11:29PM 8 How to Dialogic 240/JCT-T1 interface with Asterisk?
9:10PM 2 dahdi on solaris
8:52PM 4 Anybody with experience with Aculab Groomer II
8:16PM 0 Reinvite to alaw after T.38 reception
2:57PM 4 Problems with ulaw/g729 translation
2:02PM 1 res_fax_digium and T.38 error correction
12:04PM 0 Reg. EMT-22 IP Phone
10:45AM 4 [NAT] * + private IP + locked-down firewalls?
9:31AM 5 SIP response 482 "Loop Detected"
7:04AM 0 Hold and Retrieve the call through AGI
2:40AM 1 Why does my IAX2 trunk between two office hangup a channel after 30 seconds? Can you share your IAX2 trunking configuration? URGENT HELP much appreciated
Sunday July 4 2010
2:40PM 2 Anyway to know when a channel is going to hangup if Dial Timeout option is used?
11:32AM 3 Asterisk for transcoding
Saturday July 3 2010
1:04PM 2 Couple of questions about modules
10:53AM 0 Join July Global via VOIP Free SW HW Culture Mtgs - BerkeleyTIP
8:59AM 0 strange issue while setting pin in MeetMe
7:09AM 1 VoIP Users Conference Recordings
6:09AM 0 [asterisk-user] gsmtolin_framein: Invalid GSM data
Friday July 2 2010
10:33PM 3 Using AMI Originate to call 2 outside numbers and connect them
5:10PM 11 iptables/ blocking brute-force attacks, and so on...
4:54PM 8 asterisk and cisco 2800
12:58PM 0 Fax T.38 passthrough failing after upgrade
9:46AM 0 Difference in dahdi between 1.4.x and trunk?
9:08AM 2 Transfer fails
3:19AM 3 DAHDI FXO calls and the 's' extension. No, Jackie-O doesn't work here--it's just an example. Sheesh!
2:59AM 4 GotoIfTime problem
1:32AM 0 SwitchVox AA355 w/ 4 Port PRI and 2 Port FXO and 2 Port FXS For Sale on eBay
Thursday July 1 2010
6:57PM 0 rename External Directory
5:20PM 1 Dial SIP channel with no registration, timeout before CONGESTION?
2:52PM 23 Remote Party ID issue
1:16PM 9 Brute force attacks
1:02PM 0 AppDial in CEL Data
12:50PM 1 mISDN install on Asterisk 1.6 failing
11:49AM 3 Originate multiple channels
7:51AM 4 p2p or p2mp for BRI
2:44AM 2 call file question
2:36AM 2 Want to retrieve the value of contact header