asterisk users - May 2011

Tuesday May 31 2011
TimeRepliesSubject
10:24PM 4 SIP Register DOS attack
10:03PM 0 Mitel PBX caller id format?
5:38PM 13 AMI buffering event output?
4:48PM 0 Dropping incompatible voice frame on DAHDI/i1/xxxxxxx of format slin since our native format has changed to 0x4 (ulaw)
2:21PM 1 queuemetrics with 1.8 queue_log
12:32PM 2 To know if the ISDN PRI E1 is UP?
5:18AM 9 BRI confiugration error
 
Monday May 30 2011
TimeRepliesSubject
10:29PM 1 Configuring ISDN PRI using DAHDI
5:30PM 7 please help
12:32PM 4 ControlPlayback's options
9:23AM 2 CLI command 'database deltree' doesn't remove family with space in its name
8:03AM 3 DAHDi installation problem
6:20AM 0 asterisk in some kind of loop
1:10AM 20 asterisk fails when DNS or internet fails
 
Sunday May 29 2011
TimeRepliesSubject
11:47AM 23 Free CNAM
8:57AM 5 Why PRI not BRI ?
 
Saturday May 28 2011
TimeRepliesSubject
8:08PM 11 Cisco registration problem with 1.8.3.3
11:34AM 3 dtmf Caller-id detection before first ring
 
Friday May 27 2011
TimeRepliesSubject
5:30PM 2 More Cores or more CPU Speed
5:12PM 0 Asterisk on FreeBSD 8.2
4:33PM 12 DAHDI span timeing source
3:45PM 2 disable sip registration
3:10PM 8 standalone PRI-to-SIP converter
2:42PM 4 DID for outbound PSTN call
1:41PM 0 Dahdi and function CHANNEL
10:28AM 1 About Redfone Configuration
9:29AM 7 how to specify the numbers to call with sip
8:31AM 4 Audio dropping
6:44AM 0 ldap questions
12:50AM 2 user asterisk does not exist - using root
 
Thursday May 26 2011
TimeRepliesSubject
9:12PM 0 Grandstream + IPv6
6:24PM 0 Dahdi channel stuck in "ringing" state
6:19PM 3 DB driven voicemail
4:55PM 7 make calls from DID
1:38PM 0 DTMF digits received, but not completely forwareded
1:09PM 6 UK English sounds packs
10:45AM 8 Is this Asterisk issue of feature
 
Wednesday May 25 2011
TimeRepliesSubject
9:33PM 2 Tr : how to user SIP realtime option
5:32PM 16 Asterisk 1..8 multiple queue
6:00AM 0 How to put your call on hold with asterisk Dialplan
5:26AM 2 asterisk hint SIP presence
4:45AM 1 synway
 
Tuesday May 24 2011
TimeRepliesSubject
11:20PM 3 How to enable the addon in the Asterisk 1.8 compilation
10:50PM 0 Asterisk SIP Trunk with CUCM Express, Disable Comfort Noise?
8:50PM 4 Skype for Asterisk - RIP
7:39PM 1 Asterisk + Patton ISDN2e gateway (UK)
4:20PM 0 Asterisk 1.8.4.1 Now Available
3:22PM 0 Realtime dbase table mods
2:07PM 0 SMS callfile
10:19AM 3 SIP per-call heartbeat?
10:12AM 0 issue with asterisk 1.4
8:19AM 0 Grandstream and setvar
 
Monday May 23 2011
TimeRepliesSubject
9:30PM 5 Sending call to specific IP address
5:36PM 4 chan_zap
2:47PM 0 Asterisk + USSD
2:41PM 2 AJAM XML output not valid xml
12:43PM 4 Latest DAHDI/libpri/Asterisk 1.8 & 1x BRI port HFC based ISDN card?
12:41PM 2 [Fwd: FW: extconfig.conf]
12:29PM 0 Asterisk 1.8 TLS with Softphone blink on Windows donĀ“t work
12:25PM 0 asterisk 1.6.2.17.2 Warning on startup
9:06AM 4 Asterisk DTMF 'talkoff' issues
7:26AM 2 SIP-T to SIP Gateway
 
Sunday May 22 2011
TimeRepliesSubject
11:05PM 9 call files .vbs
 
Saturday May 21 2011
TimeRepliesSubject
12:28PM 3 how to user SIP realtime option
12:19PM 3 difference between SIP peer and SIP user ?
 
Friday May 20 2011
TimeRepliesSubject
11:43PM 0 looking for testers for app_meetme AMI patch
10:04PM 0 AstLinux 0.7.8 Release
6:37PM 0 Playback noanswer & SIP
6:20PM 4 Faxing with Asterisk 1.8.4 & T.38
6:10PM 7 Restart asterisk destroy all registered SIP peers
4:10PM 0 Static agent in queue
1:42PM 3 SIP Diversion RDNIS - how to get reason parameter?
1:37PM 14 AstManProxy
10:07AM 1 Hints custom:abcdef
9:59AM 1 *8 pickup and CLI presentation
9:57AM 0 VUC: Sangoma NetBorder 4.0 and an Androis SIP client from Media5 Corp
9:33AM 0 first dtmf is not detected
5:19AM 0 Using a feature from AMI or CLI
 
Thursday May 19 2011
TimeRepliesSubject
10:13PM 3 [Fwd: FW: realtime mysql - p4]
9:10PM 5 Agent (Invalid) has taken no calls yet
5:24PM 1 Polycom IP335 3.3.1 Call Waiting
4:44PM 11 click to call with php
4:41PM 1 Static Vs Dynamic queue confusion
4:23PM 3 Dropping incompatible voice frame
4:05PM 3 Manager logged on/off messages
3:10PM 1 Getting 603 Declined after AGI execution
1:39PM 8 ConfBridge - Failed to find a bridge technology to satisfy capabilities
6:05AM 1 SIP 603 Declined after AGI execution
2:05AM 6 Pridialplan/ prilocaldialplan
12:01AM 3 v1.8.4: Extension Not found in Context?
 
Wednesday May 18 2011
TimeRepliesSubject
8:40PM 9 asterisk's zombie processes
8:32PM 1 asterisk18 - realtime/mysql - take 3
2:05PM 2 Failover trunks
12:07PM 7 Using Asterisk/Digium repos => Astribank firmware not found
9:22AM 0 Make Multiple Calls using Chan_alsa module
7:57AM 0 Sending SRTP to Asterisk Gateway ends up in authentication failure
5:21AM 3 SRTP of Asterisk
12:12AM 0 Log off all agents from all queues...
 
Tuesday May 17 2011
TimeRepliesSubject
10:25PM 2 Queues.conf Language Agents
8:21PM 2 script to trim sip.conf
6:50PM 1 Question on AMI
5:30PM 5 Skype-like dialing from web page
5:12PM 7 OT, free software for SIP ladder diagrams?
3:46PM 1 Name or service not known
2:36PM 1 mysql call stored procedure
2:16PM 5 how to know how many calls are on hold
1:45PM 0 Type of number in outgoing SETUP frame
1:17PM 0 put multiple call on hold by dialplan in asterisk
12:18PM 1 OT - Which XMPP server for Jingle-enabled XMPP service ?
8:43AM 9 Automatic dialing + SMS
1:27AM 0 3. Re: ITSP Multi IPs (Alex Balashov) Asterisk-users Digest, Vol 82, Issue 33
 
Monday May 16 2011
TimeRepliesSubject
9:54PM 5 Reporting Tool: To show who is login, queue, ... etc
6:41PM 5 dahdi command not available
4:05PM 2 Step by step guide
4:03PM 2 Missing Config Files under /etc/asterisk
3:25PM 1 AMI check if connection is alive
1:36PM 1 question on digium repo
1:00PM 3 Different box for SIP and RTP
12:56PM 2 AMD tweaking
12:14PM 26 AMI perl daemon
10:50AM 0 1.8.4 quitting console
10:19AM 0 1.8.4 keeps quitting console by itself
 
Sunday May 15 2011
TimeRepliesSubject
11:26PM 0 Asterisk 1.8 and 1.4 SlackBuilds for Slackware Linux
7:59PM 0 Alarms Sound files
7:12PM 0 asterisk-users Digest, Vol 82, Issue 52
 
Saturday May 14 2011
TimeRepliesSubject
11:51PM 12 iptables for Asterisk - Any good guides out there?
3:37PM 4 Asterisk 1.41 - Warning and Notice about contact info and stale nonce
3:31PM 0 How to install the new cdr-stats?
3:59AM 21 Asterisk-cpu utilization > 60 %
 
Friday May 13 2011
TimeRepliesSubject
6:58PM 4 OPTIONS Keep alive - Reply: 481 No subscription
6:37PM 0 Unusual message
5:32PM 4 Asterisk 1.6: Custom Name for Recordings file
3:28PM 0 Asterisk 1.6 - voice quality becomes poor after several minutes.
3:14PM 7 res_timing_timerfd.so Vs res_timing_dahdi.so
2:26PM 1 outbound calls via google voice not answered by toll free numbers with ivrs
2:24PM 5 Backport of DEVICE_STATE to 1.4
2:19PM 0 Unknown Agent Status on DAHDI
1:02PM 9 DAHDI Error
9:46AM 0 Asterisk 1.8 realtime tables.
4:54AM 0 Blocking multiple SIP registration
4:38AM 1 1.8.4 Core Dump after installing from source
3:08AM 2 asterisk 1.8 + google voice
1:44AM 3 undefined symbol: cap_set_proc on several modules after installation from source
 
Thursday May 12 2011
TimeRepliesSubject
8:57PM 0 asterisk 1.8 somehow dead
8:27PM 1 how to reload agents.conf ?
7:40PM 3 lead time for RPM's?
7:29PM 0 regarding core modules
6:16PM 1 Problem with PSTN calls (Asterisk as SIP client on embedded device)
4:50PM 2 Realtime - ara180
4:50PM 22 Light indicator managed by Asterisk
2:33PM 3 ConfBridge for 1.8 ?
1:11PM 0 log full of Name or service not known
10:07AM 3 multiple calls into hold
7:40AM 0 Friday VUC: Discussion of Mobile SIP, Microsoft Lync
7:10AM 0 About minimum requirements to install PSTN GW+SIP Client
5:57AM 2 Different IP addresss for SIP and RTP
1:54AM 2 Disabling echo cancellation by software
12:29AM 2 Higher CPU usage on 1.6.1 than 1.4?
 
Wednesday May 11 2011
TimeRepliesSubject
5:30PM 2 Asterisk SIP Trunking with Cisco UC 560
5:21PM 0 kernel: dahdi: Master changed to B4/0/x
4:57PM 8 concurrent call tracking
3:48PM 6 CLI - displaying all channel variables
3:06PM 2 With what options is asterisk compiled in rpm's
12:04PM 2 no audio with SIP:INFO in meetme
7:16AM 0 obd call drops after few seconds : only for mobile numbers
 
Tuesday May 10 2011
TimeRepliesSubject
8:23PM 37 When someone helps you, at least let them know if the problem is resolved or not
7:27PM 1 iax2 Max retries exceeded to host
4:50PM 4 About X100P and TDM400P analog card in China
3:37PM 1 Using MixMonitor()
2:38PM 8 Asterisk 1.8.4 Now Available
12:21PM 1 Plotting fxotune dump
1:57AM 19 1.8 and prematuremedia problem
1:12AM 1 ITSP Multi IPs
 
Monday May 9 2011
TimeRepliesSubject
9:03PM 0 Call ends when using SendDTMF(*)
9:00PM 4 Voicemail Configuration
8:40PM 3 Really, really loud ringers
7:53PM 1 Need help defining a stackexchange (i.e. stackoverflow) for telephony
6:28PM 4 Rates Importer Tool
5:28PM 0 high PDD
5:25PM 2 iax2 issue in asterisk
3:48PM 0 Free Alarms sound
2:32PM 5 asterisk syntax highlighting for gedit
1:11PM 7 Trying out a new version with sangoma card
1:01PM 0 Ustream feed as MOH
12:26PM 6 40sec between dial execution and sending SIP request
12:10PM 9 OT - Which Android handset with Wifi-only ?
10:49AM 0 conf syntax highlighting for gedit
9:22AM 0 RTP Path and t or T option
8:03AM 22 OUTBOUND CALLER ID
7:47AM 4 Slightly OT: Android phone as sip-gw?
7:41AM 7 how to play music when dial fail or time out
 
Sunday May 8 2011
TimeRepliesSubject
9:19PM 12 Cisco 7940 phone and tftpd provisioning - for ever?
2:31PM 0 txgain no effect
11:43AM 1 no ringback tone on outgoing call PRI line
12:59AM 9 Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS
 
Saturday May 7 2011
TimeRepliesSubject
11:08PM 0 asterisk-users Digest, Vol 82, Issue 27
5:04PM 3 record call from iax to sip
3:24PM 3 [SOT] Virtualising Asterisk
1:05AM 3 Tricky: Progress, Delay, DTMF / background calling
 
Friday May 6 2011
TimeRepliesSubject
8:51PM 4 Blacklist with *30
6:48PM 5 Supermicro X7SPE (Atom) as Asterisk server?
6:14PM 4 question on ways to activate voicemail light on polycom
5:49PM 0 polycom page custom ring
5:03PM 0 Gateway GSM x Comercio Indevido ?
4:49PM 4 Configuring Voicemail in Asterisk 1.8
4:11PM 1 QueueCallerAbandon is not triggering after 1.8.3.3...
3:58PM 5 Cannot install dahdi-linux on (old) PAE kernel.
3:32PM 0 Cannot built kmod-dahdi-linux for PAE kvariant from SRPM
3:30PM 10 Background music during a call
3:29PM 0 Missed call when call is answered by other phone
3:22PM 6 Asterisk 1.6.2.18, Cisco 79XX not registering
1:04PM 3 TCP Trigger on incoming call request
8:33AM 1 is res_timing_timerfd module stable in 1.8?
4:37AM 0 Audiocodes MP-114 - modem dial not going through
2:54AM 0 how to let the call play audio when the dial fail
 
Thursday May 5 2011
TimeRepliesSubject
10:51PM 0 feedback mechanism
9:14PM 4 ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer
7:11PM 5 Why is PQMSTATUS empty?
6:41PM 1 estimated queue hold time
4:36PM 1 asterisk for g729 to g711
3:51PM 0 Asterisk 10 / Trunk and RecieveFax "F" Option
3:33PM 1 Queues, pickup and transfers
3:10PM 0 Does IAX2 support call completion or callback ?
2:13PM 12 Asterisk 1.8 latest branch safe for production ?
1:30PM 2 Auto dialing Polycoms and other SIP phones
12:46PM 4 [Asterisk 1.8.3.2] Mixmonitor not working on member(calling part) channel of Queue.
12:08PM 7 SIP secruity: username and password
10:10AM 0 Could not place calls through IAX
7:49AM 0 Voicemail message storage in db w/o ODBC?
3:37AM 7 Issue with Asterisk & Aastra 57i at v3.2
 
Wednesday May 4 2011
TimeRepliesSubject
10:24PM 0 Park a call when sip phone becomes unreachable?
10:21PM 9 Cordless VoIP Phones and Access Point hand-off?
5:12PM 3 asterisk-1.8 crash if no extension specified in Dial
5:10PM 3 Remove "name" part of SIP From header
5:01PM 2 Sangoma A400 background noise after a while
4:01PM 0 Compiling extra modules
1:42PM 2 pickup question
9:56AM 2 Invalid use of undefined type when make dahdi
7:43AM 2 asterisk HA for queue calls
1:55AM 2 asterisk 1.4.35 to 1.4.41
 
Tuesday May 3 2011
TimeRepliesSubject
9:31PM 1 Having redundancy, so if first IP failed then send for the other
7:20PM 50 receive faxes
6:13PM 7 asterisk call forwarding
5:57PM 2 dial from voicemail
4:43PM 3 Asterisk 1.6 Questions
3:52PM 0 record call transfered from IAX softphone to SIP hardphone
2:32PM 4 Multiple cards using same IRQ - getting IRQ errors and hissing
2:03PM 8 audiohook.c: Failed to get 160 samples from write factory
1:28PM 0 asterisk 1.8 rpms and additional modules from source
12:34PM 2 Asterisk, bicolor BLF and DEVSTATE
10:16AM 4 Fading voice problem
5:10AM 4 How to debug MixMonitor misbehaviour
 
Monday May 2 2011
TimeRepliesSubject
11:50PM 11 ATA refuses to answer a call?
8:56PM 6 sip busy detect
5:19PM 4 asterisk call completion issue
5:01PM 0 music on hold skipping
1:59PM 8 Retrieving/Streaming audio/video files from DB using over AGI
12:40PM 2 Asterisk repository: asterisk14-addons-mysql
11:33AM 4 out of the blue one way audio
10:09AM 3 default context overrides context of peer
7:20AM 0 queue member invalid
7:15AM 6 Retrieving sound files from DB as opposed to filesystem
 
Sunday May 1 2011
TimeRepliesSubject
6:46PM 4 Odd error in libpri
10:41AM 2 Join and listen to conference call through web-interface
10:20AM 1 Queue Setup