asterisk users - May 2012

Thursday May 31 2012
7:55PM 2 meetme and dtmf
7:37PM 1 Anybody knows a good SBC to download?
7:33PM 0 McBee SC
4:28PM 1 appears to be down right now
3:48PM 9 Queue callers with Callback option without lose their place
3:39PM 2 looking for some quality testers for zoiper softphone for android.
1:21PM 0 Error about codecs AMR-NB.
11:33AM 4 Getting unwanted pager email from Asterisk voicemail
7:05AM 4 PSTN termination in Virtualized Asterisk Environment
Wednesday May 30 2012
9:12PM 0 IAX ATA can't register
6:55PM 0 Asterisk 10.4.2 Now Available
6:54PM 0 Asterisk Now Available
6:34PM 6 Sangoma Card Issue
6:02PM 4 Asterisk 10.4.0 GotoIf to label problem when DUNDi active
3:24PM 0 Introducing Limesco
10:03AM 0 Group call from DAHDI
Tuesday May 29 2012
9:55PM 0 AST-2012-008: Skinny Channel Driver Remote Crash Vulnerability
9:55PM 0 AST-2012-007: Remote crash vulnerability in IAX2 channel driver.
7:55PM 4 unable to create channel of type 'SIP'
5:00PM 0 Certified Asterisk 1.8.11-cert2; Asterisk, 10.4.1 Now Available (Security Release)
3:26PM 0 (no subject)
1:45PM 0 Allowing transfer of only incoming calls
12:57PM 3 axfer with simple CDR
8:10AM 14 Fax Server for Asterisk
4:21AM 3 IMAP Voicemail
3:15AM 0 audiohook errors
Monday May 28 2012
4:10PM 0 [HELP] How to configure Asterisk 1.8 to run with IPv6 in a LAN network
10:18AM 2 How to tell VPM presence without restarting?
5:16AM 0 Which combination of codecs are required?
Sunday May 27 2012
5:11PM 9 Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error
3:14PM 0 (no subject)
12:48PM 0 NAT problem: "Retransmission timeout reached on transmission … for seqno 2 (Critical Response)"
9:31AM 0 Call Forwarding
Saturday May 26 2012
10:13PM 3 Common/Reasonable Assumption on DID/Channel over-subscription
5:08PM 1 Linksys PAPT2
8:49AM 1 unsubscribe
5:15AM 0 Help ! Audio not stored .
Friday May 25 2012
9:30PM 3 Loss of RTP stream during DTMF collection
7:11PM 0 URA
4:17PM 0 Huawei K3765-HV with Asterisk?
4:10PM 6 Dual- or Quad ISDN cards for PCI-X Slots
2:41PM 0 Digium's new Community Support Manager - Rusty Newton
7:43AM 0 Asterisk Atxfer
7:18AM 2 Function not Registered??
Thursday May 24 2012
8:10PM 5 Asterisk MixMonitor starts recording 44 bytes file
5:51PM 1 extension status using AMI
5:24PM 0 talkoff problem - relaxDTMF is off
2:54PM 4 T.38 debug logs
8:38AM 1 Call Record File and Play
1:41AM 7 Detecting Fax Tones over IAX2
Wednesday May 23 2012
7:28PM 2 SIP endpoints CANCEL when PRI receives Cause Code 31
6:46PM 8 twenty thousands (20, 000) users, which asterisk and how many servers
4:18PM 1 Disable All Asterisk Features (blind xfer, disconnect, etc)
4:13PM 2 No caller id when using cadence with DAHDI
2:44PM 0 Planned service outage for community services
12:16PM 3 twenty thousands (20, 000) users, which asterisk and how many servers?
11:57AM 17 Vitelity Setup
9:02AM 1 Unable to execute 'dahdi_scan > /etc/asterisk/dahdi_scan.conf'
7:35AM 1 Transfer call issue
3:32AM 2 How to stop ringing when incoming PSTN call is answered externally?
Tuesday May 22 2012
9:54PM 11 Deleting OLD Voicemails
7:08PM 0 multiple sip on same ip
5:48PM 2 Asterisk doesn't catch SIP packet from time to time
5:14PM 0 Queue Log data weirdness with Feature Transfers
2:58PM 0 Asterisk AMI SIP channel detect phone
12:02PM 2 sip show peers
4:43AM 1 gr-303
Monday May 21 2012
2:51PM 6 asterisk voicemail
12:10PM 2 add new sip account in sip.conf with API Action UpdateConfig with php
12:03PM 11 Asterisk and the media path
7:26AM 0 BYE message is not relayed to the UAC
6:31AM 0 Call Recording Stream
6:04AM 2 Recommendations on FXS Bank
4:29AM 3 DPMA for Digium Phones
Sunday May 20 2012
10:16PM 1 SSM
5:22PM 0 Save Call Detail state on second leg of a calls
3:14PM 0 Jumping inside a macro with AEL
Saturday May 19 2012
6:16PM 6 IAX2 passing back and forth variables
5:33PM 3 SET SIP_CODEC and Video issues
4:15PM 3 Realtime peers and trunks coming from the same IP
4:10PM 2 make and receive call from dial-up modem
12:33PM 1 Testing for media?
9:26AM 1 Slow AMI Originate
7:48AM 3 Extensions routing
1:48AM 0 BroadVoice Unlimited World PLUS - Dialplan Update (18/May/2012)
Friday May 18 2012
5:00PM 5 hangup not detected?
1:10PM 0 special digits * # on sip dial string
12:57PM 4 Best practices to route calls according holidays
12:32PM 1 Transfer CDRs
11:00AM 4 Fwd: RTP stats explaination
8:56AM 2 Asterisk 1.8 canreinvite
Thursday May 17 2012
4:47PM 0 dlz-ldap-enum - expose LDAP data to Asterisk via ENUM
12:38PM 2 realtime configuration for /etc/dahdi/system.conf
10:02AM 0 groups and categories
Wednesday May 16 2012
7:14PM 0 Asterisk 10.1.3 on SUSE 10 SP2 - Resolved
5:44PM 5 Fax Problem on direct FXO port
5:07PM 0 (no subject)
4:29PM 0 Asterisk 10.1.3 on SUSE 10 SP2
3:56PM 5 OT - Incoming fax cuts ADSL line
3:07PM 2 50% of time SendDTMF failed
1:03PM 0 Asterisk queue: announce in more than one language or announceoverride
12:29PM 0 AsteriskNow: Does it support call details records?
11:57AM 1 Wrong SIP to SIP SIP Cause mapping
10:28AM 19 how to set iaxmodem receiving speed
Tuesday May 15 2012
9:00PM 9 Asterisk forward call
5:58PM 0 Difference between Asterisk/libPRI/DAHDI versions breaks Caller ID?
Monday May 14 2012
9:39AM 2 How to script cdr_mysql enabling in menuselect ?
Sunday May 13 2012
4:05PM 6 Configuring OpenVOX A400P issues
12:39PM 0 Setting channel variable using AMI
Saturday May 12 2012
4:09PM 4 GSM gateway or PCI Card recommendation?
12:40PM 0 CoreCess Equipment
3:46AM 4 R-Series with NON-DIGIUM card on servers
Friday May 11 2012
1:15PM 7 Least Machine Specs to run a production asterisk server
3:32AM 2 Can run from shell but not from Asterisk System command
Thursday May 10 2012
10:52PM 5 Event response (AMI)
2:45PM 11 SLA - Shared Line Appearance - Polycom
2:39PM 4 enabling dialing by sip uri
1:20PM 0 Email-to-Fax
12:17PM 0 Community event: Open Source Realtime Dinner in Barcelona - June 13th
11:11AM 6 Increasing voice volume without getting echo or entered digit problem
10:08AM 1 where can i find code documentation
8:49AM 32 looking for "solid state" like PC suitable for Asterisk
1:38AM 5 Digium IP Phones
Wednesday May 9 2012
11:17PM 0 Why do I get call twice in one go?
5:59PM 7 Belgian BRI (euroisdn): what to use for a B410P
2:17PM 5 No compatible codecs, not accepting this offer! - after upgrading to 1.8.11
12:47PM 0 Replacing PBX with Asterisk, need feedback on my new architecture.
Tuesday May 8 2012
8:11PM 6 Why did it Hangup?
4:46PM 23 British Telecom ISDN BRI line issues
12:13PM 7 Asterisk 1.8 Transfer CallerID
Monday May 7 2012
3:38PM 3 1.8 busypatterns
12:03PM 0 How to hang up a call after sending SendDTMF() ?
8:24AM 17 using Wifi smartphones as SIP clients
Sunday May 6 2012
5:56PM 0 fake auth rejection??
4:42PM 8 TDM400P: Lifetime & Replacement
11:46AM 7 Why SendDTMF is not working?
3:20AM 1 Problem with SendDTMF
Saturday May 5 2012
6:26PM 0 Asterisk as a SIP trunk termination point.
10:48AM 0 What is the best way to upgrade DAHDI?
9:53AM 2 Mysql identifier not found
9:01AM 4 Asterisk 1.6.2 > 1.8.12
8:16AM 0 Need Hosted Predictive Dialer
4:07AM 16 A worth reading Tutorial for Asterisk Hardware and software configuration
Friday May 4 2012
9:46PM 0 Sound file format and Asterisk 1.8.11-cert1
6:34PM 7 Asterisk backtrace
4:24PM 3 chan_dahdi with asterisk 1.4 and new Linux versions
12:37PM 1 IP address of remote SIP host
8:48AM 1 Asterisk 1.6.2 backtrace
7:11AM 2 Broadvoice Got SIP response 503 Service Unavailable
Thursday May 3 2012
8:28PM 5 Fax .pdf from Asterisk
5:45PM 3 AMI disconnects
3:41PM 14 Asterisk Capacity
12:27PM 5 Asterisk Vs FreeSWITCH for Fax
9:59AM 3 call-back php/agi script
Wednesday May 2 2012
9:41PM 2 Asterisk AMI SIP channel detect phone ringing
6:40PM 0 Asterisk 10.4.0 Now Available
6:40PM 0 Asterisk Now Available
5:48PM 4 parsing issue
4:20PM 3 detecting intl. CLI with +
4:18PM 2 hangup problem on T1 span
12:09PM 3 Asterisk 8 and mixmonitor
12:04PM 6 realtime config for general settings in sip.conf
7:12AM 0 CallerId back to incoming
6:39AM 0 VP8 Codec integration in Asterisk
Tuesday May 1 2012
5:55PM 3 FUNC_ODBC expr must be non-null
4:35PM 0 FW: Auto answer Asterisk ; Unable to create channel of type
4:15PM 7 asterisk go to "holiday" extension but hoiday is not defined
9:56AM 0 Syntax highlight in emacs for editing extensions.ael