asterisk users - Mar 2011

Thursday March 31 2011
TimeRepliesSubject
7:40PM 7 Huawei K3765 + Internet + SMS + Telephone
6:36PM 0 asterisk-users Digest, Vol 80, Issue 73
4:29PM 1 Transfer feature dialing out after one digit
2:46PM 0 moh questions - quality and semi-streaming
2:33PM 0 TDM800P not detecting answer fast enough
1:50PM 1 Atcom/Rowetel IP04 Asterisk Appliance US Source
1:39PM 3 ** to disconnect and make a new call
12:52PM 1 is downloads.asterisk.org down?
12:34PM 0 Asterisk 1.8 Dimensioning.
 
Wednesday March 30 2011
TimeRepliesSubject
7:34PM 5 CDR Mysql adaptive Colum
6:32PM 6 chan_dahdi unknown dependency problem
3:46PM 3 dtmf_2833_1.pcap: what PCM codec? ulaw or alaw?
2:17PM 0 Updated: 10 Minutes: Asterisk PBX on Amazon EC2
2:15PM 0 Asterisk 1.8.3.2 core dump chan_sip.c
9:34AM 0 asterisk and COLP
2:28AM 0 Discover when remote phone answers through IAX2
 
Tuesday March 29 2011
TimeRepliesSubject
10:16PM 5 Cisco IP Phones and Asterisk
10:11PM 1 E1 PRI configuration: DAHDI and LIBPRI
5:52PM 5 wrong from URI in options message
2:05PM 3 Debugging not going to log file
1:19PM 1 Get phone number from SIP header PAI
10:57AM 0 Asterisk Transfer Extensions
9:38AM 0 disconnecting destination channel
 
Monday March 28 2011
TimeRepliesSubject
5:27PM 0 Is History-Info (RFC4244) supported ?
4:19PM 0 special control 16
4:04PM 5 s extension not working
1:26PM 0 Asterisk SS7 error
1:17PM 0 Channel status with AMI originate calls
12:55PM 0 DAHDI, IAX2 and SIP considerations for Early-Media / Alerting
12:54PM 3 DTMF input while waiting in queue...
12:41PM 5 Variable. AMI and dialplan
12:20PM 51 asterisk and fail2ban
11:44AM 3 Dialplan help: hang up incoming call and call the number back
11:21AM 0 Queue(): how to Perform operations at the time of call sent to Queue member but not answered.
10:22AM 1 problems with blind transfer on GXP-2000 - Multi tenant asterisk !!
8:23AM 2 AMI redirect from Queue to MeetMe
12:36AM 12 CDR MYSQL missing field data
 
Sunday March 27 2011
TimeRepliesSubject
9:48PM 2 spa8000 spa2102 t38 faxing
7:54PM 0 DAHDI custom ring cadences in 1.8.3
12:54PM 0 Jabber/Jingle to Google users via local XMPP server
 
Saturday March 26 2011
TimeRepliesSubject
3:36PM 1 Asterisks with ss7 problem
9:50AM 8 Checking status of a cell phone
4:15AM 10 pbx.c: We were unable to say the number
 
Friday March 25 2011
TimeRepliesSubject
11:05PM 2 White papers or success cases to convince a customer?
10:33PM 4 Removing Polycom Transfer Softkey
8:19PM 0 Asterisk with FXO card only, no network
7:40PM 15 Back-to-back asterisk PRI issue
6:49PM 5 reload command not availeble asterisk 1.8.x
6:08PM 0 3com 3102
4:15PM 2 checking dahdi channels
2:36PM 6 Why shouldn't I use 1.8?
1:51PM 2 asterisk 1.8 question
6:56AM 0 Today on VUC, Dan York on Google Voice + SIP
 
Thursday March 24 2011
TimeRepliesSubject
10:08PM 3 Streaming Hold Music
8:58PM 6 Sox and bad quality when converting to 8 kHz
7:19PM 2 Linux Based Billing and CDR
4:27PM 0 Asterisk Tech Tips: Calling With Google Starts At Noon Central (30 minutes from now)
3:58PM 10 Filtering on from caller id
1:46PM 0 Mail list issues?
11:18AM 0 Digium TC400 cards query
8:55AM 5 Fwd: Asterisk 1.6.2.10 & CDR custom added field
5:49AM 1 SIP Invite and Asterisk API/Variable
3:53AM 7 Issues with Digum Repos / AsteriskNOW & Bad Packages
 
Wednesday March 23 2011
TimeRepliesSubject
11:18PM 3 Forwarding XXXX to XXXX prevented.
10:46PM 3 spa8000 t38 faxing
10:32PM 5 Asterisk 1.8 Packages for Debian and Ubuntu
10:05PM 9 asking for some help
9:03PM 3 dahdi restart warning
8:38PM 1 OT: Have unused DID's; where to warehouse?
8:27PM 2 using ${EXTEN} with waitexten
7:15PM 1 Hang using Festival application
6:36PM 1 Asterisk Queue ACD when the queues and agents has the same priority/weight
5:56PM 1 Sangoma A102D wanpiple issue with dahdi
4:58PM 15 What is the most stable version of asterisk?
10:40AM 2 Asterisk using as a SIP client
6:01AM 5 Problems Extension with a Call In on Asterisk 1.6
 
Tuesday March 22 2011
TimeRepliesSubject
8:06PM 3 Sangoma wapipe installation error
7:18PM 0 conference room ideas
4:57PM 1 How to use Atxfer in AMI
4:53PM 3 Asterisk PRI back-to-back connect
4:41PM 3 Act! Integration
2:54PM 2 question on show channels
1:05PM 0 Asterisk 1.6.2.10 & CDR custom added field
5:56AM 4 Usage of lock in CDR
1:17AM 0 Play different voice-mail messages based oncertain conditions
1:17AM 0 Multi-Tenant Hosted PBX system with Resellerfunctionality
1:05AM 3 Play different voice-mail messages based on certain conditions
 
Monday March 21 2011
TimeRepliesSubject
11:49PM 3 IAX Call token revisited
11:38PM 7 Multi-Tenant Hosted PBX system with Reseller functionality
7:55PM 3 record individual callers in confbridge
5:28PM 0 Reminder: Asterisk Tech Tips: Calling With Google on Thursday at 12PM CDT
4:24PM 7 Queue pause vs logged out ?
4:22PM 5 1.8 realtime - segfault
11:45AM 10 wrong time retrieved from system command
9:38AM 1 iax2 sound problem
9:04AM 0 Problem routing call to fax machine on DAHDI FXSport
2:43AM 0 Record individual callers in ConfBridge?
 
Sunday March 20 2011
TimeRepliesSubject
8:24PM 1 why does "core show channels" on 1.8 not show the channel
8:03AM 5 Asterisk
5:47AM 0 switch statement in extensions.conf
 
Saturday March 19 2011
TimeRepliesSubject
8:38PM 0 Single vendor for IMAP VM storage
6:10AM 2 Getting No Antenna bar when behind a NAT
 
Friday March 18 2011
TimeRepliesSubject
5:52PM 10 One PRI card with 2 (or more) Telcos
10:33AM 0 DISA DTMF problem
3:02AM 4 Problem routing call to fax machine on DAHDI FXS port
 
Thursday March 17 2011
TimeRepliesSubject
5:35PM 0 Asterisk 1.6.1.24, 1.6.2.17.2, and 1.8.3.2 Now Available (Security Releases)
3:45PM 2 Status of Queue Members
3:44PM 2 [1.6] Where to put "options wctdm opermode"?
3:36PM 0 blind transfer from AGI triggered call -> dropped
2:18PM 4 [1.6.2.5] Asterisk can't find MOH file
1:57PM 1 Getting the missed calls using Asterisk Manager
1:12PM 6 Answering machine detection for a second leg call generated by a call file.
11:29AM 0 Trying to turn off TLS....
11:24AM 0 Asterisk not logging originating IP of a brute force attack
10:59AM 0 why asterisk sip dial procedure take long time
9:59AM 2 [1.6/Ubuntu] What packages for * + Dahdi?
9:50AM 0 Passing an argument to a macro within an Originatecommand
5:37AM 1 SIP registration DoS but no logs in messages
2:07AM 7 Call are established, but voices can't be heard
 
Wednesday March 16 2011
TimeRepliesSubject
11:00PM 0 Asterisk 1.6.1.23, 1.6.1.17.1 and 1.8.3.1 Now Available (Security Releases)
10:50PM 0 AST-2011-004:
10:50PM 0 AST-2011-003:
10:05PM 0 Multiple Parking Lots Being Redirected to the Wrong Parking Lot
7:39PM 2 Trunk form asterisk1 to asterisk2 fails
3:11PM 0 Connecting Asterisk to Siemens Hipath, 3750
2:44PM 0 (no subject)
2:22PM 17 Pushing info to a Polycom phone - from outside of the local network
1:44PM 1 Discover held channel?
1:24PM 0 Setting up 1.6.2.9 on Debian 6.0 Squeeze
11:09AM 2 chan_sip.c:3115 __sip_xmit of 0x108d33c0 (len 523) to xxx.xxx.xxx.xxx:0 returned -1: Invalid argument
8:34AM 0 SIP Call setup time monitoring in Asterisk
7:38AM 5 Extract Remote-Party-ID from incoming INVITE in dialplan
 
Tuesday March 15 2011
TimeRepliesSubject
10:19PM 15 Multiple Asterisk
9:52PM 2 Is asterisk 1.8 stable version to upgrade from asterisk 1.6 on live server?
9:35PM 1 Passing an argument to a macro within an Originate command
9:19PM 5 signal amplified by asterisk
8:59PM 1 Auto Answer in manager
6:06PM 3 call file for page auto-call
5:51PM 0 FW: [newtech-1] Uncovering Spoken Phrases in Encrypted Voice over IP Conversations
4:30PM 13 [1.4] Failed callfile doesn't jump to "failed" extension
2:11PM 11 Some errors
1:59PM 3 call being rejected
1:54PM 6 [1.4] Asterisk doesn't hang up?
1:48PM 1 AMI Timestamps unit
9:38AM 1 Ast 1.8_CentOS5.5 with timerfd as timing source
9:18AM 3 How to send Hold invite from asterisk to other
 
Monday March 14 2011
TimeRepliesSubject
9:52PM 0 New Webinar Series For Asterisk Users: Asterisk Tech-Tips
7:19PM 3 Asterisk -rx command not returning data - Version 1.4.33.1
5:17PM 0 Anyone (else) need an asynchronous asterisk event->action framework ?
3:01PM 10 Asterisk 1.8 paging with ploycom
8:07AM 1 sip show channel and t.38
 
Saturday March 12 2011
TimeRepliesSubject
7:11PM 0 Hi all
6:22PM 2 Call flash transfer
3:43PM 0 Need to hire full-time linux/asterisk tech in North Atlanta
2:27PM 1 G.711.0
10:00AM 1 Asterisk and PlayBack
 
Friday March 11 2011
TimeRepliesSubject
8:35PM 5 Asterisk 1.8 AGI error ast_carefulwrite: write() returned error
8:35PM 1 Anyway to monitor SIP debug from originator and terminator separate of each other on two screens?
7:02PM 0 dnsmgr_lookup
3:16PM 4 How do you handle queues with AMI?
12:17PM 0 Exceptionally long voice queue length in asterisk 1.6.2
9:58AM 1 Automatically unpause a paused queue memeber - bad idea?
1:37AM 0 regarding authentication with out challenge.
 
Thursday March 10 2011
TimeRepliesSubject
9:30PM 7 Asterisk queues : command to run when a call is being bridged
7:31PM 15 Is H323 supported when installing Asterisk from Digium Yum repository?
5:52PM 4 Dialplan: funcionality testing
4:53PM 2 [1.4.21.2] Read() disconnects half-way through?
3:02PM 1 Metaswitch to Asterisk problems
1:21PM 0 console.conf.sample in 1.8.3
12:52PM 9 Connecting Asterisk to Siemens Hipath 3750
10:55AM 0 Experience with Phones, Asterisk and other pbx, cloud services, etc
10:49AM 1 Is this true for Asterisk as SBC?
10:13AM 3 problem with crashing Asterisk 1.8
7:50AM 0 Display something on the top line of Polycom SPIP 3.1 screen
2:28AM 1 ChanSpy with alphanumeric SIP channels [1.6.2]
2:04AM 9 [1.8] Unable to Register: Registration denied because of contact ACL
 
Wednesday March 9 2011
TimeRepliesSubject
10:40PM 17 One Way Audio
9:53PM 0 Anyone have BRI working with Asterisk 1.8, Latest DAHDI, LibPRI?
9:37PM 0 1.8 and no alsa input
7:26PM 3 Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw)
5:16PM 1 asterisk 1.8 still need dahdi
5:07PM 1 No audio after 15 minutes on Asterisk 1.8
4:39PM 1 HK DIDs
4:09PM 0 BLF, Directed pickup and Polycom 601 with SIP 3.1.6
1:13PM 1 Asterisk pri card replecement
11:14AM 6 Multiple SIP endpoint registrations
11:01AM 9 [Opinion Request] SIP phones that work well with Asterisk
9:17AM 12 SIPAddHeader not working
5:35AM 7 doorphone?
 
Tuesday March 8 2011
TimeRepliesSubject
3:40PM 5 Weird PRI error on an QUAD E1 span: Ring requested on unconfigured channel 255/255
12:22PM 20 [1.4] Reading phone number the French way?
11:31AM 3 (fast) AGI and AMI synchronization ?
10:05AM 3 CDR and call transfers :)
2:22AM 2 TDM410P & dahdi driver == no lights?
12:51AM 2 Sip/google
 
Monday March 7 2011
TimeRepliesSubject
10:52PM 3 Asterisk 1.6 MySQL Realtime fails to connect with working username and password.
10:49PM 5 Error compiling Asterisk 1.8.3 on Sun SPARC x64 w/Debian Squeeze
10:15PM 8 1.8.3 - IAX - echo - jitterbuffer
9:35PM 6 [1.8.3] Error compiling Asterisk: __sync_fetch_and_add
6:58PM 2 Error loading module 'res_fax_digium.so'
2:16PM 2 Help on incoming
10:04AM 2 Cisco 7942G IP Phone firmware conversion from SCCP to SIP.
5:10AM 3 Mirrors in Australia?
 
Sunday March 6 2011
TimeRepliesSubject
11:14PM 0 AstLinux 0.7.7 Release
7:35PM 1 Inadyn error
5:59PM 2 Any good tutorials for setting up Asterisk SNMP and Cacti for remote monitoring?
5:10PM 0 Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit.
6:54AM 4 fail2ban + asterisk
3:47AM 0 imsdroid on droidX to asterisk: No matching peer found
3:06AM 2 ignore this test
2:51AM 2 Early codec selection / negotiation
 
Saturday March 5 2011
TimeRepliesSubject
6:14PM 4 Configuration for Multiple PRI cards
5:26PM 4 Prepaid Billing other than A2Billing
7:28AM 4 Asterisk, Sent accountcode between 2 asterisk
7:25AM 3 Help Asterisk / API / Perl
3:31AM 0 [announce] jkSMS
2:49AM 6 can anyone tell me how to set asterisk to record all phonecall
12:42AM 1 2 ip phones and 1 normal, can't neither send nor receive calls at all...
 
Friday March 4 2011
TimeRepliesSubject
6:49PM 4 OT: OpenSIPS vs Kamailio -- which do you use and why?
5:13PM 5 server performance....
2:46PM 1 GXW4004 - lines get stuck
2:07PM 7 Asterisk <-> Lync / Call Center Transfer / Refer
11:37AM 3 Gosub and 'h' (again?)
8:30AM 11 Loudness of recorded wav-audio
5:50AM 1 How is Libpri developped ?
5:00AM 0 DAHDI-Linux 2.4.1 and DAHDI-Tools 2.4.1 Released
 
Thursday March 3 2011
TimeRepliesSubject
5:46PM 0 Friday #vuc at 12 Noon EST
4:22PM 8 SIP Provider Recommendation in US
4:20PM 1 Contact Directory on Polycom phones
4:07PM 1 [ASK] can't make call
3:02PM 2 Sangoma PCI vs PCI Express card
2:42PM 18 mySQL connection testing
2:13PM 7 [1.4] Forcing Asterisk/Zaptel to wait until callee answers?
11:15AM 3 VoIP Bandwidth Calculator
8:04AM 1 asterisk dump core when i try to record my name on the voicemail
7:02AM 13 Testing from where number is...
5:59AM 2 chan_skinny and Cisco 793X (7936) support in 1.8
5:20AM 3 Converting MP3 files to wav for Asterisk
1:17AM 4 How do I find a phone numbers issued by Rogers?
 
Wednesday March 2 2011
TimeRepliesSubject
10:29PM 5 how to use qualify times to route calls
9:54PM 2 Functionality Questions
9:33PM 4 Question on Asterisk 1.8 and Wait()
2:46PM 2 asterisk behind nat
2:34PM 2 Doubt about cdr on asterisk
2:11PM 0 Missing audio
1:25PM 2 [1.4] Comparing value of string with spaces?
12:43PM 0 Hardware recommendation needed
12:39PM 0 Asterisk 1.6 and windows RTC
11:47AM 1 Registering Cisco 7942G IP phone with Asterisk!.
9:54AM 3 [1.4] Call progress for Zaptel 1.4.3.1?
8:57AM 1 Asterisk 1.8 SIP realtime and NAT
8:43AM 1 GSM-Card for Asterisk / recommendation needed
1:18AM 0 Intermitent voice issues
 
Tuesday March 1 2011
TimeRepliesSubject
5:23PM 2 Caller ID
4:34PM 9 records inbound and outbound calls
12:08PM 3 [zapata.conf] What is "wink"?
11:24AM 0 IRQ 0 on BRI card (B200E)
10:08AM 0 [1.4] Simple way to bridge two channels?
9:20AM 8 wav files are not playing asterisk
9:04AM 3 two questions regarding incoming call
2:49AM 1 Question about how traffic passes from phones
2:10AM 1 duplicate keys change from zaptel to dahdi 2.4.0
1:19AM 4 TLS/SRTP calls go to circuit busy.