asterisk users - Nov 2008

Sunday November 30 2008
11:02PM 1 Oslec issue
5:27PM 3 DTMF Tones
12:39AM 1 DAHDI issue in dialplan
Saturday November 29 2008
10:54PM 0 pp_each_user(), pp_each_extension()
7:33PM 0 asterisk-users Digest, Vol 52, Issue 81
1:19PM 1 GSM gateways - which one ?
5:43AM 1 cannot open shared object file: No such file or directory
4:18AM 2 Trixbox OpenR2
3:02AM 0 received wrong state events for originate command
Friday November 28 2008
7:06PM 1 How to disable trunk from the cli?
4:20PM 0 Asterisk and multicast RTP
4:11PM 0 Friday at 12 Noon ET, the VoIP Users Conference reminder
4:00PM 1 Asterisk SIP security
3:56PM 0 Calls drop after a couple of minutes.
3:07PM 1 RTCP too short
1:32PM 0 length of field names
12:24PM 0 [SPAM] - Asterisk and S-Bus - Email found in subject
11:13AM 1 Priority between calls from different queues
10:18AM 0 [SPAM] - Re: FW: did notregister itselfduringload - Email found in subject
9:56AM 1 MixMonitor with non-20ms packets
5:51AM 1 Anonymous callerid
3:01AM 1 Windows Mobile 6 SIP client: Remote host can't match request NOTIFY to call
2:25AM 2 force channel hangup
Thursday November 27 2008
4:02PM 1 originate problem
1:05PM 2 Wellgate & Asterisk
11:09AM 0 trunk peer not registering after migrating installation
11:03AM 5 Any 1.6 SendFAX example ?
10:14AM 0 Disable Transfer
8:24AM 1 Unknown signalling method 'bri_cpe
6:11AM 0 Softphones with RPID and BLF
Wednesday November 26 2008
9:36PM 0 CDR Hangupcause
9:14PM 0 Softphone IP publico x privado
9:04PM 3 2 Asterisks to one PBX - E1 conection
7:09PM 0 spandsp not recognized by menuselect on Lenny [SOLVED]
5:24PM 0 Problems with Rhino Channelbank...
4:18PM 1 sip MWI Messages-Waiting: always reports no messages
4:02PM 6 spandsp not recognized by menuselect on Lenny
3:48PM 1 Customized CDR Records
3:29PM 0 1.4.x Strange Vocemail delay
2:41PM 0 MS Exchange IMAP Voicemail
2:19PM 7 Dahdi, b410p and looping from 1 port to another
2:15PM 1 language and meetme issue
2:10PM 8 Mobile as FXO
12:33PM 1 Hints stopped working suddently
11:47AM 1 Channel variable to identify the calling SIP peer
11:06AM 1 SVN
8:30AM 0 Asterisk voicemail and Lotus Notes
4:26AM 2 Ring/Off-hook in strange state 6 channel X
1:14AM 1 bridging - Didn't get a frame from channel
Tuesday November 25 2008
10:56PM 8 Connecting AS5350XM with Asterisk
9:17PM 2 half channel audio after upgrade to 1.4.18
6:33PM 4 MOH Realtime
3:15PM 1 AsteriskNOW 1.5 upgrade from 1.4 to 1.6
1:04PM 2 Disabling Call-Waiting
11:22AM 0 Set a specific registration expiry value to a given peer without touching defaultexpiry in sip.conf ?
7:10AM 1 cdr mysql error
Monday November 24 2008
5:44PM 0 reducing iax packet size
12:51PM 1 play sound while executing agi script
10:12AM 1 SendText and non-ASCII characters
Sunday November 23 2008
10:47PM 2 asterisk and vlc
10:44PM 2 Problem with DAHDI and OSLEC integration.
8:28PM 1 Asterisk 1.6, IMAP Voicemail and externnotify
2:42PM 14 CDR Design
12:19PM 1 Asterisk 1.6 mysql cdr log problem
4:17AM 2 How does IMAP notify Asterisk that I've read a message?
3:22AM 1 SendImage()
3:09AM 0 Large Asterisk installations (~10, 000 extensions), preferably at universities
Saturday November 22 2008
1:55PM 1 IMAP voicemail with Exchange (was: A way to run extenrnotify when IMAP events take place...)
7:03AM 1 Upgrade 1.4.19 to 1.6 => segementation fault
6:42AM 1 Asterisk Instant message passing with eyebeam
6:08AM 2 Need Recording Solution in Asterisk
4:02AM 5 CDR Desgin
Friday November 21 2008
11:22PM 0 MoH in a loop
11:09PM 2 MOH Realtime Problem
9:12PM 2 MozIAX - Mozilla IAX2 soft-phone 3sec delay
8:49PM 1 Setting up to reveive faxes.
6:09PM 2 TrixBox problem...
5:51PM 2 hint priority with 50 channels
5:28PM 2 Log level of 500 Server Internal Error.
4:32PM 0 OT - SIP message encoding to enhance text display
4:28PM 4 upgrade from 1.2 to 1.4 and now half channel audio
2:59PM 1 Ping
2:06PM 2 SPA2100 transfer to ASTERISK CID
10:09AM 0 PSTN Gateway setup
7:53AM 2 sip trunking and call transfer
4:54AM 4 Large Asterisk installarions (~10, 000 extensions), preferably at universities
4:15AM 0 Group count not being preserved when transferring a call into a conference
Thursday November 20 2008
11:31PM 2 SVN - DIGIUM
11:20PM 2 A way to run extenrnotify when IMAP events take place...
10:55PM 2 ISDN Cause codes
10:33PM 1 Playback using AMI
10:27PM 0 OT: ATA causes random DTMF in stream
9:58PM 2 Limit the number of users in a meetme conference?
9:33PM 0 DTMF issue
8:55PM 0 dial console/dsp hear crackling in headset
8:52PM 0 Elastix workshop in Toronto; Wed Nov 26th, 2008
8:14PM 1 A question about how much an Asterisk Dcap consultant and a Sipmaster make
7:07PM 0 Rv: sas
6:54PM 0 VB 6 developer needed
6:01PM 0 Subversion Mirror Down for Maintenance
5:49PM 0 Disable native bridge?
5:47PM 1 Low RX volume and half duplex/"walkie-talkie" on AEX-804E
5:33PM 1 Sending / Receiving sms messages with Portech 370
5:02PM 4 SIP to IAX2 with delayed echo
4:02PM 4 Using MAC or extension number as SIP identifier
3:57PM 1 Macro conversion in 1.6
2:32PM 2 Collect digits from the Callee after the Call is connected.
1:51PM 1 Voicemail in Real Time
1:46PM 1 Load balancing Asterisk.
12:45PM 0 DTMF payload
11:47AM 1 Voice Mail
9:11AM 1 jitterbuffer
8:15AM 1 echo cancellation for sip phones
2:40AM 2 Any other "free" toll free SIP providers out there?
Wednesday November 19 2008
11:14PM 1 dahdi_test drops after restarting Sangoma driver
7:57PM 3 puzzle
7:20PM 1 Upgrading Asterisk and FreePBX from 1.2 to 1.4
7:07PM 3 TDM400 (?) zap hangup
6:07PM 2 VoiceMail - audio problem
5:22PM 1 Howto grab back call transfered from SIP phone
1:39PM 1 Asterisk NOW - Where to start
1:13PM 1 IF else
11:10AM 3 P2P
10:33AM 4 Role of asterisk
8:12AM 1 presence with polycom DND
5:51AM 1 Aeterisk NOW 1.5beta1 - CDR problem....
3:30AM 4 question about connecting with Mobile Base Station
12:56AM 5 help with dahdi
12:00AM 1 Asterisk 1.6 call files Disposition=NO ANSWER
Tuesday November 18 2008
10:46PM 2 Fwd: Polycom phone time behind one hour.
8:42PM 0 Realtime MOH
6:19PM 1 setting up callback
6:00PM 2 Do Digium Digital Cards Handle Remote Loopback Command?
5:33PM 1 diax debian package
5:30PM 1 sound quality between two back-to-back asterisk
3:55PM 1 Incoming Transfer
3:45PM 1 Configuring Sangoma BRI with zaptel?
2:22PM 0 Crash when rebooting or unload xorcom modules
1:54PM 1 Asterisk and gtalk2voip
1:08PM 1 meetme command from 1.4 to 1.6
12:56PM 1 Asterisk not reading fast DTMFs, was: PBX -> PRI -> * -> Telco not working
12:29PM 2 Asterisk with or without OpenSER
11:14AM 2 FOP with Asterisk 1.6. No call Information.
10:30AM 4 busy-level / busy-limit Asterisk 1.4.22
9:05AM 0 Asterisk Realtime and device contexts
5:28AM 1 How to Barge specific extensions
2:11AM 2 HPEC performance
Monday November 17 2008
11:55PM 0 Message 12216
11:54PM 2 two sip listening ports for single asterisk
11:04PM 1 Picked up calls die in exactly 20 seconds
8:03PM 1 MixMonitor Problem
5:31PM 1 Deny FOP originated calls
5:16PM 1 AMI Events disabling.
4:59PM 0 dahdi and asterisk 1.4.22
4:55PM 4 Digium Card Noice issue
4:38PM 1 Hints and realtime
3:04PM 2 Full Duplex
2:43PM 1 asterisk conference
2:32PM 1 ALL of DIDx Down?
1:08PM 0 E1/channels
8:26AM 4 Debugging Asterisk
6:37AM 0 IAX2 client for 'eee pc 1000'
2:41AM 2 upgrade to 1.6
Sunday November 16 2008
11:33PM 0 Record Application
9:49PM 2 dahdi compile error on svn
9:04PM 0 DialPlan
7:04PM 1 Caching Asterisk SIP useragent info?
5:07PM 1 iPhone SIP or IAX client (without proxy)?
4:47PM 0 What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux software)?
11:45AM 0 Info about dstchannel
9:28AM 6 * + Legacy PBX works but strange problem
Saturday November 15 2008
10:55PM 0 RV: MixMonitor and Queues
10:49PM 3 IAX2 client for "eee pc 1000"
10:18PM 1 Asterisk GUI and SIP registration
7:10PM 0 MixMonitor and Queues
6:43AM 2 Polycom low volume
4:25AM 1 PBX -> PRI -> * -> Telco not working
2:17AM 2 Best way to handle include files?
Friday November 14 2008
9:56PM 1 installation
9:04PM 0 Originate on AMI
8:21PM 0 Linksys SPA 400, 901 and 921 with asterisk
6:50PM 0 Manilla inbound DID
6:28PM 0 PRI users, please read
4:29PM 1 no dial to busy sip line
2:58PM 1 Queue App - Set monitoring dynamically
2:02PM 4 Looking for a good lightweight Linux softPhone
1:23PM 1 kick from conference message on 1.2.23
12:59PM 3 asterisk/E1
10:24AM 1 RTP LOG
8:46AM 1 Dedicated Servers
7:52AM 0 openLDAP
7:07AM 0 Virtual Question
5:30AM 1 ParkandAnnounce?
3:32AM 2 Preserving DID numbers on PRI pass through
Thursday November 13 2008
10:05PM 1 Asterisk and Zaptel version numbers -- how close is close enough?
2:16PM 2 How long will Asterisk 1.4.x supported/maintained
2:16PM 5 database queries from extensions.conf
1:36PM 0 [Fwd: [OpenSIPS-Devel] RFC: new opensips design]
1:04PM 0 Tos_sip
11:26AM 0 Problems with Licensed g729a codec from Digium
9:25AM 0 cisco voice gw / cisco call manager /asterisk for voice record, ivr
7:57AM 1 Parking help - causing Asterisk crash
6:18AM 2 asterisk setup w/ voIP phones
12:03AM 0 1.4.22 CALLERID(num)
Wednesday November 12 2008
11:03PM 1 List eating mail again?
9:29PM 6 Why Nat=yes Nat=no Option?
8:39PM 3 SIP provider and NAT
6:17PM 1 How to get correct dial result for outgoing calls thru ISDN?
5:25PM 4 E1 PRI to and from SIP screeching
5:04PM 1 What are the minimum realtime fields for sipusers?
4:44PM 1 QueueLog from AMI
1:03PM 1 Query about Call Recording with Asterisk / Freeswitch in Cisco IPCC deployment
9:55AM 4 test OpenVox B400P and junghans card for dahdi BRI wcb4xxp
9:30AM 4 The sound is played but I did not hear
2:26AM 1 Use DECT GAP handsets with Snom M3 base?
2:02AM 2 AS5200 <-> T100P - No alarms but no calls either...
12:27AM 3 Grandstream and pickup
Tuesday November 11 2008
11:17PM 1 AsteriskNOW 1.5 - won't talk to IMAP server
10:54PM 1 Request for testing of new driver for B410P Quad-Port BRI
10:24PM 2 play file from url
9:19PM 3 Use the NEW ulaw/alaw codecs (slower, but cleaner)
7:05PM 1 ztdummy: rtc: lost some interrupts at 1024Hz
5:34PM 1 ztdummy: rtc: lost some interrupts at 1024Hz.
4:21PM 0 Asterisk CDR Error ??
3:41PM 3 OT: Polycom Firmware available (by accident?)
3:10PM 1 view the current calls and their codec
2:24PM 0 help with call with no sound via PSTN
11:59AM 1 Dial outside number using the E1 Link
8:49AM 2 TE410P alarms stay RED with 1.4.22
6:35AM 7 music on hold
2:03AM 0 dial a number while play the sound
1:05AM 2 Server for 25-30 phones, sip trunks over the net
12:34AM 1 What makes TDM400 FXS Connection to TELCO go into Off Hook State?
Monday November 10 2008
7:55PM 1 Voicemail IMAP ./configure error
5:26PM 3 console/dsp asterisk seg fault
5:10PM 3 Asterisk daemon dies about once per day
2:50PM 0 SRTP support in asterisk 1.6
2:11PM 1 Using AMI to determine PRI Channels Used
1:13PM 3 directrtpsetup without reinvite
12:11PM 0 analog issues using xen virtualization
11:56AM 2 GEN-GEN and Manual Ring-Down (MRD)?
11:16AM 6 changing the size of voice packets
Sunday November 9 2008
6:22PM 2 Codec problems when using G.723
3:07AM 3 set(CALLERID(name) not working
Saturday November 8 2008
11:18PM 3 Rolled Distro?
7:13AM 0 ISDN for Dahdi get hangs
Friday November 7 2008
7:35PM 1 Outgoing SIP calls dropped after 30 seconds.
5:25PM 0 REFER problems with Asterisk and OpenSER
5:12PM 1 Providing Ringback
4:29PM 1 DNS A queries for channel
3:17PM 2 help with dialplan
2:35PM 3 TE121B Doesn't Fit PCI-E Slot
2:23PM 4 1.6 Production ready??
2:14PM 0 T.38 without port changes
2:04PM 1 Help with asterisk and avaya SIP trunking
1:58PM 1 is it possible to deactivate RTCP?
1:26PM 0 AEL NoOp not working [SOLVED]
9:22AM 0 asterisk - avaya ip office SIP trunking
7:46AM 0 [OT] Reporting Spam
7:01AM 0 Use of optional new number in ISDN release 22
Thursday November 6 2008
6:27PM 4 ODBCExec and Asterisk 1.6 New Thread
6:07PM 0 [OT] Capitalism (was: Spam from DIDForSale <>)
4:27PM 0 Asking again about busylevel
4:12PM 2 Variable Scope Question
3:42PM 0 Asterisk trunking
12:18PM 2 tired of "midget packet received" warnings
10:57AM 3 RFC: multiple packages editing asterisk config files
10:46AM 2 crashes after upgrade from 1.2.16 to
9:31AM 2 Spam from DIDForSale <>
9:30AM 0 HD Voice conference Friday Nov 7th @ 12 Noon EST
8:56AM 0 asterisk.conf ====> maxload
4:33AM 1 ISDN Cause Code 100, Bosch Integral Management Connection
3:56AM 4 Recommend Wireless IP Phone
3:50AM 0 Missing callerid
1:40AM 0 Agent Question
1:17AM 1 Polycom's lose BLF after Asterisk restart
1:06AM 1 Asterisk Realtime Configuration
12:01AM 2 TDM400 with FXS some handsets not ringing
Wednesday November 5 2008
11:26PM 1 Cisco 776M Any good for connection to local Asterisk server?
10:44PM 1 Type 102 Millwatt Test Line
9:20PM 0 b410p mIDSN - RNIS signaling problems
8:53PM 0 a callerid question
7:30PM 1 Inbound/Outbound undesired behavior
5:39PM 2 ExtenSpy? am I doing it correctly?
5:04PM 1 SER/Asterisk interworking mailing list.
4:49PM 0 Maintenance and Downtime
3:42PM 0 br.Doctor Ester
3:26PM 5 Phishing attempt
2:40PM 1 800 origination
2:33PM 2 Dundi Issues
2:20PM 1 sas
2:18PM 0 SIP Qualify is not working with Postgres
10:39AM 1 AEL NoOp not working
8:01AM 1 How is it best to initialize specific SIP peer settings
Tuesday November 4 2008
11:50PM 2 Sendmail using SMTP authorization
11:26PM 0 WARNING message when calls get into a queue with realtime members (Local channel)
9:31PM 1 shared voicemail box
8:17PM 1 dahdi trunk does not compile with kernel 2.6.27
6:14PM 1 Some progress, anyway...
5:12PM 0 What is the best way to resale termination/origination?
4:30PM 1 users.conf and hints
3:39PM 0 Is SIPPEER curcalls working for you ? [SOLVED]
3:34PM 5 VoIP Users Conference Call Friday Nov 7 On Wideband Voice & Conferencing
2:17PM 2 SPA-962 Time on Asterisk
2:08PM 3 SPA-962 & Asterisk
11:33AM 1 Is SIPPEER curcalls working for you ?
8:10AM 0 manager event privilege: call, all? what is?
Monday November 3 2008
8:41PM 0 asterisk src=dst
6:09PM 0 CLI dial and echo recorder
5:38PM 0 busylevel question
5:11PM 2 Looking for a web video phone?
4:14PM 1 Call quality issue across VPN-> POTS vs SIP
3:24PM 0 loading misdn.conf strange error regarding out of range
1:56PM 1 help with debugging phone call
12:30PM 1 say load new
4:03AM 1 asterisk and bigmem kernel
12:16AM 1 Polycom 430 no hangup after SIP BYE, Status 481 instead
Sunday November 2 2008
4:53PM 0 Returned mail: see transcript for details
1:11PM 1 Question regarding keywords in sip.conf/users.conf
12:11PM 0 Asterisk and Media gateway controller
1:03AM 5 Ztdummy and Asterisk
Saturday November 1 2008
9:58PM 0 Gizmo ok but no audio when incoming
7:17PM 0 asterisk 1.2 and Dial with LIMIT_WARNING_FILE
6:08PM 0 asterisk-users Digest, Vol 52, Issue 1
5:01PM 0 2 Or (971) Numbers for sale
3:15PM 1 SPA3102 interdigit timers bug?
6:25AM 1 Wierd queue question
4:36AM 1 VoIP traffic shaping