asterisk users - Nov 2007

Friday November 30 2007
11:44PM 3 Only call me once
11:26PM 0 v33 of codec_g729a released
11:20PM 0 Asterisk-addons 1.4.5 Released
10:56PM 2 Problem registering Cisco 7970 phone with Asterisk 1.4 running FreePBX
7:47PM 1 Outgoing PSTN calls , unusable voice quality
7:09PM 1 OT - How to add a new TAPI driver on an XP system ?
6:26PM 2 Remote Office, Centrally Shared Voicemail
6:16PM 3 How to setup redundant SIP peers
4:25PM 1 Simple Asterisk to Asterisk SIP Call Setup?
3:12PM 3 Do While loop
2:57PM 1 Asterisk 1.4.15 crash without generating core file
2:20PM 1 Nov 28, 2007 Asterisk Poll Results
1:30PM 1 Off-Topic: Avaya
1:26PM 1 Suppressing certain queue announcement voice prompts
10:06AM 0 OT - Which TAPI driver to use ?
9:17AM 2 My AsteriskNo unable to registration
8:52AM 4 IAX complaints? What are they?
7:35AM 4 How to originate a call from console CLI ?
3:30AM 0 Sip 1.4.x DTMF detection not working
2:07AM 2 Newb Question
Thursday November 29 2007
11:10PM 1 Call Parking/Pickup on a single button
11:07PM 0 AST-2007-026 - SQL Injection issue in cdr_pgsql
10:54PM 0 AST-2007-025 - SQL Injection issue in res_config_pgsql
10:10PM 0 Asterisk 1.4.15 and 1.2.25 Released
7:29PM 1 Adding new recorded phrases to the release
7:29PM 2 Using existing extensions.conf macros, and co-habitation
7:17PM 1 Transfering IAX context
6:05PM 0 Protection switching on PRIs.
5:31PM 1 SLA: Handling of errors in outgoing call
5:28PM 1 Problems with Asterisk 1.4.14 and Queue app
4:59PM 1 least cost routing and asterisk-1.4
1:46PM 1 roundrobin and rrmemory with pre-defined agent order
1:39PM 0 asterisk-users Digest, Vol 40, Issue 82
11:59AM 0 Is it better to use debian binary or compiled version?
11:32AM 1 Hylafax
10:12AM 0 Needed Hardware
8:17AM 0 CDR n Dial A option
7:52AM 0 [Copfilter] Copy of quarantined email - *** SPAM *** [6.5/6.0] Asterisk <-> Nortel Phone Switch
7:52AM 0 [Copfilter] Copy of quarantined email - *** SPAM *** [6.6/6.0] Problems getting Asterisk to detect call in
7:49AM 1 IP Trunk and increasing volume level on diguim card
7:48AM 0 [Copfilter] Copy of quarantined email - *** SPAM *** [7.4/6.0] Re: Asterisk <-> Nortel Phone Switch
6:13AM 0 queue_log duration=NULL
2:20AM 0 FSK signal start after second ring
12:56AM 2 Realtime SIP & BLF
Wednesday November 28 2007
10:27PM 1 Cross-compiling asterisk-1.4 for Debian on a slug
9:42PM 4 G729/MOH Quality
9:18PM 1 Polycom MWI's will not turn off
9:01PM 0 No ACK on 200 OK
8:49PM 1 Unable to lookup host in c= line,
8:41PM 2 What is voice format 8
8:06PM 1 Asterisk <-> Nortel Phone Switch
6:56PM 0 troubles with res_pgsql
6:49PM 3 Bad audio quality in 1.4-SVN when encoding to alaw/ulaw
5:52PM 1 Digium TE120P versus Sangoma A101D-X
4:52PM 4 Sangoma Question
4:21PM 5 To DB or not to DB?
3:08PM 2 Shared line appearance phones?
2:10PM 1 Fw: Remove a TDM Card
2:02PM 1 test
1:52PM 0 Outbound calls through iaxy ATA not hearing ring + appending carrier PIN codes
10:07AM 5 retrieve last number dialled
9:58AM 3 Asterisk on multi-homed systems
9:12AM 0 1 FXS module / PCI express
8:47AM 2 DTMF not recognized on ISDN with Siemens -not IP- phone
8:37AM 2 cvs or svn
6:50AM 6 G729 on wrong bus
4:27AM 2 Billing/Call Control engine : AGI scripts/ AstMan API
3:21AM 0 Re :Recommendations for 100 Wifi SIP phone
2:05AM 3 Multiple Return Values from func_odbc
Tuesday November 27 2007
11:27PM 1 Lost setting up IAXmodem after drive crash
7:53PM 0 Zaptel 1.2.22 and 1.4.7 released
6:59PM 1 Semi-OT Part 2: Videophone
6:27PM 5 Copy or Make + Make Install
6:23PM 0 ResetCDR Options v, a - Asterisk 1.4
6:02PM 3 Urgent question.
5:00PM 1 Can Asterisk act like an ISP dialin server to data callls from Sipura 3000 or other ATA connected devices ?
4:13PM 2 Restricting the manager interface to a number?
4:05PM 4 Snom phones, blinking lights and call pickup
3:33PM 2 Finding the status of an extension
3:31PM 2 Attended transfer to Queue
2:24PM 1 Asterisk API Manager
2:20PM 5 SIP port 5060 closed - how do I open it?
2:01PM 3 Sip to ATA?
1:10PM 10 Asterisk behind a PIX firewall?
6:34AM 0 CDR issue need help
4:51AM 0 Dial application response code--help required
4:10AM 1 Voice mail & Uniden UIP-200 phones
Monday November 26 2007
9:56PM 0 SIP Trunk Problems
9:51PM 3 Correct syntax for IF()?
9:07PM 1 Filesharing + video + voice supported Soft phone
9:06PM 1 VMukti - Filesharing + video + voice supported Soft phone
8:58PM 0 Digium b410p + mISDN echo
8:08PM 2 Broadcast dialing/playback
7:33PM 0 Interesting Conference Request - Can this be done ?
7:30PM 1 Semi-OT: Best Speakerphone
5:41PM 0 Queue with cell phones
4:28PM 2 Asterisk Recording
3:25PM 0 Remove a TDM Card
3:22PM 4 Digium E1 and Digium TDM400P (2xFXO) Help!
3:06PM 2 Possible Conflicts with Junghanns 4 Port BRI and 8 Port Sangoma Analog in Same Box?
2:19PM 3 Problems getting Asterisk to detect call in SUSE9.3, with FritzCard
1:51PM 1 OT: Best firmware for Linksys Router that is "SIP AWARE"
1:28PM 1 iptables requirements for SIP
12:47PM 0 Asterisk B2BUA patch useful??
10:14AM 2 Asterisk version survey
9:26AM 0 How to manage several AMI connections to an Asteris server ?
6:50AM 1 Agi manager session.
6:20AM 2 Get IP address of an incoming or outgoing SIP call
3:55AM 1 passing DTMF upon call answering
Sunday November 25 2007
11:44PM 1 [Record() function] Script stops if user doesn't hit # after msg
9:49PM 0 Recommendation for 100 SIP WiFi phone setup
8:26PM 4 Recommendations for 100 Wifi SIP phone setup
Saturday November 24 2007
8:48PM 3 MSSQL ODBC Connections
2:20PM 1 dial in group
10:33AM 0 Voip Users Conference moves up to 12:00 EST
1:17AM 2 Annoying PRI Channels Restarting Message
12:06AM 3 Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.
Friday November 23 2007
5:03PM 1 Best Prepaid Application?
4:29PM 1 AMI Newstate Ringing events -- Inconsistent caller id ?
4:17PM 1 OT - 3Com and IBM iSeries
3:03PM 1 SIP detects loop when forwarding to voicemail
10:33AM 2 TDM808B 8 port FXO setting problem
7:44AM 0 Enough Turkey? Voip Users Conference today at 12:30 EST to help digest it all
7:38AM 2 How to bridge two connected calls
Thursday November 22 2007
11:12PM 0 Work
9:24PM 1 NAT keep-alive
9:02PM 5 Odd bug in Siemens C460IP ?
9:02PM 4 Calling with hidden callerid
5:34PM 1 Dial problem
2:56PM 1 Toll fraud detection/password script
1:48PM 0 mailbox name length
1:03PM 6 Digium and Asterisk
8:57AM 4 Phones Not Registering
7:28AM 1 common/shared voicemail box
5:34AM 0 Asterisk support V5.2 protocal
2:37AM 0 Vicidial + Unicall mfcr2
Wednesday November 21 2007
11:13PM 3 spandsp as T.38 termination?
10:31PM 1 Problem dialing certain numbers with an E1 PRI
7:24PM 1 sip proxy failover
6:52PM 1 Caller ID Question
6:47PM 3 Aastra 480i CT - No Incoming Calls
6:39PM 1 Queue Drops to Voicemail
3:50PM 0 [DB] Insert only one prefix for multiple numbers?
3:45PM 1 Help Dial extention
3:09PM 1 Problem installing Asterisk
2:33PM 0 Need help in selecting DTMF Mode
9:30AM 0 chan_ss7 0.10.1
9:29AM 5 Softphone to be installed on the Mobile
7:47AM 2 Zaptel 1.4 spec file
2:42AM 1 [1.4 - Record] How to tell if user did leave a msg?
2:24AM 1 quality after call transfer
12:33AM 1 Building an Asterisk 1.4 RPM
Tuesday November 20 2007
10:33PM 1 Asterisk-Users: Termination
9:52PM 2 Music on Hold Problem w/ Transfers
8:23PM 1 How to receive manager events from commands made by an AGI script?
7:46PM 0 FXO incomming call hangup problem
7:36PM 0 Cisco phones and 32 directory object limit
6:59PM 1 Bugtracker to use with Asterisk?
6:49PM 0 automatic blind transfer calls
6:35PM 1 [asterisk-dev] trunk working under windows!
6:27PM 2 e911
6:17PM 0 iaxpeers from Realtime
6:01PM 1 FXO Hangs up automatically
5:33PM 2 Reporting bugs
3:16PM 1 ACD functionality , Skills for agents
2:51PM 1 Realtime extensions configuration - calling user filtering
2:43PM 1 OT - What is Alarm receiver feature ?
12:56PM 1 store 2 separate records in cdr when a call is transferd
11:19AM 0 not sending bye
11:05AM 2 SMS Feature In Asterisk
9:53AM 0 sl75 wlan not able of being pickuped?
9:42AM 1 Realtime - mysql query gives wrong results??
9:21AM 0 MediaHandling--Help Required
8:07AM 1 Problems with losing D-Channel on
2:59AM 1 Switch to Multi-Proc -> Choppy sound?
2:59AM 1 SIP - ooh323 Bridging
2:58AM 1 Interface with NEC NEAX 2400
Monday November 19 2007
10:48PM 1 asterisk manager and perl
6:23PM 3 dialplan design - unknown extension length
6:17PM 0 Natural Microsystems AG Quad
5:27PM 1 AstLinux WebSite Problem
4:51PM 7 asterisk as non-root/best practices
2:14PM 2 blind transfer dumping calls
1:33PM 3 How to enable res_config_mysql
12:45PM 3 Gigaset S450ip and simultaneous calls
9:49AM 5 Registration problem: UA -> SER -> Asterisk
8:45AM 2 Asterisk Sound File
3:53AM 4 Help: How to configure SIP domain on SPA942
Sunday November 18 2007
9:14PM 6 Asterisk on Pcengines Alix board
4:20PM 6 Conference Call Dial-Out to a participant
1:58PM 1 [IAX] Does the client have to use UDP4569 as source port?
1:08PM 0 facilityenable in zapata.conf
8:44AM 1 sip + jitter buffer
4:06AM 2 Trouble with asterisk-users mailman
2:46AM 2 problem with tdm2400p configuration
12:47AM 1 p2p t1 with sangoma hw
12:02AM 0 Connecting Ericsson 4422 or similar set to Asterisk ?
Saturday November 17 2007
10:53PM 0 Polycom Provisioning Tool Source Code Released
9:28PM 1 Building and running mISDN for B410P on Ubuntu 7.04
9:28PM 1 Multiple B410P's in one machine
7:05PM 1 Page Command
3:19PM 1 chan_ss7 0.10
2:41PM 0 Astmanproxy Yahoo Group
1:43PM 0 Blackberry MVS and Asterisk.
1:32PM 1 California based PSTN connections
10:03AM 0 The call does not disconnect at the softphone when caller hangup the mobile
12:38AM 3 modifying a dialed exension before dialplan processing
Friday November 16 2007
11:51PM 0 Asterisk 1.4.14 Released
6:40PM 1 Dumb AGI question
3:58PM 0 Polycom softkey transfer issue
3:00PM 1 Help with Polycom 320
1:28PM 2 Changing audio message to text message
1:06PM 1 channels to destroy
1:03PM 2 Change the Voice promps in asterisk 1.4
8:31AM 2 Which files to be copied
3:32AM 0 dtmf detection
12:16AM 1 Asterisk 1.4 with LDAP
Thursday November 15 2007
11:43PM 1 Help on strange problem...
11:31PM 0 Building an Asterisk 1.4 RPM.
6:51PM 2 Two B-Channel Transfer (2BCT/TBCT) Trobule on DMS100 PRI
6:32PM 1 Music on Hold -- Error
6:29PM 1 OT - best policy for logs
6:18PM 1 Pass CallerID when call forwards to PSTN?
5:47PM 2 make config update-rc.d
5:40PM 0 SPA-2100 into Paging System "Hangs"
5:33PM 2 Asterisk Program Closes
5:32PM 1 Lists dead?
4:49PM 1 Centos 5 Asterisk 1.4 FreePBX Install script
4:37PM 2 Dialing time-out
3:15PM 1 Shared gsm files
1:12PM 2 reload command
1:06PM 1 TE210P Vs TE220P difference
12:14PM 1 DTMF Problem
11:03AM 1 Friday Conference reminder: AGI example Nov 16th at 12:30 PM EST
9:51AM 0 Problems with positions in callqueues
8:38AM 0 pipemedia
6:41AM 0 Queue
3:23AM 0 Integration of Asterisk with MS Dynamics CRM
1:27AM 1 asterisk integration with panasonic analog pbx
Wednesday November 14 2007
10:15PM 3 asterisk-stat problem
9:27PM 1 pip tones in Monitor or MixMonitor
8:24PM 0 Real Time CDR
7:40PM 0 Routing Anonymous Callerid
7:02PM 0 Error: inserting return line in dialing strings
6:32PM 0 Help in getting a dialplan to produce the right CDR info
6:02PM 0 PBX Testing Framework
3:47PM 1 "Whats New at Digium the Asterisk Company" -- Junk?
3:25PM 0 IVR Tree Best Practices
1:59PM 4 Problem with AGI Script
12:51PM 1 Using php exec() in agi script
12:24PM 1 Linksys 942 Call Transfer
10:54AM 1 Asterisk ignoring manager events when busy
6:31AM 0 Asterisk trunk and manager redirect problem
5:21AM 4 What is wrong with this mailing list
4:14AM 0 asterisk-users Digest, Vol 40, Issue 37
3:58AM 1 How to pay for libpri development
3:52AM 2 Nortel digital FXO channel bank? Exists?
3:34AM 6 function voicemailmain
Tuesday November 13 2007
8:51PM 2 Call Forward on SIP unreachable (network failure)
7:05PM 0 Can I connect device on FXS of Sipura 3000 to internet virtually ? - it can only call ISPs numbers on POTS line
6:14PM 1 route INVITE
3:11PM 1 Install Scripts for CentOS 4
2:58PM 2 How to integrate Asterisk with Avaya
2:38PM 1 Conference rooms
2:26PM 4 How to play Asterisk .raw file
2:10PM 1 [Fwd: Re: VoiceMail hangup]
1:44PM 4 Cisco 7911/7941/7970/7971 Softkey XML Files
11:58AM 1 Default mohmp3 : free of rights ?
11:06AM 0 Fwd: Re: Grandstream GXP2020 + Asterisk 1.4.11
9:49AM 0 chan_alsa issue
9:25AM 0 Fwd: Re: Grandstream GXP2020 + Asterisk 1.4.11
8:26AM 1 Toshiba DK - Asterisk Integration
7:22AM 3 Stress-Testing Asterisk
5:07AM 1 Chatterbug
4:04AM 1 MOH Codec Issue
12:14AM 1 ODBC connection to Microsoft SQL Server
Monday November 12 2007
7:28PM 4 VoiceMail hangup
6:54PM 0 No sound from playback and voicemail (Atis Lezdins)
6:47PM 0 No sound from playback and voicemail (Carlos Chavez)
2:46PM 3 No sound from playback and voicemail
1:13PM 1 sip_chan - how to use value of the SIP 'To:' header field for extension logic
1:03PM 1 Internal CallerID problem
12:16PM 2 'h' extension on call-out
11:12AM 1 Grandstream GXP2020 + Asterisk 1.4.11
Sunday November 11 2007
5:35PM 1 IMAP Voicemail -- HELP! Asterisk not playing Greeting!
10:38AM 3 detect asterisk pbx via sip
9:15AM 3 sangoma zaptel patches
5:51AM 0 sangoma asterisk patches
1:35AM 1 RTP traffic not being forwarded
Saturday November 10 2007
11:34PM 2 Record() : How to get filename created with %d?
8:17PM 5 r2 multiframe error
7:26PM 4 mpg123 on Thecus N2100
1:04PM 2 sidetone
12:34PM 5 'Traditional' Faxing
10:46AM 1 Asterisk direct dialing
8:13AM 1 PHP - Queues - etc.
8:12AM 4 asterisk 1.4 prereq
Friday November 9 2007
10:57PM 0 Copying the needed configuration files to be used on new installation
9:14PM 1 H323 registeration and routing the calls
8:11PM 1 Asterisk on Zonbu, impact of transcoding
5:04PM 1 asterisk ODBC dependencies
4:50PM 0 Weekly Conference Reminder: Friday Nov 9th 12:30 PM Eastern Time
2:33PM 0 svn+chan_mobile+Asterisk+Siemens FXO no voice
12:07PM 3 How to get ten-digit number?
10:27AM 0 What can I do with Jabber?
7:55AM 1 Your favorite desktop wifi sip hardphone ?
5:39AM 2 Kernel Native PCIE Network Cards?
12:34AM 4 If caller id is null set to a specific number
12:11AM 4 Wanted: tutorial on troubleshooting SIP issues
Thursday November 8 2007
11:39PM 3 Asterisk as a SIP to XMPP Jingle voice gateway
10:01PM 0 AST-2007-024 - Fallacious security advisory spread on the Internet involving buffer overflow in Zaptel's sethdlc application
9:37PM 1 Switchvox Space Requirements
8:01PM 2 Asterisk and OBDC
7:37PM 3 Cisco IP Communicator with Asterisk
3:56PM 3 'a' extension
3:46PM 2 time on polycom 501
2:13PM 1 Channel variables, any difference with SIP vs. IAX?
12:13PM 2 asterisk and installing rpm
12:10PM 0 dtmfmode RFC2833 and inband
10:50AM 0 make h323 native transfer on stablished call
9:22AM 1 Snom 320 with TDM02B and echo problems
7:27AM 3 Client lost on skinny
7:23AM 0 Polycom IP601 call parking
6:49AM 0 Polycom IP601 (mac)-directory.xml changes don't update phone
6:13AM 2 weird 185 secs timeout call problem
5:03AM 1 extensions.conf pattern match info
4:06AM 0 DeStar finalist in Les Trophées du Libre 2007
Wednesday November 7 2007
9:35PM 3 ztdummy, zttest
8:53PM 0 Little OT: Compilation of EICON driver, fails with capi errors
5:01PM 0 Cisco phone 7911g restarts
4:33PM 0 accessing variables when using SIP vs. IAX
3:31PM 1 Polycom SoundStation VTX 1000 with Asterisk?
2:26PM 1 CDR on channel not posted
1:26PM 1 SIP: "To:" header?
1:07PM 0 Audiocodes over Sat link. and delay
10:56AM 1 Board configuration - specification or recommendation
10:33AM 5 What do you do to keep asterisk alive?
9:29AM 1 Call terminated with error message logged
8:47AM 2 Determination of billsec
8:30AM 1 grandstream troubles
8:08AM 2 OT: Aastra 57i configuration via TFTP problem
7:58AM 1 detecting voltage on fxo
3:52AM 2 wifi
Tuesday November 6 2007
10:52PM 1 Extracting custom headers from SIP REFER
10:12PM 1 dtmf / misdn
10:04PM 2 Pickup Command not working
8:34PM 1 Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID
8:29PM 2 Selecting OSLEC for zaptel-1.4.6
7:25PM 3 Asterisk Help
7:09PM 1 Sangoma S200 and Digium TDM400P together
6:39PM 1 Help: Asterisk info
6:36PM 1 Asterisk 1.4 + Presence
5:30PM 5 asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking
4:09PM 4 MeetMe CPU resources
11:49AM 2 Recording just first part of call?
11:09AM 1 Asterisk and Grandstream both behind different NAT
10:37AM 5 Linksys SPA-941 Unavailable
2:35AM 1 1.4 SIP Jitter Buffer
2:10AM 2 Asterisk & OpenVZ
Monday November 5 2007
11:03PM 0 Two B410P cards in one machine
10:55PM 1 Please explain the correct LED color for B410P
9:54PM 2 Queue Statistics reporting
9:06PM 1 PRI dialout problem with some numbers...
6:39PM 1 Arbitrary limit on length of email address?
6:27PM 1 Help: Static and dropped calls
6:16PM 1 Testcall
6:14PM 2 Free T1 Card?
5:40PM 2 Problem with CDR userfield not being set
5:38PM 1 Meetme - how to protect the conference?
3:34PM 0 Parameters effect on the success registeration
3:21PM 1 OT: Which SIP method to use for this specific behaviour ?
1:21PM 2 Which Variable???
11:15AM 2 How to delete voice mail messages?
10:52AM 0 How to disable Asterisk 407 Proxy Authentication Required Challenge response
8:19AM 1 Not Hearing hello-world Play
8:10AM 2 TE220 PCI express performnace
6:47AM 0 AGI connected to meetme conf gives Failed to write frame error
6:42AM 0 crash
4:35AM 2 Dynamic Queue Members - Auto Logoff
3:42AM 1 asterisk-users Digest, Vol 40, Issue 5
12:51AM 1 Are the ATAs which can allow multiple extensions from one network connection?
Sunday November 4 2007
11:52PM 2 Need Reference sites
11:26PM 3 7960 Queue Issue
5:01PM 1 Issues after upgrading from 1.2 to 1.4: hangup immediately
1:10PM 5 Restart when convenient
Saturday November 3 2007
8:43PM 2 Asterisk versions and H323
8:05PM 0 F3000 not connecting to Netgear router
5:11PM 0 F3000 not connecting to Netgear routing
2:05PM 0 Grandstream 488 and DTMF with a call file
1:52PM 1 Wifi handover/roaming
1:37PM 0 [Fwd: voicemail locked up Asterisk 1.4.13]
1:10PM 0 OT: Snom 300 losing config?
7:50AM 0 3pcc/click to dial accounting
3:39AM 1 Asterisk SIP Channels Bridge
Friday November 2 2007
8:04PM 3 use dial plan passed arg value in C agi code
8:01PM 1 one way RTP using NAT
7:46PM 2 sip show peers in 1.4.13
6:18PM 1 Off-Topic: add GSM codec to X-Lite
5:51PM 3 ztdummy and BackGround
5:38PM 0 steps for installing on Debian Etch
4:44PM 1 RealTime register for iax DID
4:35PM 1 SIP Channels
4:32PM 1 Jitterbuffer issues
3:25PM 1 Asterisk as a SIP to SIP Gateway
2:47PM 2 Route an incoming call by ANI*DNIS
1:46PM 1 res_mysql versus res_odbc
11:37AM 1 AEX800 (TDM800 Express) - not detected
9:26AM 2 asterisk as a gateway
5:21AM 2 zaptel.conf missing
4:05AM 3 __sip_xmit problem
1:50AM 1 Get value from linux terminal to dialplan in Asterisk ?
12:31AM 3 Two PRI setup questions
Thursday November 1 2007
11:56PM 1 OpenSER for Asterisk Load balance
11:39PM 1 Polycom Park Button
11:21PM 1 ring group containing external 10-digit numbers
10:41PM 1 Help: How does one determine the length of an outbound/dialout MESSAGE to be delivered
7:22PM 0 Chanspy attaching to a caller ID entry?
7:02PM 1 AsteriskNOW and TDM800P
6:32PM 3 Outgoing PRI CID?
5:43PM 5 DST
4:57PM 0 libpri & tie line vs trunk
4:00PM 0 Parking speed
3:42PM 1 Help
12:34PM 5 SER/OpenSER as registrar to Asterisk (1500 SIP users)
9:56AM 3 Video Call
9:11AM 1 Autodialing
8:05AM 2 Spam Filter News
8:01AM 2 SER with Asterisk intergration
1:03AM 1 Call Failed
12:57AM 1 Connection astrisk to a RAS (portmaster)
12:49AM 2 hostname in MySQL CDR records