asterisk users - Feb 2008

Friday February 29 2008
10:24PM 0 Skewed RTP timestamps in SIP calls on Asterisk 1.4.18
7:07PM 1 IAX2's JB and DTMF
5:20PM 0 Request for testing: New wctdm24xxp and wcte12xp drivers.
3:17PM 1 Page app, Polycom IP 601, 60 SIP peers, Interesting Issue
3:05PM 1 Detecting SIT (Special Information Tone) on outbound calls
2:47PM 1 Gtalk with asterisk
2:38PM 1
2:33PM 0 load balancing and high availability
2:10PM 10 when we try to add CURL code to file channel.c we get an error - undefined reference to curl_easy_init
12:30PM 1 asterisk queue agent problem
12:12PM 14 which phones to use ??
11:33AM 1 basic installation
11:09AM 2 Cisco 7965g and asterisk
7:30AM 1 Can call in but cannot call out (CHANUNAVAIL): TE410 + Asterisk 1.4.13 + Zaptel 1.4.6 + Libpri 1.4.2
5:39AM 7 Polycom IP600 + PC share same switch port with VLAN
3:56AM 0 I would like to hire someone to automate my asterisk for hosted PBX service
3:39AM 0 1EZphone is only two way browser softphone - SIP Softphones and Citrix ?
2:01AM 7 load balancing
Thursday February 28 2008
11:30PM 5 Problems with removing zaptel
10:18PM 2 Asterisk Voicemail for iPhone
10:04PM 1 GLOBAL function - introduced at what version?
7:08PM 1 Problems with setting up Zaptel
3:54PM 2 quickfix for building zaptel with 2.6.24?
3:30PM 0 Digium certified asterisk professional linkedin group
1:32PM 1 Asterisk monitor() zap channel problem
11:48AM 5 New Interested services to be added for Telephoney Service Provider
11:13AM 4 Friday Feb 29th Leap Year Special wih Aastra
10:49AM 0 OT : OpenSER Summit & Pavilion - 17th to 19th of March, 2008 , San Jose, US
10:21AM 5 Asterisk and Cisco Unity?
7:14AM 1 C Code to connect to Asterisk Manager Interface
2:03AM 4 Unicall mfcr2 testcall issues in mexico outgoing:ok | incoming: fail.
Wednesday February 27 2008
8:55PM 9 Simultaneous Inbound and Outbound calls on analog lines...
7:41PM 3 Attended transfers through a GUI
6:18PM 1 simultaneous ring problem
5:25PM 5 About faxes recived through a PRI and passed to a fax machine connected to a FXS port
5:01PM 2 What causes SIP 486?
4:52PM 0 Problems with Refered Call and Accountcode using siptapi
4:42PM 0 Attended transfers and orginal caller ID
4:36PM 21 Customer complains of noise on line I cannot reproduce.
3:53PM 4 Can AMD detect Service Provider Message.
3:30PM 1 Asterisk as SMSC to GSM-Phones
2:51PM 2 OT But I Would Rather See People Running Asterisk on a "Real" Server than an Emachine
2:41PM 5 SPA3102 registration problem
2:29PM 1 best practice
12:40PM 0 Entering code to restart the machine or reload iax
9:38AM 5 Zap Call deflection on PRI
9:24AM 5 Entering code to restart the machine
8:32AM 2 Danish callerid on a x100p card
8:14AM 0 Configuring modem pools in Asterisk
8:03AM 0 res_config_ldap in asterisk 1.6.0-beta2/4
7:38AM 0 duplicated voicemail messages
7:20AM 2 Call recording problems from queue
Tuesday February 26 2008
11:10PM 0 Anything like SipT38SwitchOver in Asterisk?
10:48PM 1 Configuring modem pools in Asterisk [WAS: Connecting a Rolm CBX to Asterisk via T1?]
10:40PM 0 How is reinvite triggered
8:21PM 1 How do I tell if T.38 was used?
8:10PM 48 Had it with Dell Garbage
7:19PM 3 How can I call cheap to UK cell phones
6:24PM 1 AMD on a SIP trunk...
5:53PM 3 Explain Cause of Error: manager.c: Accept returned -1: Too many open files
5:31PM 4 Sip trunk mystery
2:44PM 0 How to transfer an unanswered call???
1:50PM 3 Asterisk as useragent registered using 2 accounts
1:03PM 1 iax trunking problem
11:32AM 0 CLIR missing in MySQL CDR records
10:44AM 17 [URGENT] Zap channels fail to load
8:59AM 6 chan_ss7 0.10
5:24AM 0 Problem with rxfax
3:41AM 3 Still can't pickup parked call
3:29AM 1 Anybody installed Asterisk in a Virtuozzo VPS system???
1:03AM 7 Parked calls - can't pickup
Monday February 25 2008
10:38PM 6 cannot dial out with latest zaptel and kernel 2.6.24
9:35PM 1 Realtime Queue Status for Agents
8:27PM 0 Considering replacing ATA with linecard.
8:27PM 7 DTMF tone crashes server (Asterisk 1.4.18 with Digium TE120P)
8:12PM 4 Problem with asterisk and aastra phones
6:12PM 5 TE120P echo cancellation problem
2:54PM 0 question on call setup
2:36PM 0 BarCampNYC3
2:22PM 1 VOIP Application on Dating Contact Service
12:20PM 3 DDNS and host: updating when destination IP changes
8:42AM 28 TDM400P dialout problem
Sunday February 24 2008
11:16PM 7 beta4: outgoing call causes Red Alarm on TDM400P
12:14PM 0 Load balancing SIP extensions.
8:01AM 3 DUNDi with two servers
6:25AM 0 Call limits per server with Iax
Saturday February 23 2008
11:29PM 1 Need some dialplan help
8:20PM 2 dundi lookup
3:36PM 2 Suggestions for reliable DID provider forCanada, USA and Europe
12:51PM 1 Fax-to-Email - Legal Issues
12:22PM 5 Suggestions for reliable DID provider for Canada, USA and Europe
9:12AM 0 SIP peers from multiple databases
12:28AM 5 MySQL Voicemail Storage Questions\Errors
Friday February 22 2008
11:03PM 1 Mexico Dids
10:08PM 1 Post call QoS in Asterisk 1.4
8:45PM 1 Polycom 301/501 Keymapping
7:57PM 0 is tos=ef same as tos=0xb8 same as DSCP ef ?
7:57PM 1 Message waiting light on polycom 301 using asterisk 1.4.14
6:58PM 4 AGI / Voicemail Que
6:33PM 8 NOKIA E series Phone for SIP-VOIP calling
2:05PM 2 Interrupt VM and Steal a call.
1:59PM 0 adjusting volume on a wildcard 100XP with zaptel's {t, r}xgain
1:51PM 3 Will this be sufficient for 20+ concurrent calls?
1:43PM 1 [VOIP-Users-Conference] Re: Opinions please: Polycom IP 430 vs 330?
1:17PM 0 Opinions please: Polycom IP 430 vs 330?
1:09PM 0 DUNDi ${NUMBER} variable not defined
12:56PM 6 Linksys SPA-942 Phones
11:42AM 9 load balancing SIP extensions
10:50AM 1 canreinvite question
10:40AM 2 Digium B410P and 8 ports connectivity
10:20AM 0 Using app_sms in South Africa
9:57AM 1 Weird Zaptel sound after anwser calls
7:37AM 0 chan_h323 build failure - `IPTOS_MINCOST' undeclared
7:02AM 0 chan_woomera tries to connect to strange host
6:58AM 0 Friday 22 FEB 08 @ 12 Noon EST ISPBX COGOBLUE
5:52AM 1 Pager ("beeper") Emulation Script
5:08AM 2 (no subject)
3:54AM 1 spandsp/tx_fax/rx_fax frustrations
3:07AM 0 Chan_h323 isn`t dropping calls comming with wrong codecs
12:58AM 1 FW: jabber
Thursday February 21 2008
11:18PM 0 Question regarding AGI
10:45PM 5 Asterisk, Zaptel and the Kernal Compatibility Matrix
10:07PM 1 cid_rewrite.php -- Caller ID Name lookup
9:12PM 3 High CPU load after upgrading to 1.4
8:21PM 0 Asterisk-addons 1.6.0-beta2 Released
8:13PM 0 Asterisk 1.6.0-beta4 Released
7:57PM 0 Asterisk-addons 1.4.6 Released
5:44PM 8 Pattern matching....
5:40PM 3 IVR No sound on other provider
5:37PM 1 Which echo-can for Digium B410P ?
4:13PM 0 Maybe OT: SIP - Missing 407 messages
4:04PM 4 Answered Call marked as "NO ANSWER"
4:00PM 0 SendDTMF not Working - Possible Echo Cancelling Issues
3:19PM 0 HoldMusic Beep
2:53PM 0 Contents of asterisk-users digest
1:42PM 1 USB ISDN interface
1:38PM 5 Allow INVITE for hold to pass through
11:37AM 0 UCS-2 Problem
9:07AM 0 Third Party Call Control - SIP to Iax Gateway
9:07AM 8 Voted most stable and easy to use phone?
8:58AM 5 chan_h323 requirements
5:57AM 3 Can asterisk support 20 user's conference?
4:59AM 2 Asterisk Nagios
2:49AM 1 Multiple Asterisk Servers. One Conference
2:31AM 2 Converence/Meetme with Manager API
2:30AM 22 How to get a clean, basic configuration?
12:53AM 0 How to Configure 1.4.17 to Store CDR's in PostgreSQL
Wednesday February 20 2008
11:48PM 0 Call drops to fast busy
10:52PM 12 Coppercom and Asterisk
10:31PM 0 Southern Alberta Canada * users.
9:26PM 41 Best ATA. Period.
8:39PM 0 debugging stuck led lights
7:47PM 4 Polycom Key Assignment
7:44PM 7 ata device but for a soundcard
7:30PM 1 Include in asterisk realtime
7:00PM 1 Receiving double DTMF
6:39PM 1 manager ignore my settings
6:12PM 1 IAX: No outgoing audio for 10 seconds
6:08PM 0 Strange NewCallerIDEvent after channel are linked
3:17PM 3 Skype Users
1:50PM 4 GXP-2020 Transfer Key
12:15PM 18 problem transferring calls some of the times
10:07AM 0 Unable to create channel of type 'Zap' with ecmg2 and kernel 2.6.23
8:28AM 3 Dial+Macro and Queue
7:14AM 1 OT - DECT-GAP Handsets with Polycom-Kirk 600/3 base station
5:31AM 1 Need to Connect offices in Dubai and Pakistan
3:50AM 5 which codec over iax => pstn
3:09AM 0 GS/* phonebook + Web-BLF
12:27AM 11 Sangoma FXO EC vs Rhino FXO EC
Tuesday February 19 2008
11:49PM 0 More on Broadvoice w/Asterisk (1.4.18)
11:13PM 0 Restricting registration for peer 'iaxmodem0' to60 seconds
10:49PM 2 Restricting registration for peer 'iaxmodem0' to 60 seconds
9:07PM 0 Detecting the digits
6:16PM 1 Zaptel version, and compatibility matrix
4:33PM 1 Connecting a UMTS module via USB to asterisk
4:05PM 2 MeetMe Admin Functions
3:59PM 0 Extension Logic Help
3:44PM 5 asterisk config file online editor
2:59PM 0 two lines written in CDR for each failed call in asterisk 1.4
2:55PM 0 ASTERISK C File
12:46PM 0 IAX registration problem
12:00PM 1 SIP Request: OPTIONS
9:59AM 1 A problem about digium TE220B
9:54AM 0 jabber
9:43AM 0 [Copfilter] Copy of quarantined email - *** SPAM *** [6.0/6.0] IAX2 client asked to authenticate against wrong
6:53AM 4 No compatible codecs!
2:43AM 0 Request for testing: Distributed device state
1:45AM 5 Asterisk 1.4 vs 1.6
Monday February 18 2008
11:52PM 2 ztscan ports = zaptel channels ??
11:10PM 0 Zaptel 1.2.24 and 1.4.9 Released
11:06PM 0 Asterisk 1.6.0-beta3 Released
10:18PM 1 Asterisk: how to limit h323 connections.
10:17PM 7 Attatch monitor recording to a voicemail
8:00PM 1 Avaya 4610sw
7:14PM 7 Cisco SIP Gateway
6:21PM 0 Pulling a variable from a shell script into Asterisk - backticks?
5:15PM 0 logging the estimated RTT using SIP
4:04PM 0 Problems with TE120, Kernel BUG
3:53PM 1 realtime table customization to track iax registrations
3:08PM 2 SPA-3000 caller ID and KPN
2:28PM 0 Changing the automon output filename
1:26PM 7 IAXModem - NDID=s
1:06PM 0 Contents of asterisk-users submissions
10:37AM 0 Please reply..Not able to call H323 using SIP client
10:24AM 10 SiP call generator
10:15AM 2 ForkCdr in 1.4.*
8:37AM 0 from address modification
8:33AM 0 Vancouver - Asterisk Event Feb 18 (Monday)
7:36AM 3 mfcr2 stuck
7:36AM 0 IAX2 client asked to authenticate against wrong peer (username)
6:42AM 0 Not able to call H.323 client by SIP client
6:11AM 1 PRI dialplan/prefix
5:35AM 9 ISDN2 facility code...
4:30AM 2 Failure of Sending Voicemail As an attachment in E-mail
2:17AM 8 Zaptel 1.4.8 breaks tor2 support on CentOS 5.1? (kernel panic)
Sunday February 17 2008
7:50PM 1 app_voicemail - Failed to open file ../tmp/xxxxx.WAV
6:22PM 0 Able to receive calls on my fxo port and not able to do calls via it
12:20PM 19 Music on hold
12:16PM 2 Asterisk reltime mode with Postgresql
9:40AM 1 IAX2 trunks unreliable becoming UNREACHABLE aftera time
12:42AM 3 Asterisk H.248 Support
Saturday February 16 2008
10:22PM 1 Zaptel and Asterisk compilation
5:46PM 4 T1 "access layer" used Cisco or new Digium
5:07PM 2 zaptel: modpost section mismatch ?
3:28PM 6 Fritz! Card/CAPI Help.
1:56AM 0 Digium stopped TDM400P production: alternatives?? ?In-Reply-To: <> ?References: <> <> ? <> ? <47B58F8B
1:44AM 0 arris tm502g cablemodem FXS ports and zaptel 1.4.8
Friday February 15 2008
8:49PM 11 Connecting a Rolm CBX to Asterisk via T1?
6:40PM 0 Zaptel compilation problems.
11:58AM 3 1.4 and IAX Trunks ...
11:29AM 25 Digium stopped TDM400P production: alternatives??
9:18AM 0 G729 transcoding and "clicking"
8:54AM 3 Communication between two asterisk server
8:35AM 0 Question about DIALSTATUS NOANSWER
6:02AM 0 How to check if a local channel member of a queue?
2:18AM 0 patch which makes Asterisk-Addons 1.4.5 work when codec negotiation patch applied to asterisk
2:08AM 2 HPEC
1:15AM 3 57iCT BLF problem
1:04AM 1 DialPlan help with Analog Fax Machine
Thursday February 14 2008
11:20PM 1 gtalk and dtmf
9:51PM 0 ExtenSpy strange behavior on Asterisk 1.4.18
7:26PM 0 Emagen (a Telrad VM solution) -- any way to replace with *?
7:04PM 6 Variable setting in AMI Originate
5:41PM 3 X100P Burnouts
5:30PM 5 SNMP monitoring
5:08PM 0 IAX load balancing
4:22PM 2 Pass arguments from extensions.conf
4:05PM 0 translating iax2 register into sip register
3:30PM 9 Monitor Asterisk
12:17PM 2 Error checking asterisk method - suggestions?
11:12AM 1 Ser, asterisk and ip2ipgw
9:39AM 11 UK -999 dialing issue
5:30AM 0 Snom Provisioning Tool Release
12:49AM 2 Touch monitor file name format
Wednesday February 13 2008
10:42PM 5 multiple host in 1 context on sip.conf
10:21PM 3 SIP over TCP
8:31PM 4 Asterisk Manager and Visual Basic
6:45PM 4 MWI problem with Siemens Gigaset S675 IP
6:03PM 10 ISDN PRIs and taking a server down for maintenance - blocking issue
4:54PM 5 Asterisk and fax
4:40PM 6 Analog DID
3:59PM 1 FOSDEM in Brussells - Feb 23-24
3:58PM 8 GXP2000 and asterisk 1.0.9
2:36PM 0 Digium's Exceptional Satisfaction Program
2:22PM 4 What is a "secure call"?
2:12PM 6 UK issue - Asterisk dialling 999... sort of
1:46PM 0 Wanted: VoIP Engineer for Switerland
12:46PM 6 [Linux/Python 2.4.2] Forking Python doesn't work
12:35PM 5 Telephone line signaling configuration in Egypt for FXO ports
12:30PM 5 Attendant phone
9:33AM 4 urgent-channels
9:15AM 1 Hardware needed
8:10AM 0 Friday Feb 15th @ 12 Noon EST: VoIP Users Conference welcomes Lumenvox
7:58AM 5 PCI32 and PCI-X compatibility
5:06AM 0 Netgear TA612V line 2 and asterisk
2:49AM 3 How to soft hangup all channels at a time .
1:58AM 21 restart asterisk daily
12:39AM 0 OT: 3rd party SMS service?
Tuesday February 12 2008
10:23PM 0 play greeting from odbc voicemail
4:04PM 0 Asterisk bridging timeout when calling out with SIP phone
3:53PM 19 LCR in Asterisk
2:56PM 1 Zhone Channel Bank
2:33PM 0 which mobile compatible with asterisk
9:11AM 0 * SIP dial out with multiple sip users
3:50AM 1 chan_ooh323 patches compatible with codec negotiation patch applied to asterisk 1.4.17
2:18AM 4 Nortel 1140E
Monday February 11 2008
11:39PM 0 OT: Recommendation for EDGE/3GPP SIP phone?
11:00PM 1 SIP Bad request protocol Packet on Asterisk 1.4.18
10:24PM 0 SPA3000 + asterisk +call waiting
10:11PM 0 Configuring Polycom SP300 -- weird problem
9:52PM 11 Grandstream GXP2000 Loses Connectivity
7:44PM 7 Automatically start after restart
7:22PM 4 Automon reliability issue
6:32PM 2 G729 without licence
4:36PM 0 Semi-OT: bluetooth conference phone?
4:20PM 3 message: !! Got Busy in Connected State !?!
3:25PM 3 differences
2:16PM 3 Asterisk as a softswith for a small ISP
1:10PM 1 Iax2 provisioning and Asterisk
11:41AM 0 asterisk-users Digest, Vol 43, Issue 30
10:19AM 0 TE412P on Intel S5000XVN and S5000VSA Motherboards
8:22AM 3 Single * multiple offices
6:24AM 13 Realtime SIP peers - reloading cached info
5:56AM 1 Asterisk Meetme & its Admin
4:45AM 0 asterisk, asterisk-addons, ooh323, codec negotiation patch in FreeBSD
12:24AM 2 how to create a standalone voicemail server
Sunday February 10 2008
11:41PM 1 Generate anonymous SIP Calls
11:18PM 1 SIP proxy/registration for *
7:33PM 7 IAX2 trunks unreliable becoming UNREACHABLE after a time
6:29PM 3 usability Testing Costa Rica, SanJose asterisk PBX / dsl/cable service
1:55PM 3 Still dropped calls :(
7:44AM 11 Disappearing B-Channels
4:20AM 0 Carrier SIP resource?
1:01AM 1 HP proliant and hpasm
Saturday February 9 2008
6:44PM 2 [asterisk-dev] Monitor Asterisk using C
5:09PM 3 Cisco phone 79xx get database information
5:06PM 6 BLF and Asterisk 1.6.0b2
11:11AM 1 SIP user registration and Asterisk Realtime
9:04AM 3 voicemail to non-default context user does not work
7:21AM 3 Dialing SIP server user extension... Dial string issue...
4:54AM 15 oneway audio with asterisk behind cisco pix 506
12:49AM 0 [asterisk-biz]SIP to SIP professional community
12:21AM 7 Sending a message from inside voicemailmain.
Friday February 8 2008
10:32PM 0 Rejected calls to Sylantro server
7:03PM 0 Dealipedia
6:32PM 4 Monitor Asterisk using C
4:46PM 0 GS/* phonebook
4:27PM 0 Interoperability between TE412P and Eurotech PRI E1 GSM & CDMA Gateway
3:58PM 0 canreinvite option - gona have problems?
3:12PM 3 Question about Asterisk versions (newbie)
2:52PM 2 Upgrade 1.2 -> 1.4 voice files
2:27PM 1 Permission denied when obtaining Status
12:24PM 2 Domainname for outgoing uri-dialing
10:11AM 1 Cosini iAN7s
10:05AM 7 Transferring a call received by an agent in a queue
7:53AM 1 (no subject)
7:41AM 0 VoIP Users Conference Call Today Friday @ 12 Noon EST
6:51AM 1 Asterisk queue not play muscinhold or hangup
1:54AM 0 Transcoded G.722 calls unintelligible with recent SVN head
Thursday February 7 2008
10:11PM 2 Asking for recommendations on Asterisk Boxes or Appliances
8:18PM 0 Asterisk 1.4.18 Released
7:37PM 2 Snom 300 MWI
6:24PM 4 Asterisk as XMPP component. How to use it ?
6:19PM 0 Asterisk trunk/1.6 and nvfaxdetect
5:40PM 5 Snom 300 Echo
5:11PM 6 Asterisk G722
3:00PM 2 How to balance traffic between 2 gateways ?
2:05PM 10 Goto in Realtime extensions
12:39PM 1 SIP / RTCP statistics logging
12:32PM 1 FW: transcoder
9:22AM 6 Two Leg CDR
9:17AM 3 Preventing IAX frame concatenation
3:33AM 0 New deployment questions
3:04AM 1 OT: POTS telephone like the SPA-942?
2:45AM 3 Need good voicemail documentation
1:48AM 0 Asterisk and Avaya phone system
12:18AM 3 Matching "+" characters in dial plan
Wednesday February 6 2008
11:03PM 0 Post Call QoS....?
9:53PM 1 TE412P and Delll PowerEdge 2900
9:09PM 2 AGI Process Count (HOWTO?)
7:54PM 1 FXO modules and polarity reverse
7:46PM 0 Need a dial rule to match and replace a number.
6:43PM 3 Polycom BLF / Speed Dial
6:22PM 1 TDM400P phone won't ring
5:00PM 4 [OT] ISDN 30 (PRI) service in the Netherlands
12:50PM 0 Problem forwarding a call with an AGI script
11:05AM 0 Directing SIP/RTP sessions b/w UA
9:28AM 0 How to register h323 users?
12:50AM 13 R2 with Alestra in Mexico...
12:39AM 4 Gemeinschaft released
Tuesday February 5 2008
11:38PM 3 Can't delete voicemail messages
8:32PM 6 Cannot hear voice through SIP Phone from one side
8:10PM 8 How to hookup to cell phone for outbound calls?
6:29PM 0 What causes this?
6:00PM 0 Post Call QoS?
4:53PM 0 meetme with ztxen - WARNING[10725]: app_meetme.c:772 build_conf: Unable to open pseudo device
4:34PM 1 Telephony Migration Hardware
3:22PM 11 is encrypted iax safe and secure?
1:52PM 17 wireless VOIP phone recommendations?
9:30AM 1 Mistake in the wiki's description of cmd Pickup() ?
8:36AM 1 Can't dial out from SIP to CAPI
8:07AM 11 [Softphones] ZoIPer vs. XLite?
7:35AM 0 Asterisk does not handle INVITE authentication by Proxy
5:54AM 2 basic server specs
5:31AM 14 External MWI question for Asterisk
Monday February 4 2008
6:27PM 2 GROUP_COUNT and Attended transfer
5:22PM 6 transcoder
4:15PM 3 Asterisk with 2 NIC cards
3:16PM 0 error while loading zaptel or ztdummy module under kernel 2.6.18-6-xen-amd64 - no sound in asterisk
2:25PM 4 one CDR instead of multiple CDR
1:53PM 0 Asterisk mishandling user busy isdn releases
11:14AM 3 OT POlycom question
10:22AM 0 AGENTDUMP lines in queue.log????
9:08AM 3 Losing CALLERID{dnid}
8:44AM 2 PRI with 20 channels
8:37AM 1 asterisk-gui installation hangs
8:25AM 1 Got SUBSCRIBE for extension ... but there is no hint for that extension.
8:18AM 0 Problem picking up a call with PickUpChan or PickUp [SOLVED]
6:14AM 2 Problem with IRQ Share
2:50AM 10 AGI: Not getting answers from get_data in a call-file call
Sunday February 3 2008
11:56PM 4 Console/dsp, makes me sound like a Dalek
8:14PM 8 switch QOS requirements
8:10PM 3 Test
5:09PM 1 Telco MWI Detection on TDM400 Interface?
1:25AM 3 Multiple SIP phones behind a Linksys firewall
Saturday February 2 2008
9:09PM 0 SIP: IP in the VIA-Header
8:06PM 5 app_valetparking.c anyone using it on 1.4?
8:46AM 12 Zaptel timer on Intel Dual Core servers
8:45AM 2 ATA with pulse dialing support over FXS
1:53AM 2 Polycom - Buddy Watch not a choice when adding Speed Dial
1:01AM 3 Echo() app doesn't work
12:20AM 0 IAX Registraion Refresh
Friday February 1 2008
10:57PM 15 "Real" API for Perl?
10:56PM 2 X-Lite Softphone keeps de-registering?
10:30PM 0 Trying to make SIP calls through Asterisk with anonymous connection
8:32PM 3 QueueMember event/LastCall Variable - Format?
8:31PM 5 It's about time! -- Digium PCI-Express Cards
7:47PM 0 Bypassing a Auth on Invite or Forbiden?
6:57PM 2 Asterisk 1.4.17 and Teliax DTMF
5:34PM 2 Asterisk 1.6 - Problems with SIP/REFER
5:29PM 1 Astersik Transcoder support
4:02PM 1 Problem picking up a call with PickUpChan or PickUp (asterisk-users Digest, Vol 43, Issue 1)
3:35PM 0 Codec preference selection, codec negotiation
3:33PM 3 Remote Call Center Agents and Asterisk?
3:12PM 5 SIP Softphones and Citrix ?
2:16PM 17 Enterprise or Fedora?
1:37PM 3 BRI card with PCI-E interface
1:24PM 2 Unicall
1:08PM 1 play promt at the same time to calling and callee
9:59AM 0 call log notice messages
9:39AM 2 meetme music on hold - when only conference member problem
6:54AM 1 Asterisk-Addons install success-Could not find ooh323.conf
4:55AM 3 h priority problem
3:40AM 2 realtime warning