| Thursday June 30 2005 |
| Time | Replies | Subject |
|---|
| 11:37PM |
0 |
Got this error after my installation when i do ztcfg -vv |
| 11:35PM |
1 |
Caller ID problem.. |
| 11:23PM |
1 |
Outbound answer on TDM400P |
| 11:23PM |
1 |
IAX DTMF Problem... |
| 10:58PM |
0 |
Flash Zap Channel |
| 9:55PM |
0 |
Re: Asterisk-Users Digest, Vol 11, Issue 181 |
| 8:58PM |
1 |
No BUSY on PRI |
| 6:31PM |
0 |
chan_oh323.c not compiling (cid not found) |
| 4:56PM |
3 |
Linux Firewall Question |
| 3:30PM |
3 |
Computer to use |
| 3:21PM |
0 |
Cell Phones reporting internation calls |
| 2:53PM |
2 |
New Setup with Analog Phone lines |
| 1:54PM |
1 |
Flash and zap and # key |
| 1:34PM |
5 |
wi-fi phone advice |
| 1:20PM |
2 |
Dial Option A(file.gsm) |
| 11:55AM |
0 |
callprogress and queues |
| 11:42AM |
0 |
Asterisk & mp3 playback while dialing |
| 11:37AM |
1 |
spandsp fax out fails |
| 11:25AM |
7 |
passing through MWI info from SBC |
| 11:09AM |
2 |
ser --> sip.conf --->extensions.conf, variable context |
| 9:55AM |
5 |
Failover question |
| 9:48AM |
3 |
Trying to do very simple Zaptel Config. NO LUCK! |
| 9:43AM |
1 |
Pickup pin |
| 9:22AM |
0 |
Strange dropped calls |
| 9:22AM |
3 |
GUI that supports virtual PBX's/users |
| 8:27AM |
0 |
Re: Asterisk-Users Digest, Vol 11, Issue 202 |
| 7:50AM |
0 |
Audiocodes MP-1xx cheat sheet |
| 7:42AM |
2 |
Asterisk failover solution |
| 7:22AM |
5 |
Logrotate |
| 7:22AM |
1 |
Cisco Voip Question |
| 6:54AM |
4 |
[Fwd: Asterisk Balancing solution] |
| 6:34AM |
0 |
Daily Asterisk News |
| 6:05AM |
2 |
[Asterisk-Dev] Developing an Application in Asterisk |
| 5:43AM |
1 |
Developing an application in Asterisk. |
| 4:59AM |
1 |
[Asterisk-Dev] C Code of Asterisk |
| 3:43AM |
1 |
Master.csv and MYSQL |
| 3:32AM |
1 |
problems in dialing in routes patterns |
| 2:55AM |
7 |
Voicemail => SMS |
| 2:41AM |
1 |
Fw: Multiple Quad Bri card |
| 2:38AM |
4 |
Getting FOP working with ICD? |
| 12:57AM |
3 |
Resolving groupcalls |
| 12:51AM |
1 |
Do any ITSPs support Speex? |
| 12:29AM |
3 |
AMP - recording call |
| 12:05AM |
3 |
R: Music oh hold |
| 12:01AM |
0 |
Sipura 3k answers then immediate busy signal |
| |
| Wednesday June 29 2005 |
| Time | Replies | Subject |
|---|
| 11:35PM |
1 |
GnuGK and Asterisk |
| 10:28PM |
1 |
Can't build cdr_addon_mysql. |
| 10:28PM |
0 |
Calls Dropping |
| 10:01PM |
1 |
Asterisk/SER/Call Manager |
| 10:00PM |
0 |
failed IAX, SIP registration - email notification |
| 8:55PM |
2 |
X100P connected as extension to Panasonic 616 EASA-PHONE |
| 8:50PM |
2 |
Problems with zaptel and voice prompts/voicemail |
| 8:36PM |
4 |
Quality of provider: VocTel |
| 8:32PM |
0 |
RE: Asterisk-Users Digest, Vol 11, Issue 198 |
| 6:42PM |
0 |
ASTCC Issues - Resolved |
| 5:41PM |
11 |
Asterisk@Home Ver 1.2 Whats new? |
| 5:33PM |
1 |
Kind of Computer to use |
| 5:19PM |
0 |
Access asterisk features from analog phone during call |
| 5:08PM |
0 |
Server hanging |
| 4:33PM |
3 |
UK SIP Provider |
| 3:40PM |
2 |
New Asterisk documentation |
| 3:25PM |
1 |
Problems connecting to and from my Asterisk server :( |
| 2:25PM |
0 |
Parking Position |
| 1:56PM |
3 |
hidecallerid on analog line |
| 1:16PM |
5 |
Extension Matching. |
| 12:53PM |
2 |
Multiple Timezones with Asterisk |
| 12:47PM |
1 |
Welcome |
| 12:39PM |
1 |
AMP or Asterisk |
| 12:27PM |
1 |
Dial ZAP Problem |
| 12:23PM |
0 |
PRI: XXX Missing handling for mandatory IE 12 (cs0, Connected Number) XXX |
| 11:48AM |
2 |
Polycom SoundPoint 501 Problem |
| 10:35AM |
4 |
Music oh hold |
| 10:26AM |
0 |
(no subject) |
| 10:21AM |
1 |
Sangoma and quad card hang up problems |
| 9:47AM |
0 |
dtmfmode=inband still broken in *-1.0.7 |
| 8:33AM |
10 |
Setting Caller ID after Dial |
| 8:13AM |
1 |
OrderlyQ installations? |
| 7:49AM |
1 |
Teliax Problems |
| 7:40AM |
5 |
Problems with OR Logic in the GotoIf Statement |
| 7:28AM |
2 |
Play an announcement to the CALLING party |
| 7:12AM |
2 |
timeout on incoming PRI call |
| 7:11AM |
0 |
How to fetch a call not in the same callgroup |
| 6:17AM |
1 |
Machine Sizing |
| 6:13AM |
4 |
PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 6:02AM |
2 |
Asterisk LAMP Developer |
| 4:54AM |
1 |
Equipment for small office setup |
| 4:40AM |
0 |
ast_rtp_read: Unknown RTP codec 100 received21 when receiving fax |
| 4:40AM |
2 |
Recommend against Teliax as primary ITSP |
| 4:18AM |
1 |
App_conference in dial plan? |
| 4:14AM |
1 |
CAPI and Caller ID name not showing. |
| 2:31AM |
1 |
GSM Hunting |
| 2:15AM |
0 |
chan_capi-cm-0.5.3 fixup release |
| 1:48AM |
0 |
CallerID Bug? |
| |
| Tuesday June 28 2005 |
| Time | Replies | Subject |
|---|
| 11:20PM |
1 |
Hop-On WIFI Phone MSRP $40 |
| 10:55PM |
1 |
Fw: Shoutcast Music On Hold problems? |
| 10:39PM |
1 |
audiocodes |
| 9:46PM |
2 |
AMP/A@H (asterisk at home) custom incoming routing |
| 8:51PM |
4 |
Anyone using SipP to produce RTP load? |
| 7:43PM |
1 |
Red Hat Enterprise 3.0 issue |
| 7:08PM |
0 |
manager api call number, pause, dial ext |
| 6:48PM |
0 |
ASDI Programming through an ATA/SIP device? |
| 4:32PM |
2 |
SIP Phone Config Generator |
| 4:14PM |
1 |
Net2Phone equipment and different VOIP providers |
| 3:45PM |
1 |
TDM04B Echo on Only One Channel |
| 3:42PM |
2 |
Asterisk RSS list feeder ready |
| 3:04PM |
0 |
BRIstuff/OctoBRI problem: Ring requested on unconfigured channel 255/255 span 5 |
| 2:49PM |
3 |
Asterisk with Lucent TNT echo |
| 2:15PM |
0 |
Inter-Tel 8662 configuration problem. |
| 2:04PM |
2 |
Trying to get *8 call pickup to work |
| 1:49PM |
0 |
Dial application timeout |
| 12:02PM |
1 |
VoipJet TOS (was Teliax and also LiveVoip) |
| 12:00PM |
1 |
Linksys WRT54GP2-NA settings for performance and low bandwidth? |
| 11:53AM |
4 |
Revision I Board TDM04b |
| 11:52AM |
0 |
Mitel SX2000 Integration |
| 11:36AM |
2 |
Asterisk Realtime and ODBC |
| 11:34AM |
1 |
enabling stun on asterisk? |
| 11:24AM |
1 |
Voicemail max time length |
| 10:53AM |
1 |
Voice Mail hangup on not messages |
| 10:37AM |
1 |
initial setup: problem |
| 10:07AM |
0 |
CPU load about at max when it should be idle. |
| 10:06AM |
1 |
problems with chan_capi 0.3.5 , divactrl, eicon diva server, and kernel 2.6.10/2.6.12 |
| 10:02AM |
1 |
Correction to Janghanns BRI problem |
| 9:55AM |
0 |
Re: Asterisk-Users Digest, Vol 11, Issue 181 |
| 9:44AM |
0 |
Re: Asterisk-Users Digest, Vol 11, Issue 183 |
| 9:22AM |
2 |
Junghanns 4 port BRI problem |
| 9:21AM |
1 |
This weeks Developer meeting |
| 9:19AM |
0 |
Speech driven crm apps |
| 9:16AM |
0 |
Re: teliax [Was: LiveVoip is Bankrupt] |
| 9:14AM |
1 |
Unable to connect to remote asterisk |
| 9:03AM |
0 |
Re: Asterisk-Users Digest, Vol 11, Issue 183 |
| 9:00AM |
0 |
Asterisk dies with Meetme |
| 8:35AM |
1 |
Hangup detection on Panasonic KXTD816 |
| 8:29AM |
1 |
TDM400 |
| 8:25AM |
1 |
list Searchability |
| 8:16AM |
2 |
MeetMe application in Asterisk V1.07 |
| 7:59AM |
4 |
How do you handle NAT? |
| 7:43AM |
1 |
ClueCon, Vote? |
| 7:35AM |
1 |
Re: teliax [Was: LiveVoip is Bankrupt] |
| 7:13AM |
0 |
help, switch off NOTICE in console |
| 6:53AM |
0 |
Asterisk & SpanDSP -> problems by sending a fax |
| 6:16AM |
4 |
Using Conferencing and Meetme |
| 5:28AM |
2 |
Using asterisk as Quality Monitoring Platform? |
| 2:35AM |
0 |
Spinlock with ZAPTEL |
| 2:35AM |
0 |
GSM/PSTN Gateway function of DIAX - feedback request |
| 2:04AM |
1 |
simultaneus calls? |
| 1:58AM |
0 |
cheap HFC card on Bristuff vs cheap HFC card oni4l vs Fritz ISDN BRI card on CAPI |
| 1:56AM |
1 |
pbx_extension_helper: No application 'agi' |
| 1:46AM |
0 |
RE: [Serusers] *** SER - Asterisk |
| 1:29AM |
1 |
AVM CAPI INSTALLATION |
| 12:59AM |
1 |
HOW TO WRITE ROUTE PATTERNS DIALPLAN |
| 12:01AM |
1 |
cheap HFC card on Bristuff vs cheap HFC card on i4l vs Fritz ISDN BRI card on CAPI |
| |
| Monday June 27 2005 |
| Time | Replies | Subject |
|---|
| 10:51PM |
3 |
Shoutcast Music On Hold problems? |
| 10:14PM |
0 |
Can anyone guide me regarding h323.cong ??? |
| 9:20PM |
0 |
Fw: shoutcast mp3 music onhold with amp portal? |
| 9:07PM |
0 |
How to get the outbound data of agent in queue |
| 8:19PM |
2 |
Eicon equipment, BRI Server or PRI? |
| 8:13PM |
1 |
Newbie Confusion on Call Forward and DBput/DBdel |
| 7:33PM |
0 |
Asterisk ended with exit status 1 |
| 6:49PM |
0 |
Disable record busy greeting option in voicemail |
| 6:31PM |
2 |
is teliax down? |
| 5:36PM |
1 |
SixTel? |
| 4:24PM |
1 |
Level 3 SIP <--> asterisk |
| 3:17PM |
0 |
Dialogic D/300pci-E1 card |
| 3:17PM |
0 |
Possible bug in meetme when hangup |
| 2:58PM |
0 |
Just let whiners whine... Please? |
| 2:17PM |
0 |
Bridging and unbridging channels |
| 2:07PM |
0 |
RE: [Serusers] *** SER - Asterisk |
| 1:55PM |
4 |
LiveVoip is Bankrupt - Why this thread |
| 1:49PM |
0 |
how to set agent to busy when agent makes a outgoing call? |
| 1:34PM |
1 |
VoIP provider in Switzerland |
| 1:19PM |
1 |
Polycom & VPN trouble |
| 1:09PM |
0 |
Snom 220 Active Call Lights |
| 1:04PM |
3 |
AGI "say number" but in french |
| 12:42PM |
8 |
OT: Good soft-phone on Linux |
| 12:17PM |
0 |
H.323 (Asterisk@Home) |
| 11:50AM |
1 |
RE: [Serusers] *** SER - Asterisk |
| 11:42AM |
2 |
PSTN IAX Connections / Line Banks |
| 11:00AM |
0 |
Re: teliax [Was: LiveVoip is Bankrupt] |
| 10:27AM |
1 |
Re: teliax [Was: LiveVoip is Bankrupt] |
| 9:51AM |
3 |
Bad Bad Performance; Max 20 Calls on Quad Proc? |
| 9:35AM |
0 |
Re: teliax [Was: LiveVoip is Bankrupt] |
| 9:34AM |
1 |
Passing called number in SIP |
| 9:31AM |
1 |
announced transfer |
| 9:11AM |
2 |
DID in Western Canada |
| 8:50AM |
0 |
dropcount |
| 8:43AM |
1 |
Asterisk and conference bridging... |
| 8:06AM |
0 |
Re: FXO as modem (was: * fax reliability between ISDN PRI andFXS ports) |
| 7:31AM |
1 |
LogWatch for Asterisk |
| 7:31AM |
0 |
???? WARNING[20313]: channel.c:531 ast_channel_walk_locked ???? |
| 6:52AM |
2 |
Accessing SIP username from AGI script |
| 6:16AM |
2 |
Comedian Mail User Setup Prompts |
| 6:00AM |
6 |
TDM card and voicemail volume |
| 5:22AM |
1 |
TE100P |
| 5:22AM |
0 |
Howto answer, hold and transfer a incoming call? |
| 4:46AM |
1 |
Native MoH patch for 1.0.8? |
| 4:46AM |
0 |
Failover Design |
| 4:34AM |
0 |
facing troubles with routes patterns dialplan |
| 3:35AM |
3 |
Fw: linksys rt31p2 test case |
| 3:15AM |
2 |
R: zaphfc: empty HDLC frame or bad CRC received |
| 2:34AM |
0 |
chan_capi-cm-0.5.2 fixup release |
| 2:22AM |
1 |
Strange behaviour with lost internet connection |
| 2:17AM |
1 |
MWI |
| 12:50AM |
0 |
bristuff-0.2.0-RC8h does not compile |
| |
| Sunday June 26 2005 |
| Time | Replies | Subject |
|---|
| 11:30PM |
0 |
No Sound at all |
| 11:18PM |
0 |
newbie here ...... regarding h323.conf |
| 9:34PM |
2 |
FXO as modem (was: * fax reliability between ISDN PRI and FXS ports) |
| 7:34PM |
0 |
TDM400P-04B fails after reboot |
| 5:53PM |
0 |
APP - ValetParking on CVS-HEAD -- instructions on its use, anyone? |
| 5:51PM |
0 |
Missing first second of voice on outgoing SIP/IAX calls |
| 2:25PM |
1 |
Changing Caller ID |
| 12:57PM |
0 |
Prepaid for mysql and simple auth |
| 12:15PM |
3 |
TDM400P (TDM02B) ceased to work... |
| 9:44AM |
0 |
need help for configuring voicemail with db |
| 9:12AM |
1 |
DID in 513 Cincinnati |
| 8:59AM |
1 |
CDR: source completed with sip domain |
| 8:40AM |
3 |
Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE! |
| 6:28AM |
0 |
failure in writing in pattern (routes) |
| 6:01AM |
1 |
help regarding h323.conf |
| 5:53AM |
0 |
Building DUNDi? |
| 5:33AM |
0 |
Bug in Mailman version 2.1.5 |
| 4:59AM |
2 |
Asterisk RealTime Voicemail |
| 4:31AM |
3 |
cdr and billing |
| 12:38AM |
30 |
LiveVoip is Bankrupt |
| |
| Saturday June 25 2005 |
| Time | Replies | Subject |
|---|
| 10:33PM |
0 |
IPSwitchBoard version 0.120 released |
| 9:27PM |
2 |
* fax reliability between ISDN PRI and FXS ports |
| 9:11PM |
0 |
Everyone is busy/congested at this time |
| 8:06PM |
0 |
iaxy device |
| 6:52PM |
0 |
How to bridge 2 calls together |
| 4:24PM |
2 |
iaxy over the public cloud |
| 12:47PM |
6 |
ASTCC not billing |
| 10:33AM |
1 |
Looking for link.exe to compile G729 codec |
| 10:01AM |
0 |
SIP registration fails with realtime |
| 9:38AM |
0 |
help for odbc storage |
| 8:58AM |
1 |
callerid in forwarded call |
| 8:11AM |
3 |
* 1.0.8: no more reacting to callerid? |
| 7:54AM |
4 |
Asterisk and Cisco CallManager Integration |
| 5:47AM |
1 |
isdn channels busy |
| 5:18AM |
0 |
OH323, RxFax and codecs |
| 2:58AM |
0 |
FW: ZAP to SIP Dial Plan |
| |
| Friday June 24 2005 |
| Time | Replies | Subject |
|---|
| 7:35PM |
1 |
Asterisk with dual WAN router |
| 7:14PM |
0 |
H323 with Asterisk |
| 7:11PM |
0 |
Running 6 copies of Asterisk on my machine |
| 5:26PM |
4 |
UTStarcom F1000 WiFi IP Phone Review |
| 2:33PM |
0 |
Help installing PyAsterisk |
| 2:26PM |
1 |
Qualify Frequency |
| 2:25PM |
0 |
Exposing Zap Channels on Server A to be UsedByServer B |
| 1:32PM |
0 |
UTStarCom F1000 SIP configuration |
| 1:31PM |
1 |
Exposing Zap Channels on Server A to be Used ByServer B |
| 1:17PM |
4 |
Tellabs Echo Canceller |
| 1:14PM |
0 |
Playtones volume control? |
| 12:47PM |
0 |
Help instaling PyAsterisk |
| 11:48AM |
0 |
New astGUIclient version released 1.1.4 |
| 11:05AM |
2 |
Set global variables without extension.. |
| 10:39AM |
1 |
Unable to open pseudo channel for timing... Sound may be choppy. |
| 9:33AM |
0 |
Distinctive Ring for Agents (Was: Re: Asterisk 1.0.8) |
| 9:07AM |
2 |
Exposing Zap Channels on Server A to be Used By Server B |
| 9:06AM |
0 |
format_base64.c released on pbxfreeware.org |
| 7:32AM |
0 |
Extensions Puzzle: Contexts Confligting with each other. |
| 6:44AM |
0 |
wcte11xp hardlock problem |
| 6:36AM |
2 |
RTP session between two end users |
| 6:21AM |
3 |
SendText |
| 6:00AM |
1 |
Asterisk server with remote monitoringcapabilities |
| 5:37AM |
1 |
Errors on SuSE 9.3 default install. |
| 5:24AM |
1 |
Dial peer preference |
| 4:55AM |
0 |
How to setup two Asterisk boxes - keeping theregistration |
| 3:39AM |
1 |
Whole configuration for SMS |
| 3:07AM |
1 |
BRIstuff/QuadBRI problem: Ring requested on unconfigured channel 255/255 span 5 |
| |
| Thursday June 23 2005 |
| Time | Replies | Subject |
|---|
| 11:34PM |
0 |
asterisk security issue |
| 11:28PM |
1 |
realtime sip confusion |
| 11:08PM |
0 |
Requirement for internal calls |
| 10:36PM |
1 |
More IP address in bindaddr directive |
| 10:05PM |
0 |
Voicemail recording cutoff when silent for 1 second |
| 9:08PM |
0 |
SPA841 Utterly Horrible, are there any good stun hardphones? |
| 6:59PM |
1 |
HDLC abort 6 error |
| 6:55PM |
1 |
Asterisk & Zoom x5v 5565 |
| 6:14PM |
0 |
Using 2 x DSL |
| 5:02PM |
0 |
Asterisk server with remote monitoring capab ilities |
| 4:46PM |
5 |
Asterisk server with remote monitoring capabilities |
| 3:13PM |
0 |
Automatically setting mailbox on registration of SIP device by SIP device's line # |
| 1:46PM |
4 |
French Audio Files |
| 1:46PM |
1 |
Driving direction sent to callers mobile phone viatext/sms ? |
| 1:41PM |
1 |
Asterisk 1.0.8 Released |
| 12:51PM |
1 |
*77 does not work .. |
| 12:45PM |
0 |
Driving direction sent to callers mobile phone via text/sms ? |
| 12:44PM |
5 |
SpanDSP - Squished Faxes |
| 12:41PM |
3 |
Chan_Woomera beta released at www.pbxfreeware.org |
| 12:40PM |
2 |
Asterisk 'losing' upstream provider registration state during small network outages. |
| 12:38PM |
0 |
Looking for Honduras DIDs, Origination, Termination |
| 12:19PM |
5 |
INBAND DTMF G729 ASTERISK |
| 12:14PM |
0 |
Pls Hlp - Sendmail handoff / relay to cable |
| 11:18AM |
4 |
12 FXO ports into Asterisk |
| 11:05AM |
1 |
PRI auto reset? |
| 10:41AM |
0 |
Asterisk Manager Interface Remote BufferOverflow Vulnerability |
| 10:21AM |
0 |
Dial a specific priority |
| 9:39AM |
7 |
mini itx |
| 8:30AM |
4 |
Monitoring Sirrix quad BRI channels |
| 8:11AM |
3 |
privacy manager |
| 8:09AM |
2 |
ChanSpy on Asterisk v1.0.7 |
| 8:07AM |
1 |
Always forward an extension? |
| 8:00AM |
0 |
RES: MFC R2 - Can this problem be solved?????????? |
| 7:50AM |
0 |
dialtone conf.of Turkey for ata186 sip |
| 7:39AM |
0 |
This cpu usage doesn't seem right. |
| 7:26AM |
1 |
Polycom display variable |
| 7:23AM |
0 |
BRI signalling Morocco |
| 7:01AM |
0 |
AGI to monitor conenction quality |
| 6:40AM |
1 |
Asterisk @ Home setup & Doc |
| 6:39AM |
2 |
Legal Requirement for Digital PBX |
| 6:25AM |
1 |
Help with Dial multiple channels simultanously |
| 5:58AM |
1 |
Zap lines-inbound,outbound calls intersect |
| 5:57AM |
2 |
Management: Reload performace & Realtime performance ? |
| 5:38AM |
1 |
MGCP Groups |
| 2:48AM |
1 |
Server Load/Capacity |
| 2:23AM |
1 |
Music on Hold Choppy |
| 1:46AM |
0 |
avm c2 correct configuration for two p2p lines |
| 1:03AM |
0 |
Welltech 4 Port FXO - Asterisk |
| 12:57AM |
7 |
Cisco 7960 firmware upgrade promblems |
| 12:19AM |
0 |
Loosing hair on connecting Panasonic PBX- *- Euroisdn Italy |
| 12:11AM |
1 |
SIP DID routing |
| 12:07AM |
0 |
Routing calls by trunk? |
| |
| Wednesday June 22 2005 |
| Time | Replies | Subject |
|---|
| 11:54PM |
3 |
flash panel only works with IP address |
| 11:50PM |
3 |
Loosing hair on connecting Panasonic PBX- * - Euroisdn Italy |
| 11:38PM |
2 |
ASTCC not making calls |
| 11:20PM |
1 |
OT: MAX TNT and PRI calling name (CNAM) facility message |
| 10:44PM |
1 |
missing cdr records |
| 10:30PM |
1 |
Error on installing oh323 on asterisk |
| 10:23PM |
0 |
Malformed/Missing.URL Error from CallManager |
| 9:26PM |
0 |
Asterisk + Asterisk-Stat as a Employee Time Clock |
| 9:03PM |
2 |
Zaptel card AND Ztdummy together? |
| 8:23PM |
1 |
Sip Sidecar Options |
| 7:34PM |
2 |
asterisk authentication issue |
| 7:17PM |
0 |
DMS-500 CID NAME Problem |
| 6:21PM |
0 |
Zaptel + IBM OpenPower Servers |
| 5:25PM |
0 |
add-on mysql cmd |
| 5:14PM |
1 |
RE: res_cepstral |
| 4:49PM |
3 |
indexing tables for dialing |
| 4:30PM |
2 |
Asterisk Manager Interface Remote Buffer Overflow Vulnerability |
| 3:47PM |
1 |
Question on bridged calls |
| 3:46PM |
1 |
Connecting extern telephones, |
| 3:30PM |
0 |
Adit600-->Asterisk Via MGCP |
| 2:45PM |
1 |
Is anyone using VOIPREACH |
| 2:41PM |
0 |
Setup suggestions/ideas |
| 1:37PM |
0 |
Wireless & Wireline Integration |
| 1:32PM |
3 |
combining calls from 2 queues |
| 1:18PM |
1 |
Zap POTS Line Problem calling outbound |
| 1:15PM |
2 |
Weird ring back |
| 11:46AM |
1 |
Using HEAD version of Zaptel with Asterisk Stable Release |
| 11:18AM |
1 |
A Simple * Answering Machine w/ Caller Screening like the olden days |
| 11:04AM |
0 |
Presentation Number |
| 10:01AM |
1 |
Can I dial a number from handset to pickup voicemail? |
| 9:57AM |
0 |
DIAX 0.9.15a with GSM gateway functionality |
| 9:20AM |
2 |
problem compile |
| 9:06AM |
0 |
Performance Monitoring. |
| 9:03AM |
0 |
ISDN (PRI) in the US and Redirect? |
| 9:01AM |
1 |
Garbled one-way audio only with ulaw |
| 8:58AM |
1 |
volume "fading in and out" |
| 8:58AM |
1 |
TE110P Card |
| 8:49AM |
4 |
automated response |
| 8:46AM |
3 |
TDM400P & Channel Group |
| 7:54AM |
0 |
(no subject) |
| 7:51AM |
4 |
TDM400P DevKit Problem |
| 7:50AM |
10 |
New Asterisk Implementation |
| 7:47AM |
4 |
Asterisk Manager Api |
| 7:34AM |
1 |
Fwd:protocol TCP/UDP question |
| 6:58AM |
1 |
FOP related questions |
| 6:41AM |
4 |
FXS interfaces |
| 6:28AM |
0 |
Asterisk ended with exit status 139 |
| 6:23AM |
1 |
gsm gateway |
| 6:13AM |
0 |
TDM400P and Dell Poweredge 1750 |
| 5:44AM |
1 |
Re: [Serusers] ASTERISK+SER+MWI |
| 5:43AM |
2 |
meetme mute status |
| 4:27AM |
2 |
Is this server sufficient? |
| 4:19AM |
0 |
is sip:%2321 valid invite? |
| 3:32AM |
2 |
OT: Asterisk and Mambo - help wanted |
| 3:29AM |
0 |
Detecting the active queue agent... |
| 3:28AM |
1 |
Dialplan Q: Dialing with Capi |
| 3:11AM |
5 |
ZapRAS |
| 2:28AM |
0 |
Variables to emailbody of voicemail |
| 2:28AM |
2 |
Spanish doc |
| 2:03AM |
2 |
Asterisk to NEC NEAX |
| 1:58AM |
1 |
call divert to TRUNK , if one number is unregistered? |
| 1:57AM |
1 |
PPPD problem please help |
| 1:43AM |
0 |
using DBGet inside extensions.ael |
| 1:16AM |
0 |
3month Internship between February end July 2006 |
| 12:54AM |
0 |
Core Dump |
| 12:45AM |
1 |
zeroconf help |
| 12:25AM |
0 |
3-way conference using zap channels -- how is it done? |
| 12:17AM |
0 |
ASTERISK+SER+MWI |
| |
| Tuesday June 21 2005 |
| Time | Replies | Subject |
|---|
| 11:50PM |
1 |
DID not working? + sendmail problems |
| 7:37PM |
3 |
FXS |
| 7:23PM |
1 |
gxp-2000 tftp cfg |
| 7:09PM |
0 |
PBXFreeware.org new res_js example order status checking script. |
| 6:58PM |
1 |
Seeking Inbound 800# Origination for Unique Prostate Cancer Support Call-In Show |
| 6:56PM |
0 |
Questions about FXO Outgoing Dialing |
| 6:37PM |
2 |
Echo Issues |
| 3:47PM |
5 |
logged in agent make an outbound call? |
| 3:43PM |
0 |
Zombie? |
| 3:41PM |
0 |
IAX protocol will not go through firewall after certain time. |
| 3:39PM |
0 |
chan_unicall and /dev/zap/channel |
| 3:19PM |
4 |
voip-info.org unreliable lately? |
| 3:16PM |
1 |
Re: New JAVA application server for Asterisk - OrderlyCalls |
| 2:15PM |
5 |
NVFaxdetect |
| 1:29PM |
0 |
Re: Asterisk-Users Digest, Vol 11, Issue 136 |
| 1:20PM |
0 |
Problem with Connecting pbx and asterisk: using TE405P Asterisk -> T1 -> PBX |
| 1:13PM |
2 |
Digium Card: Echo, Echo and more Echo |
| 12:52PM |
2 |
403 forbidden on SIP register |
| 12:47PM |
0 |
Astricon Europe Media Post! |
| 12:47PM |
5 |
Problem with Connecting PBX to Asterisk |
| 12:26PM |
0 |
Intermittent audio issues with Asterisk behind symmetrical firewa ll |
| 12:21PM |
1 |
Asterisk in India? |
| 12:14PM |
4 |
Grandstream 100 pricing question |
| 11:58AM |
1 |
Asterisk answers with high pitch sound |
| 11:54AM |
2 |
Polycom and CallerID |
| 11:48AM |
0 |
Best Echo Canceller. |
| 11:01AM |
2 |
Re: New JAVA application server for Asterisk - OrderlyCalls |
| 10:06AM |
0 |
[ot] wifi 3G gsm phone |
| 9:41AM |
1 |
MeetMe Problems |
| 9:17AM |
0 |
communication between IAX softphones |
| 9:09AM |
0 |
asterisk-api |
| 8:35AM |
1 |
Asterisk died - exactly every 60 minutes |
| 7:24AM |
1 |
GSM ISDN gateway to quadbri NT port under bristuffed Asterisk - where to set gain (I have unreliable inband dtmf recognition )? |
| 7:22AM |
1 |
chan_unicall, bug in 1.0.X - 99% CPU |
| 7:10AM |
5 |
app_changrab.c released on pbxfreeware.org |
| 6:56AM |
1 |
Cisco 7750 |
| 6:47AM |
1 |
modprobe wctdm waiting for ever |
| 6:22AM |
2 |
MFC R2 - Can this problem be solved?????????? |
| 6:16AM |
0 |
MFC R2 - Can this be solved??? |
| 6:00AM |
1 |
ast_data help |
| 5:28AM |
0 |
Call file calling twice |
| 5:26AM |
0 |
Looking for PRI Outbound Caller ID Configura tion |
| 5:26AM |
1 |
Ground Start on Asterisk |
| 4:22AM |
0 |
Problems with wew FXO modules for TDM400P |
| 2:35AM |
0 |
ipswitchboard |
| 12:37AM |
1 |
Asterisk and Sirrix PCI4S0 echo cancellation |
| 12:18AM |
0 |
Re: Asterisk-Users Digest, Vol 11, Issue 131 |
| 12:04AM |
0 |
[sourceforge.net abuse] New JAVA application server for Asterisk - OrderlyCalls |
| |
| Monday June 20 2005 |
| Time | Replies | Subject |
|---|
| 11:25PM |
2 |
Asterisk does not function without a DNS ser ver |
| 11:03PM |
0 |
asterisk and radius? |
| 10:42PM |
1 |
storing CDR records in a MySQL database |
| 10:09PM |
0 |
Re: app_valetparking.c for * STABLE |
| 9:23PM |
3 |
accountcode not present in cdr |
| 7:27PM |
6 |
Extension Configuration Best Practice |
| 6:05PM |
1 |
the table tariffrate is empty in Areskicc ! |
| 6:04PM |
1 |
Unable to reconfigure channel |
| 5:48PM |
0 |
IAXy G729 Translation |
| 5:08PM |
0 |
Prepaid systems considerations |
| 4:37PM |
3 |
service scan |
| 4:24PM |
1 |
Zaptel Disconnect Tone |
| 3:51PM |
1 |
voicemail system |
| 3:14PM |
4 |
ee1000 Ethernet in Dell 1850 |
| 3:07PM |
0 |
Contexts Calling Each Other |
| 2:56PM |
1 |
Asterisk does not function without a DNS server |
| 2:49PM |
1 |
Compilation Problem with asterisk-addons |
| 2:31PM |
1 |
How can you check that eg TDM04B hardware installed and drivers OK |
| 1:39PM |
1 |
Looking for PRI Outbound Caller ID Configuration |
| 1:37PM |
0 |
Re: app_valetparking.c for * STABLE |
| 1:21PM |
2 |
FXO/FXS cpu spikes, data loss and ztclock. |
| 1:19PM |
2 |
Automatic Agent Login |
| 12:33PM |
0 |
Suggestions for using AbsoluteTimeout |
| 12:22PM |
1 |
SIP Ad-Hoc Conferencing with Asterisk |
| 11:53AM |
1 |
RCAPI ISDN Support |
| 11:23AM |
1 |
PBXfreeware.org Open for business! / JavaScript module for Asterisk Unveiled! |
| 11:20AM |
3 |
QuadBRI: How to set the outgoing callerid (KPN - NL) |
| 10:46AM |
0 |
What should I get for SOHO TDM card or sipura3000? |
| 10:15AM |
1 |
Re: app_valetparking.c for * STABLE (1.0.X) |
| 8:15AM |
0 |
Can't get TDM04B to work! |
| 7:47AM |
1 |
TxFax: can't get a fax to destination (log inside) |
| 7:44AM |
0 |
LookupCIDName on outgoing calls |
| 7:21AM |
1 |
chan_h323 vs chan_oh323 & chan_ooh323 |
| 7:17AM |
2 |
app_valetparking.c |
| 6:33AM |
0 |
MGCP and SIP clients |
| 5:23AM |
0 |
second isdn line doesn't work with avm c2 card |
| 4:11AM |
1 |
call file ignored? |
| 4:11AM |
3 |
AGI/PHP errors |
| 3:27AM |
0 |
CPU load 100% when SIP register |
| 2:53AM |
1 |
oneTouchVoicemail issue with Polycom 1.5.2 |
| 2:46AM |
1 |
$0-per-month (pay as you go) provider with T.38? |
| 1:45AM |
1 |
sipredirect question |
| 12:46AM |
3 |
Debian Vs Fedora |
| 12:25AM |
2 |
webvmail debian package |
| 12:12AM |
0 |
No CDR Records |
| |
| Sunday June 19 2005 |
| Time | Replies | Subject |
|---|
| 11:57PM |
1 |
help for making several calls at the same time.. |
| 11:43PM |
2 |
Zaptel HEAD with * Stable? |
| 11:17PM |
0 |
Premptible Linux Kernel |
| 11:02PM |
0 |
Scratchy audio on Bridged PRI Calls |
| 9:40PM |
4 |
Polycom 500 Sound Problem |
| 3:22PM |
3 |
tos problem |
| 3:17PM |
2 |
NAT/Proxy advise |
| 1:42PM |
0 |
chanisavail...not workin with SIP and IAX |
| 1:19PM |
0 |
Zaptel and Zapata Conf's |
| 1:17PM |
0 |
Problem with astperl primitives say... in astcc |
| 12:54PM |
1 |
MySQL to static .conf |
| 12:46PM |
1 |
*67 with Sipura 3000 |
| 11:00AM |
0 |
chan_capi-cm-0.5.1 fixup release |
| 10:26AM |
2 |
outgoing call routing |
| 10:11AM |
0 |
asterisk and fayn.cz |
| 7:08AM |
0 |
I don't get a queue_log with my version of asterisk (0.7.1). |
| 6:23AM |
0 |
missing mysql cdr records |
| 6:01AM |
0 |
app_curl.so: Can't locate module sound-service-0-3 |
| 5:54AM |
4 |
bluetooth audio and asterisk |
| 5:51AM |
3 |
Libtiff 3.5.7 - recommended version for spandsp |
| 5:24AM |
0 |
stale nonce received |
| 4:29AM |
0 |
Now it is working |
| 3:13AM |
0 |
Is actually somebody doing it: load balancing 20 asterisk servers |
| 12:45AM |
0 |
Asterisk on vpbx exited on signal 11. Might want to take a peek. |
| |
| Saturday June 18 2005 |
| Time | Replies | Subject |
|---|
| 8:35PM |
1 |
channel.c:1884 set_format: Unable to find a path from g729 to gsm |
| 7:45PM |
0 |
How to setup two Asterisk boxes - keeping the registration |
| 5:25PM |
0 |
New Voip-info.org mirror/translation |
| 4:45PM |
0 |
Three way calling with Cisco 12SP+ |
| 2:29PM |
0 |
H323 implementations |
| 1:24PM |
0 |
[Fwd: IAX with shaw cable not going through] |
| 1:16PM |
4 |
IAX with shaw cable not going through |
| 12:00PM |
1 |
Want to test my * behind firewall |
| 11:55AM |
2 |
Unable to make outbound calls |
| 11:39AM |
0 |
Re: Asterisk-Users Digest, Vol 11, Issue 68 |
| 8:00AM |
3 |
FATAL: Error running install command for wctdm |
| 3:45AM |
0 |
UK SMS Config problems |
| 3:18AM |
0 |
E1 to E1 connection |
| 1:54AM |
0 |
TTS |
| 12:59AM |
1 |
How to view all call detail in asterisk server |
| 12:11AM |
2 |
allowing outside dialing only for SIP users |
| |
| Friday June 17 2005 |
| Time | Replies | Subject |
|---|
| 11:15PM |
0 |
AreskiCC and asterisk |
| 8:20PM |
1 |
Unable to find a path from g729 to gsm |
| 8:17PM |
1 |
Local numbers |
| 6:40PM |
0 |
Queue/How to get the number of incoming calls |
| 5:19PM |
2 |
ASTCC Rate Calculation |
| 4:59PM |
1 |
Asterisk ael files |
| 3:48PM |
1 |
callqueues confused :( |
| 2:04PM |
0 |
MGCP files for Polycom |
| 1:34PM |
6 |
Console ALSA Sound |
| 12:26PM |
0 |
Multiple phones on a Zap FXS extension |
| 11:56AM |
0 |
Phone lookup |
| 11:50AM |
0 |
Spandsp - fax problem |
| 11:48AM |
2 |
PLEASE HELP X100P no responding |
| 11:29AM |
0 |
Phantom problem authenticating IAX2 with RSA |
| 10:29AM |
1 |
PIX Firewall Ports and Access-Lists |
| 10:25AM |
0 |
txfax 18Kb file problem |
| 9:36AM |
5 |
tdm400p not working after cvs-head update |
| 9:15AM |
2 |
Can't switch span to E1-mode |
| 8:51AM |
5 |
Presence and IM? |
| 7:59AM |
0 |
Programinng Aplication with Music on Hold |
| 7:54AM |
2 |
Calculating the lenght of time in a call queue? |
| 6:58AM |
0 |
No ringing tone on outgoing SIP trunk |
| 6:55AM |
0 |
auto-dial dial status |
| 6:41AM |
0 |
Asterisk box as a billing machine in a PSTN network |
| 6:12AM |
0 |
Need Last App-Fax Source |
| 5:48AM |
0 |
Agents/Queues Contexts |
| 5:19AM |
4 |
Analog modems behind an Asterisk server? |
| 5:05AM |
2 |
Dell PowerEdge + TDM |
| 4:52AM |
0 |
Call group channel limits |
| 4:02AM |
0 |
Miax: Digital voice channel when connecting to asterisk |
| 3:31AM |
0 |
Sip INFO DTMF over satellite |
| 2:30AM |
1 |
bristuff-0.2.0-RC8g: zaptel error in suse 9.2 |
| 2:08AM |
1 |
Dial timeout when server down |
| 1:58AM |
0 |
ata186 IVR problem |
| 1:06AM |
2 |
Junk at the beginning of frame |
| |
| Thursday June 16 2005 |
| Time | Replies | Subject |
|---|
| 11:58PM |
0 |
inbound agent recording filename |
| 11:10PM |
1 |
Re: Dell PowerEdge SC420 interrupt issue |
| 10:42PM |
0 |
How to detect telco Message Waiting Indicator (WMI) |
| 8:49PM |
0 |
Problem with monitor. |
| 8:24PM |
3 |
Dial Commands "D" Option Question |
| 7:39PM |
1 |
Routing SIP to Cisco routers running IOS 12.3+ |
| 5:37PM |
1 |
Newbie question about pressing a key to, be connected to the caller |
| 5:27PM |
1 |
Viva Madrid! |
| 4:40PM |
5 |
meetme - conf-invalid |
| 3:53PM |
3 |
rxfax problem - libspandsp issue? |
| 3:41PM |
2 |
Multiple Sipura 3000 |
| 1:16PM |
2 |
@Home AMP call recording documentation |
| 1:04PM |
0 |
SIP connection |
| 12:40PM |
4 |
Sipura 3000 help |
| 11:35AM |
1 |
faxdetect config issues |
| 11:12AM |
1 |
iax2 registry - auto reconnect ? |
| 10:45AM |
0 |
How to get started, what do I need? |
| 10:23AM |
0 |
Intelligent maximum channels solution? |
| 10:15AM |
0 |
have asterisk box #2 pick up calls. |
| 9:45AM |
1 |
Coding a telemarketing call blocker |
| 9:01AM |
0 |
misdn and call hangup problem |
| 8:28AM |
2 |
Error when compiling in freeTDS support |
| 8:23AM |
1 |
Nobody picked up in 30000 ms |
| 7:43AM |
0 |
Problem with 2 digium cards |
| 7:27AM |
6 |
Case studies for Asterisk Voicemail |
| 7:05AM |
0 |
Subject: asterisk gsm gateway hardware |
| 6:38AM |
1 |
Cisco 7960 (SIP) with Asterisk: how to get # to work during a call |
| 6:13AM |
1 |
reload from dialplan |
| 5:43AM |
0 |
Fall back dialing |
| 5:36AM |
3 |
SER and Asterisk question |
| 5:34AM |
0 |
How to dimension Asterisk - that is used solely as callback server - only sending untranscoded voice between two ISDN channels on PRI ? |
| 4:15AM |
1 |
unamble to dialout to mobiles and others "special" numbers |
| 4:06AM |
1 |
Do includes include the includes |
| 3:39AM |
0 |
Zaphfc unable to dial out |
| 3:38AM |
0 |
Grandstream phones losing registration withserver. |
| 3:27AM |
0 |
chan_h323 context |
| 3:12AM |
0 |
Features.conf Set Language |
| 3:03AM |
0 |
Error on incoming calls |
| 2:53AM |
0 |
Busy, differences between SIP and Zaptel(bristuff) |
| 1:46AM |
1 |
MeetMe ERROR "Unable to dup channel" |
| 1:25AM |
1 |
Grandstream phones losing registration with server. |
| 1:08AM |
0 |
Asterisk Problems with FXO Ground Start Trunks and DID Wink Start Trunks |
| 12:32AM |
0 |
Asterisk on Fedora Cora 3 |
| 12:22AM |
0 |
SER with Asterisk Problem |
| 12:10AM |
1 |
How to stop Asterisk from changing the SDP? |
| 12:02AM |
9 |
chan_capi-cm-0.5 release announcement |
| |
| Wednesday June 15 2005 |
| Time | Replies | Subject |
|---|
| 11:59PM |
0 |
Asterisk Integration with an SBC-410 phone system |
| 9:53PM |
2 |
iax2 can't listen on virtual interface |
| 9:42PM |
0 |
asterisk with eurovoice.ro? |
| 9:03PM |
2 |
terminating DID to FWD |
| 8:40PM |
1 |
Bridged-appearances |
| 8:36PM |
2 |
VoiceXML? question |
| 8:11PM |
2 |
Bill seconds |
| 7:22PM |
2 |
SIP to PRI |
| 7:10PM |
0 |
Problem with slin |
| 7:06PM |
1 |
Strange Inbound Ring Handling |
| 7:00PM |
1 |
Gnet Phones |
| 6:26PM |
1 |
This mailing list is being spam filtered on my site. |
| 5:48PM |
3 |
Includes include the includes? |
| 5:19PM |
2 |
1-800 DID in Alberta |
| 4:04PM |
3 |
Grandstream ATA Toasted |
| 3:52PM |
0 |
zaptel witch smp |
| 2:46PM |
0 |
SIP to ZAP dialout without pre"0" (Asterisk <-> HiCom) |
| 2:44PM |
1 |
echo cancellation on an iax2 channel |
| 2:05PM |
1 |
Changing caller ID on a Zap channel |
| 2:03PM |
0 |
Dundi - Multiple Results |
| 1:48PM |
1 |
Sipura 841 Ringtones |
| 1:18PM |
1 |
Caller ID on TelaSIP SIP Channel |
| 12:41PM |
0 |
Re: Asterisk-Users Digest, Vol 11, Issue 100 |
| 12:30PM |
1 |
Problem with overlap dialing |
| 12:27PM |
1 |
CellPhone BlueTooth adapater with Wireless Profile ?? |
| 12:13PM |
0 |
Config files under CVS versioning system |
| 12:07PM |
1 |
phantom answer |
| 12:00PM |
12 |
WiFi IP Phones |
| 11:47AM |
6 |
Help with Cron and Reload |
| 11:09AM |
1 |
SIP transfer/REFER to voicemail problem |
| 10:51AM |
1 |
Question on cdr_odbc |
| 9:34AM |
0 |
asterisk gsm gateway hardware recommendation? |
| 8:16AM |
0 |
Error installing Asterisk with zaptel and libpri |
| 8:15AM |
0 |
Handling -1 in dialplans |
| 8:04AM |
0 |
user web interface |
| 7:55AM |
2 |
Asterisk and Max TNT |
| 6:50AM |
0 |
RE: Call being answered, but no audio on either end |
| 6:37AM |
1 |
Broadvoice and Inbound DTMF |
| 6:17AM |
1 |
[Help] ZT_CHANCONFIG failed on channel 25 |
| 6:01AM |
1 |
Port Inquiry |
| 5:45AM |
0 |
Load problems |
| 5:43AM |
1 |
asterisk security |
| 5:40AM |
2 |
Nasty little incident ... |
| 5:28AM |
0 |
empty HDLC frame or bad CRC received |
| 5:10AM |
0 |
CDR's -> ODBC and logging IP's |
| 4:13AM |
2 |
SIP call doesn't execute the 's'-extension |
| 3:47AM |
0 |
Asterisk slow transferring calls |
| 3:24AM |
1 |
Sipura SPA 3000 FXO Setting India |
| 3:20AM |
4 |
Dial more then 9 digits |
| 3:13AM |
0 |
Personalised Unavail / busy messages no longer play |
| 3:08AM |
1 |
app_dial.c:977 dial_exec_full: Unable to create channel of type 'Zap' (cause 0) |
| 2:56AM |
0 |
Stable Versions |
| 2:40AM |
1 |
How to restrict access to * for a specific soft/hard phone model? |
| 1:47AM |
1 |
Old but Gold |
| 1:45AM |
0 |
SIP REFER method. |
| 1:35AM |
1 |
newbie question.. |
| 12:59AM |
0 |
TAPI for Dos (xbase / Clipper) |
| 12:54AM |
0 |
SV: Asterisk and Panasonic KX-TD1232 |
| 12:09AM |
0 |
localize ${VM_DATE} ? |
| |
| Tuesday June 14 2005 |
| Time | Replies | Subject |
|---|
| 8:12PM |
0 |
upgrade problems... |
| 7:35PM |
0 |
Asterisk & outbound proxy? |
| 7:18PM |
1 |
Newbie question about pressing a key to be connected to the caller |
| 7:17PM |
3 |
Calling on all Polycom Experts |
| 7:07PM |
2 |
[PRI] TE110P |
| 6:52PM |
1 |
canreinvite=yes not working with sipura device. |
| 6:43PM |
0 |
Asterisk@Home and intermittent ring sound on callers end |
| 6:13PM |
0 |
RE: Call being answered, but no audio on either end |
| 6:09PM |
1 |
how to make a dialplan on bases of Caller |
| 6:05PM |
0 |
Snom hardware quality |
| 6:02PM |
0 |
dialplan appdata separators |
| 5:39PM |
1 |
Asterisk and grandstream weird call probs |
| 4:10PM |
0 |
ATA186 & X100P - detect hangup |
| 3:56PM |
0 |
Digit Map for IP500 - prepend digits from phone |
| 3:10PM |
1 |
OH323 Packetization |
| 2:30PM |
0 |
Call being answered, but no audio on either end (Intermittent) |
| 1:04PM |
0 |
AW: Should I choose DSL @ 1.5 or a full T1? |
| 12:59PM |
0 |
how to remove + in CIDNumber |
| 12:56PM |
0 |
chan_sip.c: Maximum retries exceeded on call ........ for seqno 1 01 (Non-critical Response) |
| 12:49PM |
0 |
Has anyone made a purchase from http://www.telephonyware.com/? |
| 11:54AM |
0 |
Cannot handle frames in 2 format |
| 11:16AM |
2 |
Call parking in multi user environment |
| 11:14AM |
0 |
RJ45 instead RJ11 in Digium's TDM20B card he lp me please |
| 11:11AM |
0 |
RJ45 instead RJ11 in Digium's TDM20B card help meplease |
| 10:46AM |
5 |
RJ45 instead RJ11 in Digium's TDM20B card help me please |
| 10:36AM |
1 |
outgoing prefix dial plan |
| 10:34AM |
0 |
Zaptel problem on BSD |
| 9:33AM |
2 |
-HEAD/--STABLE using 100% cpu |
| 9:08AM |
2 |
Questions about contexts |
| 9:04AM |
8 |
Making Asterisk NOT Pickup a Line when Ringing? |
| 8:21AM |
0 |
Info on ACD in Asterisk |
| 8:19AM |
0 |
Transfers on PRI connected channel banks and legacy PBX's |
| 7:39AM |
4 |
488 Not Acceptable Here |
| 7:27AM |
6 |
VOIP-INFO down? |
| 6:43AM |
2 |
SIP_HEADER - anybody using it? |
| 6:28AM |
5 |
HT-488 vs. SPA-3000? |
| 6:00AM |
3 |
How to setup a test number to know my extension number |
| 5:56AM |
2 |
Asterisk and Panasonic KX-TD1232 |
| 5:48AM |
1 |
Long time to detect hang-up |
| 5:26AM |
2 |
# no longer working |
| 4:57AM |
2 |
Features.conf for secretary function |
| 4:54AM |
0 |
ERROR[6504]: chan_zap.c:6710 mkintf: Channel 24 is reserved for D-channel. |
| 4:44AM |
0 |
Is there a problem when we want to transfer anincoming call to an external phone number |
| 4:10AM |
1 |
Re: Asterisk-Users Digest, Vol 11, Issue 93 |
| 4:01AM |
3 |
RTP Forwarding |
| 3:31AM |
1 |
SIP to ZAP Dialplan |
| 2:57AM |
0 |
How to connect to LVDX / opentelecoms.org |
| 2:35AM |
2 |
ERROR[5539]: chan_zap.c:9750 setup_zap: Unable to load config zapata.conf |
| 2:32AM |
2 |
AVAYA & Asteris & H323 chanel |
| 2:11AM |
0 |
Static CLID |
| 1:59AM |
0 |
Max Retries Exceeded - IAX2. Network problem? |
| 1:16AM |
1 |
Is there a problem when we want to transfer an incoming call to an external phone number |
| 12:06AM |
0 |
No mans problem? |
| |
| Monday June 13 2005 |
| Time | Replies | Subject |
|---|
| 10:42PM |
2 |
Adtran TA 750 FXO Groundstart Mode |
| 10:22PM |
7 |
Keeping users, extensions, voicemail and so on in DB |
| 9:45PM |
9 |
SIP Listen to multiple ports |
| 8:54PM |
0 |
Re: Asterisk forking, Was: Digium Website Update:Asterisk Business Edition |
| 7:31PM |
2 |
ztcfg server crash |
| 7:00PM |
1 |
Need help connecting two * pcs with *@home |
| 6:58PM |
0 |
Reading about G729 |
| 6:19PM |
0 |
IAX Issues... |
| 5:00PM |
3 |
problem with pf and asterisk |
| 3:59PM |
7 |
MCI vs. XO/Allegiance |
| 2:09PM |
1 |
Re: Re: Digium Website Update: Asterisk Busi ness Edition |
| 2:03PM |
0 |
Re: Asterisk forking, Was: Digium Website Up date:Asterisk Business Edition |
| 1:55PM |
0 |
Re: Voicemail and MS Exchange Synchronizatio n |
| 1:27PM |
0 |
Unable to support trunking .... without zaptel timing |
| 1:14PM |
0 |
Asterisk connecting remote villages in westernUganda |
| 12:39PM |
0 |
DID in AMP with 2+ incoming lines |
| 11:52AM |
0 |
Hiss patch |
| 10:55AM |
2 |
Asterisk connecting remote villages in western Uganda |
| 10:42AM |
1 |
Zaptel modules |
| 9:57AM |
1 |
More on the IAD connection |
| 9:01AM |
1 |
DNIS and DID seeking confirmation |
| 8:56AM |
1 |
Interfacing to an IAD |
| 8:43AM |
0 |
T1 multiplexer (or ?) for failover in largeinstallation |
| 8:42AM |
2 |
snom 190: dial tone without registration? |
| 8:35AM |
2 |
T1 multiplexer (or ?) for failover in large installation |
| 8:15AM |
1 |
Components and suggestions for an asterisk server with 9 to 17 POTS. |
| 8:04AM |
1 |
wiki server session limit? |
| 7:20AM |
1 |
presence and video conference |
| 6:56AM |
0 |
nativ bridging problem with ilbc!! |
| 6:04AM |
3 |
Oh323 and Caller ID missing |
| 5:47AM |
1 |
Cepstral partnership with Digium |
| 5:25AM |
0 |
Guidance , for which card to buy |
| 5:17AM |
2 |
SNOM, Asterisk and Attended transfer (bug?) |
| 4:59AM |
0 |
MySQL: max realistic size of extensions table. |
| 3:37AM |
1 |
about timeouts |
| 2:08AM |
1 |
Problem with DTMF Relay and Oh323 |
| 1:53AM |
0 |
Asterisk installation error after CVS update |
| 12:32AM |
2 |
Need Help with pickup *8 |
| 12:27AM |
0 |
Phantom incoming calls on x100p |
| |
| Sunday June 12 2005 |
| Time | Replies | Subject |
|---|
| 11:35PM |
0 |
Macro support in realtime |
| 10:41PM |
2 |
POLYCOM IP 500 Setup |
| 7:54PM |
0 |
phone rings but caller doest hear it |
| 5:11PM |
0 |
Asterisk Community Meeting in Sydney Australia |
| 3:48PM |
0 |
*66 auto redial emulation? |
| 2:51PM |
0 |
ZAP channel (X100P) won't detect call waiting |
| 1:40PM |
1 |
how to tell |
| 12:31PM |
0 |
Unable To Register a SIP phone ... Help Needed |
| 10:34AM |
3 |
GSM -> ULAW sound conversion |
| 10:10AM |
1 |
Not answering inbound a line used for outboun |
| 9:50AM |
0 |
Re: Asterisk-Users Digest, Vol 11, Issue 81 |
| 8:03AM |
0 |
IAXy Pulse/Flash timing |
| 7:52AM |
1 |
DID Issue |
| 7:32AM |
3 |
Best bet ... IAX vs SIP |
| 6:07AM |
1 |
Changing Email Templates |
| 1:56AM |
1 |
Load balancing for each protocol |
| |
| Saturday June 11 2005 |
| Time | Replies | Subject |
|---|
| 11:55PM |
3 |
Not answering inbound a line used for outbound |
| 8:07PM |
4 |
PRI Trouble |
| 6:29PM |
0 |
SPA-2001 features on analog side |
| 5:28PM |
0 |
LCDC Integration/bounty |
| 4:32PM |
1 |
ISDN Sub-Address |
| 4:27PM |
0 |
Help with denighing access to certain numbersbyCallerID |
| 3:46PM |
2 |
Help with Oh323 |
| 3:45PM |
0 |
Help with denighing access to certain numbers byCallerID |
| 3:39PM |
0 |
Help with denighing access to certain numbers by CallerID |
| 3:14PM |
0 |
Re: ztdummy/rtc - staticy audio |
| 3:05PM |
1 |
SIP-H.323 dial tone and busy tone problem. |
| 2:46PM |
0 |
Re: ztdummy/rtc - staticy audio |
| 2:33PM |
1 |
Problems with IAX Trunks |
| 2:25PM |
1 |
SIP Connection Timing Out BroadVoice |
| 2:04PM |
0 |
Transcoding GSM to G723.1 |
| 2:02PM |
0 |
Flash hook not going through SPA-2002 |
| 1:33PM |
1 |
AreskiCC Calling Problem |
| 11:13AM |
3 |
how to limit simultaneous calls |
| 11:01AM |
0 |
Voice quality of Softphones vs. IP Phones an d Gateways. |
| 10:56AM |
0 |
Shorewall Configuration for Asterisk Box |
| 10:44AM |
0 |
SIP_HEADER example |
| 10:35AM |
0 |
Re: Asterisk-Users Digest, Vol 11, Issue 77 |
| 10:04AM |
3 |
ztdummy/rtc |
| 8:55AM |
0 |
Caller ID transforms |
| 8:49AM |
0 |
Deleting Unavail Message |
| 8:09AM |
0 |
In Dial Application, reading the L(x[:y][:z]) parameter from database. |
| 7:55AM |
3 |
No path to translate from SIP/615-25c8(256) to SIP/601-27b6(4) |
| 7:23AM |
0 |
Asterisk Users & Developers on their way to Madrid - Meet us there! |
| 7:03AM |
0 |
Voice quality of Softphones vs. IP Phones and Gateways. |
| 4:22AM |
4 |
Best platform |
| 3:02AM |
1 |
Why does my name not show in the from address |
| 2:48AM |
1 |
Manager API timestamps of events |
| 12:21AM |
0 |
Newbie Here..... Unable To Register A SIP phone |
| |
| Friday June 10 2005 |
| Time | Replies | Subject |
|---|
| 10:50PM |
2 |
what is asteriskathome-1.0.iso? |
| 10:35PM |
1 |
VoicePulse DTMF Problems Anyone? |
| 9:36PM |
1 |
Is it necessary that i need to have TDM01B for PC-to-PC intercom calls? |
| 7:45PM |
2 |
Asterisk@Home connecting through firewall/router |
| 6:49PM |
1 |
[newbie] configuration for IAX server to server |
| 6:26PM |
19 |
Should I choose DSL @ 1.5 or a full T1? |
| 6:23PM |
0 |
First Asterisk community meeting in Sydney |
| 4:48PM |
1 |
Wildly inaccurate CDR records |
| 4:12PM |
0 |
Open telecoms.org |
| 3:12PM |
0 |
voiceblue gsm/sip |
| 3:00PM |
1 |
Polycoms Go Silent after a a handful of calls. |
| 2:47PM |
1 |
Convert extensions.conf INTO MySQL script |
| 2:27PM |
1 |
Newie Questions |
| 2:11PM |
1 |
Unable to register Zyxel WIFI Phone as SIP Client to Asterisk |
| 2:05PM |
3 |
DMS-500 CID name not in CDR |
| 1:36PM |
0 |
Unable To Register A SIP phone |
| 11:31AM |
2 |
Toll Free DIDs |
| 10:19AM |
0 |
Call disconnect |
| 10:12AM |
0 |
Re: Asterisk-Users Digest, Vol 11, Issue 71 |
| 9:23AM |
0 |
Dropping Frame of G729 |
| 9:15AM |
0 |
blindtransfers with IAX |
| 8:58AM |
2 |
G711 ( alaw or ulaw ) pass-thru |
| 8:48AM |
0 |
AAH 1.1 cannot call between extensions (xten lite softphones) |
| 8:44AM |
4 |
Best BootRom & SIP Code for Poly600? |
| 8:08AM |
1 |
config problem |
| 7:45AM |
0 |
asterisk and mpg123 lock up |
| 7:41AM |
0 |
Re: Asterisk-Users Digest, Vol 11, Issue 69 |
| 7:41AM |
0 |
Is it possible to have a remote Phone work behindNat without a VPN? |
| 7:20AM |
0 |
D-Link DVG-1402S |
| 7:07AM |
1 |
Re: Voicemail and MS Exchange Synchronizatio n |
| 6:55AM |
1 |
ATTN: Keith - Seriously OT |
| 6:53AM |
0 |
SoftPhone - Solaris |
| 6:48AM |
1 |
404 not found |
| 5:56AM |
2 |
Asterisk Evening in Melbourne (again!) next Thursday |
| 5:51AM |
0 |
g729 support |
| 5:41AM |
0 |
SpanDSP wownt compile |
| 5:24AM |
1 |
TE410P and Siemens HIPATH 3750 |
| 5:01AM |
2 |
Cell redirect |
| 4:47AM |
0 |
Handytone-488 FXO - PSTN in calls to Asterisk, sip.conf? (fwd) |
| 4:35AM |
3 |
chan_unicall, dtmf bug in * 1.0.x - 99.9% CPU |
| 3:17AM |
1 |
Call inband progress indication and zaphfc |
| 2:09AM |
5 |
lost g729 lic |
| 1:58AM |
1 |
Request OPTION and 404 Sjphone Xlite |
| 1:53AM |
2 |
G.729AB codec support |
| 12:13AM |
0 |
sirrix NT mode |
| |
| Thursday June 9 2005 |
| Time | Replies | Subject |
|---|
| 11:45PM |
1 |
PHPAGI Swift Escape Digits |
| 11:30PM |
1 |
Zap Clocking - Frame Slips - tdm400p wcfxo zttest cpu spikes spandsp |
| 11:18PM |
2 |
Is it possible to have a remote Phone work behind Nat without a VPN? |
| 10:21PM |
1 |
IAX2 Max Retries dropped calls Firefly |
| 8:09PM |
0 |
"auto-dial out" not waiting for answer |
| 7:35PM |
0 |
Conversations cuts: "didn't get a frame from Channel: SIP/..." |
| 7:22PM |
1 |
compile error cannot find -lidn |
| 6:58PM |
0 |
Multiple Digium cards? |
| 6:19PM |
0 |
GXP-2000 Wiki update.. |
| 6:04PM |
0 |
Handytone-488 FXO - PSTN in calls to Asterisk, sip.conf? |
| 5:40PM |
2 |
Sixtel is still alive? |
| 3:43PM |
0 |
Flash Hook won't work with Asterisk@Home and SPS-2002 |
| 3:28PM |
2 |
VOIP-INFO.ORG |
| 2:33PM |
12 |
VOIP-INFO |
| 2:12PM |
0 |
Parked Call queue function key notify |
| 1:26PM |
1 |
Phantom (ghost) Calls with Wildcard TDM400P |
| 1:25PM |
1 |
voicemail check for busy message |
| 1:07PM |
5 |
Voicemail and MS Exchange |
| 12:53PM |
1 |
Inbound provider in Canada |
| 11:46AM |
0 |
Getting a Quintum AS200 to connect with Asterisk using SIP? |
| 11:04AM |
0 |
OT: SpamFiltering (used to be: ATTN: Keith) |
| 10:49AM |
0 |
Digium vs. Sangoma: Performance |
| 10:39AM |
4 |
ATTN: Keith |
| 10:38AM |
1 |
Asterisk to Cisco Voip System Unity |
| 10:18AM |
23 |
Voicemail and MS Exchange Synchronization |
| 10:18AM |
1 |
astGUIclient installation problem |
| 9:51AM |
0 |
Agent refuses to log out |
| 8:59AM |
0 |
Polycom IP-500 & 600 Nat settings. |
| 8:51AM |
8 |
howto write CDRs on two mysql servers |
| 8:46AM |
1 |
REPOSTED: Polycom 500 "Group Call Pickup Feature" and * |
| 8:16AM |
1 |
Cisco 7960 and Skinny |
| 8:08AM |
2 |
E1 and SS7 |
| 8:04AM |
2 |
having to reload asterix after internet connection failure |
| 7:02AM |
4 |
Lingo(.com) and Asterisk |
| 5:49AM |
1 |
3COM NBX SuperStack 3 |
| 5:12AM |
3 |
Pickup problem |
| 4:10AM |
3 |
Asterisk to Cisco Unity |
| 1:50AM |
3 |
Softphone for Linux desktops |
| 1:22AM |
0 |
New version 1.013 of Asterisk VConfig |
| |
| Wednesday June 8 2005 |
| Time | Replies | Subject |
|---|
| 11:44PM |
1 |
TDM400P strangeness |
| 11:26PM |
0 |
Asterisk Engineer/Programmer required |
| 11:05PM |
1 |
Thank you for the timely suggestion |
| 10:48PM |
3 |
Play MP3 during Record |
| 10:23PM |
5 |
GXP2000 and hint LED's |
| 10:01PM |
1 |
tdm04b slow response |
| 7:37PM |
2 |
[ADMIN]: subscription failure |
| 6:15PM |
3 |
More than one account from the same provider? |
| 6:01PM |
2 |
Ringing a few phones |
| 5:39PM |
1 |
Cisco 7960 mic generating noise on other end |
| 5:31PM |
2 |
Incoming call stops at random with Teliax |
| 3:57PM |
2 |
format g729 and Voxee.com |
| 3:04PM |
3 |
AgentCallBacklogin (logout continued...) |
| 2:35PM |
0 |
Load per server? |
| 2:28PM |
2 |
IP PHONE iareaphone x100, tested?? |
| 2:08PM |
13 |
Anyone noticed Voipjet voice quality problems? |
| 12:23PM |
1 |
Do I need a ring capacitor to use TDM400P cards in UK |
| 11:25AM |
1 |
rxfax not working |
| 11:11AM |
0 |
Number of AGI's running at the same time |
| 10:54AM |
1 |
EuroISDN Italy - quadbri - zaptel.conf - what settings work ? |
| 10:23AM |
8 |
TDM04B |
| 9:33AM |
1 |
Remote CDR logging on mysql: |
| 9:23AM |
0 |
CVS Head, Flex 2.5.31 or higher? READ THIS! |
| 8:19AM |
7 |
Clicks in audio with TE100P PRI |
| 8:14AM |
0 |
Asterisk and Alcatel 4200 PBX |
| 8:05AM |
1 |
Latest CVS and app_rxfax |
| 7:57AM |
3 |
TDM400P... ignoring hanguponpolarityswitch |
| 7:28AM |
10 |
* @ Home: All Circuits busy |
| 7:00AM |
0 |
sip to sip echo with meetme, timing |
| 6:59AM |
0 |
Polycom 500 "Group Call Pickup Feature" and * |
| 6:40AM |
0 |
Fax + Fritz + Capi + detection |
| 5:38AM |
2 |
Station Lines |
| 4:04AM |
0 |
Faxing error rtp.c:504 ast_rtp_read: Unknown RTP codec 100 received |
| 3:55AM |
2 |
How to handle one incoming call on multiple lines? |
| 2:39AM |
1 |
performance of * in several scenarios |
| 2:21AM |
1 |
no DTMF pass-thru |
| 2:09AM |
0 |
file.c:1073 ast_waitstream_full: Wait failed (Interrupted system call) |
| 1:28AM |
5 |
Xlite not communicating with Asterisk |
| 1:03AM |
2 |
bypass incoming ring..is it possible? |
| 12:54AM |
1 |
error message: INIT: Id "s0" respawning toofast:disable for 5 minutes |
| 12:44AM |
1 |
Newbie on asterisk ask for configuratio help |
| 12:33AM |
1 |
Asterisk to Avaya PBX using TDM cards |
| |
| Tuesday June 7 2005 |
| Time | Replies | Subject |
|---|
| 10:39PM |
1 |
MGCP Useragent |
| 10:07PM |
1 |
Message Playback |
| 10:02PM |
2 |
so what are the additional hardware components needed? |
| 9:13PM |
3 |
Help Connecting Cisco AS5300 to Asterisk |
| 9:05PM |
2 |
Books |
| 8:46PM |
1 |
error message: INIT: Id "s0" respawning too fast: disable for 5 minutes |
| 7:54PM |
0 |
Incoming voice "disappears" |
| 6:55PM |
2 |
codec preference |
| 5:17PM |
0 |
meetme recording of one user in the conference |
| 4:54PM |
3 |
FXO Gateway recommendation |
| 4:49PM |
2 |
Gnudialer |
| 4:22PM |
2 |
ASTCC what has been changed |
| 3:40PM |
1 |
connecting Asterisk to NEC NEAX system |
| 3:20PM |
1 |
Fax problem with Asterisk @home ver 1.0.7 |
| 3:11PM |
1 |
DID on SIP channel |
| 2:25PM |
1 |
DISA Help |
| 1:53PM |
2 |
Help! Zap echo on bridged calls |
| 1:19PM |
0 |
Incorrect FAX detection. |
| 12:54PM |
0 |
X100P long delay before dial |
| 11:42AM |
1 |
SPA-2002 and NAT |
| 11:38AM |
1 |
Problem in Reloading the asterisk server ! |
| 11:25AM |
0 |
Re: Asterisk-Users Digest, Vol 11, Issue 48 |
| 10:59AM |
1 |
TE410P |
| 10:32AM |
0 |
AgentCallbackLogin (logout) |
| 9:24AM |
3 |
Polycom Phones & shorter than /24 netmasks |
| 8:49AM |
1 |
New Asterisk Manager Proxy -- astmanproxy 1.0 |
| 8:45AM |
3 |
rxfax not answering |
| 8:13AM |
0 |
Monitor and failing Fax |
| 8:08AM |
1 |
realtime & nat |
| 7:53AM |
3 |
AAH 1.1 - CRM Setup |
| 7:43AM |
4 |
Queue Log |
| 7:13AM |
2 |
Multiple E1s on one box |
| 6:36AM |
2 |
PRI Lines not being answered (No User Responding) |
| 6:18AM |
0 |
Sounds |
| 6:07AM |
1 |
Polycom 500 'SERVICE'S' key |
| 5:50AM |
0 |
(no subject) |
| 5:34AM |
0 |
3com 3105 Attendant DSS Console (SIP??) |
| 5:27AM |
1 |
CallerID/chan_sccp |
| 5:23AM |
0 |
NEWBIE: sip subscriptions |
| 5:01AM |
2 |
Call Routing based on number dialed (using S IP) |
| 4:29AM |
1 |
RE: Asterisk-Users] te405p and dell poweredge |
| 3:26AM |
4 |
I want to move the MySQL server out to another machine |
| 3:20AM |
1 |
How to configure 2 asterisks |
| 3:05AM |
1 |
D-link DPH-80 (SIP) call to asterisk problem |
| 2:22AM |
3 |
te405p and dell poweredge |
| 2:10AM |
1 |
connecting Asterisk with Siemens HiPath4000 |
| 1:25AM |
0 |
Duplicate Calls |
| 1:21AM |
2 |
Problems with Junghanns QuadBri |
| 1:05AM |
2 |
How to allow multiple codecs in A@H |
| 1:02AM |
3 |
run a script on completion of call |
| 12:44AM |
1 |
USB phones... |
| 12:14AM |
0 |
Re: chan_sccp / 7960: "External call" and more |
| |
| Monday June 6 2005 |
| Time | Replies | Subject |
|---|
| 9:46PM |
1 |
RE: LOA for CFA . . work up "pencil copy" |
| 7:04PM |
1 |
Service Unavailble, Sipura 3000, CheckGroup, what the heck?? |
| 5:34PM |
1 |
Debugging SIP Connection |
| 5:19PM |
1 |
NAT & RealTime |
| 3:35PM |
0 |
About BillSec when having conferences |
| 3:32PM |
1 |
Transfer differences between BudgeTone101 and Snom190 |
| 3:19PM |
1 |
ADSI over SIP |
| 3:03PM |
2 |
ENUM NL dead ? |
| 2:57PM |
1 |
Servers Compatible with Digium HW |
| 2:39PM |
1 |
Double NAT issues with SIP and workaround (?) |
| 2:37PM |
4 |
*@home .conf files request |
| 1:49PM |
3 |
Asterisk eating up 99.8% cpu |
| 12:39PM |
1 |
CLUELESS NEWBIE needs help making an outboundsip call to PSTN |
| 12:36PM |
0 |
dial-a-string (e.g. 2=a,b,c, 3=d,e,f) |
| 12:30PM |
2 |
Erro message - Received mini frame before first full voice frame |
| 11:44AM |
0 |
Newbee, help with cdr/odbc/mysql logging problem |
| 11:36AM |
0 |
Unable to Configure NetPhone IP phone |
| 11:32AM |
5 |
Asterisk Live! CF |
| 11:17AM |
0 |
Degraded voice without packet loss |
| 11:09AM |
0 |
Dial(SIP/xyz&zap/r1/123) with different Caller Ids ?! |
| 11:08AM |
0 |
How to make Polycom phones work with Asterisk asaSIP Client? |
| 10:17AM |
5 |
OT: Please comment on Dvorak's troll |
| 10:16AM |
1 |
Hangupcode == 44 |
| 9:43AM |
0 |
How to make Polycom phones work with Asterisk as aSIP Client? |
| 9:08AM |
0 |
D channel initialization |
| 9:08AM |
0 |
How to make Polycom phones work with Asterisk as a SIP Client? |
| 8:24AM |
0 |
OT: WAS: * found in Iraq!! NOW: Asterisk bus iness sightings |
| 8:20AM |
1 |
Asterisk at Home ... |
| 8:01AM |
1 |
IAX Phone Pro - Open Beta Test |
| 7:35AM |
5 |
Polycom 500... |
| 7:09AM |
1 |
Zaptel comple on FC2 |
| 6:35AM |
0 |
Any thoughts |
| 6:30AM |
5 |
IRQ Problems |
| 6:12AM |
0 |
Echo Issues via SIP |
| 5:53AM |
2 |
Features.conf - atxfer |
| 5:34AM |
0 |
snom 360 conference button |
| 5:24AM |
0 |
[SPAM] - what hardware components do i need? - Email found in subject |
| 4:45AM |
1 |
AMP and custom application |
| 4:37AM |
2 |
How to Playback a file continuously during conversation? |
| 3:53AM |
1 |
Compiling asterisk-addons-1.0.7 on Debian Sarge with asterisk-packages installed |
| 3:00AM |
1 |
what hardware components do i need? |
| 2:47AM |
1 |
Quotation request: 12 KHz signal generation for billing purposes. |
| 2:29AM |
1 |
Issue with SIP inter-op |
| 2:10AM |
2 |
Variables and status problems in AGI application |
| 1:41AM |
2 |
No DTMF interpretation on outgoing calls |
| 1:28AM |
2 |
mISDN + chan_misdn.so + winbond issue |
| 12:17AM |
0 |
SIP changes in CVS head |
| |
| Sunday June 5 2005 |
| Time | Replies | Subject |
|---|
| 11:40PM |
1 |
Little help with MySQL please |
| 11:26PM |
0 |
Re: Bison, Flex, Conditional Expression |
| 9:15PM |
0 |
RXFax and Hangup context Question. |
| 9:11PM |
2 |
TDM400P Polarity reversal detection |
| 8:46PM |
1 |
Problems getting VoicePulse Connect working |
| 7:36PM |
1 |
Accountcode being ignored? |
| 5:48PM |
1 |
TDM20B FXS card configuration? |
| 5:34PM |
1 |
Voice Dtect |
| 4:26PM |
0 |
sipura3000 problems in callcenter |
| 4:04PM |
2 |
Disa - how it returns on user not dialing any numbers ? |
| 3:57PM |
0 |
Adtran 600 channel bank |
| 3:49PM |
0 |
VoiceMail Termporary greeting option |
| 3:41PM |
0 |
Examples of Asterisk deployments with 100-500 users? |
| 2:09PM |
1 |
IAXtel update! |
| 2:06PM |
3 |
ISDN 4 BRI card for UK |
| 1:14PM |
2 |
te410p not working after cvs-head update |
| 12:47PM |
0 |
Outgoing TDM400P FXO calls always answered |
| 11:45AM |
0 |
UK call disconnects during record |
| 11:18AM |
0 |
ACD Login |
| 9:33AM |
1 |
DTMF Tone Lengths |
| 8:31AM |
4 |
Digium G729 licensing - is it worth the trouble? |
| 7:07AM |
2 |
180 Ringing? |
| 2:50AM |
2 |
Compilation on Debian with support for HFC-chip based ISDN-cards |
| 2:38AM |
0 |
CDR records.. How do you deal with them? |
| 2:10AM |
1 |
New version of Asterisk VConfig |
| 1:17AM |
1 |
Unable to create channel of type SIP-please help |
| |
| Saturday June 4 2005 |
| Time | Replies | Subject |
|---|
| 11:40PM |
0 |
New version of IPSwitchBoard |
| 10:41PM |
1 |
Extension 'hint' info please? |
| 10:39PM |
1 |
SetCallerID based on extension |
| 10:37PM |
3 |
zap to zap bridging not hanging up |
| 3:10PM |
0 |
Satelite Internet connection |
| 2:48PM |
0 |
facing problems with TDM400P |
| 1:39PM |
1 |
Asterisk@Home Forum Suggestions |
| 12:20PM |
0 |
Garbled speech - strange problem. |
| 11:57AM |
4 |
X100P installed OK, after added TDM400P Asterisk would no longer start |
| 11:44AM |
1 |
How to quickly replace ',' with '|' in dialplans? |
| 5:31AM |
2 |
chan_sip + MD5 encryption: WARNING Format for authentication entry is user[:secret]@realm |
| 5:26AM |
0 |
Problem with X100P (ZT_SPANCONFIG failed) |
| 4:25AM |
1 |
call queues problem |
| 4:24AM |
0 |
Apologies |
| 4:02AM |
0 |
chan_sccp / 7960: Messages key, line / speeddial keys |
| 3:31AM |
0 |
Re: [Asterisk-Dev] Is there any SCCP patent issue? |
| 3:28AM |
1 |
Incoming SIP calls with no extension |
| 3:27AM |
0 |
[Fwd: [Fwd: Help Cisco 12+]] |
| 3:27AM |
0 |
[Fwd: [Fwd: Use with Octtel 8 port fxs device]] |
| 3:25AM |
0 |
[Fwd: Use with Octtel 8 port fxs device] |
| 3:25AM |
0 |
[Fwd: Help Cisco 12+] |
| 2:51AM |
3 |
Automatic callback feature *66 |
| 2:21AM |
0 |
SMS + SMSQ |
| 2:13AM |
0 |
Is there any SCCP patent issue? |
| 1:52AM |
3 |
SNOM extension lights programmable, eg. based on asterisk variable setting? |
| 12:19AM |
2 |
Zap channel not hangingup |
| |
| Friday June 3 2005 |
| Time | Replies | Subject |
|---|
| 9:25PM |
1 |
ztdummy errors on WBEL4 |
| 9:23PM |
1 |
ARESKICC DOESN'T make a CALL!!! |
| 8:16PM |
1 |
Caller ID Routing using VoicePulseConnect |
| 7:05PM |
0 |
spam filter |
| 4:55PM |
1 |
Asterisk and Audiocodes 108 FXS |
| 4:53PM |
1 |
Problem starting RX_FAX and TX_FAX Module |
| 4:16PM |
0 |
(no subject) |
| 3:07PM |
6 |
Livevoip 800 Choppy Audio |
| 2:56PM |
1 |
login/logout of call queue |
| 1:53PM |
0 |
Call Routing based on number dialed (using SIP) |
| 1:49PM |
0 |
Use with Octtel 8 port fxs device |
| 1:44PM |
0 |
Help Cisco 12+ |
| 12:58PM |
1 |
Can an open source project get acquired? |
| 12:51PM |
2 |
Setting up calls through the manager interface |
| 12:48PM |
0 |
add/remove PRI card without rebooting |
| 11:39AM |
0 |
Asterisk - >SIP -> DNIS |
| 11:37AM |
1 |
AgentLogin already on? |
| 11:20AM |
1 |
Call parking on Polycom 500 doesn't transfer, stays on hold |
| 10:39AM |
1 |
Last astcc/* versions working? |
| 10:01AM |
0 |
Dazed and confused on refresh |
| 9:48AM |
0 |
New astGUIclient version released 1.1.1 |
| 9:33AM |
1 |
.call files in outgoing dont get run |
| 9:27AM |
0 |
ring requested on channel 0/23 already in use on span |
| 9:11AM |
1 |
chan_sip notices |
| 9:09AM |
0 |
voicemail errors |
| 9:01AM |
1 |
oh-323 / Cisco AS5300 problem |
| 7:58AM |
0 |
* found in Iraq!! |
| 7:22AM |
0 |
Anybody knows how to setup chan_misdn incoming calls |
| 7:09AM |
2 |
Everyone-- the scoop on Bison/Flex -- |
| 7:03AM |
0 |
Asterisk @ Home 1.1 Released |
| 6:59AM |
1 |
G.729 with RVA |
| 6:54AM |
0 |
Digium TDM400 Trouble Shooting Tip |
| 6:31AM |
2 |
Simple sip.conf question |
| 6:21AM |
0 |
SIP_CODEC, reinvites, and changing codecs |
| 5:27AM |
0 |
Installation of Asterisk addons 1.0.7 fails (longish) |
| 5:25AM |
1 |
Any ideas on an Interactive IVR? |
| 4:55AM |
1 |
How to use same h323-conf-id in incoming and outgoing legs? |
| 4:47AM |
0 |
PAP2-NA with Panasonic KX-TD1232 CE |
| 4:44AM |
3 |
4 port BRI options ? |
| 3:15AM |
3 |
secretary function |
| 2:50AM |
3 |
911 context, is this right? |
| 2:33AM |
1 |
Asterisk Realtime - How to enable the debug message for SIP users query |
| 2:25AM |
2 |
Inactivity restart segmentation fault |
| 2:01AM |
4 |
Portable USB headset for VoIP |
| 1:17AM |
3 |
Sip UA behind NAT |
| 12:32AM |
0 |
[OT] The Voice of Asterisk |
| 12:11AM |
0 |
Followup: MP3Player cmd issue (for Asterisk OS X users) |
| 12:03AM |
0 |
ISDN Data Calls stop working ? |
| |
| Thursday June 2 2005 |
| Time | Replies | Subject |
|---|
| 11:51PM |
0 |
Connecting Asterisk with Microsoft LCS (Live Communication Server) |
| 11:08PM |
0 |
MP3Player could not play remote stream |
| 9:59PM |
0 |
Newbie MP3Player() cmd questions |
| 9:27PM |
2 |
Asterisk 1.0.7 on Gentoo |
| 9:08PM |
1 |
Teliax is DOWN |
| 8:43PM |
2 |
Ring but now audio on answer |
| 7:36PM |
0 |
IAX2 and Queues Problem? |
| 7:29PM |
3 |
Pricing for DS3000P |
| 7:26PM |
2 |
voip provider request |
| 7:18PM |
1 |
Asterisk RealTime Voicemail Not Working |
| 7:08PM |
0 |
Re: Asterisk-Users Digest, Vol 11, Issue 17 |
| 6:44PM |
1 |
Zaptel not found error during modprobe |
| 6:00PM |
1 |
iax went away |
| 4:18PM |
2 |
Announce: Asterisk virtual configuration |
| 4:13PM |
0 |
what about dCap certification? |
| 3:54PM |
1 |
How to disable Digium card ? |
| 2:54PM |
3 |
CLUELESS NEWBIE needs help making an outbound sip call to PSTN |
| 1:06PM |
2 |
asterisk sipura and g726 codec |
| 1:03PM |
0 |
Host Authentication Problems |
| 12:42PM |
1 |
DID Routing over SIP |
| 11:09AM |
0 |
application sdp message and not answering call |
| 10:59AM |
0 |
Call Manager & Asterisk for VM - MWI not working |
| 10:46AM |
1 |
compile asterisk |
| 10:27AM |
0 |
Asterisk connecting to nortel CS 1000 as sip trunk Need help with final piece (incoming call) outgoing works. |
| 9:51AM |
3 |
asterisk on internet sip phone behind nat - doessomeone even have this working |
| 9:39AM |
1 |
asterisk on internet sip phone behind nat - does someone even have this working |
| 9:21AM |
2 |
bison/flex version warning |
| 9:19AM |
5 |
2 incoming lines and Asterisk@home... |
| 9:14AM |
1 |
Replacing SIP Trunking With IAX Trunking |
| 8:51AM |
3 |
ProSLIC 3210 version 2 is too old. |
| 8:29AM |
0 |
FW: Help with Kpn e1 settings please |
| 8:28AM |
0 |
connect to SIP trunk getting unable to create/find channel |
| 7:34AM |
1 |
Does Debian Bristuffed Asterisk work ignore Beronet cards ? |
| 7:32AM |
0 |
connecting to nortel CS1000 (half way there) |
| 7:23AM |
1 |
asterisk like modems access server |
| 5:04AM |
0 |
gsm call-hunting [OT] |
| 4:59AM |
2 |
Call Meeting VS Call Confrence |
| 4:05AM |
0 |
Astricon Europe :: Tutorial Agenda now published |
| 3:51AM |
7 |
a simple call to my girlfriend |
| 3:21AM |
1 |
H323 trunk with cisco gatekeeper |
| 3:20AM |
0 |
Script to test channel bank |
| 3:07AM |
1 |
trunk timing on 2.6.x |
| 2:06AM |
0 |
How to connect to Asterisk to IPTEL.ORG |
| 1:29AM |
1 |
Will my CPU/RAM be sufficient? |
| 12:52AM |
1 |
Newbie :Call Forwarding problem |
| 12:09AM |
0 |
chan_capi + mISDN + Fritz PTP |
| |
| Wednesday June 1 2005 |
| Time | Replies | Subject |
|---|
| 11:48PM |
2 |
SIP or IAX |
| 10:51PM |
4 |
4+ Port FXS Analog Device |
| 8:59PM |
1 |
Voice recognition application - VoIP/Open Source |
| 8:57PM |
0 |
RTP Read too short |
| 8:27PM |
8 |
Asterisk Box as a Router, Firewall and DHCP Server |
| 8:26PM |
1 |
Incoming and Outgoing |
| 7:59PM |
0 |
Legacy PBX -> * -> Voip Calls problems |
| 7:54PM |
0 |
Issue with Not Capturing All Key Presses |
| 7:11PM |
1 |
Re: Obtaining Cisco Firmware painlessly and LEGITIMATELY? |
| 6:48PM |
1 |
Supervised/Attended transfers |
| 6:40PM |
2 |
IAX2 analog telephone adapter |
| 6:38PM |
1 |
does asterisk work with other processors |
| 6:35PM |
2 |
Realtime+IAX2 and RSA |
| 6:20PM |
5 |
Reccomendations for connecting to 3-4 PSTN lines? |
| 6:14PM |
2 |
Does Asterisk Realtime require the use of CVS HEAD ??? |
| 6:11PM |
1 |
CVS HEAD won't compile for me |
| 4:45PM |
0 |
Re: Obtaining Cisco Firmware painlessly andLEGITIMATELY? |
| 4:36PM |
1 |
Newbie Question: HOWTO make outgoing call on SIP account from internal extensions? |
| 4:34PM |
0 |
Areskicc v2 login issues |
| 3:43PM |
0 |
Inject Audio into Existing Call |
| 3:18PM |
4 |
1.0.8 Release Candidate |
| 2:54PM |
2 |
wrong numbers message |
| 2:34PM |
0 |
chan_zap.c error |
| 2:25PM |
5 |
Broadvoice - Customer feedback |
| 2:00PM |
7 |
SNOM 360 extension lights |
| 1:46PM |
0 |
Cannot find module (NET-SNMP-EXTEND-MIB) |
| 1:24PM |
2 |
SetGroup CheckGroup |
| 1:03PM |
1 |
RFC2833 & firewall problems? (16-byte UDP packets) |
| 12:41PM |
2 |
voice-coloring with asterisk |
| 12:34PM |
2 |
A Way to Write DTMF Digits as text to CDR? |
| 12:33PM |
0 |
Cannot receive incoming calls via ISDN |
| 12:12PM |
0 |
Pri restarting randomly (TE110P or TE405P) |
| 11:59AM |
0 |
99% cpu on asterisk with chan_unicall and low traffic |
| 11:54AM |
0 |
Alternate DID |
| 11:22AM |
0 |
tellme hiring VXML |
| 10:15AM |
0 |
TDM400P Channels stop answering after some time . |
| 10:09AM |
4 |
list down? |
| 9:45AM |
1 |
Astcc does not work - no repeat metering |
| 9:30AM |
3 |
DTMF not working |
| 9:27AM |
0 |
astapi memory errors? |
| 9:25AM |
1 |
rxfax problems - cont. |
| 9:20AM |
0 |
Last of the servers forsale cheap |
| 9:10AM |
2 |
ARESKICC - Another issue |
| 9:04AM |
0 |
Segmentation Fautl / Core Dump with G.729 |
| 8:15AM |
0 |
[q] About chan_misdn, latest mISDNuser and asterisk cvs |
| 8:06AM |
7 |
Pass-through |
| 7:51AM |
2 |
MOH Jittery Voice |
| 7:46AM |
0 |
Large installation with Asterisk |
| 7:37AM |
0 |
Setting up a TDM |
| 7:10AM |
1 |
Asterisk Google API applications - $4500 bounties available |
| 6:51AM |
1 |
FW: TellMe pay-as-you-go? - UPDATE |
| 4:43AM |
1 |
R: R: R: R: R: AT-320 + supervised transfer |
| 4:33AM |
0 |
Launching an application from within Asterisk |
| 3:38AM |
0 |
Problem with codec negotiation |
| 3:05AM |
0 |
hang up a SIP channel |
| 2:57AM |
0 |
debugging zap channel |
| 2:54AM |
1 |
send and receive MMS |
| 2:23AM |
2 |
Problems hanging up PSTN line |
| 1:55AM |
0 |
BT101 new firmware problem (1.0.6.3) |
| 1:50AM |
1 |
Unreliable DTMF detection with DISA on incoming Zap channel on bristuffed * and GSM gateway |
| 1:08AM |
0 |
Hardware questions |
| 1:04AM |
2 |
IVR Load |
| 1:02AM |
0 |
When to use 'Answer' and when NOT to... |
| 12:49AM |
1 |
Dynamic IAX Server |
| 12:21AM |
0 |
newbie with kphone and asterisk |