| Monday July 31 2006 |
| Time | Replies | Subject |
|---|
| 11:06PM |
0 |
Re: If you prefer to read this mail list asa forum ... |
| 10:33PM |
0 |
VM integration to panasonic kx-td500 |
| 8:15PM |
1 |
FC4(2.6.16-1.2108_FC4smp) meetme 404 Not Found |
| 7:31PM |
0 |
Selective Router (route based on Caller-ID) configuration |
| 7:03PM |
1 |
Asterisk SIP problems with Nokia E61 |
| 6:45PM |
0 |
SIP response 400 Bad request |
| 4:58PM |
0 |
Disconnection During Incoming Call |
| 3:13PM |
1 |
AGI Scripts and CDR |
| 2:51PM |
1 |
RemoveQueueMember isn't working. |
| 2:33PM |
0 |
Goldmine sip client revisited |
| 2:32PM |
4 |
sip phone networking question [possibly OT] |
| 2:11PM |
3 |
IAX over two T1 connections bad quality |
| 1:32PM |
5 |
MWI from Asterisk to Meridian |
| 1:11PM |
0 |
Do zttest results matter without telephony hardware? |
| 12:03PM |
0 |
Playfile waiting for N digits |
| 11:59AM |
1 |
Automatic deletion of voicemail messages older than N days? |
| 10:14AM |
1 |
Problems with supervised transfer and agents |
| 10:08AM |
1 |
music ring (CRBT) |
| 9:30AM |
1 |
not reaching at the destination number I dialed |
| 7:51AM |
0 |
Call Disconnects |
| 6:34AM |
0 |
Port doubler for 8 port BRI cards |
| 6:12AM |
2 |
Voice mail limit |
| 6:05AM |
1 |
Asterisk Polycom_acd_functions and G729 |
| 6:03AM |
0 |
Can't load ztdummy |
| 4:52AM |
1 |
app background |
| 4:25AM |
1 |
Testers for ISDN AOC (Advice of Charge (Gespraechsgebuehren)) needed |
| 4:15AM |
1 |
Compiling zaptel on CentOS x86_64 |
| 3:41AM |
3 |
asterisk 1.4 download |
| 3:20AM |
0 |
Cisco 2610 RTP port forwarding |
| 3:15AM |
1 |
DNS lookups failing for SIP register |
| 3:10AM |
3 |
Canreinvite and remotely registered devices |
| 2:35AM |
2 |
Voicemail dial pattern from old pbx |
| 2:17AM |
0 |
Postpaid |
| 2:14AM |
0 |
Invalid Conference Number - Meetme Created via FreePBX GUI |
| 1:37AM |
1 |
SIP channel problem |
| 12:22AM |
1 |
Multiple dialing |
| |
| Sunday July 30 2006 |
| Time | Replies | Subject |
|---|
| 11:38PM |
1 |
Disable native bridge between two zap trunks |
| 11:14PM |
2 |
MeetMe recordings in mp3 format. |
| 10:22PM |
1 |
freepbx and a2billing |
| 10:17PM |
0 |
VoiceMail Name Variable in Dial Plan |
| 8:50PM |
2 |
New Asterisk GUI |
| 3:33PM |
0 |
Server for Asterisk PCI |
| 2:59AM |
0 |
Hangup detection with Sangoma A200 in the UK? |
| 2:51AM |
1 |
Zap Problem |
| 2:35AM |
0 |
Error On brdging Call |
| |
| Saturday July 29 2006 |
| Time | Replies | Subject |
|---|
| 11:05PM |
2 |
FYI - first release of alarm response code. |
| 6:02PM |
0 |
voice format changed to 4 |
| 3:37PM |
2 |
Polycom 1.6.7 Firmware Messages Button |
| 11:26AM |
0 |
agentcallbacklogin Asterisk V1.210 and v1.4 |
| 10:07AM |
1 |
How do you recompile individual source modules? |
| 3:07AM |
1 |
where to read stderr.out from an agi script |
| |
| Friday July 28 2006 |
| Time | Replies | Subject |
|---|
| 5:43PM |
1 |
can't retake call after dialing through Zap/E1 wich doesn't answer |
| 5:10PM |
1 |
Asterisk AGI cmd Record |
| 1:56PM |
3 |
AEL2 Looping |
| 1:12PM |
0 |
Hairpin Detection issues |
| 1:02PM |
3 |
need a pointer regarding scripting asterisk |
| 12:45PM |
1 |
Announce queue? |
| 11:49AM |
11 |
VoipNow 1.2.0 Beta |
| 10:30AM |
0 |
Clicking noise when load XP100 zaptel driver (at boot time) |
| 10:26AM |
1 |
SendText() & displaying text messages on a SIP handset's screen |
| 8:53AM |
2 |
Source Directory of ASterisk |
| 8:45AM |
0 |
asterisk cdr shows "FAILED" |
| 8:37AM |
1 |
wav49 for voicemail attachment not playing |
| 8:02AM |
0 |
R: Canreinvite |
| 7:55AM |
0 |
Extending call parking to display park extension on the handset display |
| 7:54AM |
0 |
SMS functionality of bristuff (0.3.0-PRE-1r) with a Junghanns "duo GSM PCI" card |
| 7:34AM |
2 |
One extension to ring on multiple outside lines |
| 6:58AM |
0 |
Which card do you recommend for heavy load application? |
| 6:47AM |
3 |
Asterisk VOIP / Mikrotik |
| 6:40AM |
1 |
Change the from@ using the voicemail.conf |
| 6:18AM |
1 |
Install asterisk-bristuff for Debian Linux |
| 6:17AM |
1 |
sendtext or sip message - where in RFC |
| 5:52AM |
4 |
Grand stream 2000 will not dial *xx |
| 5:04AM |
1 |
Zaptel trunk failed to compile |
| 4:45AM |
0 |
Weird E1 problem |
| 4:41AM |
10 |
cmd DIAL - Who picked up the call? |
| 4:19AM |
1 |
stream file outputs only silence, even with asterisk example gsm file |
| 3:45AM |
1 |
Voicmail Question |
| 3:35AM |
1 |
registration process |
| 2:57AM |
1 |
Transfer call in SIP |
| 2:34AM |
2 |
CDR IP Authorization |
| 1:59AM |
0 |
Sending email after voicemail |
| 1:55AM |
0 |
asterisk+ooh323.. one way audio issue |
| 1:11AM |
2 |
PAP2T always busy on incoming calls with zaptel |
| 12:34AM |
1 |
FreePBX Inbound Route |
| 12:25AM |
4 |
Fritz!Box Fon ATA |
| |
| Thursday July 27 2006 |
| Time | Replies | Subject |
|---|
| 11:12PM |
0 |
asterisk with CSTA using VAIL SIP TIM |
| 11:03PM |
0 |
CSTA support for astersik |
| 9:34PM |
0 |
Asteriskguru switchboard |
| 6:44PM |
2 |
Looking for carrier grade redundant solution |
| 6:41PM |
1 |
Rate engine AGI? |
| 6:18PM |
9 |
Strange behaviour Panasonic KX-TD1232 |
| 4:26PM |
2 |
Trunk transferring? |
| 4:02PM |
1 |
accessing dialplan global variables in agi |
| 3:50PM |
1 |
Asterisk 1.4 Schedule and Features/Changes |
| 2:39PM |
3 |
long distance ethernet & Asterisk |
| 2:06PM |
5 |
Getting no Audio with G729 |
| 1:59PM |
2 |
gxp-2000 configure line appearances |
| 1:43PM |
1 |
IAX2 Connection fails over time... |
| 12:56PM |
1 |
Detecting voicemail from CO on FXO port and passing to H.323 phone. Possible? |
| 12:36PM |
3 |
Anyone tried vitelity? |
| 12:14PM |
0 |
Re: Goldmine SIP client/softphone questions continued: (Dan Elder) |
| 12:02PM |
0 |
DTMF Dial Tone |
| 11:49AM |
0 |
SIP phone w/ 'modem/data' port? |
| 10:54AM |
0 |
Goldmine SIP client/softphone questions continued: |
| 9:41AM |
0 |
OFF-TRACK: Is VOIP -PSTN integration legal inChina |
| 8:09AM |
2 |
SIP client with video??? |
| 6:33AM |
1 |
Linksys SPA-3102 |
| 5:36AM |
1 |
Problem with call receiving (Asterisk+PSTN+Digium TDM04B) |
| 5:35AM |
1 |
playing a sound into a meetme conf |
| 5:26AM |
1 |
Rxfax and squashed TIFF |
| 4:06AM |
3 |
dropping calls in the middle of conversation |
| 3:47AM |
6 |
Manager interface |
| 3:11AM |
3 |
alcatel ip touch 4068 ... sip? |
| 3:07AM |
1 |
Nokia E61/E70 not always answering voip calls |
| 2:40AM |
0 |
Malformed/Missing URL Problem with Cisco Callmanager 4.1 |
| 2:36AM |
0 |
[oh323]FastStart/H245Tunnelling/H245inSetup |
| 2:35AM |
2 |
Mobile SIP Client |
| 2:00AM |
1 |
SV: Sip phone settings set when user registers |
| 1:39AM |
3 |
Sip phone settings set when user registers |
| 1:32AM |
1 |
french promt |
| 1:13AM |
0 |
CDR dest question |
| 12:35AM |
1 |
Multi Asterisk Server to relay call request |
| 12:04AM |
2 |
Reload of wct4xxp without restarting of Asterisk? |
| |
| Wednesday July 26 2006 |
| Time | Replies | Subject |
|---|
| 10:48PM |
1 |
Determining what gets written to the dst field for a CDR |
| 10:43PM |
1 |
OFF-TRACK: Is VOIP -PSTN integration legal in China |
| 9:30PM |
1 |
Top Users in a MeetMe room??? |
| 8:01PM |
0 |
Polycom 501 provisioning : how to secure valueslocated in plein text files |
| 7:22PM |
0 |
playing a sound into a meetme conference |
| 7:03PM |
2 |
Polycom 501 provisioning : how to secure values located in plein text files |
| 6:58PM |
0 |
Developing VoIP with Asterisk (hardphones & softphones) |
| 6:50PM |
1 |
Cisco 7960 Call Waiting Beep |
| 5:06PM |
3 |
HP DL380 and the TE4xxP cards |
| 2:40PM |
5 |
Strange Error when calling |
| 1:09PM |
0 |
problems with IAX, extension recognition and Asterisk 1.2.9.1 |
| 12:49PM |
2 |
Developing VoIP with Asterisk |
| 12:07PM |
0 |
Sip phone settings according to logged in user |
| 11:44AM |
1 |
Sony Ericsson F250m, Sipura 3000 and Asterisk |
| 11:11AM |
0 |
Just bought a Polycom 501 - I feel likemyGXP-2000was better... |
| 10:18AM |
1 |
Which ATA to test T.38 ? What about Linksys 3102 |
| 9:58AM |
1 |
SIP is not working sometimes. IAX is working fine. Why? |
| 9:55AM |
1 |
Dial exited non-zero, only if PSTN/ZAP/E1 hangs up first. not if voip hangs up. |
| 8:24AM |
2 |
2 * servers, IAX connection between to dial extensions across IAX - not working |
| 8:15AM |
0 |
FS: 2 x Asterisk X100M (red) daughterboard cards - brand new. |
| 7:19AM |
2 |
Message waiting question... |
| 6:48AM |
4 |
CSTA support for asterisk |
| 5:50AM |
0 |
wip-300 question on audio dial out with tdm2402e |
| 4:29AM |
3 |
Zip code, city and area codes |
| 2:38AM |
8 |
Ringing timer |
| 2:17AM |
3 |
E1 connectivity question |
| 1:40AM |
1 |
Asterisk with Linksys SPA-3000 |
| 1:32AM |
1 |
Extension planning |
| 12:05AM |
1 |
Fwd: Problem with chan_zap.so |
| |
| Tuesday July 25 2006 |
| Time | Replies | Subject |
|---|
| 11:58PM |
2 |
Queue announcement issues |
| 9:13PM |
2 |
MWI from Octel to Asterisk |
| 8:58PM |
0 |
Just bought a Polycom 501 - I feellike myGXP-2000 was better... |
| 7:04PM |
2 |
odd sound between SIP & IAX clients |
| 6:37PM |
1 |
T.38 call with t38 in original SDP fails |
| 6:33PM |
1 |
Play sounds to the callee and the caller synchronously when call begins |
| 6:06PM |
1 |
Change current working directory to /tmp |
| 3:26PM |
0 |
sounds format |
| 3:24PM |
0 |
How to send a signal via E1/T1 ISDN to asterisk, to ask the call to be moved. |
| 2:34PM |
0 |
PRI died and Asterisk crashed |
| 2:34PM |
3 |
Rookie voicemail user question |
| 2:12PM |
4 |
Sangoma Stops Receiving Calls |
| 2:03PM |
2 |
sip realtime |
| 1:42PM |
1 |
Voicemail Forwarding |
| 12:44PM |
0 |
sdp multipart information nortel |
| 12:25PM |
4 |
PRI vs "Digital Trunk" |
| 11:49AM |
0 |
All Extensions Dropped |
| 11:25AM |
19 |
Caller ID on Transfers |
| 10:53AM |
1 |
transfers from an E1 using 2b-channel or similar anyone? (QSIG?) |
| 10:06AM |
0 |
netstats like command for sip , Is there one ? |
| 9:41AM |
0 |
MoH clicks and pops |
| 9:27AM |
0 |
Call transfer asterisk + with SPA-1001 |
| 8:59AM |
2 |
Connecting branch offices through IPsec tunnel --latency effects? |
| 8:28AM |
2 |
New message |
| 8:27AM |
2 |
SIP and podcasts |
| 8:25AM |
5 |
Connecting branch offices through IPsec tunnel -- latency effects? |
| 8:23AM |
2 |
Recommend hard phone which supports IAX2? |
| 8:23AM |
3 |
Still voice with echo |
| 8:13AM |
1 |
RE: Just bought a Polycom 501 - I feel like myGXP-2000 was |
| 8:02AM |
1 |
FW: IP CDR |
| 7:47AM |
0 |
IAX2 Variables |
| 7:43AM |
3 |
vegastream 50 FXO DTMF Problem |
| 7:36AM |
1 |
G729 License to Bridge calls through VOIP provider? |
| 7:19AM |
3 |
One way "screech" or tone |
| 6:15AM |
0 |
IAX ATA with FXO |
| 3:18AM |
0 |
SIP user deny and permit for calls through Asterisk |
| 3:12AM |
6 |
Binary/unreadable configuration files? |
| 2:45AM |
0 |
Double Ring on Asterisk 1.2.x (fwd) |
| 2:21AM |
0 |
Conference help |
| 1:19AM |
0 |
Re: FW: meetme application doubt |
| 1:01AM |
1 |
Force peer source ip |
| |
| Monday July 24 2006 |
| Time | Replies | Subject |
|---|
| 11:53PM |
2 |
TDM01B -1 FXO card not working. |
| 10:32PM |
1 |
Asterisk/GPL and G.729 licensing |
| 10:28PM |
0 |
kernel: Error! while loading wct4xxp module |
| 10:22PM |
1 |
Just bought a Polycom 501 - I feel likemyGXP-2000 was better... |
| 7:52PM |
1 |
Unicall reload problem |
| 5:50PM |
3 |
Voice with echo |
| 2:02PM |
1 |
ERROR 1045 (28000): Access denied for user |
| 2:02PM |
0 |
sipbuddies realtime fields and latest documentation |
| 1:56PM |
0 |
sms on wifi phones |
| 1:55PM |
3 |
Just bought a Polycom 501 - I feel like my GXP-2000 was better... |
| 1:47PM |
2 |
RDNIS and IAX2 |
| 1:40PM |
1 |
create custom cdr's |
| 12:51PM |
1 |
Urgent source code changes needed |
| 12:03PM |
2 |
Goldmine CRM softphone + asterisk |
| 11:31AM |
0 |
Zap DMTF detect error |
| 11:24AM |
3 |
Polycom_acd_functions SIP trouble |
| 10:42AM |
0 |
Lots of Asterisk child processes |
| 9:59AM |
0 |
core dumps when phpagi script ends? |
| 9:47AM |
2 |
Asterisk Realtime Macros |
| 9:46AM |
2 |
Intercom feature on Polycom phones |
| 8:41AM |
3 |
G729 Softphone |
| 8:29AM |
0 |
Odd SIP timeout |
| 8:20AM |
3 |
Clocking Multiple T1 Cards |
| 7:59AM |
0 |
Asterisk and Vigortalk problem |
| 7:58AM |
3 |
Circuit/channel Congestion |
| 7:38AM |
9 |
Transfers - No ringback or moh |
| 7:37AM |
4 |
Operator in Voicemail |
| 7:33AM |
1 |
reboots itlself |
| 7:30AM |
1 |
AstLinux 0.4.2 Released |
| 7:29AM |
0 |
Astrisks compatable cards |
| 7:24AM |
0 |
playback / stream file |
| 6:54AM |
2 |
How to receive a phone call each time you receive an email ? |
| 6:50AM |
1 |
Asterisk, IAXModem and Hylafax |
| 6:24AM |
1 |
H.323 an IAX |
| 6:16AM |
1 |
asterisk extra sounds: what for? |
| 6:06AM |
0 |
PRI got event: HDLC Abort (6) on Primary D-channel of span 1 (fwd) |
| 6:05AM |
0 |
compain |
| 5:57AM |
0 |
Asterisk and Phonesystems ... |
| 5:20AM |
1 |
Connecting Asterisk to a Metaswitch |
| 5:01AM |
5 |
Voicemail not sent via email |
| 4:11AM |
2 |
Regular expression problem |
| 4:10AM |
4 |
Mitel 3300 + * |
| 3:16AM |
0 |
Multiuser and analog port |
| 2:35AM |
0 |
Transfering Problem |
| 1:29AM |
0 |
Asterisk Jobs.com Updates |
| 1:14AM |
8 |
overlapdial and DID not always working |
| 12:09AM |
1 |
Solution init.d scripts for CentOS 4.3 |
| |
| Sunday July 23 2006 |
| Time | Replies | Subject |
|---|
| 10:08PM |
1 |
Missing close quote in CallerID breaks SIP. . .workaround? |
| 10:01PM |
0 |
MeetMe in Realtime |
| 9:46PM |
0 |
(no subject) |
| 7:30PM |
4 |
Asterisk autoloading of card modules |
| 4:32PM |
0 |
Problems with freePBX and Fax reception |
| 11:30AM |
1 |
How to connect XLite with another public IP? |
| 7:50AM |
2 |
SIP Woes |
| 5:52AM |
1 |
G726 codec softphone |
| 1:05AM |
0 |
Request for some help.... |
| |
| Saturday July 22 2006 |
| Time | Replies | Subject |
|---|
| 11:24PM |
1 |
Trouble configuring TDM400P on Dell SC420 |
| 7:23PM |
0 |
SNOM missed call. |
| 4:20PM |
3 |
newbbie question |
| 3:45PM |
3 |
Operator Console(s)/Shared Call Appearances |
| 1:00PM |
6 |
Asterisk Dial Plan to Play Message |
| 12:02PM |
2 |
X100P clone not working |
| 8:47AM |
0 |
SIP reinvite _and_ NAT |
| 6:47AM |
1 |
Upgrading my office - Need help |
| 3:00AM |
1 |
cannot received calls in pstn line |
| 2:34AM |
0 |
NAT and externip problem or bug |
| |
| Friday July 21 2006 |
| Time | Replies | Subject |
|---|
| 10:06PM |
1 |
Cyberdata paging speakers - anyone use them? |
| 4:49PM |
2 |
Information about Softphone support G729 ? |
| 3:37PM |
1 |
Digium TE110P IRQ |
| 3:20PM |
0 |
China |
| 2:43PM |
0 |
Yate client |
| 2:26PM |
5 |
Associate manager events to a previous Originate action |
| 1:13PM |
0 |
Weird Hold Problem |
| 12:57PM |
0 |
MFCR2 Patch |
| 12:53PM |
2 |
Invalid module format (ztdummy) |
| 12:08PM |
0 |
AGI record_file |
| 11:36AM |
7 |
Germany VOIP provider |
| 10:24AM |
1 |
[OT] Windows softphone with handset support? |
| 9:37AM |
1 |
Error in ubuntu dapper |
| 9:24AM |
7 |
ftp setup for Polycom phones |
| 8:38AM |
3 |
How to connect 2 AAH |
| 8:23AM |
8 |
Sipura ATA's Forwarding PSTN Calls to Asterisk |
| 7:49AM |
2 |
Queue Persistence with queue.log |
| 7:43AM |
0 |
MySQL question |
| 7:20AM |
1 |
did sometimes not working |
| 7:07AM |
0 |
Asterisk internal extensions caller ID |
| 7:02AM |
5 |
question about asterisk DB |
| 5:49AM |
0 |
problem with iax -> sip across 2 asterisks |
| 5:36AM |
0 |
help for SPA-2100 |
| 5:28AM |
0 |
Transfering a caller_in_queue to a conference room |
| 5:26AM |
0 |
I: ooh323c - cdr problem |
| 4:51AM |
1 |
asterisk-1.2.9 / chan-oh323.so |
| 4:02AM |
1 |
How to connect XLite with public IP? |
| 3:47AM |
1 |
Problem with NAT |
| |
| Thursday July 20 2006 |
| Time | Replies | Subject |
|---|
| 11:47PM |
0 |
Voicemail volume patch |
| 8:46PM |
0 |
Asterisk / Avaya 70XX |
| 8:35PM |
10 |
Typical Asterisk Company |
| 7:53PM |
9 |
If you prefer to read this mail list as a forum ... |
| 7:26PM |
0 |
Re: OT: Project Management & Collaboration Software |
| 7:16PM |
1 |
Asterisk dead-air issues with Digium TE110P and IVR/meetme/internal directory- |
| 7:10PM |
0 |
realtime function |
| 4:41PM |
1 |
OT: FOP examples |
| 4:08PM |
0 |
Asterisk and PacketCable PROJECT |
| 3:39PM |
4 |
A very lost newbie. |
| 2:43PM |
2 |
Asterisk fails to register, when the full logging is turned on |
| 1:56PM |
0 |
Re: [Asterisk-java-users] A newbie introduction |
| 1:40PM |
2 |
Source Clock |
| 12:46PM |
1 |
Overriding # at the end |
| 12:30PM |
0 |
Re: asterisk-users Digest, Vol 24, Issue 116 |
| 12:30PM |
1 |
Re: asterisk-users Digest, Vol 24, Issue 116 |
| 12:27PM |
3 |
Unicall, not HOW but WHY |
| 12:04PM |
1 |
Interested in IVR information |
| 11:47AM |
1 |
Cisco 7960 - automated send DTMF digits after dialing? |
| 10:43AM |
1 |
Aastra 9133i w/NAT and Asterisk |
| 9:49AM |
0 |
regexten / Realtime WAS DUNDI / regcontext |
| 9:45AM |
8 |
Redundant Ethernet |
| 9:10AM |
0 |
Polycom IP301 and Queue questions, deployed environments |
| 9:01AM |
1 |
Agent Attended Transfer Without DTMF |
| 8:55AM |
0 |
PRI channels filling up |
| 8:44AM |
0 |
Problem handling agents and queues vía RealTime |
| 8:05AM |
2 |
Two phone numbers, one SIP provider |
| 8:02AM |
0 |
Calls waiting announcement with two or more queues? |
| 7:50AM |
1 |
Fast busy after one digit dialled? - 7970 SIP 8.0.3 |
| 7:05AM |
0 |
agentcallbacklogin (logging out of) |
| 6:26AM |
3 |
ACD Queues Agents logout |
| 6:10AM |
7 |
*****SPAM***** Load balenced (ADSL) network connections, is it possible? |
| 5:33AM |
1 |
Has anybody in here created their own softphones? |
| 5:16AM |
1 |
Voismart GSM - no billsecs |
| 4:51AM |
0 |
Has anyone programmed their own user\client software for asterisk? |
| 4:06AM |
1 |
Automating the registration process |
| 4:05AM |
2 |
Writing own applications for asterisk - read CALLERIDNUM |
| 3:57AM |
1 |
meetme application doubt |
| 3:53AM |
1 |
Macro help needed!!!! |
| 2:31AM |
1 |
all call forward |
| 1:10AM |
1 |
setting call-limits |
| 12:23AM |
2 |
IP CDR |
| |
| Wednesday July 19 2006 |
| Time | Replies | Subject |
|---|
| 11:48PM |
0 |
!! Got a UA, but i'm in state 1 |
| 11:41PM |
0 |
[Fwd: [Fwd: polarityswitch: no ringback]] |
| 10:30PM |
0 |
Polycom Silent ring |
| 9:29PM |
0 |
question about function realtime |
| 8:17PM |
2 |
Unicall in Australia |
| 6:29PM |
1 |
Help with sip debug? |
| 6:25PM |
0 |
Asterisk process run amock |
| 5:20PM |
0 |
Realtime, ODBC Voicemail, and multiple asterisk servers? |
| 4:45PM |
1 |
RE: [asterisk-dev] How to send DNIS(B-party number) in IAX trunk |
| 4:37PM |
0 |
RE: [asterisk-dev] How to send DNIS(B-party number) in IAX trunk |
| 3:27PM |
3 |
Warm transfer issues in 1.2.10 |
| 3:10PM |
1 |
RE: $3,000 server |
| 2:37PM |
1 |
OH323 registration with gatekeeper problem |
| 2:19PM |
1 |
SIP Registration conundrum |
| 1:52PM |
0 |
SipAddHeaders Question |
| 12:40PM |
1 |
Identifying invoking party for a feature |
| 12:39PM |
0 |
Server locking up again |
| 12:13PM |
1 |
Is dmtfmode used/valid in iax.conf contexts? |
| 11:17AM |
1 |
MoH from Sound Card: Does it actually work? |
| 11:09AM |
0 |
inbound sip rtcp hangup |
| 10:39AM |
1 |
Can't get blind transfer to work |
| 10:28AM |
1 |
Stuck ACD Agents |
| 9:58AM |
0 |
Asterisk patches for packetcable |
| 9:49AM |
6 |
Simple But important question (for me) |
| 9:34AM |
1 |
SV: Queue hold position in other language? |
| 9:16AM |
0 |
asterisk core dumps on a Sipura forwarded to a queue/moh |
| 9:15AM |
1 |
Zaptel Problem - Unable to create channel of type 'Zap' |
| 8:48AM |
1 |
Queue hold position in other language? |
| 8:45AM |
2 |
Zap channel faxing in or out fails but phone calls work. |
| 8:37AM |
0 |
Choppy/Jittery playback at beginning of calls |
| 8:35AM |
0 |
Problems after upgrade asterisk |
| 8:27AM |
1 |
Callback: Dial(dummy) 10 seconds rining without costs? |
| 6:27AM |
5 |
Don't Hit # after 9 to get PSTN line |
| 6:05AM |
0 |
emulating key system - pick up so and so on line1 |
| 4:59AM |
2 |
QueueMetrics 1.2.1 released today |
| 4:48AM |
0 |
-- Going to extension s|1 because of immediate=yes, but immediate is 'no' |
| 3:19AM |
1 |
Issues with MeetMe |
| 3:12AM |
1 |
Finding far end echo in Verizon network |
| 3:11AM |
0 |
Dynamic Queue Members never called |
| 2:53AM |
1 |
BudgeTone BT-102 not registering to Asterisk |
| 2:50AM |
1 |
QuadBRI + TDM + GSM hangup problems |
| 2:34AM |
1 |
Issue with g729 codec |
| 1:33AM |
1 |
Unicall libmfcr |
| 1:14AM |
1 |
Alternative (?) ways to handle G.729 and annexb |
| 1:10AM |
1 |
Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat? |
| 1:05AM |
1 |
Zaptel Compilation Error |
| 12:46AM |
4 |
header replacement |
| 12:19AM |
0 |
Uni Call |
| |
| Tuesday July 18 2006 |
| Time | Replies | Subject |
|---|
| 11:21PM |
1 |
Keep Zap Channel from answering |
| 10:58PM |
8 |
Please suggest me Best VoIP Service Provider |
| 9:57PM |
0 |
FW: How to send DNIS(B-party number) in IAX trunk |
| 6:48PM |
1 |
Problem with MFCR2 |
| 5:28PM |
0 |
zapata.conf pri |
| 4:32PM |
0 |
AW: Using dproxy to solve "no DNS hangs everything"problem? |
| 4:22PM |
2 |
Using dproxy to solve "no DNS hangs everything" problem? |
| 3:11PM |
1 |
Install H323 |
| 2:04PM |
1 |
Polycom 601 and Paging |
| 1:37PM |
1 |
Macro call uniqueid |
| 1:34PM |
1 |
ISDN Protocol |
| 1:09PM |
1 |
Hints to help me debug cdr_odbc not inserting |
| 1:05PM |
1 |
Serveremail Setting Does Not Work for Text Messages |
| 12:24PM |
3 |
emulating key system - pick up so and so on line 1 |
| 12:02PM |
1 |
Astribank? |
| 12:00PM |
0 |
Ignore This |
| 11:50AM |
0 |
SPA-2000, Asterisk 1.2.4 & Incoming call success? Anyone? |
| 11:36AM |
1 |
Error: Dropping incompatible voice frame |
| 11:10AM |
1 |
Tf.voipmich.com - Broken? |
| 11:01AM |
0 |
rxfax Got hangup |
| 10:20AM |
0 |
Net::CSTA on CPAN |
| 10:08AM |
2 |
extensions.conf 4 digit dialing question |
| 9:51AM |
3 |
Asterisk 1.2.7.1 Crashing |
| 9:41AM |
1 |
Examples of handeling input from phones with PHP |
| 9:22AM |
0 |
call-limit and problem with freezy phones. also freezy zap channels with x101p card. |
| 8:46AM |
2 |
Reload clears queue stats |
| 8:27AM |
0 |
Buch Bekanntmachung: Der Weg zu VoIP Asterisk von A bis Z |
| 7:39AM |
1 |
PAP2 TUI Configuration Menu |
| 7:28AM |
3 |
GSM gateway flooded cell - how to detect? |
| 7:03AM |
0 |
Asterisk Trunk Name Problem |
| 6:49AM |
0 |
External call press 1 |
| 5:46AM |
0 |
Reinvite and NAT -> Problem |
| 5:40AM |
0 |
ooh323c - cdr problem |
| 5:40AM |
0 |
Called party cannot hear caller |
| 5:16AM |
0 |
Other phone continues to ring when pick up a call with *8 on SVN HEAD |
| 4:16AM |
0 |
GSM Module not picking up DTMF digits from VOIP FXO Gateway |
| 3:30AM |
2 |
CentOS 4.3 and Zaptel-1.2.7 |
| 3:16AM |
1 |
realtime oracle dialplan select |
| 3:06AM |
0 |
usage of ast db |
| 2:35AM |
0 |
Asterisk v/s other Telephonic plants |
| 2:17AM |
0 |
how to enable users on other iax server call my iax users |
| 1:56AM |
6 |
call forwarding to mobile phone |
| 1:33AM |
1 |
SIP ATA Channels for outbound calls - How to select in dialplan |
| 1:28AM |
0 |
Forward call |
| 1:15AM |
2 |
don't hear start/begin of voiceprompts |
| 12:19AM |
1 |
link quality is poor |
| 12:05AM |
3 |
polycom 601 manual config? |
| |
| Monday July 17 2006 |
| Time | Replies | Subject |
|---|
| 8:15PM |
0 |
app-callforward |
| 6:40PM |
0 |
Hardware/Software suggestions, Supermicro 6024H-TR? |
| 6:29PM |
4 |
PRI and Asterisk |
| 4:55PM |
1 |
Dlink DVG 1120S/Asterisk VoIP to PSTN |
| 4:42PM |
1 |
phpagi problem |
| 4:26PM |
1 |
asterisk 1.2.9.1 and spandsp and rxfax |
| 3:31PM |
1 |
Asterisk H323 and Alcatel 4400 |
| 3:26PM |
1 |
Voicemail and Polycom ip301 |
| 3:13PM |
1 |
Two security holes fixed in latest versions of Asterisk |
| 2:52PM |
0 |
Unable to find extension in context '' |
| 2:24PM |
1 |
RE: asterisk-users Digest, Vol 24, Issue 86 |
| 1:03PM |
1 |
Extensions Register but don't ring when called, can call others though |
| 11:30AM |
2 |
Passing Variables with IAX |
| 11:29AM |
0 |
Queue Penalty |
| 11:19AM |
1 |
show channels |
| 10:31AM |
1 |
an ATA with lamp support |
| 10:14AM |
0 |
Call information on blind transfers |
| 10:10AM |
2 |
Setvar=var=val in sip.conf |
| 10:00AM |
2 |
ooh323c - cdr |
| 9:40AM |
2 |
How many users on an asterisk box behind a dsl can you have |
| 9:06AM |
1 |
Cisco 7960 SIP 8-3-0 |
| 8:36AM |
0 |
Cisco 7960 SIP 8-3-0 getting "Got SIP response 400" |
| 8:35AM |
0 |
R: R: Called number on ISDN |
| 8:29AM |
0 |
Queue Transfers |
| 8:25AM |
0 |
MOH With Asterisk Controlled Transfers |
| 8:03AM |
0 |
Current radius patches |
| 7:56AM |
0 |
One extension can transfer internal calls, can't transfer incoming external calls |
| 7:25AM |
1 |
What is ZapRas used for ? |
| 7:25AM |
2 |
can no more compile zaptel !!! |
| 6:17AM |
1 |
[Fwd: where is the error?] |
| 5:46AM |
0 |
asterisk sending connects when it shouldn't (is there a q931-INFORMATION equivalent in IAX2 ?) |
| 3:12AM |
0 |
question ast db |
| 3:03AM |
1 |
asterisk sending connects when it shouldn't |
| 2:16AM |
4 |
problems to call brazil from germany |
| 1:24AM |
1 |
DTMF in QUEUES dont work |
| 1:22AM |
2 |
Parked calls |
| |
| Sunday July 16 2006 |
| Time | Replies | Subject |
|---|
| 11:05PM |
6 |
Testing 911? |
| 10:05PM |
3 |
zaptel on dual processor, How? |
| 10:02PM |
2 |
Sphinx and Asterisk Integration, How? |
| 6:51PM |
1 |
Polycom phone cycles between UNREACHABLE and REACHABLE |
| 6:18PM |
2 |
Vicidial + Unicall mfcr2 |
| 4:15PM |
5 |
Polycom IP301 and Queues |
| 3:31PM |
3 |
Regression testing dialplan changes |
| 1:54PM |
1 |
Queue RoundRobin |
| 1:29PM |
0 |
Automation of call initiation |
| 11:36AM |
1 |
Setting a threshold for asterisk to take ZAP line off hook ? |
| 9:54AM |
1 |
Injecting prerecorded audio into active call |
| 9:41AM |
1 |
sending flash using DTMF |
| 8:56AM |
1 |
OT: Skype protocol cracked? |
| 8:08AM |
1 |
7970 SIP configs |
| 1:53AM |
4 |
SRTP enabling |
| |
| Saturday July 15 2006 |
| Time | Replies | Subject |
|---|
| 6:13PM |
1 |
How to create or test tone configuration to include them in zaptel |
| 2:59PM |
2 |
DUNDI / regcontext |
| 1:00PM |
1 |
compiling zaptel 1.2.7 error: stray '\194' in program |
| 5:22AM |
4 |
PRI dropouts - solution I hope... |
| 4:18AM |
1 |
Blog about asterisk and voip techology |
| 1:28AM |
1 |
Manager action "hold" missing? |
| |
| Friday July 14 2006 |
| Time | Replies | Subject |
|---|
| 8:40PM |
1 |
Tough time getting Polycom phones to register after router reboot |
| 8:18PM |
3 |
SIP configuration by group |
| 7:19PM |
0 |
489 Bad Event |
| 6:04PM |
2 |
PRI dropouts |
| 3:33PM |
0 |
THOR-2 support |
| 3:21PM |
14 |
Hitting # to Transfer out of a Queue |
| 3:14PM |
7 |
Asterisk 1.2.10 and Zaptel 1.2.7 released! |
| 2:20PM |
0 |
Can not check voicemail from outside |
| 2:11PM |
0 |
Transferring out of Queues |
| 2:01PM |
2 |
Clearing variables in the dialplan? |
| 1:50PM |
0 |
Update for trunk? |
| 1:33PM |
1 |
ATCOM's AG-188 |
| 1:28PM |
1 |
Polycom - simpler transfers? |
| 1:13PM |
4 |
Snom 300 headset with static noise |
| 12:11PM |
0 |
Install Asterisk on VPS |
| 10:43AM |
0 |
Linksys SPA941 - low Static Noise? or some parameter in hands |
| 10:16AM |
4 |
Polycom config file location |
| 8:14AM |
0 |
Transfer ACCEPT followed by DECLINE |
| 8:09AM |
1 |
Cisco Gateway & CallerID Name |
| 8:07AM |
3 |
"Legacy" analog data modems and Asterisk |
| 8:04AM |
0 |
Caller ID on a Sangoma |
| 7:36AM |
1 |
Can incoming alternate rings be discriminated? |
| 6:59AM |
0 |
SIP-> H323 |
| 6:44AM |
2 |
Contacts for Chan_gsm_bt maintainer? |
| 6:34AM |
2 |
Again on ISDN - MSN in Italy |
| 6:34AM |
3 |
R: Called number on ISDN |
| 5:55AM |
1 |
Called number on ISDN |
| 5:07AM |
1 |
Call queue drops call after 1 min |
| 4:01AM |
4 |
asterisk + centos 4.3 |
| 3:57AM |
2 |
astbill + mysql 5 |
| 3:54AM |
0 |
Nokia Primicell and Asterisk ? Hangup and Answer detection ? |
| 3:47AM |
1 |
billed calls when cellullar phone is unreachable |
| 2:15AM |
0 |
ACD rejected calls with out going to Voicemail |
| |
| Thursday July 13 2006 |
| Time | Replies | Subject |
|---|
| 7:56PM |
2 |
need a pointer about scripting asterisk |
| 7:42PM |
1 |
No ringing on outgoing SIP calls. |
| 6:35PM |
0 |
Faxing over CCM SIP trunk to asterisk |
| 5:20PM |
0 |
SPA-3000 XML Config File |
| 5:10PM |
0 |
CT3 cards |
| 4:35PM |
1 |
DUNDi 'Unable to Find Key' |
| 2:44PM |
1 |
Wrong account code from iax_buddies |
| 1:43PM |
1 |
Asterisk instances on VPS |
| 1:29PM |
2 |
FW: Are FreePBX Extensions not being created in asterisk? & FOP question. |
| 1:18PM |
0 |
Delay on ring after dial Out |
| 12:37PM |
0 |
Extensions not busy showing as busy |
| 12:28PM |
0 |
Voicemail Getting Cut Off after 5 seconds |
| 11:04AM |
0 |
SIP adapters questions |
| 10:47AM |
3 |
quad T1 pri |
| 10:11AM |
0 |
Mediatrix 1204 and Asterisk 1.2.9 stops working intermittently |
| 9:50AM |
4 |
How do you harden an Asterisk install? |
| 8:44AM |
2 |
New York city Asterisk consultants |
| 7:56AM |
1 |
cdr functions change between * 1.2.4 and 1.2.9.1 (agi) |
| 7:51AM |
1 |
Voicemail & CallerID |
| 7:17AM |
0 |
Asterisk Console Colorization Question |
| 6:53AM |
1 |
Can I register multiple TERMINATORS to a single account on IAX? |
| 5:59AM |
1 |
Connect to 'agi://blablabla' failed: Operation now in progress |
| 4:11AM |
3 |
SIP To: header |
| 3:35AM |
2 |
asterisk dual servers through iax: Accepting UNAUTHENTICATED call |
| 3:02AM |
0 |
H323 implementation |
| 2:55AM |
2 |
Using DUNDi with TrixBox mini HOWTO |
| 2:43AM |
1 |
sending out fax using asterisk |
| 1:45AM |
1 |
Very bad quality withAVMFritz!cardPCIandchan_capi |
| 1:12AM |
1 |
IAX2 vs TDMoE |
| 1:09AM |
2 |
Channel Redirect |
| 1:05AM |
0 |
Cisco 7912 IP Phone - Convert SIP to SCCP |
| 1:03AM |
1 |
CDRTools please help |
| |
| Wednesday July 12 2006 |
| Time | Replies | Subject |
|---|
| 11:17PM |
8 |
priority problem |
| 10:20PM |
2 |
Inc.com Names Mark Spencer of Digium to its “30 Under 30: America’s Coolest Young Entrepreneurs” |
| 8:49PM |
0 |
console/dsp and autoanswer |
| 7:20PM |
6 |
Polycom compatible phone for Asterisk |
| 6:42PM |
0 |
IGNORE: test email |
| 3:59PM |
1 |
Recording/Monitor after xfer |
| 3:32PM |
2 |
unhappy-about-VoIP-providers@googlegroups.com founded |
| 3:17PM |
2 |
DTMF detection and Sangoma cards |
| 2:53PM |
1 |
Cisco 7940 dialplan.xml |
| 1:34PM |
1 |
sip, dbsecret, and dundi |
| 1:28PM |
0 |
ttp question getting connection timeout. |
| 1:16PM |
1 |
Trouble with call file |
| 1:09PM |
1 |
FW: $3,000 server |
| 12:55PM |
0 |
Very OT: For the Record |
| 12:51PM |
4 |
RE: $3,000 server |
| 12:37PM |
0 |
Agent login problem with MP 124 |
| 12:34PM |
0 |
(no subject) |
| 10:39AM |
2 |
FXS adapters and Polycom phones |
| 10:33AM |
1 |
an operational scenario |
| 10:20AM |
1 |
where the bottleneck lies ? (was: Serverredundancy) |
| 10:17AM |
1 |
Exclude a certain route from using a trunk |
| 10:13AM |
3 |
PCMCIA card support |
| 9:34AM |
0 |
Call Parking breaks suddenly |
| 8:06AM |
0 |
Hardware... dimensioning ?? |
| 8:05AM |
1 |
Email notification of voicemail |
| 7:45AM |
4 |
comcast info -- somewhat offtopic |
| 7:11AM |
0 |
Lets All Get Smart... |
| 7:08AM |
0 |
Option D in dial doesnt seem to be working |
| 6:48AM |
0 |
Echo on PRI |
| 6:31AM |
5 |
Asterisk version: 1.2.9.1 or older? |
| 6:29AM |
0 |
waitexten only provides one digit in chan_zap |
| 6:22AM |
3 |
Problem with making outgoing calls |
| 6:15AM |
8 |
1000s of extensions in one context? |
| 5:39AM |
0 |
Automatic Hangup problem on IAX2 communication to Asterisk |
| 5:28AM |
0 |
Problem incoming calls from sipphone/giztmo |
| 4:46AM |
0 |
IVR with LDAP query for phone number and mobile number?? |
| 3:49AM |
2 |
Queue menu |
| 3:34AM |
3 |
Possible polycom_acd_functions BUG |
| 3:12AM |
1 |
Urgent context |
| 2:39AM |
1 |
asterisk + nite affiliates |
| 1:59AM |
0 |
dial plan -- help |
| 1:58AM |
1 |
Polycom ACD, Asterisk, Kernel 2.6 - now SIP does not register |
| 1:30AM |
2 |
IAX2 trunking problems |
| 12:56AM |
0 |
Urgent call forward |
| |
| Tuesday July 11 2006 |
| Time | Replies | Subject |
|---|
| 9:55PM |
0 |
TE110P configuration problem |
| 7:16PM |
1 |
Polycom, TFTP, and DHCP |
| 7:16PM |
0 |
Problem - Can't pickup call |
| 6:27PM |
0 |
register process flow |
| 5:03PM |
0 |
multiple authentication realms |
| 3:40PM |
14 |
NuFone, please send the log file |
| 3:29PM |
2 |
Intercom mode on Polycom and/or SPA9xx |
| 2:54PM |
0 |
taskset with asterisk |
| 2:26PM |
0 |
2 legs and cdr's |
| 2:21PM |
0 |
Question on event AgentComplete of Manager API |
| 1:33PM |
0 |
CDR Call Status |
| 1:26PM |
3 |
Polycom ACD, Asterisk, Kernel 2.6 |
| 1:15PM |
0 |
Inconsistent call detail records |
| 11:51AM |
2 |
So many configuration files! |
| 11:09AM |
0 |
several asterisk servers questions |
| 9:49AM |
2 |
MFC/R2 country and carrier specific protocol variants |
| 9:13AM |
1 |
what single PRI interface, from which manufacturer |
| 8:45AM |
0 |
[announcement] kansas city asterisk user group |
| 8:29AM |
3 |
Issues with making Transfers |
| 8:20AM |
4 |
Asterisk stops abruptly |
| 8:13AM |
0 |
IPKALL direct to asterisk bypassing FWD |
| 8:04AM |
1 |
RE: [Asterisk-video] Asterisk as an MCU |
| 8:02AM |
2 |
How to do load balancing (1:1) with IAX and two different ISPs |
| 7:48AM |
1 |
WARNING[30954]: chan_sip.c:2734 sip_indicate: Don't know how to indicate condition 9 |
| 7:44AM |
2 |
Server Optimization and Load Balancing |
| 7:17AM |
1 |
Yet another problem with incoming SIP calls and 407 |
| 6:40AM |
1 |
Rate or rank ITSP |
| 6:23AM |
4 |
New Asterisk server crashes daily |
| 6:14AM |
0 |
stuck/phantom zap channels |
| 5:59AM |
1 |
Having trouble to receive fax from samsung sf3200 |
| 4:54AM |
6 |
Provider UNREACHABLE |
| 3:57AM |
0 |
WG: CDR ist getting wrong status |
| 3:55AM |
1 |
Anyone out there using Junghanns ISDNguard? |
| 1:41AM |
0 |
SRTP or zrtp |
| 12:47AM |
0 |
sip_poke_noanswer: Peer xxx is now unreachable |
| |
| Monday July 10 2006 |
| Time | Replies | Subject |
|---|
| 11:52PM |
1 |
Asterisk Servers problem? |
| 11:13PM |
1 |
2 NICs; Asterisk receives on eth1 and replieson eth0 |
| 9:31PM |
2 |
2 NICs; Asterisk receives on eth1 and replies on eth0 |
| 8:54PM |
3 |
Text priority labels not working for me |
| 7:29PM |
2 |
Asterisk and NEC NEAX 2000 IPS |
| 6:40PM |
2 |
Problem with GotoIf in dialplan |
| 6:30PM |
8 |
Server redundancy |
| 3:47PM |
0 |
timing sources |
| 1:53PM |
1 |
Blended? |
| 1:31PM |
1 |
Dialing timeouts |
| 12:14PM |
0 |
Keeping stable 1.2.9.1 updated with patches |
| 10:15AM |
2 |
Mutiple Homes one asterisk box |
| 9:30AM |
5 |
OT: 3Com 3C10222 POE 24 Port Ethernet |
| 9:20AM |
7 |
Mandriva 2006 Cooker RPM for Asterisk 1.2.9 |
| 8:40AM |
0 |
I need help patching source |
| 8:25AM |
0 |
multiple calls |
| 8:05AM |
0 |
loading graphic on a Cisco 7960 |
| 8:03AM |
1 |
zaphfc - problem |
| 7:24AM |
1 |
Very bad quality with AVMFritz!cardPCIandchan_capi |
| 7:17AM |
0 |
Sip No Audio Both Side |
| 6:46AM |
1 |
QueuePauseMember(|Agent/) question |
| 6:38AM |
0 |
IAX2 failed to authenticate as priv (DUNDi) |
| 6:37AM |
0 |
Dial command option D(digits) |
| 6:05AM |
3 |
outgoing call problem |
| 5:45AM |
2 |
Unable to configure my DID number |
| 5:41AM |
1 |
Call-limit and internal transfer |
| 5:05AM |
0 |
channel bank log |
| 3:09AM |
3 |
Certain fax types cause problems |
| 2:51AM |
1 |
Which Fax Solution really works on IAX or SIP? |
| 2:50AM |
2 |
Encrypting the Conversation |
| 2:39AM |
0 |
Error on dial_exec_full |
| 1:59AM |
0 |
CDR calls started via AstManProxy |
| 1:31AM |
5 |
AGI tutorials |
| 1:23AM |
1 |
FXS: No ringtone |
| 1:15AM |
0 |
SV: setting up an email to fax with asterisk |
| 12:35AM |
7 |
setting up an email to fax with asterisk |
| 12:07AM |
1 |
spa941 call pickup? |
| |
| Sunday July 9 2006 |
| Time | Replies | Subject |
|---|
| 11:47PM |
1 |
Urgent Upgrade |
| 11:37PM |
7 |
IVR DTMF |
| 10:35PM |
0 |
PRI Random Disconnected |
| 9:33PM |
0 |
spandsp and app_*fax.c |
| 7:28PM |
0 |
How to transfer other sessions |
| 5:04PM |
1 |
NuFone suggests to use Vonage!!!! |
| 2:07PM |
2 |
Global variables and AGI |
| 11:49AM |
6 |
Choppy MOH (Cisco gateway) |
| 11:17AM |
2 |
2 Handsets, Same extension |
| 7:01AM |
2 |
Can one SIP extension be used for two phones? |
| 6:59AM |
1 |
zap and fax |
| 1:22AM |
4 |
What's the story with X10*P FXO cards? |
| |
| Saturday July 8 2006 |
| Time | Replies | Subject |
|---|
| 11:56PM |
1 |
Suggesstion Required |
| 11:52PM |
0 |
packet8 dta 310 power supply question |
| 7:04PM |
1 |
Help with router setup on new asterisk box |
| 1:37PM |
1 |
PHP AGI |
| 1:09PM |
1 |
Freeware sip/iax client windows mobile |
| 12:30PM |
2 |
trouble with * and # infront of a phonenumber |
| 10:45AM |
1 |
setting of volume |
| 7:30AM |
0 |
voicemail realtime and MWI |
| 6:59AM |
3 |
Asterisk with ISDN Fritz PCI card |
| 2:59AM |
1 |
CallerID in UK on TalkTalk - different to BT? |
| 1:29AM |
2 |
Outgoing MSNs and chan_misdn |
| |
| Friday July 7 2006 |
| Time | Replies | Subject |
|---|
| 11:46PM |
1 |
Uninstalling Asterisk? No make uninstall? |
| 7:28PM |
0 |
Play sound mid way through call |
| 6:42PM |
1 |
Disable the flash hook hold capability on a SIP-to-SIP or SIP-to-ZAP call? |
| 5:57PM |
1 |
Asterisk with Analogue cards |
| 3:34PM |
1 |
zaptel errors |
| 12:57PM |
2 |
test tone |
| 12:39PM |
6 |
Fonality vs TrixBox UI |
| 12:12PM |
3 |
prob with debian and chan_zap |
| 11:50AM |
0 |
Re: Feasability of using * for smallappartmentbuilding? |
| 11:44AM |
1 |
Re: Feasability of using * for small appartmentbuilding? |
| 11:39AM |
5 |
[tip]semicolon trouble: System($(sleep 4; cp 1.call out)&) not working, but System($( sleep 4 && cp 1.call out)&) ; ) |
| 11:35AM |
4 |
Voicemails randomly not deleting in 1.2.9.1 ?? |
| 11:33AM |
1 |
Asterisk and NFS |
| 10:52AM |
0 |
Re: Feasability of using * for smallappartmentbuilding? |
| 10:50AM |
2 |
Re: Feasability of using * for small appartmentbuilding? |
| 10:36AM |
2 |
Help with MusicOnHold!!! |
| 10:31AM |
0 |
E1 additional calling party number |
| 9:31AM |
1 |
Metermaid phone compatibility |
| 9:20AM |
2 |
ASTCC: inuse flag still hangs! |
| 9:17AM |
0 |
ASTCC: how can I limit to xxx minutes per week? |
| 9:14AM |
1 |
Incoming Call matching to peer |
| 9:11AM |
2 |
New GTK Gui for Monitoring and Administration |
| 8:47AM |
1 |
Asterisk stops accepting calls |
| 8:36AM |
6 |
Feasability of using * for small appartment building? |
| 8:16AM |
0 |
SIP account not available in queue ringall |
| 7:57AM |
3 |
ztmonitor in numeric mode |
| 7:48AM |
2 |
qozap w/ 1.2.9.1 |
| 7:33AM |
1 |
OT: Sipura SPA-3000 ATA Directing Calls toAsterisk |
| 7:20AM |
0 |
Best method for detecting state of a sip trunk |
| 7:08AM |
4 |
Do you need a licence to connect a Cisco hardphone to Asterisk ? |
| 7:04AM |
1 |
mgcp trouble |
| 6:38AM |
2 |
Test E1 channel |
| 6:28AM |
3 |
Dell PowerEdge 830 |
| 6:15AM |
4 |
IVR - Automatic Attendant database query |
| 5:46AM |
0 |
Multiple issues |
| 5:44AM |
3 |
Problem With Transfering Calls. |
| 4:40AM |
3 |
SV: How to collect Call duration, Dialout Call files? |
| 2:22AM |
0 |
How to collect Call duration, Dialout Call files? |
| 12:11AM |
2 |
Best practices with Asterisk |
| 12:05AM |
0 |
2.6.18 Kernels |
| |
| Thursday July 6 2006 |
| Time | Replies | Subject |
|---|
| 11:33PM |
0 |
sip.conf, extensions.conf |
| 10:57PM |
2 |
menu system - configurator |
| 9:08PM |
0 |
Please ignore ... |
| 4:46PM |
5 |
Help with MusicOnHold |
| 4:35PM |
0 |
Help troubleshooting "deadlocked" Asterisk |
| 4:19PM |
0 |
Dropped Calls Need Help |
| 3:38PM |
2 |
Tadiran Coral IP PBX to Asterisk |
| 3:27PM |
0 |
fxo lines bridged on a new call once! |
| 3:00PM |
2 |
OT: Sipura SPA-3000 ATA Directing Calls to Asterisk |
| 1:57PM |
3 |
NOT logging Callerid/Call Data? |
| 1:55PM |
2 |
asterisk and sip nat problems |
| 12:57PM |
2 |
Zap Channel not hanging up on Telco side |
| 12:44PM |
10 |
for you guys setting up customer offices... |
| 11:58AM |
0 |
How to plot/graph fxotune -d data |
| 11:53AM |
0 |
xlite softphones: Got SUBSCRIBE for extensions without hint. Please add hint to 1001 in context |
| 10:33AM |
2 |
Phones cutting out.....again - PLEASE HELP!! ! |
| 9:58AM |
1 |
audio session start delay |
| 9:57AM |
0 |
Asterisk Home on 64bit? |
| 9:51AM |
6 |
Phones cutting out.....again - PLEASE HELP!!! |
| 8:55AM |
3 |
Cisco SIP Firmware |
| 7:26AM |
1 |
spa941 and sip "bye" |
| 6:38AM |
3 |
Cisco 7941/7961/7971 wont register with asterisk |
| 6:25AM |
0 |
SOLVED: Re: Calling Extensions generates congestion when call answered |
| 6:22AM |
0 |
SOLVED: Re: Extensions dialing but fails on pickup |
| 6:19AM |
4 |
mISDN configuration |
| 6:17AM |
0 |
Using outboundproxy in sip.conf |
| 6:09AM |
0 |
WG: CDR Accounting wrong |
| 6:03AM |
3 |
Invite someone to Conference |
| 4:00AM |
2 |
Sip voip call termination in Nigeria |
| 3:04AM |
2 |
Unable to find good link to configure Polycom 501 with Asterisk (Plz send good link) |
| 3:01AM |
0 |
SIP connections |
| 2:57AM |
11 |
Tired of fax calls... :-/ |
| 1:47AM |
0 |
(no subject) |
| 1:46AM |
0 |
SV: B2BUA Webbased and Click 2 dial apps |
| 1:40AM |
4 |
B2BUA Webbased and Click 2 dial apps |
| 1:32AM |
1 |
Rockwell Modem |
| 12:59AM |
0 |
Polycom with Asterisk |
| 12:01AM |
1 |
control during registration process |
| |
| Wednesday July 5 2006 |
| Time | Replies | Subject |
|---|
| 10:46PM |
0 |
Help! Zap Startup failure: why is libpri not defined ? |
| 10:11PM |
0 |
fax to HP machine |
| 9:01PM |
2 |
Cisco Buddies |
| 8:48PM |
0 |
Echo cancellation doesn't work after inbound calls are transferred to another extension |
| 8:46PM |
0 |
Voicemail Contexts |
| 7:09PM |
1 |
SIP conf |
| 6:25PM |
3 |
buy X100p card in singapore |
| 6:11PM |
0 |
Got Mediatrix 1204 to work! now MWI and Poly com |
| 6:10PM |
1 |
PRI issues with telco access codes |
| 5:31PM |
1 |
sip codec convertion on the fly |
| 4:52PM |
3 |
Any Polycom dealers willing help out? |
| 4:14PM |
0 |
tormenta2 drivers |
| 3:54PM |
0 |
Weird transcoding error (SIP, local channels): sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 64/256) |
| 3:27PM |
1 |
Got Mediatrix 1204 to work! now MWI and Polycom |
| 2:12PM |
0 |
New mailing list: asterisk-speech-rec |
| 2:04PM |
0 |
Zaptel For new TE412P |
| 1:29PM |
0 |
test to see if I can get any message through |
| 1:14PM |
2 |
Possible Bug? |
| 1:03PM |
1 |
Looking for an asterisk guru |
| 1:00PM |
7 |
Asterisk in Seattle |
| 12:57PM |
0 |
strange Fax or modem like tone when tdm400 answers pstn |
| 12:40PM |
0 |
is ooh323 RAS/ASN.1 broken? |
| 12:15PM |
1 |
Caller Prompts in a Queue?? |
| 11:42AM |
0 |
delay and jitter issues.. |
| 10:47AM |
4 |
'sip debug' |
| 10:05AM |
2 |
[Asteirsk-Users]TE110P configuration problem |
| 10:02AM |
1 |
Cisco 7960 Softkey templates |
| 9:55AM |
1 |
CFWD Status with PHP |
| 9:34AM |
0 |
Sangoma A200 and hangup detection with Asterisk. |
| 9:08AM |
1 |
Agent penality for dynamic agents |
| 9:05AM |
1 |
Asterisk UAc / Request-URI |
| 8:41AM |
0 |
Performance of Database Storage Vs Clustered File System |
| 8:35AM |
2 |
Troubleshooting Random PRI disconnects |
| 8:22AM |
0 |
AGI: Channel status |
| 8:11AM |
0 |
meetme issue with high cpu usage and "hung" conference rooms |
| 7:44AM |
1 |
Queues and qview.pl script |
| 7:39AM |
0 |
DEBUG[13314]: Didn't get a frame from channel: SIP/ |
| 7:11AM |
2 |
International Dialing setup in extensions.conf |
| 7:07AM |
5 |
intel vs amd motherboards |
| 6:58AM |
0 |
Extensions dialing but fails on pickup |
| 6:26AM |
0 |
zaptel Disabled echo canceller because of tone (rx) on channel 2 work? |
| 5:24AM |
0 |
ZAP channel for outbound calls. |
| 4:56AM |
0 |
g729.1 + g723.1 codec conversion |
| 4:46AM |
1 |
Bug in chan_sip mysql support and canreinvite? |
| 4:00AM |
1 |
SV: SV: Nokia E61 |
| 3:45AM |
0 |
0000491... |
| 3:18AM |
3 |
Skype gateway |
| 2:29AM |
0 |
Hanging SIP Channels |
| 2:18AM |
2 |
Intel E7220 chipset? |
| 2:12AM |
2 |
SV: HP Proliant server? |
| 1:48AM |
7 |
HP Proliant server? |
| 1:39AM |
0 |
Dynamic realtime with MWI working |
| 12:34AM |
4 |
SV: Nokia E61 |
| 12:05AM |
0 |
Bridging Prob:::I guess |
| |
| Tuesday July 4 2006 |
| Time | Replies | Subject |
|---|
| 11:57PM |
0 |
Asterisk Shutdown !!! |
| 8:53PM |
1 |
tdm04b strange noise when answering calls |
| 8:10PM |
3 |
RE: Is there a search feature? |
| 7:18PM |
2 |
H.264 and Asterik? |
| 4:51PM |
2 |
More g729 calls than licenses? |
| 4:13PM |
0 |
Sample PRI and FXS channel bank zap files for zaptel and asterisk. |
| 2:21PM |
1 |
Sangoma A200 woes |
| 2:03PM |
0 |
vserver with no /dev/tty* how to run "asterisk-c"for a colored CLI? |
| 1:40PM |
1 |
Page() command and file playback |
| 1:34PM |
0 |
vserver with no /dev/tty* how to run "asterisk -c"for a colored CLI? |
| 1:10PM |
2 |
vserver with no /dev/tty* how to run "asterisk -c" for a colored CLI? |
| 12:32PM |
7 |
MediatrixclientauthenticationfailedEFAILURE_REASON_AUTHENTICATION |
| 12:09PM |
0 |
Mediatrix clientauthenticationfailedEFAILURE_REASON_AUTHENTICATION |
| 11:53AM |
0 |
Mediatrix client authenticationfailedEFAILURE_REASON_AUTHENTICATION |
| 11:27AM |
0 |
Mediatrix client authentication failedEFAILURE_REASON_AUTHENTICATION |
| 11:25AM |
0 |
please remove the autoresponder |
| 11:12AM |
0 |
Mediatrix client authentication failed EFAILURE_REASON_AUTHENTICATION |
| 10:39AM |
11 |
SOLVED: IAX jitter / clocking problem |
| 10:36AM |
11 |
voip-magazinearticle"UsingDUNDiwithaClusterofAsteriskServers" |
| 10:22AM |
1 |
voip-magazinearticle"UsingDUNDiwithaClusterofAsterisk Servers" |
| 10:19AM |
0 |
voip-magazine article"UsingDUNDiwithaClusterofAsterisk Servers" |
| 10:16AM |
0 |
voip-magazine article "UsingDUNDiwithaClusterofAsterisk Servers" |
| 10:13AM |
1 |
voip-magazine article "Using DUNDiwithaClusterofAsterisk Servers" |
| 10:07AM |
0 |
voip-magazine article "Using DUNDiwithaClusterof Asterisk Servers" |
| 10:03AM |
0 |
voip-magazine article "Using DUNDi withaClusterof Asterisk Servers" |
| 9:59AM |
0 |
voip-magazine article "Using DUNDi with aClusterof Asterisk Servers" |
| 9:55AM |
0 |
voip-magazine article "Using DUNDi with aCluster of Asterisk Servers" |
| 8:26AM |
0 |
Recommendations for best Voicemail application manager? |
| 8:10AM |
2 |
vserver (Debian) - no tty: howto use /usr/sbin/safe_asterisk with "-c" for color CLI? |
| 8:01AM |
0 |
I am looking for a (graphical) statistic program |
| 7:41AM |
0 |
SV: SV: Running 40 active calls (too much för CPU?) |
| 7:36AM |
0 |
how to send flash command from asterisk to old pbx when pressing button on phone |
| 7:35AM |
0 |
SIP <--> H323 RTP Questions (1 WAY Audio only) |
| 7:11AM |
1 |
H323 Asterisk best practices |
| 7:06AM |
3 |
Zaptel 1.2.6 / Upgrade Problem |
| 6:52AM |
0 |
Quintum A400 Configuration |
| 6:31AM |
2 |
Libpri + Zaptel + Asterisk polycom_acd_functions error message |
| 6:30AM |
2 |
Help getting International Dialing setup in extensions.conf |
| 6:05AM |
0 |
Quintum A400 Call Establishment Prob |
| 5:54AM |
9 |
time variable |
| 5:43AM |
4 |
Need help with config-files |
| 4:09AM |
14 |
Does asterisk support outbound fax? |
| 3:49AM |
1 |
Calling Extensions generates congestion when call answered |
| 1:40AM |
3 |
trixbox 1.1 download |
| 1:36AM |
1 |
AW: Putting a call recording into a mailbox |
| 1:17AM |
1 |
Putting a call recording into a mailbox |
| 12:52AM |
0 |
FW: SRTP |
| 12:50AM |
0 |
Qsig-Link * to Meridian 81c |
| 12:49AM |
3 |
SV: Running 40 active calls (too much för CPU?) |
| 12:41AM |
1 |
Running 40 active calls (too much för CPU?) |
| |
| Monday July 3 2006 |
| Time | Replies | Subject |
|---|
| 8:57PM |
0 |
Howto: Gentoo + Hudlite + Scratch Asterisk Install |
| 5:11PM |
1 |
Nokia E61 |
| 1:53PM |
1 |
Trouble Setting Up International Dialing in extensions.conf |
| 11:04AM |
1 |
The Asterisk console on a Dell D820 with Intel High Definition Audio. |
| 8:50AM |
1 |
SV: SV: SV: How to configure NOKIA N70 with Asterisk? |
| 8:12AM |
2 |
TDM Installation error |
| 7:59AM |
0 |
PacketCable and Asterisk |
| 7:14AM |
1 |
can't dial Scotland ... |
| 7:11AM |
3 |
Polycom Soundpoint IP 301 w/ MGCP |
| 7:01AM |
0 |
file.c: Unexpected control subclass '14' |
| 6:25AM |
5 |
flash button on asterisk + legacy pbx system |
| 6:22AM |
9 |
SRTP |
| 6:09AM |
1 |
callwaiting |
| 4:41AM |
2 |
Aastra phones - disable call waiting |
| 3:56AM |
2 |
Queues and annoucements |
| 3:48AM |
2 |
Help with IVR menu. |
| 1:12AM |
1 |
Call waiting using free PBX |
| 12:57AM |
1 |
SV: SV: How to configure NOKIA N70 with Asterisk? |
| 12:44AM |
1 |
Duration for billing |
| 12:30AM |
2 |
SV: How to configure NOKIA N70 with Asterisk? |
| |
| Sunday July 2 2006 |
| Time | Replies | Subject |
|---|
| 11:28PM |
1 |
performance & reliabulity of asterisk voicemail using odbc storage |
| 11:24PM |
0 |
How to configure NOKIA N70 with Asterisk? |
| 8:55PM |
1 |
SIP debug logging |
| 8:41PM |
0 |
What does it mean? |
| 6:50PM |
1 |
Motorola and Asterisk |
| 6:36PM |
1 |
Latest SVN of asterisk-addons doesn't compile |
| 12:42PM |
2 |
how to ask for number to dial and then dial it? |
| 12:12PM |
0 |
H323 to SIP Gateway |
| 11:52AM |
0 |
to.gsm and the.gsm |
| 11:37AM |
0 |
setting cdr userfield in .call file |
| 9:59AM |
3 |
dtmfmode=inband but SDP also indicates rfc2833 |
| 8:44AM |
1 |
channel shows to be in use |
| 8:24AM |
2 |
How to continue after a match in an include |
| |
| Saturday July 1 2006 |
| Time | Replies | Subject |
|---|
| 9:21PM |
0 |
ooh323 svn updated |
| 1:48PM |
0 |
Cant seem to send cidname to snom 320 |
| 12:45PM |
1 |
can't run "cat $filename" inside scripts with system() |
| 8:33AM |
3 |
Nufone Tollfree Port |
| 5:00AM |
1 |
svn trunk and call hold / transfers |
| 3:22AM |
1 |
IVR menus on different DIDs |
| 2:02AM |
1 |
callwaiting in queues |
| 1:33AM |
0 |
Asterisk and HiSax |