| Tuesday January 31 2006 |
| Time | Replies | Subject |
|---|
| 11:51PM |
1 |
Leftover sound on isdn modem channel |
| 11:41PM |
0 |
About Meetme and CDRcustom |
| 11:06PM |
0 |
Asterisk and Fax ? |
| 10:57PM |
2 |
Comedian Mail Wont Take Password |
| 10:12PM |
1 |
T.38 patch instruactions |
| 10:10PM |
1 |
Strange echo phenomenon (double tandem) |
| 9:32PM |
0 |
Cisco Gateway - Context Issues |
| 8:31PM |
0 |
Ast<->Ast: IAX2 error w/no audio |
| 8:07PM |
4 |
Asterisk Registering with SER question |
| 7:58PM |
0 |
problem loading zaptel drivers |
| 7:19PM |
0 |
AirCard GSM and Asterisk PBX |
| 6:37PM |
0 |
continuing the context after caller hangsup |
| 6:19PM |
1 |
playing audio to one of several bridged channels |
| 6:03PM |
0 |
RE: [Asterisk-Announce] Asterisk 1.2.4 and Zaptel1.2.3 |
| 5:47PM |
0 |
RE: [Asterisk-Announce] Asterisk 1.2.4 and Zaptel1.2.3 |
| 5:41PM |
3 |
MOH sourced from a sound card? |
| 4:18PM |
0 |
Help with sip setup because can't receive calls!!!!!! |
| 3:07PM |
1 |
Networking voicemail |
| 2:19PM |
7 |
Teliax - Codec Preference effective? |
| 2:11PM |
0 |
Lucent MAX TNT faxing |
| 1:57PM |
3 |
ZAP <--> sip(polycom301) can not hear each other |
| 1:53PM |
1 |
Test Test |
| 1:19PM |
0 |
feedback on grandstream budgetone |
| 1:08PM |
1 |
Fw: Codec preference selection? |
| 1:00PM |
1 |
Channels Codecs |
| 12:57PM |
4 |
broadvoice?? |
| 12:23PM |
1 |
R2 implementation problem |
| 12:17PM |
1 |
E1 PRI Error: Provider error messages. |
| 12:13PM |
3 |
Linking Asterisk Boxes with Sip |
| 11:54AM |
0 |
Snom 360 Message Waiting indicator |
| 11:49AM |
0 |
idefisk 1.31 - Voicemail Button |
| 11:41AM |
0 |
What is causing this error? |
| 11:04AM |
2 |
Asterisk hangs on 1.2.1 |
| 10:56AM |
2 |
Canadian Termination $0.0039 / Minute |
| 10:31AM |
1 |
RE: Euro-ISDN |
| 10:06AM |
0 |
How to start a playback after the called partypicks up? |
| 9:55AM |
1 |
Asterisk 1.2.1 + TDM400P + fax machine unreliable ? |
| 9:52AM |
1 |
international caller id on UK (BT) PRI |
| 9:20AM |
1 |
Forwarding issue. |
| 9:09AM |
0 |
dialing 2 channelsatthesametimewithdifferentcaller ID number? |
| 9:03AM |
1 |
Voicemail greetings |
| 8:54AM |
0 |
unable to register using SIP |
| 8:23AM |
1 |
dialing 2 channels atthesametimewithdifferentcaller ID number? |
| 8:18AM |
5 |
Polycom IP501 Endless Loop |
| 8:14AM |
0 |
Asterisk 1.2 1 FXO Problem |
| 8:09AM |
1 |
dialing 2 channels at thesametimewithdifferentcaller ID number? |
| 8:02AM |
0 |
dialing 2 channels at the sametimewithdifferentcaller ID number? |
| 7:52AM |
1 |
Polycom IP301: Pass-through ethernet port unusable? |
| 7:23AM |
2 |
Gain adjustment |
| 6:45AM |
1 |
How to start a playback after the called party picks up? |
| 6:35AM |
1 |
Voipbuster incoming |
| 6:20AM |
0 |
newbie dial problem, |
| 5:49AM |
5 |
Queue() with timeout=0 |
| 5:03AM |
3 |
Individual SIP account how to make it Trunk |
| 4:35AM |
2 |
Preventing Asterisk from transfering the call |
| 4:28AM |
1 |
meetme and dtmf |
| 4:25AM |
0 |
information on how to use asterisk for telephony boards other than given ones |
| 4:10AM |
2 |
Asterisk hardware. |
| 4:03AM |
0 |
New GXP-2000 Beta firmware available |
| 3:08AM |
1 |
missing pre pattern matching feature |
| 3:01AM |
3 |
Default value for ASTERISK_VERSION_NUM |
| 2:12AM |
7 |
Interface card for Euro-ISDN (BRI) |
| 2:09AM |
0 |
SV: Set caller id on Swedish PRI (euroisdn) |
| 1:57AM |
1 |
Forward a call from AGI/PHP script |
| 1:38AM |
2 |
R: Kirk IP600 |
| 1:32AM |
0 |
Sharing a dialplan |
| |
| Monday January 30 2006 |
| Time | Replies | Subject |
|---|
| 11:48PM |
0 |
dialing 2 channels at the same timewithdifferentcaller ID number? |
| 11:20PM |
0 |
dialing 2 channels at the same timewithdifferent caller ID number? |
| 11:06PM |
0 |
dialing 2 channels at the same time withdifferent caller ID number? |
| 10:50PM |
1 |
Grandstream Budgetone BT-101 audio problems |
| 10:37PM |
0 |
dialing 2 channels at the same time with different caller ID number? |
| 10:11PM |
0 |
Meetmee weirdness |
| 9:47PM |
2 |
RE: [Asterisk-Announce] Asterisk 1.2.4 and Zaptel 1.2.3 |
| 9:33PM |
0 |
Asterisk 1.2.4 and Zaptel 1.2.3 |
| 7:47PM |
1 |
Asterisk SIP phones to Cisco Unity via CCM4.0SIP Trunk |
| 5:57PM |
3 |
polycom ip601 attendant console |
| 4:10PM |
0 |
mISDN errors on asterisk CLI |
| 3:40PM |
2 |
Asterisk Evening in Melbourne: Feb 2! |
| 2:28PM |
0 |
sip domain |
| 1:51PM |
0 |
load balancing |
| 1:37PM |
0 |
Dlink DVG-3004S ? |
| 12:45PM |
0 |
re: help with redirect from SER |
| 12:43PM |
0 |
Codec preference selection? |
| 12:43PM |
1 |
help with iaxy - one way sound |
| 12:42PM |
8 |
Analog with channel bank - Inbound works, outbound doesn't |
| 11:56AM |
2 |
Cisco 7940 not reading SIP image file |
| 11:40AM |
1 |
Kirk IP600 |
| 11:35AM |
0 |
About Extensions |
| 11:17AM |
1 |
Connecting the two servers |
| 10:23AM |
1 |
Live CD? |
| 9:55AM |
5 |
How many TDM2400P's will a server take? |
| 9:42AM |
1 |
Cant compile asterisk #error "You need newer libpri" |
| 9:34AM |
2 |
Most Popular FREE SoftPhone for Windows |
| 9:34AM |
1 |
Need to recompile * after changing zap echo method? |
| 9:21AM |
4 |
DID over analog? |
| 9:13AM |
0 |
Question on SIP Domains and registration |
| 8:50AM |
0 |
Unable to do anonymous outbound calling |
| 7:43AM |
3 |
Set caller id on Swedish PRI (euroisdn) |
| 7:18AM |
1 |
Manage api- Matching 'Newchannel' event with the 'Originate' command |
| 7:14AM |
1 |
Gateways |
| 7:13AM |
1 |
Help configuring Asterisk server |
| 6:37AM |
1 |
Asterisk and LCS ? |
| 6:26AM |
0 |
backgrounddetect and busy |
| 6:23AM |
1 |
SIP-H323 translation |
| 4:33AM |
0 |
Realtime Queue not realtime anymore in Asterisk 1.2.3?! |
| 4:30AM |
1 |
Playing music while transfering |
| 3:59AM |
3 |
adress book |
| 3:33AM |
1 |
app_snmp |
| 2:42AM |
5 |
Grandstream Budgetone mass deployment? |
| 2:08AM |
0 |
intel 536 EP as x100p clone? |
| 1:59AM |
3 |
How many digium cards per server ? |
| |
| Sunday January 29 2006 |
| Time | Replies | Subject |
|---|
| 9:29PM |
0 |
dialing 2 channels at the same time with differentcallerID number? |
| 9:08PM |
0 |
dialing 2 channels at the same time with different callerID number? |
| 8:41PM |
0 |
Transfer (SIP REFER) - AccountCode not available? |
| 8:38PM |
0 |
Dialogic / Voip Forum |
| 6:29PM |
10 |
Web interface |
| 3:58PM |
2 |
Access Codes |
| 3:40PM |
4 |
Asterisk + XEN does it make sense? |
| 3:00PM |
0 |
Cisco VG200 as FXO for * ? |
| 12:43PM |
1 |
New C7960 won't tftp? |
| 11:40AM |
1 |
Unable to get IP of eth0 |
| 11:36AM |
1 |
HandyTone 488 ata? |
| 11:18AM |
2 |
simulating a few thousand SIP clients? |
| 11:04AM |
0 |
strange performance issue |
| 8:32AM |
2 |
username not stabled? |
| 7:39AM |
4 |
How to remove first ring tone on FXO? |
| 7:01AM |
0 |
Modprobe Zaptel error |
| 6:58AM |
1 |
file.c:509 ast_openstream_full: File 100 does not exist in any format |
| 6:46AM |
1 |
Moprobe Zaptel error |
| 6:16AM |
3 |
Wifi phone set-up |
| 5:49AM |
0 |
Real-time: username |
| 4:11AM |
1 |
changing displayed call info on snom 360 |
| 3:25AM |
1 |
Asterisk as SIP endpoint ? |
| |
| Saturday January 28 2006 |
| Time | Replies | Subject |
|---|
| 8:54PM |
0 |
Adjusting gain, Milliwatt and ztmonitor |
| 7:51PM |
3 |
Urgent: Unable To Execute after updating from SVN |
| 5:29PM |
3 |
Multiple Subscriptions to SIP accounts at Same Domain |
| 4:18PM |
1 |
english snom support forums ? |
| 4:14PM |
2 |
RoadRunner |
| 3:53PM |
0 |
voicetronix FXOs with * ? |
| 3:06PM |
0 |
AutoDialing with VOP USING SIPURA 2100'S |
| 2:08PM |
0 |
Help with Music on Hold during transfer |
| 1:42PM |
1 |
Installing the none commercialintelg729codecsinto Asterisk@Home 2.2? |
| 12:03PM |
0 |
How to Unregister? |
| 11:21AM |
1 |
Looking for 150 SIP desktop phones with power over ethernet that will work with Plantronics HL-10 Handset Lifter for Remote Answering |
| 10:28AM |
2 |
Best CoDec for high network latency |
| 9:46AM |
2 |
VOIP carriers and asterisk |
| 9:45AM |
0 |
Other side disconects when using TxFAX |
| 8:51AM |
1 |
Can't send DTMF transfer code from called SIP phone |
| 8:00AM |
0 |
Re: 5, 000 concurrent calls system rollout question |
| 7:01AM |
0 |
Re: Lockups since upgrade 1.2.3 - anyone else? Anyideas? |
| 6:18AM |
1 |
regarding connecting to AMP>> |
| 5:39AM |
1 |
double ringing tone on asterisk 1.2 (workaround) |
| 2:32AM |
2 |
Trunk is not released |
| 1:34AM |
3 |
No IN and OUT on ISDN line at the same |
| 1:27AM |
3 |
Simple question about ringing multiple phones (extensions)? |
| 1:09AM |
3 |
(Un)PauseQeueMamber usage |
| |
| Friday January 27 2006 |
| Time | Replies | Subject |
|---|
| 10:44PM |
1 |
Installing the none commercial intel g729 codecsinto Asterisk@Home 2.2? |
| 10:13PM |
2 |
Name/username (sip show peers) |
| 10:13PM |
1 |
shared fxo line |
| 10:07PM |
3 |
G729 Commercial Licenses. |
| 9:28PM |
23 |
5,000 concurrent calls system rollout question |
| 8:40PM |
2 |
DTMF's indescipherable, but voice clean! |
| 4:45PM |
1 |
Agent counts |
| 4:38PM |
3 |
sip qualify=yes interval |
| 3:33PM |
0 |
Re: Asterisk-Users Digest, Vol 18, Issue 181 |
| 3:16PM |
2 |
fxo/fxs cards with 8 ports |
| 3:02PM |
2 |
VOXEE Caller ID.. |
| 2:58PM |
0 |
FlashTransfer to Bridge |
| 2:55PM |
0 |
Page() and Asterisk 1.2.3 Problems? |
| 2:46PM |
0 |
moh & clock |
| 2:46PM |
1 |
How's the best way to set up agents... |
| 2:22PM |
4 |
CDR reporting between two Asterisk servers |
| 2:20PM |
1 |
Help with Congestion error |
| 1:45PM |
0 |
Good provider of Polycom Phones (mostly for accessto latest/greatest firmware) |
| 1:35PM |
0 |
starvox communications |
| 12:41PM |
6 |
Lockups since upgrade 1.2.3 - anyone else? Any ideas? |
| 11:48AM |
0 |
SIP channel not diconnecting on hangup |
| 11:35AM |
1 |
chan_bluetooth: successful compile and outbound cell calls: Still tweaking inbound setup. WAS: Cannot compile chan_bluetooth on Asterisk 1.2.1 |
| 10:33AM |
1 |
802.1p |
| 9:45AM |
1 |
SIP incoming calls |
| 9:29AM |
0 |
Digium Wildcard TDM400P call pickup timing |
| 9:24AM |
1 |
No IN and OUT on ISDN line at the same time? |
| 9:19AM |
0 |
Problems with MFC/R2 in Brazil |
| 8:41AM |
5 |
SER redirect |
| 8:26AM |
1 |
Good provider of Polycom Phones (mostly for access to latest/greatest firmware) |
| 8:09AM |
5 |
External IAX2 phone defined as internal behaving as from PSTN |
| 8:06AM |
3 |
OT?: International number parsing |
| 7:16AM |
0 |
Caller Presentation |
| 7:13AM |
7 |
AAH out bound routing problem |
| 6:48AM |
0 |
Newbie SIP trunk question... |
| 6:27AM |
1 |
T38 providers |
| 5:46AM |
0 |
ATA's ??? |
| 5:00AM |
0 |
wcfxo md3200 problem... |
| 4:43AM |
0 |
pb with callerid |
| 4:29AM |
2 |
Spa3k and ISDN |
| 4:13AM |
1 |
Outgoing FXO and CDR |
| 3:48AM |
2 |
WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38 |
| 2:04AM |
0 |
ODBC Problem with voicemail. |
| 1:57AM |
0 |
No matching peer or user based on IP address |
| 1:32AM |
0 |
Offtopic: Auto provioning Snom 360 |
| 1:31AM |
1 |
Packeting multiple GSM frames in one IP packet - Help needed. |
| 12:49AM |
3 |
Max concurrent calls |
| 12:23AM |
3 |
paging agi |
| 12:05AM |
0 |
How to put peers into Realtime |
| |
| Thursday January 26 2006 |
| Time | Replies | Subject |
|---|
| 11:58PM |
0 |
Alex Tew interview made possible because of Simon @ Simwood eSMS |
| 11:47PM |
1 |
S100-FX v2.0 |
| 11:42PM |
3 |
Chan_capi on builds 7955>8320 strangeness |
| 8:15PM |
2 |
Shared Line Appearance |
| 8:08PM |
1 |
Current viewpoints on the Sayson/Aastra 480i |
| 7:59PM |
0 |
Using Flash |
| 7:31PM |
1 |
ISAC Codec Support |
| 6:07PM |
4 |
extension to extension dialing |
| 5:55PM |
2 |
PRI restarting each hour? |
| 5:17PM |
0 |
Webphone with Asterisk?? |
| 4:43PM |
5 |
Is Voxee down? |
| 4:30PM |
0 |
Anybody will to share a sm bus polycom dialplan to share? |
| 4:20PM |
1 |
Round Robin Call Distribution |
| 3:43PM |
0 |
Re: OT: Legacy systems / fax |
| 2:36PM |
0 |
To which context a "registered peer" is sent? |
| 2:29PM |
0 |
How set up "call forward on no answer" for allextensions? |
| 2:27PM |
2 |
Transferring Using Flash |
| 2:19PM |
0 |
Hamachi with Asterisk |
| 1:25PM |
2 |
checking voicemail via trunk |
| 1:08PM |
1 |
Manager API mailing list |
| 1:00PM |
1 |
CDR problems |
| 12:43PM |
5 |
Skype-to-Asterisk(SIP): progress |
| 12:33PM |
0 |
Aastra 9112i or 9133i Ring Tones |
| 12:02PM |
1 |
fax detect DID variable |
| 11:41AM |
0 |
Asterisk 1.2.3 CentOS 4.x RPMS <--SpanDSP Ba ckport? |
| 11:29AM |
2 |
Announcement: Snom 360 with integrated XML O bjects |
| 11:23AM |
0 |
Pause/UnpauseQueueMember |
| 11:06AM |
5 |
Asterisk 1.2.3 CentOS 4.x RPMS |
| 11:06AM |
3 |
snom 320 echo problems |
| 11:02AM |
1 |
Announcement: Snom 360 with integrated XML Objects |
| 10:40AM |
0 |
app_background and app_cepstral |
| 10:13AM |
0 |
Local Channel Call Looping |
| 10:00AM |
2 |
Dynamically disabling echo cancellation (Zap). |
| 9:50AM |
1 |
CDR logging in /var/log/asterisk instead of MySQL DB |
| 9:42AM |
0 |
Plea to support a much needed function for Call Centers in Asterisk. |
| 9:38AM |
0 |
[Fwd: Asterisk as an Ascend box] |
| 8:52AM |
6 |
Fail over to Pri on VoIP connection failure |
| 8:31AM |
2 |
Snom360 Sidecar & Asterisk |
| 7:58AM |
6 |
* point to point t1 solution? / alternatives |
| 4:51AM |
0 |
0h323 - one way audio |
| 4:26AM |
1 |
Calls pickup |
| 3:54AM |
2 |
using sangoma cards as a timesource? |
| 3:52AM |
0 |
Good switchboard solution? |
| 2:24AM |
0 |
codec selection based on call prefix |
| 2:08AM |
1 |
TDM400 pinout |
| 1:30AM |
0 |
Missing meetme recordings. |
| 1:02AM |
7 |
Bootable CD? |
| 12:42AM |
3 |
VOIP Router |
| 12:16AM |
1 |
Asterisk Setup Question -- Please Help |
| |
| Wednesday January 25 2006 |
| Time | Replies | Subject |
|---|
| 11:53PM |
20 |
* point to point t1 solution? |
| 10:49PM |
1 |
asterisk 1.2.3 call problem |
| 10:46PM |
5 |
transfer, recording ... |
| 10:24PM |
1 |
Adding number prefix on Polycom SP300 phone |
| 10:19PM |
2 |
Voipbuster/voipstunt -- what a crap service |
| 10:18PM |
0 |
Free calls to UK, US and Germany??? |
| 9:33PM |
0 |
Want to automatically park call and have callerhear ring tones |
| 9:14PM |
0 |
Received fax "offset" in tif file? |
| 8:54PM |
2 |
Changing Asterisk install location... |
| 8:53PM |
0 |
Re: Asterisk-Users Digest, Vol 18, Issue 158 |
| 8:37PM |
0 |
Re: Asterisk-Users Digest, Vol 18, Issue 158 |
| 7:45PM |
3 |
Fast AGI Options. Eeeek! |
| 7:30PM |
1 |
Want to automatically park call and have caller hear ring tones |
| 6:58PM |
1 |
Speech playback getting cut off |
| 6:18PM |
0 |
include from database |
| 5:44PM |
0 |
Monitor and * 1.2.3: Sync issues? |
| 3:23PM |
1 |
Dial String Questions |
| 3:03PM |
2 |
Best FXO hardware for home use |
| 2:37PM |
0 |
asterisk 1.2 with grandstream ht-496 2nd port registration issues |
| 2:04PM |
0 |
Steal with MusicOnHold |
| 1:39PM |
4 |
VoIP in India |
| 1:15PM |
1 |
Disregard: Looking for the .xml file format for idleURL for Cisco 79xx |
| 1:03PM |
0 |
SIP register vs SIP with a fixed IP |
| 12:47PM |
10 |
Asterisk 1.2.3 Released - Critical Update |
| 12:46PM |
0 |
Parking from external PBX |
| 12:31PM |
0 |
SIP re-invites ignored by other end |
| 11:10AM |
0 |
feature transfer on PRI |
| 11:05AM |
4 |
Setting ringtone on Polycoms |
| 10:58AM |
1 |
BroadVoice subscribers and Asterisk 1.2.3 |
| 9:22AM |
0 |
Looking for the .xml file format for idleURL for Cisco 79xx |
| 9:11AM |
0 |
chan ooh323 choppy sound |
| 9:10AM |
0 |
Echo while using Headset with Polycom IP 501 / 601 Asterisk 1.2.1 |
| 8:21AM |
1 |
NEAX 2000 IVS Integration |
| 7:55AM |
0 |
Polycom 601 Bricked? |
| 7:47AM |
0 |
Problem in auto dialing through call files |
| 5:58AM |
2 |
Help with sip setup because can't receive calls |
| 5:41AM |
0 |
ISDN / Analog |
| 5:25AM |
1 |
Asterisk + Ericsson PBX |
| 2:59AM |
14 |
No audio? Update your Asterisk |
| 2:49AM |
1 |
ISDN D-channel disconnects for a minute every 5 minutes |
| 2:49AM |
5 |
trunk to trunk forwarding |
| 2:10AM |
0 |
need an bench-marking tool |
| 2:08AM |
0 |
dailplan questions |
| 1:22AM |
0 |
(no subject) |
| 1:15AM |
1 |
ACT-P104S SIP Firmware |
| 12:52AM |
1 |
jitterbuffer causes no sound? |
| 12:49AM |
2 |
Suddenly No audio |
| 12:25AM |
0 |
Very low audio levels after asterisk answers inbound calls |
| |
| Tuesday January 24 2006 |
| Time | Replies | Subject |
|---|
| 11:55PM |
0 |
(no subject) |
| 7:27PM |
0 |
Re: Anyone using verizon fios ftth foranalogvoice?Any echo? |
| 5:55PM |
0 |
Re: Anyone using verizon fios ftth for analogvoice?Any echo? |
| 5:51PM |
0 |
Asterisk and IP Aliases |
| 5:34PM |
0 |
Astbill and Wholesale |
| 4:53PM |
0 |
Including files in AEL file |
| 4:40PM |
3 |
Linksys SPA-941 multiple line appearences |
| 4:14PM |
1 |
E1 -> T1 native bridging for fax, will it work? |
| 4:04PM |
0 |
No ringing |
| 2:57PM |
1 |
Hunting for DIDs in Kenya/Nigeria |
| 2:51PM |
0 |
SIP call failover |
| 2:38PM |
0 |
analog channels answer detection anything new in1.2.X |
| 2:15PM |
1 |
AAH 2.0 fax problems continued |
| 2:00PM |
1 |
Paging HardPhones |
| 1:05PM |
5 |
Looking for Q.Sig success story |
| 12:52PM |
1 |
oh323 and asterisk v1.2.2 |
| 12:38PM |
3 |
ZAP - Can't pickup calls on Analog Trunk |
| 12:18PM |
1 |
Mini frame before first full voice frame (IAX) |
| 12:08PM |
4 |
which gui for asterisk on web |
| 11:54AM |
0 |
Disable music on hold per user |
| 10:38AM |
1 |
CAPI crash/lockups? |
| 10:31AM |
1 |
Re: Anyone using verizon fios ftth for analog voice?Any echo? |
| 9:50AM |
1 |
Call Parking - Set ID on return |
| 9:36AM |
0 |
OT: testing email routing |
| 9:12AM |
0 |
How to keep Asterisk (1.2) out of the media path |
| 8:36AM |
0 |
pulsedial on fxo signalling |
| 8:35AM |
2 |
txfax application problem |
| 8:26AM |
0 |
PhpAgiTutrial |
| 8:07AM |
13 |
Nortel Meridian Opt 81C and PRI |
| 7:54AM |
0 |
Microsoft Office Communicator 2005 as SIP client? |
| 7:26AM |
4 |
Asterisk with SuSe 10 |
| 7:15AM |
0 |
Help compiling bristuff on FC3 |
| 6:54AM |
2 |
Re: Asterisk-Users Digest, Vol 18, Issue 134 |
| 6:54AM |
1 |
cannot change distinctive ring polycom phones |
| 6:34AM |
0 |
Nortel IP2000 |
| 5:51AM |
0 |
Re: Asterisk-Users Digest, Vol 18, Issue 144 |
| 5:48AM |
0 |
Problem: have no RTP streams from Asterisk |
| 5:01AM |
6 |
iax provider |
| 4:35AM |
0 |
H.264 and AAC codecs |
| 4:09AM |
1 |
Voipbuster problem |
| 3:36AM |
3 |
Simple setup ... |
| 3:16AM |
5 |
Is it possible ? |
| 2:39AM |
1 |
iaxphone for ubuntu 5.10 |
| 2:17AM |
1 |
suggest a gsm router |
| 2:13AM |
8 |
UK Provider |
| 1:47AM |
1 |
MOH begin behavior |
| 1:44AM |
0 |
Anyone using verizon fios ftth for analog voice? Any echo? |
| 1:20AM |
0 |
What happens to global and channel variables? |
| 12:58AM |
1 |
need help asterisk and AS5300 |
| |
| Monday January 23 2006 |
| Time | Replies | Subject |
|---|
| 11:18PM |
3 |
MOH Server |
| 10:18PM |
0 |
Jumping on the asterisk bandwagon |
| 9:03PM |
2 |
weird zttest result |
| 6:46PM |
14 |
Polycom 501 horrible echo |
| 6:42PM |
3 |
Config File Storage |
| 6:01PM |
0 |
asterisk fax to pdf, blank pdfs? |
| 5:19PM |
0 |
DTMF not working on overseas cellphone calls |
| 4:48PM |
1 |
chan_capi - B3 Error |
| 4:24PM |
3 |
canreinvite always =no * no matter what we try :-( |
| 4:18PM |
0 |
Firewall/Embeded System/CF/etc |
| 3:52PM |
1 |
OFF TOPIC: Core router upgrade for a voip colocation center |
| 2:40PM |
2 |
Newer version of Zaptel with 1.0 branch of * |
| 2:26PM |
2 |
Polycom phones and dynamic IP for NAT |
| 2:09PM |
4 |
make linux26 |
| 2:07PM |
0 |
Help with bad audio using MPC.. |
| 1:48PM |
1 |
Asterisk SIP phones to Cisco Unity via CCM 4.0 SIP Trunk |
| 1:34PM |
2 |
Fw: setting outgoing caller ID by the queue an extension is logged into |
| 1:26PM |
1 |
background SayDigits()? |
| 1:21PM |
3 |
SPA-3000 - the party's over :-( |
| 1:06PM |
0 |
user not seen |
| 1:01PM |
2 |
analog channels answer detection anything new in 1.2.X |
| 12:47PM |
1 |
Video Conferencing. |
| 12:29PM |
1 |
Installing the none commercial intel g729 codecs into Asterisk@Home 2.2? |
| 12:27PM |
5 |
dial out and message playback |
| 12:19PM |
0 |
Call Waiting CallerID |
| 11:47AM |
1 |
Answering Service Add-on? |
| 11:44AM |
0 |
Odd asterisk behavoir |
| 11:40AM |
0 |
H.323 videoconferencing with asterisk? |
| 11:26AM |
3 |
How to view Q.931 Disconnect code |
| 11:08AM |
2 |
Home Test! |
| 10:48AM |
0 |
Polycom videoconferencing with asterisk? |
| 10:20AM |
0 |
Problem with Codecs |
| 10:16AM |
7 |
G729a Pass-Through and Recording/Monitoring |
| 10:06AM |
1 |
SIP over TCP: latest news? |
| 9:58AM |
3 |
Announcing PodMail 1.0 (GPL) |
| 9:31AM |
1 |
not able to start asterisk |
| 7:55AM |
0 |
openH323 from cvs |
| 7:30AM |
1 |
Testing List (JUST A TEST) |
| 6:47AM |
1 |
debug with ser |
| 5:30AM |
0 |
Sip Extensions |
| 4:59AM |
0 |
SIP response 300 "Multiple choice" ??? |
| 2:38AM |
0 |
Xlite set-up program |
| 2:26AM |
1 |
How to set-up LCR |
| 1:53AM |
0 |
Re: Asterisk-1.2.1.tar on Suse Linux 9 (Atif Nadeem) |
| 12:53AM |
5 |
Bug in attended transfer or as expected? |
| |
| Sunday January 22 2006 |
| Time | Replies | Subject |
|---|
| 11:16PM |
0 |
changing agent passwords without reloading asterisk |
| 9:45PM |
0 |
Forwarding out to cellular phone's voicemail with AMP |
| 8:42PM |
1 |
SNOM 190 Daylight Savings |
| 8:33PM |
3 |
how to set caller id? |
| 7:56PM |
1 |
macro-faxreceive |
| 7:19PM |
6 |
spandsp Error |
| 7:13PM |
0 |
Finding good, objective reviews of major VoIP phones |
| 4:43PM |
0 |
IP SIP Phone/2.0.6 |
| 4:05PM |
4 |
Detection of Answering Machine |
| 2:40PM |
0 |
SIPDiscount inbound number |
| 2:19PM |
4 |
Snom 320 and message retrieve key |
| 12:44PM |
0 |
Interrupting ring to go to voicemail pickup -- How to ring after Answer()? |
| 12:39PM |
1 |
asterisk 1.2.2 and zap channel voice detection |
| 12:11PM |
1 |
Fail over using CHANAVAIL |
| 11:45AM |
1 |
Gen. Question |
| 11:40AM |
3 |
Installing the none commercial intel g729codecs into Asterisk@Home 2.2? |
| 10:29AM |
1 |
Distinctive ring detection using SIP - Broadvoice addon line detection |
| 9:42AM |
1 |
wildcard matching in dialplan |
| 8:52AM |
0 |
Thanks for all your messages |
| 8:04AM |
0 |
RE: Asterisk-Users Digest, Vol 18, Issue 131 |
| 6:42AM |
5 |
T3 Mux and Asterisk Question |
| 6:37AM |
1 |
Installing the none commercial intel g729codecsinto Asterisk@Home 2.2? |
| 5:09AM |
0 |
Asterisk cut offs on TE110P |
| 5:00AM |
1 |
Asterisk TS-1 |
| 4:59AM |
1 |
Gateway TIMEOUT |
| 4:30AM |
1 |
Asterisk-1.2.1.tar on Suse Linux 9 |
| 2:46AM |
2 |
Disposition codes in CDR |
| |
| Saturday January 21 2006 |
| Time | Replies | Subject |
|---|
| 11:52PM |
0 |
Extensions for in-bound faxes w/o properly-provisioned T1. |
| 10:56PM |
1 |
Can you disable Forward on a Polycom phone? |
| 9:21PM |
1 |
Compiling app_cepstral.c into Asterisk - failing |
| 6:14PM |
3 |
cvs asterisk compile failed (newer libpri) |
| 5:54PM |
2 |
Tellabs 2572 EC Photos here. |
| 3:57PM |
2 |
How to disable WARNINGS in CLI |
| 3:36PM |
0 |
[Announce] Mark Spencer interview |
| 3:01PM |
0 |
sip outgoing calls over proxy |
| 2:52PM |
1 |
h323 configuration |
| 2:12PM |
1 |
TE110P + PRI incoming + outgoing extensionsquestion |
| 1:49PM |
0 |
Anyone interested in getting a basic training course together for the greater NYC area? |
| 12:42PM |
1 |
Is sip1.voipbuster.com corking reliably for others on list? |
| 12:30PM |
0 |
Dialstatus Oddity in 1.2 |
| 11:56AM |
1 |
SIP and NAT - best practices? |
| 11:30AM |
1 |
Caller ID and Sipura Router |
| 11:02AM |
4 |
asterisk + usb celular |
| 10:42AM |
0 |
jitterbuffer on zap channel |
| 9:35AM |
1 |
Asterisk 1.2.2 - Double Quote on CallerID Causing SIP Problem (7940) |
| 7:47AM |
0 |
Rotated Logs Per SIGXFSZ every few seconds |
| 5:37AM |
0 |
Providers with jitter buffer |
| 4:53AM |
7 |
MeetMe Dialplan question |
| 3:26AM |
3 |
Asterisk always uses 127.0.0.1 address |
| |
| Friday January 20 2006 |
| Time | Replies | Subject |
|---|
| 10:29PM |
0 |
Activate Call Waiting by default |
| 10:27PM |
0 |
Custom cdr trouble, help this newbie |
| 9:49PM |
1 |
SIP problem picking up the call |
| 9:31PM |
1 |
Need a good extensions.conf sm bus config w/polycom phones |
| 8:56PM |
0 |
Japanese J1 question |
| 7:28PM |
2 |
TE110P + PRI incoming + outgoing extensions question |
| 6:06PM |
0 |
Need a good extensions.conf sm bus config w/ polycom phones |
| 5:10PM |
3 |
OT:Snom 360 prompt for registration pwd? |
| 4:45PM |
1 |
Why is agents.conf not utilized? (aka: can't find good info on agents and queues for AMP) |
| 4:41PM |
1 |
Teliax Down? |
| 4:40PM |
7 |
Asterisk Development and Release Cycle |
| 4:15PM |
5 |
When/whether to use SER? |
| 3:40PM |
1 |
How to Clear SIP Channels |
| 3:26PM |
1 |
Calling MySQL 5 stored procedures from app_mysql |
| 2:56PM |
1 |
SPA-941 auto-answer capability |
| 2:51PM |
0 |
Queues & All Agents Busy |
| 2:51PM |
1 |
applicationmap |
| 2:34PM |
1 |
cisco 7940g, 7960g phone screen sizes? |
| 2:09PM |
1 |
Dell PowerConnect 2724 Switch and QoS for VOIP? |
| 1:48PM |
5 |
Asterisk in SPA9000? |
| 1:46PM |
1 |
AIX calls with sipdiscount |
| 1:40PM |
0 |
sip notify on sipura? |
| 1:36PM |
1 |
HardPhone Dilemma |
| 1:36PM |
2 |
Agressive echo cancelation |
| 1:33PM |
0 |
Problems with incoming PSTN calls |
| 1:28PM |
0 |
Cisco 7912G SIP phone and Asterisk double RTP packets |
| 1:18PM |
1 |
SIP, NAT and Firewalls |
| 1:15PM |
2 |
ztdummy on opteron |
| 1:14PM |
1 |
Hardwiring a Tellabs echo canceller - help req |
| 1:10PM |
0 |
h extension |
| 12:46PM |
0 |
Help with poor audio using SIP |
| 11:58AM |
1 |
Can TE406P provide PRI to other VoIP gateways? |
| 11:47AM |
2 |
Asterisk bounty PRI 2B channel transfer for NI2 PRI line |
| 11:09AM |
2 |
Conversation interrupted by fax |
| 11:02AM |
5 |
iDEFISK (mac iax2 softphone) release |
| 10:49AM |
1 |
How to have a phone ring another extension as soonas off-hook? |
| 10:41AM |
1 |
Dial command not executing following priority when caller hangs up |
| 10:32AM |
2 |
How to have a phone ring another extension as soon as off-hook? |
| 10:13AM |
1 |
quality and delay test |
| 9:43AM |
0 |
Realtime - reading values from registred family name |
| 9:40AM |
0 |
newbie cdr_custom and cdr_csv2 problem, please help |
| 8:50AM |
0 |
Double Progress Tone |
| 8:36AM |
2 |
no nat, but one way only audio (more info) |
| 8:22AM |
1 |
IAX and call transfer |
| 7:53AM |
0 |
can wengophone or gizmo be used directly with asterisk??? |
| 7:45AM |
0 |
multithreading for res_perl |
| 7:24AM |
1 |
2400P Pinouts |
| 7:21AM |
0 |
Mapoing extensions to specific trunks |
| 7:19AM |
1 |
Connecting a TE to a NT BRI isdn |
| 6:52AM |
0 |
R: Dect to SIP PCI card |
| 6:40AM |
2 |
no nat, but one way only audio |
| 6:28AM |
0 |
Asterisk and Cisco GW |
| 6:16AM |
1 |
instant fallback to zap in case of sip/iax/xyz-failure |
| 5:54AM |
0 |
No translator path: iax2 calls not possible |
| 5:51AM |
2 |
AVM C4, asterisk-1.0.8, /etc/asterisk/capi.conf |
| 5:44AM |
1 |
more voicemail frustrations (was: realtimevoicemail) |
| 5:36AM |
0 |
AstTAPI and CallerID Popups |
| 4:50AM |
1 |
SIP phone receiving but not transmitting |
| 4:49AM |
3 |
Detecting a PRI failure from dialplan |
| 4:35AM |
1 |
How to Confiure Voicetronix V4PCI16 in asterisk |
| 3:15AM |
1 |
No congestion |
| 2:55AM |
2 |
'h' in CDR |
| 2:12AM |
0 |
[ANNOUNCE] Asterisk::LCR released on CPAN |
| 1:52AM |
3 |
Dect to SIP PCI card |
| 1:43AM |
1 |
RTCP XR support (RFC 3611) |
| |
| Thursday January 19 2006 |
| Time | Replies | Subject |
|---|
| 11:54PM |
0 |
sipura 3000 help needed |
| 8:41PM |
1 |
Port forwarding on a DLink Di-604 |
| 5:14PM |
1 |
Problems with Module |
| 3:53PM |
1 |
TDM400P zttest not working |
| 3:48PM |
0 |
PRI |
| 3:32PM |
1 |
Cannot compile chan_bluetooth on Asterisk 1.2.1 |
| 3:03PM |
0 |
New astGUIclient/VICIDIAL release: 1.1.9 |
| 2:13PM |
13 |
Polycom FW |
| 1:28PM |
0 |
DTMF not recognized on overseas call from cellphone |
| 1:04PM |
0 |
AudioCodes Unreliable DTMF Detection |
| 12:45PM |
2 |
Dial() Jumping behaviour and Vesrsion 1.2 |
| 12:01PM |
0 |
Fw: chanspy |
| 11:56AM |
0 |
AGI Tx error 510 |
| 10:52AM |
4 |
Disabling zap echo cancellor from dialplan |
| 10:45AM |
1 |
Sound issue with Asterisk |
| 10:40AM |
1 |
Problem with rxfax - Dropping incompatible voice frame? |
| 10:40AM |
0 |
transfer and zap |
| 9:30AM |
1 |
(newbie) using dtmf during a call |
| 9:11AM |
1 |
DTMF # ? |
| 9:10AM |
0 |
Incoming fax on voipbuster |
| 7:23AM |
0 |
Asterisk least cost routing expert needed |
| 6:34AM |
0 |
sipTAPI and usernames |
| 6:24AM |
0 |
Connection pooling |
| 6:23AM |
3 |
Processor Size |
| 6:02AM |
0 |
Loud Tone issue, still having problems |
| 5:33AM |
0 |
Problem configuring Asterisk |
| 3:48AM |
1 |
CDR Accounting Question |
| 3:14AM |
0 |
A problem in recieving voice on one side |
| 3:12AM |
2 |
Brief silences during calls |
| 3:03AM |
1 |
DTMF Simultaneous Inband and RFC2833 performedby Asterisk => Duplicate tones |
| 1:30AM |
0 |
spandsp-0.0.2pre22 not working! |
| 1:13AM |
2 |
Asterisk and Linux-HA |
| 12:36AM |
0 |
Fax sending with asterisk 1.2.1 |
| 12:10AM |
0 |
SoCal Asterisk Users Group Thursday Evening |
| 12:03AM |
0 |
agi_callingpres: 0 |
| |
| Wednesday January 18 2006 |
| Time | Replies | Subject |
|---|
| 11:45PM |
1 |
SIP RTP Negotiation |
| 9:37PM |
1 |
speex in asterisk 1.0.10 |
| 9:12PM |
1 |
bug in Authenticate application ? |
| 9:08PM |
5 |
SMS to fixed phone line |
| 8:13PM |
3 |
asterisk 1.2.2 RPMS for CentOS 4.x |
| 7:34PM |
0 |
IAX2 between two * server not working |
| 7:24PM |
1 |
Sip phone with Bluetooth - does it exist? |
| 7:19PM |
0 |
Asterisk with GnuGK |
| 6:58PM |
0 |
OT: Network Wire Brand |
| 6:43PM |
0 |
Re: Asterisk-Users Digest, Vol 18, Issue 115 |
| 6:33PM |
1 |
Iaxmodem and Efax? |
| 5:58PM |
1 |
festival-script.pl... howto change language? |
| 5:13PM |
1 |
I see Asterisk 1.2.2 into the ftp or was a vision? |
| 5:02PM |
1 |
Asterisk 1.2.2 Released! |
| 4:57PM |
2 |
SipAddHeader bug? |
| 4:56PM |
0 |
RTP error problem |
| 4:18PM |
1 |
Still the LDAP Realtime extension |
| 4:10PM |
0 |
Re: Asterisk-Users Digest, Vol 18, Issue 114 |
| 3:35PM |
1 |
chan_sip.c:5262 sip_reg_timeout Probably a DNS error for registration |
| 3:10PM |
1 |
Polycom 301 DTMF |
| 3:06PM |
0 |
O'Reilly's Etel Conference |
| 2:33PM |
1 |
DTMF Simultaneous Inband and RFC2833 performed by Asterisk => Duplicate tones |
| 2:31PM |
2 |
modem simulation |
| 2:00PM |
4 |
sipura ata 3000 UK (BT) CAllerid |
| 1:26PM |
1 |
Bugs that Need Your Input! |
| 1:02PM |
0 |
Call Waiting CallerID not showing up |
| 12:26PM |
5 |
SAN Devices |
| 10:54AM |
1 |
Disconnecting call 'SIP/X.X.X.X-085340d0' for lack of RTP activity in 11 seconds |
| 10:36AM |
0 |
Atcom AT320: SIP or IAX? |
| 10:22AM |
2 |
1.2 in production w/100+ phones? |
| 10:13AM |
0 |
misdn svn |
| 9:48AM |
0 |
Problem with Vonage and Asterisk, Please help me |
| 8:53AM |
1 |
detect when grab up the phone |
| 8:47AM |
1 |
Dial Rules in localprefixes.conf |
| 7:57AM |
1 |
Web Conferencing |
| 7:44AM |
2 |
Asterisk Sound Issue |
| 7:33AM |
2 |
CALLERIDNAME/CALLERIDNUM Deprecation |
| 7:30AM |
0 |
PING users of Manuel Guesdon's LDAP extensions |
| 6:32AM |
0 |
asterisk 1.2 bristuff and sms |
| 6:20AM |
1 |
LDAP direct authentication Problem |
| 6:10AM |
0 |
Force Port Number on INVITE |
| 5:16AM |
0 |
Asterisk Fax part 2 |
| 4:32AM |
1 |
Australian Asterisk Job Listing |
| 3:49AM |
0 |
Problem with DIAX and Asterisk and Vonage |
| 3:00AM |
0 |
CPU utilization in general |
| 2:49AM |
1 |
PRI D-channel errors |
| 2:48AM |
0 |
get only GHOST fax |
| 2:48AM |
1 |
SpanDSP not sending to fax extension. |
| 1:54AM |
0 |
rtcachefriends and REALTIME + MWI |
| 1:20AM |
0 |
SIP IP Phone is not registering [urgent] |
| 12:12AM |
1 |
Attended transfer reconnect when goes to voicemail? |
| |
| Tuesday January 17 2006 |
| Time | Replies | Subject |
|---|
| 11:00PM |
0 |
Problem with Asterisk and DIAX, Please help me |
| 10:41PM |
1 |
Qwest can't/won't |
| 7:25PM |
0 |
A few straightforward questions about 1.2 |
| 7:00PM |
0 |
Loud Tone When Key Pressed |
| 6:50PM |
0 |
ruby-agi 1.1.0 released |
| 6:33PM |
0 |
rx/txgain per device? |
| 5:07PM |
2 |
MeetMe Listen Only flag (|m) |
| 4:55PM |
2 |
How do you deal with subprefixes with LCR? |
| 4:39PM |
0 |
Line transfering calls back to asterisk system from another pbx |
| 4:21PM |
2 |
chan_sccp crashes Asterisk on startup |
| 4:21PM |
0 |
How to compile and install just one module? |
| 4:18PM |
3 |
[Asterisk-Dev] WAS: click-to-call cleint NOW: XML Manager I/F str aw poll |
| 3:46PM |
1 |
Make SIP calls go back out a zap trunk |
| 2:55PM |
2 |
Re: Choosing an FXO card, Asterisk-Users Digest, Vol 18, Issue 100 |
| 2:36PM |
1 |
Asterisk and Fax part 2 |
| 2:35PM |
1 |
Asterisk & Genesys integration |
| 1:41PM |
1 |
Voicebroadcasting with Asterisk? |
| 1:25PM |
1 |
Slightly OT: Plantronics headset quick connectorwiring |
| 1:07PM |
2 |
idefisk 4 linux now available for download |
| 1:02PM |
1 |
red alarm? |
| 12:34PM |
0 |
Includes affecting menu, zaptel transfers |
| 12:05PM |
0 |
RE: Building from scratch would like the benefit of (TOO LONG...) |
| 11:35AM |
2 |
Problem with ISDN HFC-S card |
| 11:10AM |
1 |
Asterisk LDAP Authentication Problem |
| 11:08AM |
0 |
asterisk.ctl limitations |
| 11:04AM |
0 |
RE: Building from scratch would like the benefit of (TOO LONG...) |
| 10:41AM |
3 |
Fritz card technology & German * |
| 10:22AM |
4 |
How to find out if a new voicemail exists |
| 10:07AM |
0 |
1.2.1 can´t register with SIP-Provider, 1.0.9 could |
| 9:54AM |
0 |
call fails first time, then succeeds |
| 9:51AM |
6 |
nwebmail |
| 9:44AM |
6 |
OT: DCAP Certification |
| 9:31AM |
0 |
Verizon DTMF Recognition |
| 9:03AM |
1 |
ACD announce-holdtime |
| 8:52AM |
0 |
Slightly OT: Plantronics headset quick connector wiring |
| 8:22AM |
2 |
auto load SIP peers on startup |
| 7:32AM |
0 |
Problem with installation of rpm's, Please, help me. |
| 7:28AM |
2 |
Building from scratch, would like the benefit of everyone's experience |
| 7:24AM |
0 |
SIP hardphones with xml/html/xhtml/microbrowsersupport? |
| 6:46AM |
2 |
Is Asterisk the right tool? |
| 6:44AM |
2 |
Problem configuring Asterisk, Please help me |
| 6:18AM |
0 |
FYI - Cisco IP Phones SYN Flood Device Reload Vulnerability |
| 4:44AM |
3 |
experiences with teliax, voipjet or junction networks? |
| 4:40AM |
1 |
Call Center sofphone |
| 3:56AM |
0 |
SVN Compile Error |
| 3:39AM |
3 |
Phone still rings while on a call |
| 3:36AM |
1 |
Asterisk under SUSE 9.2/VMWARE 5.5.1 |
| 3:00AM |
1 |
Is there a key sequence to stop a call as its ringing? |
| 2:10AM |
2 |
IAX/SIP and openser problem. IAX bug? |
| 1:54AM |
1 |
Hold on with Asterisk Manager |
| 12:53AM |
0 |
Possible Job |
| 12:01AM |
0 |
Asterisk RELAY |
| |
| Monday January 16 2006 |
| Time | Replies | Subject |
|---|
| 10:47PM |
0 |
Quadra software - Changing to Opensource |
| 10:44PM |
3 |
CVS HEAD chanisavail not working for sip channel? |
| 10:14PM |
1 |
Problem with installation of rpm's, Please help me. |
| 9:04PM |
1 |
RTP redirect system usage |
| 7:00PM |
2 |
question about zttest |
| 6:45PM |
1 |
Incoming call: Got SIP response 503 "Server error" back from xxx.xxx.xxx.xxx |
| 5:55PM |
5 |
SIP hardphones with xml/html/xhtml/microbrowser support? |
| 5:50PM |
1 |
modify a cdr values.. |
| 4:59PM |
2 |
automon - one touch record |
| 4:56PM |
1 |
I've sent a message to the list 6 hours ago and it's still not showing up |
| 4:41PM |
1 |
setting Cisco 7940 to factory default |
| 4:36PM |
2 |
Agents getting logged off agressively |
| 4:08PM |
2 |
Call Center and Predictive dialing |
| 4:01PM |
1 |
cisco 7940 firmware upgrade |
| 2:22PM |
0 |
Looking for Full Time technicians |
| 2:20PM |
0 |
asterisk 1.2.1 crashed |
| 1:39PM |
1 |
Zapata.conf and Realtime |
| 12:46PM |
5 |
Dundi Examples |
| 12:41PM |
1 |
TE210P Trade |
| 12:36PM |
1 |
making wakeup feature call phone number, not extension? |
| 12:23PM |
0 |
FW: Exited non-zero |
| 12:20PM |
0 |
Asterisk RTP Bridging |
| 11:53AM |
1 |
IAX voice distortion with full upload channel /SIP ok |
| 11:10AM |
0 |
Set(LANGUAGE()=language throwing warnings |
| 11:05AM |
2 |
New RPM packages for CentOS4.0 |
| 11:02AM |
2 |
MeetMe greeting message. |
| 10:47AM |
1 |
Support for RFC3323? |
| 10:30AM |
2 |
Problem with calls starting from a legacy PBX |
| 10:03AM |
0 |
How to put someone on hold with Astersik Manager |
| 9:43AM |
2 |
ztdummy inaccuracy on linux-2.6 |
| 9:36AM |
1 |
Asterisk for Call Center (missing reference) |
| 9:26AM |
0 |
Asterisk for Call Center |
| 8:34AM |
0 |
Asterisk with Cisco |
| 8:33AM |
2 |
AGI variables |
| 7:57AM |
3 |
Max Number of #include statements |
| 7:56AM |
4 |
new in asterisk world |
| 7:43AM |
2 |
Pickup Button |
| 7:33AM |
0 |
FW: confirmation |
| 7:28AM |
0 |
strange voicemail issue |
| 7:25AM |
2 |
cmd Dial parameters |
| 7:20AM |
0 |
OT: ignore me, just a test |
| 7:13AM |
1 |
Dynamic conference - add participants |
| 6:51AM |
2 |
agi debug - unable to set normal priority |
| 5:02AM |
3 |
asterisk down because of cdr |
| 4:27AM |
3 |
distorted native music on hold |
| 4:05AM |
4 |
problems with a pri (E1) |
| 3:28AM |
1 |
chan_capi-cm and DID |
| 3:23AM |
0 |
Pre-made E1 crossver cables for the UK |
| 3:16AM |
1 |
Test to see if I'm still on list... |
| 2:58AM |
0 |
SIP Error 401 Problem |
| 2:37AM |
0 |
dnid |
| 1:20AM |
0 |
asterisk1.2.1/PRI-E1 outbound call issues |
| 12:07AM |
0 |
Zapata.conf Realtime? |
| |
| Sunday January 15 2006 |
| Time | Replies | Subject |
|---|
| 7:20PM |
6 |
uplink call quality issues |
| 5:18PM |
2 |
Choosing an FXO card |
| 1:37PM |
1 |
Oooh / ahhh . . . 5 tellabs boards on ebay. |
| 9:06AM |
3 |
MoH trouble with latest bristuff (0.3.0-PRE-1f) |
| 7:40AM |
2 |
RX/TXgain on bristuff/zaptel ? |
| 6:25AM |
2 |
Save the Quintum before I throw it out a window.... |
| 5:16AM |
3 |
Detecting Long PDD |
| 12:51AM |
0 |
passing user information problem |
| 12:10AM |
2 |
zaptel echo canceller preload patch |
| |
| Saturday January 14 2006 |
| Time | Replies | Subject |
|---|
| 11:35PM |
0 |
codec_gsm.c:194 gsmtolin_framein: Invalid GSM data |
| 11:00PM |
1 |
No "native bridge" on outbound SIP channels |
| 10:54PM |
3 |
Reducing echo on FXS port |
| 8:56PM |
4 |
Ugly echo cancel, with Bristuff/Zaphfc |
| 7:59PM |
3 |
SIP RTP |
| 4:41PM |
0 |
Mediatrix windows-based setup? |
| 4:20PM |
2 |
Advice Of Charge (AOC) ? |
| 4:16PM |
4 |
tuning an x100p in Australia for echocancellation |
| 4:10PM |
4 |
echo tail stats |
| 12:24PM |
0 |
oh323 h245 tunneling not working |
| 10:36AM |
2 |
1.2.1 "Silence suppression is disabled" whatthehell? |
| 10:22AM |
1 |
I need feed back on how an Aastra VentureIP 4FXO |
| 9:55AM |
0 |
RE: read.what else to do ? |
| 9:26AM |
3 |
1.2.1 "Silence suppression is disabled" what the hell? |
| 8:38AM |
1 |
call file result |
| 8:37AM |
1 |
Problem with just one number! |
| 8:11AM |
0 |
RE: Mediatrix Unit Manager Express needed |
| 7:07AM |
3 |
rxgain/txgain test numbers in Germany? |
| 3:47AM |
2 |
IAX voice distortion with full upload channel / SIP ok |
| 2:03AM |
4 |
"Catch all" extension |
| 1:48AM |
0 |
class 5 softphone |
| |
| Friday January 13 2006 |
| Time | Replies | Subject |
|---|
| 11:03PM |
0 |
Asterisk/Zaptel 1.2 |
| 9:46PM |
2 |
"auto fallthrough" hangup on 1.2.1 |
| 8:14PM |
1 |
linksys pap2 automatically connect on liftinghandset |
| 8:09PM |
1 |
CALLERIDNUM::3 do not working on 1.2.1 |
| 7:50PM |
0 |
Extensions.conf error - 'MaximumInclude level(10) exceeded' |
| 7:30PM |
1 |
linksys pap2 automatically connect on lifting handset |
| 6:18PM |
0 |
configuring asterisk to send and recieve fax using hylafax |
| 5:50PM |
9 |
loading zaptel drivers automatically upon reboot |
| 5:36PM |
1 |
tuning an x100p in Australia for echo cancellation |
| 5:07PM |
2 |
Extensions.conf error - 'Maximum Include level(10) exceeded' |
| 4:46PM |
0 |
Extensions.conf error - 'Maximum Include level (10) exceeded' |
| 3:22PM |
1 |
SIP NOTIFY on REALTIME USERS/PEERS |
| 2:57PM |
4 |
PHPAGI daemon/background task? |
| 2:50PM |
1 |
uip200 transfer calls |
| 2:40PM |
1 |
ZAP Digit Timeout |
| 2:17PM |
2 |
ILBC to G711 transcoding experince ? |
| 1:58PM |
0 |
SIPDiscount credit card details |
| 12:24PM |
2 |
zapata.conf for non pri T1? |
| 12:14PM |
1 |
[Fwd: SipDiscount END-OF-LIFE announcement for old IAX2/SIP servers] |
| 10:57AM |
1 |
MINNESOTA: TwinCities Asterisk Users Group - Saturday 01/14/2006 |
| 10:34AM |
1 |
queus & agents |
| 10:23AM |
1 |
Re: <Ben Higley> Can you send us your AGI CDR logging application? |
| 9:54AM |
2 |
AEL2 -- The Future -- |
| 9:40AM |
1 |
TDMoE - best signalling method? |
| 9:39AM |
0 |
NOTIFY authentication |
| 9:21AM |
2 |
Asterisk echo & fxotune |
| 8:28AM |
1 |
Calls through madiatrix with incorrect disposition |
| 8:16AM |
3 |
FastAGI Command Execution |
| 7:41AM |
2 |
Use Grandstream ATA as trunk |
| 6:58AM |
0 |
Low Speed Software Modem |
| 6:36AM |
1 |
double ringing tone on asterisk 1.2 |
| 5:59AM |
1 |
pause between queue calls for agents |
| 5:19AM |
1 |
Cepstral in AGI problem |
| 4:56AM |
0 |
R: RE: RE: Spandsp |
| 3:39AM |
1 |
Agent monitoring - join files |
| 3:05AM |
0 |
How to configure (make) Fax over IP with T38 and Asterisk? |
| 3:00AM |
1 |
dnid support? |
| 2:26AM |
0 |
resinstall |
| 2:15AM |
0 |
Variable |
| 1:38AM |
0 |
Voicemail indication fails |
| 1:20AM |
2 |
X-web Lite |
| 12:54AM |
2 |
PrimuX Cards with chan_capi-cm |
| |
| Thursday January 12 2006 |
| Time | Replies | Subject |
|---|
| 11:30PM |
3 |
Asterisk Prepaid Solution |
| 10:07PM |
0 |
Configuration of SIP Mysql peers. |
| 9:14PM |
2 |
Random Disconnects |
| 9:14PM |
0 |
SOLVED: SIP phones can't pick up incoming call on analog (PSTN) trunk - signalling problem? |
| 7:50PM |
0 |
SIP phones can't pick up incoming call on analog trunk - signalling problem? |
| 6:10PM |
0 |
Re: Asterisk-Users Digest, Vol 18, Issue 77 |
| 4:54PM |
3 |
D-Link announces Asterisk on Router/DSL-Modem |
| 3:45PM |
0 |
Sending commands to Asterisk via FastAGI |
| 3:42PM |
1 |
Why can Asterisk Auto Attendant pick up on firstring? |
| 3:38PM |
0 |
latest openh323...still compile error |
| 3:34PM |
0 |
latest pwlib...still compile error |
| 2:42PM |
0 |
trigger event on asterisk warning/error |
| 2:33PM |
1 |
PRI and QSIG |
| 2:03PM |
0 |
safe_asterisk not working? |
| 2:03PM |
1 |
spandsp and page orientation |
| 2:01PM |
3 |
linksys SPA-941 |
| 1:56PM |
2 |
interfacing w/ a legacy InterTel PBX |
| 1:24PM |
5 |
[Announce] Web-MeetMe v2.0.0 |
| 1:09PM |
2 |
Server Specification |
| 1:05PM |
2 |
dimensioning: Where is the CPU vs Asterisk load table |
| 12:53PM |
2 |
SIP phones unbeatable echo |
| 12:50PM |
2 |
Easy to Access Telephone Directory AGI |
| 12:48PM |
3 |
Bridging app |
| 12:33PM |
0 |
Second edition of my * book has been release d |
| 12:20PM |
0 |
Re: Transfer issue with a Cisco CCM/phone (Peckham, Christopher) |
| 12:09PM |
0 |
How to register a SIP phone on Asterisk behind NAT |
| 11:43AM |
3 |
Using an extension to send a linux command |
| 11:02AM |
0 |
(Trunk) in production |
| 10:41AM |
0 |
cisco as5400, sip, asterisk. cisco won't detect that the call is answered |
| 10:33AM |
2 |
Where do I find *asterisk-capi* |
| 10:03AM |
0 |
PBX making ENUM lookups |
| 10:02AM |
2 |
Asterisk crossed lines? |
| 9:58AM |
1 |
Problem with an automatic responder |
| 9:58AM |
1 |
R: app_rxfax.so and app_txfax.so |
| 9:38AM |
2 |
Company directory not finding names... sometimes. |
| 9:32AM |
2 |
DTMF Issues With Asterisk 1.2 IVR |
| 9:25AM |
2 |
Adit 600 and echo |
| 9:01AM |
0 |
Transfer issue with a Cisco CCM/phone |
| 8:39AM |
6 |
app_rxfax.so and app_txfax.so |
| 7:33AM |
2 |
conditional canreinvite |
| 7:26AM |
4 |
dCAp |
| 6:47AM |
2 |
Build Error - ZT_EVENT_DTMFDIGIT |
| 6:35AM |
0 |
(no subject) |
| 6:15AM |
1 |
No D-channels available! Using Primary on channel 16 anyway! |
| 6:01AM |
1 |
GSM codec problem - Windows messenger 5.1 |
| 4:57AM |
0 |
Fwd: voip - forwarding ports |
| 3:56AM |
0 |
Avoided initial deadlock |
| 3:56AM |
3 |
read .what else to do ? |
| 3:41AM |
0 |
Catv ATA problem |
| 2:25AM |
0 |
Re: Issue calling other PBX systems using VoIPwithPolycom 501 |
| 12:45AM |
2 |
Zaptel SVN |
| |
| Wednesday January 11 2006 |
| Time | Replies | Subject |
|---|
| 11:25PM |
2 |
Dial application newbie help |
| 10:04PM |
0 |
Turning off 2100 Hz tone detection without editing zconfig.h and recompiling |
| 8:46PM |
0 |
patton smartnode 2400 with ic-t1v |
| 8:16PM |
1 |
chan_bluetooth problems |
| 7:16PM |
0 |
problems with installing app_odbcexec into dialplan |
| 6:53PM |
0 |
AlarmReceiver? |
| 4:46PM |
0 |
Remote Trunk setup |
| 3:58PM |
1 |
a2blling billing system |
| 3:19PM |
1 |
OOH323 Configuration with Cisco FSX ports, no Gatekeeper |
| 3:17PM |
1 |
Zaptel modules load, but Asterisk fails at s tartup |
| 3:01PM |
0 |
FW: Enchance Me 1.004 Released! |
| 2:48PM |
0 |
Enchance Me 1.004 Released! |
| 2:20PM |
1 |
where to get app_cepstral.c |
| 2:06PM |
21 |
FXS or VOIP |
| 1:47PM |
1 |
asterisk with an external predictive dialer |
| 1:43PM |
0 |
Execute command on pickup the phone |
| 1:08PM |
1 |
Re: setting up asterisk to handle incoming SIP URI |
| 12:55PM |
1 |
OT- Sangoma Question |
| 12:30PM |
1 |
Asterisk and Radius |
| 12:21PM |
1 |
Fax RX and SIP/IAX |
| 11:35AM |
1 |
Zaptel modules load, but Asterisk fails at startup |
| 11:21AM |
1 |
Issue calling other PBX systems using VoIP with Polycom 501 |
| 11:21AM |
17 |
Nested MySQL Commands |
| 10:53AM |
0 |
ruby-agi-1.0.2 released ! |
| 10:45AM |
0 |
China DID Wanted |
| 10:28AM |
4 |
Echo on phones... |
| 8:59AM |
0 |
Asterisk Manager API and ZapBarge or ChanSpy |
| 8:59AM |
0 |
Asterisk doesn't detect answer for some numbers |
| 8:59AM |
1 |
patching asterisk with tzafrir patch for voicemail permission does not work |
| 8:31AM |
1 |
Call Parking... |
| 8:20AM |
1 |
Signaling the status of the line on the phone |
| 7:57AM |
1 |
Better solution to mysql reconnect timeout |
| 7:46AM |
4 |
Why remotely reboot SIP phones? |
| 6:59AM |
3 |
Web based SIP client |
| 6:58AM |
1 |
SIP standard for flash |
| 6:52AM |
0 |
Connecting to a legacy PBX extension |
| 6:45AM |
1 |
[suse-isdn] Major Problems UTStarcom F1000 registering -- pls help |
| 6:43AM |
6 |
Failover Device? |
| 6:42AM |
0 |
Errors with bristuff-0.3.0-PRE-1e and asterisk cores |
| 6:26AM |
3 |
video development |
| 6:01AM |
5 |
Recommend Fax Hardware for T1 PRI |
| 4:46AM |
2 |
Transfer sounds - notifications |
| 3:24AM |
0 |
Incoming PSTN Calls - Can't interrupt Main Menu |
| 2:31AM |
1 |
Transfer to meetme on different server |
| 2:27AM |
1 |
Asterisk REGISTERs |
| 2:12AM |
0 |
does anyone know how to use 1.2 CVS setgroup in CAGI script |
| |
| Tuesday January 10 2006 |
| Time | Replies | Subject |
|---|
| 11:52PM |
3 |
IAX & CallerID |
| 10:03PM |
0 |
New Freelance Site for Asterisk Consultantsand Those who Need Projects Done |
| 9:10PM |
1 |
SOLVED: Hung Zap channels connected to old key system |
| 7:50PM |
3 |
New Freelance Site for Asterisk Consultants and Those who Need Projects Done |
| 6:38PM |
0 |
register to a peer register => from database |
| 3:40PM |
0 |
TDM04B odd problem |
| 2:38PM |
2 |
TE405p -- loopback for the phone company? |
| 2:26PM |
0 |
The second edition of my Asterisk book is nowavailable |
| 2:21PM |
0 |
testmail |
| 1:47PM |
4 |
Help with amportal: asterisk ended with exit status 127 |
| 1:28PM |
3 |
The second edition of my Asterisk book is now available |
| 1:24PM |
1 |
Still an open Seat in London for Next Weeks Signate intro to Asterisk Course |
| 1:19PM |
1 |
Eid Mubarak |
| 1:19PM |
0 |
Sipura SPA-2100 / 3000 provisioning .xml examples / xml variable list |
| 12:59PM |
1 |
GrandSTream 488/Asterisk |
| 12:16PM |
0 |
outboundproxy issue |
| 10:56AM |
1 |
Asterisk voicemail support |
| 10:49AM |
1 |
pattern mach doubt |
| 9:58AM |
0 |
Sacramento Asterisk Users Group |
| 9:54AM |
0 |
Re: 32 e1's with asterisk |
| 9:53AM |
1 |
Another cisco question |
| 9:02AM |
0 |
Besides the ISDN Guard what options? |
| 8:42AM |
0 |
ASTCC Voice Prompts in Spanish |
| 8:38AM |
1 |
avoided deadlock/channel already in use |
| 8:25AM |
4 |
web sip client |
| 8:10AM |
1 |
32 E1's in one Asterisk 'box' |
| 8:04AM |
0 |
Need help testing IAX based web conferencing tool |
| 8:02AM |
1 |
FW: Re: hangup detection |
| 7:56AM |
1 |
VMauthenticate always asks for mailbox |
| 7:48AM |
1 |
busydetect |
| 7:37AM |
0 |
Asterisk configuration using Database..! |
| 7:28AM |
1 |
Disconnected calls |
| 7:20AM |
0 |
Austin User Group |
| 6:04AM |
0 |
Live Demo of DRUID Asterisk Management Interface |
| 5:56AM |
0 |
Max calls & IAX2 trunking |
| 5:17AM |
1 |
Sip Behind Proxy |
| 5:09AM |
0 |
Setting up Asterisk using Mysql. |
| 4:58AM |
3 |
CDR problem - incorrect time |
| 4:06AM |
2 |
Asterisk Archives: BUG? |
| 12:10AM |
2 |
Problem with Action:Originate with ASterisk Manager |
| |
| Monday January 9 2006 |
| Time | Replies | Subject |
|---|
| 10:10PM |
0 |
(no subject) |
| 10:09PM |
1 |
Second edition of my * book has been released |
| 9:50PM |
0 |
Call Rules |
| 9:22PM |
1 |
How does the PCI bus effect latency and echo? |
| 9:20PM |
1 |
Tellabs echo can, can someone wire mine up for $? |
| 8:06PM |
3 |
Incoming Zap channels not behaving as expected. Rejecting call on channel.... |
| 6:45PM |
7 |
Presence support on GrandStream GXP-2000 |
| 6:44PM |
9 |
Recommendations on a WiFi phone for *? |
| 6:29PM |
1 |
to another country |
| 5:30PM |
0 |
RE: Help needed ("...Broken pipe" error |
| 5:15PM |
2 |
mpg123 removal |
| 4:36PM |
2 |
TDM400 (TDM11B) configuration |
| 4:26PM |
1 |
how to adjust volume |
| 3:46PM |
2 |
ZAP - configure not to answer? |
| 3:31PM |
15 |
MTU and Voice Delay (latency??) |
| 3:14PM |
0 |
Stanaphone Configuration |
| 2:56PM |
1 |
OT: IAXModem in inittab causes modem to be u nres ponsive, running from console it's OK |
| 2:20PM |
8 |
Pri Gateway Hardware |
| 2:08PM |
0 |
Asterisk 1.2 - sip_buddies restrictid problem. |
| 2:06PM |
0 |
zaphfc and T0 ISDN to Alcatel PBX |
| 1:47PM |
2 |
Cisco phones 7940 |
| 1:44PM |
1 |
Unable to connect to Asterisk |
| 1:28PM |
7 |
"Decent" sub-$100 SIP phone. |
| 12:58PM |
3 |
Problem Compiling Zaptel 1.2.1 |
| 12:53PM |
1 |
OT: IAXModem in inittab causes modem to be unres ponsive, running from console it's OK |
| 12:50PM |
1 |
Zaptel errors (power alarm?) |
| 12:46PM |
1 |
Asterisk featdmf signalling. |
| 12:26PM |
0 |
Answer call waiting / flash with Zaptel POTS and VOIP |
| 12:16PM |
0 |
ectoolkit |
| 10:54AM |
1 |
SPA-841 spontaneous voicemail problem |
| 9:52AM |
1 |
Voicemail emailed volume |
| 9:32AM |
0 |
asterisk stops unexpected, no crash, but " clean" exit |
| 9:32AM |
1 |
ATA failover between datacenters |
| 8:52AM |
1 |
PrivacyManager & CallerID not passing |
| 8:44AM |
3 |
Same Zap channel in multiple groups |
| 8:23AM |
1 |
PSTN line quality |
| 8:10AM |
0 |
Asterisk over 3Com |
| 8:02AM |
1 |
Chanspy options in Asterisk Manager API |
| 7:41AM |
0 |
Dialtone detection help needed |
| 7:36AM |
0 |
Agents in 1.2.1 |
| 7:00AM |
0 |
Snom Idleline XML |
| 6:33AM |
2 |
Lost my Zap's |
| 6:06AM |
1 |
snom programmable buttons |
| 5:32AM |
1 |
Is it Wildcard 406 |
| 4:40AM |
2 |
dual IP connections |
| 4:21AM |
2 |
call files, fax |
| 4:21AM |
3 |
SNOM Hotdesking... |
| 3:17AM |
0 |
GradStream Budge Tone - 100 / PLease help |
| 1:35AM |
1 |
ISDN beronet: cannot send digits during outbound calls |
| 12:57AM |
0 |
SIP-SIP transfer via the REFER/NOTIFY method |
| |
| Sunday January 8 2006 |
| Time | Replies | Subject |
|---|
| 11:32PM |
0 |
Re: Asterisk-Users Digest, Vol 18, Issue 46 |
| 11:15PM |
0 |
problems with app_odbcexec |
| 11:04PM |
1 |
Asterisk crashing system |
| 10:28PM |
0 |
Re: Asterisk-Users Digest, Vol 18, Issue 46 |
| 10:19PM |
0 |
Call forwarding for particular extension when line 1 is busy |
| 9:20PM |
1 |
Successfully Ported Asterisk On ARM Platform |
| 8:13PM |
1 |
FastAGI available? |
| 8:05PM |
1 |
spandsp, rxfax, TDM400/zaptel, missed frames, any help? |
| 7:18PM |
1 |
JiveMessenger HOWTO |
| 7:17PM |
0 |
DialPlan for Call Limit, Call Duration, And Group Call |
| 7:11PM |
0 |
FWT - LSP-350T - Asterisk |
| 4:42PM |
0 |
DTMF relay problem |
| 3:35PM |
0 |
spandsp for 1.2.1 - libspandsp.so.0: cannot openshared object file: No such file or directory |
| 3:24PM |
0 |
spandsp for 1.2.1 - libspandsp.so.0: cannot open shared object file: No such file or directory |
| 1:52PM |
1 |
PolyCom phones with blinking clock and wrong time |
| 1:51PM |
0 |
Fwd: Problems with R2 Support |
| 12:39PM |
3 |
Monitor Logged in Agent's conversation |
| 11:41AM |
1 |
new AMPortal and Asterisk debs |
| 8:01AM |
1 |
Dialogic VFX/41JCT-LS found i a drawer |
| 6:50AM |
2 |
Zaptel make install error |
| 5:22AM |
8 |
Cisco 801 and rcapi |
| 5:08AM |
2 |
3 PSTN lines, 3 IP Phones |
| 4:01AM |
0 |
2 small issues with Cisco 1760 gateway and Asterisk |
| 2:11AM |
0 |
Advice on fax support |
| 12:28AM |
1 |
Processor Update? |
| |
| Saturday January 7 2006 |
| Time | Replies | Subject |
|---|
| 11:37PM |
0 |
Line Sharing or Better Call Pickup |
| 10:12PM |
0 |
Agi Perl Talk Time |
| 9:27PM |
1 |
Immediate routing on "0" (DNIS)? |
| 9:03PM |
1 |
Some advice on routing DID's |
| 8:33PM |
0 |
Up to 4 seconds delay to play prompt? |
| 7:45PM |
1 |
Kudzu and Zaptel Cards |
| 7:23PM |
0 |
cisco 8xx ISDN router |
| 7:11PM |
0 |
Asterisk Market Share |
| 5:11PM |
2 |
how to configure iax account for iaxmodem? |
| 4:47PM |
14 |
Asterisk Jobs |
| 4:42PM |
1 |
Possible bug with GotoIfTime |
| 2:23PM |
1 |
choppy music on hold - only on PRI PSTN |
| 6:00AM |
4 |
Draytek Vigor 2900 & Asterisk |
| 5:51AM |
1 |
Problens to link 2 * servers |
| 5:19AM |
2 |
wich IAX soft client allow to specify a different server port? |
| 1:26AM |
1 |
re: where can i find all .C files |
| |
| Friday January 6 2006 |
| Time | Replies | Subject |
|---|
| 8:47PM |
0 |
--- AEL 2 --- Try it out! |
| 6:18PM |
1 |
Aastra 9133i and NAT: Can it work? |
| 6:02PM |
0 |
h323 gatekeeper?? |
| 4:39PM |
3 |
transfer application |
| 4:15PM |
7 |
Fax, txfax -bizarre thing |
| 3:56PM |
1 |
Latency |
| 3:30PM |
0 |
Bristuffed asterisk 1.2.1 on Suse 10 - problem with zaphfc module |
| 2:56PM |
6 |
Non-PRI T1 |
| 2:52PM |
3 |
Help Connecting server districts |
| 2:24PM |
3 |
Asterisk initialization |
| 2:24PM |
2 |
Voice mail messages aren't sent to e-mail |
| 2:16PM |
0 |
How to properly use GROUP |
| 2:06PM |
0 |
Not Able to Connect Two Asterisk Servers Usi ng IAX2 |
| 1:57PM |
2 |
Using local\number |
| 1:05PM |
0 |
SJPhone with external ringer |
| 12:57PM |
2 |
controlling SIP subscriptions from SNOM phones |
| 12:20PM |
2 |
Not Able to Connect Two Asterisk Servers Using IAX2 |
| 12:06PM |
2 |
SPA-3000 is translating vocal sounds into DTMF |
| 11:33AM |
1 |
Alphanumeric pattern match in extensions.conf |
| 11:24AM |
1 |
server recommendations |
| 9:54AM |
0 |
Problem with Call Monitoring |
| 9:28AM |
1 |
Annoying Notice Message: "Don't know what to do with control frame 15" |
| 8:51AM |
4 |
Problem with show channels |
| 8:47AM |
5 |
3RD REQUEST - Any Help Is Appreciated |
| 8:28AM |
2 |
Incoming PSTN Calls - Stumped |
| 7:57AM |
0 |
IAX2->SIP dropped calls |
| 7:46AM |
3 |
Announcing a call transfer |
| 7:37AM |
3 |
Recording Calls at the phone |
| 7:35AM |
1 |
How To - Building a VoIP-PSTN Gateway with Asterisk |
| 6:53AM |
0 |
Xs4all VoIP service - SIP config? |
| 5:51AM |
2 |
Call forwarding for particular extension |
| 5:23AM |
1 |
Problem with integrating ISDN PBX using NT mode |
| 4:57AM |
0 |
RE:how many calls Asterisk gateway can handle |
| 4:10AM |
2 |
Budge Tone-100 as a Ext in the LAN |
| 3:27AM |
3 |
Macro DialPlan |
| 2:27AM |
3 |
bayhamsystems.com experience |
| 1:45AM |
0 |
cisco/asterisk interop issues? |
| 12:01AM |
1 |
Sharing SIP Info with Realtime |
| |
| Thursday January 5 2006 |
| Time | Replies | Subject |
|---|
| 10:50PM |
0 |
DNIS dropping digits. |
| 9:05PM |
1 |
CD (call deflection) on Bristuff/zaphfc? |
| 7:22PM |
1 |
bristuff/zaphfc disturbing other ISDN phones |
| 6:50PM |
2 |
Screening incoming calls. |
| 6:49PM |
0 |
Call Limit[Local, InterLocal, International, Group, Time, Duration, Etc] |
| 5:42PM |
1 |
open h323 compile error |
| 5:41PM |
1 |
fax detection on TE406P |
| 5:01PM |
2 |
Integrating with Toshiba Strata DK40i KSU |
| 3:36PM |
0 |
phpagi stream_file |
| 2:56PM |
1 |
In search of Headset Compatible Analog Phone |
| 2:48PM |
3 |
TE110p and pri_cpe signalling not recognized |
| 2:47PM |
1 |
Re: Has anyone tried using flash() in features.conf (applicationmap) - RESOLVED |
| 2:37PM |
1 |
Polycom 501 netboot not working. |
| 1:32PM |
1 |
troubleshooting hangups? |
| 1:31PM |
0 |
Trailing silence in voicemail messages |
| 1:10PM |
0 |
PRI deadlock problem is 1.2.1 |
| 12:50PM |
0 |
Meetme user join/leave |
| 12:42PM |
0 |
Bizarre Answering Problem - 2ND REQUEST |
| 12:13PM |
2 |
DEFAULT_USERAGENT |
| 11:49AM |
1 |
Call logging |
| 11:38AM |
1 |
ChanSpy via external application |
| 11:28AM |
0 |
Hardware Manual |
| 10:43AM |
1 |
UserEvent() with multiple body lines |
| 10:19AM |
8 |
Asterisk Debugging |
| 9:52AM |
0 |
Problem when i make a DATA CALL |
| 9:41AM |
0 |
red alarm when modprobe wcte11xp |
| 9:14AM |
5 |
OT: SIP aware firewalls? |
| 9:02AM |
2 |
Call Group Limit |
| 8:24AM |
3 |
Fax with Asterisk and Sipura 2100 |
| 8:10AM |
0 |
Reading sound and recognizing DTMF sounds in eagi script ? |
| 8:05AM |
1 |
zaptel does not compile with kernel 2.6.15 |
| 7:50AM |
1 |
Bizarre Answering Behavior |
| 7:29AM |
0 |
Re: Problem with blind transfer and Polycom phones !! more info |
| 7:11AM |
3 |
Remotely reboot SIP Phones ? |
| 6:59AM |
2 |
Asterisk CLI | more |
| 6:15AM |
1 |
Iaxy Ringtone |
| 6:07AM |
3 |
Problem with blind transfer and Polycom phones |
| 5:35AM |
1 |
Incoming PSTN Calls |
| 4:44AM |
0 |
Regular Crashes - Partially Solved |
| 3:30AM |
1 |
Virtuozzo - G729 |
| 3:23AM |
0 |
Incoming calls grind to a halt |
| 2:22AM |
0 |
SIP/IAX softphones for use in callcentre environments |
| 2:17AM |
1 |
Bind asterisk to multiple IPs (reply problem) |
| 12:48AM |
1 |
TDM400P modules not found |
| 12:39AM |
4 |
Zap channel instances |
| |
| Wednesday January 4 2006 |
| Time | Replies | Subject |
|---|
| 11:32PM |
0 |
RE: how many call an Asterisk gateway can handle |
| 10:37PM |
3 |
SIP/IAX softphones for use in call centre environments |
| 10:00PM |
0 |
Some WARNINGS |
| 8:51PM |
0 |
Has anyone tried using flash() in features.conf (applicationmap) |
| 7:21PM |
2 |
Cisco phone issue |
| 7:00PM |
1 |
New Mail Message Waiting |
| 6:52PM |
0 |
Sip peers got disconnect |
| 4:35PM |
2 |
H323 compilation Help needed |
| 3:11PM |
2 |
[Web-MeetM] Seeking Beta testers |
| 2:18PM |
0 |
Asterisk Dial problem |
| 1:48PM |
1 |
local exchange dialtone on ISDN/bristuff? |
| 12:53PM |
3 |
Email2fax big problemo |
| 12:22PM |
0 |
NOTICE[3589]: frame.c:128 __ast_smoother_feed: Dropping extra fr ame of G.729 since we already have a VAD frame at the end |
| 11:59AM |
5 |
Grandstream web configuration utility |
| 11:27AM |
0 |
Strange AGI behavior on Mac OS X |
| 11:26AM |
0 |
confusion about contexts - SER |
| 11:24AM |
2 |
Dial(Console/dsp) and option g doesnt appear to work |
| 10:19AM |
0 |
TE411P in a HP DL360 - which BIOS settings work? |
| 9:59AM |
1 |
M0n0Wall traffic shaping rules |
| 9:59AM |
1 |
chan_oh323.so freeze my box on unload |
| 9:43AM |
2 |
VoiceMailMain Pass Mailbox |
| 8:43AM |
0 |
SUSE 10.1 |
| 8:13AM |
1 |
AMP: Losing backslash characters in config files |
| 8:01AM |
0 |
how do i play a prerecorded message in the middle of a conversation ? |
| 7:41AM |
1 |
FYI new aricle on asteisk |
| 7:39AM |
0 |
Unknown digits |
| 7:22AM |
2 |
Using *RT for HA purposes was: RealtimeMultipleAsterisk boxes, iaxusers |
| 7:20AM |
0 |
Entry level IP phone |
| 7:08AM |
2 |
suddenly iax calls don't work anymore |
| 6:32AM |
0 |
Can i compile Asterik on Fedora 4 x86 64 and which hardware could you support ? |
| 6:30AM |
0 |
Compilation of OpenH323 libraries under CYGWIN... |
| 6:08AM |
1 |
RxFax : Change FAX Resolution |
| 4:23AM |
0 |
Anybody successfully using vISDN on A@H? |
| 3:43AM |
0 |
USB fxo/fxs devices |
| 3:31AM |
2 |
remote sip client fail to register |
| 3:28AM |
1 |
RBT enable/disable |
| 2:42AM |
2 |
call monitoring from 3th phone |
| 2:42AM |
0 |
Can I call another S0 bus device (BRI) locally without taking 2 channels through Telco provider ? |
| 1:57AM |
1 |
SIP security |
| 1:47AM |
0 |
Is it possible to get caller and callednumberwith Asterisk Manager |
| 1:32AM |
0 |
MusicOnHold don't start at begin |
| 12:19AM |
2 |
Ominiis Asterisk TAPI driver |
| |
| Tuesday January 3 2006 |
| Time | Replies | Subject |
|---|
| 11:16PM |
1 |
Detect a forwarded incoming call? |
| 10:33PM |
0 |
Asterisk::LDAP |
| 9:31PM |
1 |
Raw Hangup messages with IAX2? |
| 9:18PM |
1 |
IAX2 channels denoted as '(None)' |
| 6:54PM |
5 |
Start recording after call started |
| 6:14PM |
2 |
integration with Meridian/Norstar ATA2 |
| 5:51PM |
0 |
How do you check whether a channel is active andthe number of calls |
| 5:33PM |
0 |
(Fwd) bridging two active calls |
| 5:11PM |
1 |
bridging two active calls |
| 4:43PM |
4 |
Problems Upgrading to 1.2.1 on Fedora 3 |
| 4:31PM |
7 |
Dialer |
| 4:30PM |
1 |
confusion about contexts |
| 4:17PM |
1 |
Resolving timing issues with dual PRIs in a TE411P |
| 3:26PM |
0 |
Uvox streams |
| 2:41PM |
1 |
Using *RT for HA purposes was: Realtime MultipleAsterisk boxes, iaxusers |
| 2:35PM |
1 |
Sipbroker? |
| 12:56PM |
1 |
Asterisk 1.2.1 Type of Service |
| 12:43PM |
0 |
Re: Asterisk-Users Digest, Vol 17, Issue 109 |
| 12:43PM |
3 |
OT: XML Content Manager for Cisco 79XX Phones |
| 12:27PM |
2 |
How do you check whether a channel is active and the number of calls |
| 12:14PM |
0 |
Experience with SetTransferCapability |
| 12:10PM |
4 |
iax2 wireless and Multicast |
| 11:42AM |
2 |
Looping Problem With Call Forwards - Do you have comments on my solution? |
| 11:29AM |
2 |
Heavy Static on incoming calls |
| 11:26AM |
0 |
cannot register whit sip client when i'm outside the PBX LAN |
| 10:51AM |
0 |
Recording Agent Calls |
| 10:39AM |
0 |
DTMF dialing |
| 10:33AM |
3 |
IAX termination services |
| 10:21AM |
5 |
Asterisk on Dell blade servers |
| 9:48AM |
0 |
Re IAXTEL |
| 9:37AM |
0 |
Sipura SPA-1001 question |
| 9:33AM |
9 |
FC3 or FC1 (or something else?) |
| 8:55AM |
0 |
Anyone heard of this company? http://www.affinityvoiptelecom.com/ |
| 8:50AM |
1 |
AEL - Using a Macro in the Dial Command in AEL |
| 8:18AM |
0 |
Asterisk realtime mysql connection |
| 8:16AM |
1 |
Problem with date & time on Aastra480isincerelease 1.3 |
| 8:05AM |
2 |
Question on SPA-2002 |
| 7:41AM |
1 |
E1 with CAS but no call signalling? |
| 7:38AM |
2 |
Problem with date & time on Aastra 480isincerelease 1.3 |
| 7:13AM |
2 |
Problem with date & time on Aastra 480i sincerelease 1.3 |
| 7:01AM |
1 |
Howto compile chan_h323 on macosx 10.3? |
| 6:06AM |
4 |
Where is the Prefix() application in Asterisk 1.2.1 ? |
| 6:01AM |
1 |
Three Way Calling with HFC PCI Card |
| 5:52AM |
0 |
outbound sip calls on asterisk |
| 5:45AM |
0 |
SetCallerPres |
| 4:41AM |
0 |
Txgain & Rxgain |
| 3:47AM |
3 |
cisco 7960 registration fails |
| 2:59AM |
1 |
AgentCallbackLogin pre-# announcement? |
| 2:56AM |
0 |
dhcp auto-provision spa-3000 like hardphones? |
| 2:06AM |
4 |
call-limit kills hints |
| 12:44AM |
0 |
How to get To-header and From-header display name |
| |
| Monday January 2 2006 |
| Time | Replies | Subject |
|---|
| 11:25PM |
0 |
Mute a given channel. |
| 7:52PM |
1 |
offline g.729 transcoding |
| 7:40PM |
2 |
Phantom Call on hangup SPA 2000 |
| 5:06PM |
1 |
Limit concurent calls per MSN on BRI (bristuff/zaphfc)? |
| 4:23PM |
2 |
Asterisk Upgrade to 1.2 |
| 2:41PM |
0 |
Memory PB. |
| 1:38PM |
1 |
Festival clicks instead of sound and disconnects. |
| 1:28PM |
1 |
Ignoring too old SIP packet packet problems |
| 1:23PM |
6 |
@Home overwrites configs on startup |
| 1:08PM |
0 |
Re: Asterisk-Users Digest, Vol 18, Issue 6 |
| 12:32PM |
5 |
SIP through freeBSD NAT |
| 9:40AM |
0 |
RE: simulator for asterisk gateway |
| 9:35AM |
0 |
R: Client SIP fo Windows Mobile |
| 9:10AM |
3 |
USB phone |
| 9:05AM |
2 |
Client SIP fo Windows Mobile |
| 8:35AM |
5 |
E911 And Routing |
| 8:27AM |
1 |
sip realtime and setvar |
| 8:20AM |
2 |
Translating between different codes |
| 7:52AM |
4 |
Asterisk PRI problems. |
| 7:46AM |
1 |
Echo after asterisk has been running for several days |
| 7:44AM |
2 |
Q: How to dial out / transfer calls with manager |
| 6:13AM |
2 |
Is it possible to get caller and called number with Asterisk Manager |
| 6:06AM |
1 |
CAPI unable to handle busy() |
| 5:03AM |
6 |
connect more the one phone to ONE sip Acoount |
| 4:30AM |
2 |
unable to execute call file |
| 4:05AM |
0 |
Happy New Asterisk Year! |
| 2:40AM |
1 |
Newbie Problem With Agents |
| |
| Sunday January 1 2006 |
| Time | Replies | Subject |
|---|
| 10:43PM |
2 |
(Fwd) hi there |
| 4:28PM |
1 |
Asterisk 1.2.1 segmentation faulting!... |
| 3:01PM |
6 |
Cell phone dock/switch as Asterisk FXO source |
| 12:42PM |
5 |
Having major issues with TDM2400 |
| 9:55AM |
1 |
Need HT488 FXO example for both inbound andoutbound. |
| 9:29AM |
1 |
Snom 190 occasionally NR, SIP 401 |
| 8:13AM |
3 |
CrystalFontz LCD display |
| 6:50AM |
2 |
Recommendations on web interface for IT staff |