| Thursday September 30 2004 |
| Time | Replies | Subject |
|---|
| 11:37PM |
0 |
Please help - jitter problem |
| 11:04PM |
1 |
problem in using sip-communicator with asterisk |
| 7:28PM |
1 |
V1.0.1 - Segmentation fault in res_crypto.so ? |
| 7:11PM |
1 |
Queue Setup almost got it |
| 6:43PM |
0 |
register expire |
| 6:35PM |
2 |
Queue Setup |
| 4:26PM |
1 |
Monitoring SIP with Whats Up Gold |
| 4:06PM |
2 |
Timing Problem |
| 3:38PM |
4 |
Caller ID Info from Cisco router to Asterisk |
| 3:13PM |
7 |
Asterisk hardware |
| 2:54PM |
0 |
cisco 7940 in mgcp |
| 2:02PM |
2 |
Time of Day Routing |
| 1:31PM |
4 |
Voice mail |
| 1:17PM |
2 |
FXO/FXS card |
| 1:02PM |
1 |
Channel banks that you've used |
| 1:00PM |
0 |
BRI cards for Asterisk in US |
| 12:52PM |
3 |
Debian: loading Zaptel modules at boot time |
| 12:49PM |
0 |
CLI color using -r |
| 12:36PM |
0 |
Re: Re: Re: Confused of London - How to associate zapchannels to extensions |
| 12:15PM |
1 |
Sip over cisco VPN |
| 12:13PM |
1 |
Re: Re: Re: Confused of London - How to associate zapchannels to extensions |
| 12:11PM |
0 |
autodial question |
| 12:00PM |
0 |
* doesn't answer |
| 11:15AM |
0 |
UK Caller ID - todays CVS update knocks out a channel |
| 10:42AM |
1 |
Anyone in digium? |
| 10:26AM |
2 |
OT: Kphone installation problem |
| 10:21AM |
0 |
Asterisk seems to have more jitter than a hardphone with SIP |
| 9:28AM |
5 |
Confused of London - How to associate zap channels to extensions |
| 9:17AM |
7 |
asterisk 407 Proxy Authentication Required |
| 9:03AM |
4 |
Setting auto-attendant to answer immediately |
| 8:29AM |
0 |
Asterisk server suddenly fronzen after many zapter errors |
| 8:09AM |
4 |
Ring Multiple SIP client at the same time |
| 7:54AM |
3 |
Sipura-3000 - silent dial out on FXO port |
| 7:25AM |
6 |
No Asterisk Sounds on SUSE ES 9/Linux 2.6 |
| 7:06AM |
0 |
Re: Asterisk-Users Digest, Vol 2, Issue 329 |
| 6:42AM |
1 |
CVS best practice |
| 6:22AM |
1 |
Strange Quality problems with Asterisk, Gentoo, Redhat and Kernels - /dev/dsp] |
| 6:13AM |
0 |
Refer Method |
| 4:08AM |
1 |
Sip Channels. |
| 3:02AM |
0 |
Oops, a seg fault =( |
| 2:16AM |
1 |
sipfriends in MySQL question/request |
| 2:13AM |
0 |
AGGRESSIVE_SUPPRESSOR |
| 2:06AM |
4 |
No Audio |
| 1:56AM |
2 |
cli command to check the codec use for the connected calls |
| 1:15AM |
1 |
easy way of add 100 extensions |
| 12:44AM |
0 |
Video mail |
| 12:36AM |
1 |
how to hung up a call immediately if it SIP response 486 "Busy Here" received |
| |
| Wednesday September 29 2004 |
| Time | Replies | Subject |
|---|
| 11:29PM |
2 |
Asterisk 1.0.1 Released |
| 10:51PM |
0 |
CDR Disposition for call queue's |
| 10:37PM |
1 |
Asterisk 1.00 Call quality problem |
| 9:31PM |
0 |
Big zaphfc issues |
| 9:24PM |
3 |
Calling Waiting on PSTN line |
| 8:52PM |
0 |
Query on DMA Config |
| 8:50PM |
1 |
Budgetone problem |
| 7:21PM |
1 |
mid-call echo |
| 6:37PM |
3 |
7912G SCCP only? |
| 6:34PM |
0 |
TDM400P (TDM20B) 'Freshmaker failed register test' |
| 5:52PM |
3 |
Dial Delay |
| 5:47PM |
1 |
I am newbie with asterisk |
| 5:36PM |
0 |
astersk-oh323 compile error make |
| 5:33PM |
0 |
no sound from jsphone, the voicemail ok |
| 5:07PM |
4 |
Wooksung Video Phones |
| 3:33PM |
0 |
Queue member still rings even if on call |
| 2:34PM |
1 |
Zaptal and Fedora Core 2 and losing GSM playback |
| 2:15PM |
3 |
HELP: Asterisk - SIP to H.323 translation |
| 1:54PM |
0 |
Stanaphone DID : user not available |
| 1:49PM |
1 |
VoicemailMain GSM |
| 11:30AM |
0 |
SIP signalling from Gateway |
| 11:26AM |
0 |
Re: Asterisk Performance |
| 11:14AM |
1 |
Channelized E1 - help ! |
| 11:03AM |
0 |
Grandstream BT101 stops ringing |
| 10:58AM |
0 |
PRI Failover Success? |
| 10:49AM |
0 |
chan_sip2 broken with FWD |
| 10:42AM |
0 |
PRI D-channel signalling error? "Ring reques ted onchannel 0/1 a lready in use on span 1. Hanging up owner." |
| 10:32AM |
4 |
Cisco 3620 PRI and Asterisk |
| 10:19AM |
3 |
X100P Unstable. |
| 10:01AM |
1 |
Call waiting does not ring phone |
| 9:40AM |
0 |
Asterisk - SIP to H.323 translation |
| 8:33AM |
1 |
E1 in Iran |
| 7:56AM |
2 |
Strange Quality problems with Asterisk, Gentoo, Redhat and Kernels - /dev/dsp |
| 7:54AM |
0 |
Compiling Zaptel on Suse ES 9.0 and torisa module errors |
| 7:29AM |
0 |
stateful vs. stateless |
| 7:21AM |
7 |
Credit Card machines / interop |
| 7:09AM |
0 |
Ang: Re: Dutch (DTMF) caller-ID |
| 6:44AM |
5 |
music on transfer |
| 5:43AM |
0 |
Anyone uses Lucent i2021 BRI Phone with Asterisk. |
| 5:17AM |
2 |
secure |
| 4:01AM |
4 |
* and Fax |
| 3:50AM |
1 |
Call Forward, DND and other standard features |
| 3:03AM |
0 |
sound dropouts during SIP re-register |
| 2:38AM |
1 |
Help termcap error |
| 1:17AM |
1 |
iax connection and 1 way distortion |
| 12:56AM |
4 |
No Sound between H.323 Gateway & Asterisk |
| 12:40AM |
0 |
PRI D-channel signalling error? "Ring requestedonchannel 0/1 a lready in use on span 1. Hanging up owner." |
| |
| Tuesday September 28 2004 |
| Time | Replies | Subject |
|---|
| 11:46PM |
1 |
too many ex-(boy|girl)friends |
| 11:18PM |
2 |
Nat Traversal help! |
| 11:18PM |
2 |
GSM phones, bluetooth and general hapiness |
| 9:55PM |
1 |
ASTCC : card generation problem |
| 9:42PM |
1 |
Newbie 2 PBX VOIP, protocol ?'s using Cisco 827 7910 |
| 8:38PM |
1 |
CIsco Gateway recommendation |
| 8:19PM |
4 |
Gatekeeper registration failed |
| 8:01PM |
2 |
7960 upgrade |
| 6:12PM |
0 |
TXfax: return code |
| 5:53PM |
1 |
binding to two IPs among five |
| 5:26PM |
1 |
ASTERISK loosing it.. stops responding to commands on TERMINAL |
| 5:00PM |
0 |
Understanding codecs and transcoder |
| 4:44PM |
2 |
Speex config |
| 3:49PM |
0 |
Leader IP10S |
| 3:39PM |
3 |
Retrieve voice mail message from outside |
| 3:09PM |
1 |
asterisks queues with static members |
| 2:41PM |
1 |
Help with Call Waiting! |
| 2:31PM |
20 |
Polycom IP500 |
| 2:27PM |
0 |
Intel Chipset MD-3200. | Works as 100XP- Clone. :) |
| 2:18PM |
1 |
Help with fax dial plan |
| 2:08PM |
0 |
PRI hiding caller id |
| 2:02PM |
1 |
Codecs and negotiations |
| 1:42PM |
0 |
Asterisk and other codecs |
| 1:00PM |
2 |
PRI D-channel signalling error? "Ring requested on channel 0/1 a lready in use on span 1. Hanging up owner." |
| 12:30PM |
0 |
Creating native bridge takes long time |
| 11:27AM |
0 |
Subscribe 403 forbidden |
| 10:46AM |
1 |
Sep 28 17:52:28 WARNING[163850]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call 2bda648025dbf8c52fd293515d98d2c2@216.252.176.45 for seqno 102 (Non-critical Request) |
| 10:14AM |
1 |
IAX softphone issues |
| 10:06AM |
3 |
CODECs and sip.conf and voice quality |
| 10:03AM |
1 |
Voicepulse quality problems |
| 10:00AM |
1 |
Looking for whoever wrote cdr_mysql |
| 9:53AM |
1 |
Is app_icd ready to replace app_queue? |
| 9:11AM |
1 |
Fast AGI |
| 9:03AM |
1 |
cisco 7960 7.1 -> 7.2 upgrade problem |
| 8:43AM |
0 |
xc-ast public beta |
| 8:37AM |
2 |
Variable Codecs Order |
| 8:18AM |
1 |
ZT_CHANCONFIG failed on channel 1 |
| 8:13AM |
1 |
chan_oh323 and DTMF |
| 7:32AM |
1 |
CDR and transfer |
| 7:14AM |
0 |
SpanDSP mail to fax |
| 7:12AM |
0 |
SRV LOOKUP |
| 7:07AM |
0 |
FW: FXO question |
| 7:01AM |
3 |
building a phone recorder |
| 6:43AM |
0 |
custom emailbody per any voicemail (mailbox) |
| 6:38AM |
0 |
HELP with IAXy provisioning........... newbie |
| 5:48AM |
2 |
SMDI Bounty - where? |
| 5:45AM |
0 |
H323 dropping connections |
| 5:21AM |
7 |
UK (British Telecom) Caller ID again |
| 3:52AM |
1 |
CAPI channels |
| 3:34AM |
2 |
Asterisk, Hylafax and T38Modem - help! |
| 2:46AM |
0 |
srv records |
| 12:11AM |
2 |
Unable to open pseudo channel |
| |
| Monday September 27 2004 |
| Time | Replies | Subject |
|---|
| 10:42PM |
1 |
Peer Review - Linuxfest Presentation Outline |
| 10:17PM |
1 |
(no subject) |
| 10:05PM |
1 |
asterisk with subnet 172.16.x.x |
| 9:05PM |
11 |
sipura over heat |
| 8:20PM |
2 |
Asterisk 1.0.0 rpms available |
| 7:11PM |
0 |
Cisco 7940 -60 firmware upgrades |
| 7:07PM |
1 |
Fedora2 and zaptel - using the udev |
| 5:45PM |
0 |
Predictive dialing and ICD |
| 4:53PM |
0 |
ASTCC with oh323 |
| 4:47PM |
3 |
CDW Part# for Cisco Software upgrade contract |
| 3:40PM |
0 |
recording a meet-me - only my voice? |
| 3:14PM |
3 |
Mutliple Line 'Buttons' |
| 1:52PM |
0 |
Re: Asterisk-Users Digest, Vol 2, Issue 281 |
| 1:49PM |
2 |
BudgeTone 100 & Call Transfer |
| 1:23PM |
5 |
Sending DTMF after recording new voicemail |
| 12:48PM |
0 |
Aastra/Sayson 480i |
| 12:45PM |
0 |
Passing DIALSTATUS between servers |
| 12:35PM |
2 |
Pass-thru port on X100P Clone |
| 12:23PM |
1 |
FC1 and FC2 RPMs now available |
| 12:02PM |
3 |
Agent Call Back LOGOUT? |
| 11:56AM |
3 |
Simple question |
| 11:50AM |
0 |
Flashing a POTS Line when you are on a call in Asterisk |
| 11:21AM |
1 |
New to Asterisk, questions about IVR and MySQL integration |
| 11:07AM |
0 |
Incomming calls are fine. No sound on outbound. |
| 10:52AM |
7 |
X100P knock-off price jump |
| 10:04AM |
2 |
How to revert back to autoattendant from a voicemail box? |
| 10:00AM |
2 |
Cisco Downloads --> was --> Re: Cisco 7960 andAsterisk...not working... |
| 9:51AM |
0 |
SNMP instrumentation and/or talk path health monitoring? |
| 9:31AM |
1 |
Beeping on messages and calls |
| 9:14AM |
0 |
Re: Complete newbie seeks start |
| 9:11AM |
0 |
spandsp question |
| 8:40AM |
1 |
Cisco Downloads --> was --> Re: Cisco 7960 andAsterisk...not working... |
| 8:21AM |
1 |
Manager QueueAdd |
| 7:51AM |
6 |
FXO question |
| 7:48AM |
1 |
Call Center Reporting Tools |
| 7:28AM |
1 |
PRI fields |
| 7:17AM |
8 |
Complete newbie seeks start . . . |
| 6:58AM |
0 |
Non-PRI T1 configuration - Asterisk-Users Digest, Vol 2, Issue 263 |
| 6:39AM |
1 |
G729 Private Licensing ?? |
| 6:23AM |
3 |
chan_capi, Eicon Diva server BRI, kernel 2.6? |
| 6:20AM |
1 |
Prepaid function with MySQL |
| 5:54AM |
2 |
stable OS |
| 5:44AM |
1 |
Dutch (DTMF) caller-ID |
| 5:31AM |
0 |
Faxdetect does not detect certain Fax machines |
| 5:29AM |
2 |
Brand New to List, requesting assistance |
| 5:27AM |
0 |
New Chipset from TI for VoIP with 802.11g/b |
| 5:23AM |
0 |
chan_sip.c 183 / 180 handling, unexpected results & playtone bug ... |
| 5:23AM |
1 |
ASTCC installation problems |
| 5:00AM |
3 |
TDM400 FXO outgoing call only |
| 4:53AM |
1 |
VoIP South Africa |
| 4:33AM |
1 |
Cisco IP phone G.723 |
| 2:47AM |
1 |
IP phone programming, |
| 2:29AM |
0 |
Asttapi - Version 0.04 |
| 2:28AM |
1 |
GS 101/102 Reboot |
| 2:17AM |
9 |
Question |
| 1:47AM |
1 |
make update and upgrade failed with `ZT_EVEN T_POLARITY' undeclared |
| 12:57AM |
1 |
music on hold file |
| 12:51AM |
0 |
ISDN line |
| 12:20AM |
0 |
Speex/ILBC buggy with * 1.0 and X-Lite/Pro? |
| 12:15AM |
3 |
Asterisk Compile error |
| |
| Sunday September 26 2004 |
| Time | Replies | Subject |
|---|
| 11:22PM |
1 |
FWD's Ed Guy notes from Astricon |
| 11:17PM |
0 |
is it possible to 'Auto Call Back' in Asterisk ?? |
| 9:53PM |
0 |
Error Compiling libunicall for MFC/R2 with spandsp |
| 9:35PM |
1 |
pri to voip |
| 8:38PM |
1 |
ASTCC Terms : Help |
| 7:24PM |
1 |
Autodial on off-hook? |
| 7:01PM |
0 |
RE: What about [TCL as ] a higher level configuration language? |
| 6:29PM |
1 |
Strange problem: can't fit into subject |
| 6:06PM |
1 |
Background call forwarding? |
| 4:02PM |
1 |
H323 with Tenor CMS Gateway |
| 3:45PM |
2 |
spandsp with TDM fxo card? |
| 2:38PM |
6 |
SIP Registration Timeout, No FW |
| 2:17PM |
0 |
zqaptel and hdlc |
| 11:35AM |
0 |
iLBC modes |
| 11:17AM |
6 |
Digium and mailing lists |
| 11:05AM |
1 |
How Do I configure zaptel for PRI in |
| 11:04AM |
4 |
IP Phones ? |
| 9:19AM |
6 |
Looking for a commercial version of an IAX2 Softphone |
| 8:31AM |
2 |
Proper Syntax |
| 7:42AM |
1 |
spandsp patch help |
| 6:52AM |
0 |
HELP with IAXy provisioning |
| 6:24AM |
1 |
voicemail /w asterisk - voicemail() problems |
| 6:03AM |
3 |
What about a higher level configuration language |
| 5:08AM |
1 |
pthread problem |
| 1:55AM |
0 |
Got SIP response 400 "Bad Request" ; Cisco 7940 inbound station/station call problem. |
| 12:41AM |
2 |
Asterisk <-> WellGate 3502a : ulaw/alaw only? |
| |
| Saturday September 25 2004 |
| Time | Replies | Subject |
|---|
| 11:39PM |
0 |
Digits being dropping when dialing from certain analog phones |
| 7:50PM |
2 |
Cisco Downloads --> was --> Re: Cisco 7960 and Asterisk...not working... |
| 4:46PM |
0 |
Cisco Soft IP Phone |
| 4:24PM |
2 |
* works, but after a few seconds audio always stops. |
| 2:53PM |
0 |
Dropping numbers on dialout through tdm400p |
| 1:32PM |
1 |
German Termination and DIDs |
| 1:15PM |
1 |
Application almost there..Dialplan challenges |
| 12:24PM |
1 |
How can I dial one unbusy channel of 4 available? |
| 11:47AM |
0 |
Ring delay |
| 11:40AM |
2 |
Cisco 7960 and Asterisk...not working... |
| 11:34AM |
3 |
Queue and Agent functionality |
| 11:01AM |
3 |
Non-PRI T1 configuration |
| 8:16AM |
4 |
Cisco PIX and Asterisk |
| 8:02AM |
1 |
Whoa.... I'm owned but found ?? |
| 7:50AM |
0 |
Codecs Problem? |
| 7:09AM |
0 |
getting variable using agi |
| 7:00AM |
1 |
Astricon Developers Conference Recordings |
| 6:28AM |
3 |
Help with dialing out with TDM400P |
| 6:18AM |
1 |
ilbc problem |
| 5:31AM |
4 |
Absolutely minimal Asterisk PSTN gateway |
| 5:24AM |
0 |
G.729 and Asterisk intellectual property issues |
| 5:03AM |
2 |
Asterisk 1.0 & Zaptel 1.0 -- False Hangup Disaster |
| 5:01AM |
0 |
Put Asterisk 1.0 mirrors into the Wiki |
| 3:48AM |
1 |
Only Accept Call After Pressing a Key '#' or '*' |
| 3:43AM |
1 |
chan_capi install problem |
| 2:58AM |
1 |
TDM400P Newbie configuration hell :-) |
| 1:12AM |
3 |
Debian Sarge, ISDN, CAPI and Asterisk blues |
| 12:19AM |
2 |
How to get Call Details Records |
| |
| Friday September 24 2004 |
| Time | Replies | Subject |
|---|
| 9:48PM |
1 |
chan_sccp.so: _use_ast_pthread_create_instead_ |
| 9:22PM |
0 |
getting dtmf in between an active conversation |
| 9:01PM |
0 |
PAP2 vs. PAP2-NA |
| 8:16PM |
2 |
What type of PRI setup is best |
| 5:52PM |
0 |
Astrison - many thanks - lots of enthusiams |
| 5:10PM |
0 |
Re: Setting [rx/tx]gain for spandsp/fax |
| 4:29PM |
1 |
Support |
| 3:33PM |
2 |
VICIDIAL and IAX |
| 2:59PM |
0 |
Two questions for Asterisk setup (Definity G3R and NFAS Trunk Gro ups) |
| 2:56PM |
3 |
ISDN (point to point) questions |
| 2:51PM |
2 |
CTI development |
| 2:44PM |
2 |
Call Groups |
| 2:11PM |
1 |
help with skinny |
| 2:05PM |
0 |
Asterisk MySQL CDR - Destination Number |
| 1:10PM |
2 |
Re: [Asterisk-Dev] Free G.729 ready for download |
| 11:51AM |
0 |
Intel IPP licensing and G.729 |
| 11:28AM |
1 |
Cisco Support Agreements |
| 11:27AM |
0 |
Calling to Broadvoice via Linux MASQ (NAT) |
| 11:18AM |
2 |
kernel: Power alarm on module 1, resetting! |
| 11:01AM |
0 |
Re: Thank you Mr. Mark Spencer and Asterisk |
| 10:53AM |
2 |
1.0 Libs |
| 10:50AM |
3 |
SMP support |
| 10:12AM |
2 |
Digium Closed Today? |
| 10:05AM |
0 |
Verso Call Manager |
| 9:50AM |
2 |
Free G.729 ready for download |
| 9:38AM |
0 |
app_queue |
| 9:31AM |
0 |
SER -- Asterisk , RTP Question. |
| 8:53AM |
5 |
TDM channel shows Offhook when I plug it to the telco |
| 8:42AM |
1 |
Cisco SIP Files |
| 8:09AM |
2 |
dev meeting bridge |
| 8:06AM |
5 |
Local Outbound Calls on PRI |
| 8:00AM |
0 |
how to put extension on hold? using h323 phones and gnu gatekeeper |
| 7:52AM |
1 |
No sound into asterisk??? |
| 7:39AM |
0 |
Asterisk skinny or sccp as softphone |
| 6:51AM |
0 |
How to transfer a call before the called party to answer |
| 6:42AM |
0 |
is the feature list online somewhere? |
| 6:38AM |
2 |
Asterisk as PSTN gateway |
| 6:27AM |
0 |
SIP - how does * decide codec order of preference |
| 6:25AM |
0 |
Fax Status |
| 5:17AM |
0 |
Fw: latest cvs / spandsp |
| 5:12AM |
0 |
latest cvs / spandsp |
| 4:13AM |
0 |
Cisco Phone 7960 |
| 4:11AM |
2 |
Case studies for 120 simultaneous calls on IVR |
| 4:00AM |
1 |
Call redirect with * |
| 1:54AM |
1 |
dynamic config |
| |
| Thursday September 23 2004 |
| Time | Replies | Subject |
|---|
| 10:18PM |
0 |
7960 Backlight project status? |
| 9:57PM |
0 |
problme with astcc |
| 8:48PM |
2 |
CallerID on Channelized T1 not working with 1.0.0 |
| 7:54PM |
2 |
How to set up a server compatible with Windows apps ? |
| 4:57PM |
1 |
IAXTel and Telesthetic |
| 3:18PM |
0 |
Duplicated INVITE in SIP session? |
| 2:15PM |
5 |
Billing Fun - anybody know where to get a NPA/NXX db? |
| 1:51PM |
2 |
viewing fax tiffs? |
| 1:37PM |
1 |
running 1.0 on macosx |
| 1:33PM |
1 |
Cisco 7960G, SIP, NAT, Qualify and Unreachable |
| 1:05PM |
1 |
Alternate MP3 Player |
| 12:56PM |
4 |
Asterisk 1.0 RPMS RH73 and RH9 |
| 12:53PM |
1 |
Diva Server PRI/E1-30 |
| 12:42PM |
1 |
new user From Canada |
| 12:07PM |
1 |
CDR (cdr_odbc) |
| 11:41AM |
2 |
New Mirror |
| 10:40AM |
2 |
Random Intermittent Noise for SIP to FX0 calls plus echo |
| 10:10AM |
8 |
GSM phones, bluetooth and general happiness |
| 9:22AM |
0 |
asterisk-1.0.0 woes |
| 9:20AM |
1 |
TDM400P FXO and Primus TalkBroadBand |
| 9:12AM |
1 |
send Flash via FXO |
| 8:07AM |
0 |
Asterisk 1.0.0 Mirror |
| 7:57AM |
3 |
app_valetparking / parking in general |
| 7:40AM |
0 |
Re: [Asterisk-Dev] Softphone for PocketPC or iPaq |
| 7:28AM |
12 |
Asterisk 1.0 released |
| 7:21AM |
10 |
MFC/R2 |
| 7:21AM |
11 |
1.0 Mirrors |
| 6:51AM |
3 |
Help with strategy for echo cancellation. |
| 6:40AM |
0 |
Monitor w/ m flag - Doesn't mux in some cases - Advice? |
| 6:37AM |
3 |
eyebeam |
| 5:32AM |
1 |
PRI(E1) Call recording with Digium cards? |
| 5:28AM |
3 |
GnomeMeeting and h323 |
| 4:30AM |
2 |
Cisco 2610XM and Asterisk |
| 3:52AM |
0 |
Redirecting incoming PRI to PSTN |
| 3:22AM |
0 |
MusicOnHold and Mp3 threads |
| 2:22AM |
2 |
Modem[i4l]/ttyI0 sent into invalid extension 's' |
| 2:03AM |
1 |
I can't solve mi problem compiling CAPI, please help |
| 1:59AM |
1 |
video via IAX or SIP |
| 1:56AM |
0 |
(no subject) |
| 1:14AM |
0 |
Eicon Diva PCI 2.02 (not server) |
| 12:33AM |
0 |
Cisco 30 VIP |
| 12:07AM |
0 |
Asterisk and Siemens HiPath |
| 12:06AM |
0 |
RE: An old problem still hanging around? |
| |
| Wednesday September 22 2004 |
| Time | Replies | Subject |
|---|
| 8:28PM |
1 |
News From Astricon |
| 8:03PM |
4 |
Softphone for PocketPC or iPaq |
| 7:33PM |
7 |
Some photos from Astricon 2004 |
| 6:55PM |
1 |
TDM400 synch issue |
| 4:10PM |
1 |
(euro)ISDN: complete silence / can't hear a word. |
| 3:36PM |
3 |
American vs English |
| 3:00PM |
2 |
SIP soft phones |
| 2:46PM |
1 |
T100P PRI Problem |
| 2:36PM |
4 |
zaphfc NT-mode can't dial outgoing |
| 2:25PM |
3 |
Galaxy Voice changed their SIP proxy |
| 2:06PM |
0 |
How an application could dialog with an external ivr ? |
| 12:44PM |
1 |
7960 SIP 7.2 keypress (not DTMF) problem |
| 12:28PM |
1 |
TE405P hardware question |
| 11:05AM |
1 |
IAX Config |
| 10:28AM |
0 |
make update and upgrade failed with `ZT_EVENT_POLARITY' undeclared |
| 10:15AM |
2 |
Problems compiling CAPI |
| 9:37AM |
2 |
MS SQL |
| 9:16AM |
0 |
SIP Clients Dont Clear |
| 9:07AM |
0 |
app_queue & cisco transfer |
| 9:05AM |
0 |
Siemens Optipoint 400 and Voice Mail |
| 8:44AM |
2 |
Cisco IP phone |
| 8:32AM |
1 |
'asterisk' displayed on my Cisco 7960 & 7912 ... |
| 7:37AM |
0 |
Re: Asterisk-Users Digest, Vol 2, Issue 216 |
| 7:20AM |
2 |
Transfering incoming calls using same line |
| 7:18AM |
4 |
PRI messages while running |
| 7:14AM |
8 |
Digium Hardware |
| 7:07AM |
0 |
IBM releases speech code as open source |
| 7:06AM |
18 |
Linksys PAP2-NA |
| 6:08AM |
1 |
OT: Hardware solutions to tie two offices together |
| 5:27AM |
1 |
Help: global (India/US) connection too expansive |
| 5:06AM |
1 |
'asterisk' displayed on my Cisco 7960 & 7912... |
| 4:52AM |
0 |
No Echo Cancellation with echocancel=yes |
| 3:45AM |
0 |
The previous reload command didn't finish yet |
| 2:46AM |
1 |
Opteron vs Xeon? |
| 2:05AM |
2 |
Grandstream bin cfg.txt generator |
| 1:40AM |
0 |
bug 2462 |
| 1:06AM |
1 |
Status of conference calls at Astricon ? |
| 12:13AM |
3 |
Optus Australia Multiline SHDSL service |
| |
| Tuesday September 21 2004 |
| Time | Replies | Subject |
|---|
| 7:23PM |
1 |
Asterisk , ISA Firewall/VPN , STUN and other |
| 6:03PM |
3 |
Uniden uip200 |
| 5:40PM |
2 |
Asterisk(OS X) & X-Lite |
| 5:20PM |
0 |
Astguiclient problems |
| 4:46PM |
1 |
iax2 notransfer=yes ignored |
| 4:43PM |
0 |
No call progress from * to E1 |
| 2:58PM |
3 |
Cisco 7905G |
| 2:53PM |
0 |
Asterisk + GnuGK :::: Unreachable Destination. |
| 2:38PM |
2 |
Asterisk , ISA Firewall/VPN , STUN and other issues |
| 2:01PM |
1 |
More than one OH323 Gatekeeper Registration |
| 1:40PM |
12 |
Astricon pictures |
| 1:35PM |
1 |
Cisco 7940/7960 and voicemailmain not able to press keys after a hold. |
| 1:33PM |
0 |
SIP Phone dropping calls, SIP Softphones working fine |
| 1:13PM |
1 |
asterisk voip only solution |
| 12:59PM |
0 |
Chan_capi: using both b-channels |
| 12:26PM |
1 |
Astricon meets? |
| 12:13PM |
1 |
Auto Attendant How To ? |
| 11:19AM |
0 |
Sayson/Aastra PT390 in Canada |
| 11:16AM |
1 |
Cisco 7905/7912 SIP image location (on Cisco's site) |
| 11:00AM |
1 |
HELP on AVM Fritz with CAPI drivers for SMP RH 9 |
| 10:43AM |
1 |
IP phones AT-723 or AT-323 |
| 10:20AM |
0 |
Zyxel P2000W or WiSIP with asterisk? |
| 10:16AM |
1 |
Polycom IP500 problem updating bootrom |
| 10:15AM |
0 |
Sanity Check --Zapras With T-1 |
| 10:05AM |
2 |
Anti Ex-Girlfriend feature for entire area codes? |
| 8:53AM |
0 |
ZAP problem / Strange State |
| 8:53AM |
0 |
Queue position and thankyou message plays even when queue is empty? |
| 8:44AM |
0 |
New astGUIclient version released 1.0.4 |
| 8:42AM |
0 |
T100P lost D channel |
| 8:23AM |
3 |
chan_sccp/SEP<mac>.cnf.xml |
| 8:17AM |
1 |
sipura registration problem |
| 8:16AM |
1 |
Need Help !! |
| 8:03AM |
0 |
more on spandsp and partially received fax |
| 7:56AM |
4 |
Voicemail forward to a remote server? |
| 7:41AM |
2 |
RC1 still broken with Cisco 7960? |
| 7:30AM |
0 |
Segmentation Fault TDM22B & TDM04B |
| 6:53AM |
0 |
Queues & Transfers |
| 6:52AM |
1 |
RDSI vs Analogic |
| 6:28AM |
0 |
Question/Future Request for Call Queues |
| 6:22AM |
2 |
Can someone suggest |
| 6:01AM |
2 |
Basic ISDN Access |
| 5:32AM |
2 |
ISDN problem: lacking dialtone |
| 5:20AM |
2 |
SIP termination in Brazil |
| 5:20AM |
3 |
FreeBSD 100% cpu |
| 4:28AM |
1 |
zaptelrtc for 2.6.x |
| 1:55AM |
2 |
TDM400P: RJ45 to RJ11 |
| 12:38AM |
1 |
Faxing thru freshtel |
| |
| Monday September 20 2004 |
| Time | Replies | Subject |
|---|
| 11:45PM |
9 |
panasonic KX-TD1232 |
| 10:08PM |
0 |
how do I get R2 signalling working with a Digium |
| 10:08PM |
1 |
spandsp / I get only garbage in my faxes |
| 10:01PM |
0 |
Notes from Newbie |
| 6:38PM |
3 |
fax autoanswer |
| 5:09PM |
3 |
asterisk install in a home with regular phones and a x100p |
| 3:33PM |
0 |
SIP peers in MySQL Database and accountcode lack |
| 3:15PM |
0 |
[QUAR] How can I make a rotative board? |
| 2:54PM |
1 |
ZapRTC loading problems |
| 2:46PM |
3 |
Question about the 'fax' extension |
| 2:22PM |
4 |
How can I make a rotative board? |
| 2:07PM |
2 |
1 extension entry for multiple purposes? |
| 1:58PM |
0 |
MySQL Directory Patch |
| 1:40PM |
0 |
Connect asterisk on classic pbx withisdn card |
| 1:35PM |
1 |
Nortel/Bell Canada Vista-350 ADSI |
| 1:28PM |
0 |
Can't Dial using perl. |
| 1:21PM |
1 |
Configure an TDM04B & TDM22B |
| 1:09PM |
0 |
Error compiling astersik-oh323 |
| 12:25PM |
6 |
Newbie has a few basic questions please. |
| 12:17PM |
1 |
wcfxo load problem |
| 12:11PM |
2 |
Voicemail Directory |
| 10:02AM |
1 |
spandsp core dumps asterisk receiving fax |
| 9:54AM |
1 |
Cannot do international dial with E1 in Germany |
| 9:43AM |
0 |
Call failed to go through, reason x |
| 9:29AM |
0 |
Spellcaster ISDN BRI card |
| 9:11AM |
0 |
Unable to request channel Zap/r1/... |
| 9:04AM |
2 |
Garbled voice on long distance calls |
| 8:39AM |
0 |
Installation problem; collect2: ld returned 1 exit status |
| 8:24AM |
6 |
SER + Asterisk |
| 8:20AM |
2 |
Polycom ip500 dial prob |
| 7:54AM |
0 |
Manager redirect action does not appear to work in some cases. |
| 7:17AM |
2 |
Cisco 76XX - How to ignore a call (silence ring) |
| 7:12AM |
4 |
spandsp / compilation errors |
| 7:04AM |
2 |
H.323 call problemm (no sound) |
| 6:59AM |
0 |
problem with dialing |
| 6:43AM |
0 |
connect asterisk to isdn bri connection |
| 5:50AM |
5 |
iax2_read: I should never be called |
| 5:41AM |
0 |
add iax user |
| 4:45AM |
1 |
can't compile chan_capi 0.3.5 under SuSE 9.0 |
| 4:37AM |
6 |
PBX CallTransfer |
| 4:06AM |
0 |
virtual T38modem with hylafax over asterisk OH323 chans |
| 3:15AM |
2 |
Update: Welltech Wellgate 3504A registration problem |
| 3:10AM |
1 |
Wait() |
| 2:46AM |
0 |
Dialplan Configuration with MYSQL |
| 1:37AM |
2 |
CallerID in Queue |
| |
| Sunday September 19 2004 |
| Time | Replies | Subject |
|---|
| 11:41PM |
1 |
how do I get R2 signalling working with a Digium E100P E1/PRA Card |
| 10:04PM |
1 |
Cisco Phones (7960, 7905G) and tdm400p with FXS for sale |
| 9:59PM |
2 |
Effectively using a telco Type 102 Milliwatt Test line with ztmon itor -v to set txgain/rxgain in zapata? |
| 7:31PM |
0 |
Timing source on SMP system - DisableRTC forzaprtc |
| 6:46PM |
0 |
Re:Asterisk and Red Hat 9 (Henry Devito) |
| 5:56PM |
2 |
passing octothorpe |
| 5:51PM |
0 |
PC Requirements |
| 5:41PM |
2 |
Setting time on ADSI phones from Asterisk |
| 5:25PM |
0 |
How does Asterisk interact with an h323 gateway |
| 4:01PM |
1 |
How To get response of command from another socket |
| 3:30PM |
7 |
Asterisk and Red Hat 9 |
| 2:48PM |
1 |
Timing source on SMP system - Disable RTC forzaprtc |
| 2:42PM |
1 |
Using queue app with external members/destinations |
| 2:21PM |
1 |
Re: X100p on VIA EPIA-V |
| 1:11PM |
0 |
New Zealand Supplier |
| 11:58AM |
2 |
Timing source on SMP system - Disable RTCforzaprtc |
| 11:13AM |
2 |
Timing source on SMP system - Disable RTC for zaprtc |
| 8:22AM |
6 |
new ATA box for sale by Linksys |
| 7:12AM |
1 |
openh323 compile for Asterisk |
| 5:29AM |
4 |
X100p on VIA EPIA-V problems |
| 2:46AM |
1 |
Is Dual CPU machine solution for using Asterisk with other general apps (like home automation, web server, ...) in home environment ? |
| 2:14AM |
1 |
Dial 0 to outbound |
| 1:12AM |
1 |
RE: [Asterisk-Dev] Hardware details for the Digium TDM400P |
| 1:11AM |
1 |
vim ftplugins for asterisk? |
| 12:53AM |
1 |
[OT^2]: Getting at the fan on an IBM Thinkpad 600E Laptop? |
| 12:03AM |
0 |
problem to reach sip client which is connected to asterisk when t he call coming from another sip server. |
| |
| Saturday September 18 2004 |
| Time | Replies | Subject |
|---|
| 8:23PM |
2 |
Timing source on SMP system |
| 8:03PM |
0 |
New MOH stream for each queue member? |
| 7:06PM |
1 |
Asterisk stopped answering the calls |
| 1:54PM |
1 |
13 sec. delay what is causing it? |
| 1:23PM |
1 |
NEWBIE - No Audio on ISDN BRI (Teles PCI) |
| 12:03PM |
3 |
uk caller id |
| 11:41AM |
2 |
IP Intercom's |
| 10:56AM |
2 |
Asterisk as an outbound call machine? |
| 10:39AM |
0 |
Quintum A800 and asterisk |
| 9:28AM |
1 |
First time asterisk installation problem |
| 3:30AM |
9 |
No sound |
| 2:47AM |
0 |
RSS Feed Added to Asterisk News Site |
| 1:31AM |
2 |
call center application |
| |
| Friday September 17 2004 |
| Time | Replies | Subject |
|---|
| 11:42PM |
2 |
Caller ID with DTMF |
| 10:52PM |
1 |
Canreinvite=??? |
| 9:05PM |
3 |
how to get caller ID |
| 8:13PM |
1 |
Connecting SPA-300 to Asterisk |
| 7:53PM |
1 |
? |
| 7:52PM |
0 |
anyone can see response of a request from other connections? |
| 5:35PM |
1 |
Zaptel compile error - unresolved symbols |
| 3:52PM |
5 |
Background() command |
| 2:50PM |
1 |
ZAPTEL Compile Problem? |
| 2:30PM |
0 |
new Cepstral voice's TEST number |
| 2:18PM |
3 |
Medium volume 100% SIP/IAX PBX. |
| 2:04PM |
0 |
MySQL Voicemail and Directory Patch |
| 1:56PM |
3 |
MySQL Voicemail Problems |
| 1:02PM |
0 |
AnnounceOveride |
| 12:51PM |
3 |
Astricon |
| 12:40PM |
3 |
FC2 zaptel compile failure |
| 12:27PM |
0 |
using Astriskfor MGCP |
| 11:32AM |
3 |
Cisco 7940/7960 QOS? |
| 11:17AM |
0 |
Re: Asterisk forum created http://ASTERISK.XVOIP.COM |
| 11:05AM |
1 |
Ackcall works for sip, not for zap |
| 10:13AM |
2 |
AB1 |
| 10:11AM |
1 |
No sound from IVR scripts, yet calls placed without any problem. |
| 10:06AM |
9 |
Asterisk forum created |
| 10:02AM |
2 |
New User Help |
| 9:20AM |
2 |
Re: Asterisk-Users Digest, Vol 2, Issue 163 |
| 9:09AM |
6 |
Agents and Queues |
| 9:00AM |
2 |
Suppressing CallerID in .call files |
| 8:30AM |
8 |
cisco 7960 CTLSEP |
| 8:20AM |
1 |
Permanently logged in agents? |
| 7:01AM |
1 |
AGI Python Clear or Channel Failure? |
| 6:26AM |
4 |
SS7 E1 cards |
| 6:13AM |
2 |
Transferring Calls |
| 6:08AM |
1 |
Asterisk and Norstar 0X32 MICS |
| 5:41AM |
0 |
dtmfmode with kphone |
| 5:35AM |
1 |
let incoming callers contact a certain extension... |
| 4:52AM |
2 |
Error in zapata/zaptel configuration |
| 4:21AM |
1 |
How would you handle a fax without T.38orG.711uLaw? |
| 3:21AM |
1 |
AW: dial '0' for outside line and get a dialtone... |
| 2:51AM |
0 |
paly answering sounds |
| 2:43AM |
2 |
dial '0' for outside line and get a dialtone... |
| 1:50AM |
1 |
caller id? |
| 1:35AM |
1 |
Silently Wait for DTMF Input |
| 1:25AM |
0 |
OT: FWD Iax |
| 12:22AM |
8 |
English vs American voice files |
| 12:17AM |
1 |
Issue with TE405P and Adaptec U160 SCSI |
| |
| Thursday September 16 2004 |
| Time | Replies | Subject |
|---|
| 11:17PM |
0 |
Problem in Dialing |
| 8:56PM |
0 |
Predictive Dialer, Web & Inbound Phone System |
| 8:02PM |
3 |
Creating conference calls from within Astman. |
| 8:00PM |
0 |
ISDN BRI termination via Cisco? |
| 7:49PM |
3 |
SIP Phone -> PBX Phone |
| 7:23PM |
2 |
FW: Polycom IP500 |
| 6:51PM |
5 |
reverse the selection order of zap channels for outgoing calls |
| 6:49PM |
0 |
apologies if last message was sent multiple times... |
| 6:03PM |
1 |
SIP channel stuck after registration |
| 5:00PM |
0 |
Conf file for an Avaya 4624 ip phone |
| 4:17PM |
0 |
Dial command r option |
| 4:12PM |
1 |
Non-PRI T1 showing red |
| 2:48PM |
3 |
Playing GSM files |
| 2:27PM |
1 |
How would you handle a fax without T.38 or G.711uLaw? |
| 2:09PM |
2 |
H323 dialing makes Asterisk crash |
| 1:57PM |
0 |
No Caller Name sent from Asterisk over Natio nal or DMS100 PRI to a Norstar MICS? |
| 12:32PM |
1 |
Sprint PCS -> Asterisk through Digium TDM400P |
| 11:42AM |
5 |
Earthlink Releases SIP Based P2P File-Sharing App |
| 11:22AM |
1 |
Static noise and server locked when using two 4FXO tdm400p pci cards |
| 10:36AM |
1 |
IAX2 only asterisk scalability |
| 10:18AM |
1 |
Unable to dial using SIP using FWD and iConnectHere |
| 9:41AM |
1 |
Transfer and Release of a call out to PSTN |
| 9:40AM |
3 |
IAX- FAX |
| 9:16AM |
1 |
ID for outgoing calls from DDI (DID) line |
| 9:06AM |
0 |
spandsp on current cvs? |
| 9:06AM |
0 |
What can you do with Asterisk in Brazil following the law |
| 9:02AM |
0 |
call parking & forwarding |
| 8:54AM |
1 |
Beyond T1 |
| 8:20AM |
0 |
Re: No Caller Name sent from Asterisk over National or DMS100? |
| 7:20AM |
2 |
Uniden UIP-200 Multiple line appearances |
| 7:08AM |
1 |
${CONTEXT} variable |
| 6:55AM |
1 |
ERROR[16384]: chan_h323.c:1987 load_module: Gatekeeper registration failed |
| 6:23AM |
0 |
Language settings Cisco 7960 |
| 6:11AM |
0 |
3 Way Calling on Snom Phones and Asterisk |
| 5:58AM |
0 |
problem connecting to icallglobe |
| 4:23AM |
2 |
Help with E1 configuration |
| 4:22AM |
2 |
Audiocodes Mediant 2000 |
| 4:17AM |
2 |
Current bristuff error report |
| 3:25AM |
1 |
ZAP Hook flash / recall on active zap interface |
| 2:48AM |
2 |
No Caller Name sent from Asterisk over National or DMS100 PRI to a Norstar MICS? |
| 2:13AM |
0 |
Thoughts on Adding Locking to db.c? |
| 1:34AM |
0 |
H323 - Control Protocol Error (Master slave Determination) |
| 1:27AM |
1 |
Problems with native h323 channel on Asterisk RC2: No early audio and codec negotiation issues |
| 1:11AM |
1 |
quality of musiconhold... |
| 12:43AM |
0 |
Receiving queue urls |
| 12:33AM |
0 |
ftp.digium.com/pub/asterisk/webmin |
| 12:20AM |
0 |
transfering a call |
| |
| Wednesday September 15 2004 |
| Time | Replies | Subject |
|---|
| 11:31PM |
0 |
codec trouble? |
| 10:02PM |
1 |
Zap: busydetect & busycount |
| 8:20PM |
3 |
SIP Options |
| 5:39PM |
0 |
Re: question on VoIP setup |
| 4:13PM |
0 |
No Audio in Voicemail |
| 3:10PM |
4 |
Fax and Asterisk |
| 2:51PM |
1 |
Asterisk is not "picking up the phone" with a x100p card |
| 1:43PM |
1 |
Channel H323, RH9, OpenH323_1.12.2, pwlib_1.5.2 +GnuGK |
| 10:38AM |
1 |
Extension based call forwarding using capiECT |
| 9:19AM |
1 |
Sending IAX2 calls back to a registered client |
| 9:05AM |
0 |
Static Problem... Ahhh! |
| 8:56AM |
3 |
Cisco 79xx + asterisk + some functions Q |
| 8:25AM |
0 |
Asterisk and Cisco MC3810 Help needed |
| 8:16AM |
2 |
Results of 13 month study on reducing telemarketing calls |
| 7:55AM |
0 |
Question calling number |
| 7:40AM |
0 |
Asterisk SIP gateway --> SCCP Phone |
| 7:23AM |
1 |
Transfer / Music-On-Hold |
| 7:21AM |
0 |
design check |
| 7:19AM |
0 |
chan_capi and outgoing calls |
| 7:10AM |
1 |
voicebox |
| 6:06AM |
3 |
call recording and CDR "feature" discovered? |
| 4:45AM |
1 |
RC2 zaptel compile problem |
| 4:39AM |
1 |
phone line "roaming" |
| 3:29AM |
1 |
* and Philips IS3090 PBX |
| 3:22AM |
1 |
Not register |
| 2:45AM |
4 |
IAX to IAX connect question |
| 2:14AM |
0 |
IAX2 call drop |
| 2:06AM |
0 |
What FXO cards (1,2 or 4 channels) in Europe ? |
| 1:43AM |
1 |
capiHOLD and capiECT |
| 1:02AM |
0 |
incoming calls to a soft phone |
| 12:44AM |
0 |
AGI didn't get var from Asterisk? |
| 12:32AM |
3 |
ztdummy on Fedora Core 2 |
| |
| Tuesday September 14 2004 |
| Time | Replies | Subject |
|---|
| 11:48PM |
3 |
Fw: Asterisk R2 Signaling |
| 11:26PM |
4 |
One Question:CLI dial cmd |
| 9:06PM |
4 |
Sending Caller ID info in MD/USA |
| 6:25PM |
2 |
Press 9 to dial by name |
| 6:13PM |
2 |
Warn before Absolute Timeout |
| 5:54PM |
1 |
Polycom IP600 and instant messaging |
| 4:57PM |
0 |
Problem with hangup |
| 3:04PM |
2 |
3-way calling |
| 2:13PM |
0 |
Cepstral available |
| 2:06PM |
1 |
ERROR: cannot load module kernelcapi |
| 12:56PM |
1 |
Patching UK Caller ID |
| 12:56PM |
1 |
i4l "1 second patch", anyone got it? |
| 12:25PM |
1 |
cannonicalizing phone num in macro |
| 12:16PM |
1 |
Agents on zap channels must acknowledge calls even with ackcall=no |
| 11:32AM |
0 |
Chanspy updated |
| 11:20AM |
2 |
Spawn extension.....exited non-zero |
| 11:10AM |
1 |
Clarification - FAX on local network |
| 10:52AM |
1 |
Detecting DTMF tones |
| 10:25AM |
1 |
Using Asterisk as a replacement for a Merlin Legend. |
| 10:21AM |
1 |
asterisk does not start... |
| 9:58AM |
2 |
Asterisk not outputting real time display |
| 9:53AM |
0 |
video softphones |
| 9:32AM |
1 |
Openswitch12 |
| 9:30AM |
1 |
Manager events logic depends on channel type? |
| 8:35AM |
0 |
MeetMe - waiting for marked user |
| 8:00AM |
0 |
SIP registrations CVS Head |
| 7:57AM |
0 |
*called* id name display? |
| 7:48AM |
1 |
Comparisons between * and sipXpbx (PingTel's open source product) |
| 7:47AM |
0 |
RE. compiling zaptel |
| 7:27AM |
0 |
Detecting DTMF reliably |
| 7:20AM |
0 |
Get Connected With Kingston A How To Guide |
| 7:15AM |
1 |
Setting up Asterisk with fwd |
| 6:54AM |
0 |
SIP call server- Too many hops |
| 6:33AM |
2 |
Mitel 5010 +5220 |
| 5:48AM |
2 |
Use ISP's SIP account for IP-PSTN gateway |
| 5:24AM |
1 |
Cheap Sams computer good for tdm400? |
| 5:17AM |
3 |
OH323 Trunking |
| 4:35AM |
1 |
cvs stable |
| 3:47AM |
1 |
Newbie question: X101P card - Asterisk - /dev/dsp0 |
| 3:26AM |
1 |
What does 'Forbidden (From header is not a Trust host or gateway)' mean? |
| 3:05AM |
1 |
Requested device 'ttyI1' does not exist |
| 2:30AM |
1 |
Wrong ID going out... |
| 1:01AM |
3 |
how to route these outgoing calls? |
| 12:23AM |
0 |
softphone crash? |
| |
| Monday September 13 2004 |
| Time | Replies | Subject |
|---|
| 10:37PM |
0 |
Making the Old PABX work with new * box |
| 9:36PM |
4 |
asterisk make |
| 8:56PM |
1 |
Read command without # |
| 8:00PM |
1 |
agents and *8 pickupgroups |
| 7:08PM |
3 |
Aasterisk SIP<->SIP No audio |
| 5:34PM |
4 |
PABX & VOIP Gateway |
| 4:42PM |
1 |
chan_sip2 Install Question |
| 4:04PM |
0 |
Sipura-3000 Assistant for Asterisk on MacOSX? Well, maybe, with your help! |
| 3:41PM |
1 |
Extending E1's over a Satellite link |
| 2:59PM |
0 |
voicepulse problems since new configs |
| 2:44PM |
0 |
Codec usage in iax.conf |
| 2:42PM |
2 |
Sip Outbound Proxy |
| 1:55PM |
0 |
Arrgh, Broadvoice, SIP.conf |
| 1:33PM |
0 |
IAXy loud static problem |
| 12:37PM |
0 |
Registering asterisk with FWD |
| 12:35PM |
2 |
allowing/disallowing codecs in dialplan? |
| 12:31PM |
7 |
festival |
| 11:43AM |
0 |
Dialplan transfer. (h323 transfer) |
| 11:43AM |
0 |
Dial-plan transfer |
| 11:09AM |
1 |
Caller ID "forwarded" to analog phone? |
| 10:38AM |
0 |
WhoIsIt -- a contributed utility |
| 10:27AM |
3 |
Alchemy branch integration, one way audio |
| 10:08AM |
1 |
Server load capabilities |
| 9:10AM |
3 |
Astersk as AVAYA IVR |
| 9:00AM |
0 |
iax2 transfer and CDRs |
| 8:50AM |
0 |
CVS lock directory still not fixed? |
| 8:23AM |
2 |
Astricon tutorials :: Open for registration again |
| 8:12AM |
0 |
Zaprtc help |
| 7:52AM |
0 |
test membership |
| 7:45AM |
0 |
CDR database. |
| 7:45AM |
0 |
Post to list |
| 7:42AM |
8 |
Playback Fileformats |
| 7:35AM |
0 |
Asterisk daemon start errors |
| 7:25AM |
1 |
IAXy DHCP lease not renewing |
| 7:22AM |
1 |
Simple Caller-ID match and block and/or play voice file saying you're calling too much or don't call! |
| 7:02AM |
2 |
unavail and busy. |
| 6:42AM |
0 |
Agentlogin incorrect |
| 6:19AM |
0 |
IBM to Open Voice Recognition Software |
| 4:53AM |
5 |
music on hold not strting |
| 4:13AM |
4 |
Unknown RTP codec 72 received |
| 3:40AM |
1 |
Red Alarm - Config Zaptel card |
| 3:39AM |
1 |
(no subject) |
| 2:57AM |
1 |
ProSLIC and measuring of PSTN parameters like Voltage, Polarity, Power (A) and Frequency (Hz) |
| 2:30AM |
1 |
SIP Remote-Party-ID |
| |
| Sunday September 12 2004 |
| Time | Replies | Subject |
|---|
| 11:38PM |
4 |
One Question |
| 10:22PM |
0 |
RE: No subject by Steve M |
| 10:01PM |
1 |
(no subject) |
| 9:54PM |
0 |
iconnecthere DTMF detection |
| 8:41PM |
2 |
New BudgeTone |
| 7:46PM |
1 |
IAX2 crash course wanted |
| 5:12PM |
1 |
detecting fax and passing it to Hylafax |
| 3:37PM |
1 |
Monitor and AGI - doesn't record much! |
| 1:50PM |
2 |
Multiple MD 3200 (Intel 537) cards on a single system. |
| 1:27PM |
1 |
SetGroup Limitation!!! |
| 12:29PM |
2 |
Overriding SIP From Header |
| 11:35AM |
2 |
(no subject) |
| 10:59AM |
3 |
Final Help on setting up x100p |
| 9:02AM |
1 |
TN405P running but with errors |
| 7:28AM |
2 |
Grandstream Budgetone 100 Caller ID shows extension, not incoming Caller ID |
| 5:27AM |
2 |
sipphone dial out problems.. |
| 1:54AM |
0 |
sip does not bind all addreses |
| 1:05AM |
1 |
Voice from one call carried on to next call |
| 12:18AM |
2 |
GSM / Radio |
| |
| Saturday September 11 2004 |
| Time | Replies | Subject |
|---|
| 11:36PM |
1 |
mknod /dev/phone0 c 100 0 |
| 10:08PM |
2 |
VoIP Telephony with Asterisk by Paul Mahler |
| 9:19PM |
25 |
Broadvoice |
| 8:52PM |
1 |
Audio from GS to asterisk double speed |
| 8:17PM |
0 |
Problems with Call Progress and fax detection on PRI |
| 7:41PM |
2 |
TDMoE questions |
| 6:51PM |
1 |
IAX not binding to the right port |
| 6:51PM |
0 |
How to make a call from command line |
| 6:34PM |
1 |
IAXy intermittent sound problem |
| 5:47PM |
0 |
h.323 Transfer |
| 5:43PM |
1 |
creating device=/dev/phone0 |
| 4:30PM |
2 |
Leading '0's and what do 'pri_dialplan', 'pridialplan' and 'prilocaldialplan' in zapata.conf do? |
| 4:20PM |
0 |
Grandstream x Asterisk 1.0 RC1 x VOIPJet |
| 1:40PM |
2 |
Questions about PRI lines for modem banks and Asterisk |
| 1:13PM |
1 |
Compilation error with 2.6 kernel |
| 11:44AM |
1 |
Final status of the call |
| 11:40AM |
2 |
Help!!!!! |
| 11:39AM |
2 |
Audio level in compressed wav files |
| 10:41AM |
0 |
Spandsp garbage |
| 9:29AM |
3 |
FWD |
| 9:00AM |
0 |
DTMF signaling with GSM codec |
| 5:19AM |
1 |
call park question |
| 2:46AM |
0 |
call forwarding when busy - single pots blues |
| 2:30AM |
0 |
Call Queues, CallerID, SIP and AutoDial |
| 12:57AM |
1 |
Questions about cdr |
| |
| Friday September 10 2004 |
| Time | Replies | Subject |
|---|
| 7:47PM |
2 |
Suggested Motherboard for TE410P |
| 6:57PM |
1 |
(Resend) Trouble with all linux sip softphones.... And asterisk/linphone/kphone SRPMs |
| 5:39PM |
0 |
chan_agent and SIP UA transfers fail |
| 5:03PM |
1 |
Can't get ChanSpy to work |
| 3:21PM |
4 |
SIP on Handhelds |
| 3:03PM |
8 |
Organization wide |
| 2:05PM |
0 |
Definity <-> Asterisk w/callerid |
| 1:53PM |
0 |
Proposal regarding the "*80" vertical service code |
| 1:41PM |
0 |
RDNIS and Q.931 |
| 1:06PM |
1 |
moh cell phones |
| 1:04PM |
0 |
SIP Dropped Calls |
| 12:53PM |
2 |
What would be required for this? |
| 12:46PM |
1 |
Sangoma S508 Rev-B |
| 12:09PM |
2 |
SpanDSP/RxFax anomalies... |
| 11:37AM |
3 |
call quality monitoring |
| 10:44AM |
1 |
Net2Phone, Asterisk, and "404 Not Found" |
| 10:21AM |
0 |
Re: Problem with Openh323 channel driver |
| 9:46AM |
0 |
Re: Asterisk-Users Digest, Vol 2, Issue 94 |
| 8:58AM |
1 |
Problem with stuttering on TE410P |
| 7:27AM |
1 |
Call Parking Problem |
| 7:21AM |
0 |
pridialplan & nationalprefix |
| 5:38AM |
14 |
Asterisk newbie questions |
| 5:32AM |
1 |
No DTMF or Audio |
| 4:55AM |
1 |
Problems with 0penh323 Channel Driver |
| 4:50AM |
4 |
sip.conf from mysql |
| 3:35AM |
2 |
Asterisk and VoDSL |
| 1:58AM |
1 |
Netmeeting i can't hear voice |
| 1:26AM |
2 |
Snom 200 updates |
| 1:06AM |
3 |
Asterisk testbed for teaching connecting to a PRI-ISDN |
| |
| Thursday September 9 2004 |
| Time | Replies | Subject |
|---|
| 11:09PM |
0 |
Asterisk server keeps crashing |
| 7:54PM |
3 |
Store data from call to database |
| 5:57PM |
1 |
DevKit TDM400P module won't load |
| 2:29PM |
1 |
Debian Sarge -- cvs vs. apt-get |
| 2:23PM |
1 |
Unknown IE 40 (cs6, Unknown Information Element) |
| 2:13PM |
2 |
Conference Phone |
| 2:06PM |
1 |
Uniden UIP 200 |
| 2:04PM |
0 |
What are ZOMBIES and why am I getting them? |
| 1:27PM |
0 |
two asterisk boxes and using outgoing spool file on second box to call out on first box. |
| 1:19PM |
1 |
astcc not working |
| 1:07PM |
4 |
IAX2 dropping call? |
| 12:59PM |
10 |
Cepstral |
| 12:55PM |
0 |
rate_engine substitue db field? |
| 12:40PM |
0 |
Polycon IP 300 SIP vs Grandstream BT-101Deployment |
| 12:34PM |
1 |
UIP-200 conference call |
| 12:06PM |
1 |
Festival Speech Synthesis 1.95:beta July 2004 Eval |
| 12:04PM |
0 |
Astricon News :: Tutorials are now fully booked |
| 11:29AM |
12 |
SNOM 200 can't conference. |
| 11:16AM |
3 |
Caller-ID name lookup via anywho.com |
| 11:10AM |
2 |
Dial Out w/ OH323 |
| 10:36AM |
0 |
Dialing Out through Provider with Authentica tion |
| 10:11AM |
0 |
OT: how much are polycom phones in the UK |
| 9:56AM |
2 |
Legacy Toshiba Phones |
| 9:55AM |
0 |
Aruba Origination |
| 9:53AM |
3 |
Polycom IP500 vs Cisco 7940 |
| 9:34AM |
0 |
Re: Asterisk-Users Digest, Vol 1, Issue 5082 |
| 9:33AM |
1 |
Virtual queue member |
| 9:26AM |
0 |
Telcordia TBCT |
| 8:53AM |
3 |
Simple question about SIP community |
| 8:15AM |
0 |
Asterisk not playing sounds after Kernel upgrade? |
| 7:48AM |
2 |
Fax relaying with T.38 |
| 6:38AM |
0 |
Queues : Rings even when the agent is on a call |
| 6:20AM |
3 |
Dialing Out through Provider with Authentication |
| 6:16AM |
3 |
weird routing(?) problem with 2 Asterisk servers |
| 4:53AM |
0 |
ser+ asterisk |
| 3:44AM |
0 |
russian sound files |
| 2:30AM |
1 |
Problems to setup ast_data with asterisk. |
| 2:04AM |
0 |
zaphfc errors |
| 1:28AM |
0 |
Ordinary phones can call into asterisk - but * does not recognize the dtmf signals |
| 1:09AM |
1 |
Dialing pstn-asterisk |
| |
| Wednesday September 8 2004 |
| Time | Replies | Subject |
|---|
| 11:50PM |
0 |
transfer on a zaptel FXO port |
| 11:26PM |
0 |
Asterisk & the Micronet SP5210 anyone? |
| 6:51PM |
2 |
Zaptel and Linux Distros |
| 6:01PM |
1 |
New ChanSpy and MOH Patch |
| 5:13PM |
0 |
IAXy/S100I reomote PBX extension provision |
| 4:40PM |
0 |
Spontaneous Hangup occuring |
| 3:05PM |
0 |
T100P calls with playback starts speaking be fore pickup |
| 2:35PM |
0 |
Disa extension entry timeout |
| 2:29PM |
4 |
Cisco GW and DTMF problems |
| 2:18PM |
0 |
T100P calls with playback starts speaking before pickup |
| 1:03PM |
2 |
My AGI is not detecting hangups on outgoing calls |
| 12:42PM |
2 |
'Hangup' not hanging-up, is this intended behaviour? |
| 12:40PM |
0 |
Changed * server to static non-nat IP from nat |
| 12:39PM |
2 |
How do I get DIDs for remote areas in Canada |
| 12:35PM |
1 |
successful echo cancellation!!! (multitech) |
| 11:44AM |
0 |
Driving MWI on Norstars (was Maximum tollera ble lag/jitter...) |
| 11:38AM |
1 |
Polycom SIP 1.3.1 & Reject Button |
| 11:28AM |
2 |
Asterisk with Primus Talkbroadband |
| 11:06AM |
1 |
zap: reroute incoming calls to dedicated channel |
| 10:53AM |
1 |
Intertex IX66 |
| 10:18AM |
0 |
stale voicemail messages / greeting |
| 8:40AM |
0 |
Directory command assistance |
| 8:14AM |
1 |
OH323 Ignoring PROGRESS indication |
| 7:50AM |
2 |
PRI issue |
| 7:41AM |
3 |
Where to post SuSE 9.x startup script? |
| 7:31AM |
1 |
Polycon IP 300 SIP vs Grandstream BT-101 Deployment |
| 6:28AM |
2 |
Help needed! |
| 6:21AM |
1 |
Problem playing file with G729A |
| 6:14AM |
1 |
accept DTMF while beeing in a queue |
| 5:55AM |
0 |
asterisk+chan_h323+redhat9 troubles |
| 5:29AM |
4 |
WellGate 3504A with Asterisk SIP authentication and config |
| 5:16AM |
1 |
SIP and */# |
| 4:54AM |
1 |
asterisk console from xinetd? |
| 4:10AM |
3 |
sendmail&hostname |
| 3:37AM |
0 |
zaphfc strange errors |
| 3:27AM |
1 |
Assigning a higher irq to a digium card |
| 2:04AM |
0 |
re: asterisk, SER and autocreatepeer |
| 1:43AM |
2 |
X-Lite & Meetme problem |
| 1:31AM |
3 |
Newbie: Only allow authenticated users to call |
| 1:08AM |
2 |
'connecting' voip-numbers to our Asterisk |
| 12:20AM |
2 |
Answer confirmation on non-Zap channels? |
| 12:14AM |
3 |
astwind has any one got this thing to work? |
| |
| Tuesday September 7 2004 |
| Time | Replies | Subject |
|---|
| 9:50PM |
0 |
Monitored outbound dialing via Zap interface ? |
| 9:16PM |
1 |
QSIG against a Nortel/Meridian PBX |
| 8:43PM |
4 |
Caller id and the number of rings |
| 8:41PM |
1 |
Got *80 working ... now some Blacklist questions |
| 8:12PM |
6 |
Problems with length of voicemail |
| 6:44PM |
1 |
astcc dont write to the table cdrs or cards |
| 5:55PM |
2 |
Maximum tollerable lag/jitter for IAX2 w/o j itterbuffer enabled? |
| 5:48PM |
1 |
REPOSTED: Problems patching Makefile in apps directory |
| 5:04PM |
1 |
Compiling on Mac OS X (10.3.5) |
| 4:54PM |
4 |
Maximum tollerable lag/jitter for IAX2 w/oji tterbuffer enabled? |
| 4:26PM |
3 |
Maximum tollerable lag/jitter for IAX2 w/o jitterbuffer enabled? |
| 2:52PM |
1 |
Monitored outbound dialing via Zap interface? |
| 2:43PM |
0 |
chan_h323: remote ip address -> context |
| 2:36PM |
0 |
T100P problem with LD T1 |
| 2:14PM |
0 |
Bristuff wackyness - not answering |
| 2:11PM |
0 |
OH323 return call from openphone to sip? |
| 1:48PM |
1 |
Checking Return Codes |
| 1:37PM |
6 |
MySQL on another host? |
| 1:25PM |
3 |
DTMF Caller ID w/o polarity inversion |
| 12:54PM |
1 |
MeetMe without ZAP? |
| 12:13PM |
1 |
Tormenta & Asterisk |
| 11:56AM |
0 |
FWD registration sort of expiring |
| 11:10AM |
0 |
new Asterisk resources site |
| 10:49AM |
0 |
OT - Experience using Gmail for AsteriskMail ingList |
| 10:13AM |
0 |
Cisco 7960G SIP Registration Timeouts |
| 9:46AM |
1 |
MOH/mpg123 broken when running asterisk as non-root? |
| 9:27AM |
2 |
TE410P in Germany |
| 9:16AM |
1 |
Nortel PC client |
| 8:34AM |
2 |
OT - Experience using Gmail for Asterisk Mailing List |
| 8:29AM |
1 |
Performance Specs |
| 8:18AM |
0 |
GRQ |
| 8:16AM |
0 |
phones and atas timeout..force * restart |
| 7:44AM |
0 |
extension mobility with cisco phones |
| 7:21AM |
1 |
Asterisk + NetJet (ISDN4Linux) |
| 6:43AM |
0 |
Problems patching Makefile in apps directory |
| 6:27AM |
1 |
Cisco 7912 issues |
| 6:24AM |
2 |
Crossed lines - a worrying problem. |
| 5:51AM |
0 |
sending SIP Message 404 out of extension.conf |
| 5:10AM |
0 |
Country specificals-- Incomplete |
| 4:57AM |
0 |
Country specificals |
| 4:37AM |
0 |
Conference Call Query? |
| 3:56AM |
0 |
voip gateway connect to a pbx |
| 3:45AM |
3 |
H323 Control Protocol Error |
| 2:11AM |
0 |
SRV lookup fails after DNS update |
| 1:47AM |
0 |
[Asterisk-User] Problem with bus s0 and analog lines |
| 12:51AM |
0 |
problem with E100P |
| |
| Monday September 6 2004 |
| Time | Replies | Subject |
|---|
| 11:46PM |
1 |
forwarding calls thru Freshtel |
| 10:38PM |
0 |
carrier connection options |
| 10:35PM |
3 |
iaxy vs sipura |
| 10:01PM |
0 |
Zaptel errors with E100P + TDM40B |
| 9:42PM |
1 |
[patch] allow the transfer keys from app_dial's't' and 'T' and hangup key 'H' to be configured via features.conf |
| 9:19PM |
1 |
[patch] allow the transfer keys from app_dial's 't' and 'T' and hangup key 'H' to be configured via features.conf |
| 6:51PM |
0 |
IAX2/GSM VOIP troubleshooting |
| 6:12PM |
9 |
Zaptel 'Under the Hood' Project |
| 5:18PM |
1 |
Wait for Dialtone syntax in Dial cmd? |
| 4:12PM |
0 |
Horrible noise instead of indications |
| 3:16PM |
1 |
Problem Loading asterisk_oh323-0.6.3b eith last *cvs... |
| 3:09PM |
2 |
Codecs for fax traffic |
| 1:29PM |
6 |
RES: Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual. |
| 12:26PM |
0 |
only hear a few ring tones |
| 10:24AM |
3 |
multiline IP hardphone w/ FDX speakerphone? |
| 10:15AM |
3 |
SIP rtp port forcing |
| 10:04AM |
2 |
DTMF information? |
| 9:48AM |
3 |
VM access |
| 8:38AM |
2 |
Placing Asterisk between existing PBX and PSTN |
| 7:37AM |
2 |
spouse-friendly spa-3000 pstn interface |
| 7:03AM |
0 |
x-lite and pound key |
| 5:05AM |
1 |
UK Callerid bug #1719 & TDM400p |
| 4:02AM |
5 |
Newby question. Basic structure |
| 3:45AM |
0 |
Wildcard TE410P still making trouble |
| 3:22AM |
1 |
cvs server problem |
| 1:44AM |
2 |
Four single-port FXO Cards in one * box |
| 1:08AM |
0 |
SIP-Channels cannot be created after a while of running asterisk ... |
| 1:01AM |
1 |
Voicetronix OpenSwitch12 |
| 12:40AM |
1 |
T.38 "pass-thru" |
| |
| Sunday September 5 2004 |
| Time | Replies | Subject |
|---|
| 9:28PM |
5 |
Asterisk Conferencing using g729 |
| 3:22PM |
0 |
DTMF with HFC-S, not supported yet? |
| 2:52PM |
4 |
Asterisk & sudo from httpd |
| 2:03PM |
1 |
res_perl |
| 1:51PM |
1 |
internal s0 using chancapi |
| 11:19AM |
1 |
Pause or Wait character in Dial command? |
| 10:49AM |
2 |
ZAP channell Dial timeout |
| 10:35AM |
0 |
iconnect and Asterisk |
| 10:26AM |
3 |
ChanSpy by anthm and more... |
| 9:32AM |
1 |
Number of digits |
| 9:07AM |
2 |
GRQ / RRQ |
| 8:14AM |
1 |
Any asterisk echo demo servers ? |
| 7:37AM |
0 |
My Cisco 7940 is not registering with Asterisk |
| 7:36AM |
2 |
FXO/FXS with T.38 over SIP |
| 1:10AM |
2 |
offtopic - channel banks |
| 12:02AM |
1 |
need help configuring dlink dvg-1120M |
| |
| Saturday September 4 2004 |
| Time | Replies | Subject |
|---|
| 10:25PM |
5 |
Wildcards and variable number of digits |
| 3:11PM |
1 |
Oh323, Please Help Newbie ;( |
| 1:57PM |
3 |
Question on echo's for Canadian Asterisk users ... |
| 12:18PM |
5 |
Free WWT (WorldWideTelco): Utopia, or just a matter of organization? |
| 11:45AM |
0 |
Terminating Multiple incoming calls on * |
| 11:18AM |
3 |
Help Running am-main.pl Perl/CGI on Apache Server |
| 10:13AM |
0 |
Wall-mounting UIP 200 and SoundPoint IP600 keepscoming off hook |
| 8:42AM |
0 |
current cvs - zap failure with tdm? |
| 6:53AM |
0 |
PBX --- > Asterisk Connection |
| 5:43AM |
1 |
call back on failed transfer or dial? |
| 4:35AM |
8 |
Linux distribution |
| 2:44AM |
0 |
Which proto. Is more stable/less resource demanding for Asterisk, SIP or H323? |
| 2:12AM |
1 |
How do you avoid or reduce false hangups on X100P? |
| 1:34AM |
1 |
SIP/IAX2 phones with builtin magnetic stripe reader |
| |
| Friday September 3 2004 |
| Time | Replies | Subject |
|---|
| 6:04PM |
0 |
D/600JCT-2E1 |
| 5:14PM |
1 |
Voicemail Size on Disk |
| 4:13PM |
3 |
Putting a call on hold |
| 2:41PM |
2 |
Wall-mounting UIP 200 and SoundPoint IP600 keeps coming off hook |
| 2:21PM |
0 |
Re: Re:New to * |
| 1:55PM |
0 |
Lower cost router suitable for VoIP? |
| 1:37PM |
0 |
Rejecting Calls in Cisco 7960 -- |
| 1:16PM |
2 |
Using AVM Fritz!PCI as zap interface |
| 12:02PM |
3 |
Help setting 2 Offices in US and India |
| 11:19AM |
2 |
X100P blows up after a while (really loud noise) |
| 10:42AM |
0 |
Sending multi-line sms text |
| 10:26AM |
0 |
Slow Robotic or like underwater voice |
| 9:22AM |
7 |
Dropping incompatible voice frame |
| 8:37AM |
0 |
Call Parking with Queues |
| 8:21AM |
0 |
Dlink Video Phone & Asterisk |
| 8:07AM |
1 |
BIG ISSUE with SIP, not sure where to go but it's killing asterisk. |
| 7:04AM |
2 |
mpg123 - multiple instances, taxing CPU |
| 6:38AM |
0 |
I forgot to add my email please contact me offline we have around 300, 000 to 1/2 million minutes per month for India and Pakistan .. can ztdummy help trunk mode? |
| 6:17AM |
1 |
RES: Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual |
| 5:49AM |
1 |
SIP / Keep alive... |
| 3:59AM |
0 |
busy signalling on PRI doesn't work... |
| 3:31AM |
5 |
Digium E100P and PMX in Germany |
| 2:35AM |
1 |
one doubt |
| 2:30AM |
5 |
Lower cost router suitable for VOIP ? |
| 1:53AM |
0 |
RC2 with OH323 or H323 |
| 1:27AM |
2 |
OH323 0.6.3b compilation problem with 1.0 RC2 on RH9 |
| 12:20AM |
1 |
zap barge restrictions |
| 12:13AM |
0 |
Kphone Can't register to ser via Asterisk |
| |
| Thursday September 2 2004 |
| Time | Replies | Subject |
|---|
| 11:37PM |
3 |
digitnetworks card issues? |
| 11:07PM |
1 |
Mediatrix APA III-4FXO (or 1204) help. Anyone with user manual? |
| 10:24PM |
0 |
Message waiting / stutter dialtone ? |
| 8:40PM |
1 |
GSM codec bandwidth |
| 7:42PM |
5 |
Any way to _always_ execute certain commands in a dialplan context? |
| 7:33PM |
2 |
Sorry, Newbie here |
| 6:46PM |
5 |
Polycom SIP INFO & Changing Ringers |
| 6:03PM |
0 |
diff between SNOM & Asterisk |
| 5:53PM |
0 |
Quad E1 card that support R2 signaling |
| 5:27PM |
0 |
Weird CallerID question |
| 2:24PM |
1 |
Analogue call answer detection |
| 2:16PM |
1 |
WG: Digum TE410P |
| 2:05PM |
0 |
MeetMe- on demand recording |
| 1:21PM |
2 |
Phone numbers for testing |
| 1:20PM |
2 |
Polycom Microbrowser |
| 11:27AM |
1 |
Hard Ground (On Ring) |
| 11:20AM |
2 |
uniden Uip300 |
| 11:16AM |
0 |
GUI VoiceMail directory question: |
| 11:06AM |
0 |
Audio dropouts w * and 7960's |
| 10:58AM |
0 |
line feedback, no dial tone |
| 10:43AM |
0 |
UIP300 |
| 9:47AM |
1 |
Incomming ring on POTS line kills ongoing voip call? |
| 9:15AM |
2 |
Please help to config tdm11b |
| 8:52AM |
0 |
ROBO-8712VLA SBC |
| 7:53AM |
0 |
isdn, pbx and * |
| 7:40AM |
1 |
Asterisk + ISDN BRI - gateway or card? |
| 7:06AM |
1 |
call back on failed transfer? |
| 6:27AM |
0 |
Problem with supply of pin number from SJPhone |
| 6:03AM |
1 |
Commercial CID spoofing system |
| 5:51AM |
1 |
BRI&DDI |
| 5:24AM |
2 |
${CALLERID} |
| 5:05AM |
2 |
How let SIP clients connect directly? |
| 5:04AM |
1 |
no dial tone when dialing out on vonage |
| 4:51AM |
1 |
Problem with HasNewVoicemail() |
| 4:01AM |
0 |
Webmin module. |
| 3:30AM |
1 |
asterisk config and root |
| 3:16AM |
1 |
why do i get this message emailed to me everytime i post? |
| 3:10AM |
1 |
Any UK PipeCall/PipeMedia users? |
| 2:08AM |
3 |
BT Easicom - Andy Powell |
| 1:46AM |
0 |
oh323 <> sip |
| 1:43AM |
0 |
X-Lite from Home |
| 12:32AM |
1 |
voicemail email problem |
| 12:21AM |
2 |
Searchable Archives? |
| |
| Wednesday September 1 2004 |
| Time | Replies | Subject |
|---|
| 11:38PM |
2 |
Hung SIP channels |
| 10:57PM |
3 |
Distinctive rings |
| 10:47PM |
0 |
Meetme delay issue |
| 10:35PM |
0 |
Audio Delay in Meetme |
| 9:57PM |
0 |
Whats the '411' on echo cancellation? |
| 9:33PM |
1 |
Odd PRI Behavior |
| 7:43PM |
1 |
HFC cards and Asterisk |
| 6:56PM |
4 |
Why are you guys promoting a Rippoff |
| 6:00PM |
1 |
Really Wierd softphone problem ... must read |
| 5:28PM |
0 |
Newbie: Asterisk Config to Replace Lucent Partner |
| 5:08PM |
1 |
Festival TTS & mbrola ? |
| 3:40PM |
0 |
Analog -> ip sip softphone on Fritz Capi - strong reverb ? |
| 3:36PM |
2 |
zaphfc crashes Linux |
| 3:09PM |
0 |
h323 - forcing user authentication |
| 2:48PM |
1 |
X100P + Call-Waiting - Flash how-to. |
| 2:17PM |
0 |
TDM40B hangup on fax or data modem carrier |
| 12:39PM |
2 |
Migrating Asterisk |
| 12:31PM |
0 |
Asterisk, newbie, fwd and is this jitter? |
| 12:19PM |
0 |
Using an analog modem through asterisk (zap channels) |
| 11:49AM |
1 |
latest CVS build won't load |
| 11:43AM |
1 |
Broken sound in VoiceMail |
| 11:01AM |
1 |
MWI light on Cisco Phones |
| 10:59AM |
2 |
Help Me - SIP Phones ( No Voice) !!!! |
| 10:41AM |
2 |
Rebooting Linux / Asterisk |
| 10:26AM |
1 |
FXO Disconnect supervision |
| 10:24AM |
1 |
Dynamic dialplan |
| 9:13AM |
1 |
NEWBIE: PWLIB Build Failure |
| 8:48AM |
0 |
Newbie - Troubles after installing e100p |
| 8:33AM |
0 |
CLI variable not set on incoming call |
| 7:35AM |
1 |
Agents Log off |
| 7:19AM |
4 |
Group Dial |
| 5:28AM |
5 |
dtmf problem |
| 4:59AM |
2 |
Lucent iMerge |
| 3:07AM |
0 |
Ring tone when busy in trunk scenario |
| 1:07AM |
1 |
international caller id support |
| 1:05AM |
0 |
has anyone the capiCD() funktion in chan_capi running? |
| 12:20AM |
6 |
Mitel 5010 |