| Saturday December 31 2005 |
| Time | Replies | Subject |
|---|
| 9:47PM |
0 |
Voicemail through outlook |
| 7:12PM |
0 |
SPA-3K FXO: Incoming and outgoing calls in different contexts? |
| 5:55PM |
0 |
Need HT488 FXO example for both inbound and outbound. |
| 3:11PM |
0 |
New Manager Client Program |
| 1:46PM |
0 |
OT Maybe: Anyone have any knowledge of v5.1/v5.2 in connection with Asterisk? |
| 11:41AM |
4 |
voicemail/privacy system |
| 7:52AM |
1 |
How to set features.conf to change the hangup key. |
| 4:57AM |
0 |
[Announcement] chan_capi-cm 0.6.2 released |
| 3:34AM |
6 |
GXP-2000 fw 1.0.1.13 and NTP |
| 12:33AM |
1 |
Multiple Realm Definitions? |
| 12:10AM |
0 |
AGI Variable |
| |
| Friday December 30 2005 |
| Time | Replies | Subject |
|---|
| 10:57PM |
0 |
answer supervision and POTS |
| 10:06PM |
1 |
Sip man in the middle |
| 10:04PM |
1 |
RE:problem with X100P card |
| 9:13PM |
0 |
Re: Go directly to new messagesfromVoiceMailMain? |
| 6:53PM |
8 |
name that vendor... |
| 6:49PM |
0 |
Dicate not completing the DialPlan? |
| 6:17PM |
0 |
Best way to terminate calls |
| 5:49PM |
0 |
sip through nat problem |
| 4:01PM |
7 |
GXP-2000 any good with * ? |
| 2:50PM |
1 |
Outputting human readable info on a VoIP call's quality? |
| 2:49PM |
1 |
voicemail .wav filename |
| 2:29PM |
2 |
Recording Calls for Specific ACD Agents |
| 2:16PM |
1 |
NOOB: Need Help Learning How to Debug PRI (U.S.) |
| 2:14PM |
1 |
Aterisk 1.2.1 zaptel module not found |
| 2:02PM |
0 |
Motherboard choice for large opteron based asterisk server? |
| 1:58PM |
1 |
Cheap FXS/USB terminal SE-B2K, can it work with asterisk? |
| 1:32PM |
0 |
Passing authentication to an analog adapter |
| 12:38PM |
1 |
Which Asterisk GUI? |
| 12:17PM |
3 |
Fax Support |
| 12:12PM |
0 |
Re: Asterisk-Users Digest, Vol 17, Issue 176 |
| 11:26AM |
1 |
ENUM trees |
| 11:16AM |
0 |
Manually Opening and Closing a Queue |
| 11:09AM |
0 |
Vonage Sip Peering |
| 10:52AM |
0 |
call sip:info@nxs.yi.org |
| 10:41AM |
2 |
Playback after Page() |
| 10:38AM |
1 |
IAX problem - Bug or Compatibility issue? |
| 10:20AM |
2 |
FOP Maximum extensions? |
| 10:11AM |
0 |
TDM400 FXO outbound issue |
| 9:34AM |
1 |
RPID Issue |
| 9:28AM |
3 |
using a Gigaset SX440isdn on a Diva 4BRI ? |
| 9:26AM |
1 |
Notifications when host fails qualify |
| 9:18AM |
0 |
Has anyone used the applicationmap in features.conf? |
| 9:15AM |
0 |
TBCT For PRI support |
| 8:58AM |
0 |
MYSQL Fetch Warning |
| 8:52AM |
0 |
No RTP Warning |
| 8:23AM |
1 |
wctdm module goes missing after a reboot - Gentoo? |
| 7:34AM |
3 |
Problem on ZAP channel |
| 6:38AM |
2 |
Queue features |
| 5:16AM |
4 |
Can we dial agents from extensions.conf |
| 5:06AM |
2 |
Howto config tdm2400 |
| 4:12AM |
3 |
Asterisk connect to voicemaster configuration 1.7 |
| 3:43AM |
0 |
Outbound call using ISDN extension disconnected after *exactly* 30 seconds |
| 2:33AM |
0 |
Re: CALLERIDNUM (Rehan AllahWala) |
| |
| Thursday December 29 2005 |
| Time | Replies | Subject |
|---|
| 11:19PM |
0 |
RE:probelem in working of X100P |
| 8:34PM |
2 |
Re: Go directly to new messagesfromVoiceMailMain? |
| 7:58PM |
0 |
silent dial/ring? |
| 7:00PM |
0 |
Sending Polarity/DTMF Caller ID in chan_zap (Sweden etc...) |
| 6:58PM |
2 |
SetAccount missing? |
| 6:47PM |
4 |
Semi-OT: porting numbers away |
| 3:22PM |
2 |
Allison on Free 411 |
| 2:43PM |
2 |
transfers using # in asterisk |
| 2:09PM |
1 |
Getting Yoda unit to register all four ports |
| 1:41PM |
2 |
voicemail storage over odbc and postgres |
| 1:19PM |
2 |
Regular modems? |
| 12:56PM |
2 |
Linksys SPA-942 |
| 12:41PM |
7 |
Realtime Multiple Asterisk boxes andrtcachefriends MWI |
| 12:15PM |
0 |
fernando_ldb@yahoo.com.ar |
| 12:14PM |
2 |
Realtime Multiple Asterisk boxes and rtcachefriends MWI |
| 11:49AM |
0 |
Gnet VP168S |
| 11:35AM |
3 |
Problem getting D channel up on Sangoma A102 |
| 11:18AM |
0 |
PRI Hangup cause |
| 11:04AM |
1 |
Asterisk SIP PORTS |
| 10:49AM |
1 |
Asterisk 1.2 + DMZ + NAT clients |
| 10:21AM |
0 |
CoreDump |
| 9:24AM |
4 |
What does "Page" application do? |
| 9:16AM |
2 |
zaptel TDM21B 4-5 second pause |
| 9:07AM |
1 |
Asterisk Server Hangs |
| 9:00AM |
1 |
Re: Go directly to new messages fromVoiceMailMain? |
| 7:32AM |
1 |
Easiest way to use HFC-S? |
| 7:31AM |
1 |
SNOM 360 locked up SOLVED |
| 7:28AM |
1 |
smsq |
| 7:23AM |
1 |
SPA-3000 + call waiting |
| 4:57AM |
0 |
Driver not configuring correctly on TE210P forCCS |
| 4:20AM |
1 |
Congestion problem |
| 4:00AM |
1 |
HELP! Asterisk 1.2.1 stops immediately - voicemail problem? |
| 1:50AM |
1 |
I thought they weren't charging - FW: [DIDx.net] Happy holidays wishes from DIDX.net. |
| |
| Wednesday December 28 2005 |
| Time | Replies | Subject |
|---|
| 11:16PM |
1 |
TDM2400 wierdness |
| 9:56PM |
2 |
Grandstream Configuration Utility available |
| 9:45PM |
1 |
Go directly to new messages from VoiceMailMain? |
| 4:06PM |
2 |
Conditional CODEC translation |
| 3:25PM |
5 |
Regular crashes |
| 2:54PM |
3 |
Polycom check-sync |
| 2:29PM |
1 |
Soundstation 4000 |
| 2:22PM |
2 |
sip debug > file.txt |
| 2:16PM |
0 |
Static (Distortion?) and noise on FXO (TDM04b) |
| 2:09PM |
6 |
Asterisk as a Gateway |
| 1:48PM |
3 |
CALLERIDNUM and SRC in CDR |
| 1:43PM |
0 |
spool calls always failing with congestion (AST_CONTROL_CONGESTION) |
| 1:30PM |
1 |
extension not ringing when dialed from DID |
| 12:38PM |
1 |
Problems with multiple outbound calls going to PSTN - Wildcard TE405P |
| 12:10PM |
0 |
how to alter cdr dst info? |
| 12:05PM |
1 |
Iterfacing with a Mitel PBX |
| 12:02PM |
1 |
Polycom Asterisk 1.2.1 and Sipura SPA3000 strange problem |
| 11:06AM |
0 |
GSM-gateway setup |
| 10:51AM |
0 |
Re: 26. RE: Stay away from Grandstream! (Bjorn Asmul) |
| 10:40AM |
1 |
SIP to SIP calls |
| 10:10AM |
2 |
Most Stable Version of Asterisk |
| 10:04AM |
0 |
Tr: Re: call test |
| 9:48AM |
1 |
Driver not configuring correctly on TE210P for CCS |
| 9:46AM |
4 |
CALLERIDNUM |
| 9:40AM |
0 |
BUG? AGI stuck in ast_waitfor_nandfds() |
| 9:39AM |
3 |
What setup |
| 9:18AM |
0 |
MWI problem |
| 9:17AM |
2 |
call test |
| 9:08AM |
1 |
CallerID info needed |
| 9:07AM |
7 |
who is online |
| 7:51AM |
3 |
voip-info: Asterisk record calls |
| 7:48AM |
2 |
oh323 configuration |
| 7:16AM |
1 |
ipVolution |
| 6:44AM |
2 |
Sipura 2002 codec preferences |
| 3:48AM |
1 |
Wrong Password????? |
| 3:36AM |
1 |
CLI execute extensions |
| 2:14AM |
1 |
billing system |
| 2:09AM |
1 |
Bad Checksum answering inbound call |
| 1:10AM |
2 |
PHP Manager |
| |
| Tuesday December 27 2005 |
| Time | Replies | Subject |
|---|
| 11:36PM |
0 |
Trial Edition of Druid Asterisk Web-interface |
| 11:33PM |
1 |
Maximizing audio quality |
| 9:54PM |
4 |
4-port external sip fxo which doesnt suck? |
| 9:36PM |
0 |
Asterisk seg fault (SVN-branch-1.2-r7641) |
| 5:26PM |
0 |
[Announce] Pending Web-MeetMe update |
| 5:07PM |
0 |
Difference between CDR dispositions.. |
| 4:22PM |
3 |
Automatic logoff of all agents at set time |
| 2:33PM |
2 |
Play soundfile before snswer |
| 2:06PM |
0 |
Polycom Soundpoint 501 outbound calls always show NO ANSWER |
| 1:57PM |
0 |
How to register a sip user/peer in real time |
| 1:31PM |
0 |
Asterisk Realtime Database Redundancy |
| 1:25PM |
1 |
agent logs |
| 1:12PM |
0 |
MSN Messenger / Windows messenger Passport service With asterisk any one ? |
| 1:11PM |
1 |
polycom sip slower than grandstream |
| 12:03PM |
4 |
UK, Disconnect supervision |
| 11:28AM |
1 |
Asterisk does not handle call from a Cisco IAD correctly |
| 10:50AM |
4 |
Blackberry SIM card |
| 10:26AM |
6 |
Realtime Static/Dynamic Preference |
| 10:23AM |
1 |
Cisco 7912G through NAT, problems with tones detection. |
| 10:18AM |
1 |
Login incorrect on ZIP2 phones when checking voicemail |
| 8:59AM |
1 |
Strange IAX messages on the console |
| 8:49AM |
0 |
Callerid ID lookup program updated (CID_rewrite v1.2) |
| 8:33AM |
1 |
"one touch record" on asterisk 1.2.1 uses monitor and not mixmonitor |
| 8:27AM |
0 |
TDD/TTY - How does one use this? |
| 8:05AM |
2 |
Cisco dtmf |
| 7:40AM |
1 |
CDR_CSV stops writing, help! |
| 7:30AM |
0 |
RE: [Asterisk- Pls. explain what happens... |
| 7:24AM |
0 |
Asterisk 1.2.1 and X100 clone Zap problem |
| 7:23AM |
4 |
spandsp & fax |
| 7:05AM |
2 |
Asterisk on VPS |
| 7:03AM |
1 |
How to check Digium TE410P firmware version? |
| 6:51AM |
1 |
Polycom IP301 time changing |
| 6:18AM |
2 |
IAX media path, forcing server to stay in the middle |
| 4:38AM |
1 |
Changing Automon filenames? |
| 4:12AM |
1 |
SIP ENUM Daemon |
| 3:32AM |
0 |
Asterisk+mgcp setup+vrg 121 |
| 1:26AM |
2 |
Pls. explain what happens... |
| |
| Monday December 26 2005 |
| Time | Replies | Subject |
|---|
| 11:37PM |
1 |
iptables rules for forwarding SIP/RTP to Asterisk server from behind nat firewall/router |
| 11:20PM |
0 |
SIP "403 Forbidden" Errors... |
| 9:51PM |
1 |
LD_LIBRARY_PATH |
| 7:40PM |
6 |
Stay away from Grandstream! |
| 3:18PM |
1 |
Operator breakout from voicemail |
| 1:12PM |
2 |
64 bit Zaptel? |
| 11:58AM |
2 |
Asterisk lines go into PBX? |
| 11:56AM |
2 |
Delays in IVR |
| 10:24AM |
0 |
NEW Asterisk Management Interface withJavaManager Live Console. |
| 8:31AM |
3 |
channel monitoring whisper mode? |
| 7:52AM |
0 |
NEW Asterisk Management Interface with JavaManager Live Console. |
| 6:04AM |
1 |
RE: how to make contribution in asterisk |
| 5:54AM |
1 |
Eicon DIVA Server V-BRI questions |
| 2:32AM |
0 |
NEW Asterisk Management Interface with Java Manager Live Console. |
| 1:05AM |
5 |
Asterisk Christmas Help request |
| 12:20AM |
0 |
No of records in calls table |
| |
| Sunday December 25 2005 |
| Time | Replies | Subject |
|---|
| 9:58PM |
4 |
Problem with date & time on Aastra 480i since release 1.3 |
| 4:39PM |
3 |
Channel bank timing |
| 4:14PM |
1 |
weird problem with sipura spa2000 and soundcardpa setup |
| 3:11PM |
1 |
weird problem with sipura spa2000 and sound cardpa setup |
| 2:48PM |
0 |
weird problem with sipura spa2000 and sound card pa setup |
| 12:32PM |
2 |
Cisco PGW-2200 OR Asterisk |
| 11:07AM |
1 |
newbie question about making outbound call |
| |
| Saturday December 24 2005 |
| Time | Replies | Subject |
|---|
| 7:03PM |
0 |
Re: Asterisk-Users Digest, Vol 17, Issue 148 |
| 5:08PM |
3 |
System(...) but how to pass parameters? |
| 12:04PM |
1 |
PRI outgoing caller ID stopped working |
| 9:18AM |
1 |
Dialling out with clone X100P board |
| 5:02AM |
1 |
CAPI and * |
| 4:53AM |
0 |
Laptop PCMCIA ISDN card |
| |
| Friday December 23 2005 |
| Time | Replies | Subject |
|---|
| 9:52PM |
0 |
Feature: Attendet transfer with original caller ID |
| 9:02PM |
1 |
Problem with Xlite free phone(Xten) |
| 8:07PM |
0 |
Asterisk With Yahoo messenger |
| 4:19PM |
1 |
Aastra firmware 1.3.x (Far-End sound level issue) |
| 3:22PM |
1 |
problem with tdm400 fxo |
| 3:18PM |
1 |
AMP stuff via CLI? |
| 3:16PM |
5 |
tdm400 fxo problem |
| 3:15PM |
1 |
ASterisk and home lines.. DGM-TDM01B or x100 ? |
| 3:04PM |
0 |
Probs with outbound calls |
| 12:24PM |
1 |
TDM2400P driver change |
| 11:00AM |
6 |
Matching SIP users and peers |
| 9:51AM |
1 |
List Of Defined Variables |
| 9:18AM |
6 |
SIP permit/deny |
| 9:18AM |
0 |
provu 2100 videophone and asterisk |
| 9:03AM |
0 |
FYI on zttool output on SMP system |
| 7:13AM |
0 |
no have dial tone |
| 6:55AM |
0 |
how to get the number of an external phone with Asterisk Manager |
| 6:41AM |
2 |
How to make Asterisk to generate and terminate calls |
| 2:23AM |
1 |
Is there a GUI for asterisk realtime |
| 2:14AM |
2 |
chinese asterisk related web site opened...... |
| 1:54AM |
1 |
Virtual Memory Usage |
| 1:25AM |
0 |
Feature: Attendet transfer with original caller ID |
| 1:22AM |
2 |
Merry Xmas to everybody! |
| 1:03AM |
7 |
Grandstream Budge Tone 102 |
| 12:52AM |
0 |
dialing outbound using te411p |
| 12:43AM |
1 |
No sound problem, chan_sip.c:3451 |
| |
| Thursday December 22 2005 |
| Time | Replies | Subject |
|---|
| 11:44PM |
1 |
Malformed CallerID freaks out SIP channel |
| 9:56PM |
1 |
Emergency Information Needed: sip.conf - bindport allow multiple ports? |
| 9:17PM |
2 |
What hardware fits my needs? |
| 5:23PM |
2 |
automon doesn't work with 1.0.9 |
| 4:27PM |
4 |
Asterisk@Home Fax to Email problems |
| 4:08PM |
0 |
FOLLOWUP: .call files on PRI / Zombie AGI proces ses in FC2 / 1.2 Beta 1 under load |
| 3:07PM |
7 |
Creating conf files from db |
| 2:18PM |
0 |
Re: Fw: Legacy PBX -> * -> Voip Calls problems |
| 1:52PM |
2 |
SNOM 360 locked up |
| 12:53PM |
1 |
"CALLED NUMBER" in IAX2 |
| 12:51PM |
0 |
Fwd: Legacy PBX -> * -> Voip Calls problems |
| 12:45PM |
0 |
forwarding a caller to a conference room |
| 12:23PM |
0 |
chan_sip.c error message |
| 12:01PM |
2 |
ast_sock_cmd: pipe commands to asterisk |
| 11:31AM |
6 |
wav to g729 |
| 11:03AM |
0 |
Asterfax beta4, Asterisk 1.2.0 and issue sending FAX |
| 10:57AM |
0 |
[POSSIBLE SPAM] RE: Identifying Frame Slips from PRI debug |
| 10:30AM |
1 |
how to follow a call in the console |
| 10:21AM |
1 |
Manager API connections - crashes? |
| 10:19AM |
0 |
Make Asterisk explicitly UNREGISTER from a SIP service? |
| 10:09AM |
0 |
Zap Error |
| 9:21AM |
1 |
recording queue calls |
| 9:01AM |
3 |
TDM2400 |
| 8:12AM |
1 |
asterisk AVM C2 again |
| 7:48AM |
2 |
chan_oss.so |
| 7:02AM |
1 |
anybody getting "No authority found" with teliaxnow? |
| 6:58AM |
1 |
anybody getting "No authority found" with teliax now? |
| 6:26AM |
1 |
need help in building dynamic conference |
| 5:55AM |
0 |
DTMF - > FSK CallerID problems |
| 5:48AM |
2 |
Anyone doing NAT through m0n0Wall? |
| 5:47AM |
3 |
snom Firmware 5.0. |
| 5:29AM |
0 |
Codec selection in dialplan |
| 5:23AM |
1 |
PRI problems: B-Channel restart |
| 4:47AM |
0 |
*1.2.1 setcidnum from Zap |
| 4:40AM |
2 |
IAX No Authority found |
| 4:28AM |
0 |
realtime & SIP |
| 3:51AM |
2 |
unplugging E1 cables while asterisk running |
| 1:48AM |
1 |
Problem with octobri and x100p clone |
| |
| Wednesday December 21 2005 |
| Time | Replies | Subject |
|---|
| 8:34PM |
3 |
How to record a call |
| 7:53PM |
1 |
Daily Phreak - Daily Telecom, Asterisk and Phreaking Updates |
| 7:45PM |
2 |
Calls not incoming to any extension from remote proxy server |
| 7:37PM |
1 |
Weird rtpmap issue |
| 6:14PM |
5 |
Semi OT - SuperMicro config question for the Linux/Hardware jedi's - $50 bounty! |
| 4:29PM |
0 |
(no subject) |
| 2:57PM |
0 |
SIP configuration for uip200 |
| 2:39PM |
4 |
caller_id and law |
| 12:56PM |
1 |
DaemonTools Supervise |
| 12:26PM |
1 |
FAX Problems - PRI, Adtran and ZetaFax |
| 12:16PM |
0 |
Asterisk/Zaptel on Kernel 2.6 and ACPI |
| 11:32AM |
2 |
New To Asterisk/POTS - Hardware Setup Questi on |
| 11:28AM |
1 |
Extension cannot match ! receiving call mISDN ... |
| 11:26AM |
1 |
New To Asterisk/POTS - Hardware Setup Question |
| 11:13AM |
1 |
MWI not working - using seperate vm and call routers: |
| 11:11AM |
1 |
chan_capi-cm 0.6.1 won't load |
| 11:02AM |
0 |
enabling while/endwhile |
| 10:55AM |
0 |
Asterisk's VoiceMail server accessed through various DIDs |
| 10:10AM |
0 |
no subject |
| 10:09AM |
0 |
Crash |
| 10:02AM |
0 |
Name file automatic |
| 9:40AM |
1 |
realtime sip firends not being updated |
| 9:31AM |
0 |
Some values ignored when using static realtime |
| 8:44AM |
0 |
Need help with script from http://www.voip-info.org/wiki/view/Polycom+auto-answer+config |
| 8:40AM |
1 |
recieve mutiple inbound calls |
| 8:18AM |
4 |
Identifying Frame Slips from PRI debug |
| 7:46AM |
0 |
Broken MOH |
| 7:31AM |
1 |
Polycom 500 IP and problems with show hints |
| 7:05AM |
0 |
port vs bindport |
| 6:50AM |
0 |
show queue |
| 6:47AM |
0 |
Using mgcp get/generate "message waiting indication" |
| 6:45AM |
1 |
Re: Re: RFC 3262 PRACK (Olle E. Johansson) |
| 6:33AM |
4 |
Asterisk Call Forwarding |
| 6:15AM |
2 |
php agi problem (perhaps problem..) |
| 6:11AM |
8 |
Asterisk server to provide virtuals IPBX |
| 6:04AM |
1 |
Postgres |
| 5:51AM |
0 |
ASTERISK CALL ROUTER |
| 5:47AM |
4 |
[offtopic] Asterisk <-IP-> Siemens HiPath 4000 |
| 5:28AM |
2 |
Instalar Ubuntu |
| 5:09AM |
3 |
Tracing a crash with CAPI calls |
| 4:53AM |
1 |
turn off message "Silence suppression ..." on Asterisk console |
| 4:44AM |
0 |
MP3 problems: MP3Player and Musiconhold |
| 4:36AM |
0 |
Problem with CDR |
| 1:10AM |
0 |
WG: Goto after Dial PRoblem |
| |
| Tuesday December 20 2005 |
| Time | Replies | Subject |
|---|
| 11:05PM |
0 |
Unicall Problem with fax |
| 9:15PM |
0 |
Digium E1 Card Modprobe problems |
| 7:28PM |
2 |
Asterisk based IVR/VoiceMail Server for a Unified Messaging suite |
| 5:22PM |
14 |
Latest Source |
| 5:01PM |
4 |
Unicall E1 Error in Mexico |
| 4:43PM |
4 |
Got SUBSCRIBE for extensions without hint |
| 4:36PM |
2 |
Meetme and ztdummy |
| 3:57PM |
1 |
SPANDSP & TX RX Fax paid support wanted. |
| 3:35PM |
1 |
Help Debugging Dropped Call Audio |
| 3:04PM |
30 |
SIP Subscriptions |
| 2:30PM |
2 |
RFC 3262 PRACK |
| 2:22PM |
3 |
IVR Capacity |
| 2:01PM |
1 |
3 Phone Call Qualtiy Issues |
| 1:17PM |
2 |
Analog terminals and modems? does it work |
| 12:33PM |
1 |
Linking existing channels through Manager interface. Is it possible? |
| 12:17PM |
3 |
Asterisk & FXO & Panasonic PBX |
| 11:28AM |
1 |
IVR and db |
| 11:16AM |
0 |
meet me room status |
| 11:03AM |
0 |
SunFire X4100 |
| 10:58AM |
2 |
G729 and Cisco IOS 12.4 |
| 10:12AM |
5 |
1.2.1 Queues |
| 10:00AM |
0 |
MOH engaged while holding for ANOTHER party (1.2.1) |
| 9:59AM |
0 |
SIP/IAX to PSTN |
| 7:04AM |
0 |
Goto after Dial PRoblem |
| 7:03AM |
5 |
Rolling dialplan... best practice? |
| 6:43AM |
0 |
TE205P E1 PRI card and other problems |
| 6:39AM |
1 |
482 Loop Detected when transferring calls back to Asterisk |
| 6:05AM |
1 |
How to get received digits from console channel |
| 6:05AM |
6 |
Soporte |
| 5:26AM |
2 |
inbound routing with amp and TDM400 |
| 4:38AM |
1 |
messages of Mobile Operator |
| |
| Monday December 19 2005 |
| Time | Replies | Subject |
|---|
| 11:58PM |
1 |
Fast AGi Variables |
| 11:38PM |
1 |
Digium TDM2400 Series Server Compatability |
| 11:29PM |
1 |
UNREACHABLE PEER |
| 11:29PM |
1 |
To write Sphinx Interface in EAGI or app_xxx.c? |
| 9:06PM |
2 |
SIP - SIP bridge dropping calls? |
| 8:34PM |
1 |
Asterisk with Uniden uip200 |
| 5:45PM |
0 |
memory not being released |
| 5:33PM |
0 |
Polycom retry interval and DNS SRV failover |
| 5:10PM |
1 |
Handling SIP clients behind NAT on a semi-dynamic IP |
| 4:34PM |
2 |
Handytone 486 Outbound problem |
| 3:59PM |
1 |
ALERT_INFO Not Working Upon Upgrade to 1.2.1 |
| 3:13PM |
0 |
Asterisk & NAT behaviour |
| 2:33PM |
0 |
queues and redirection. |
| 2:19PM |
3 |
IBM eServers? |
| 2:07PM |
1 |
Mulitple voicemail on mulitple phones |
| 1:31PM |
2 |
Simulate incoming line |
| 1:23PM |
1 |
VoIP/VPN providers in Switzerland |
| 12:57PM |
1 |
Originate a call to a Queue? |
| 11:25AM |
1 |
Variable Help |
| 11:16AM |
0 |
problem with automatic attender calls |
| 10:44AM |
1 |
Re: Asterisk-Users Digest, Vol 17, Issue 111 |
| 9:55AM |
1 |
iax2 on a server behind a linux based stateful firewall |
| 9:15AM |
0 |
MixMonitor error exit |
| 9:08AM |
2 |
Can't pass variables using Originate in PHPAGI 2.14 |
| 7:40AM |
1 |
Problem using Queue and Sip Soft |
| 7:29AM |
3 |
Can't call out on ZAP channel - need help |
| 6:47AM |
0 |
Callware VoiceOne released: a new, easy web GUI |
| 5:34AM |
0 |
ACD with polycom ip phones (resent) |
| 4:45AM |
1 |
DTMFMODE with grandstream |
| 4:20AM |
0 |
Looking for a spare LCD display SNOM 220 |
| 3:33AM |
2 |
Asterisk <-> Skype anywhere/anyhow? |
| |
| Sunday December 18 2005 |
| Time | Replies | Subject |
|---|
| 9:21PM |
1 |
Asterisk Voice mail-reg |
| 7:51PM |
0 |
SIP Remote Call Control |
| 7:48PM |
1 |
[Fwd: Odd problem with Encore 201-SA (r2 converter) with asterisk] |
| 7:47PM |
0 |
Odd problem with Encore 201-SA (r2 converter) with asterisk |
| 6:47PM |
1 |
Anybody having trouble terminating calls at Voxee? <eom> |
| 5:20PM |
0 |
Extension processing misunderstanding |
| 3:28PM |
0 |
iaxmodem through zaphfc |
| 2:15PM |
0 |
FOP led Colors |
| 12:05PM |
3 |
Is it me, or is 1.2.1 slower than 1.0.9? |
| 10:39AM |
12 |
ACD with polycom ip phones |
| 10:24AM |
3 |
New voicemail alert options for Cisco 7960 SIP phones |
| 10:19AM |
1 |
SIP Watchdog |
| 8:27AM |
0 |
Asterisk <-> Avaya system |
| 7:35AM |
4 |
Is this possible in Asterisk? |
| 6:25AM |
1 |
Too high volume on Music on Hold |
| 5:23AM |
1 |
asterisk 1.2.1 and mixmonitor problem |
| 5:22AM |
0 |
Can't pickup call when dialing *8 extension (resent) |
| 3:13AM |
2 |
PERL AGI DIALSTATUS |
| |
| Saturday December 17 2005 |
| Time | Replies | Subject |
|---|
| 10:35PM |
4 |
ztdummy problem !!! |
| 7:59PM |
1 |
aastra.cfg & mac.cfg examples Firmware version 1.3 |
| 7:11PM |
1 |
SIP and echo cancel |
| 6:28PM |
7 |
Toll Free Providers |
| 4:42PM |
3 |
Teliax billing question |
| 3:48PM |
1 |
Can't pickup call when dialing *8 extension |
| 1:43PM |
0 |
placing a call in one or several call groups |
| 1:05PM |
2 |
Grandstream GXP-2000 Auto Answer |
| 11:41AM |
1 |
Linksys PAP2 and Asterisk |
| 11:14AM |
0 |
Terminating calls externally via SER |
| 9:21AM |
0 |
i can't register to my sip service(but x-lite can) |
| 9:20AM |
2 |
Alarm panel through ATA |
| 9:10AM |
1 |
Strange problem with sjphone and 1.2.1 |
| 9:05AM |
0 |
Cisco 79xx display as busy-lamp field |
| 8:44AM |
0 |
Re: [Astguiclient-users] [PATCH][RFC] Quiet debugging messages in Net::MySQL Perl module |
| 8:42AM |
0 |
multiple ALSA devices and Asterisk |
| 7:57AM |
0 |
Cid_rewrite update |
| 7:06AM |
0 |
A2billing Trunk |
| 6:41AM |
0 |
asterisk 1.2.1 realtime mysql.4.1.xx report errors |
| 5:19AM |
0 |
Key R (Flash) and Asterisk |
| 4:33AM |
0 |
I need syntax on applicationmap in features.conf |
| 2:00AM |
1 |
Can Asterisk replace Cisco 5350? |
| 12:24AM |
0 |
asterisk and h323 problems |
| |
| Friday December 16 2005 |
| Time | Replies | Subject |
|---|
| 8:42PM |
5 |
What is the best Dell Machine for Asterisk? |
| 7:13PM |
2 |
Redency of Asterisk |
| 5:08PM |
2 |
Codecs. |
| 5:02PM |
2 |
Asterisk Redundancy |
| 4:58PM |
4 |
TDM01B answering issue |
| 2:52PM |
1 |
DIGIUM |
| 1:24PM |
0 |
Asterisk-1.2.1 incomplete DID number on incoming T1 line |
| 1:06PM |
1 |
.call files on PRI not waiting for answer in de sired context <--ResponseTimeout the best answer? |
| 12:54PM |
0 |
.call files on PRI not waiting for answer in desired context |
| 12:35PM |
0 |
Merlin Legend mode codes |
| 12:35PM |
1 |
.call files on PRI not waiting for answer in de sired context |
| 11:43AM |
0 |
Digium TE205 Card |
| 11:39AM |
0 |
Amtelco Infinity |
| 11:29AM |
4 |
CID lookup from an Exchange Public folder |
| 9:33AM |
0 |
Having trouble calling out from Zap channel |
| 9:19AM |
1 |
1.2.0 queue.conf exit context |
| 8:04AM |
2 |
Mediatrix 1204 help please. |
| 7:20AM |
0 |
Connecting Meridian M8x24-DS to Asterisk - NoDTMFtones |
| 6:41AM |
2 |
Experience sharing on Planet VIP-450 + Asterisk |
| 6:37AM |
1 |
incoming dtmf handling by ATA devices ? |
| 6:04AM |
2 |
Central Registration mechanism |
| 5:29AM |
2 |
ztdummy / timer problem with kernel 2.6.14 |
| 4:28AM |
1 |
Romania/Rumania setup |
| 3:52AM |
8 |
HW Echo Cancellers |
| 3:48AM |
2 |
Does hardware like this exist...? |
| 2:59AM |
1 |
Configuration of two Asterisk server |
| 2:35AM |
1 |
Meetme option Ax |
| 2:19AM |
0 |
asterisk 1.2 mysql cdr garbage |
| |
| Thursday December 15 2005 |
| Time | Replies | Subject |
|---|
| 11:29PM |
1 |
Raltime database schemas |
| 10:57PM |
1 |
CallerID/Extension Matching with RealtimeExtensions |
| 10:50PM |
1 |
CallerID/Extension Matching with Realtime Extensions |
| 8:51PM |
2 |
Alternative source for Asterisk-IM |
| 8:25PM |
0 |
Echo & TDM11B |
| 7:31PM |
1 |
SIP Trunk please help |
| 7:23PM |
1 |
Connecting Meridian M8x24-DS to Asterisk - No DTMFtones |
| 7:06PM |
3 |
Weird IAX trunking/7960/ILBC quality issue |
| 5:04PM |
0 |
Connecting Meridian M8x24-DS to Asterisk - No DTMF tones |
| 3:50PM |
1 |
Will ooh323 ever move from addons? |
| 3:44PM |
2 |
looking for hardphone configuration info |
| 1:57PM |
0 |
Script to detect corrupted faxes from SpanDSP |
| 1:05PM |
3 |
Echo Canceller usage |
| 12:27PM |
1 |
Disposition Failed still happening |
| 11:36AM |
0 |
Asterisk Realtime connection failed |
| 11:28AM |
0 |
Originating calls to a channel groups |
| 11:26AM |
8 |
Google Analytics and voip-info.org |
| 11:02AM |
1 |
E1 Echo (was: Small explanation of txgain rx gain statement please) |
| 10:55AM |
1 |
Voipsupply - my experience |
| 10:44AM |
0 |
Handyton 486 Outbound problem |
| 10:38AM |
4 |
Shutting down Asterisk when not in RTP Stream |
| 10:12AM |
1 |
Can you time limit access to a trunk? |
| 10:10AM |
2 |
Outbound Routing |
| 9:04AM |
4 |
E1 Echo (was: Small explanation of txgain rxgain statement please) |
| 8:19AM |
1 |
astcc issue |
| 7:50AM |
2 |
function cut() |
| 7:14AM |
2 |
ChanIsAvail() |
| 7:13AM |
0 |
Sip configuration for make and receive calls |
| 6:33AM |
1 |
voicemail cutting out |
| 6:26AM |
1 |
hint on Zap channels |
| 6:09AM |
1 |
chan-capi avm b1 and capi.conf problems |
| 5:58AM |
0 |
RE: how to forward call within office |
| 5:45AM |
1 |
screen safe_asterisk does'nt spawn asterisk |
| 5:38AM |
1 |
Firewall Ports forward |
| 5:31AM |
0 |
EXITWITHQUEUE on queue_log |
| 5:22AM |
1 |
Small explanation of txgain rxgain statement please |
| 5:18AM |
0 |
oh323 : which versions recommended for asterisk 1.2? |
| 5:16AM |
1 |
RE: how to forward call within office |
| 4:55AM |
0 |
RE: Asterisk-Users Digest, Vol 17, Issue 89 |
| 4:50AM |
2 |
How to change the Dial command H option to ## ? |
| 4:30AM |
0 |
again - show queue info |
| 4:14AM |
0 |
Help with mgcp |
| 4:02AM |
0 |
How to tell if Authenticate failed without using j in 1.2 |
| 4:00AM |
1 |
background music... |
| 3:50AM |
0 |
QueueMetrics 1.0 rc 1 out today |
| 2:42AM |
3 |
AoC (Advice of Charge) |
| 1:14AM |
0 |
Anyone with VIP-450 |
| |
| Wednesday December 14 2005 |
| Time | Replies | Subject |
|---|
| 10:21PM |
2 |
2 PBX linked via internet |
| 10:12PM |
1 |
I don't want ilbc, i just want G.711 |
| 9:44PM |
1 |
Starting RTP with Dial and MusicOnHold |
| 9:33PM |
3 |
Asterisk & STUN |
| 9:29PM |
0 |
why sql error in asterisk 1.2.1 with realtime with mysql 4.1.x |
| 9:22PM |
0 |
WishList - Devices that are (probably) not available yet [OFF-TOPIC] |
| 5:24PM |
0 |
Problem with bridging SIP to OH323 and SIP to SIP: Bridge stops bridging |
| 3:57PM |
2 |
How to disable sip Native bridge |
| 3:56PM |
1 |
"Context Picker" for interception and redirection |
| 1:55PM |
0 |
Prevent Logging "reload" verbose |
| 12:43PM |
1 |
1.2.1 Compile Error |
| 12:36PM |
1 |
ChanIsAvail() and SIP |
| 11:33AM |
3 |
HOWOT transfer call from mobile back to extension? |
| 10:56AM |
2 |
asterisk + H323 + 723 |
| 10:20AM |
5 |
traffic shaping |
| 10:20AM |
1 |
Background() followed by Read - something wrong? |
| 10:11AM |
1 |
Headset Phones? |
| 10:08AM |
1 |
Cisco 7940 Time Source |
| 9:44AM |
5 |
OT: Linux on treo 650 |
| 9:11AM |
1 |
appradius |
| 9:02AM |
2 |
hardware echo cancellation for TDM card |
| 8:51AM |
0 |
Gateway crashes when transferring to external lines |
| 8:49AM |
3 |
Best way to automatically stop and start Asterisk nightly |
| 8:44AM |
1 |
Blind transferred user does not hear phone ring while waiting for phone to be picked up. |
| 7:50AM |
0 |
Need help with Sipura 3000 |
| 7:50AM |
0 |
Video calls (MS Messenger, Tandberg) |
| 7:42AM |
2 |
Need help with sipura |
| 7:37AM |
0 |
MGCP Unable to find key |
| 7:19AM |
0 |
'#' (fast foward) and '*' (Rewind) not working in VoicemailMain |
| 5:58AM |
0 |
Exceptionally long queue in SIP Channel |
| 5:42AM |
1 |
subscription |
| 5:30AM |
2 |
Unable to find key |
| 4:58AM |
0 |
quadbri, isnd, netherlands: callerid not working |
| 4:48AM |
0 |
Help:asterisk 1.2.1 release compile |
| 4:35AM |
0 |
[help] problem in astersik |
| 3:57AM |
1 |
[Fwd: Re: Re: [helpp] Problem in astersik] |
| 3:31AM |
1 |
Dial multiple destinations |
| 3:26AM |
2 |
voicemail boxes |
| 3:09AM |
3 |
Wildcard TDM2400P: comments |
| 2:19AM |
2 |
Join when empty problem, in queue |
| 1:46AM |
2 |
capi.conf - AVM C4 P2P or P2MP |
| 1:45AM |
0 |
PSTN gateway & Asterisk -> Virtual Switchboard??? |
| 12:59AM |
0 |
SIP peer vs. user-- how is the USER ever selected? |
| 12:41AM |
1 |
send SMS via own SMS Service |
| 12:22AM |
0 |
RealTime and automatic extension registration. |
| |
| Tuesday December 13 2005 |
| Time | Replies | Subject |
|---|
| 11:10PM |
1 |
fxs woes... |
| 10:59PM |
2 |
SIP Subscription Storage Location |
| 10:28PM |
0 |
Cisco 7960/ATA/MultiTech MVP200 FXS/FXO to H323 gateways on ebay |
| 8:07PM |
0 |
Pattern Matching, speed and memory.... |
| 6:24PM |
2 |
VizuFon CIP-4500 with Asterisk through SIP |
| 5:53PM |
0 |
Asterisk 1.2 SIP register problems |
| 4:59PM |
0 |
nice -n 19 called from shell script through Syst em() gives "Permission Denied" |
| 3:01PM |
0 |
Entering Digits |
| 2:57PM |
2 |
Asterisk 1.2.1 |
| 2:51PM |
0 |
empty line sip_notify.conf |
| 2:33PM |
2 |
December VON Magazine |
| 1:39PM |
1 |
pb !Astrisk 1.2 Card TE411P |
| 1:18PM |
1 |
IAX error message |
| 12:56PM |
0 |
Meetme Conference sound problems |
| 12:41PM |
0 |
Question on having asterisk put calls into a meetme. |
| 12:12PM |
1 |
format_mp3 & uninstalling mpg123 |
| 11:43AM |
0 |
queues & music on hold |
| 11:21AM |
1 |
cdr_addon_mysql can't find libmysqlclient.so |
| 11:12AM |
0 |
CID name & number contain unwanted quotes in CDR |
| 10:59AM |
1 |
Very high memory consumption when high number of calls are processed |
| 10:47AM |
1 |
mISDN & chan_misdn on Fedora Core 4 - problems |
| 10:33AM |
1 |
Tellabs manuals |
| 10:28AM |
0 |
Asterisk Feature Request: app_bridgeme |
| 9:55AM |
2 |
Bonded ethernet ports and * |
| 9:44AM |
1 |
mISDN Caller ID problem |
| 8:49AM |
2 |
extension seen as busy when it is not |
| 8:43AM |
0 |
FXOTUNE Error on channel 2 |
| 8:39AM |
0 |
408 Request Timeout vs. 403 Forbidden |
| 8:20AM |
5 |
Partial PRI pass thru? |
| 8:05AM |
1 |
RE: 1.2.1 has broken voicemail realtime |
| 7:39AM |
1 |
Testing 10.0.0.203 with 10.0.0.0 |
| 7:20AM |
2 |
calls forwarded to busy agent |
| 6:34AM |
1 |
1.2.1 has broken voicemail realtime switching |
| 5:57AM |
0 |
NAT/Qualify/RTP bug |
| 5:47AM |
3 |
IAX2 show channels show Channel (NONE) |
| 5:34AM |
2 |
g729 translation to zap (ISDN) doesn´t work |
| 4:18AM |
0 |
OOH323 -> IAX2 : no sound |
| 4:08AM |
0 |
queue_log Vs show queue abandon calls discrepancy |
| 3:54AM |
0 |
Call Disconnecting |
| 3:05AM |
2 |
Info request from Sangoma users |
| 2:10AM |
5 |
chan_capi AVM C2 |
| 1:35AM |
2 |
SPA-3000: Dual Registrations? |
| 12:49AM |
1 |
AGI GET Variable Problem |
| 12:33AM |
1 |
Setting Language |
| |
| Monday December 12 2005 |
| Time | Replies | Subject |
|---|
| 11:47PM |
1 |
ENUM For Presence |
| 11:26PM |
2 |
Patch zaptel.init to support debian |
| 10:47PM |
0 |
NAT Issues? |
| 10:31PM |
0 |
subscribing |
| 9:57PM |
0 |
Patch to zaptel Makefile |
| 8:44PM |
2 |
X100p echo guide |
| 7:13PM |
2 |
No outgoing sound...sometimes |
| 6:54PM |
0 |
Invoking blind SIP REFER transfer asterisk |
| 6:35PM |
2 |
Skips and Pops in Call Recordings |
| 6:16PM |
0 |
(Montreal Users) Call for technical presentations |
| 5:02PM |
1 |
new asterisk 1.2 setup doesn't react when I press any numbers |
| 4:42PM |
0 |
Interesting Article on Echo Cancellation |
| 4:39PM |
0 |
analog FXS card for dialup modem and fax |
| 4:30PM |
0 |
E1 PRI cause codes |
| 4:29PM |
0 |
NOTIFY Messages |
| 4:24PM |
2 |
Softphone with Hint support? |
| 3:45PM |
0 |
Is clearglobalvars=no really working in asterisk 1.2.1? |
| 2:31PM |
1 |
busypattern tones? |
| 2:17PM |
6 |
asterisk in real estate developments |
| 2:02PM |
1 |
Turning off hardware echo can on TE411P |
| 12:41PM |
0 |
Call Monitoring / Ext to Ext with Sipura-841 |
| 12:22PM |
3 |
trying to get SIP to work remotly. |
| 11:55AM |
1 |
Dlink DI-102 QOS Thingy? |
| 11:41AM |
3 |
Make list of incoming and outgoing calls |
| 11:40AM |
0 |
capi incoming call timeout |
| 11:33AM |
1 |
Need advice on BRI |
| 11:30AM |
0 |
Zultys ZIP2 + asterisk + DTMF on other end? (i.e. ivrs, autoattendants, etc) |
| 10:41AM |
1 |
executing a reload under stress in Asterisk |
| 10:36AM |
1 |
Cisco 7940 Reboot |
| 10:33AM |
1 |
uniqueid with multiple asterisk hosts |
| 10:04AM |
0 |
Outgoing data call |
| 8:52AM |
0 |
Unable to prevent SIP to SIP calls from removing Asterisk from Media path |
| 8:42AM |
0 |
Dial Cmd Outbound CLID Failure (* 1.2.1) |
| 7:47AM |
3 |
How do I remove the temp greeting?!?! |
| 7:16AM |
1 |
ASTCC/ASTCC anything wrong with that? |
| 7:07AM |
1 |
Zap Transfer |
| 7:01AM |
0 |
ChefSec function |
| 6:36AM |
0 |
persistentagents, persistentmembers |
| 5:00AM |
1 |
Digium PCI-X timeline |
| 4:57AM |
1 |
Production Upgrades |
| 3:19AM |
5 |
PRI E1 - HDLC Bad FCS / HDLC Abort errors |
| 2:45AM |
2 |
Problem with Speex |
| 2:40AM |
1 |
click to dial applications |
| 2:35AM |
0 |
Variables naming, may be a BUG?? |
| 2:27AM |
1 |
Cisco 7941 difference |
| 2:14AM |
1 |
"Got clone lock for masquerade" crash |
| 2:11AM |
1 |
Problems with current chan-capi-cm |
| 1:56AM |
5 |
[helpp] Problem in astersik |
| 1:17AM |
0 |
asterisk1.2.1+realtimedb+voicemail+contexts |
| 12:59AM |
2 |
CallerID Transfer |
| 12:15AM |
2 |
Long and variable echo |
| |
| Sunday December 11 2005 |
| Time | Replies | Subject |
|---|
| 11:32PM |
2 |
Mechanisms for Implementing a Common ContactDatabase |
| 11:03PM |
2 |
New Product ID. |
| 10:53PM |
1 |
Mechanisms for Implementing a Common Contact Database |
| 7:56PM |
0 |
Originate action script causes Asterisk to hang |
| 7:20PM |
1 |
Cleaning up CLID on incoming PRI lines |
| 3:23PM |
2 |
Small / embedded system recommendations |
| 3:22PM |
4 |
Realtime Subscribecontext |
| 1:37PM |
14 |
Regexten |
| 12:39PM |
3 |
Attack dialing |
| 11:52AM |
0 |
[Asterisk-Dev] C++ AGI debuggin |
| 11:50AM |
8 |
Asterisk Dynamic DNS |
| 11:47AM |
5 |
SIP Qualify |
| 8:59AM |
1 |
Dialing analog extensions from SIP? |
| 8:34AM |
3 |
SRV Lookups *ARRGH!* |
| 7:18AM |
2 |
outgoing calls that last an unreasonably long time |
| |
| Saturday December 10 2005 |
| Time | Replies | Subject |
|---|
| 9:42PM |
2 |
bristuff use without BRI/PRI |
| 6:58PM |
2 |
Setting Request URI |
| 4:31PM |
1 |
extensions and regular expressions ( probablyan easy question ) |
| 3:06PM |
1 |
Using SIP_HEADER() Function correctly |
| 2:24PM |
0 |
RE: chan_grab.c for version 1.0X |
| 1:42PM |
0 |
Echo on incoming sip provider- asterisk - sip phone / ata |
| 1:40PM |
1 |
extensions and regular expressions ( probably an easy question ) |
| 11:09AM |
0 |
Good Dialing Macros |
| 10:56AM |
0 |
Problems with zaptel channels not properly beinganswered... |
| 10:52AM |
1 |
agi variables list |
| 10:46AM |
1 |
Problems with zaptel channels not properly being answered... |
| 10:22AM |
3 |
Re: Asterisk-Users Digest, Vol 17, Issue 61 |
| 9:08AM |
1 |
Re: Asterisk-Users Digest, Vol 17, Issue 61 |
| 6:10AM |
1 |
Asterisk not Replying on Port Specified in the VIA header |
| 5:48AM |
1 |
Possible bug in record? |
| 5:44AM |
6 |
Email to voice? |
| 5:34AM |
2 |
Channel 0/1, span 1 got hangup request |
| 4:56AM |
0 |
Sending a recorded message to voicemail |
| 4:13AM |
0 |
Q regarding dialling multiple phones (fwd) |
| 3:49AM |
0 |
Predefined Channel Variables |
| 1:19AM |
4 |
Via Epia |
| |
| Friday December 9 2005 |
| Time | Replies | Subject |
|---|
| 10:23PM |
1 |
Dial Command Doesn't return Correctly! (Bug?) |
| 9:23PM |
1 |
What's the best opensource web interface for customer portal |
| 8:55PM |
0 |
Asterisk without RTP streams vs. SER in statefull mode |
| 8:17PM |
3 |
Synthesized Voice for Asterisk |
| 6:39PM |
0 |
Major Advantages of Asterisk vs Nortel PBX Systems |
| 6:36PM |
0 |
sellvoip.net service |
| 5:34PM |
3 |
Asterisk, Small Business, and Teliax |
| 4:31PM |
4 |
Daily Reboot Script for Asterisk Question |
| 4:13PM |
0 |
Anyone else experiencing Voicepulse outage? |
| 4:04PM |
2 |
IAX Jitterbuffer and trunking |
| 3:45PM |
0 |
Unable to receive DTMF inbound |
| 3:41PM |
0 |
Stop Dial but no Hangup |
| 3:30PM |
0 |
NY Clec Listing |
| 2:55PM |
1 |
Wait for X rings before answering? |
| 2:52PM |
0 |
Asterisk Consulant - Call Centre Proposal |
| 2:46PM |
1 |
callerid international-format |
| 2:03PM |
2 |
Call Routing from GnuGK to Asterisk |
| 1:31PM |
1 |
Adit 600 and Centrex |
| 1:05PM |
0 |
number of users in a meetme conference |
| 1:04PM |
2 |
Asterisk 1.2 chan_misdn modules.conf |
| 12:40PM |
1 |
IAX2 Status monitoring |
| 12:34PM |
1 |
T.38/PRI Gateway Recommendations |
| 12:24PM |
0 |
Unknown RTP codec 96 received |
| 11:19AM |
0 |
Hardware based echo can from NMS Communications |
| 10:23AM |
1 |
Re: Asterisk-Users Digest, Vol 17, Issue 56 |
| 10:13AM |
0 |
testers needed for channel.c jitter buffer (better known as SIP jitter buffer) |
| 9:53AM |
0 |
t38 support in latest asterisk release |
| 9:36AM |
2 |
Asterisk vs Nortel, Northstar and Mitel |
| 9:10AM |
0 |
Asteriskguru Queue Statistics version 0.7 released |
| 8:48AM |
1 |
Echo PSTN Asterisk@Home 2.0 Digium TDM11B & DSL |
| 8:16AM |
4 |
Teliax experiences |
| 7:59AM |
2 |
Phone Information |
| 7:07AM |
1 |
Low Layer Compatibility (LLC) not forwarded? |
| 7:02AM |
4 |
Change time when * is running |
| 5:22AM |
0 |
connection between asterisk and cisco |
| 3:51AM |
1 |
CIDNUM CIDNAME |
| 3:40AM |
1 |
Queue routing - calls return to agent which previously handled call |
| 3:26AM |
0 |
Hangup after dialing |
| 2:59AM |
2 |
/dev/zap/ctl or /dev/zapctl cause ztdummy in init.d failed |
| 2:33AM |
0 |
Aastra firmware 1.3.x. Solution to Far-End sound level issue |
| 2:05AM |
0 |
Re: Is Polycom 500CS with P# 2201.11500.001 SIP capable? |
| 2:04AM |
0 |
PRI billing signalization |
| 1:09AM |
1 |
DID Providers |
| |
| Thursday December 8 2005 |
| Time | Replies | Subject |
|---|
| 10:36PM |
3 |
Porting a phone number to a voip provider |
| 9:51PM |
1 |
Can Asterisk accept and relay calls |
| 9:03PM |
11 |
Asterisk Dial Failover |
| 8:01PM |
1 |
Core dumps since 1.2.0 |
| 7:55PM |
2 |
does asterisk-oh323-0.6.7support asterisk1.2 |
| 7:14PM |
1 |
cant start the conference bride |
| 7:11PM |
1 |
ztdummy on FC4 |
| 6:27PM |
2 |
Why Won't Asterisk REINVITE? |
| 5:40PM |
1 |
Asterisk and Adtran TA 750 Channel Bank -- odd behavior (help!) |
| 4:52PM |
1 |
Re: Meetme and Sipura SPA-941 -badjitter/distortion |
| 4:45PM |
2 |
OOH323 towards cisco gateway(2691)callsetupfailsat q931: Mandatory information element ismissing (96) |
| 4:04PM |
1 |
Where to find Digium products in Mexico? |
| 2:54PM |
1 |
OOH323 towards cisco gateway (2691)callsetupfails at q931: Mandatory information element is missing (96) |
| 1:20PM |
2 |
Call screening script |
| 12:39PM |
0 |
question about priorities? |
| 12:18PM |
0 |
app_md5.so compile problem |
| 12:11PM |
0 |
Compile modules app_rxfax.c app_txfax.c for asterisk 1.2.1 |
| 11:53AM |
0 |
AstLinux 0.3.0 Released |
| 11:16AM |
0 |
Re: [Asterisk-biz] Help with learning Asterisk for the real world.. |
| 11:15AM |
1 |
How do I set up extensions.conf to dial out on analog telephone line? |
| 11:08AM |
2 |
Leave one voicemail for multiple recipients |
| 10:51AM |
0 |
Lucent MAX TNT - how do I route a DID to my sip trunk |
| 10:43AM |
4 |
Asterisk Bounty Pool |
| 10:13AM |
1 |
OOH323 towards cisco gateway (2691) call setupfails at q931: Mandatory information element is missing (96) |
| 9:55AM |
2 |
Realtime Replication of a Single File |
| 9:40AM |
1 |
SIP.conf Technical Documentation - Help |
| 9:27AM |
3 |
Meetme and Sipura SPA-941 - bad jitter/distortion |
| 9:23AM |
0 |
Voicemail context |
| 9:23AM |
1 |
New GSM 1-8 ports Gateway / Terminal for sale(with SMS Feature and Many more) |
| 9:21AM |
2 |
Maximum Calls handled |
| 9:20AM |
0 |
OOH323 towards cisco gateway (2691) call setup fails at q931: Mandatory information element is missing (96) |
| 9:17AM |
0 |
Zombie AGI processes in FC2 / 1.2 Beta 1 under l oad |
| 8:56AM |
2 |
Exit Voicemail |
| 8:41AM |
1 |
New GSM 1-8 ports Gateway / Terminal for sale (with SMS Feature and Many more) |
| 8:24AM |
0 |
Polycom SIP part numbers |
| 8:21AM |
2 |
Octo Bri card together te405p and bristuff |
| 7:54AM |
0 |
Hardware combination and type of asterisk configuration |
| 7:26AM |
1 |
Dynamic IAX2 hosting in the UK |
| 6:43AM |
5 |
Asterisk Call Recording and SIP canreinvite |
| 6:15AM |
3 |
Call simulators |
| 5:21AM |
1 |
Forwarding only at certain times |
| 4:27AM |
1 |
about g729 |
| 4:15AM |
0 |
about * and CM/CME |
| 4:14AM |
1 |
multiple line registrations on attendant console |
| 4:09AM |
1 |
No application 'MeetMe' for extension |
| 3:48AM |
0 |
IAX stress test |
| 3:35AM |
1 |
Change Inbound CALL ID "Asterisk" |
| 3:22AM |
0 |
GROUP_COUNT and AGI |
| 3:11AM |
0 |
CDR manipulation in macros |
| 3:01AM |
0 |
Wrong caller id num on Swissvoice IP10S |
| 2:34AM |
0 |
SVN Revision 7230 |
| 2:30AM |
0 |
MYSQL cmd with Asterisk Realtime |
| 2:00AM |
1 |
Asterisk as sipclient |
| 1:55AM |
0 |
Integration of external (ZAP) agents into queue |
| 12:38AM |
1 |
Asterisk as a gatekeeper |
| 12:19AM |
1 |
Recording Volume on Zap Channel |
| |
| Wednesday December 7 2005 |
| Time | Replies | Subject |
|---|
| 11:53PM |
2 |
RE:how to listen voicemail messages |
| 9:31PM |
0 |
ZT_CHANCONFIG failed on channel 1: No such device or address |
| 8:46PM |
0 |
local not ring,,,, |
| 7:56PM |
0 |
London DID 30 Cents a Number available - 60 Channels, ULAW |
| 7:45PM |
3 |
Aastra 9133i Configurations - are the file names to be lower case or upper case or does it matter? |
| 7:32PM |
1 |
HOW TO: CDR Customer IP address where call came in from |
| 5:47PM |
5 |
Asterisk Hardware recomendation |
| 4:33PM |
4 |
A company that sells Toll Free Number in USA |
| 3:37PM |
0 |
Asterisk 1.2.1 and queue_log |
| 2:55PM |
0 |
AstManProxy Segmentation Faults |
| 2:49PM |
2 |
Announcement only Devices that work with Asterisk for Dedicated 24/7 Conferencing |
| 2:18PM |
1 |
IAX2: Don't know any of 0xf800 formats |
| 1:43PM |
8 |
Sip behind the NAT |
| 1:33PM |
2 |
app_queue on 1.2 ? |
| 1:20PM |
4 |
Unable to compile zaptel / ztdummy |
| 12:38PM |
2 |
Door Phones |
| 12:23PM |
3 |
Can Asterisk act as a media gateway? |
| 12:02PM |
5 |
Recording a call |
| 11:42AM |
0 |
chan_bluetooth Audio Sensitivity |
| 11:21AM |
0 |
Zaphfc as a timing source? |
| 10:48AM |
0 |
VoIP US - Toll Free Origination / Termination Providers |
| 10:38AM |
3 |
Asterisk 1.2.1 released |
| 10:10AM |
3 |
IConnecthere dial out problems |
| 10:07AM |
2 |
Lucent TNT / Asterisk Help |
| 9:34AM |
2 |
Polycom 501 remapping keys |
| 9:04AM |
0 |
Wanted Japan DID |
| 8:41AM |
1 |
E1/T1 configurations |
| 8:28AM |
0 |
how to detect hangup |
| 8:22AM |
0 |
Config Attended Transfer |
| 6:58AM |
1 |
asterisk with EWSD v16 |
| 6:45AM |
0 |
Sending credit card thorugh network/sipura 3000 |
| 5:57AM |
0 |
VoIP to GSM? |
| 5:41AM |
3 |
Busy recognition |
| 5:32AM |
2 |
UK ISDN2e with DDI? |
| 5:10AM |
0 |
Goldstar GDK 186 voicemail |
| 4:00AM |
2 |
HDLC link unstable, yellow alarm on |
| 3:58AM |
2 |
Asterisk modules description |
| 3:18AM |
1 |
Feature implemention |
| 2:31AM |
0 |
res_perl error when loading asterisk |
| 2:11AM |
2 |
Ringtone when dialing |
| 12:36AM |
0 |
audio lost on incoming calls |
| |
| Tuesday December 6 2005 |
| Time | Replies | Subject |
|---|
| 11:42PM |
1 |
Connecting asterisk over consumer wifi network |
| 10:51PM |
0 |
dialplan activated Toll restriction |
| 10:31PM |
2 |
Win up to $2000 for AsteriskEnterpriseReferences! |
| 10:01PM |
1 |
Win up to $2000 for Asterisk EnterpriseReferences! |
| 8:41PM |
0 |
Snom 360 and 320 AutoAnswer |
| 8:17PM |
4 |
Hint Priority for Polycom Phones |
| 8:09PM |
2 |
Asterisk as a Softswitch |
| 7:23PM |
1 |
Win up to $2000 for Asterisk Enterprise References! |
| 6:24PM |
1 |
Flash operation on a call on a ZAP interface... |
| 5:35PM |
1 |
DSP-based echo cancellation (T1). |
| 5:14PM |
0 |
Re: VoipBuster / Finarea |
| 3:59PM |
8 |
Nortel Meridian Option81C to TE405P |
| 3:33PM |
2 |
Snom monitoring of extensions not working |
| 3:24PM |
0 |
ATA Registration problems |
| 2:05PM |
0 |
Outgoing fax detection |
| 1:11PM |
5 |
Packeteer ? Edgemark ? How to not re-cable ? |
| 12:43PM |
0 |
How to setup Connected number |
| 12:35PM |
1 |
Re: Asterisk-Users Digest, Vol 17, Issue 7 |
| 12:19PM |
1 |
Toll-free number on a PRI |
| 11:46AM |
2 |
Re: VoipBuster / Finarea |
| 11:10AM |
1 |
VoIPJet issue == No one is available to answer at this time |
| 11:03AM |
2 |
snom 320 'retrieve' button |
| 10:35AM |
2 |
can * translate DTMF from rfc2833 to inband? |
| 10:17AM |
3 |
Per Extension Password for Outgoing Routing |
| 10:08AM |
0 |
E1 and hardware Test. |
| 9:40AM |
11 |
Complicated Dialing plan routing |
| 8:54AM |
1 |
Credit Card Terminals |
| 8:47AM |
3 |
Odd DTMF issue over PRI |
| 8:12AM |
0 |
Dial application "g" option |
| 8:00AM |
0 |
What would prevent logs from being recreated if they are deleted? |
| 7:33AM |
2 |
Call Forwarding with Account Code.. can it be done? |
| 6:27AM |
0 |
PRI and Dialed Number |
| 5:33AM |
0 |
Primary D-Channel on span 1 up |
| 3:38AM |
0 |
Problem with a second incoming call on a BRIZapChannel |
| 2:59AM |
0 |
CallParking and chan_capi-cm-0.6 |
| 2:48AM |
1 |
RE:Is it possible to install ZAPTEL after installation of Asterisk |
| 2:42AM |
1 |
Problem with a second incoming call on a BRI ZapChannel |
| 2:05AM |
0 |
Asterisk and Video |
| 1:34AM |
1 |
zaptel : unresolved symbol zt_unregister ... |
| 1:30AM |
4 |
SNOM Shared line DevState |
| 12:37AM |
3 |
OH323 user configuration |
| |
| Monday December 5 2005 |
| Time | Replies | Subject |
|---|
| 11:31PM |
1 |
How to restric user to call only specified country |
| 10:58PM |
5 |
Echo cancellation over satellite link |
| 10:48PM |
2 |
EAGI Audio Capture |
| 10:03PM |
3 |
Realtime SIP Lookups |
| 9:50PM |
0 |
2 leg bridged call not hanging up until both legs hangup |
| 9:26PM |
0 |
Please help in writing AGI script |
| 6:40PM |
2 |
Asterisk on PPC & chan_capi issue |
| 6:37PM |
0 |
sipura "Vertical Service Activation Codes" |
| 6:04PM |
2 |
uip200 phone not work with 1.2 |
| 5:44PM |
1 |
logging performance, important impact? |
| 5:01PM |
1 |
ADIT 600 T1 with DNIS digits problem |
| 4:38PM |
1 |
Messages button on a Polycom 501 |
| 3:17PM |
5 |
Best Switch for VOIP Applications |
| 3:00PM |
3 |
PRI indications. |
| 2:08PM |
1 |
Preventing incoming calls from ringing SIP lines |
| 2:07PM |
1 |
Linksys SPA-941 DTMF failure with asterisk v.1.2 |
| 1:21PM |
0 |
transfers from Polycom 501 involving Sipura 300 and asterisk 1.2 |
| 12:28PM |
2 |
Anyone know anything about the new Linksys One product - does it use Asterisk? |
| 12:26PM |
0 |
video phones |
| 11:30AM |
0 |
Panasonic DBS DISA |
| 11:21AM |
0 |
Asterisk Queues Tutorial updated... |
| 10:54AM |
2 |
Include a variable from another file in configfiles |
| 10:25AM |
0 |
asterisk won't answer malformed caller id |
| 10:09AM |
5 |
Looking for advice on cell carrier's default "Un avaliable" message |
| 8:45AM |
0 |
kernel lockup with Fedora Core 4.0 2.6.14-1.1637 |
| 8:36AM |
2 |
Restore logging functionality... |
| 7:36AM |
4 |
h323 vs oh323 |
| 7:35AM |
1 |
Transfer/take call to/from other phone |
| 6:50AM |
0 |
Problem with a second incoming call on a BRI Zap Channel |
| 5:10AM |
1 |
A@H with a2Billing |
| 3:56AM |
2 |
Include a variable from another file in config files |
| 3:51AM |
0 |
oh323 installation |
| 3:35AM |
1 |
SIP INVITE with no 'Contact' field and RealTime support. |
| 3:31AM |
0 |
Calls to DISA over ISDN PRI don't get CONNECT ACKNOLEDGE |
| 3:19AM |
0 |
(warning) iaxy.bin fails checksum |
| 2:28AM |
1 |
Chat Lines/ Party Line Solutions for Asterisk |
| 2:28AM |
0 |
asterisk.h |
| 1:04AM |
0 |
VegaStream 400 |
| 1:02AM |
0 |
Asterisk 1.2.0 - TE210P - "...Control Frame 15..." |
| 12:42AM |
4 |
Error when compiling asterisk |
| 12:37AM |
0 |
voicexml vendors |
| 12:21AM |
1 |
[Amportal-users] AMP queues, AddQueueMember and 'Wrapup Time' |
| 12:18AM |
1 |
Re: sound problem in X-Lite phone with asterisk server |
| |
| Sunday December 4 2005 |
| Time | Replies | Subject |
|---|
| 11:26PM |
2 |
Sipura 3000 Call waiting on the PSTN line |
| 10:11PM |
2 |
Connecting 2 Asterisk using SIP |
| 3:51PM |
1 |
UK DID 0208 £1 per month |
| 3:44PM |
2 |
DISA function |
| 2:23PM |
0 |
Sending data over ZAPHFC D-channel? |
| 2:19PM |
2 |
Getting started with Asterisk and Aastra 9133i |
| 1:16PM |
0 |
RE: how to remove asterisk 1.2 from Red Hat 9 |
| 1:00PM |
0 |
Why does musiconhold.conf changes require a reboot? |
| 11:39AM |
1 |
ISDN 2e Cards |
| 8:48AM |
1 |
re: Help required on asterisk |
| 7:01AM |
1 |
replace hard-phone if soft-phone is online |
| 2:42AM |
1 |
UK Patches for Asterisk 1.2 |
| 2:25AM |
1 |
Sipura 3000 Disconnect Singnel |
| |
| Saturday December 3 2005 |
| Time | Replies | Subject |
|---|
| 11:41PM |
2 |
Re: Asterisk-Users Digest, Vol 17, Issue 18 |
| 8:52PM |
1 |
Converted mp3 files to raw for musiconhold and still does not work... |
| 3:48PM |
1 |
Call queues, agents with DND status set. |
| 1:23PM |
0 |
Order of ports on rear of Sangoma card and pictures in a mini-itx chassis. |
| 12:06PM |
2 |
Can I escape queue with a '*'? |
| 8:42AM |
2 |
Asterisk 1.2 and weird ZAP interface behaviour |
| 8:23AM |
0 |
Fwd: Queue Statistics |
| 7:13AM |
0 |
No CID Info an TE405P with zaptel 1.2.0 |
| 6:50AM |
0 |
Rates for Asian countries |
| 2:21AM |
1 |
IAX Conf Realtime? |
| 2:18AM |
0 |
Voip providers with trunked iax2 |
| 2:14AM |
1 |
Iax2 connection failed |
| 2:09AM |
1 |
Can't compile chan_zap.c on fresh cvs 1.20 checkout |
| |
| Friday December 2 2005 |
| Time | Replies | Subject |
|---|
| 11:15PM |
0 |
Softphone setup |
| 11:06PM |
1 |
Fw: Re: Re: Zaptel errors on Debian |
| 10:12PM |
0 |
Need Good UAE calling rates |
| 8:54PM |
3 |
Broadband VoIP Startup with Asterisk |
| 8:47PM |
1 |
Anyone have experience with SellVoip.net |
| 8:23PM |
0 |
Suggestions on Echo Problems |
| 6:24PM |
1 |
chan_blutooth |
| 5:56PM |
4 |
Linksys SPA-841 Missing Calls |
| 5:51PM |
0 |
Problem, to register an ata on an asterisk |
| 4:48PM |
0 |
Testing Sangoma A2022-SO card with Asterisk 1.2 |
| 4:30PM |
0 |
Prodding channel failed |
| 4:28PM |
1 |
echo canceling algorithm |
| 3:06PM |
0 |
USB/Bluetooth Speakerphones |
| 2:43PM |
1 |
Re: Asterisk 1.2 problems (tneuwert@formos.com) |
| 2:20PM |
0 |
What kind of extension numbers can be used in the exit context of a queue? |
| 2:07PM |
3 |
Context confict question?? |
| 1:54PM |
3 |
Fax Service |
| 1:47PM |
1 |
callback script |
| 1:43PM |
0 |
Subject: Re: Linksys SPA-941 Admin Guide |
| 1:17PM |
1 |
ZapHFC cards not maintaining sync?! |
| 12:53PM |
1 |
Can I get to a menu system while in a queue?? |
| 12:22PM |
1 |
Asterisk 1.2 problems |
| 12:00PM |
4 |
Failover Registration |
| 11:22AM |
1 |
Kernel upgrade causes ztdummy to refuse to run |
| 11:18AM |
1 |
DIAXY to DIAXY problems |
| 10:50AM |
1 |
"hint" priority in AEL? |
| 10:44AM |
0 |
Help with a Company or Site for a DEMO. AYUDA con una empresa para una DEMO |
| 10:40AM |
0 |
v1.2 and cdr badly written |
| 10:03AM |
3 |
Sangoma & Asterisk at home |
| 9:46AM |
0 |
Re: Asterisk-Users Digest, Vol 17, Issue 11 |
| 9:38AM |
1 |
Meetme option 'b' |
| 9:30AM |
2 |
dial-out and variable inheritance problems |
| 9:30AM |
0 |
Originate calls but can't receive them on a SIP trunk |
| 8:46AM |
1 |
sip invite timeouts |
| 7:53AM |
1 |
Queue and agent transfer |
| 7:17AM |
3 |
what is your echo solution |
| 7:12AM |
0 |
ISDN card Sirrix.PCI4S0 |
| 7:12AM |
0 |
DTMF is choppy on the receive |
| 7:04AM |
0 |
Polycom DTMF after connection not working |
| 5:03AM |
1 |
change priority by time |
| 4:59AM |
1 |
equal priority trunks for balancing |
| 4:54AM |
1 |
RE:how to solve error : cannot find extension context 'from-sip' |
| 3:55AM |
0 |
Asterisk-users |
| 3:33AM |
0 |
BT - DSS |
| 3:00AM |
1 |
Limiting DID calls |
| 2:26AM |
1 |
voice problems under 8 concurent calles |
| 2:22AM |
3 |
Asterisk Users Newsgroup |
| 1:10AM |
0 |
DTMF on Planet VIP153 |
| 12:47AM |
1 |
(no subject) |
| |
| Thursday December 1 2005 |
| Time | Replies | Subject |
|---|
| 9:43PM |
6 |
IAX trunking frequency parameter works only on initiator side |
| 8:27PM |
1 |
Hint: how to include dialplan files from remotesystems |
| 7:51PM |
1 |
Hint: how to include dialplan files from remote systems |
| 6:32PM |
2 |
Running asterisk within screen |
| 6:04PM |
1 |
call center dial plan |
| 4:05PM |
0 |
config Polycom with both SIP provider and Asterisk |
| 3:58PM |
0 |
Transfer problem... |
| 3:51PM |
0 |
Asterisk with ooh323 registers with a cisco gatekeeper but disconnects after 300 |
| 3:51PM |
1 |
Very Weird problem with MeetMe, SIP, Zap and the combo of the three |
| 3:36PM |
2 |
Can Asterisk do This? |
| 3:32PM |
1 |
Asterisk as a gateway to Index PBX |
| 3:28PM |
0 |
ITSP in a box demo updated |
| 3:14PM |
5 |
Two Phones - Same extension? |
| 3:09PM |
1 |
Linksys SPA-941 Admin Guide |
| 3:02PM |
0 |
Shoutcast For MOH with Asterisk 1.2 |
| 2:13PM |
0 |
A worrying article |
| 2:01PM |
0 |
showing the hardware status of an * system |
| 1:56PM |
2 |
version 1.2 with chan_bluetooth |
| 1:52PM |
2 |
Write to text file in dialplan |
| 1:10PM |
1 |
default user name and password for a2billing |
| 1:06PM |
0 |
CID text stripped over IAX |
| 12:25PM |
7 |
sixtel |
| 12:24PM |
1 |
Sip trunk between Avaya S8700 and Asterisk |
| 12:21PM |
1 |
chan_bluetooth and Ericsson/SonyEricsson models |
| 10:48AM |
2 |
Altering Incoming CallerID |
| 10:31AM |
2 |
Problem compiling libmfcr2 on FC4 |
| 9:36AM |
1 |
Asterisk Perl AGI, bug with stream_file() ? |
| 8:31AM |
0 |
Asterisk Realtime 2 Servers calling each other |
| 8:31AM |
0 |
meet me message |
| 8:20AM |
5 |
voipbuster |
| 7:50AM |
1 |
Better transfer |
| 7:23AM |
3 |
Complete Removal of Asterisk |
| 7:09AM |
0 |
Call transfer error |
| 6:46AM |
3 |
WG: App_rxfax problem |
| 6:31AM |
0 |
show queue in BE |
| 6:27AM |
1 |
Error on using queue. |
| 6:17AM |
0 |
mail2fax and fax2mail |
| 4:55AM |
1 |
MGCP problem when through internet |
| 4:24AM |
0 |
Compiling Asterisk 1.2 from Source on |
| 4:04AM |
6 |
Codec Problem |
| 3:16AM |
1 |
cannot dial on console on asterisk 1.2 |
| 2:53AM |
8 |
unable to make calls out using ast@home |
| 2:24AM |
1 |
Compiling Asterisk 1.2 from Source on Debian Sarge- Problems |
| 2:22AM |
0 |
optipoint 410 and MWI |
| 12:51AM |
4 |
prepaid application |
| 12:34AM |
0 |
App_rxfax problem |