| Sunday September 30 2007 |
| Time | Replies | Subject |
|---|
| 2:15PM |
0 |
Asterisk Dropping Calls (Richard Young) |
| 1:19PM |
0 |
Asterisk 1.4, h.323, OpenCom 1010 |
| 9:34AM |
1 |
Selecting a specific line from Zap/g |
| |
| Saturday September 29 2007 |
| Time | Replies | Subject |
|---|
| 9:06PM |
0 |
Big problems with TDM2400 :( |
| 12:56PM |
3 |
FAX detection not working |
| 9:52AM |
1 |
VoIP, Asterisk, What Do you really need ? |
| 1:22AM |
3 |
meetme conference using g729? |
| |
| Friday September 28 2007 |
| Time | Replies | Subject |
|---|
| 6:53PM |
1 |
odd audio problem |
| 6:36PM |
0 |
Nano syntax highlighting. |
| 6:02PM |
0 |
Proximity detection versus GSM receiver |
| 5:57PM |
1 |
Proximity Detection: Motorola Q + Bluetooth + Asterisk |
| 4:28PM |
2 |
Changing contexts "on the fly" |
| 2:41PM |
0 |
VoIP, What Do you really need ? |
| 2:39PM |
1 |
Recommend Digium Hardware? |
| 1:40PM |
1 |
Non-USASCII chars in sip.conf? |
| 1:13PM |
1 |
Multiple Meet me conferences |
| 12:55PM |
4 |
. (period): Wildcard match; matches one or more characters |
| 12:52PM |
1 |
How can I know if I wrote the configuration like correctly |
| 11:57AM |
0 |
Proper trunk to connect two systems. |
| 8:57AM |
1 |
Ringing Groups, SIP Forward and looping problem |
| 7:16AM |
0 |
Conference call today at 12:30 PM EDT |
| 3:39AM |
1 |
call relation in call transfer |
| 12:58AM |
0 |
Asterisk Appliance with VoIPStreet |
| |
| Thursday September 27 2007 |
| Time | Replies | Subject |
|---|
| 10:30PM |
1 |
Polycom 501 won't take new bootrom.ld or sip.ld... |
| 10:27PM |
0 |
Problems Connecting Two Asterisk Installs ViaISDN PRI Cards |
| 7:56PM |
0 |
Timeout issues |
| 7:07PM |
1 |
Zap channel stuck in conference |
| 5:57PM |
0 |
SIP interface status |
| 4:48PM |
0 |
TE120p and music on hold |
| 4:03PM |
15 |
Cisco 7940G licensing with asterisk |
| 3:48PM |
5 |
Which Asterisk version to use? |
| 3:25PM |
3 |
Digium acquires Switchvox |
| 2:23PM |
2 |
IAX configuration |
| 12:45PM |
0 |
astcc sometimes doesnt write on mysql |
| 12:26PM |
0 |
ADIT TDM T1 <> Asterisk MGCP |
| 11:24AM |
3 |
Music On Hold - How to increase volume ? |
| 10:34AM |
4 |
3-way calling |
| 3:56AM |
1 |
help with channelbank audiocodes MP-124 |
| 3:35AM |
0 |
Doesn't seem to want to transcode. |
| 3:05AM |
0 |
voip hacking article |
| 12:35AM |
0 |
h.323 out of media path |
| |
| Wednesday September 26 2007 |
| Time | Replies | Subject |
|---|
| 11:39PM |
1 |
asterisk audits |
| 11:38PM |
4 |
Asterisk realtime error |
| 8:48PM |
2 |
ChanSpy issue |
| 8:05PM |
0 |
Networking Question |
| 7:54PM |
3 |
How to "busy out" zap channels |
| 7:49PM |
2 |
Ast_log |
| 4:10PM |
1 |
IAX gsm bandwith calls |
| 3:58PM |
1 |
Routing issue |
| 3:45PM |
1 |
Asterisk - Spandsp Fax not working? |
| 3:31PM |
1 |
faster timeout in ENUMLOOKUP() function |
| 3:04PM |
1 |
Manager Originate Action and Cancel |
| 1:36PM |
1 |
Supermicro PDSME+ and TE110P [ ref:00D36mPe.50033qy57:ref ] NEW CASE 22828 |
| 1:31PM |
1 |
chan_sip falls over with undefined symbol ast_pickup_ext |
| 1:24PM |
2 |
My G729 problem re-visited |
| 12:48PM |
0 |
Grandstream HT 502 ATA stops receiving calls |
| 12:08PM |
0 |
configuration of wanpipe for asterisk. |
| 10:42AM |
1 |
Busy problem |
| 9:08AM |
1 |
DTMF signalling, SIP, and Background() |
| 7:07AM |
4 |
Music On Hold |
| 4:53AM |
2 |
SIP Panel? |
| 4:17AM |
0 |
Grandstream GXW-4008 |
| 2:25AM |
0 |
asterisk-users Digest, Vol 38, Issue 83 |
| |
| Tuesday September 25 2007 |
| Time | Replies | Subject |
|---|
| 10:37PM |
1 |
Configure one call per line on Cisco 7941/7961 |
| 9:19PM |
1 |
Help with Sip Registration |
| 8:40PM |
1 |
Dutch Number for Inbound |
| 6:34PM |
2 |
Point-to-Point SIP link without registration |
| 2:57PM |
2 |
swift.conf - cepstral voice quality adjustment options |
| 2:41PM |
1 |
Multiple Home system with SIP |
| 2:32PM |
2 |
show queue (queue name) |
| 2:22PM |
3 |
Zaptel-1.4.5.1 Compile Error |
| 12:55PM |
5 |
Do I need to run #modprobe zaptel for Digium |
| 9:27AM |
9 |
Asterisk Redundancy |
| 9:14AM |
4 |
Anyone else having problems with the list |
| 9:09AM |
1 |
Backuping VoIP provider with PRI |
| 8:59AM |
4 |
Grandstream GXP2020 / 2000 |
| 8:35AM |
0 |
[EVENT] Asterisk and VoIP in enterprise |
| 7:54AM |
2 |
HOWTO/FAQ question (Location: Sweden) |
| 6:44AM |
2 |
running twice |
| 5:37AM |
1 |
Completing my Configuration |
| 3:52AM |
0 |
Analog Telephone Adapter |
| 2:36AM |
1 |
ExternNotify Voicemail |
| 12:49AM |
2 |
Yikes! Polycom 501 chokes on BootRom 4.0.0? |
| |
| Monday September 24 2007 |
| Time | Replies | Subject |
|---|
| 11:38PM |
0 |
CID spill after second ring |
| 10:36PM |
0 |
Extensions Configuration |
| 9:16PM |
0 |
Asterisk with multi-line appearence? How? |
| 9:14PM |
0 |
Asterisk as Media Server |
| 8:57PM |
0 |
TDM2400 answer detection |
| 8:56PM |
1 |
DTMF dropping digits |
| 6:36PM |
0 |
Spur error with Siemens Hi Path |
| 6:31PM |
3 |
CallerID problem Asterisk 1.4.2 |
| 3:22PM |
0 |
Adding allowed codecs to Asterisk |
| 2:57PM |
0 |
Anyone use the Linksys phones? (Zeeshan Zakaria) |
| 2:18PM |
2 |
Asterisk 1.4.12 Release? |
| 11:30AM |
0 |
asterisk crash |
| 11:29AM |
0 |
Asterisk Dropping Calls |
| 11:28AM |
3 |
Asterisk and OCS integration |
| 9:38AM |
2 |
Virtual server Solution |
| 9:07AM |
2 |
Sangoma or digium ? |
| 7:21AM |
1 |
# to transfer calls |
| 6:12AM |
0 |
asterisk canreinvite option questions |
| 5:59AM |
0 |
Call hangup after 60seconds |
| 1:04AM |
2 |
asterisk cli - vi keybindings ? |
| 12:25AM |
1 |
Help with log entries. |
| |
| Sunday September 23 2007 |
| Time | Replies | Subject |
|---|
| 5:13PM |
5 |
Anyone use the Linksys phones? |
| |
| Saturday September 22 2007 |
| Time | Replies | Subject |
|---|
| 8:40PM |
0 |
Looking for a reliable source for DIDs from Mexico, Colombia and Venezuela |
| 5:48PM |
1 |
Looking for recommendations on Nufone and Gamachi |
| 5:21PM |
1 |
PCI: UNABLE TO HANDLE 64-bit ADDRESS SPACE FOR |
| 3:51PM |
2 |
error messages related to mysql in asterisk CLI |
| 2:44PM |
2 |
errors messages in asterisk CLI |
| 10:47AM |
0 |
HITBSecConf2007 - Malaysia Materials & Photos are up ! |
| 4:34AM |
1 |
prepaid application recommendation |
| 3:07AM |
2 |
Realtime table columns |
| |
| Friday September 21 2007 |
| Time | Replies | Subject |
|---|
| 9:36PM |
4 |
Polycom 501 Phones Rebooting |
| 8:44PM |
0 |
Confused about Asterisk 1.4 RTPQOS... |
| 7:06PM |
1 |
SIP and Firewall |
| 7:03PM |
1 |
UNICALL MFC/R2 + Asterisk 1.4 |
| 4:07PM |
1 |
Authenticate() application and CDR |
| 3:18PM |
3 |
SIP over TCP |
| 2:18PM |
3 |
Asterisk 1.2.13 and presence |
| 1:21PM |
0 |
"HiarPinning" via TDM400 in the UK ... |
| 1:08PM |
1 |
call limit |
| 12:55PM |
3 |
Asterisk and MS Exchange 2007 |
| 12:49PM |
0 |
Problems bringing up ZAP trunks via PRI |
| 6:24AM |
0 |
Voxalot User and Peer details. |
| 3:22AM |
1 |
Dialing an external number and then passing it to an extension... |
| |
| Thursday September 20 2007 |
| Time | Replies | Subject |
|---|
| 10:31PM |
4 |
Asterisk 1.2.24 simultaneous call limits. |
| 10:18PM |
1 |
GROUP() issues for me |
| 9:37PM |
1 |
Paging MEETME_RECORDINGFILE Variable |
| 7:31PM |
1 |
Polycom 330 + Asterisk, phone locks up. * key will do it |
| 7:23PM |
0 |
Astricon Ride From Airport to Conf Hotel |
| 5:48PM |
10 |
IAX Java Softphone? |
| 5:20PM |
9 |
Problems Connecting Two Asterisk Installs Via ISDN PRI Cards |
| 4:52PM |
1 |
Queue Question |
| 3:14PM |
2 |
The device state is still 'Not in Use' ... check UPGRADE.txt |
| 2:30PM |
2 |
Outgoing SIP packets out of order? |
| 12:33PM |
0 |
Ghost calls from phones |
| 10:49AM |
5 |
Horrible problem - calls losing sound |
| 8:48AM |
0 |
loop detected |
| 8:34AM |
1 |
asterisk crash and core dump: format_mp3.so |
| 6:30AM |
4 |
Newcomer Question |
| 3:32AM |
0 |
Video doesn't work for outgoing call? |
| 1:57AM |
1 |
OT: Samsung Sprint CDMAoIP |
| |
| Wednesday September 19 2007 |
| Time | Replies | Subject |
|---|
| 9:54PM |
1 |
Building an RPM from Asterisk 1.4 |
| 9:43PM |
2 |
AMI extension states |
| 9:33PM |
2 |
Asterisk on Fedora Core 4 |
| 8:48PM |
1 |
crash after callbackagent ackcall |
| 6:11PM |
1 |
Short Audio Drop Out During Calls |
| 5:17PM |
3 |
Dial() Command Parameter L Overflow? |
| 3:56PM |
1 |
Freeswitch Vs Asterisk |
| 3:54PM |
0 |
How many SIP phone connect with asterisk |
| 3:41PM |
1 |
How to cancel the password check in VoicemailMain() |
| 3:20PM |
2 |
Hfcmulti and B410P Digium Card |
| 3:02PM |
2 |
what is softswitch |
| 1:01PM |
2 |
Supermicro PDSME+ and TE110P |
| 12:37PM |
1 |
dtmf issues on PRI and 1.4.11 |
| 12:25PM |
0 |
Configuring Loose routing method |
| 11:24AM |
0 |
Vedio confrancing with asterisk |
| 10:31AM |
2 |
Multi-sip rings |
| 10:20AM |
1 |
off-topic: Avaya 46xx, release 032207 ... help |
| 10:03AM |
1 |
Queue serializes call delivery ? |
| 9:51AM |
0 |
Howto pickup call from queue? |
| 8:50AM |
0 |
RTP Read too short with T.38 |
| 8:43AM |
1 |
AgentCalbackLogin not loging in race condition ? |
| 8:40AM |
0 |
openser/ser/Asterisk user meeting (beer drinking in Vienna) |
| 5:43AM |
0 |
asterisk directory dialing |
| 5:32AM |
3 |
VoIP Provider for business |
| 3:11AM |
18 |
sip.conf best practices? |
| |
| Tuesday September 18 2007 |
| Time | Replies | Subject |
|---|
| 8:26PM |
6 |
Limiting Simultaneous calls |
| 8:19PM |
3 |
Comfort noice sample (gsm/mp3) |
| 8:03PM |
4 |
Linux limits |
| 4:41PM |
2 |
ISDN PRI debug in Asterisk |
| 4:27PM |
1 |
Queue agents w/ DUNDi |
| 4:07PM |
2 |
ISDN data packets |
| 3:55PM |
1 |
Dell Power Edge 1900 |
| 3:53PM |
3 |
Interesting Conference Request - Can this be done ? |
| 1:27PM |
0 |
T1/PRI pricing |
| 1:01PM |
0 |
Bug labs |
| 12:15PM |
2 |
asterisk crash and core dump |
| 10:44AM |
1 |
stanaphone issues. can someone verify my config? |
| 8:20AM |
2 |
Randomly half-voice at sip/zap |
| 6:59AM |
0 |
Issue with Asterisk realtime |
| 12:56AM |
1 |
Asterisk 1.4 and Cepstral |
| 12:27AM |
1 |
Chan_SCCP vs. Chan_Skinny |
| |
| Monday September 17 2007 |
| Time | Replies | Subject |
|---|
| 10:34PM |
0 |
Asterisk and HPC Cluster |
| 10:19PM |
3 |
Enabling MySQL UNIQUE from cdr.conf |
| 7:50PM |
1 |
Softphone RTP Session Start-up Delay |
| 6:58PM |
7 |
Why does everyone seem to dislike *now? |
| 5:06PM |
2 |
Call Center SoftPhone with Auto Answer |
| 4:05PM |
0 |
analog topex and Digium cards (maybe OT??) |
| 2:08PM |
1 |
RTP Call Disconnect |
| 12:09PM |
2 |
Filesharing + video + voice supported Soft phone |
| 10:32AM |
1 |
Problem with asterisk-perl-0.08 and Asterisk >= 1.2.20 |
| 7:50AM |
0 |
[SOLVED] fax machine detection for outgoingcall on DIVAcard |
| 6:29AM |
1 |
extensions for conference call |
| 6:09AM |
0 |
Asterisk 1.4.11 over NAT |
| 5:02AM |
1 |
Wondering why I can't post |
| 3:33AM |
3 |
Voicemail.conf |
| |
| Sunday September 16 2007 |
| Time | Replies | Subject |
|---|
| 2:24PM |
0 |
Problem with asterisk 1.4.11 and playing files to meetme conference |
| 9:08AM |
0 |
Replacing an SPA 3000 |
| |
| Saturday September 15 2007 |
| Time | Replies | Subject |
|---|
| 6:18PM |
2 |
Astribank and caller ID from PSTN |
| 4:21PM |
2 |
AGI/PHP: missing arguments |
| 1:55PM |
1 |
External FXO port. |
| |
| Friday September 14 2007 |
| Time | Replies | Subject |
|---|
| 11:04PM |
0 |
Zaptel ztdummy module causes playback to fail |
| 9:28PM |
0 |
3 way Calling |
| 9:00PM |
2 |
AGI script fails on IAX channels (from call file). |
| 8:38PM |
6 |
Force a new user to configure Comedian mail? |
| 8:13PM |
1 |
g729 on 1.4.10.1 |
| 7:05PM |
3 |
ztdummy kills audio |
| 5:58PM |
1 |
MOH Files Volume |
| 5:23PM |
4 |
how to route outgoing calls on IP-level |
| 4:53PM |
0 |
AstLinux 0.4.8 Released |
| 4:49PM |
2 |
Prompt for extension with standard dial-tone. |
| 4:39PM |
4 |
Can Asterisk match a literal "*" in extensions.conf |
| 3:52PM |
1 |
Help Drop Calls |
| 2:10PM |
2 |
DISA and DTMF detection problem w/ FXO port on a TDM400 |
| 1:55PM |
0 |
AsteriskNOW + legacy PBX integration |
| 1:02PM |
2 |
outgoing call restriction in extention.conf |
| 12:45PM |
1 |
Mutipoint Conferencing? |
| 12:15PM |
1 |
[Serusers] user meeting (beer drinking in Vienna) |
| 5:56AM |
0 |
VOIP Users Conference Friday September 14 @ 12:30PM EDT |
| 5:47AM |
1 |
Asterisk voice quality tuning |
| 1:07AM |
1 |
Include zaptel in kernel... |
| 12:05AM |
6 |
DECT SIP phones |
| |
| Thursday September 13 2007 |
| Time | Replies | Subject |
|---|
| 9:10PM |
3 |
Voicemail in 1.4? |
| 8:56PM |
1 |
FreePBX (2.3) - Good? Bad? Ugly? |
| 8:55PM |
2 |
TDM400P |
| 7:34PM |
1 |
TCP connection to AMI broken after 15 minutes |
| 7:32PM |
5 |
CallWithUs Service? |
| 6:31PM |
0 |
ZAP to invalid SIP device call looping |
| 6:30PM |
0 |
Very fast playback |
| 6:26PM |
0 |
problem |
| 5:32PM |
2 |
Paging to external speaker like in airports etc... |
| 5:18PM |
1 |
Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX |
| 3:32PM |
1 |
how to disable wctdm auto modprobe during boot |
| 2:16PM |
2 |
DTMF error on asterisk |
| 1:51PM |
1 |
SMS in France - allways get "NAK" |
| 1:45PM |
2 |
Fwd: Bad FCS error |
| 1:30PM |
1 |
how to determine if a SIP extension has DND on or off |
| 11:11AM |
0 |
[phpAGI] generate a call from a SIP device to Asterisk |
| 10:19AM |
3 |
Asterisk cli |
| 10:03AM |
2 |
ztdummy problem in fedora7, kernel 2.6.22.5-76.fc7 |
| 9:08AM |
1 |
fax machine detection for outgoing call on DIVA card |
| 7:13AM |
0 |
Support of simple E1 CAS signaling (MFCR-like) |
| 6:37AM |
0 |
asterisk call back dail plan |
| 6:24AM |
0 |
Licensing and provisionning |
| 4:33AM |
1 |
call transfer detection in dial plan |
| 3:38AM |
2 |
FW: Problems with two trunks |
| 3:18AM |
1 |
Trunk & Outbound Route for a Cisco VOIP router? |
| 1:08AM |
1 |
Zap channels: no sound with certain call paths |
| 12:44AM |
1 |
Problems with two trunks |
| |
| Wednesday September 12 2007 |
| Time | Replies | Subject |
|---|
| 10:26PM |
0 |
AsteriskNOW |
| 8:45PM |
3 |
Agent Callback Login in 1.4 |
| 8:44PM |
2 |
Conference bridge. |
| 7:45PM |
1 |
Looking for Asterisk Consultant in San Franicsco |
| 7:43PM |
0 |
(no subject) |
| 5:44PM |
2 |
Callback for unanswered transfers... |
| 4:52PM |
2 |
Digium Appliance |
| 3:57PM |
2 |
Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris? |
| 3:51PM |
1 |
res_snmp |
| 3:49PM |
0 |
Solution: Sysmaster and Asterisk |
| 2:59PM |
1 |
Astribank 32 and Far End Disconnection |
| 2:55PM |
1 |
Direct dialing to correct extension from analog lines |
| 1:57PM |
0 |
Wanted: VoIP Engineer for Warsaw ! |
| 1:23PM |
2 |
Problems with Asterisk behind a firewall |
| 12:21PM |
0 |
fax and answer machine detection for outgoing call on DIVA card |
| 10:46AM |
1 |
TDM2400P: Power alarm error on boot |
| 10:42AM |
1 |
TE405P intermittent yellow alarm |
| 10:01AM |
0 |
AAI2UUI - how? |
| 7:57AM |
2 |
Generating an old-fashioned dialtone |
| |
| Tuesday September 11 2007 |
| Time | Replies | Subject |
|---|
| 10:06PM |
1 |
IAX2 NAT issues |
| 8:50PM |
1 |
Chan_sip Entry |
| 8:26PM |
1 |
TDM400P periodic sound clicks on FXS |
| 7:17PM |
2 |
bug in 1.2.24 |
| 6:31PM |
1 |
TDM400P not answering or making calls |
| 5:02PM |
0 |
SIPAddHeader cmd from Realtime MySQL, not getting all the 'appdata' field |
| 4:50PM |
1 |
Mark Spencer: Digium is Growing Up (VONMAG) |
| 4:18PM |
0 |
dtmfmode rfc2833 and info |
| 3:38PM |
1 |
exit ChanSpy with DTMF |
| 2:32PM |
1 |
Linux-HA and Asterisk |
| 12:56PM |
2 |
Another State Of The Punctuation Mark question - Vonage |
| 10:30AM |
3 |
Prevent multiple sip registrations |
| 9:38AM |
4 |
Installing Asterisk on to CentOS 4 |
| 9:04AM |
5 |
Flash IDE |
| 7:22AM |
2 |
Asterisk 1.4.11, res_features.so, SegFault |
| 6:39AM |
1 |
Asterisk on NGINX Server? |
| |
| Monday September 10 2007 |
| Time | Replies | Subject |
|---|
| 11:26PM |
0 |
rtptimeout on Asterisk 1.4.x |
| 9:33PM |
0 |
DTMF |
| 9:01PM |
5 |
Asterisk Manager API - Originate command |
| 7:25PM |
1 |
Cisco UC 500 |
| 6:27PM |
4 |
Partitioning DSL input |
| 5:28PM |
2 |
Siemans SIP/PSTN phone S450 |
| 3:12PM |
2 |
Failover SIP logic |
| 10:18AM |
1 |
56k modem configuration |
| 9:09AM |
1 |
USA Termination |
| 8:04AM |
1 |
New Project: AskoziaPBX |
| 6:37AM |
5 |
online active call watching |
| 5:07AM |
5 |
Connecting Legacy Pbx With Asterisk With FXS. |
| |
| Sunday September 9 2007 |
| Time | Replies | Subject |
|---|
| 10:35PM |
1 |
SIP endpoint that does not register |
| 10:31PM |
3 |
canreinvite |
| 10:28PM |
3 |
nat=yes |
| 10:26PM |
2 |
What is the difference between increasing the verbose level and the debug level? |
| 10:14PM |
1 |
Maximum retries exceeded on transmission |
| 7:35PM |
1 |
DTMF bug in dsp.c and 1.4.11 |
| 6:16PM |
4 |
Strange Behaviour |
| 5:28PM |
1 |
Which cause less CPU usage: GSM or wav?? |
| 3:45PM |
1 |
Softkeys wrong with chan_skinny |
| 1:55PM |
0 |
what is the usable feature in DUNDi? |
| 1:40PM |
0 |
[mythtv-users] Real Time Clock Alarm Broken with 2.6.22+ kernel |
| 2:50AM |
1 |
Difference in show channels |
| 12:32AM |
2 |
Asterisk on Ubuntu Feisty |
| |
| Saturday September 8 2007 |
| Time | Replies | Subject |
|---|
| 3:22PM |
0 |
Forgotten SIP session |
| 8:52AM |
1 |
redendent asterisk server for backup |
| 7:58AM |
3 |
Configure extension by software |
| 12:02AM |
1 |
Musiconhold instead ringing |
| |
| Friday September 7 2007 |
| Time | Replies | Subject |
|---|
| 11:41PM |
0 |
SIP INFO request in asterisk |
| 9:03PM |
0 |
voice to text |
| 8:52PM |
0 |
chan_misdn |
| 6:31PM |
3 |
T1 to SIP conversion, standalone device |
| 5:44PM |
1 |
Asterisk + Realtime + Manager reload = crash |
| 5:21PM |
2 |
Meridian S1 to Asterisk via T1 |
| 4:03PM |
1 |
Channels in use? |
| 3:52PM |
3 |
Show Callee name on Display |
| 3:45PM |
1 |
New Installed X100p |
| 3:21PM |
1 |
Manager connection timeout |
| 2:29PM |
0 |
Errors: Too many SIP headers and Unknown SDP media type in offer: video 10702 RTP/AVP 34 31 |
| 1:15PM |
0 |
Sysmaster and Asterisk |
| 1:08PM |
1 |
queue static agents |
| 12:56PM |
1 |
Broken UDP streams |
| 3:39AM |
1 |
how to DUNDi branch office with area code? |
| 1:42AM |
0 |
MINNESOTA: TwinCities Asterisk Users Group Meeting - This Saturday Sep 8th, 2007 (Only hours away) |
| 1:08AM |
0 |
Connecting Asterisk to Alcatel OmniPCX |
| |
| Thursday September 6 2007 |
| Time | Replies | Subject |
|---|
| 8:15PM |
1 |
Random Double Digits |
| 7:44PM |
0 |
Inbound SIP issues |
| 7:05PM |
6 |
Build your own "appliance" concept |
| 6:44PM |
1 |
Cascading queues & calls not joining unavailable queues. |
| 5:43PM |
2 |
Register Extension |
| 5:36PM |
1 |
Dead SIP channels |
| 5:16PM |
0 |
DTMF Problem with International Calls |
| 4:57PM |
3 |
Skype + Asterisk |
| 3:49PM |
0 |
Help needed - ISDN is "redialling" |
| 3:38PM |
0 |
Digium Innovation Awards |
| 3:35PM |
1 |
Asterisk Users Conference Friday @ 12:30PM EDT |
| 3:05PM |
2 |
Different Networks |
| 2:46PM |
0 |
(txfax+spandsp) fax is successfully sent, but Asterisk keeps sending DelayedRetries. |
| 1:58PM |
7 |
SIP Debugging to separate log file |
| 1:55PM |
0 |
31 seconds because it is directly bridged to another RTP stream |
| 11:58AM |
3 |
Multitenant or Multiple virtual machines |
| 11:30AM |
0 |
Asterisk on UML (User Mode Linux) |
| 9:00AM |
2 |
bridge on DIVA card and how to see it |
| 8:07AM |
2 |
asterisk voicemail to email and relaying |
| 5:58AM |
2 |
FAX machine connect with audiocode SIP device |
| 5:28AM |
1 |
Choppy sound while converting alaw to ulaw |
| 4:46AM |
2 |
alphabetical extension patterns |
| 1:46AM |
0 |
Asterisk 1.4 Ignoring SIP ACK's on 487 Responses |
| 12:18AM |
1 |
14. Re: ztcfg error : TE110p error with " CAS signalling on span1 conflicts with HDLC with ... (Carlos Chavez) |
| |
| Wednesday September 5 2007 |
| Time | Replies | Subject |
|---|
| 7:04PM |
0 |
Benchmark |
| 5:32PM |
7 |
Can asterisk give half-ring periodically for MWI? |
| 4:44PM |
4 |
special kind of billing |
| 4:34PM |
1 |
Asterisk + LDAP or RADIUS |
| 4:32PM |
2 |
DTMF Relay Problems |
| 4:20PM |
0 |
ANNOUNCEMENT: Asterisk-Java 0.3.1 released |
| 3:09PM |
1 |
Dialplan regexp |
| 3:05PM |
0 |
Presentation and mISDN |
| 2:56PM |
4 |
ztcfg error : TE110p error with " CAS signalling on span 1 conflicts with HDLC with ... |
| 2:09PM |
2 |
No Dial tone came from fxs modules |
| 1:53PM |
1 |
rxfax() problem - fax signal seems to be ignored |
| 1:31PM |
1 |
Overhead paging over IP |
| 1:13PM |
1 |
Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0 |
| 12:23PM |
8 |
Ping |
| 10:23AM |
2 |
TDM400P (TDM22P) and aux power. |
| 6:53AM |
1 |
Issue with calling queues |
| 6:37AM |
0 |
FAX with asterisk |
| 6:35AM |
0 |
outgoing call restriction |
| 4:36AM |
3 |
E1 Line Tapping |
| 3:04AM |
0 |
CID debugging help needed |
| 3:01AM |
0 |
about ChanSpy,thank you!!! |
| 2:58AM |
5 |
How to make call from asterisk? |
| 1:48AM |
0 |
Tone detection while Dialing |
| |
| Tuesday September 4 2007 |
| Time | Replies | Subject |
|---|
| 11:07PM |
6 |
Overhead paging over IP... |
| 8:39PM |
6 |
Udev issue on zaptel install |
| 4:49PM |
2 |
Asterisk w/MS SQL Server 2005 |
| 4:38PM |
1 |
Cisco 79xx XML Apps (was: Re: Cisco Directory Format) |
| 4:23PM |
4 |
Asterisk Died message |
| 1:46PM |
1 |
SIPBroker vs SIPgate |
| 10:30AM |
0 |
NAT-troubles with RTP |
| 6:47AM |
11 |
stop log/debug messages into /var/log/messages |
| 6:41AM |
1 |
Asterisk Manager Interface, reliably monitor NewCall for an extension |
| 4:24AM |
1 |
unsuscribe |
| 3:50AM |
1 |
VSP authentication to incorrect context |
| 2:13AM |
1 |
FW: Account Registration Failed |
| |
| Monday September 3 2007 |
| Time | Replies | Subject |
|---|
| 6:03PM |
3 |
Manager Originate without phone off hook? |
| 3:23PM |
0 |
Append Extension number sounds to Voice Mail Message? |
| 3:21PM |
1 |
ADIT 600 & CMG <=> Asterisk question |
| 2:49PM |
0 |
Digium 4FXO PCI card or Grandstream 4FXO Gateway? |
| 2:08PM |
1 |
Asterisk with app_RPT question |
| 12:52PM |
1 |
Dificult macro, please advise |
| 12:10PM |
1 |
Setting Callerid with chan_misdn |
| 8:18AM |
0 |
Wanted: VoIP Engineer for Warsaw! |
| 8:14AM |
0 |
Account Registration Failed |
| 6:41AM |
0 |
enter menu |
| 6:25AM |
1 |
unnumbered priorities |
| 12:48AM |
0 |
Grandstream GXW-4104 ??? |
| 12:45AM |
1 |
Wireless VOIP Keysets? Recommendations? |
| |
| Sunday September 2 2007 |
| Time | Replies | Subject |
|---|
| 10:15PM |
1 |
off-hook warning tone |
| 2:03PM |
1 |
How can i send my sip channel 3 to mailbox 2? Please Help! |
| 1:32PM |
2 |
Cisco 7960 or 7960G |
| |
| Saturday September 1 2007 |
| Time | Replies | Subject |
|---|
| 1:16PM |
0 |
phone as control interface (was 99 bottles of beer) |
| 9:28AM |
2 |
Escape characer for Digit Timeout |
| 7:55AM |
0 |
Asterisk MESSAGE Method for SMS |
| 7:30AM |
3 |
Zaptel modules are being installed in different directory |
| 6:38AM |
1 |
asterisk 1.2 or 1.4 for conference call service |
| 6:35AM |
1 |
A102d sangoma's card and ztdummy |
| 5:59AM |
0 |
OT - How to script softphone installation |
| 1:15AM |
0 |
Problems with Asterisk 1.2.23 and Polycom 601 |