| Tuesday September 30 2008 |
| Time | Replies | Subject |
|---|
| 11:05PM |
4 |
Asterisk in VM. |
| 8:24PM |
0 |
Transfer a call without announce : no sound |
| 7:32PM |
1 |
OT- NIU Framing |
| 7:28PM |
0 |
How to tell the underlying carrier for your ITSP. |
| 6:08PM |
0 |
Using AMI to View ZAP Channels |
| 3:49PM |
0 |
asterisk-users Digest, Vol 50, Issue 89 |
| 3:32PM |
4 |
Asterisk Documentation now on voip-info.org Wiki |
| 2:22PM |
1 |
asterisk app store |
| 2:14PM |
1 |
Question about Asterisk and Java |
| 10:54AM |
0 |
Cisco 7911g |
| 4:42AM |
1 |
OT: real 2 line phone vs. 1 line and call waiting |
| 2:50AM |
1 |
problem with my softphone |
| 12:47AM |
3 |
Maybe OT - routing calls in PSTN |
| |
| Monday September 29 2008 |
| Time | Replies | Subject |
|---|
| 7:59PM |
1 |
Channel variables "materializing" ... |
| 5:46PM |
0 |
Cheap FXO Card? |
| 5:03PM |
4 |
How can Block a pri channel |
| 3:33PM |
1 |
Source of SIP "Remote host can't match request NOTIFY" |
| 3:08PM |
2 |
Zaptel Lines - How many are in use.. |
| 2:16PM |
0 |
CLI and verbosity level [SOLVED] |
| 1:34PM |
1 |
CLI and verbosity level |
| 12:03PM |
0 |
SIP/IAX Interworking ip CANCEL behavior |
| 11:26AM |
3 |
OT - Avantages of ISDN PtP and PtmP |
| 10:25AM |
1 |
Disable CDR? |
| 10:11AM |
3 |
Knowing incoming call technology and channel [SOLVED] |
| 9:17AM |
2 |
Knowing incoming call technology and channel |
| 8:24AM |
0 |
identify/find a channel to pick it up |
| 8:06AM |
10 |
ATA for large networks |
| 7:41AM |
4 |
Creating Asterisk Binary Package |
| 7:31AM |
0 |
AGI defunct processes + GSM Playback - HELP! |
| 6:00AM |
3 |
uk tole-free dids? |
| |
| Sunday September 28 2008 |
| Time | Replies | Subject |
|---|
| 6:11PM |
0 |
Need help with Cisco 7960 |
| 8:21AM |
1 |
Conferencing Hardware |
| 2:46AM |
1 |
Vividial issue |
| 1:10AM |
0 |
Eye P Media Soft Phone? |
| 12:54AM |
1 |
G.722 between Eyebeam and a Polycom IP650 |
| |
| Saturday September 27 2008 |
| Time | Replies | Subject |
|---|
| 10:52PM |
4 |
credit card processing |
| 10:16PM |
0 |
Keeps Ringing After Answer |
| 9:54PM |
3 |
Troubleshooting one-way voice... how to peek into SIP RTP? |
| 8:03PM |
0 |
rtpkeepalive problem ? |
| 7:11PM |
4 |
FW: Google Alert - "dean collins" |
| 4:11PM |
0 |
Asterisk and VoIP educational resources |
| 4:02PM |
0 |
Philippines |
| 11:15AM |
1 |
Set A-Number in Sip Header |
| 10:58AM |
2 |
running out of disk space |
| 9:08AM |
4 |
Problem with pickup extension *8 from features.conf using IAX |
| 7:41AM |
3 |
test call generator |
| 2:20AM |
1 |
Split incoming call volume across queues on several asterisk servers |
| 2:11AM |
3 |
iPhone Sip App |
| |
| Friday September 26 2008 |
| Time | Replies | Subject |
|---|
| 11:55PM |
1 |
Audio Files |
| 8:23PM |
2 |
Extremely OT: I need someone who can parse a MS Word or PDF or RTF document |
| 7:04PM |
2 |
Bizarre international call problem. |
| 7:01PM |
2 |
Voicemail retention |
| 6:54PM |
1 |
Dial issue |
| 5:25PM |
2 |
server and 2 uniden phones no ringing |
| 2:28PM |
0 |
PRI TE110P Configuration (Solved) |
| 1:38PM |
0 |
Friday 2008-09-26 12:00:00 Asterisk + Skype on your box |
| 1:28PM |
0 |
Incoming URL handling Problem (Asterisk problem ?) |
| 12:53PM |
1 |
Get Call Length of Calls |
| 12:27PM |
1 |
setting DNID |
| 11:27AM |
1 |
Push presence from one asterisk to another |
| 9:15AM |
0 |
T38 fax gateway announcement |
| 6:05AM |
0 |
Skype channel beta |
| 3:06AM |
1 |
ZAP not answering call |
| 3:04AM |
1 |
Monitoring simul calls |
| 2:59AM |
1 |
Sip reload casuing issues |
| 2:58AM |
5 |
Music on hold for sub tenants |
| |
| Thursday September 25 2008 |
| Time | Replies | Subject |
|---|
| 10:18PM |
1 |
users.conf behavior |
| 10:17PM |
0 |
Mysql Command and number rows returned |
| 10:08PM |
0 |
Skype + Asterisk Interview at Astricon |
| 8:29PM |
1 |
Create virtual extension |
| 6:29PM |
2 |
sip forking needed for ekiga 3.0 |
| 5:38PM |
1 |
PRI TE110P Configuration |
| 4:58PM |
0 |
Skype-asterisk connection announced (was Astricon people please post the announcement) |
| 4:17PM |
7 |
Astricon people please post the announcement |
| 2:21PM |
2 |
Dial Plan Issues |
| 1:17PM |
0 |
Monitoring trunk |
| 1:01PM |
1 |
OT: Do You Know What the Problem With CDMA is? |
| 12:52PM |
1 |
Ringing after console dsp hangup |
| 12:05PM |
0 |
SIP TLS |
| 10:45AM |
0 |
IMAP voicemail import |
| 10:27AM |
0 |
appconference low quality g729 |
| 9:10AM |
0 |
Current available allarms in the Asterisk |
| 7:31AM |
2 |
Terrible Experience Net2phone A-Z termination |
| 6:50AM |
0 |
Problem making international calls |
| 2:35AM |
0 |
What happened to the register= setting in sip.conf? |
| 1:21AM |
4 |
g729 capacity |
| 1:12AM |
1 |
Asterisk 1.4 is asking me for Mailbox # |
| 12:28AM |
4 |
Asterisk on VMware Workstation 6 |
| |
| Wednesday September 24 2008 |
| Time | Replies | Subject |
|---|
| 10:23PM |
1 |
Zaptel/DAHDI ztdummy only |
| 8:18PM |
0 |
Astricon 08 Videos & interviews & Voiceroute twitters on astricon |
| 6:14PM |
0 |
Timeout question |
| 4:00PM |
1 |
DID mode |
| 2:20PM |
1 |
Asterisk is covering the peers IP address in SIP and SDP messages |
| 11:30AM |
2 |
Voicemail cutting out after about 30 seconds |
| 8:50AM |
1 |
IAX Hangup floods link with repeated VNAK and HANGUP |
| 6:47AM |
1 |
asterisk console: "quit" is twice in history |
| 3:26AM |
1 |
Asterisk mysql CDR |
| |
| Tuesday September 23 2008 |
| Time | Replies | Subject |
|---|
| 11:00PM |
1 |
How to send indicating call privacy using P-Asserted-Identity? |
| 9:25PM |
0 |
A2Billing Callback Hangup after/about 20 sec! |
| 9:02PM |
5 |
"No route to destination" error |
| 7:05PM |
6 |
Asterisk 1.4 or 1.6 |
| 6:11PM |
2 |
Short question: CPU hardware requirements for Asterisk |
| 6:11PM |
2 |
extension definition |
| 4:36PM |
0 |
Connecting TE212p to NEC XenMaster |
| 3:37PM |
0 |
Linksys 3102 with rfc2833 - NOT WORKING |
| 3:16PM |
3 |
Fwd: more on Free World Dialup groups and FWDLive |
| 2:34PM |
0 |
PRI incoming call forward / call redirect |
| 12:28PM |
5 |
Extension registration |
| 9:57AM |
2 |
chan_misdn troubles |
| 9:52AM |
1 |
AGI and prepaid billing |
| 9:51AM |
0 |
Registration by IP address |
| 8:52AM |
0 |
t38modem on OpenSuse |
| 8:51AM |
6 |
Fax with asterisk |
| 8:36AM |
1 |
Transcoding G.729 files |
| 8:22AM |
1 |
[1.4.21.2] Checking that already off-hook? |
| 3:13AM |
2 |
PSTN Simulator |
| 1:52AM |
0 |
Send us your suggestions on exhibits & tutorials to cover (video) at Voiceroute |
| 12:58AM |
1 |
How to hangup a channel immediately so that it doesn't get charged on cell phone |
| 12:30AM |
0 |
ast_func_write: Function not registered |
| |
| Monday September 22 2008 |
| Time | Replies | Subject |
|---|
| 8:46PM |
0 |
E&M wink/no audio |
| 6:02PM |
1 |
I can't call my remote users? |
| 4:58PM |
0 |
GotoIfTime and timezone specification |
| 1:46PM |
3 |
Problem using AJAM on asterisk 1.4.17 |
| 12:48PM |
1 |
setvar for outgoing SIP channels? |
| 10:30AM |
2 |
Astricon news online? |
| 4:56AM |
8 |
Seemingly easy question: NPA/NXX |
| |
| Sunday September 21 2008 |
| Time | Replies | Subject |
|---|
| 9:00PM |
6 |
How to notify an event to every user |
| 12:28PM |
1 |
Asterisk weird behavior after upgrading |
| 4:49AM |
0 |
Astricon: Throw the dice, give a talk |
| |
| Saturday September 20 2008 |
| Time | Replies | Subject |
|---|
| 10:21PM |
1 |
how to add extensions and sip registrations dynamically |
| 5:48PM |
2 |
broadcast ability |
| 3:17PM |
0 |
callwaiting callerid |
| 1:25PM |
1 |
Cisco acquires Jabber |
| 9:28AM |
1 |
1.6.0-rc6 - SIP hold logic broken? |
| 6:24AM |
0 |
[CID] Unknown IE 18/21? |
| |
| Friday September 19 2008 |
| Time | Replies | Subject |
|---|
| 11:27PM |
2 |
Specific SIP answers on incoming calls? |
| 9:52PM |
1 |
SVN 1.6.0 / current does not compile |
| 9:05PM |
1 |
Loud noise on Zap port... |
| 8:03PM |
0 |
Last 2 days for early bird tickets to DruidCON 2008, 1-2 Oct in Atlanta GA |
| 7:54PM |
2 |
getting results messages from CLI commands via -rx |
| 7:54PM |
2 |
Dropping Phone Calls |
| 4:40PM |
0 |
Weird "permissions" issue when permissions check out... |
| 2:47PM |
1 |
TE110P or TE120P |
| 2:47PM |
0 |
VoIP Users Friday Conference @12 Noon EDT: Astricon run up meeting and more |
| 2:05PM |
0 |
dundi and zap devices |
| 12:31PM |
1 |
Preventing a call forward |
| 9:29AM |
3 |
SIP request send me 482 error |
| 7:41AM |
1 |
Dialing a 60anything number issue! |
| 7:22AM |
0 |
T38 FAX over a Broadsoft |
| 6:26AM |
3 |
PRI E1 Inbound calls hangup with busy after a few seconds |
| 4:57AM |
0 |
T100P detection. |
| 2:49AM |
1 |
what codec is sip using? |
| 2:36AM |
1 |
Follow Me app question |
| |
| Thursday September 18 2008 |
| Time | Replies | Subject |
|---|
| 8:51PM |
0 |
Polycom phones and DNS SRV |
| 7:19PM |
1 |
Old voicemail bounces users |
| 5:09PM |
1 |
device probe order question |
| 4:20PM |
4 |
OT: Cisco 1841 - Can it be made SIP aware? |
| 3:01PM |
4 |
OT - How to stream a A-Law/wav file to a browser ? |
| 2:54PM |
2 |
Custom Voicemail emails |
| 11:42AM |
2 |
Pre-paid Billing |
| 11:41AM |
1 |
How to make a Outgoing Call from Asterisk ? |
| 11:34AM |
1 |
rxfax and txfax |
| 10:25AM |
1 |
Verbosity best practice |
| 10:16AM |
4 |
BRI or PRI callerid |
| 10:16AM |
1 |
Get rid of "Really destroying SIP dialog" |
| 9:46AM |
0 |
482 Loop Detected |
| 7:05AM |
1 |
how to detect pickup... |
| 1:03AM |
0 |
Speech recognition on simultaneous SIP / PSTN calls |
| |
| Wednesday September 17 2008 |
| Time | Replies | Subject |
|---|
| 11:49PM |
1 |
strategy for measuring conference audio delay |
| 10:14PM |
0 |
Understanding of SIP Info Messages |
| 8:02PM |
3 |
app_confrence with loud voices |
| 6:37PM |
1 |
Digium training course |
| 5:58PM |
2 |
Restrict SIP registration to one ip address only? |
| 5:25PM |
0 |
Format ulaw|h ? |
| 5:18PM |
0 |
Part of some calls does not get recorded |
| 4:25PM |
1 |
DTMF detection problem on DISA |
| 4:23PM |
0 |
How to remove dialtone from DISA? |
| 1:57PM |
1 |
chan_iax2.c: No more space |
| 1:28PM |
1 |
pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use |
| 1:12PM |
2 |
codec of channels |
| 9:50AM |
3 |
dtmf passthru |
| 9:45AM |
1 |
SIP URI Forwarding |
| 8:58AM |
0 |
Asterisk 1.6.0-beta5 voicemail problem |
| 8:57AM |
1 |
realtime queue asterisk 1.6.0-beta5 |
| 8:04AM |
1 |
Cellroute setup with asterisk |
| 6:27AM |
2 |
Help with MFC/R2 |
| |
| Tuesday September 16 2008 |
| Time | Replies | Subject |
|---|
| 6:39PM |
0 |
iax reject using domain name |
| 6:27PM |
1 |
Parked Calls |
| 6:04PM |
3 |
Cisco + Asterisk |
| 5:10PM |
1 |
addons will not load when compiling Asterisk with DEBUG_THREADS |
| 3:44PM |
2 |
What is worng with that include in contrast to the example |
| 2:21PM |
1 |
Snom phones and P-asserted |
| 2:20PM |
0 |
One Week Until AstriCon 2008 |
| 12:52PM |
1 |
Voicemail: Thunderbird extension to play wav file in attachment? |
| 12:50PM |
1 |
how to force Asterisk 1.4 to use soxmix |
| 9:51AM |
5 |
What is in practice the maximum no of simultaneous calls that Asterisk 1.4 can handle |
| 9:49AM |
0 |
Redirecting SIP RTP with one Asterisk behind a NAT. The other is on Public IP |
| 2:44AM |
1 |
dundi |
| 2:11AM |
1 |
Context always defaults to PSTN |
| |
| Monday September 15 2008 |
| Time | Replies | Subject |
|---|
| 11:38PM |
0 |
rc6: Dunno what to do with STUN message 0101 ?? |
| 10:37PM |
5 |
RTCP-XR |
| 8:13PM |
0 |
Looking for an Asterisk consultant recommendation - LONDON, ON, CANADA |
| 7:54PM |
0 |
What characters can be present in SIP dial string passwords? |
| 7:46PM |
1 |
Need help with PHP script to authenticate user from database |
| 5:22PM |
1 |
FW: open source PBX survey |
| 4:47PM |
1 |
[FreeBSD] Right way to upgrade Zaptel from ports? |
| 1:50PM |
6 |
Callcenter monitoring tool |
| 12:22PM |
4 |
PBX appliances |
| 11:06AM |
1 |
UK call initiating party hangup control on analog home lines |
| 11:00AM |
2 |
Asterisk |
| 9:25AM |
1 |
call files hacking... |
| 9:20AM |
0 |
[OT] email netiquette (was: Re: Re: Asterisk realtime MySQL clients from same IP problem) |
| 3:48AM |
1 |
Setting up Asterisk to make calls using a VoIP provider and the regular phone line |
| 1:43AM |
0 |
asterisk-users Digest, Vol 50, Issue 38 |
| 1:28AM |
1 |
CallerID Resolution |
| 12:45AM |
0 |
asterisk-users Digest, Vol 50, Issue 37 |
| 12:29AM |
0 |
Degum Hardware Echo Cancellation |
| |
| Sunday September 14 2008 |
| Time | Replies | Subject |
|---|
| 10:59PM |
0 |
MGCP Configuration <=> ADIT 600 <=> T1 Port |
| 8:34PM |
1 |
MoH with an Aastra 9112i |
| 11:46AM |
2 |
Read dublicate dtmf |
| 4:56AM |
9 |
Streaming MoH on 1.4 |
| 12:40AM |
3 |
Can someone give a plain english explanation of the HASH function? |
| |
| Saturday September 13 2008 |
| Time | Replies | Subject |
|---|
| 10:19PM |
1 |
What if some phone picks up |
| 9:53PM |
1 |
Queue Calls getting stuck in there |
| 9:52PM |
0 |
Getting realtime ASR and ACD from Asterisk |
| 8:14PM |
0 |
Help...Failed to initialize G.729 copy protection! |
| 7:25PM |
2 |
rc6: can't get a dialtone |
| 2:36PM |
0 |
how to monitor traffic against specific Calling ID |
| 2:22PM |
1 |
Newber |
| 11:58AM |
0 |
Which internet phone protocol best to, choose |
| 10:43AM |
0 |
app_conference |
| 1:50AM |
0 |
Can the outbound SIP leg Call-ID be set to match the inbound SIP leg Call-ID? |
| 1:47AM |
1 |
Is there a way to get the remote User-Agent info from an outbound leg? |
| 1:39AM |
2 |
Append String to CIDNAME |
| 12:27AM |
2 |
cdr_adaptive_odbc writing CDR before h extension is processed |
| 12:13AM |
2 |
Sip Info events |
| |
| Friday September 12 2008 |
| Time | Replies | Subject |
|---|
| 10:44PM |
1 |
[FreeBSD 6.3/Ports] Make does nothing |
| 9:35PM |
0 |
OpenStage20 Problem |
| 9:00PM |
1 |
SIp Signalling |
| 7:55PM |
0 |
Transfer via AMI |
| 6:10PM |
0 |
echo cancellation problem with dahdi |
| 5:15PM |
2 |
Setup speed dials on Cisco 7921 |
| 5:03PM |
0 |
Encrypted IP phone compatible with Asterisk |
| 4:51PM |
1 |
SCCP - max lines per phone limit |
| 3:28PM |
4 |
Which internet phone protocol best to choose |
| 3:19PM |
2 |
SCCP port numbers used for audio stram? |
| 2:41PM |
1 |
Extension not found |
| 12:29PM |
0 |
Dial function, and no telephone line fixed in the fxo port |
| 11:52AM |
0 |
show g729 seems to no longer work in latest 1.4 version. What do I use please? |
| 7:59AM |
1 |
Amazing "show uptime" |
| 6:59AM |
0 |
VoIP Users Conference today at 12 Noon EDT |
| 5:07AM |
2 |
how to pass a variable in extensions.conf to AGI file |
| 3:40AM |
5 |
Asterisk and Fedora 9 |
| |
| Thursday September 11 2008 |
| Time | Replies | Subject |
|---|
| 9:57PM |
0 |
Possible Packet loss but need an opinion |
| 8:18PM |
5 |
Unable to run make menuselect for asterisk-addons |
| 7:29PM |
1 |
Probably very simple... call a number and play a sound? |
| 7:18PM |
1 |
about application Jack and its runtime |
| 3:56PM |
0 |
Asterisk calleri id resolution |
| 3:46PM |
1 |
dahdi vs zap with latest version of asterisk -- having some problems |
| 3:23PM |
0 |
g729 passthrough |
| 2:58PM |
0 |
[Re: Asterisk CDR Problem for Export CSV (Asterisk-stat-v2)] |
| 2:57PM |
2 |
asterisk 1.6.0rc6 make menuselect failed. |
| 2:00PM |
1 |
IVR response of the pound key |
| 1:31PM |
3 |
Outside SIP Caller accessing voivemail |
| 1:30PM |
0 |
Sarfaraz has invited you to join iDeezire - Keeps Connected! |
| 1:28PM |
0 |
redirection of called |
| 11:37AM |
5 |
BLF call pickup on Linksys SPA932 |
| 9:58AM |
2 |
meetme without zaptel |
| 8:09AM |
1 |
SHELL function strangeness |
| 7:59AM |
0 |
Language for app_queue, chan_local, chan_agent or whatever? |
| 1:10AM |
1 |
distinguish trunk from same host |
| |
| Wednesday September 10 2008 |
| Time | Replies | Subject |
|---|
| 11:56PM |
0 |
Zero time (nearly) call transfer. |
| 10:54PM |
0 |
Is there a way to get the Call-ID into the CDR? |
| 6:22PM |
3 |
Write Asterisk CDR MySQL records to multiple servers |
| 5:21PM |
2 |
Bell Canada (Nortel DMS100) PRI Outbound CNAM issue |
| 1:02PM |
1 |
Resilience using DNS or phone feature ? |
| 11:55AM |
1 |
How to make outgoing call from Pri ? |
| 8:10AM |
3 |
Newbie AEL2: Syntax for Hint |
| 2:56AM |
4 |
Asterisk and cloud computing (amazon EC2 + S3) |
| |
| Tuesday September 9 2008 |
| Time | Replies | Subject |
|---|
| 11:30PM |
0 |
nagios check_sip plugin |
| 10:23PM |
2 |
Asterisk phone conferencing performance |
| 10:04PM |
3 |
Pressing 0 to get an external line |
| 9:40PM |
1 |
Asterisk REFER |
| 9:34PM |
0 |
Asterisk 1.4.22-rc5, 1.6.0-rc6 and Zaptel 1.4.12.1 Released |
| 7:19PM |
2 |
SIP to IAX? |
| 7:00PM |
1 |
DruidCON 2008, 1-2 Oct in Atlanta GA, 2 free DruidCON conference passes to be given away! |
| 6:15PM |
0 |
Choppy Audio in One Direction |
| 4:49PM |
0 |
CLI and AGI question |
| 4:30PM |
2 |
Asterisk - Operator switch billing |
| 3:30PM |
0 |
AstriCon 2008 - Two Weeks To Go - Register Today |
| 3:13PM |
2 |
PRI auto-configure - continued from DEV list |
| 1:42PM |
1 |
Does X-Lite 'remember' Congestion state? (halfway OT) |
| 12:50PM |
5 |
Asterisk and Network Monitoring |
| 11:25AM |
0 |
Manager API -> call to agi |
| 11:15AM |
0 |
Call-Limit on Asterisk Cluster |
| 10:50AM |
2 |
Asterisk CDR Problem for Export CSV (Asterisk-stat-v2) |
| 9:17AM |
0 |
Session Progress |
| 8:50AM |
0 |
cli-originate and features |
| 6:41AM |
1 |
Dundi Help |
| 4:34AM |
1 |
Help needed creating gateway |
| 4:25AM |
0 |
Aastra Phone Parts |
| |
| Monday September 8 2008 |
| Time | Replies | Subject |
|---|
| 11:50PM |
1 |
Digium Hardware Echo Cancellation |
| 11:12PM |
5 |
OT: ARI |
| 9:45PM |
0 |
New Versions of Asterisk and DAHDI |
| 8:25PM |
0 |
Auto Attendant help |
| 7:43PM |
3 |
SIP Extension Config Issue |
| 7:08PM |
1 |
Asterisk T38 and Dialogic DMG 2000 |
| 6:11PM |
1 |
Multihomed Server Issues |
| 5:43PM |
2 |
Pointers to replace astdb |
| 5:07PM |
0 |
Newbie questions: seting up extension for miSDN |
| 4:32PM |
1 |
[OT] Re: Asterisk realtime MySQL clients from same IP problem |
| 4:00PM |
0 |
How to read DTFMs from MEETME_AGI_BACKGROUND without blocking? |
| 3:11PM |
1 |
Originate |
| 3:11PM |
1 |
Help Astmanproxy - AutoFilter |
| 2:15PM |
1 |
Video on Hold? |
| 1:51PM |
3 |
Asterisk realtime MySQL clients from same IP problem |
| 12:50PM |
1 |
Help about the Rxfax on asterisk |
| 11:06AM |
2 |
mISDN or BRIstuff ... |
| 9:35AM |
1 |
Which kernel mISDN to choose |
| 8:39AM |
3 |
how to disallow the native bridge between the two channel |
| 6:01AM |
0 |
CPU Usage 100% when Voicemail Notification is sent |
| 5:32AM |
0 |
Streaming live music into a conference room |
| |
| Sunday September 7 2008 |
| Time | Replies | Subject |
|---|
| 8:38PM |
5 |
iLBC and G729 codecs |
| 8:34PM |
2 |
IAX vs SIP |
| 1:22PM |
1 |
error while trying to compile dahdi-tools-trunk |
| 1:52AM |
3 |
Problems with 2 Asterisk servers on same LAN |
| |
| Saturday September 6 2008 |
| Time | Replies | Subject |
|---|
| 6:46PM |
1 |
realtime queue reload |
| 5:17PM |
1 |
[FreeBSD 6.3] Right-way to recover Zaptel? |
| 1:55PM |
0 |
call-limit problem |
| |
| Friday September 5 2008 |
| Time | Replies | Subject |
|---|
| 4:19PM |
2 |
FAX over T1 Question |
| 2:38PM |
1 |
Dear asterisk-users@lists.digium.com 79% OFF on Pfizer |
| 2:32PM |
1 |
svn branches for dhadi and its tools |
| 1:36PM |
0 |
FW: Vivox SLim |
| 12:20PM |
1 |
Call-leg stays on MusicOnHold forever |
| 11:37AM |
0 |
Grandstream Video Phones & Asterisk.. |
| 11:27AM |
2 |
Bridge 2 incoming calls |
| 2:58AM |
1 |
dahdi & tdm400p: no luck |
| 2:40AM |
0 |
Linksys 3102 - Call Waiting |
| 1:43AM |
0 |
libpri 1.4.5 priindication |
| 1:43AM |
1 |
The question about the M(X)option of Dial |
| 12:09AM |
2 |
Polycom BLF - multiple buddies |
| |
| Thursday September 4 2008 |
| Time | Replies | Subject |
|---|
| 8:44PM |
2 |
How to setup SIP so that RTP traffic flows from Source to destination |
| 7:28PM |
1 |
DAHDI FAQ not up. Anyplace else? |
| 5:59PM |
1 |
1.6rc4 chan_iax2 messages |
| 5:45PM |
1 |
extensions.conf programming? |
| 3:16PM |
7 |
ringback when the channel is answered |
| 2:36PM |
0 |
strange transfer problem |
| 2:01PM |
0 |
conf files for dahdi |
| 1:22PM |
0 |
Logs: messages, events, queue |
| 1:03PM |
0 |
New Install using DAHDI |
| 12:10PM |
0 |
MixMonitor + Originate |
| 11:14AM |
1 |
dial out via fxo gateway |
| 10:55AM |
1 |
iLBC codec |
| 10:28AM |
1 |
How to check mailbox exists (Received SIP subscribe for peer without mailbox) |
| 10:02AM |
0 |
Installing ValetParking? |
| 9:25AM |
1 |
#include changes in 1.4 |
| 9:12AM |
1 |
Stability problems in Asterisk 1.4.18 (and other 1.4.xx versions) |
| 8:15AM |
1 |
Dial L( x [: y ][: z ]) option truncates colon (:) using AGI /_ |
| 5:42AM |
1 |
Z-Wave or Zigbee for Office or Home automation using XML Browser enabled Screen Phone |
| 3:58AM |
4 |
ASTERISK supported Video phone |
| 2:46AM |
0 |
Voicemail "from an unknown caller" |
| 12:12AM |
2 |
All calls want to go out only on interface ZAP/g0 |
| |
| Wednesday September 3 2008 |
| Time | Replies | Subject |
|---|
| 9:31PM |
0 |
Combine sip audio and video from different sources |
| 9:18PM |
1 |
New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI |
| 8:37PM |
1 |
Asterisk with E1 interface vs IP PBX |
| 8:20PM |
3 |
DID number |
| 8:12PM |
2 |
Ringing on Console after a page |
| 7:41PM |
0 |
Asterisk voicemail message order |
| 6:55PM |
2 |
G722 and Asterisk 1.6 |
| 5:57PM |
0 |
sip to sip unplanned conference! help!! |
| 4:53PM |
0 |
SIP TLS / Nokia E51 |
| 3:24PM |
0 |
res_cepstral.so |
| 2:14PM |
2 |
Asterisk Crash |
| 1:18PM |
1 |
Live operator as a service? |
| 8:59AM |
1 |
MixMonitor-Saving Recorded file with AgentId. |
| 7:13AM |
1 |
multiple passwords for one meetme! |
| 6:50AM |
1 |
Newbie Polycom: ACD AgentLogin display on phone |
| 5:30AM |
0 |
Offering FIFO service to receptionist with LIFO hardphone ... |
| |
| Tuesday September 2 2008 |
| Time | Replies | Subject |
|---|
| 11:20PM |
1 |
Mark Spencer on TWiT's FLOSS Weekly with Leo LaPorte & Randal Schwartz |
| 11:16PM |
1 |
Selectively disable echo cancellation? |
| 6:58PM |
1 |
Dial timeout to cell phones |
| 4:41PM |
0 |
SALE 71% OFF on Pfizer |
| 3:07PM |
4 |
AgentCallbackLogin AddQueueMember |
| 12:56PM |
3 |
Asterisk Trunk and normal |
| 12:05PM |
0 |
zaptel 1.2.27 ? |
| 9:36AM |
1 |
SetCallerPres |
| 8:44AM |
1 |
play remote file |
| 5:19AM |
0 |
Dialplan terminates when the caller hangs up |
| 2:01AM |
2 |
Redundant PSTN PRI Gateways using Asterisk |
| |
| Monday September 1 2008 |
| Time | Replies | Subject |
|---|
| 11:08PM |
0 |
Still badly in need for mISDN help! |
| 5:11PM |
1 |
Documentation of users.conf |
| 2:15PM |
2 |
Asterisk 1.6 beta |
| 10:43AM |
2 |
lists.digium.com monthly reminders |
| 9:07AM |
1 |
Problematic Trunk SIP: Got SIP response 405 "Method not allowed" |
| 8:57AM |
0 |
not able to make call to landline no...to mobile works fine |
| 8:41AM |
0 |
Penalties for agents |
| 7:26AM |
3 |
Gateway errors |
| 3:28AM |
0 |
PhoneControl integrations |