Wednesday April 30 2008 |
Time | Replies | Subject |
10:55PM |
0 |
Zaptel Channel Numbering |
9:39PM |
2 |
StatusComplete is getting me sick !! |
7:59PM |
0 |
Sobre retencion de liquidos... Interesante... |
7:07PM |
10 |
Discover connected Zap lines |
6:00PM |
2 |
Zaptel Compatibility |
5:27PM |
1 |
Connecting Analog to SIP gateway to asterisk server |
5:08PM |
1 |
One way audio... |
5:06PM |
0 |
AVAYA 8300 integration with asterisk 1.2.x |
4:57PM |
0 |
OT: anyone have a Jitterbuffer patch for Asterisk 1.2.28? |
4:48PM |
0 |
Queue setup and phones ringing |
4:42PM |
1 |
one way audio after call transfer |
4:33PM |
2 |
Customize Music On Hold |
4:01PM |
5 |
Asterisk - CRM Integration |
3:56PM |
2 |
Shared Line Appearance |
2:34PM |
0 |
Asterisk 1.4.17 and ExtensionStatus |
2:08PM |
0 |
asterisk-users Digest, Vol 45, Issue 85 |
1:36PM |
0 |
murf@parsetree.com Please Make Room on Your Hard Drive |
12:22PM |
1 |
Zap channels on TDM400P report to be unimplemented after upgrade from Asterisk 1.2 to 1.4 - (cause 66 - Channel not implemented error) |
12:11PM |
3 |
Asterisk on Xen or Dedicated |
9:34AM |
2 |
DTMF Issues |
5:20AM |
3 |
Portability in Asterisk |
4:49AM |
2 |
Sending caller name out PRI? |
2:13AM |
0 |
Jitter buffer not used in SIP -> chan_local -> ZAP path even with /nj for local channels |
|
Tuesday April 29 2008 |
Time | Replies | Subject |
7:31PM |
0 |
ooh323 asterisk 1.2.x |
6:28PM |
1 |
AJAM event subscription - Was: func_odbc creating records or best practice |
3:51PM |
0 |
Odd Zaptel Issue - Strange State 6? |
2:49PM |
2 |
Outbound international calls over BT ISDN30 |
2:49PM |
1 |
Debugging DTMF |
1:48PM |
1 |
need a monitor for asterisk |
12:49PM |
0 |
Digium TE420B (Four Port E1 PRI PCI-E x1 with Echo Cancellation), URGENT |
12:36PM |
6 |
RAS with Asterisk and PRI |
10:47AM |
0 |
changing of ssrc between early-media and call media |
10:30AM |
1 |
Annoying Sipura problem? |
9:13AM |
0 |
PRI CallerID - leading zero added |
8:25AM |
0 |
Polycom 320 working in direct mode, characters dialing and timeout |
7:02AM |
0 |
AddQueueMember() and PersistentMembers |
6:09AM |
0 |
Asterisk V1.6.0 SVN debug WARNING(6830) a bug or deliberate? |
12:49AM |
2 |
Anyone have pricing on the Color Polycom Phone? |
|
Monday April 28 2008 |
Time | Replies | Subject |
10:30PM |
2 |
func_odbc creating records or best practice |
10:03PM |
2 |
OT: Polycom 3.0 |
6:48PM |
0 |
Question on bridging |
6:09PM |
0 |
misdn, no free channels, similar to FAQ one |
5:34PM |
1 |
realtime queue callers |
3:07PM |
11 |
tftp issue |
1:29PM |
2 |
PRI hangup certain outgoing calls |
1:21PM |
4 |
Dell 1950 |
12:46PM |
3 |
(no subject) |
|
Sunday April 27 2008 |
Time | Replies | Subject |
3:41PM |
6 |
zap not coming online on fedora 8 |
1:32PM |
1 |
Sangoma A200 to A200D |
12:17PM |
2 |
Siemens Gigaset S685IP Review |
11:55AM |
0 |
Ploycom 320: characters and need to press DIAL if not registered |
6:46AM |
2 |
psql |
|
Saturday April 26 2008 |
Time | Replies | Subject |
10:41PM |
9 |
Manual Wardialer |
6:08PM |
1 |
[asterisk-dev] zap not coming online on fedora 8 |
4:31PM |
4 |
yum install for specific kernel, how? And zaptel on fedora core 8 |
4:02PM |
0 |
Channel variable settings |
2:53PM |
1 |
Outside call not coming through |
1:32PM |
3 |
Roaming callback? |
1:01PM |
0 |
how to retrieve sip tag from dialplan |
12:11PM |
4 |
Need comments on CRM development / Asterisk Customization |
|
Friday April 25 2008 |
Time | Replies | Subject |
11:40PM |
1 |
choopy audio when both side talk at the same time |
9:55PM |
3 |
Graphing Jitter Packet loss and Latency for SIP Calls |
8:23PM |
1 |
Asterisk using 100% of CPU |
5:58PM |
0 |
E-mail date is wrong |
2:31PM |
0 |
Friday Apr 25 @ 12 Noon EDT VoIP Users Conference |
11:55AM |
2 |
Upgrading to 1.4 |
11:31AM |
2 |
Cisco 7960 odd behaviour ... |
11:03AM |
16 |
Asterisk for larg |
9:33AM |
2 |
Cisco to Asterisk migration |
9:23AM |
0 |
SIP response 400 on attended transfer |
9:14AM |
0 |
DNS Problems during zaptel upgrade |
8:08AM |
0 |
using "m" switch in dialplan |
8:06AM |
3 |
noisy analog lines |
7:47AM |
0 |
Play sounds to both caller and callee at the same time |
6:23AM |
2 |
Playback / Background / Read choppy, but musiconhold fine, even with ztdummy |
1:57AM |
1 |
followme scenarios |
12:41AM |
0 |
No DTMF on Sip Connection between two asterisk boxes? |
|
Thursday April 24 2008 |
Time | Replies | Subject |
8:58PM |
1 |
Full queue issues |
8:48PM |
1 |
ATA FXO / FXS - can forward to sip ? |
7:43PM |
3 |
GoToIfTime problem |
7:37PM |
1 |
No CallerID Transfer Problem |
4:58PM |
6 |
Digium B410P or Sangoma A502D? |
4:51PM |
3 |
IAX issues with 1.4.19.1 |
4:26PM |
0 |
ring group question |
4:02PM |
2 |
No DTMF on Sip Trunk? |
4:01PM |
2 |
help...i cant do more... |
3:50PM |
2 |
Playing mp3-files – will it be OK? |
2:47PM |
1 |
Macro/Goto Help |
1:30PM |
1 |
T.38 VoIP providers |
10:18AM |
1 |
G723 pass thru |
8:51AM |
6 |
Forking in Dialplan |
8:11AM |
1 |
Newbie Polycom: Instant Messaging |
7:28AM |
3 |
MFC/R2 in chan_zap , Testers Wanted |
6:43AM |
0 |
Newbie Polycom: can Speed Dial display last name first? |
|
Wednesday April 23 2008 |
Time | Replies | Subject |
2:54PM |
0 |
cdr_pgsql and spool file |
2:17PM |
0 |
Asterisk SNOM and DTMF |
2:11PM |
0 |
ZapRAS, pppd plugin option |
1:37PM |
3 |
how to copy a variable without interpretation of the content |
12:37PM |
2 |
AEL: how to copy a variable without interpretation of the content |
11:37AM |
1 |
Next step in extensions.conf after answer the phone in Queue |
8:08AM |
0 |
Anyone using HKBN / 2b and have DTMF working? |
5:59AM |
2 |
prepaid on the trunks |
3:08AM |
0 |
question about asterisk setup... |
|
Tuesday April 22 2008 |
Time | Replies | Subject |
11:05PM |
1 |
Asterisk 1.2.28, 1.4.19.1, and 1.6.0-beta8 Released |
10:58PM |
1 |
AST-2008-006 - 3-way handshake in IAX2 incomplete |
9:02PM |
0 |
Can't transfer call |
8:13PM |
4 |
need examples of asterisk and mysql integration |
7:43PM |
0 |
OT - How to configure TAPI client-server ? |
2:12PM |
1 |
Cisco 7961 + 7914, speeddials, BLF & Asterisk 1.4? |
1:47PM |
0 |
Bip viol. in E1 cards |
1:10PM |
0 |
Cisco 79X1 speaker issue |
12:41PM |
2 |
Asterisk sends 486 Busy Here instead of 600 Busy Everywhere |
11:56AM |
2 |
WARNING: Remote host can't match request NOTIFY to call on Audiocodes MP-124 FXS |
11:26AM |
0 |
Check the answered channel in simultaneous sip call |
11:10AM |
18 |
Quality problems with ISDN PRI |
10:28AM |
1 |
lots of warnings from translate.c |
10:22AM |
1 |
features.conf Problem with DTMF_sequence |
10:01AM |
3 |
Parsing incoming extension till first @ |
9:11AM |
1 |
OT: Linksys devices send incorrect REGISTER |
8:55AM |
3 |
Conditional include=> ? |
7:58AM |
0 |
Strange SIP packet |
7:13AM |
0 |
caller groups |
3:17AM |
5 |
Can I roll my own E911? |
|
Monday April 21 2008 |
Time | Replies | Subject |
9:30PM |
2 |
Disable transfer on all calls |
7:29PM |
5 |
Switch recommendation? |
7:08PM |
1 |
Monitor v/s MixMonitor |
6:51PM |
1 |
Phone notification? |
6:47PM |
1 |
Dual Interface config |
3:41PM |
2 |
Monitor not merging calls |
1:34PM |
0 |
Dialplan Visualization (Extensions.conf orDialplan Show) |
1:31PM |
4 |
Click-to-talk (Java application) |
12:50PM |
0 |
RTCP stats |
12:14PM |
1 |
UPDATED Asterisk Jingle Extensions.conf |
11:33AM |
0 |
Asterisk Jingle<->SIP GW Question |
10:23AM |
1 |
re-Invite post call establishment (for RTP bypass) |
10:02AM |
1 |
OT: UMA in UK, any use? |
8:34AM |
2 |
re-invite (bypass asterisk) post call establishment |
8:20AM |
0 |
API Originate - action on reject/busy/congestion |
8:17AM |
0 |
sip channel - detect ringing (nvlinedetect??) |
7:58AM |
1 |
Basic Possiblity Question. |
6:53AM |
1 |
Digium TDM410P Cards |
2:38AM |
0 |
echo on a B410p |
|
Sunday April 20 2008 |
Time | Replies | Subject |
11:12PM |
2 |
imaps - voicemail |
8:23PM |
2 |
Problems with Quality Voice in a Asterisk-E1-Unicall |
8:11PM |
1 |
Outbound PRI ISDN 30 problems |
6:47PM |
1 |
FRAUD: BE AWARE |
4:44PM |
0 |
ISDN card freeze |
11:20AM |
1 |
chan_sip.c:3966 sip_indicate: Don't know how to indicate condition 9 |
|
Saturday April 19 2008 |
Time | Replies | Subject |
6:21PM |
2 |
meetme with time condition |
3:42PM |
2 |
func_curl.so Error on load |
|
Friday April 18 2008 |
Time | Replies | Subject |
10:56PM |
1 |
Asterisk PBX using Outbound proxy |
7:17PM |
1 |
Polycom RTP port range |
6:56PM |
1 |
REGISTER Outboundproxy |
6:53PM |
1 |
Polycom LDAP Corporate Directory |
6:48PM |
1 |
Dialplan extension priorities |
6:21PM |
1 |
WAN and LAN connections |
5:31PM |
0 |
videoconferencing with asterisk issues |
4:30PM |
1 |
Question on groups |
2:18PM |
0 |
OT Nice IBM 1U Server Gets Along w/Old and New Digium Boards Cheap X305 $199 |
8:39AM |
0 |
Friday @12Noon EDT: VoIP Users Conference on the Internet |
8:28AM |
3 |
Dialplan Visualization (Extensions.conf or Dialplan Show) |
8:15AM |
0 |
database connections question |
8:13AM |
1 |
Newbie Polycom: Subscription/Presence Problem |
5:53AM |
0 |
Guess I shoulda put a subject - sip diversionheader |
4:25AM |
2 |
SIP outboundproxy for asterisk |
|
Thursday April 17 2008 |
Time | Replies | Subject |
10:19PM |
0 |
(no subject) |
8:47PM |
1 |
multiple users collisions |
6:15PM |
1 |
End to end call monitoring? |
5:14PM |
1 |
users.conf and voicemail |
4:14PM |
2 |
G729 license count... |
4:03PM |
3 |
questions running 2 asterisk under the same LAN |
4:01PM |
0 |
status line header |
3:08PM |
1 |
sometimes UNREACHABLE!/REACHABLE doesn't appear in the log when it should |
2:48PM |
1 |
Differents routes for differents extensions |
2:45PM |
1 |
Sip or IAX device with professional balanced audio out |
2:34PM |
1 |
QOS for outgoing SIP ... Who needs QoS anyway! |
12:31PM |
0 |
FSX gateways |
11:41AM |
0 |
Constant ''CHANUNAVAIL' on PRI for Outgoing Only |
11:03AM |
2 |
buying cards from pakistan |
10:31AM |
1 |
call forking feature |
8:59AM |
2 |
Asterisk Warning 2512 |
8:59AM |
1 |
imap voicemail |
7:42AM |
0 |
keep incoming codec same as outgoing on sip proxy |
7:40AM |
0 |
keep incoming codec same as outcoming on sip proxy |
5:55AM |
2 |
keep one line open |
12:30AM |
0 |
Asterisk and LVS |
|
Wednesday April 16 2008 |
Time | Replies | Subject |
11:44PM |
1 |
lightweight prepaid app using Dial and extentions.conf |
10:49PM |
5 |
QOS for outgoing SIP calls |
8:25PM |
2 |
extenspy and chanspy |
7:51PM |
2 |
chan_zap error 1.4.19 tone duration |
6:27PM |
7 |
Drag and Drop transfer application |
6:20PM |
1 |
Problem with hints (1.4.19) |
5:51PM |
2 |
Chanspy on Asterisk 1.4.19 |
5:33PM |
2 |
Using Chanspy |
5:25PM |
3 |
PSTN to SIP |
2:36PM |
5 |
Best Click-to-call client |
2:19PM |
0 |
Callerid Error |
1:41PM |
1 |
DUNDi and SIP |
1:40PM |
3 |
Busy (congestion) signal and cell phones |
1:10PM |
1 |
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), |
12:57PM |
0 |
Version FIOS MWI Detection - asterisk-1.6-beta7 |
12:56PM |
1 |
Hangup conundrum with RxFAX |
10:47AM |
2 |
Simple queue announcements |
7:53AM |
1 |
Problem with B410P |
4:19AM |
0 |
[asterisk-announce] Zaptel 1.2.25 and 1.4.10 |
12:50AM |
4 |
X-Lite and Presence? |
12:30AM |
1 |
wcfxo and X100P card won't play nice. |
|
Tuesday April 15 2008 |
Time | Replies | Subject |
10:33PM |
0 |
Good article about VoIP, etc. |
9:46PM |
1 |
Global call limit |
8:51PM |
4 |
CDR and transfers! :( |
5:38PM |
0 |
Problem with SIP, attended transfer and GROUP_COUNT |
5:07PM |
2 |
app_swift v1.6.1 released for Asterisk 1.6 |
3:54PM |
1 |
PBX Console |
3:52PM |
0 |
asterisk online indicator |
3:52PM |
2 |
dialed number notify at invalid dial situation |
3:06PM |
0 |
[VOIP-Users-Conference] Re: Free FAX license from Pika |
2:58PM |
2 |
polycom 501 stopped working |
2:43PM |
0 |
Asterisk & Sys Admin in Chicago IL (full time) |
2:19PM |
1 |
gotoif syntax error |
2:07PM |
0 |
Asterisk on EC2 |
12:23PM |
1 |
problem with Asterisk 1.4.19 - accountcode dissapearing |
12:12PM |
0 |
Lypp/37 Signals mashup contest |
11:55AM |
2 |
What kind of Specs for Conference server |
11:08AM |
0 |
Polycom phone reboots |
10:06AM |
3 |
SIP response 480 "Do Not Disturb" |
8:22AM |
1 |
voicemail odbc storage |
7:10AM |
0 |
Patch for call deflection with libpri |
4:39AM |
5 |
Conferencing.. |
|
Monday April 14 2008 |
Time | Replies | Subject |
11:20PM |
2 |
Problem with SPA3000 -- dropping calls |
10:54PM |
0 |
CallerID in NZ |
10:00PM |
8 |
zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ? |
9:39PM |
3 |
Zap Codec |
8:06PM |
2 |
polycom auto answer |
7:05PM |
3 |
E911 Recommendations? |
4:52PM |
4 |
do cards just instantly go bad |
2:19PM |
1 |
sip.conf wont load completely |
9:24AM |
6 |
Recommend some good Click 2 Dial Application |
9:02AM |
4 |
Unable to load module chan_zap.so |
|
Sunday April 13 2008 |
Time | Replies | Subject |
5:40PM |
1 |
compilation of asterisk 1.4.19 with ilbc already on system |
3:46PM |
1 |
Similar option as promiscredir to use in transfer (REFER) |
4:11AM |
0 |
Voicemail ODBC bug or feature? |
3:34AM |
1 |
cdr_custom outout to serial port |
12:38AM |
6 |
way to inquire status of T1 link |
|
Saturday April 12 2008 |
Time | Replies | Subject |
8:27PM |
0 |
How to automaticaly close calls when Asterisk didnt receive the bye request |
8:00PM |
3 |
problem TDM01B |
3:43PM |
0 |
ATA for Fax with BroadVoice? |
1:02PM |
0 |
PSTN gateway alternatives |
2:43AM |
2 |
X100M never goes on-hook state |
12:32AM |
0 |
No compatible codecs / static noise |
|
Friday April 11 2008 |
Time | Replies | Subject |
11:25PM |
5 |
NAT issue with Fortinet Firewall |
10:32PM |
1 |
Asterisk and the Mitel SX 200 integration |
7:38PM |
1 |
ZD Net article |
5:57PM |
1 |
Correlating queue_logs and cdr for abandoned calls |
5:16PM |
4 |
Asterisk temporary hangs when no internet connection |
5:00PM |
0 |
Need good voicemail documentation |
4:59PM |
0 |
Asterisk trunk/1.6 and nvfaxdetect |
4:59PM |
0 |
nvfaxdetect, nvvoicemail, and others |
4:59PM |
0 |
SIP KEEPALIVES, with QUALIFY and without making UNREACHABLE |
3:05PM |
0 |
Cisco 7905 / 7911G Reviews |
2:56PM |
0 |
problems in REFER request to a different machine |
2:28PM |
0 |
testing the list |
2:23PM |
1 |
manger hangup call |
1:47PM |
1 |
Friday April 11th @ 12 Noon EDT VoIP Users Conference |
10:22AM |
2 |
Strange CLI behaviour |
6:40AM |
2 |
OT - How to check HPET is on and working before installing Asterisk ? |
6:29AM |
0 |
REALTIME and MACRO |
6:03AM |
6 |
bandwidth required for Asterisk running on T1 |
5:50AM |
1 |
Loosing SIP registration. |
5:48AM |
2 |
tdm410p w/ echo - no full duplex |
5:11AM |
1 |
TDM400P Dialtone problem |
4:39AM |
0 |
Running asterisk + T1 + ztdummy on Debian vserver |
3:36AM |
0 |
Newbie Polycom: Will a big 0000-directory.xml crash the phone? |
2:21AM |
1 |
odd error compiling zaptel-1.4.10 |
|
Thursday April 10 2008 |
Time | Replies | Subject |
11:26PM |
0 |
Any body tried MysqlPool-1.4 |
10:14PM |
0 |
Excellent Paper on ECHO in VOIP Environment |
8:48PM |
2 |
Digium T1 Card Crashing Server (Dell 2950) |
6:46PM |
7 |
Is Asterisk really good?? |
6:41PM |
1 |
Queue member state 'Not in use |
5:14PM |
2 |
Voicemail: afternoon audio file is missing |
5:00PM |
2 |
best way for call detail logging |
3:24PM |
2 |
Phantom Rings |
1:09PM |
2 |
[1.4.17] Disabled modules -> Zaptel stops picking up calls |
11:13AM |
0 |
A Simple Question |
8:27AM |
3 |
question about queue |
8:25AM |
0 |
followme than voicemail |
8:24AM |
0 |
Time out to disconnec: IAX trunk, SIP Trunk, Zaptel Channel |
7:04AM |
0 |
Festival in Italian does not work |
6:55AM |
3 |
Removing "Parsing /etc/asterisk/manager.conf" from CLI |
4:50AM |
0 |
Custom multilingual VM announcements & writing VM to 'default' context |
3:45AM |
0 |
Maximum Include level (10) exceeded- question |
3:29AM |
1 |
features on dial pad |
|
Wednesday April 9 2008 |
Time | Replies | Subject |
10:18PM |
1 |
multiple simultaneous access to single voice mail box |
8:49PM |
6 |
Jumped from 1.2.7 to 1.4.19, missing CLI colors |
8:02PM |
1 |
dial tree crawler? |
7:12PM |
1 |
setting dtmf mode for a particular peer |
5:37PM |
0 |
Magic Jack>>>> usable in Asterisk ? |
4:39PM |
2 |
zaptel 1.2.25 compilation error |
4:05PM |
1 |
Connecting Asterisk to Nortel Succession 4.0 sip... |
3:56PM |
1 |
[VOIP-Users-Conference] Potential subject for Friday - Does the Asterisk community need a 3rd party commercial software ecosystem? |
3:50PM |
0 |
PrivacyManager not working |
2:17PM |
1 |
Attrafax |
1:50PM |
1 |
*21*<number> # diverting |
1:49PM |
1 |
Queues +Exiting |
12:41PM |
1 |
Catch end of Eagi script when caller hung up...HELP ME PLEASE!! |
12:16PM |
2 |
MixMonitor fdiles |
11:52AM |
0 |
[VOIP-Users-Conference] Subject or Guest for Friday? |
10:30AM |
0 |
m switch in dialplan |
9:33AM |
0 |
Question about custom Asterisk billing engine |
8:07AM |
1 |
For Your Information - Our Experience with ATCom Phones... |
8:01AM |
3 |
queue logging |
5:49AM |
0 |
call forwarding |
5:37AM |
2 |
Message waiting indication(MWI) for voicemail - to H323 endpoints |
5:35AM |
0 |
MWI for voicemail - H323 |
|
Tuesday April 8 2008 |
Time | Replies | Subject |
11:54PM |
3 |
RTCP not being sent when on hold |
11:11PM |
3 |
Zaptel 1.2.25 and 1.4.10 released |
3:31PM |
1 |
Digium HPEC license counting |
11:23AM |
1 |
testing please ignore |
5:06AM |
1 |
Newbie Polycom: Where is SoundPointIPWelcome.wav used? |
2:01AM |
1 |
Anyone have a method of keeping an incremental tally of calls? |
12:05AM |
0 |
Quiet recordings |
|
Monday April 7 2008 |
Time | Replies | Subject |
8:47PM |
1 |
new instalation os asterisk |
6:16PM |
0 |
Eagi |
5:12PM |
1 |
Astribank |
3:47PM |
2 |
DTMF between Asterisk servers. |
2:31PM |
1 |
Laying out things correctly |
12:38PM |
1 |
exited non-zero on 'Zap/1-1' in macro ... |
12:36PM |
0 |
Parked calls and callerid |
12:25PM |
4 |
Asterisk 1.4.19 crash with Realtime using SIP peers |
7:33AM |
0 |
Timing issue for Music On Hold - has *anyone* come across this? |
7:12AM |
0 |
callerid problem |
5:27AM |
1 |
One Touch Recording |
3:31AM |
0 |
Problems with Unicall and TE122B |
|
Sunday April 6 2008 |
Time | Replies | Subject |
11:58PM |
1 |
Newbie Polycom: Headset Suggestion for IP601 |
9:02PM |
3 |
Need help with Cisco 7960 |
6:34PM |
0 |
OT Good Price $.099 - Voip/ Skype PC Handset UP609 "Regular $29.99" |
5:30PM |
1 |
UK POTS - Is there a better card than TDM400P available ? |
3:42PM |
1 |
Help, problems with calls sent from nextone gateway |
7:01AM |
7 |
Where is the Digium DS3 card? |
4:48AM |
1 |
Half-duplex call on TDM2400p with VPMADT032 |
|
Saturday April 5 2008 |
Time | Replies | Subject |
11:11PM |
1 |
Zaptel data mode not supported? |
11:07PM |
1 |
Question about Cisco IP phone + Asterisk + channels |
8:20PM |
9 |
Paging for analoge devices |
4:45PM |
0 |
TDM410 Callerid UK |
3:34PM |
1 |
TDM400P UK CID ISSUE |
12:16PM |
2 |
IAX IP Phone |
2:19AM |
1 |
SellVOIP |
12:55AM |
3 |
iaxmodem + hylafax w/ DID routing |
|
Friday April 4 2008 |
Time | Replies | Subject |
9:46PM |
4 |
Is there a distro with hlyafax rolled in? |
9:03PM |
0 |
One-way audio after music on hold |
8:23PM |
1 |
howto debug bad iax voice quality? |
8:06PM |
0 |
Low Volume on Recorded Voicemail Messages |
6:34PM |
0 |
Transfer BACK to CallManager over SIP trunk? |
6:24PM |
1 |
rxfax issue |
3:54PM |
0 |
discrepancy between CDR clid and Polycom IP601 clid |
3:02PM |
0 |
Problem about calling from atrixbox to pbx extension |
12:43PM |
0 |
Sample configuration files for ATAs |
12:18PM |
0 |
Communication between two asterisk servers on two different subnets |
11:51AM |
1 |
tokbox - voice and video in the browser |
11:14AM |
2 |
Quick Help, Anyone? EMERGENCY |
11:11AM |
0 |
Problems with E400P Cards |
10:35AM |
5 |
Ring back when free? |
10:09AM |
1 |
Next Move - Hosting |
8:26AM |
0 |
Problems with Analog - SIP phone conversations |
8:04AM |
1 |
Friday April 4th @ 12 Noon EDT: VoIP Users Conference (Asterisk!) |
7:28AM |
1 |
rxfax crashes Asterisk (segmentation fault) |
6:09AM |
0 |
Need help install rxfax/txfax |
5:46AM |
0 |
Forking using Openser And Asterisk |
5:19AM |
4 |
Advice on best operator phone (with attendant console) |
3:30AM |
2 |
SJphone behind NAT/Firewall without sound |
2:39AM |
2 |
Click to call |
|
Thursday April 3 2008 |
Time | Replies | Subject |
11:17PM |
0 |
About outdail SIPCALLID |
10:06PM |
1 |
Hearing "transfer" during call |
9:52PM |
0 |
Vitelity and AsteriskNOW |
9:28PM |
4 |
C# SIP API to Comiunicate with Asterisk |
6:53PM |
0 |
IAX - FWD status |
6:05PM |
0 |
Problems with analog <-> SIP phone confif\gurations |
5:40PM |
0 |
NAT when outbound call leg is not a local subscriber? |
5:34PM |
1 |
Sending audio to a channel |
5:10PM |
0 |
Asterisk (or maybe Zaptel) goes to "sleep" after inactivity? |
4:55PM |
0 |
Config file for 'make menuselect' available? |
4:38PM |
3 |
ISPBX Announces COGOBLUE Interface and PBX Appliances |
3:07PM |
3 |
Wait for dialtone feature on FXO device |
2:12PM |
2 |
Send DTMF digit every 15 seconds during a call |
2:11PM |
12 |
Web page to show online extensions? |
1:27PM |
1 |
Listening on conversations for training? |
11:56AM |
6 |
ztdummy |
11:10AM |
0 |
transfer the call from zap/1 to zap/2 (FXS) |
10:56AM |
1 |
rmirror.digium.com host unreachable |
10:36AM |
0 |
Asterisk i18n |
9:43AM |
0 |
call transfer issue |
9:39AM |
4 |
AsteriskNOW and IE |
8:17AM |
0 |
Strange problem with VoicemailMain |
7:01AM |
0 |
How does zaptel and Sangoma Media Gateway compare ? |
6:46AM |
1 |
Combined patch fixing queue-state and bug12127 for 1.4.x |
12:10AM |
0 |
problem with Kewlstart hangup detection |
|
Wednesday April 2 2008 |
Time | Replies | Subject |
9:17PM |
1 |
BRI hardware supported by 1.6 libpri ? |
8:45PM |
0 |
Asterisk parked calls and callerid |
8:37PM |
1 |
FW: [newtech-1] Skype 24 Hour Rolling Analytics |
8:26PM |
1 |
TE205P |
8:09PM |
0 |
Receiving 404 Not Found on all outbound calls when attempting SIP trunking between * 1.2.22 and Mitel 3300 CX |
7:18PM |
2 |
Analog modem as phone |
5:51PM |
0 |
OpenFrame |
5:06PM |
0 |
Asterisk 1.4.19 and Asterisk-addons 1.6.0-beta3 Released |
4:20PM |
0 |
zaptel_vldtmf |
4:01PM |
0 |
RTP no sound on asterisk |
2:37PM |
2 |
Problems with DELL 1600 |
1:28PM |
3 |
zaptel alarm |
12:56PM |
0 |
misdn config warnings in Asterisk 1.4.18.1 |
11:12AM |
1 |
CentPBX mirror? |
8:40AM |
1 |
show uptime and last reload |
8:11AM |
2 |
Howto connect to Cirpack softswitch with Asterisk ? |
7:03AM |
2 |
How to Hangup after DIAL is completed |
3:49AM |
2 |
How to wait before sending DTMF in DIAL command |
1:20AM |
1 |
FXS, Power and Sangoma |
12:46AM |
1 |
Virtual or Hardware SIP Modem |
12:42AM |
5 |
Asterisk on iPhone |
12:12AM |
1 |
call files |
|
Tuesday April 1 2008 |
Time | Replies | Subject |
10:48PM |
1 |
Asterisk and radius |
10:11PM |
1 |
TDM410E card, 1 FXO module - how to dial Out |
8:29PM |
1 |
g729 encoder/decoder |
8:15PM |
1 |
Calls randomly being placed on hold... |
7:31PM |
4 |
Voicemail- Recorded Mesage Low Volume |
6:31PM |
1 |
Digium B410P, bristuff and BRI support in 1.6 |
5:37PM |
3 |
Zaptel support removed from Asterisk |
5:24PM |
2 |
help with no audio |
2:29PM |
2 |
breaking into asterisk channel |
2:21PM |
0 |
Hangup problem with meetme |
2:09PM |
5 |
ZAPTEL |
1:25PM |
0 |
ECT implimentation |
10:14AM |
1 |
interrupting MOH |
9:31AM |
2 |
Realtime MOH |
12:38AM |
1 |
Unicall + incomplete DNIS on international calls |