Monday May 31 2010 |
Time | Replies | Subject |
8:08PM |
6 |
Voicemail : mail attachment to multiple mail-addresses |
7:17PM |
1 |
Definie gtalk troubles over here |
6:59PM |
1 |
Reloading queue members (realtime DB) |
3:13PM |
0 |
testing my asterisk 1.6.2.8-rc1 with gtalk (and JACK) - please help |
1:32PM |
3 |
Read and set the UUI in asterisk |
12:46PM |
1 |
Suddenly "HDLC Bad FCS (8)" errors on ISDN-PRI, changed nothing |
11:58AM |
0 |
Asterisk 1.4.31 IAX2/RSA BUG |
9:33AM |
2 |
Queue ringall problem. |
8:43AM |
1 |
Why Manager account log on and log off alternatively all the time? |
8:18AM |
0 |
IAX2 Load test |
4:31AM |
0 |
Dahdi PRI T1 Setup for TE210P |
|
Sunday May 30 2010 |
Time | Replies | Subject |
3:19PM |
1 |
Wierd behavior of illegal extension |
2:06PM |
6 |
How to use one single IP as origination |
1:12AM |
1 |
DID's for Chatham, ON |
|
Saturday May 29 2010 |
Time | Replies | Subject |
9:02PM |
6 |
Best way to limit outgoing calls per trunk |
12:20PM |
1 |
asterisk-users Digest, Vol 70, Issue 63 |
5:56AM |
2 |
Switchvox vs Asterisk codebase |
|
Friday May 28 2010 |
Time | Replies | Subject |
9:05PM |
6 |
Normalizing called numbers |
6:24PM |
2 |
Asterisk 1.6.2.7 + app_fax + OpenBSD 4.7 minor issue |
4:40PM |
3 |
DAHDI Help (made a cardinal sin :() |
8:43AM |
1 |
[Dahdi/system.conf] "fxsks = 1" deprecated? |
8:15AM |
0 |
Dead air between answer and packet2packet bridge (Bug 12708?) |
7:21AM |
0 |
ADA 1.1 features |
7:08AM |
0 |
Can't specify connection port with ADA |
6:44AM |
0 |
Is AstManProxy still recommended with 1.6 and later ? |
5:47AM |
1 |
call-waiting |
5:34AM |
0 |
Asterisk-based "Incredible PBX" |
12:00AM |
1 |
"pri show version" still shows old version despite doing a make && make clean && make install for v1.4.11 |
|
Thursday May 27 2010 |
Time | Replies | Subject |
11:12PM |
0 |
[X100P+Dahdi 2.3.0] Couple of questions |
9:24PM |
0 |
Aastra i740 and Asterisk |
9:17PM |
2 |
Pattern matching - how to ignore numbers after 10 digits |
7:38PM |
2 |
How to have Asterisk respond from the IP address used for registration |
7:06PM |
1 |
OpenVox B200P and D410P under Asterisk 1.6 |
6:15PM |
0 |
GoogleTalk to Asterisk - choosing voice menu options |
5:55PM |
0 |
IAX2 Call Transfer |
9:45AM |
0 |
N900 video with Asterisk? |
8:05AM |
1 |
Meetmee user introduction disabled |
3:12AM |
2 |
BRI card(B800P) doesn's work with DAHDI(wcb4xxp) in NT mode |
|
Wednesday May 26 2010 |
Time | Replies | Subject |
10:57PM |
0 |
meetme changes between asterisk 1.6.2.6 and 1.6.2.7 |
6:35PM |
0 |
Extension state can get stuck in 'Ringing' state |
6:34PM |
2 |
Music on Hold |
6:34PM |
0 |
call droped if second caller enter meetme conference |
5:18PM |
2 |
Getting 'username' of sip peer |
4:36PM |
3 |
"ring splash" |
4:27PM |
1 |
Libpri 1.4.11 Released |
4:20PM |
1 |
VoIP over virtualized VPN |
3:24PM |
1 |
Better AMD module |
3:17PM |
1 |
[Dahdi] "DAHDI_CHANCONFIG failed on channel 1"? |
2:59PM |
5 |
OT: Windows TAPI command-line driver |
12:55PM |
4 |
Help with IP Routing |
11:06AM |
1 |
Jack in /usr/local/ means failure for asterisk |
3:05AM |
1 |
Getting "ghost" transfer or music on hold |
2:48AM |
1 |
Error compiling DAHDI... |
|
Tuesday May 25 2010 |
Time | Replies | Subject |
10:59PM |
0 |
Using Sangoma Call Progress Analysis behind NAT router |
10:17PM |
0 |
DAHDI Linux 2.3.0.1 and DAHDI Linux 2.2.1.2 Released |
6:07PM |
1 |
How to get ConfBridge user count |
4:08PM |
0 |
Converting video files into .h263 |
3:50PM |
0 |
asterisk-users Digest, Vol 70, Issue 54 |
3:40PM |
0 |
Is it possible to get video working on h323 calls? |
3:14PM |
1 |
SuSE Firewall2 - Port Forward Command |
3:09PM |
0 |
1.6.2.8: need sip reload to reach peers. |
1:39PM |
1 |
nortel meridian question |
10:48AM |
2 |
Little t38 bug? |
5:05AM |
0 |
app_page.so was missing |
|
Monday May 24 2010 |
Time | Replies | Subject |
9:00PM |
1 |
mISDN compiling error |
8:56PM |
4 |
convert zaptel to dahdi? |
5:56PM |
1 |
sip and SSL |
4:16PM |
0 |
Agent Privacy - chan_local |
3:24PM |
2 |
TTS for asterisk |
3:08PM |
1 |
routing of calls |
1:51PM |
1 |
State of JACK support i9n Asterisk |
1:09PM |
2 |
Delay in IVR |
11:31AM |
1 |
[0017330] 1.6.1 and 1.6.2 + MySQL crases on ODBC Query (via func_odbc or sip realtime) |
11:05AM |
0 |
zap calls are getting dropped (unexpected disconnect message) |
10:29AM |
0 |
About testing Call transfer in asterisk |
4:40AM |
0 |
[Asterisk-User] Asterisk Video support |
|
Sunday May 23 2010 |
Time | Replies | Subject |
8:54AM |
2 |
Dahdi problems with kernel 2.6.32 |
|
Saturday May 22 2010 |
Time | Replies | Subject |
7:28PM |
4 |
US "Truth in caller id act"... and it's impact on services |
4:15PM |
1 |
Manual Web-meetme |
2:44PM |
1 |
Caller ID questions |
7:30AM |
1 |
OT - How to query vcard-like data for CTI app |
3:01AM |
1 |
voice recognition suggestion |
1:34AM |
2 |
About Sangoma cards and Asterisk integration with other PBX |
|
Friday May 21 2010 |
Time | Replies | Subject |
7:16PM |
0 |
Question about 1.6: multiple IP on a single Asterisk box / multi ISP routing |
5:29PM |
0 |
FollowMe dials numbers but can't confirm the call or hear anything |
3:40PM |
0 |
Can't load ooh323 on Centos x86_64: capabilities failure |
2:58PM |
1 |
SPA-932 BLF Stays solid red for single extension |
2:19PM |
2 |
Connecting 1-2 GSM ports to asterisk? |
12:41PM |
1 |
Hanging up call - no reply to our critical packet |
9:23AM |
0 |
quality of the call at the hangup |
7:48AM |
2 |
Using unix socket to connect with database |
7:34AM |
3 |
CANCEL Reason |
4:38AM |
0 |
file command with alaw file |
|
Thursday May 20 2010 |
Time | Replies | Subject |
7:51PM |
0 |
Asterisk 1.6.0.28 and 1.6.1.20 Now Available |
7:32PM |
0 |
Asterisk transfer to a conference using feature code? |
6:01PM |
0 |
Attended Transfer using AMI |
5:35PM |
3 |
Softphones on thin clients... |
4:43PM |
3 |
Checking blank CallerID in Dialplan |
3:41PM |
10 |
Which issue is keeping you from updrading to 1.6.2 ? |
12:58PM |
0 |
Friday @12 Noon and 1PM |
12:38PM |
1 |
Asterisk T.38 Gateway code testing |
11:47AM |
1 |
run extensions after call moved to queue and answered by member |
8:00AM |
1 |
Sending fake auth rejection for user |
7:12AM |
0 |
Early injecting Jack between call parties |
7:08AM |
1 |
DAHDI and ESXi |
|
Wednesday May 19 2010 |
Time | Replies | Subject |
8:00PM |
2 |
Cause and cure for "Exceptionally long voice queue length queuing to Local"? |
6:28PM |
1 |
Asterisk and RFC 3261 |
4:42PM |
3 |
DTMF Input from the User |
8:37AM |
0 |
Re-invite from Asterisk Server: Port number changes |
4:38AM |
2 |
Asterisk Cluster |
3:47AM |
2 |
a2billing DID and Queues |
|
Tuesday May 18 2010 |
Time | Replies | Subject |
7:36PM |
1 |
quick question on conf bridge |
6:20PM |
0 |
Location with PRI / Analog lines |
5:13PM |
2 |
NPA NXX Database |
5:10PM |
0 |
Peering with a Taqua T7000 |
3:26PM |
0 |
Faxes from website works, but from regular don't: cause 16 |
2:15PM |
3 |
About option U in Dial Ast version 1.6.2 |
10:14AM |
2 |
Asterisk 1.4.30 & T38 |
8:17AM |
1 |
automon filename does not follow the docs. |
6:52AM |
1 |
[ASTERISK-USER] Meetme Recording issue, recording is 2 times Faster then normal recording. |
6:28AM |
3 |
OT - Which SIP hardphone with 100 BLF ? |
1:26AM |
1 |
Callerid with DAHDI |
|
Monday May 17 2010 |
Time | Replies | Subject |
10:14PM |
0 |
new way to capture audio streams in calls |
7:22PM |
1 |
Commands to upgrade to latest Libpri? can I upgrade without touching Asterisk? |
7:13PM |
1 |
SIP SRV Registration problem |
6:01PM |
1 |
PRI down due to chan_zap.c: No more room in scheduler....Got SABME and Sending Unnumbered Acknowledgement...Any thoughts? |
4:31PM |
0 |
180 with SDP |
1:01PM |
0 |
ControlPlayback skip forward fails on mp3 file |
12:08PM |
1 |
R: new way of asterisk and kamailio(openser) realtime integration |
10:28AM |
3 |
Adding a context from the console |
10:00AM |
1 |
new way of asterisk and kamailio (openser) realtime integration |
9:16AM |
0 |
Hi |
8:46AM |
4 |
identify caller hangup or callee hangup? |
4:36AM |
3 |
Microsoft Response Point Voip server discontinued |
|
Sunday May 16 2010 |
Time | Replies | Subject |
7:00PM |
1 |
Digium TE121P + DAHDI |
2:15PM |
7 |
OK, I'm stumped |
11:16AM |
1 |
play a sound file directly to a caller channel |
7:52AM |
1 |
Aastra SIP phone regisration problems |
5:22AM |
2 |
Problems with Asterisk and two Linksys SPA941 |
12:28AM |
4 |
ISDN config: LBO values |
|
Saturday May 15 2010 |
Time | Replies | Subject |
8:32PM |
1 |
Re-compiling q931.c |
8:14PM |
1 |
q931.c modifications for CLID Presentation |
2:23AM |
0 |
Problem with Music on hold |
|
Friday May 14 2010 |
Time | Replies | Subject |
9:09PM |
1 |
1.6.2.7 SIP realtime problem |
4:56PM |
1 |
realtime queues "membername" problem |
4:31PM |
0 |
SIP and codec negotiation |
11:35AM |
1 |
Agents |
8:45AM |
1 |
is my PHPAGI Soap code right? |
4:53AM |
1 |
Do you think my server is being attacked? |
4:38AM |
1 |
aastra pt 480e phone |
4:05AM |
0 |
Channel cannot be released |
3:44AM |
0 |
Delay on DTMF with SpeechBackground and Vestec |
12:09AM |
2 |
Are there AMI commands to manipulate a voice mailbox? |
|
Thursday May 13 2010 |
Time | Replies | Subject |
11:24PM |
0 |
Asterisk Call Recording *1 Status Indication |
6:41PM |
1 |
What does Asterisk give to reject a re-invite? |
4:12PM |
0 |
Sip session timers. |
3:19PM |
0 |
Sending SIP credentials in INVITE |
3:10PM |
0 |
Skype for Asterisk and instant messages |
2:52PM |
0 |
Asterisk Sip Proxies and SIP persistence |
2:30PM |
0 |
asterisk-users Digest, Vol 70, Issue 30 |
2:14PM |
1 |
app_addon_sql_mysql.c:116 find_identifier |
1:52PM |
0 |
Asterisk Crashing with ERROR[1906] astobj2.c: refcount -1 on object 0xb1aab758 Ast Ver 1.6.2.6 |
8:17AM |
2 |
LAN IAX2 trunk bad audio quality vs. LAN SIP trunk good audio quality |
3:45AM |
1 |
Error at start of asterisk with cdr_addon_mysql.o |
|
Wednesday May 12 2010 |
Time | Replies | Subject |
9:03PM |
1 |
problems with unicall |
8:28PM |
0 |
Ringback |
8:26PM |
2 |
IAX2 - providers discontinuing support |
8:04PM |
1 |
Sangoma A101D PRI failing with ERROR - -- Got SABME from network peer. Sending Unnumbered Acknowledgement |
6:04PM |
1 |
pattern containing an asterisk |
6:03PM |
2 |
include sip configuration from another file in sip.conf |
2:59PM |
1 |
"bad magic number" log messages |
2:14PM |
2 |
asterisk-users Digest, Vol 70, Issue 25 |
1:59PM |
2 |
Stress Test new system |
1:12PM |
3 |
Asterisk core dumping on SendFax with FFA |
12:56PM |
1 |
Help finding online training |
11:38AM |
0 |
Additional CDR values |
10:36AM |
0 |
Have a macro update a channel variable |
10:07AM |
0 |
One way audio problem, a=sendonly and a re-invite |
9:51AM |
3 |
SIP trunk between two Asterisk servers |
9:29AM |
1 |
Voicemail() app not available? |
8:51AM |
0 |
problem of "Cannot release Channel" |
8:45AM |
0 |
Could Asterisk PHP agi be a SOAP Client? |
8:08AM |
1 |
No ringtone when going from queue to dial-command |
12:03AM |
0 |
Customizing Asterisknow distribution |
|
Tuesday May 11 2010 |
Time | Replies | Subject |
10:10PM |
1 |
iax calls via checkbox.cc |
7:39PM |
1 |
Digits and Vestec |
7:30PM |
5 |
Need fax solution for 1.4.xx |
6:25PM |
2 |
Lookup ${EXTEN} in database, update context/route if found... AGI? |
5:36PM |
0 |
E1 information |
3:32PM |
3 |
Problem with callerid(dnid) and queue |
3:23PM |
1 |
asterisk-users Digest, Vol 70, Issue 24 |
2:48PM |
1 |
conf files vs astdb |
2:31PM |
0 |
queue member state in asterisk 1.4 |
12:43PM |
0 |
AskoziaPBX 2.0 Released |
10:51AM |
1 |
bug in asterisk |
9:21AM |
2 |
Creating a HTTP Request on missed call? |
9:09AM |
1 |
Xorcom Astribank One call and dies |
8:57AM |
1 |
asterisk-users Digest, Vol 70, Issue 23 |
4:05AM |
4 |
AGI and Severe Weather Alerts |
2:31AM |
1 |
AsteriskNow |
|
Monday May 10 2010 |
Time | Replies | Subject |
11:19PM |
2 |
Speech/DTMF mix? |
10:30PM |
0 |
Recording with extension and agent in queue |
6:32PM |
1 |
Using Asterisk? Get on the press list! |
5:54PM |
0 |
Pass MWI from analog line through DAHDI? |
5:41PM |
1 |
Manipulating the Blacklist database |
5:38PM |
1 |
Working with Blacklist database |
3:54PM |
1 |
shoretel pbx |
2:35PM |
2 |
DAHDI not detecting hangup |
2:35PM |
1 |
More clarification on outbound sip channels. |
2:14PM |
1 |
Dialing a SIP Peer without using register strin |
12:46PM |
0 |
Sometimes called party answers, but callee keep hear ringing, called party hears nothing! |
12:36PM |
1 |
Continue dialplan is source channel hangs up |
12:29PM |
0 |
Unable to hear voice when called from PSTN phone line |
12:19PM |
2 |
Records sets and ODBC |
11:57AM |
1 |
Simulating a commercial SIP provider |
2:24AM |
0 |
Problem of hearing transfer' s sound |
2:18AM |
0 |
Problem of hearing attended transfer' s sound |
|
Sunday May 9 2010 |
Time | Replies | Subject |
1:15PM |
2 |
Re TrixBox |
10:14AM |
1 |
B410P and Patton smartnode : any success ? |
9:45AM |
0 |
Bug or feature: comments in chan_dahdi.conf.sample |
|
Saturday May 8 2010 |
Time | Replies | Subject |
8:04PM |
1 |
Conflict between jumper settings and dmesg with OctoBRI |
7:41PM |
0 |
OT - Newbie: Do I have a shared IRQ ? |
1:41PM |
1 |
JNET's qozap, dahdi and PCI-E Quad |
12:05AM |
3 |
text |
|
Friday May 7 2010 |
Time | Replies | Subject |
11:01PM |
1 |
DAHDI astribank Channel Unavailable |
8:52PM |
2 |
voipmonitor.org |
7:25PM |
0 |
SIP REGISTER header not containing Allow-Events or Allow |
6:17PM |
1 |
Multiple SIP lines. |
5:16PM |
1 |
Execute AGI, Then Continue |
4:37PM |
2 |
Asterisk Bible? |
3:35PM |
1 |
"Contact header appears incorrect on this invite" Asterisk registering with another PBX |
12:04PM |
0 |
asterisk and gnokii on same server: scratchy sound |
8:20AM |
3 |
Getting presence working in 1.6.2 |
7:24AM |
0 |
Issues with remote call setup |
5:23AM |
1 |
Video in Skype for Asterisk |
4:33AM |
1 |
Problem of "Playing 'pbx-transfer'" |
|
Thursday May 6 2010 |
Time | Replies | Subject |
11:30PM |
0 |
Contact header gets url decoded? |
10:46PM |
1 |
T.38 Fax With Flowroute SIP Provider |
9:36PM |
3 |
Possible bug in chan_sip:add_sdp |
8:11PM |
2 |
Questions About Fax for Asterisk |
1:44PM |
0 |
REALTIME in 1.2 |
1:00PM |
2 |
problem with trustrpid |
12:47PM |
1 |
Make the call finish after executing Dial(G()) |
|
Wednesday May 5 2010 |
Time | Replies | Subject |
10:44PM |
0 |
Still true: only first peer matched on incoming call? |
9:03PM |
0 |
T38 trunk configuration for relay appears to affect default trunks for voip |
6:33PM |
1 |
IAX2 Auto-congesting call due to slow response |
6:29PM |
2 |
Channels In Use |
5:01PM |
2 |
Registering a Cisco 7965 on 1.4.26 |
4:10PM |
4 |
VoIP Termination in Japan |
4:00PM |
4 |
What is billsec in CDR? |
2:17PM |
0 |
res_config_mysql - maximum field length for appdata |
1:39PM |
4 |
OT: NAT in SPA922 |
1:22PM |
2 |
Hash Dial Pattern Problems |
12:30PM |
1 |
SIP - SIP over PBX no audio when canreinvite=no |
11:51AM |
3 |
CDR to MS-SQL via ODBC issue |
11:46AM |
1 |
Confirm answering a call |
7:22AM |
0 |
BAD ROUND TIME FOR ANSWEREDTIME |
6:35AM |
1 |
Getting calee audio in Asterisk (real time) |
6:19AM |
0 |
Forwarding inbound mobiles |
4:58AM |
1 |
Transfer calls using ## |
3:29AM |
0 |
Problems with Asterisk 1.6.2.1 working in Realtime with PostgreSQL |
|
Tuesday May 4 2010 |
Time | Replies | Subject |
9:42PM |
1 |
Productivity Suite on Polycom IP7000 |
7:54PM |
1 |
problem with ringinuse=no, queue members receive randomly two calls |
5:59PM |
2 |
Asterisk 1.6.2.7 Now Available |
5:58PM |
0 |
Asterisk 1.6.1.19 Now Available |
5:57PM |
0 |
Asterisk 1.6.0.27 Now Available |
5:57PM |
0 |
Asterisk 1.4.31 Now Available |
5:56PM |
0 |
Bridging old system (ESI IVX E) with new Asterisk server - it is robbery! |
4:46PM |
3 |
client-server encryption |
4:23PM |
0 |
queue members |
1:41PM |
0 |
Problem with AMI Originate |
8:25AM |
0 |
Check if extension loaded over AMI |
8:15AM |
0 |
DSCP QoS value in YeaLink phone settings |
4:18AM |
6 |
Interesting email project. |
2:44AM |
1 |
Channel failover |
|
Monday May 3 2010 |
Time | Replies | Subject |
8:18PM |
1 |
Run a script after Page application |
4:59PM |
1 |
sending T.38 fax negotiation problem |
2:30PM |
4 |
Bridging old system (ESI IVX E) with new Asterisk server |
1:41PM |
2 |
Reading the CDR |
1:31PM |
0 |
CallerID problem with astribank |
1:22PM |
0 |
Spy on Asterisk 1.2 |
1:18PM |
1 |
BADTIME FOR ANSWEREDTIME |
11:15AM |
4 |
RTP ports |
10:04AM |
0 |
Hangup Detection |
9:00AM |
0 |
Parking problem with outgoing calls |
4:47AM |
2 |
Calling a RESTful Web service from Dialplan???? |
|
Sunday May 2 2010 |
Time | Replies | Subject |
7:41PM |
1 |
working example of t38 fax w/ 1.6.2? |