Tuesday June 30 2009 |
Time | Replies | Subject |
10:00PM |
1 |
Authentication Issue Between Servers |
8:39PM |
1 |
MeetMe not prompting for PIN |
8:36PM |
0 |
Asterisk & Adit 600 Configuration |
8:17PM |
4 |
Extension status as XML for an Aastra 57i |
7:21PM |
2 |
Intercepting a Call while ringing a device |
5:42PM |
1 |
Reception of vocal SMSs to landlines. |
2:47PM |
2 |
IAX2 help needed... |
2:33PM |
1 |
Setting CDR(userfield) from Macro called from feature doesn't work with cdr_mysql |
1:52PM |
0 |
Problem with DTMF detection in ast_app_getdata (*1.2) |
1:21PM |
1 |
Asterisk 1.6 WaitForSilence Problem |
12:57PM |
4 |
Asterisk module trouble |
10:51AM |
0 |
Redirect with ExtraChannel on Bridged call give AMI event with second channel name AsyncGoto/...<ZOMBIE> |
10:45AM |
4 |
Echo and static on PRI with errors |
10:01AM |
1 |
DUNDi Errors (ENCREJ) |
9:22AM |
1 |
Opensips+asterisk problem |
9:13AM |
2 |
Dial Chan_local Usage |
9:02AM |
2 |
Echo and static on PRI with errors. |
9:00AM |
0 |
Can I add one h323 endpoint to register at asterisk? |
7:03AM |
1 |
Question regarding SIP 183 "Session Progress" handling in Asterisk |
6:38AM |
1 |
Remote UNIX Connection Hanging Asterisk |
2:42AM |
2 |
cisco phone 7911 & |
1:59AM |
0 |
Resetting CDRs on inbound calls |
1:22AM |
0 |
Restricting domains with SIP Trunking |
12:44AM |
5 |
Queue Issue (1.4.21.1) |
|
Monday June 29 2009 |
Time | Replies | Subject |
11:24PM |
2 |
Calling Number Verification Number? for BellSouth/AT&T |
8:01PM |
0 |
Asterisk ended with exit status 134... Asterisk exited on signal 6. |
6:13PM |
2 |
OT: Mobile voip - WCell |
5:26PM |
2 |
CRMy type app? |
2:47PM |
1 |
ISP< ->Asterisk <-> ATA <->DIALUP |
2:33PM |
1 |
Force Authentication |
1:49PM |
3 |
Calling non-extension numbers issue |
1:32PM |
0 |
Transfer dropping calls |
12:15PM |
0 |
asterisk 1.4.21.2 a caller waited in queue, after connect to agent hears silence |
9:51AM |
4 |
how to sniff RTP and SIP traffic only |
9:07AM |
1 |
unwanted locution |
7:35AM |
0 |
underlying sound during sip calls |
6:50AM |
1 |
about monitored calls storing |
6:49AM |
2 |
Callwithus.com is discontinuing IAX service |
1:45AM |
0 |
FW: re: Asterisk Outbound with Failover, alarm notification, dial status and hangupcause capturing to CDR from Dialplan |
|
Sunday June 28 2009 |
Time | Replies | Subject |
8:57PM |
0 |
BUG in Asterisk 1.6.1.0 and issue in DAHDI 2.1.0.4 |
4:54PM |
0 |
Recommendation / doubt about building of dialplan |
4:36PM |
0 |
Looking for real world uses of CallbackAgentLogin() |
9:33AM |
1 |
Sangoma A200 |
|
Saturday June 27 2009 |
Time | Replies | Subject |
8:25PM |
0 |
IPv6 |
6:48PM |
0 |
1.6.1: unable to create channel IAX2 to Junction |
6:19PM |
0 |
Autoreply: Re: using http to provision a Grandstrea GXP2000phone |
5:27PM |
2 |
using http to provision a Grandstrea GXP2000 phone |
9:33AM |
1 |
2 problems I can't solve without any help |
9:06AM |
3 |
Skype for Asterisk. Any return of experience ? |
5:50AM |
1 |
Multiple parking lots use default park positions |
3:01AM |
1 |
Call Parking timeout fails |
2:00AM |
0 |
Audio distorted local side only |
|
Friday June 26 2009 |
Time | Replies | Subject |
11:48PM |
3 |
IAX for internet file transfer? |
7:17PM |
1 |
G.729 licence in devices connected to Asterisk |
7:12PM |
2 |
Sounds format: GSM to G.729 |
6:56PM |
0 |
Higher Ed/University users of Asterisk? Free gift! |
4:51PM |
0 |
format_mp3.so in 1.6.1 |
4:03PM |
2 |
HW recommendations for small, cheap, reliable server |
4:00PM |
1 |
registration failed, not a local domain |
3:23PM |
0 |
Problem with RetryDial |
3:10PM |
4 |
T38 Fax Gateway for Asterisk 1.6 |
2:44PM |
2 |
Normalize Voicemail Volume? |
2:34PM |
0 |
Background command not interrupted as desired |
1:55PM |
0 |
Friday at 12 Noon EDT: VICIDIAL |
12:40PM |
1 |
NOT chan_mobile |
12:20PM |
0 |
Problem loss 2 seconds audio when Packet2Packet bridging |
8:42AM |
1 |
Centrale FastAgi server down |
1:55AM |
1 |
Calls dropping |
|
Thursday June 25 2009 |
Time | Replies | Subject |
9:09PM |
1 |
Persistent dynamic queue members |
7:47PM |
1 |
res_cepstral, register & existing Cepstral licenses. |
5:56PM |
2 |
video call doesn work |
5:17PM |
0 |
asterisk-users Digest, Vol 59, Issue 62 (Should be my PHP/AGI problem and odd behavior) |
3:25PM |
1 |
SIP registration fails |
3:05PM |
1 |
asterisk-users Digest, Vol 59, Issue 62 |
2:08PM |
0 |
AMI Transfer? |
11:43AM |
0 |
iaxclient softphone: quality? |
9:40AM |
1 |
hotdesk and voicemail |
|
Wednesday June 24 2009 |
Time | Replies | Subject |
10:53PM |
3 |
dahdi-linux-2.2.0 compile problem |
10:37PM |
0 |
How to set the II DIgits? |
9:11PM |
3 |
Removing line 2 from CISCO 7940g |
7:43PM |
7 |
PHP AGI Not Working and Odd Behavior |
7:20PM |
3 |
GUI for Asterisk |
3:10PM |
0 |
DAHDI Linux 2.2.0 and Tools 2.2.0 Release Announcement |
2:10PM |
0 |
Sangoma A200 and bristuff how install |
11:53AM |
0 |
Working chan_mobile/bluez anyone? |
11:45AM |
2 |
Announcement: Howler-optimised G.729A Solution for Asterisk |
11:01AM |
1 |
Outgoing CallerID for KPN in Belgium |
10:21AM |
1 |
Message Waiting Indication Astersk and kamailio |
9:52AM |
2 |
Asterisk + Jabber |
8:08AM |
0 |
problem with sangoma card a108d |
5:12AM |
0 |
Avaya 5610 SIP Firmware |
5:10AM |
0 |
Avaya 4620 SW SIP Config - not setting Proxy/Registrar |
4:22AM |
0 |
Are there any patches for chan_h323/chan_ooh323 to support video? |
|
Tuesday June 23 2009 |
Time | Replies | Subject |
9:10PM |
4 |
1000Hz kernel |
4:03PM |
1 |
ADM v. homemade code |
3:53PM |
0 |
PRI cause code discrepancy |
2:49PM |
1 |
[extensions.conf] Any idea why not working as it should? |
2:16PM |
0 |
Asterisk/Digium Press Opportunities: Community Contacts |
12:09PM |
1 |
SIP 482 Loop detected |
10:17AM |
4 |
GSM mobile trunks |
8:31AM |
5 |
error in playback of voiceprompt???? |
7:31AM |
2 |
music on hold file formats |
1:00AM |
1 |
How to force TDM410 to alaw |
|
Monday June 22 2009 |
Time | Replies | Subject |
9:12PM |
0 |
Documenting configuration with Real Time |
5:20PM |
6 |
Learn Asterisk |
3:57PM |
1 |
RTP/SIP traffic prioritization and Linux issues |
2:25PM |
4 |
Different inbound routes for each interface on a TDM800P card. |
1:27PM |
1 |
Crash process Asterisk |
6:51AM |
2 |
Realtime extensions |
5:19AM |
1 |
[hylafax-users] "No Carrier detected" sendig fax with Hylafax-Iaxmodem-Asterisk |
|
Sunday June 21 2009 |
Time | Replies | Subject |
9:23PM |
0 |
Asterisk and h323 gatekeeper functionality |
11:33AM |
0 |
Remote-Party-ID implementation on trunk |
3:49AM |
4 |
Nobody picked up in 20000 ms |
2:15AM |
3 |
Limit transfers |
12:15AM |
1 |
Meetme Talker Optimization |
|
Saturday June 20 2009 |
Time | Replies | Subject |
10:51PM |
0 |
Configuring Deltacom pri in Florida |
10:15PM |
1 |
PRI cause codes |
7:48AM |
0 |
Join Asterisk Global Meeting Sunday June 21 using VOIP - BerkeleyTIP |
6:22AM |
1 |
Fw: RE:Nagios under *[solved] |
3:24AM |
2 |
newbie questions |
|
Friday June 19 2009 |
Time | Replies | Subject |
8:54PM |
1 |
asterisk 1.6 and mISDN |
7:08PM |
2 |
IMAP voice mail storage |
5:02PM |
1 |
Switchvox HA options |
3:25PM |
1 |
Strange res_config_odbc error messages in 1.6.1.1 |
11:43AM |
0 |
Asterisk and EC2 today at 12 Noon EDT |
8:58AM |
2 |
agent login status visual clue on Polycom? |
8:34AM |
2 |
Cisco 7941G & Auth |
8:04AM |
1 |
help setting tone zone |
6:45AM |
5 |
Dail in modem |
5:59AM |
0 |
Asterisk Flite Problem |
4:28AM |
1 |
Anonymous Connection form IP to use specific Context |
4:24AM |
2 |
Recompiling dahdi-linux after kernel update - To minimize downtime |
3:18AM |
0 |
Analogue card recommendation |
|
Thursday June 18 2009 |
Time | Replies | Subject |
9:47PM |
5 |
asterisk-gui: http://id_address:8088/asterisk/static/config/cfgadvanced.html |
9:18PM |
1 |
Multiple Outgoing Lines: extensions.conf (Ioan Indreias) |
6:27PM |
3 |
asterisk-gui: read/write in the conf files or db |
6:15PM |
3 |
Nagios under * |
5:49PM |
2 |
Voicemail Password |
5:40PM |
2 |
dahdi and overlapdial problem |
4:41PM |
2 |
Speex problem installing on CentOS 5.3 |
3:08PM |
2 |
Asterisk + mySQL |
2:21PM |
1 |
Configuring Asterisk behind a SIP Proxy |
11:25AM |
2 |
snom mass deploy help |
10:19AM |
2 |
Multiple Outgoing Lines: extensions.conf |
10:08AM |
2 |
Asterisk on AVR32 |
9:12AM |
1 |
Asterisk 1.6.1 and dahdichanname = no |
8:31AM |
2 |
asterisk and openvpn and sip |
6:35AM |
0 |
failover trunk config. |
6:16AM |
2 |
how can I get Better natural Voice in Festival |
3:06AM |
2 |
Incoming SIP and the 's' extension |
2:41AM |
1 |
help setting up transfering |
|
Wednesday June 17 2009 |
Time | Replies | Subject |
11:10PM |
1 |
Redundant Connectivity |
11:07PM |
1 |
Wideband (G722) MeetMe |
10:48PM |
2 |
What causes this error? |
10:35PM |
2 |
asterisk-gui: read/write in the conf files or db? |
10:06PM |
1 |
Function IMPORT and Local channels |
9:48PM |
0 |
File Permissions On Voicemails Left To Multiple Recipients |
7:20PM |
1 |
Incoming Call trouble with new *Now 1.5 setup |
6:34PM |
2 |
Nagios Asterisk |
6:03PM |
3 |
gap between Playback and Queue |
4:14PM |
3 |
Asterisks, Sip to Local PRI/PTSN issue |
3:27PM |
1 |
Polycom Stop Downloading Config |
1:03PM |
1 |
Debug: how to print a variable? |
12:16PM |
2 |
Scaling |
10:13AM |
1 |
MP3 File Play In Read application |
7:22AM |
2 |
modifying CID for different trunks |
2:09AM |
1 |
Installing LUA |
|
Tuesday June 16 2009 |
Time | Replies | Subject |
8:57PM |
1 |
OT: Possible Fraud-Mike Low/Zigit/ZonFon/CallCheap |
8:15PM |
0 |
the correct way to setup a transfer with REFER in SIP |
1:39PM |
2 |
no sdp or contact replacement using externip |
11:22AM |
1 |
missing chan_dahdi.o in debian asterisk 1.4.x |
11:02AM |
2 |
Update Caller-ID after Dial() |
7:29AM |
1 |
Unable to use # as feature key prefix |
7:23AM |
2 |
feature keys no longer work after a call has been parked |
3:46AM |
1 |
No exten available after pass between servers |
1:00AM |
2 |
tdm loosing interrupts and latency |
|
Monday June 15 2009 |
Time | Replies | Subject |
5:28PM |
2 |
Newbie, Question on making a PSTN call.. |
5:06PM |
0 |
Asterisk 1.6.2.0-beta3 Now Available |
4:31PM |
0 |
Bug or feature : how to customize SIP REFER from dialplan |
3:25PM |
0 |
Open Source Call Statistics / Metrics Packages |
1:46PM |
1 |
Suggest Multi-tenant Predictive Dialer ? |
1:44PM |
2 |
asterisk and google talk |
12:31PM |
0 |
Suggest Multi-tenant Hosted PBX ? |
11:55AM |
2 |
Click-to-dial CTI for Windows |
9:47AM |
0 |
external RTP IP address |
9:13AM |
0 |
OT - Aastra - mapping transfer key |
9:06AM |
2 |
How to remove a GLOBAL variable from diaplan ? |
9:03AM |
1 |
Function IMPORT |
8:37AM |
0 |
setting codecs on the fly |
7:57AM |
1 |
Opinion on Attended transfer in features.conf |
|
Sunday June 14 2009 |
Time | Replies | Subject |
9:31PM |
0 |
DNS queries based on channel name? |
6:13PM |
6 |
Open Source Soft Phone |
1:15PM |
0 |
No voice from the callee |
2:13AM |
2 |
FXS - TDM400 - No dial tone |
|
Saturday June 13 2009 |
Time | Replies | Subject |
11:31PM |
2 |
Polycom registration errors |
3:27PM |
1 |
Dial with r option doesn't use 'ring' tone as defined in indications.conf |
1:59PM |
1 |
Is it possible to do this? (forward a call w/ 3-way calling)? |
11:24AM |
1 |
Preventing MOH from restarting the tune when a call is parked |
4:23AM |
1 |
1.6.0.10: core restart on ReceiveFax() |
1:01AM |
0 |
Fedora Core 10 and g729 codec |
|
Friday June 12 2009 |
Time | Replies | Subject |
7:48PM |
1 |
AmooCon video recordings online |
3:40PM |
1 |
Asterisk + TC400B - Clock Trouble |
3:15PM |
2 |
sending sip info messages |
1:58PM |
5 |
Help building dahdi for debian |
12:01PM |
0 |
Friday 12th June @ 12 Noon EDT: VoIP Users Conference Skype to ZipDX |
8:49AM |
1 |
multiple PRI's in one group ..how?? |
8:06AM |
1 |
Simple Queue Problem |
7:38AM |
0 |
PRI connection with ZTE exchange over T1 PRI |
7:19AM |
0 |
Dynamic DNS (was asterisk-users Digest, Vol 59, Issue 28) |
6:51AM |
1 |
asterisk-users Digest, Vol 59, Issue 28 |
5:03AM |
0 |
FXO and fax-on-demand |
4:11AM |
0 |
Problems with ReceiveFAX (asterisk 1.6.0.3 and t38) |
1:39AM |
2 |
Current possible values for DIALSTATUS? |
12:50AM |
2 |
Writing for asterisk |
|
Thursday June 11 2009 |
Time | Replies | Subject |
9:17PM |
0 |
Polycom Digitmap |
7:30PM |
3 |
SIP hacked connection? |
5:47PM |
2 |
Automatic Calling Feature? |
11:45AM |
0 |
Asterisk 1.4.26-rc2 Now Available |
7:59AM |
1 |
cant use h,1 at cancel! |
7:21AM |
2 |
OT - Aastra phones provisioning |
6:03AM |
1 |
cisco MC3810 weirdness with asterisk |
12:02AM |
2 |
In Dahdi: what we use instead of /sbin/ztcfg -vv |
|
Wednesday June 10 2009 |
Time | Replies | Subject |
11:46PM |
1 |
PrivacyManager no longer working properly |
10:00PM |
2 |
Asterisk - SIP - TCP and Exchange 2007 Unified Messaging |
8:44PM |
1 |
problem with transfer application (REFER) |
3:14PM |
2 |
T38 support |
3:04PM |
1 |
Dialer program |
2:42PM |
0 |
Dial option limit call duration |
2:24PM |
0 |
sip calls not going through |
2:00PM |
2 |
Chameleon Mail |
1:40PM |
1 |
Resetting Marker Bits |
1:19PM |
1 |
External PRI Appliance |
1:16PM |
1 |
Rhino analog cards |
11:17AM |
3 |
Query about tdm410 cards |
10:42AM |
0 |
Problem with attended transfers |
9:48AM |
5 |
Problem with voicemail and NFS |
8:27AM |
0 |
optimising asterisk sounds for g722 |
6:38AM |
0 |
DAHDI and ZAPTEL for automatically start (rc.local) |
|
Tuesday June 9 2009 |
Time | Replies | Subject |
9:55PM |
0 |
zap not coming online on fedora 8 |
7:58PM |
0 |
Help - create_addr_from_peer: 'UDP' is not a valid transport for 'exten1'. we only use 'TLS'! ending call. |
6:58PM |
5 |
IAX2 issue? |
1:32PM |
5 |
voicemail |
12:32PM |
0 |
FXO- no dial tone- no call progressing |
10:51AM |
2 |
hfcpci with 1.6 ? |
8:08AM |
1 |
Bitwise AND |
2:42AM |
0 |
alsa no input |
|
Monday June 8 2009 |
Time | Replies | Subject |
7:32PM |
1 |
Help with asterisk core dump |
5:48PM |
0 |
SendText and sipsak |
4:34PM |
0 |
How to use Dial G option in AEL |
4:29PM |
1 |
OT: Grandstream, call pickup, ... |
1:39PM |
2 |
Asterisk VM and Android phone? |
1:18PM |
1 |
MeetMe: Mute All Lines Automatically? |
12:32PM |
1 |
SIP Strict Routing and canreinvite |
12:18PM |
2 |
Asterisk manager login with java not working |
11:57AM |
4 |
Best free text to speech.. |
11:32AM |
1 |
Timeout when dialing dead peer |
11:06AM |
3 |
T.38 pass-through 488 handling problem |
10:25AM |
0 |
Push to Talk with Call Drop-Out? |
8:03AM |
2 |
Snom, Asterisk and Patton SN1400 - sending bye instead of hold |
5:43AM |
1 |
Achoring MEdia |
1:10AM |
0 |
remote queue members |
|
Sunday June 7 2009 |
Time | Replies | Subject |
4:51PM |
2 |
Call recording in - out |
4:21PM |
0 |
BUSYDETECT_* flags |
2:35PM |
1 |
ANI |
12:53PM |
0 |
Callback with a2billing |
10:19AM |
1 |
chan_dahdi missing in * 1.6.1.1 |
8:20AM |
1 |
Called party name with Cisco-2,811 gateway |
|
Saturday June 6 2009 |
Time | Replies | Subject |
7:05PM |
1 |
Teliax: where's the space in CALLERID(num) from? |
6:47PM |
2 |
Sangoma A200 wt HW echo on 64bit Centos , 4 fxo and crosstalk! |
3:14PM |
1 |
What does it mean rc in the release version |
1:08PM |
1 |
FXO clock |
9:33AM |
5 |
DAHDI, and 64 bit machine |
|
Friday June 5 2009 |
Time | Replies | Subject |
10:31PM |
0 |
Asterisk 1.2.33, Asterisk 1.4.25.1, Asterisk 1.6.0.10, and Asterisk 1.6.1.1 Now Available |
9:25PM |
5 |
How run AsyncAGI commands in background |
10:55AM |
1 |
Help with inbound dialplan |
8:06AM |
0 |
Friday June 5th @12 Noon EDT: Sipgate invades the USA, more HD Voice, Video chat |
1:34AM |
1 |
DTMF Problem w/ MeetMe |
|
Thursday June 4 2009 |
Time | Replies | Subject |
10:12PM |
3 |
PHP/AGI/SetVar Issue |
7:26PM |
2 |
DECT USB dongle - an Asterisk channel? |
6:24PM |
2 |
broken pipe in perl agi |
6:08PM |
6 |
Phones dropping registration, but asterisk thinks phones are still registered |
4:15PM |
7 |
Asterisk AGI issues (at high load) |
3:23PM |
3 |
Question about core CDR system for multilpe servers |
12:56PM |
2 |
Digium Fax Driver |
12:00PM |
1 |
problem install Asterisk-FastAGI |
9:27AM |
1 |
how we can put anybody on hold using Asterisk with analog phone |
7:40AM |
1 |
CDR question |
4:38AM |
1 |
Asterisk eventually fails when connection dies |
4:22AM |
1 |
Meetme timeout |
|
Wednesday June 3 2009 |
Time | Replies | Subject |
6:38PM |
1 |
Using DIALSTATUS question |
5:01PM |
0 |
RES: RES: SIP Response 181 - Is it possible in A steri sk? |
3:34PM |
1 |
IAX2 Channel Information |
2:00PM |
2 |
Can asterisk work here |
11:51AM |
0 |
Could not stop autoservice on calling channel |
7:37AM |
3 |
IP phone recommendation |
|
Tuesday June 2 2009 |
Time | Replies | Subject |
8:13PM |
1 |
does transcoding take place when a SIP call (both ends using same codec) gets forwarded over T1? |
7:15PM |
0 |
PSTN Interface Card |
4:14PM |
0 |
Segfault on unload of chan_h323 in asterisk-1.4.25 |
3:51PM |
0 |
Delivery reports about your e-mail |
3:14PM |
4 |
Realtime LDAP passwords |
2:41PM |
0 |
DAHDI Linux 2.2.0-rc5 and Tool 2.2.0-rc3 Release Announcement |
2:20PM |
4 |
RES: SIP Response 181 - Is it possible in Asteri sk? |
1:38PM |
2 |
SIP Response 181 - Is it possible in Asterisk? |
1:31PM |
1 |
Asterisk maximum user |
1:14PM |
3 |
Call quality - how to debug |
10:07AM |
0 |
problem with outgoing calls |
8:52AM |
1 |
zaptel to dahdi |
8:06AM |
2 |
error with dial timeout |
6:10AM |
0 |
Polycom IP321? |
12:08AM |
1 |
First ever Open Source Asterisk / Wave bounty |
|
Monday June 1 2009 |
Time | Replies | Subject |
11:07PM |
0 |
Wave and Asterisk |
7:47PM |
2 |
extensions not being detected consistently |
6:52PM |
0 |
Suddenly the voice became karbage (like robot) using Asterisk |
6:47PM |
1 |
Asterisk 1.4.26-rc1 Now Available |
6:31PM |
0 |
Astlinux 0.6.6 Release |
4:08PM |
0 |
Playing sounds on local channels in Asterisk 1.6.1.0 |
4:02PM |
2 |
SVN vs "Regular" Asterisk |
2:23PM |
1 |
CPU usage vs compiler flags |
10:42AM |
1 |
Digits timeout (ISDN) |
9:52AM |
2 |
Transfer call from analog telephone |
9:43AM |
6 |
MeetMe and setting conference timeout |
9:27AM |
3 |
[Atcom] Asterisk + LAMP on 128MB RAM? |
5:36AM |
1 |
IAX2 trunking with Older Asterisk, version ? |
12:57AM |
1 |
Suddenly the voice became garbage (like robot) using Asterisk 1.4.19.2 |
12:46AM |
2 |
Asterisk 1.4.25 and zapata.conf |