| Friday November 30 2007 |
| Time | Replies | Subject |
|---|
| 11:44PM |
3 |
Only call me once |
| 11:26PM |
0 |
v33 of codec_g729a released |
| 11:20PM |
0 |
Asterisk-addons 1.4.5 Released |
| 10:56PM |
2 |
Problem registering Cisco 7970 phone with Asterisk 1.4 running FreePBX |
| 7:47PM |
1 |
Outgoing PSTN calls , unusable voice quality |
| 7:09PM |
1 |
OT - How to add a new TAPI driver on an XP system ? |
| 6:26PM |
2 |
Remote Office, Centrally Shared Voicemail |
| 6:16PM |
3 |
How to setup redundant SIP peers |
| 4:25PM |
1 |
Simple Asterisk to Asterisk SIP Call Setup? |
| 3:12PM |
3 |
Do While loop |
| 2:57PM |
1 |
Asterisk 1.4.15 crash without generating core file |
| 2:20PM |
1 |
Nov 28, 2007 Asterisk Poll Results |
| 1:30PM |
1 |
Off-Topic: Avaya |
| 1:26PM |
1 |
Suppressing certain queue announcement voice prompts |
| 10:06AM |
0 |
OT - Which TAPI driver to use ? |
| 9:17AM |
2 |
My AsteriskNo unable to registration |
| 8:52AM |
4 |
IAX complaints? What are they? |
| 7:35AM |
4 |
How to originate a call from console CLI ? |
| 3:30AM |
0 |
Sip 1.4.x DTMF detection not working |
| 2:07AM |
2 |
Newb Question |
| |
| Thursday November 29 2007 |
| Time | Replies | Subject |
|---|
| 11:10PM |
1 |
Call Parking/Pickup on a single button |
| 11:07PM |
0 |
AST-2007-026 - SQL Injection issue in cdr_pgsql |
| 10:54PM |
0 |
AST-2007-025 - SQL Injection issue in res_config_pgsql |
| 10:10PM |
0 |
Asterisk 1.4.15 and 1.2.25 Released |
| 7:29PM |
1 |
Adding new recorded phrases to the release |
| 7:29PM |
2 |
Using existing extensions.conf macros, and co-habitation |
| 7:17PM |
1 |
Transfering IAX context |
| 6:05PM |
0 |
Protection switching on PRIs. |
| 5:31PM |
1 |
SLA: Handling of errors in outgoing call |
| 5:28PM |
1 |
Problems with Asterisk 1.4.14 and Queue app |
| 4:59PM |
1 |
least cost routing and asterisk-1.4 |
| 1:46PM |
1 |
roundrobin and rrmemory with pre-defined agent order |
| 1:39PM |
0 |
asterisk-users Digest, Vol 40, Issue 82 |
| 11:59AM |
0 |
Is it better to use debian binary or compiled version? |
| 11:32AM |
1 |
Hylafax |
| 10:12AM |
0 |
Needed Hardware |
| 8:17AM |
0 |
CDR n Dial A option |
| 7:52AM |
0 |
[Copfilter] Copy of quarantined email - *** SPAM *** [6.5/6.0] Asterisk <-> Nortel Phone Switch |
| 7:52AM |
0 |
[Copfilter] Copy of quarantined email - *** SPAM *** [6.6/6.0] Problems getting Asterisk to detect call in |
| 7:49AM |
1 |
IP Trunk and increasing volume level on diguim card |
| 7:48AM |
0 |
[Copfilter] Copy of quarantined email - *** SPAM *** [7.4/6.0] Re: Asterisk <-> Nortel Phone Switch |
| 6:13AM |
0 |
queue_log duration=NULL |
| 2:20AM |
0 |
FSK signal start after second ring |
| 12:56AM |
2 |
Realtime SIP & BLF |
| |
| Wednesday November 28 2007 |
| Time | Replies | Subject |
|---|
| 10:27PM |
1 |
Cross-compiling asterisk-1.4 for Debian on a slug |
| 9:42PM |
4 |
G729/MOH Quality |
| 9:18PM |
1 |
Polycom MWI's will not turn off |
| 9:01PM |
0 |
No ACK on 200 OK |
| 8:49PM |
1 |
Unable to lookup host in c= line, |
| 8:41PM |
2 |
What is voice format 8 |
| 8:06PM |
1 |
Asterisk <-> Nortel Phone Switch |
| 6:56PM |
0 |
troubles with res_pgsql |
| 6:49PM |
3 |
Bad audio quality in 1.4-SVN when encoding to alaw/ulaw |
| 5:52PM |
1 |
Digium TE120P versus Sangoma A101D-X |
| 4:52PM |
4 |
Sangoma Question |
| 4:21PM |
5 |
To DB or not to DB? |
| 3:08PM |
2 |
Shared line appearance phones? |
| 2:10PM |
1 |
Fw: Remove a TDM Card |
| 2:02PM |
1 |
test |
| 1:52PM |
0 |
Outbound calls through iaxy ATA not hearing ring + appending carrier PIN codes |
| 10:07AM |
5 |
retrieve last number dialled |
| 9:58AM |
3 |
Asterisk on multi-homed systems |
| 9:12AM |
0 |
1 FXS module / PCI express |
| 8:47AM |
2 |
DTMF not recognized on ISDN with Siemens -not IP- phone |
| 8:37AM |
2 |
cvs or svn |
| 6:50AM |
6 |
G729 on wrong bus |
| 4:27AM |
2 |
Billing/Call Control engine : AGI scripts/ AstMan API |
| 3:21AM |
0 |
Re :Recommendations for 100 Wifi SIP phone |
| 2:05AM |
3 |
Multiple Return Values from func_odbc |
| |
| Tuesday November 27 2007 |
| Time | Replies | Subject |
|---|
| 11:27PM |
1 |
Lost setting up IAXmodem after drive crash |
| 7:53PM |
0 |
Zaptel 1.2.22 and 1.4.7 released |
| 6:59PM |
1 |
Semi-OT Part 2: Videophone |
| 6:27PM |
5 |
Copy or Make + Make Install |
| 6:23PM |
0 |
ResetCDR Options v, a - Asterisk 1.4 |
| 6:02PM |
3 |
Urgent question. |
| 5:00PM |
1 |
Can Asterisk act like an ISP dialin server to data callls from Sipura 3000 or other ATA connected devices ? |
| 4:13PM |
2 |
Restricting the manager interface to a number? |
| 4:05PM |
4 |
Snom phones, blinking lights and call pickup |
| 3:33PM |
2 |
Finding the status of an extension |
| 3:31PM |
2 |
Attended transfer to Queue |
| 2:24PM |
1 |
Asterisk API Manager |
| 2:20PM |
5 |
SIP port 5060 closed - how do I open it? |
| 2:01PM |
3 |
Sip to ATA? |
| 1:10PM |
10 |
Asterisk behind a PIX firewall? |
| 6:34AM |
0 |
CDR issue need help |
| 4:51AM |
0 |
Dial application response code--help required |
| 4:10AM |
1 |
Voice mail & Uniden UIP-200 phones |
| |
| Monday November 26 2007 |
| Time | Replies | Subject |
|---|
| 9:56PM |
0 |
SIP Trunk Problems |
| 9:51PM |
3 |
Correct syntax for IF()? |
| 9:07PM |
1 |
Filesharing + video + voice supported Soft phone |
| 9:06PM |
1 |
VMukti - Filesharing + video + voice supported Soft phone |
| 8:58PM |
0 |
Digium b410p + mISDN echo |
| 8:08PM |
2 |
Broadcast dialing/playback |
| 7:33PM |
0 |
Interesting Conference Request - Can this be done ? |
| 7:30PM |
1 |
Semi-OT: Best Speakerphone |
| 5:41PM |
0 |
Queue with cell phones |
| 4:28PM |
2 |
Asterisk Recording |
| 3:25PM |
0 |
Remove a TDM Card |
| 3:22PM |
4 |
Digium E1 and Digium TDM400P (2xFXO) Help! |
| 3:06PM |
2 |
Possible Conflicts with Junghanns 4 Port BRI and 8 Port Sangoma Analog in Same Box? |
| 2:19PM |
3 |
Problems getting Asterisk to detect call in SUSE9.3, with FritzCard |
| 1:51PM |
1 |
OT: Best firmware for Linksys Router that is "SIP AWARE" |
| 1:28PM |
1 |
iptables requirements for SIP |
| 12:47PM |
0 |
Asterisk B2BUA patch useful?? |
| 10:14AM |
2 |
Asterisk version survey |
| 9:26AM |
0 |
How to manage several AMI connections to an Asteris server ? |
| 6:50AM |
1 |
Agi manager session. |
| 6:20AM |
2 |
Get IP address of an incoming or outgoing SIP call |
| 3:55AM |
1 |
passing DTMF upon call answering |
| |
| Sunday November 25 2007 |
| Time | Replies | Subject |
|---|
| 11:44PM |
1 |
[Record() function] Script stops if user doesn't hit # after msg |
| 9:49PM |
0 |
Recommendation for 100 SIP WiFi phone setup |
| 8:26PM |
4 |
Recommendations for 100 Wifi SIP phone setup |
| |
| Saturday November 24 2007 |
| Time | Replies | Subject |
|---|
| 8:48PM |
3 |
MSSQL ODBC Connections |
| 2:20PM |
1 |
dial in group |
| 10:33AM |
0 |
Voip Users Conference moves up to 12:00 EST |
| 1:17AM |
2 |
Annoying PRI Channels Restarting Message |
| 12:06AM |
3 |
Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial. |
| |
| Friday November 23 2007 |
| Time | Replies | Subject |
|---|
| 5:03PM |
1 |
Best Prepaid Application? |
| 4:29PM |
1 |
AMI Newstate Ringing events -- Inconsistent caller id ? |
| 4:17PM |
1 |
OT - 3Com and IBM iSeries |
| 3:03PM |
1 |
SIP detects loop when forwarding to voicemail |
| 10:33AM |
2 |
TDM808B 8 port FXO setting problem |
| 7:44AM |
0 |
Enough Turkey? Voip Users Conference today at 12:30 EST to help digest it all |
| 7:38AM |
2 |
How to bridge two connected calls |
| |
| Thursday November 22 2007 |
| Time | Replies | Subject |
|---|
| 11:12PM |
0 |
Work |
| 9:24PM |
1 |
NAT keep-alive |
| 9:02PM |
5 |
Odd bug in Siemens C460IP ? |
| 9:02PM |
4 |
Calling with hidden callerid |
| 5:34PM |
1 |
Dial problem |
| 2:56PM |
1 |
Toll fraud detection/password script |
| 1:48PM |
0 |
mailbox name length |
| 1:03PM |
6 |
Digium and Asterisk |
| 8:57AM |
4 |
Phones Not Registering |
| 7:28AM |
1 |
common/shared voicemail box |
| 5:34AM |
0 |
Asterisk support V5.2 protocal |
| 2:37AM |
0 |
Vicidial + Unicall mfcr2 |
| |
| Wednesday November 21 2007 |
| Time | Replies | Subject |
|---|
| 11:13PM |
3 |
spandsp as T.38 termination? |
| 10:31PM |
1 |
Problem dialing certain numbers with an E1 PRI |
| 7:24PM |
1 |
sip proxy failover |
| 6:52PM |
1 |
Caller ID Question |
| 6:47PM |
3 |
Aastra 480i CT - No Incoming Calls |
| 6:39PM |
1 |
Queue Drops to Voicemail |
| 3:50PM |
0 |
[DB] Insert only one prefix for multiple numbers? |
| 3:45PM |
1 |
Help Dial extention |
| 3:09PM |
1 |
Problem installing Asterisk |
| 2:33PM |
0 |
Need help in selecting DTMF Mode |
| 9:30AM |
0 |
chan_ss7 0.10.1 |
| 9:29AM |
5 |
Softphone to be installed on the Mobile |
| 7:47AM |
2 |
Zaptel 1.4 spec file |
| 2:42AM |
1 |
[1.4 - Record] How to tell if user did leave a msg? |
| 2:24AM |
1 |
quality after call transfer |
| 12:33AM |
1 |
Building an Asterisk 1.4 RPM |
| |
| Tuesday November 20 2007 |
| Time | Replies | Subject |
|---|
| 10:33PM |
1 |
Asterisk-Users: Termination |
| 9:52PM |
2 |
Music on Hold Problem w/ Transfers |
| 8:23PM |
1 |
How to receive manager events from commands made by an AGI script? |
| 7:46PM |
0 |
FXO incomming call hangup problem |
| 7:36PM |
0 |
Cisco phones and 32 directory object limit |
| 6:59PM |
1 |
Bugtracker to use with Asterisk? |
| 6:49PM |
0 |
automatic blind transfer calls |
| 6:35PM |
1 |
[asterisk-dev] trunk working under windows! |
| 6:27PM |
2 |
e911 |
| 6:17PM |
0 |
iaxpeers from Realtime |
| 6:01PM |
1 |
FXO Hangs up automatically |
| 5:33PM |
2 |
Reporting bugs |
| 3:16PM |
1 |
ACD functionality , Skills for agents |
| 2:51PM |
1 |
Realtime extensions configuration - calling user filtering |
| 2:43PM |
1 |
OT - What is Alarm receiver feature ? |
| 12:56PM |
1 |
store 2 separate records in cdr when a call is transferd |
| 11:19AM |
0 |
not sending bye |
| 11:05AM |
2 |
SMS Feature In Asterisk |
| 9:53AM |
0 |
sl75 wlan not able of being pickuped? |
| 9:42AM |
1 |
Realtime - mysql query gives wrong results?? |
| 9:21AM |
0 |
MediaHandling--Help Required |
| 8:07AM |
1 |
Problems with losing D-Channel on |
| 2:59AM |
1 |
Switch to Multi-Proc -> Choppy sound? |
| 2:59AM |
1 |
SIP - ooh323 Bridging |
| 2:58AM |
1 |
Interface with NEC NEAX 2400 |
| |
| Monday November 19 2007 |
| Time | Replies | Subject |
|---|
| 10:48PM |
1 |
asterisk manager and perl |
| 6:23PM |
3 |
dialplan design - unknown extension length |
| 6:17PM |
0 |
Natural Microsystems AG Quad |
| 5:27PM |
1 |
AstLinux WebSite Problem |
| 4:51PM |
7 |
asterisk as non-root/best practices |
| 2:14PM |
2 |
blind transfer dumping calls |
| 1:33PM |
3 |
How to enable res_config_mysql |
| 12:45PM |
3 |
Gigaset S450ip and simultaneous calls |
| 9:49AM |
5 |
Registration problem: UA -> SER -> Asterisk |
| 8:45AM |
2 |
Asterisk Sound File |
| 3:53AM |
4 |
Help: How to configure SIP domain on SPA942 |
| |
| Sunday November 18 2007 |
| Time | Replies | Subject |
|---|
| 9:14PM |
6 |
Asterisk on Pcengines Alix board |
| 4:20PM |
6 |
Conference Call Dial-Out to a participant |
| 1:58PM |
1 |
[IAX] Does the client have to use UDP4569 as source port? |
| 1:08PM |
0 |
facilityenable in zapata.conf |
| 8:44AM |
1 |
sip + jitter buffer |
| 4:06AM |
2 |
Trouble with asterisk-users mailman |
| 2:46AM |
2 |
problem with tdm2400p configuration |
| 12:47AM |
1 |
p2p t1 with sangoma hw |
| 12:02AM |
0 |
Connecting Ericsson 4422 or similar set to Asterisk ? |
| |
| Saturday November 17 2007 |
| Time | Replies | Subject |
|---|
| 10:53PM |
0 |
Polycom Provisioning Tool Source Code Released |
| 9:28PM |
1 |
Building and running mISDN for B410P on Ubuntu 7.04 |
| 9:28PM |
1 |
Multiple B410P's in one machine |
| 7:05PM |
1 |
Page Command |
| 3:19PM |
1 |
chan_ss7 0.10 |
| 2:41PM |
0 |
Astmanproxy Yahoo Group |
| 1:43PM |
0 |
Blackberry MVS and Asterisk. |
| 1:32PM |
1 |
California based PSTN connections |
| 10:03AM |
0 |
The call does not disconnect at the softphone when caller hangup the mobile |
| 12:38AM |
3 |
modifying a dialed exension before dialplan processing |
| |
| Friday November 16 2007 |
| Time | Replies | Subject |
|---|
| 11:51PM |
0 |
Asterisk 1.4.14 Released |
| 6:40PM |
1 |
Dumb AGI question |
| 3:58PM |
0 |
Polycom softkey transfer issue |
| 3:00PM |
1 |
Help with Polycom 320 |
| 1:28PM |
2 |
Changing audio message to text message |
| 1:06PM |
1 |
channels to destroy |
| 1:03PM |
2 |
Change the Voice promps in asterisk 1.4 |
| 8:31AM |
2 |
Which files to be copied |
| 3:32AM |
0 |
dtmf detection |
| 12:16AM |
1 |
Asterisk 1.4 with LDAP |
| |
| Thursday November 15 2007 |
| Time | Replies | Subject |
|---|
| 11:43PM |
1 |
Help on strange problem... |
| 11:31PM |
0 |
Building an Asterisk 1.4 RPM. |
| 6:51PM |
2 |
Two B-Channel Transfer (2BCT/TBCT) Trobule on DMS100 PRI |
| 6:32PM |
1 |
Music on Hold -- Error |
| 6:29PM |
1 |
OT - best policy for logs |
| 6:18PM |
1 |
Pass CallerID when call forwards to PSTN? |
| 5:47PM |
2 |
make config update-rc.d |
| 5:40PM |
0 |
SPA-2100 into Paging System "Hangs" |
| 5:33PM |
2 |
Asterisk Program Closes |
| 5:32PM |
1 |
Lists dead? |
| 4:49PM |
1 |
Centos 5 Asterisk 1.4 FreePBX Install script |
| 4:37PM |
2 |
Dialing time-out |
| 3:15PM |
1 |
Shared gsm files |
| 1:12PM |
2 |
reload command |
| 1:06PM |
1 |
TE210P Vs TE220P difference |
| 12:14PM |
1 |
DTMF Problem |
| 11:03AM |
1 |
Friday Conference reminder: AGI example Nov 16th at 12:30 PM EST |
| 9:51AM |
0 |
Problems with positions in callqueues |
| 8:38AM |
0 |
pipemedia |
| 6:41AM |
0 |
Queue |
| 3:23AM |
0 |
Integration of Asterisk with MS Dynamics CRM |
| 1:27AM |
1 |
asterisk integration with panasonic analog pbx |
| |
| Wednesday November 14 2007 |
| Time | Replies | Subject |
|---|
| 10:15PM |
3 |
asterisk-stat problem |
| 9:27PM |
1 |
pip tones in Monitor or MixMonitor |
| 8:24PM |
0 |
Real Time CDR |
| 7:40PM |
0 |
Routing Anonymous Callerid |
| 7:02PM |
0 |
Error: inserting return line in dialing strings |
| 6:32PM |
0 |
Help in getting a dialplan to produce the right CDR info |
| 6:02PM |
0 |
PBX Testing Framework |
| 3:47PM |
1 |
"Whats New at Digium the Asterisk Company" -- Junk? |
| 3:25PM |
0 |
IVR Tree Best Practices |
| 1:59PM |
4 |
Problem with AGI Script |
| 12:51PM |
1 |
Using php exec() in agi script |
| 12:24PM |
1 |
Linksys 942 Call Transfer |
| 10:54AM |
1 |
Asterisk ignoring manager events when busy |
| 6:31AM |
0 |
Asterisk trunk and manager redirect problem |
| 5:21AM |
4 |
What is wrong with this mailing list |
| 4:14AM |
0 |
asterisk-users Digest, Vol 40, Issue 37 |
| 3:58AM |
1 |
How to pay for libpri development |
| 3:52AM |
2 |
Nortel digital FXO channel bank? Exists? |
| 3:34AM |
6 |
function voicemailmain |
| |
| Tuesday November 13 2007 |
| Time | Replies | Subject |
|---|
| 8:51PM |
2 |
Call Forward on SIP unreachable (network failure) |
| 7:05PM |
0 |
Can I connect device on FXS of Sipura 3000 to internet virtually ? - it can only call ISPs numbers on POTS line |
| 6:14PM |
1 |
route INVITE sip:s@sip.test.fr |
| 3:11PM |
1 |
Install Scripts for CentOS 4 |
| 2:58PM |
2 |
How to integrate Asterisk with Avaya |
| 2:38PM |
1 |
Conference rooms |
| 2:26PM |
4 |
How to play Asterisk .raw file |
| 2:10PM |
1 |
[Fwd: Re: VoiceMail hangup] |
| 1:44PM |
4 |
Cisco 7911/7941/7970/7971 Softkey XML Files |
| 11:58AM |
1 |
Default mohmp3 : free of rights ? |
| 11:06AM |
0 |
Fwd: Re: Grandstream GXP2020 + Asterisk 1.4.11 |
| 9:49AM |
0 |
chan_alsa issue |
| 9:25AM |
0 |
Fwd: Re: Grandstream GXP2020 + Asterisk 1.4.11 |
| 8:26AM |
1 |
Toshiba DK - Asterisk Integration |
| 7:22AM |
3 |
Stress-Testing Asterisk |
| 5:07AM |
1 |
Chatterbug |
| 4:04AM |
1 |
MOH Codec Issue |
| 12:14AM |
1 |
ODBC connection to Microsoft SQL Server |
| |
| Monday November 12 2007 |
| Time | Replies | Subject |
|---|
| 7:28PM |
4 |
VoiceMail hangup |
| 6:54PM |
0 |
No sound from playback and voicemail (Atis Lezdins) |
| 6:47PM |
0 |
No sound from playback and voicemail (Carlos Chavez) |
| 2:46PM |
3 |
No sound from playback and voicemail |
| 1:13PM |
1 |
sip_chan - how to use value of the SIP 'To:' header field for extension logic |
| 1:03PM |
1 |
Internal CallerID problem |
| 12:16PM |
2 |
'h' extension on call-out |
| 11:12AM |
1 |
Grandstream GXP2020 + Asterisk 1.4.11 |
| |
| Sunday November 11 2007 |
| Time | Replies | Subject |
|---|
| 5:35PM |
1 |
IMAP Voicemail -- HELP! Asterisk not playing Greeting! |
| 10:38AM |
3 |
detect asterisk pbx via sip |
| 9:15AM |
3 |
sangoma zaptel patches |
| 5:51AM |
0 |
sangoma asterisk patches |
| 1:35AM |
1 |
RTP traffic not being forwarded |
| |
| Saturday November 10 2007 |
| Time | Replies | Subject |
|---|
| 11:34PM |
2 |
Record() : How to get filename created with %d? |
| 8:17PM |
5 |
r2 multiframe error |
| 7:26PM |
4 |
mpg123 on Thecus N2100 |
| 1:04PM |
2 |
sidetone |
| 12:34PM |
5 |
'Traditional' Faxing |
| 10:46AM |
1 |
Asterisk direct dialing |
| 8:13AM |
1 |
PHP - Queues - etc. |
| 8:12AM |
4 |
asterisk 1.4 prereq |
| |
| Friday November 9 2007 |
| Time | Replies | Subject |
|---|
| 10:57PM |
0 |
Copying the needed configuration files to be used on new installation |
| 9:14PM |
1 |
H323 registeration and routing the calls |
| 8:11PM |
1 |
Asterisk on Zonbu, impact of transcoding |
| 5:04PM |
1 |
asterisk ODBC dependencies |
| 4:50PM |
0 |
Weekly Conference Reminder: Friday Nov 9th 12:30 PM Eastern Time |
| 2:33PM |
0 |
svn+chan_mobile+Asterisk+Siemens FXO no voice |
| 12:07PM |
3 |
How to get ten-digit number? |
| 10:27AM |
0 |
What can I do with Jabber? |
| 7:55AM |
1 |
Your favorite desktop wifi sip hardphone ? |
| 5:39AM |
2 |
Kernel Native PCIE Network Cards? |
| 12:34AM |
4 |
If caller id is null set to a specific number |
| 12:11AM |
4 |
Wanted: tutorial on troubleshooting SIP issues |
| |
| Thursday November 8 2007 |
| Time | Replies | Subject |
|---|
| 11:39PM |
3 |
Asterisk as a SIP to XMPP Jingle voice gateway |
| 10:01PM |
0 |
AST-2007-024 - Fallacious security advisory spread on the Internet involving buffer overflow in Zaptel's sethdlc application |
| 9:37PM |
1 |
Switchvox Space Requirements |
| 8:01PM |
2 |
Asterisk and OBDC |
| 7:37PM |
3 |
Cisco IP Communicator with Asterisk |
| 3:56PM |
3 |
'a' extension |
| 3:46PM |
2 |
time on polycom 501 |
| 2:13PM |
1 |
Channel variables, any difference with SIP vs. IAX? |
| 12:13PM |
2 |
asterisk and installing chan_h323.so rpm |
| 12:10PM |
0 |
dtmfmode RFC2833 and inband |
| 10:50AM |
0 |
make h323 native transfer on stablished call |
| 9:22AM |
1 |
Snom 320 with TDM02B and echo problems |
| 7:27AM |
3 |
Client lost on skinny |
| 7:23AM |
0 |
Polycom IP601 call parking |
| 6:49AM |
0 |
Polycom IP601 (mac)-directory.xml changes don't update phone |
| 6:13AM |
2 |
weird 185 secs timeout call problem |
| 5:03AM |
1 |
extensions.conf pattern match info |
| 4:06AM |
0 |
DeStar finalist in Les Trophées du Libre 2007 |
| |
| Wednesday November 7 2007 |
| Time | Replies | Subject |
|---|
| 9:35PM |
3 |
ztdummy, zttest |
| 8:53PM |
0 |
Little OT: Compilation of EICON driver, fails with capi errors |
| 5:01PM |
0 |
Cisco phone 7911g restarts |
| 4:33PM |
0 |
accessing variables when using SIP vs. IAX |
| 3:31PM |
1 |
Polycom SoundStation VTX 1000 with Asterisk? |
| 2:26PM |
1 |
CDR on channel not posted |
| 1:26PM |
1 |
SIP: "To:" header? |
| 1:07PM |
0 |
Audiocodes over Sat link. and delay |
| 10:56AM |
1 |
Board configuration - specification or recommendation |
| 10:33AM |
5 |
What do you do to keep asterisk alive? |
| 9:29AM |
1 |
Call terminated with error message logged |
| 8:47AM |
2 |
Determination of billsec |
| 8:30AM |
1 |
grandstream troubles |
| 8:08AM |
2 |
OT: Aastra 57i configuration via TFTP problem |
| 7:58AM |
1 |
detecting voltage on fxo |
| 3:52AM |
2 |
wifi |
| |
| Tuesday November 6 2007 |
| Time | Replies | Subject |
|---|
| 10:52PM |
1 |
Extracting custom headers from SIP REFER |
| 10:12PM |
1 |
dtmf / misdn |
| 10:04PM |
2 |
Pickup Command not working |
| 8:34PM |
1 |
Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID |
| 8:29PM |
2 |
Selecting OSLEC for zaptel-1.4.6 |
| 7:25PM |
3 |
Asterisk Help |
| 7:09PM |
1 |
Sangoma S200 and Digium TDM400P together |
| 6:39PM |
1 |
Help: Asterisk info |
| 6:36PM |
1 |
Asterisk 1.4 + Presence |
| 5:30PM |
5 |
asterisk 1.4.10 on linux kernel 2.6 needs timing device for trunking |
| 4:09PM |
4 |
MeetMe CPU resources |
| 11:49AM |
2 |
Recording just first part of call? |
| 11:09AM |
1 |
Asterisk and Grandstream both behind different NAT |
| 10:37AM |
5 |
Linksys SPA-941 Unavailable |
| 2:35AM |
1 |
1.4 SIP Jitter Buffer |
| 2:10AM |
2 |
Asterisk & OpenVZ |
| |
| Monday November 5 2007 |
| Time | Replies | Subject |
|---|
| 11:03PM |
0 |
Two B410P cards in one machine |
| 10:55PM |
1 |
Please explain the correct LED color for B410P |
| 9:54PM |
2 |
Queue Statistics reporting |
| 9:06PM |
1 |
PRI dialout problem with some numbers... |
| 6:39PM |
1 |
Arbitrary limit on length of email address? |
| 6:27PM |
1 |
Help: Static and dropped calls |
| 6:16PM |
1 |
Testcall |
| 6:14PM |
2 |
Free T1 Card? |
| 5:40PM |
2 |
Problem with CDR userfield not being set |
| 5:38PM |
1 |
Meetme - how to protect the conference? |
| 3:34PM |
0 |
Parameters effect on the success registeration |
| 3:21PM |
1 |
OT: Which SIP method to use for this specific behaviour ? |
| 1:21PM |
2 |
Which Variable??? |
| 11:15AM |
2 |
How to delete voice mail messages? |
| 10:52AM |
0 |
How to disable Asterisk 407 Proxy Authentication Required Challenge response |
| 8:19AM |
1 |
Not Hearing hello-world Play |
| 8:10AM |
2 |
TE220 PCI express performnace |
| 6:47AM |
0 |
AGI connected to meetme conf gives Failed to write frame error |
| 6:42AM |
0 |
crash |
| 4:35AM |
2 |
Dynamic Queue Members - Auto Logoff |
| 3:42AM |
1 |
asterisk-users Digest, Vol 40, Issue 5 |
| 12:51AM |
1 |
Are the ATAs which can allow multiple extensions from one network connection? |
| |
| Sunday November 4 2007 |
| Time | Replies | Subject |
|---|
| 11:52PM |
2 |
Need Reference sites |
| 11:26PM |
3 |
7960 Queue Issue |
| 5:01PM |
1 |
Issues after upgrading from 1.2 to 1.4: hangup immediately |
| 1:10PM |
5 |
Restart when convenient |
| |
| Saturday November 3 2007 |
| Time | Replies | Subject |
|---|
| 8:43PM |
2 |
Asterisk versions and H323 |
| 8:05PM |
0 |
F3000 not connecting to Netgear router |
| 5:11PM |
0 |
F3000 not connecting to Netgear routing |
| 2:05PM |
0 |
Grandstream 488 and DTMF with a call file |
| 1:52PM |
1 |
Wifi handover/roaming |
| 1:37PM |
0 |
[Fwd: voicemail locked up Asterisk 1.4.13] |
| 1:10PM |
0 |
OT: Snom 300 losing config? |
| 7:50AM |
0 |
3pcc/click to dial accounting |
| 3:39AM |
1 |
Asterisk SIP Channels Bridge |
| |
| Friday November 2 2007 |
| Time | Replies | Subject |
|---|
| 8:04PM |
3 |
use dial plan passed arg value in C agi code |
| 8:01PM |
1 |
one way RTP using NAT |
| 7:46PM |
2 |
sip show peers in 1.4.13 |
| 6:18PM |
1 |
Off-Topic: add GSM codec to X-Lite |
| 5:51PM |
3 |
ztdummy and BackGround |
| 5:38PM |
0 |
steps for installing on Debian Etch |
| 4:44PM |
1 |
RealTime register for iax DID |
| 4:35PM |
1 |
SIP Channels |
| 4:32PM |
1 |
Jitterbuffer issues |
| 3:25PM |
1 |
Asterisk as a SIP to SIP Gateway |
| 2:47PM |
2 |
Route an incoming call by ANI*DNIS |
| 1:46PM |
1 |
res_mysql versus res_odbc |
| 11:37AM |
1 |
AEX800 (TDM800 Express) - not detected |
| 9:26AM |
2 |
asterisk as a gateway |
| 5:21AM |
2 |
zaptel.conf missing |
| 4:05AM |
3 |
__sip_xmit problem |
| 1:50AM |
1 |
Get value from linux terminal to dialplan in Asterisk ? |
| 12:31AM |
3 |
Two PRI setup questions |
| |
| Thursday November 1 2007 |
| Time | Replies | Subject |
|---|
| 11:56PM |
1 |
OpenSER for Asterisk Load balance |
| 11:39PM |
1 |
Polycom Park Button |
| 11:21PM |
1 |
ring group containing external 10-digit numbers |
| 10:41PM |
1 |
Help: How does one determine the length of an outbound/dialout MESSAGE to be delivered |
| 7:22PM |
0 |
Chanspy attaching to a caller ID entry? |
| 7:02PM |
1 |
AsteriskNOW and TDM800P |
| 6:32PM |
3 |
Outgoing PRI CID? |
| 5:43PM |
5 |
DST |
| 4:57PM |
0 |
libpri & tie line vs trunk |
| 4:00PM |
0 |
Parking speed |
| 3:42PM |
1 |
Help |
| 12:34PM |
5 |
SER/OpenSER as registrar to Asterisk (1500 SIP users) |
| 9:56AM |
3 |
Video Call |
| 9:11AM |
1 |
Autodialing |
| 8:05AM |
2 |
Spam Filter News |
| 8:01AM |
2 |
SER with Asterisk intergration |
| 1:03AM |
1 |
Call Failed |
| 12:57AM |
1 |
Connection astrisk to a RAS (portmaster) |
| 12:49AM |
2 |
hostname in MySQL CDR records |