| Wednesday June 30 2010 |
| Time | Replies | Subject |
|---|
| 11:32PM |
2 |
Pbx_lua vs. calling lua thru AGI? |
| 10:56PM |
2 |
Warning spamming for any unsynchronized ISDN port with dahdi-2.3.0.1 |
| 6:39PM |
2 |
Return agi script. |
| 3:28PM |
3 |
Problem with extensions in IVR and queues |
| 3:24PM |
3 |
Problem in establish call from a2billing users. |
| 2:26PM |
2 |
How to work Asterisk with Video Conference |
| 2:20PM |
1 |
queue command in asterisk 1.4 with macro-argument |
| 12:33PM |
4 |
Minimum modules required to run VoIP and CDR |
| 10:32AM |
1 |
Adding Congestion to CDR logs |
| 8:28AM |
2 |
Echo problem in VoIP-calls |
| 2:58AM |
1 |
RE How to break pri DID to multiple SIP Trunks |
| 12:23AM |
0 |
Long shot... Order Logix |
| |
| Tuesday June 29 2010 |
| Time | Replies | Subject |
|---|
| 11:54PM |
2 |
Anyone can share their config file for Cisco phone please? |
| 9:50PM |
1 |
Dial options not working |
| 9:23PM |
1 |
Can't call my extension |
| 9:14PM |
1 |
Carrier needs more call examples |
| 8:28PM |
1 |
Asterisk 1.6 (and 1.4) DTMF problems using RFC2833 |
| 8:24PM |
0 |
libpri 1.4.11.3 Now Available |
| 7:50PM |
0 |
Migrating from key system to asterisk |
| 6:36PM |
1 |
Strange Asterisk/SIP call forwarding behavior |
| 4:51PM |
1 |
Voiceprompts i.e. voicemail and conferencing in multiple codecs |
| 2:39PM |
3 |
Find a way to block brute force attacks. |
| 2:06PM |
3 |
SIP Delay with remote stations? |
| 1:53PM |
8 |
What TERMINAL software do you use for MS Windows platform and WHY? |
| 1:10PM |
0 |
OrderlyStats SE 1.6.2l now available. |
| 12:32PM |
3 |
cmd Authenticate |
| 12:12PM |
1 |
Problem with GoToIfTime |
| 12:02PM |
3 |
peer IP address in CDR |
| 9:19AM |
2 |
Is Centos 64 bit or 32 bit better? |
| 8:11AM |
1 |
Hot to configure trunk in asterisk with a2billing. |
| 7:23AM |
1 |
How to Add IP address to SIP Domain |
| 6:41AM |
1 |
transfering active call to user's voicemail |
| 6:04AM |
0 |
T.38 Peer Negotiation Fails |
| 5:15AM |
0 |
How to configure key sequence in features.conf. |
| 2:20AM |
1 |
What are the guts of AsteriskNOW and how it compares to other popular flavors available? |
| 2:04AM |
5 |
What‘s the best operating system suggest for Asterisk 1.6.2.9 |
| 12:40AM |
1 |
Update the LCD with the callee's name after dialing |
| |
| Monday June 28 2010 |
| Time | Replies | Subject |
|---|
| 11:58PM |
2 |
restricting sip users to a certain useragent |
| 8:27PM |
1 |
Problem with TE411P and DAHDI |
| 7:39PM |
1 |
Asterisk 1.6 and multiple parking |
| 6:49PM |
3 |
sip server |
| 4:00PM |
1 |
Handling DTMF for number 4 |
| 3:45PM |
1 |
Never seen Problem !!! |
| 2:00PM |
3 |
Pickup a ringing Queue member |
| 12:08PM |
2 |
sip add header |
| 9:15AM |
2 |
Problem attended transfer with ilbc |
| 7:16AM |
1 |
Use one group for ISN truncs |
| 7:07AM |
0 |
Use one ring-group for ISN truncs |
| |
| Sunday June 27 2010 |
| Time | Replies | Subject |
|---|
| 10:23PM |
0 |
CID |
| 4:58AM |
0 |
Vestec Tech Support |
| 2:09AM |
2 |
append CID label |
| |
| Saturday June 26 2010 |
| Time | Replies | Subject |
|---|
| 11:25PM |
1 |
Support from Vestec |
| 9:08PM |
2 |
Codec negotiation |
| 4:57PM |
0 |
Truth in advertising |
| 1:21PM |
2 |
[voice mail] Estimating file size? |
| 12:25PM |
1 |
Up-to-date list of Asterisk appliances? |
| 11:33AM |
2 |
Detecting hook flash in asterisk |
| 9:28AM |
1 |
Error - Failed to extend from xxx to xxx |
| |
| Friday June 25 2010 |
| Time | Replies | Subject |
|---|
| 11:07PM |
1 |
Non-native codecs - MELPe? |
| 7:29PM |
1 |
sip_xmit: sip_xmit returned -1: Operation not permitted |
| 5:03PM |
0 |
Meetme delay - normal? |
| 4:57PM |
2 |
Call drops on group paging asterisk - 1.4.22.1 |
| 3:23PM |
1 |
Configure WAN Phone |
| 11:25AM |
5 |
Is there a default dial plan that is not in extention.conf? |
| 7:48AM |
2 |
G729 license key registration |
| 6:59AM |
4 |
[CRON] Right way to restart Asterisk and Zaptel? |
| 3:24AM |
2 |
Big time system |
| |
| Thursday June 24 2010 |
| Time | Replies | Subject |
|---|
| 7:46PM |
4 |
OT: Bandwidth calculations |
| 7:32PM |
2 |
T.38 on a MAX/Lucent/Ascend TNT |
| 6:54PM |
1 |
SPA8000 outbound CID problem |
| 5:55PM |
4 |
Dialplan for conference |
| 5:53PM |
0 |
A lot of : doing dnsmgr_lookup for - Asterisk installed from YUM |
| 3:49PM |
2 |
dialplan reload 1.4.33 |
| 11:36AM |
2 |
Friday at 1PM: SIPVicious has a new tool: svcrash |
| 10:26AM |
1 |
Astersik can not detect DTMF key |
| 7:34AM |
3 |
Very strange registration problem |
| 5:52AM |
0 |
87.230.80.186 - Trying to register |
| 4:10AM |
0 |
parking on ast 1.6.2.8 |
| |
| Wednesday June 23 2010 |
| Time | Replies | Subject |
|---|
| 11:21PM |
0 |
50 mantis issues marked 'Ready for Testing' |
| 8:45PM |
2 |
"Hidden" memory leak |
| 4:08PM |
6 |
one for your filters |
| 3:05PM |
1 |
I look ARI (Asterisk Recording Interface) |
| 12:44PM |
2 |
help with sip 401 unauthorized |
| 12:04PM |
0 |
Hangup Detection Problem In Turkey |
| 11:20AM |
4 |
Need USA DIDs |
| 7:21AM |
0 |
CallWaiting |
| |
| Tuesday June 22 2010 |
| Time | Replies | Subject |
|---|
| 11:06PM |
0 |
Asterisk 1.4.33.1 Released |
| 9:55PM |
0 |
SMS in landline |
| 8:38PM |
1 |
Xorocom Missing files...where to get it? astribank_upgrade |
| 7:31PM |
3 |
joining 2 conferences together |
| 6:21PM |
0 |
Endless loop with asterisk directory |
| 6:21PM |
6 |
Asterisk distribution for a Call Center |
| 6:04PM |
2 |
Workaround for bug in Linksys Firmware 6.1.3(a) (or greater) |
| 5:34PM |
0 |
SkypeKit |
| 5:27PM |
1 |
Running SIP on non-standard ports |
| 5:24PM |
1 |
Internal timing bad for Fax? |
| 5:02PM |
1 |
Call file structure and syntax |
| 4:41PM |
1 |
Sangoma - how to show channels in use? |
| 4:03PM |
0 |
Video not working with PortSIP SDK |
| 1:30PM |
3 |
PRI span problem - no D channel |
| 12:01PM |
1 |
Unregister and register SIP phones by using num pad on phones? |
| 11:26AM |
4 |
Anybody using TE410P on BT ISDN with DAHDI? |
| 10:36AM |
1 |
UDPTL T38 via NAT |
| 9:07AM |
0 |
asterisk-users Digest, Vol 71, Issue 36 |
| 8:51AM |
1 |
storing DTMF inputs |
| 8:13AM |
4 |
Local channel usage |
| 8:09AM |
1 |
NO ANSWER before playback or background function? |
| 7:47AM |
0 |
Unable to set callerid for incoming skype calls |
| 6:11AM |
2 |
Generate cdr on Hangup |
| 4:58AM |
1 |
Can't make calls out of Astribank - show regdump doesn't show voltage - Getting warning due to old driver |
| 1:41AM |
0 |
Update to chan_ooh323 wrapper |
| |
| Monday June 21 2010 |
| Time | Replies | Subject |
|---|
| 9:38PM |
1 |
AMD |
| 9:09PM |
3 |
How do I access the Dialstatus numeric code received? |
| 7:04PM |
5 |
when to use e1/t1 card? |
| 5:39PM |
1 |
How to tell if a dropped call is my fault |
| 5:10PM |
3 |
Polycom firmware: split vs. combined |
| 5:04PM |
1 |
How to find a single call in logs |
| 4:25PM |
3 |
Create Conference and exit myself |
| 4:08PM |
1 |
DAHDI: Inbound BRI call, DDI not presented |
| 3:45PM |
2 |
What is the voicemail "u option" |
| 3:39PM |
0 |
Switchboad like application |
| 3:08PM |
1 |
Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels? |
| 12:54PM |
1 |
using call file |
| 12:48PM |
3 |
[AGI] What scripting language for embedded hardware? |
| 12:32PM |
1 |
ISP down internal phones become unavailable |
| 7:27AM |
3 |
features.conf - parkedcalls - transfer |
| 7:25AM |
0 |
call dialing |
| 4:06AM |
1 |
Asterisk 1.6 + Jabber crashes |
| |
| Sunday June 20 2010 |
| Time | Replies | Subject |
|---|
| 11:45PM |
1 |
Compiling H323 |
| 4:21PM |
0 |
Deleting some of the CDR data - How to do it safely? |
| 3:36PM |
0 |
load balance meetme |
| |
| Saturday June 19 2010 |
| Time | Replies | Subject |
|---|
| 3:47PM |
3 |
dahdi span |
| 2:58PM |
2 |
Using SetVar with System() is it possible? |
| 2:34PM |
0 |
OT - Explain RDNIS |
| 12:38PM |
0 |
Asterisk ODBC |
| 9:21AM |
2 |
Muti Asterisk |
| 9:05AM |
0 |
playing file when using call file in /var/spool/asterisk/outgoing in asterisk |
| 8:29AM |
2 |
Voicemail ODBC |
| 7:09AM |
2 |
asterisk appache issue |
| 5:19AM |
1 |
Can sip clients connect with each other directly (RTP session) ? |
| 1:58AM |
1 |
Linksys SPA94x keep-alive reply replies to wrong address (1.4.32) |
| 1:43AM |
2 |
dahdi modules installed wrong location |
| |
| Friday June 18 2010 |
| Time | Replies | Subject |
|---|
| 9:03PM |
1 |
Asterisk 1.6.2.9 Now Available |
| 9:03PM |
0 |
Asterisk 1.4.33 Now Available |
| 7:22PM |
1 |
How to get asterisk to playback personal greetings using grandstream gxp-2000 |
| 4:40PM |
6 |
Why asterisk down when inet server down? |
| 3:09PM |
0 |
Relaunch of the Kansas City Asterisk User Group |
| 1:54PM |
1 |
question on nortel sip connection |
| 12:30PM |
1 |
Error trying to add context: Context 'internal' tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*' |
| 12:24PM |
1 |
OT: Physical SIP phone with inbuilt VPN support |
| 10:38AM |
0 |
Friday June 18th at 12 Noon EDT: Session Border Controllers, 1PM Bria iPhone SIP app |
| 10:12AM |
6 |
asterisk issue |
| 8:51AM |
1 |
Automatic attendant - Error in CLI. |
| 8:36AM |
3 |
CDRs not getting generated on Free PBX |
| 8:21AM |
1 |
What's diff between ${CDR(start)} , ${CDR(answer)} , ${CDR(calldate)} |
| 7:20AM |
0 |
device or sound card busy |
| 6:54AM |
0 |
Weaknesses of Asterisk still there? |
| |
| Thursday June 17 2010 |
| Time | Replies | Subject |
|---|
| 8:26PM |
0 |
calls dropped after 20 seconds in a non NAT situation |
| 7:23PM |
3 |
Music on Hold problema |
| 5:21PM |
1 |
Asterisk SIP/IAX peers can't connect after Firewall change? |
| 4:59PM |
1 |
applicationmap and ChannelRedirect |
| 3:00PM |
0 |
app_swift v2.0 released |
| 2:58PM |
2 |
Slightly OT: Cisco SPA525G and network errors |
| 2:54PM |
1 |
IVR extension dialing error |
| 2:21PM |
1 |
Asterisk no audio on calls problem. |
| 1:37PM |
4 |
Check if variable contains strings |
| 12:24PM |
1 |
VAD and cRTP, any thing else? |
| 11:53AM |
1 |
DTMF detection issues |
| 9:06AM |
1 |
calling machine over sip |
| 8:46AM |
0 |
error message in CLI regarding SET Timeout |
| 7:18AM |
0 |
writing echo in inbound file |
| 1:38AM |
4 |
Asterisk + Dahdi does not work with BRI NT mode |
| |
| Wednesday June 16 2010 |
| Time | Replies | Subject |
|---|
| 9:54PM |
2 |
DAHDI PRI error message |
| 7:40PM |
0 |
Call hangs up after exactly 1 minute |
| 3:58PM |
2 |
read data from file system and put in a variable |
| 3:38PM |
0 |
H323 Trunk Problem calling from Asterisk to Avaya PBX |
| 3:31PM |
2 |
TDD/TTY Support |
| 3:25PM |
1 |
Blind transfer feature |
| 1:21PM |
2 |
ring no answer / RONA versus HangUp |
| 12:21PM |
4 |
Asterisk + E1 card |
| 8:28AM |
1 |
Problem with dahdi and with freepbx |
| 8:15AM |
0 |
Asterisk +Dahdi does not work with BRI NT |
| 6:21AM |
0 |
asterisk sip trunk configure |
| 2:07AM |
0 |
Fwd: [INSTALL #RKZ-745226]: Digium Support Survey, Partial Faxes |
| |
| Tuesday June 15 2010 |
| Time | Replies | Subject |
|---|
| 10:00PM |
1 |
numbers |
| 9:21PM |
0 |
Extract user part from SIP URI |
| 8:36PM |
1 |
Voicemail vm-intro played even when temp greeting is setup |
| 6:29PM |
1 |
Asterisk hangs up for some calls |
| 4:09PM |
4 |
Unable to pickup an extension, tryi |
| 2:50PM |
1 |
Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com' |
| 2:24PM |
0 |
Asterisk reject SIP INTITE from different |
| 1:57PM |
1 |
Corba interface |
| 1:22PM |
4 |
Cutting the CallerID(RDNIS) |
| 11:43AM |
3 |
Asterisk reject SIP INTITE from different source ports |
| 8:07AM |
4 |
can't seem to register, status unmonitored |
| 5:22AM |
1 |
Skype for SIP |
| 5:17AM |
1 |
Skype for Asterisk - what processors/platforms does it run on? |
| 1:31AM |
2 |
a2billing for residential voip usage |
| |
| Monday June 14 2010 |
| Time | Replies | Subject |
|---|
| 10:02PM |
1 |
Configure Voicemail for Large Systems |
| 9:59PM |
0 |
Small PC for Asterisk appliance to support 2 x Sangoma A200 (2 x PCIe standard cards) |
| 4:00PM |
2 |
How to pass variable back and forth from dialplan to php file? |
| 3:47PM |
2 |
How to disable day light saving on Snom 360 phones? |
| 3:45PM |
4 |
Unable to pickup an extension, trying everything |
| 3:22PM |
0 |
Hint priority in RealTime |
| 2:50PM |
0 |
Multiple parking lots - 1.6 |
| 12:19PM |
2 |
calling peer from server |
| 11:41AM |
1 |
Call queues - issues, can't make it work. |
| 10:04AM |
1 |
logging stopped suddenly |
| 7:47AM |
0 |
debug message: Internal timing is disabled |
| 7:26AM |
6 |
Small PC to build and run Asterisk |
| 7:10AM |
1 |
Issues running Asterisk + Iaxmodem + Hylafax on same machine |
| 6:35AM |
1 |
PSTN call hunting |
| 1:19AM |
0 |
No 2nd invite from asterisk after challenging original invite |
| |
| Sunday June 13 2010 |
| Time | Replies | Subject |
|---|
| 6:59PM |
1 |
AGI library for C/C++ |
| 6:23PM |
0 |
Asterisk AMI |
| 10:35AM |
2 |
bug with Moh on MeetMe ? |
| |
| Saturday June 12 2010 |
| Time | Replies | Subject |
|---|
| 3:30PM |
1 |
MeetMe problem |
| 7:17AM |
2 |
Qwest PRIs |
| |
| Friday June 11 2010 |
| Time | Replies | Subject |
|---|
| 10:15PM |
0 |
Using 5th gen TE420 with Asterisk 1.2? |
| 9:55PM |
7 |
How to stop intruder from registering sip? |
| 8:32PM |
1 |
OT: Free DID/SIP accounts |
| 8:11PM |
2 |
Call ended after 31 seconds |
| 6:35PM |
1 |
WARNING message when play |
| 4:43PM |
1 |
contacting |
| 4:22PM |
0 |
Asterisk SIP realtime and realtime DB tools |
| 4:08PM |
2 |
asterisk log problem |
| 2:39PM |
1 |
MeetMe |
| 2:31PM |
3 |
no ring back 180 with SDP |
| 1:49PM |
0 |
ZA16E and FXO-200 modules with asterisk |
| 10:40AM |
1 |
HDLC Bad FCS (8) on Primary D-channel |
| 10:10AM |
0 |
Pri show span and PtMP mode |
| 8:22AM |
1 |
chan_dahdi compilation with embedded |
| 12:19AM |
4 |
Dual Atom mobo - call capacity |
| |
| Thursday June 10 2010 |
| Time | Replies | Subject |
|---|
| 9:05PM |
0 |
Eyebeam hangs when you dial an unavailable number |
| 7:24PM |
2 |
Priority between calls in different queues |
| 6:59PM |
2 |
ISDN -> SIP |
| 4:49PM |
2 |
tuning software echo cancellation |
| 3:10PM |
1 |
Am I having problems with codecs? or am I not receiving an invite at all from my DID provider? |
| 1:46PM |
1 |
understand which asterisk thread is consuming CPU |
| 1:02PM |
3 |
Ring + Music on Hold in the same call |
| 11:52AM |
1 |
warning : sip_xmit |
| 10:56AM |
1 |
Group call limit |
| 10:32AM |
0 |
Loud Noise when trying to call through PSTN. |
| 8:38AM |
1 |
asterisk registration |
| 7:05AM |
0 |
Dial with MOH |
| 6:49AM |
0 |
How to kick/mute using ConfBridge application |
| 5:15AM |
1 |
CDR in case of CallForwarding |
| |
| Wednesday June 9 2010 |
| Time | Replies | Subject |
|---|
| 9:28PM |
1 |
[compat] section in asterisk.conf : compatibility with pipe delimiter |
| 6:41PM |
0 |
1.6 how to use groupcount and counteronpeer in queues to avoid ringinuse |
| 1:49PM |
2 |
SIP Witch |
| 1:30PM |
1 |
OT - Astmanproxy download broken ? |
| 12:19PM |
1 |
get Asterisk version from within dialplan |
| 12:19PM |
2 |
PSTN-IVR call |
| 9:19AM |
0 |
AMI Queue information about incoming call's channel before link |
| 1:50AM |
0 |
CID name in Facility message for Q.SIG |
| |
| Tuesday June 8 2010 |
| Time | Replies | Subject |
|---|
| 11:11PM |
1 |
early media issue from phone co. |
| 9:06PM |
0 |
NMI received for unknown reason |
| 8:25PM |
0 |
(no subject) |
| 7:33PM |
5 |
own Caller ID |
| 7:30PM |
0 |
memory leak |
| 6:20PM |
1 |
Asterisk-Addons 1.6.0.6 and 1.6.1.4 Now Available |
| 5:48PM |
0 |
libpri 1.4.11.2 Now Available |
| 3:41PM |
1 |
LumenVox *.gram reload |
| 3:40PM |
6 |
reloading realtime sip peers |
| 1:50PM |
3 |
Deleting extension makes it usable? |
| 1:24PM |
0 |
Problem with iax2/rsa registration |
| 1:09PM |
6 |
Out of Office |
| 12:52PM |
3 |
Limit total length of calls to a specifig SIP peer |
| 10:38AM |
0 |
Unavailable issue with SIP realtime and app_queue (*-1.4) |
| 3:44AM |
1 |
Issues with Vestec ASR |
| |
| Monday June 7 2010 |
| Time | Replies | Subject |
|---|
| 9:10PM |
0 |
Announcement before absolute timeout / how to terminate a meetme conf? |
| 5:27PM |
1 |
IAXmodem in dialplan |
| 3:22PM |
0 |
Still no(isy) app_jack in the box |
| 8:33AM |
1 |
How to play Floating point numbers? |
| |
| Sunday June 6 2010 |
| Time | Replies | Subject |
|---|
| 6:48PM |
1 |
Error of FreePBX after installing from Yum Repository of Asterisk |
| 6:46PM |
1 |
problem with port 5090 registration |
| 4:27PM |
1 |
Assign dhadi channel to several groups |
| 1:33AM |
0 |
Strange problem with zap channel. |
| |
| Saturday June 5 2010 |
| Time | Replies | Subject |
|---|
| 11:41PM |
5 |
Controlling calls |
| 8:16PM |
5 |
Still sipping frustration - only getting state ACK |
| 4:41PM |
0 |
Queue with PopUP screen for customer |
| 4:26PM |
1 |
Can one adjust the voicemail-menu when using VoiceMailMain() ? |
| 4:23PM |
0 |
dsp.c: digit_state.current_len |
| 2:13PM |
1 |
Problem with GROUP() |
| |
| Friday June 4 2010 |
| Time | Replies | Subject |
|---|
| 11:54PM |
1 |
originating a sip call from the CLI |
| 9:58PM |
2 |
Press twice * |
| 8:38PM |
2 |
1.6 issues |
| 4:09PM |
1 |
Using Local in queues a good idea? (or at least not a very bad idea?) |
| 12:40PM |
4 |
Asterisk on Ubuntu |
| 10:58AM |
0 |
BerkeleyTIP Join June Global Free SW HW Culture Mtgs via VOIP or in Berkeley |
| 8:52AM |
2 |
Create dialplan restrictions based on the IP Address of the SIP Client? |
| 2:28AM |
1 |
Wierd error when compiling 1.6.2 branch from SVN |
| |
| Thursday June 3 2010 |
| Time | Replies | Subject |
|---|
| 11:00PM |
0 |
Small VoIP company looking for Asterisk Scalability and Maintenance Engineer |
| 9:04PM |
1 |
other codecs |
| 7:56PM |
2 |
problem with inserting records into cdr |
| 7:52PM |
1 |
11.6.2 segfaults after dtmf on dahdi channel |
| 6:40PM |
0 |
SIP: match_auth_username=yes doesn't seem to work |
| 6:07PM |
1 |
<UsingWaitorPlaybackinhextension@gmail.com>, |
| 5:32PM |
0 |
OT: Cisco ATA 186 |
| 4:09PM |
1 |
Codec G.129 A vs A/B |
| 1:24PM |
5 |
how to get call duration |
| 1:16PM |
5 |
Is this failed Asterisk setup typical? |
| 11:12AM |
0 |
how to run deadagi script after "status: expired" |
| |
| Wednesday June 2 2010 |
| Time | Replies | Subject |
|---|
| 11:37PM |
0 |
SIP message problems - retransmit and lost messages |
| 9:35PM |
0 |
libpri 1.4.11.1 Now Available |
| 9:35PM |
4 |
DAHDI volume |
| 6:57PM |
1 |
Persuing the gtalk issue - not only jack-related |
| 5:03PM |
1 |
timeout problem with basic conf |
| 3:52PM |
0 |
sipconnect 1.0 |
| 11:03AM |
6 |
How do you hangup a call without terminating your session? |
| 5:57AM |
11 |
HElP me I am a beginner |
| |
| Tuesday June 1 2010 |
| Time | Replies | Subject |
|---|
| 9:00PM |
1 |
Definite app_jack trouble - unsolvable |
| 8:51PM |
2 |
Voicemail bug(?) with Asterisk 1.6.2.8-rc1 |
| 8:38PM |
0 |
Asterisk 1.4.32 Now Available |
| 8:36PM |
2 |
Asterisk 1.6.2.8 Now Available |
| 7:19PM |
5 |
no sound between extensions |
| 5:51PM |
0 |
Caller id, sip header from problem |
| 4:21PM |
1 |
Asterisk and gtalk part 2 |
| 2:27PM |
4 |
Slightly OT: trying to mangle packets from Asterisk for a multiple ISP setup (reward) |
| 1:42PM |
0 |
Silence suppression and internal timing |
| 10:37AM |
0 |
Getting ANI on UK BT ISDN - Is SS& required? |