| Wednesday March 31 2010 |
| Time | Replies | Subject |
|---|
| 9:29PM |
2 |
Necessary hardware |
| 8:29PM |
2 |
How to run Music while looking for the caller in Database |
| 7:56PM |
0 |
Upcoming Asterisk 1.6.0 and 1.6.1 Maintenance Changes |
| 6:57PM |
2 |
Multicast Paging |
| 6:00PM |
0 |
meetme() and dahdi_dummy on an embedded system |
| 3:57PM |
2 |
Reset personal voicemail settings |
| 1:09PM |
1 |
Unable to login to voicemail with Ekiga |
| 12:06PM |
1 |
Jitter Buffer and MeetMe. |
| 1:05AM |
2 |
Asterisk hangup all outging calls after 32 seconds |
| 12:27AM |
0 |
app_txfax.c |
| |
| Tuesday March 30 2010 |
| Time | Replies | Subject |
|---|
| 10:09PM |
0 |
E1 card w/o echo cancellation |
| 8:46PM |
0 |
E-mails from Asterisk coming from root |
| 8:16PM |
2 |
convert from wav or mp3 to gsm |
| 7:54PM |
2 |
Dropped Calls |
| 6:08PM |
1 |
DAHDI 2.2.1, Asterisk 1.6.2.6 - Channel unacceptable (6) |
| 6:02PM |
2 |
Priority based softhangup |
| 5:38PM |
5 |
Confusion on call forwarding |
| 10:18AM |
0 |
Asterisk realtime ldap:active directory |
| 7:10AM |
0 |
How can install and use Async AGI |
| 6:17AM |
0 |
Inbound configuration |
| 2:08AM |
1 |
a2billing wont pass the number |
| 1:57AM |
0 |
Diameter for Asterisk, Traffix Diameter stack ? |
| 12:30AM |
1 |
How are your PRI interrupts balanced? (+ Soft lockup BUG) |
| 12:29AM |
0 |
Asterisk and Call files |
| |
| Monday March 29 2010 |
| Time | Replies | Subject |
|---|
| 10:42PM |
1 |
Trying to get reason for ending of AGI call recording |
| 9:10PM |
0 |
amr |
| 8:46PM |
3 |
Asterisk system for church call center |
| 7:03PM |
1 |
Asterisk, IAX, & Sub interfaces |
| 5:32PM |
0 |
No audio when calling via PSTN, before remote answers (with polarity reversal) |
| 4:08PM |
3 |
Foip solution |
| 1:42PM |
1 |
Realtime Issue |
| 1:03PM |
3 |
Slightly more advanced dialling.. |
| 10:18AM |
0 |
queue autopause status |
| 9:33AM |
0 |
MixMonitor and StopMixMonitor |
| 9:13AM |
5 |
Continue a dialplan when the client hang up the call |
| 7:34AM |
1 |
is it possible to connect Digium TE420 and Cisco card? |
| 12:55AM |
9 |
24 FXS Port Voip Gateway and Asterisk |
| |
| Sunday March 28 2010 |
| Time | Replies | Subject |
|---|
| 3:19PM |
1 |
Updating Asterisk and its use with MySQL |
| 3:13PM |
1 |
Libtonezone |
| |
| Saturday March 27 2010 |
| Time | Replies | Subject |
|---|
| 9:48PM |
3 |
Trying to configure xorcom on Suse 11 |
| 8:33PM |
1 |
migration |
| 3:17PM |
4 |
Cisco 7960 become UNREACHABLE behind pix firewall |
| |
| Friday March 26 2010 |
| Time | Replies | Subject |
|---|
| 9:41PM |
2 |
dnd not working correctly |
| 8:22PM |
2 |
What does this error message mean |
| 6:14PM |
0 |
Re :Re: Sip module and dns (Alyed) |
| 5:58PM |
0 |
Re :Re: Sip module and dns (Alyed) |
| 2:01PM |
1 |
no voicemail on pstn line |
| 1:18PM |
1 |
problem with polarity reverse |
| 10:29AM |
2 |
Is there any Diguim distributor in Lahore |
| 10:10AM |
2 |
need help on setup rtp directly between 2 sip clients |
| 9:13AM |
0 |
Delay on sip channel |
| 8:36AM |
7 |
Asterisk load balancing and failover |
| 6:39AM |
1 |
[VUC] Voipathon 24-hour online party begins in 30 mintes |
| 6:22AM |
1 |
SIP/2.0 403 Forbidden |
| 3:55AM |
1 |
"Failed to play transfer sound! " during attended transfer |
| 2:46AM |
0 |
Time counting while playback |
| 2:37AM |
1 |
send a call from A to B use sip trunk prablem |
| |
| Thursday March 25 2010 |
| Time | Replies | Subject |
|---|
| 6:17PM |
4 |
Background noise |
| 4:48PM |
0 |
call not routed |
| 3:00PM |
1 |
Static linking |
| 1:07PM |
0 |
intergration of Diameter |
| 12:14PM |
2 |
rtp.conf ports for inbound or outbound? |
| 9:58AM |
1 |
configure the sound for inbound calls |
| 9:26AM |
2 |
Attended transfer and callerID updates forSiemens Openstage phones |
| 3:29AM |
3 |
How to get Sip response codes in Dialplan? |
| 3:26AM |
0 |
Music class default requested but no musiconhold loaded |
| 12:42AM |
9 |
Maximum number of PRI calls on 1 asterisk box (no HW echo) |
| |
| Wednesday March 24 2010 |
| Time | Replies | Subject |
|---|
| 11:07PM |
1 |
Aastra weirds IP 169.x.x.x |
| 9:56PM |
2 |
new server install errors starting asterisk |
| 7:42PM |
0 |
Dahdi-linux & Dahdi-tools 2.2.1.1 Release Announcement |
| 7:37PM |
1 |
This is a test, hijack this |
| 7:18PM |
1 |
installing dahdi card |
| 6:42PM |
5 |
Asterisk 1.6 and OpenVPN RTP problem |
| 5:51PM |
0 |
AstLinux 0.7.1 released |
| 4:26PM |
1 |
software version (lets try it again) |
| 4:26PM |
0 |
chan_h323 and ToS |
| 4:02PM |
6 |
Restarting Asterisk using a script - Thanks to all - |
| 3:28PM |
2 |
software version |
| 10:33AM |
1 |
Firewall & audio : need a wide range to work ! |
| 9:32AM |
0 |
Hook playback or ControlPlayBack cmd |
| 8:48AM |
1 |
G.729 Codec problem. |
| 4:29AM |
1 |
pstn calls not picked up |
| 4:13AM |
1 |
Mobile phone shut down, but Queue() Ring as usual |
| 2:06AM |
3 |
AMD reporting NOTSURE most of the time |
| 1:33AM |
0 |
Asterisk 1.6.1.12 with Grandstream HT502 T38 Fax |
| |
| Tuesday March 23 2010 |
| Time | Replies | Subject |
|---|
| 11:16PM |
4 |
Safe_asterisk doesn't exists??? |
| 8:40PM |
1 |
permit/deny in sip.conf iax.conf |
| 7:41PM |
5 |
G.711a or G.711u ??? |
| 5:31PM |
2 |
Sip module and dns |
| 5:06PM |
0 |
Strange Meetme disconnects |
| 4:59PM |
0 |
In Berlin this week? Kamailio/Asterisk community dinner on Thursday |
| 4:45PM |
0 |
Classic NO AUDIO problem - DD-WRT and NAT forwarding - HELP PLEASE! |
| 4:21PM |
1 |
Minimalize jitter in VoIP calls |
| 3:07PM |
0 |
distribuited ACD on many asterisk nodes |
| 2:16PM |
1 |
Asterisk crash - segmentation fault |
| 12:57PM |
5 |
Install dahdi on Xen virtual console |
| 8:53AM |
0 |
[asterisk-ss7]Chan_ss7 issue |
| 8:50AM |
1 |
chan_ss7 issue |
| 3:14AM |
0 |
(no subject) |
| 2:43AM |
3 |
How to make upgrades with Asterisk |
| 1:38AM |
3 |
Which folder for sounds? |
| |
| Monday March 22 2010 |
| Time | Replies | Subject |
|---|
| 11:40PM |
3 |
Can I call myself on the same machine |
| 8:57PM |
1 |
Play music to caller after answer, before dial |
| 5:33PM |
2 |
Transcoding question |
| 5:26PM |
2 |
requirecalltoken & receiving IAX calls |
| 2:31PM |
0 |
DUNDi Confusion |
| 1:56PM |
2 |
Context vs. Custom Context |
| 1:48PM |
1 |
Call files : call multiple SIP-accounts |
| 11:36AM |
2 |
voicemail problem |
| 11:25AM |
1 |
PRI lines do not have CallerID activated yet it is |
| |
| Sunday March 21 2010 |
| Time | Replies | Subject |
|---|
| 8:57PM |
1 |
Invalid Makefiles to install asterisk with ldap |
| 8:27PM |
1 |
Asterisk Manager Interface (AMI) proxy recommendation |
| 6:23PM |
1 |
Asterisk Died - Ver-1.6.2.6. |
| 1:35PM |
0 |
dahdi_monitor doesn't show data on RX & TX: broken card or cable? |
| 1:06PM |
1 |
test |
| 12:52PM |
1 |
How to get Asterisk to make batch calls? |
| 12:30PM |
6 |
Do i really need Dahdi and Libpri. |
| 12:58AM |
1 |
Early audio problem in chan_dahdi |
| |
| Saturday March 20 2010 |
| Time | Replies | Subject |
|---|
| 11:52PM |
1 |
Voicemail, Asterisk and Grandstream BT200 |
| 9:35PM |
1 |
1.6.1.18 -> 1.6.2.6 T38 Fax: call drops |
| 5:20PM |
1 |
how to start callerid for india |
| 2:34PM |
1 |
SIP signal through one IP and media through different IPs |
| 2:31PM |
3 |
Asterisk general Timeout for digits |
| 1:08PM |
2 |
8Port Junghanns BRI card under Dahdi |
| 10:24AM |
1 |
basic pc to pc voip in lan |
| 10:06AM |
0 |
Elastix 1.6 continuos ring |
| |
| Friday March 19 2010 |
| Time | Replies | Subject |
|---|
| 8:11PM |
4 |
Call Drops while doing assisted transfer from remote location |
| 4:12PM |
0 |
Setting Caller ID for attended transfer |
| 3:37PM |
2 |
register => 2345:password@sip_proxy/1234 |
| 2:50PM |
0 |
SPA3102 + asterisk drop call and loop (was SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?) ) |
| 9:52AM |
1 |
too much sockets open by asterisk |
| 8:36AM |
1 |
Strange initial RING |
| 6:50AM |
0 |
rtp connection remained when call busy using agi for call control |
| 6:11AM |
1 |
how to configure caller id |
| 3:20AM |
2 |
confbridge not working? |
| 2:05AM |
1 |
Define an array of sip number in sip.conf |
| 12:19AM |
6 |
(no subject) |
| |
| Thursday March 18 2010 |
| Time | Replies | Subject |
|---|
| 9:27PM |
3 |
Free Daily Asterisk News iPhone and iPod Touch app |
| 7:22PM |
1 |
SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?) |
| 4:49PM |
0 |
Problem with forwarding: Now forwarding SIP/ XX to Local/ |
| 3:04PM |
0 |
SIP Router Project |
| 2:02PM |
6 |
Asterisk DIES with no trace. PLEASE |
| 1:39PM |
0 |
Software for my laptop to send Fax via H.323 ? |
| 12:18PM |
1 |
Voicemail Remote Access |
| 9:21AM |
2 |
Live Audio Streaming- From Aux interface-Online resource |
| 8:31AM |
2 |
How to detect a PSTN telephone is busy or not? |
| 4:29AM |
0 |
Asterisk and OOo Smart Tags |
| 12:49AM |
2 |
Wanted: free DID number and provider feedback |
| |
| Wednesday March 17 2010 |
| Time | Replies | Subject |
|---|
| 11:39PM |
2 |
DID number |
| 11:09PM |
0 |
Need help with auto-forwarding virtual extensions (Asterisk 1.4/GUI 2.0) |
| 8:08PM |
7 |
Asterisk DIES with no trace. PLEASE HELP! |
| 5:41PM |
1 |
BT ISDN-30 Call Failures |
| 4:29PM |
0 |
monitor SIP jitter buffer |
| 4:08PM |
1 |
Adding an external dial code |
| 3:37PM |
3 |
SIP codec negotiation / manipulation |
| 3:35PM |
2 |
Call Filtering |
| 2:16PM |
2 |
Asterisk running on a Xen Centos Server challenge!!! |
| 1:13PM |
2 |
Asterisk as a skinny/sccp "client"? |
| 11:43AM |
2 |
Asterisk 1.6.0.5 and app_system FAILED using TRYSYSTEM |
| 9:40AM |
3 |
asterisk fax handeling |
| 5:48AM |
2 |
sip send image |
| 3:13AM |
0 |
[NEWBIE] Simple hunt group on SIP -- need confirmation |
| |
| Tuesday March 16 2010 |
| Time | Replies | Subject |
|---|
| 8:02PM |
1 |
Outbound route prefixes |
| 7:55PM |
1 |
Asterisk hangup all incoming calls after 10 seconds |
| 7:39PM |
1 |
softhangup |
| 2:02PM |
0 |
Asterisk to be used with Ciscs media gateways |
| 1:59PM |
1 |
Asterisk + Sip Phone + BLF |
| 1:29PM |
1 |
different authentication requirement |
| 11:59AM |
2 |
DID/CID doesn't match "." (dot) in CID field |
| 2:01AM |
3 |
Asterisk 1.4.24 DUNDi CLI commands not found |
| |
| Monday March 15 2010 |
| Time | Replies | Subject |
|---|
| 9:53PM |
1 |
Article - a method on how to evaluate an Asterisk server |
| 9:15PM |
0 |
How to find Asterisk compile time options for building app_swift module |
| 7:40PM |
1 |
Installing cdr_pgsql on asterisk 1.6.0.26 |
| 5:44PM |
1 |
dnd |
| 2:25PM |
0 |
asterisk-users Digest, Vol 68, Issue 33 |
| 1:15PM |
2 |
Android Phones ;-) |
| 12:32PM |
1 |
AEL in 1.6 and Gosub |
| 8:25AM |
1 |
CDR: Add Dialed Number Identifierfield (DNID) field into MySQL |
| 5:27AM |
2 |
High Availability Asterisk PBX |
| 5:20AM |
3 |
USING ASTERISK AS AVAYA DEFINITY RECORDING SERVER |
| |
| Sunday March 14 2010 |
| Time | Replies | Subject |
|---|
| 10:56PM |
0 |
Change SIP Release Code |
| 10:23PM |
1 |
queue MOH |
| 10:10PM |
2 |
Help with playing a recorded message in a conference. |
| 5:38PM |
0 |
Strange audio problem with Digium Wildcard B410 |
| 5:32PM |
2 |
dahdi-linux-complete-2.2.1+2.2.1 failed to compile |
| 2:30PM |
1 |
Debugging log rotation problem |
| 2:26PM |
0 |
ooh323_indicate: Don't know how to indicate condition 20 |
| 1:38PM |
0 |
DECT phone wont stop ringing |
| |
| Saturday March 13 2010 |
| Time | Replies | Subject |
|---|
| 9:40PM |
2 |
DID forwarding ? |
| 7:08PM |
0 |
How to test my Dial(SIP/...) ? |
| 6:40PM |
0 |
func_devstate with latest 1.4... |
| 4:33PM |
1 |
IAX2 peer question |
| 3:14PM |
0 |
SIP debug on a per call base |
| 2:40PM |
2 |
Asterisk on MPLS VPN |
| 1:03PM |
1 |
adding agent with 2 phones to a queue |
| 9:14AM |
0 |
PBX_DUNDI question |
| 6:00AM |
0 |
DUNDILOOKUP doesn't return record |
| 4:00AM |
0 |
Skype for Asterisk and regular expressions |
| |
| Friday March 12 2010 |
| Time | Replies | Subject |
|---|
| 10:36PM |
1 |
Setting up RTP to flow between endpoints directly bypassing Asterisk |
| 8:30PM |
0 |
Asterisk 1.6.2.6 Now Available |
| 8:29PM |
0 |
Asterisk 1.6.1.18 Now Available |
| 8:24PM |
0 |
Asterisk 1.6.0.26 Now Available |
| 8:23PM |
0 |
Asterisk 1.4.30 Now Available |
| 7:19PM |
1 |
Asterisk 1.6.2.5 x64 with Skype and DTMF on skype-out. |
| 6:23PM |
0 |
Installing chan_H323 by yum? |
| 6:21PM |
0 |
modem config & pots & documentation |
| 6:08PM |
2 |
Polycom not updating the directory list |
| 4:00PM |
1 |
t38 ATA |
| 3:31PM |
2 |
ExtenSpy Problem |
| 2:54PM |
0 |
Regarding - P-Asserted identity and Privacy - SOLVED |
| 1:17PM |
0 |
Asterisk 1.2 crash: gdb trace on core dump |
| 12:17PM |
1 |
1.2 to 1.6 and bristuff |
| 9:19AM |
4 |
Can not enable sip debug because CLI flooded |
| 8:03AM |
3 |
Time counting down and # detect |
| 7:59AM |
0 |
Fri March 12th @ 12 noon EST: SIP scanning, security and attacks + Hosted vs on-site voip |
| 2:57AM |
0 |
Running DEADAGI from h extension |
| 12:07AM |
2 |
Fwd: Switchvox SOHO 4.5 is Here |
| |
| Thursday March 11 2010 |
| Time | Replies | Subject |
|---|
| 7:54PM |
1 |
Digium TE4xx T1 Bonding |
| 5:09PM |
2 |
Codec preference |
| 1:44PM |
1 |
Testers Need Issue #0016965: [patch] DBGet response does not end with a 'Complete' event |
| 10:33AM |
2 |
Is there a way for a peer to clear its registration from a server? |
| 6:16AM |
2 |
How to add custom CDR fields to MySQL |
| 2:27AM |
2 |
Phones won't stop ringing |
| 2:19AM |
0 |
Unable to forward voice or dtmf |
| 2:07AM |
2 |
press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license |
| |
| Wednesday March 10 2010 |
| Time | Replies | Subject |
|---|
| 10:58PM |
1 |
Diaplan reload command not working |
| 10:35PM |
1 |
Phishing attempt posing as digium |
| 8:37PM |
0 |
Meetme Closes Conference After One Hour |
| 8:32PM |
2 |
PGSQL application |
| 5:26PM |
1 |
BLF and realtime SIP buddies |
| 3:35PM |
1 |
multiple RTP port ranges for SIP |
| 3:34PM |
1 |
00h323 cant get gatekeeper to connect |
| 1:53PM |
1 |
dtmf payload 100 |
| 12:58PM |
2 |
Passing a parameter to voicemail |
| 11:16AM |
0 |
I loose incoming call after transfer |
| 9:38AM |
4 |
Extensions.conf changed but not take effect |
| 9:29AM |
1 |
callerid change name |
| 8:09AM |
1 |
func odbc and mult iquery |
| 6:31AM |
0 |
call features affected by native bridging between sip phones |
| 6:13AM |
0 |
CLI not working properly - Asterisk Freez |
| |
| Tuesday March 9 2010 |
| Time | Replies | Subject |
|---|
| 11:20PM |
1 |
Which spandsp to use with 1.6.2? |
| 10:27PM |
0 |
Queue Member stuck in Ring+InUse? |
| 9:50PM |
1 |
Asterisk SMDI for Nortel Option 11 |
| 6:34PM |
1 |
confbridge manager/cli |
| 4:15PM |
3 |
Snom Provisioning |
| 11:31AM |
0 |
Disable echo canceller Fonebridge |
| 10:58AM |
1 |
asterisk peer uses 5060 to send and 5061 to receive |
| 10:54AM |
0 |
Asterisk 1.6.2.5 crash with chan_capi upon calling to PSTN |
| 9:29AM |
1 |
app_queue problem with Ringing state |
| 8:51AM |
0 |
DUNDI Sip authentication failure |
| 1:08AM |
1 |
Aastra, Asterisk 1.4 and Voicemail |
| |
| Monday March 8 2010 |
| Time | Replies | Subject |
|---|
| 11:57PM |
1 |
SIP handset + SLA example |
| 11:31PM |
2 |
fax & spandsp |
| 7:45PM |
5 |
Dialplan behaviour |
| 6:25PM |
0 |
Voip Users Conference March 26th |
| 4:53PM |
1 |
Turning off DNIS on T1 set to FXO_LS protocol |
| 1:49PM |
0 |
Is it possible to configure Asterisk so that it does the Q.SIG Path Replacement Feature ? |
| 10:10AM |
3 |
Calculating R Factor and MOS metrics for VoIP |
| 9:33AM |
1 |
Play an audio file from a remote host |
| 7:50AM |
0 |
Dail of meetme options |
| 6:18AM |
3 |
dahdi not available in Asterisk |
| |
| Sunday March 7 2010 |
| Time | Replies | Subject |
|---|
| 8:18PM |
1 |
Grandstream HT 503 Outoing 403 Forbidden |
| 6:14PM |
1 |
Attended transfer broken in 1.6.0.25 |
| 5:42PM |
1 |
Caller Presentation Confusion |
| 2:21PM |
3 |
Callcenter open source program |
| 9:52AM |
4 |
press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license. |
| 3:18AM |
5 |
dahdi-2.2.1 & kernel-2.6.32: working for anyone? |
| |
| Saturday March 6 2010 |
| Time | Replies | Subject |
|---|
| 11:50PM |
2 |
Mail-2-Fax and Fax-2-Mail solution for Asterisk with T38 |
| 10:09PM |
1 |
Custom App |
| 8:14PM |
2 |
saving pressed keys |
| 7:50PM |
0 |
SIP, internet calling, per-peer contexts, and multitenancy |
| 5:46PM |
1 |
MOH over IAX2 - NOT working |
| 10:41AM |
0 |
Audio problems ins conference zap->sip |
| 9:45AM |
0 |
SIPit 26 in Sweden - organized by Edvina |
| |
| Friday March 5 2010 |
| Time | Replies | Subject |
|---|
| 10:29PM |
2 |
MOH Oddity |
| 8:06PM |
2 |
app_confbridge production ready? |
| 5:40PM |
1 |
Observation about DAHDI, FAX and Echo cancellation |
| 5:18PM |
0 |
Regarding - P-Asserted identity |
| 5:00PM |
1 |
State of 64 bits applications in Asterisk |
| 4:17PM |
3 |
Denial of Service Attack |
| 4:10PM |
1 |
AMI logs |
| 4:08PM |
3 |
Hardware requirements question. |
| 3:33PM |
2 |
FollowMe / Asterisk 1.4 Question |
| 3:15PM |
1 |
Asterisk 1.4 Followme Question |
| 2:41PM |
0 |
Follow-up to CALLERID(num) not working |
| 2:18PM |
1 |
Playback in h extension |
| 12:38PM |
0 |
MGCP FXO endpoint |
| 11:20AM |
4 |
Deadlock in Asterisk 1.4.29.1 |
| 10:38AM |
3 |
Having problems with BLF |
| 9:50AM |
1 |
Asterisk Management API |
| 9:27AM |
1 |
iLBC installation problem |
| 1:42AM |
1 |
SIP / Echo Cancellation |
| 12:22AM |
0 |
PHPAGI and Asterisk 1.6 |
| |
| Thursday March 4 2010 |
| Time | Replies | Subject |
|---|
| 11:35PM |
1 |
Remote Agents |
| 10:51PM |
9 |
30 mins GSM file |
| 10:02PM |
1 |
InterPBX communication using SIP |
| 3:46PM |
1 |
time/date over POTS? |
| 3:00PM |
2 |
UK CallerID -v- Wildcard W100P |
| 2:59PM |
0 |
Asterisk & Sofaware & Polycom |
| 2:19PM |
0 |
Availstatus returns 20 ? |
| 6:43AM |
1 |
No Audio on pstn call |
| 6:02AM |
1 |
[asterisk-user] SIP / Echo Cancellation |
| 3:37AM |
1 |
how to create a dummy call |
| 1:32AM |
0 |
CallerID and distinctive ring detection |
| |
| Wednesday March 3 2010 |
| Time | Replies | Subject |
|---|
| 10:41PM |
6 |
Identify scripts connecting to the asterisk manager |
| 9:23PM |
0 |
Looking for a configuration guru to collaborate with |
| 9:23PM |
1 |
Free 'Locked up' Channels |
| 5:34PM |
1 |
asterisk SIP, SIPAddHeader() and Cisco GED-125 |
| 5:22PM |
2 |
Best practise for ISDN Video Conferencing.. |
| 5:21PM |
1 |
911, channel full |
| 4:30PM |
1 |
forward problem! |
| 3:27PM |
0 |
Is this a bug? |
| 2:22PM |
0 |
CALLERID(num) not working |
| 10:41AM |
0 |
how can I release trunks after transferring 2 calls connected on trunks between the same machines. |
| 10:13AM |
2 |
Getting verbose or debug tracing in Asterisk |
| 6:01AM |
3 |
dahdi and oslec |
| 4:41AM |
0 |
how to play background music during record |
| 4:32AM |
0 |
asterisk-users] how to create a dummy call |
| 1:23AM |
0 |
Dial timeout problem with OpenVox A1200P Card / FXS module |
| |
| Tuesday March 2 2010 |
| Time | Replies | Subject |
|---|
| 11:27PM |
1 |
Uverse, Asterisk and SIP |
| 7:48PM |
0 |
FW: ARI problem with monitor |
| 7:37PM |
1 |
Asterisk and cellphone/GSM voicemailbox |
| 6:36PM |
5 |
MWI and 1.6.1 |
| 5:04PM |
0 |
asterisk-users Digest, Vol 68, Issue 4 |
| 4:54PM |
0 |
ARI problem with monitor |
| 4:28PM |
1 |
realtime call peers status |
| 3:05PM |
1 |
Hide time consuming processed by prompt |
| 12:56PM |
1 |
dialplan reload: not working with large dialplans |
| 11:44AM |
2 |
cli_originate malfunction after upgrade from 1.6.2.0 to 1.6.2.1-5 |
| 11:14AM |
1 |
Sip module problem |
| 10:48AM |
0 |
1.4 chan_sip use internal IP for dialog-info+xml SUBSCRIBE, why? |
| 10:14AM |
6 |
Echo cancellation on DAHDI |
| 7:50AM |
3 |
SIP Trunk with "multiple" remote ip-addresses |
| 6:26AM |
1 |
Does Asterisk 1.6.2.1 Support SIP TLS encryption |
| |
| Monday March 1 2010 |
| Time | Replies | Subject |
|---|
| 11:57PM |
1 |
help with install |
| 11:31PM |
3 |
help!!! Internal extensions not connect |
| 10:46PM |
3 |
User on PC? |
| 10:20PM |
0 |
Solved:Re: OT:4 Line DECT Cordless phone without answering machine |
| 10:08PM |
1 |
OT:4 Line DECT Cordless phone without answering machine |
| 5:04PM |
0 |
Asterisk / Trixbox 2.6 Streaming MOH Problems |
| 4:39PM |
2 |
Unable to register a sip account with x-lite |
| 4:00PM |
1 |
Fwd: Erika DeBenedictis-Recommendation |
| 3:39PM |
1 |
AVM Fritz! mISDN with Kernel 2.6.32 - Any experiences? - Email found in subject |
| 2:34PM |
0 |
Attended transfer: transferring a call as soon as the destination starts ringing |
| 2:31PM |
0 |
SPA3102 Firmware Upgrade via TFTP fails |
| 2:25PM |
3 |
Asterisk and Cisco DTMF |
| 2:06PM |
1 |
rtcachefriends & qualify |
| 1:05PM |
1 |
Swift from eagi, problems with prosody rate |
| 11:42AM |
2 |
MeetMe and usernum |
| 10:22AM |
2 |
Is answer() necessary ? |