| Saturday October 31 2009 |
| Time | Replies | Subject |
|---|
| 9:22PM |
1 |
Determining extension's sip.conf default mailbox |
| 6:56PM |
1 |
Long pause during dialing to IVR |
| 6:44PM |
2 |
Asterisk, Realtime and specify MySQL Table Name ? |
| 6:36PM |
0 |
PRI line resetting on incoming call |
| 4:04PM |
1 |
!<command> from Manager |
| 3:41PM |
2 |
Calls disconnects after short time |
| 3:20PM |
1 |
Disconnecting during the call, analog lines |
| 1:33PM |
3 |
OT - Number Portability |
| 6:57AM |
0 |
Local channel that runs a custom app... why immediate hangup? |
| 12:46AM |
0 |
strange dialing HELP ! |
| |
| Friday October 30 2009 |
| Time | Replies | Subject |
|---|
| 11:28PM |
1 |
Asterisk-1.6.1.8 DTMF with SIP is not working |
| 9:31PM |
3 |
AEX800P on HP Prolaint ML115 kernel panic |
| 7:23PM |
3 |
voicmail: no entry in voicemail config |
| 6:05PM |
1 |
asterisk 1.6 - doing dnsmgr lookup for... / call fails |
| 6:01PM |
1 |
Cannot make calls |
| 5:22PM |
1 |
SNOM 870 |
| 3:57PM |
2 |
Real replacement for AgentCallBackLogin() on Asterisk 1.6 |
| 3:05PM |
7 |
Voicemail file |
| 1:24PM |
1 |
Queue device state problem |
| 11:55AM |
4 |
[IAX] Recommended soft- and hardphones? |
| 11:54AM |
2 |
DAHDI/ZAP overlap dialing |
| 11:08AM |
1 |
inbound routes |
| 6:05AM |
2 |
asterisk 1.6 enable cdr_mysql |
| |
| Thursday October 29 2009 |
| Time | Replies | Subject |
|---|
| 11:36PM |
3 |
Unable to set TOS to 184? |
| 9:49PM |
1 |
Astreicon presentations |
| 9:16PM |
0 |
AsteriskForge Now Open |
| 8:50PM |
1 |
Booting Error for /dev/kmem |
| 8:26PM |
0 |
!! Unknown IE 50 (cs5, Unknown Information Element) on console. |
| 10:49AM |
1 |
Zap inbound hangup problem |
| 7:48AM |
1 |
Async Agi problem |
| 1:05AM |
5 |
Dynamic DNS trunk |
| 12:29AM |
2 |
GUI for hunt groups? |
| |
| Wednesday October 28 2009 |
| Time | Replies | Subject |
|---|
| 10:57PM |
3 |
Mysql CDR in Addons 1.6.2.0-rc1 does not record CLID |
| 10:46PM |
1 |
Asterisk 302 Moved Temporarily |
| 9:26PM |
0 |
Asterisk/Cisco AS5300 => Two problems in |
| 8:24PM |
1 |
MOH |
| 6:16PM |
1 |
SIP 18x Messages |
| 2:59PM |
5 |
need a local tech |
| 2:44PM |
1 |
Clear pending SIP channels |
| 2:07PM |
2 |
deploying asterisk |
| 11:41AM |
0 |
sip fullcontact and port values |
| 11:34AM |
1 |
CDR(billsec) |
| 11:27AM |
1 |
SIP Peers still ping with SIP OPTIONS on a reload |
| 10:53AM |
1 |
The SIP in the Mobile Phones are not able to register on asterisk |
| 8:50AM |
1 |
Asterisk Server with Panasonic PBX |
| 6:29AM |
1 |
SIP client MAC address. |
| 5:50AM |
1 |
Dialing out a T1 |
| 5:09AM |
2 |
Asterisk/Cisco AS5300 => Two problems in incoming (extension not found) |
| 3:03AM |
2 |
Confusion on caller-ID with SIP provider |
| |
| Tuesday October 27 2009 |
| Time | Replies | Subject |
|---|
| 11:29PM |
2 |
need to find firmware for cisco ata-188 |
| 9:13PM |
5 |
Software for PC-PC voice comunication |
| 8:58PM |
1 |
The Mobile devices are not able to register on my asterisk |
| 7:26PM |
0 |
Codecs with MixMonitor (,a) option |
| 2:18PM |
10 |
How to dial multiple extensions at once like in a ring group and put them in conference? |
| 1:59PM |
1 |
pri intense debug span 1 |
| 1:10PM |
0 |
Fax with a AEx410P and Beronet BN4S0 => Sending Problem |
| 12:56PM |
1 |
RTP timestamps |
| 12:35PM |
0 |
[OT] Snom M9 |
| 9:49AM |
3 |
installing |
| 9:34AM |
3 |
Installing Asterisk |
| |
| Monday October 26 2009 |
| Time | Replies | Subject |
|---|
| 11:36PM |
0 |
Asterisk 1.6.1.8 Now Available |
| 8:26PM |
0 |
AST-2009-007: ACL not respected on SIP INVITE |
| 8:05PM |
2 |
What is the best way to configure this? |
| 7:59PM |
1 |
DAHDI not detecting RINGING Status on the Channel |
| 7:52PM |
1 |
Answer call from another device |
| 6:43PM |
1 |
state_interface backport issue |
| 4:59PM |
1 |
Cancel attended transfer |
| 1:07PM |
1 |
IAX jitterbufer oddity |
| 12:49PM |
0 |
Call Record. |
| 12:02PM |
3 |
No tone, one way communcation. |
| 10:10AM |
1 |
Common Community for exchange the routes via Asterisk boxes |
| |
| Sunday October 25 2009 |
| Time | Replies | Subject |
|---|
| 11:39PM |
1 |
some issue with libpri cant go past 1.4.1 |
| 9:10PM |
1 |
chan_echolink |
| 8:58PM |
2 |
Asterisk as the recording server for Avaya Definity |
| 4:17PM |
0 |
FW: Queue Transfers |
| 4:16PM |
0 |
Queue Transfers |
| 2:47PM |
1 |
Asterisk-stat! - help needed (once again due to mailserver problem) |
| 2:19PM |
2 |
SIP interconnection problem |
| 1:51PM |
2 |
help sip show on CLI : no such command |
| 12:33AM |
2 |
test |
| |
| Saturday October 24 2009 |
| Time | Replies | Subject |
|---|
| 4:26PM |
3 |
OT - mISDN and B410P questions |
| 2:57PM |
2 |
SVN newbie - No trunk/ in http://svn.asterisk.org/svn/libpri/ |
| 2:40PM |
0 |
Tim Panton Astricon Presentation "recreated" |
| 10:24AM |
0 |
Manage users & administrator for the asterisk-gui |
| 10:05AM |
0 |
AMI script.. |
| |
| Friday October 23 2009 |
| Time | Replies | Subject |
|---|
| 9:24PM |
1 |
IVR reports? |
| 8:44PM |
1 |
how to announce the agent answering in a queue to the caller |
| 8:26PM |
3 |
SIREN14 call setup and record/playback |
| 7:23PM |
0 |
Crash with app_mixmonitor |
| 6:21PM |
0 |
Asterisk SIP to Cisco IAD2430 Series? |
| 4:20PM |
1 |
Strange IAX2 / Iaxmodem problem |
| 3:19PM |
1 |
GUI for asterix management |
| 1:33PM |
1 |
Recording management for IVR |
| 11:36AM |
2 |
How to generate 183 Session Progress |
| 11:01AM |
0 |
asterisk crashes when calling gtalk user |
| 6:48AM |
2 |
interfacing asterisk with a legacy PBX |
| 6:33AM |
1 |
AstriCon videos: a question of method (Robin) |
| |
| Thursday October 22 2009 |
| Time | Replies | Subject |
|---|
| 11:54PM |
1 |
Asterisk MOH playing old audio for first 30 to 60 seconds |
| 11:19PM |
1 |
OSLEC with DAHDI and Linksys/Sipura |
| 11:04PM |
1 |
Poor VoIP voice quality in one direction from three providers |
| 10:34PM |
4 |
AstriCon videos: a question of method |
| 5:12PM |
2 |
hangup from which side |
| 4:42PM |
1 |
GSM 6.10 codec for Asterisk |
| 4:33PM |
2 |
IAX Hardphones. |
| 3:48PM |
1 |
Can't configure Cisco 7942 avec factory reset |
| 3:07PM |
2 |
ChanSpy in Asterisk 1.2.24 |
| 2:50PM |
4 |
OT - How to organize TFTP root directory ? |
| 2:30PM |
2 |
carefulwrite: write() returned error: Broken pipe |
| 11:32AM |
1 |
queues autopause |
| 9:45AM |
1 |
Audio issue in skype for asterisk |
| 6:43AM |
2 |
AGI STREAM FILE and not blocking execution |
| 4:07AM |
2 |
ivr menu not hanging up call |
| 2:48AM |
0 |
Asterisk 1.6.1.6 crashing -- multiple phone entries |
| |
| Wednesday October 21 2009 |
| Time | Replies | Subject |
|---|
| 8:19PM |
1 |
Incorrect voice mail format on transfer |
| 6:50PM |
5 |
Asterisk and Nuance Vocalizer TTS Engine |
| 6:30PM |
4 |
Concurrent calls including mysql taking lot of time for execution |
| 4:54PM |
1 |
OT - Gigaset Chagall - How to download firmware without Internet access ? |
| 4:10PM |
0 |
DAHDI: TCM PCI Master abort |
| 3:08PM |
0 |
Intermittent Low volume |
| 2:57PM |
1 |
polarity on some channels |
| 12:42PM |
0 |
ringing... or lack thereof |
| 11:39AM |
0 |
TxFax works only with one of 2 PRI |
| 11:05AM |
3 |
Need Help |
| 10:46AM |
1 |
RAMDisk vs Extarnal server for recording |
| 9:47AM |
3 |
Searching on how to keep local calls... local |
| 8:04AM |
1 |
error - sources for the 2.6.18-92.1.22.el5xen kernel |
| 2:06AM |
1 |
ChannelStateDesc: Ring ? |
| |
| Tuesday October 20 2009 |
| Time | Replies | Subject |
|---|
| 11:34PM |
4 |
Linksys 962 |
| 10:36PM |
1 |
OutCALL |
| 10:26PM |
1 |
Is there a way to force a codec on an incoming sip uri call? |
| 10:25PM |
2 |
Kernel panic w/ DAHDI 2.x/Digium TE220B |
| 9:50PM |
1 |
Libpri-1.4.10.2 Released |
| 8:40PM |
3 |
High Volume Call Center SIP versus IAX2 |
| 6:32PM |
3 |
troubleshooting NAT |
| 2:08PM |
0 |
srtp crypto lifetime |
| 12:24PM |
6 |
Syncronizing files on different Asterisk servers |
| 11:59AM |
0 |
cascaded pickup |
| 11:01AM |
2 |
AMI 1.0 -> 1.1 with originate. |
| 9:43AM |
0 |
need help with card for IVR |
| 8:55AM |
3 |
Cisco 7921 |
| 8:03AM |
2 |
(no subject) |
| 6:26AM |
0 |
Voip interconnection with ACME SBC-asterisk-users Digest, Vol 63, Issue 52 |
| 12:30AM |
2 |
all our circuits are busy now |
| |
| Monday October 19 2009 |
| Time | Replies | Subject |
|---|
| 10:42PM |
1 |
Cisco 1751 setup with asterisk |
| 9:51PM |
0 |
ANN: Asterisk-Java 1.0.0.M3 Released |
| 9:37PM |
3 |
Dial a external number with extension |
| 9:29PM |
3 |
IMAP voicemail using subfolders fails. |
| 7:08PM |
1 |
Looking for the asterisk 'off' sound file |
| 6:06PM |
3 |
delay in processing dtmf |
| 4:17PM |
3 |
asterisk services not starting up |
| 3:58PM |
2 |
Astricon talk on wideband codecs |
| 3:47PM |
3 |
update CDRs in mysql during a call |
| 2:18PM |
2 |
Problems replaying G.726 - only noise |
| 1:55PM |
2 |
VoIP interconnection with Acme packet SBC |
| 1:42PM |
0 |
Missing digits from CallerID on TDM400P? |
| 11:20AM |
0 |
question about getting instance ringing member in queue |
| 10:12AM |
0 |
announcement tone to callees of app_page |
| |
| Sunday October 18 2009 |
| Time | Replies | Subject |
|---|
| 11:34PM |
2 |
BTS |
| 11:20PM |
4 |
Customising Firmware |
| 10:18PM |
1 |
SIP Headers |
| 8:27PM |
1 |
SIP debugging enabled : written to log |
| 6:43PM |
1 |
Asterisk Expert Freelancer in Karachi |
| 6:05PM |
1 |
Call from 'sip-id' to extension 'sip-id' rejected because extension not found ? |
| 2:24PM |
1 |
Asterisk+Sphinx4 for simple mobile phone <-> server speech recognition |
| 2:06PM |
0 |
Kernel Timer |
| 10:32AM |
4 |
Snom870 sidecar |
| 8:46AM |
0 |
Sunday 18th 12N-3P PDST Global Asterisk Mtg - BerkeleyTIP - for forwarding |
| 8:24AM |
1 |
SIP debugging enabled : written to log ?? |
| 2:29AM |
4 |
Astricon |
| 12:30AM |
7 |
Asterisk Monitoring |
| 12:01AM |
0 |
Best practices for reliable carrier grade telephony |
| |
| Saturday October 17 2009 |
| Time | Replies | Subject |
|---|
| 6:26PM |
3 |
Possible bug in app_meetme.c |
| 3:18PM |
1 |
sched_settime: Request to schedule in the past?!?! |
| 2:02PM |
3 |
OT - DECT SIP Phones |
| 1:45PM |
4 |
how to limit the calls leaving a queue? |
| 1:02PM |
0 |
OT - KVM and PCI telephony cards |
| |
| Friday October 16 2009 |
| Time | Replies | Subject |
|---|
| 11:46PM |
0 |
Nehalem & Digium Wildcard issues? |
| 11:37PM |
5 |
IVR |
| 11:06PM |
1 |
Whither asterisk-addons? |
| 8:04PM |
2 |
SIP to IAX to SIP |
| 6:28PM |
1 |
can i use Asterisk to send sms to my database users? |
| 4:33PM |
0 |
OT Old Sipura OK - Linksys (junk) |
| 4:20PM |
0 |
Origin of "Exceptionally long voice queue length queuing to IAX2/blahblah" messages |
| 2:40PM |
1 |
inquire if SIP connections are active or not |
| 1:41PM |
0 |
app_swift issue |
| 10:56AM |
2 |
Soft phone not registering |
| 10:07AM |
2 |
Invite after bye? |
| 10:04AM |
1 |
Check if a variable is set |
| 6:16AM |
3 |
Linux/Asterisk on game consoles? |
| 5:13AM |
1 |
The City of Amsterdam has been deploying asterisk throughout the city! |
| 2:37AM |
1 |
Mixing SIP/TDM in MeetMe |
| |
| Thursday October 15 2009 |
| Time | Replies | Subject |
|---|
| 10:19PM |
2 |
OT wanted old Sipura firmware 2.0.13 |
| 9:52PM |
0 |
Asterisk and FreePBX Amazon EC2 instances are now available in Europe |
| 9:21PM |
2 |
question on SIP and call manager |
| 7:58PM |
2 |
2 IPs for an Asterisk server. |
| 7:50PM |
2 |
A little OT but need an opinion on Aastra 57i CT |
| 7:24PM |
1 |
OT - Can't upgrade Cisco 7942 to SIP |
| 7:20PM |
3 |
DS3 capacity calls using asterisk |
| 6:57PM |
0 |
Best way to detect fax in Asterisk 1.6?? |
| 6:57PM |
1 |
Testing the Timing Device |
| 4:42PM |
1 |
best way to make 5-10 simultaneous calls to the same did at a set time of day |
| 4:41PM |
1 |
sporadic one-way audio |
| 4:34PM |
4 |
PSTN to SIP line ratio |
| 3:16PM |
0 |
Does ADA 1.1 or ADA Pro exists ? |
| 1:28PM |
1 |
Where to find IMAP storage doc ? |
| 12:50PM |
2 |
hi |
| 10:27AM |
2 |
Asterisk with a Cisco AS5300 gateway |
| 8:05AM |
4 |
Calls hang up after 20 seconds |
| 2:56AM |
1 |
Callpickup works for outside calls but not inside calls |
| 2:41AM |
2 |
MWI for multiple voice mail boxes |
| |
| Wednesday October 14 2009 |
| Time | Replies | Subject |
|---|
| 9:24PM |
3 |
Extension Paging |
| 8:52PM |
1 |
Door Phones |
| 7:53PM |
0 |
WaitForSilence doesn't work unless Background is called? |
| 6:44PM |
5 |
multiple call |
| 6:00PM |
1 |
Cisco router |
| 4:33PM |
1 |
PostgreSQL problems |
| 4:04PM |
2 |
ACD & ASR |
| 2:39PM |
2 |
Config Files |
| 1:24PM |
1 |
no outbound calls |
| 1:01PM |
1 |
Asterisk 1.4 vs 1.6 |
| 11:58AM |
2 |
Queues with unavailable members |
| 11:19AM |
0 |
SIP RealTime defaultuser Field Cleared |
| 9:36AM |
1 |
DTMF failing in some calls |
| 7:27AM |
2 |
FXS to SIP gateway |
| 6:10AM |
2 |
DAHDI Dummy for Linux VServers |
| 3:19AM |
1 |
ChanSpy on asterisk 1.6 |
| 12:42AM |
2 |
T38 negotiations in RTP |
| 12:22AM |
8 |
Asterisk in the Cloud |
| |
| Tuesday October 13 2009 |
| Time | Replies | Subject |
|---|
| 11:56PM |
4 |
AMI input streams limit? |
| 6:08PM |
0 |
Bridge command in 1.6 |
| 4:04PM |
11 |
Best Firewall Suggestions? |
| 3:07PM |
2 |
Dial Delay |
| 9:31AM |
0 |
missing CDR records in cdr while kick from meetme |
| 8:19AM |
3 |
strange transcoding values |
| |
| Monday October 12 2009 |
| Time | Replies | Subject |
|---|
| 9:53PM |
0 |
asterisk dialplan to share fax line |
| 8:36PM |
0 |
video support with voicemail question |
| 6:19PM |
0 |
How to send a digit to a channel?? |
| 5:36PM |
5 |
G729 in asterisk upgrade issue |
| 4:05PM |
0 |
Asterisk 1.6 with TDM2400 and 2 FXS modules |
| 3:56PM |
0 |
live audio streaming using monitor, mixmonitor or chanspy |
| 1:21PM |
0 |
libss7 problem with dialing a non numeric string |
| 12:24PM |
1 |
tealtime static |
| 12:21PM |
0 |
meetme and confbridge |
| 10:51AM |
2 |
SPRINTF option : format %1$s not supported |
| 3:25AM |
1 |
How to do a 3 party Warm Transfer in Asteriks 1.4 |
| |
| Sunday October 11 2009 |
| Time | Replies | Subject |
|---|
| 11:42PM |
0 |
Grandstream 2010 |
| 11:15PM |
5 |
Call Recording and Posting |
| |
| Saturday October 10 2009 |
| Time | Replies | Subject |
|---|
| 10:25PM |
3 |
Method to use SOX inside a Dialplan |
| 9:18PM |
1 |
Asterisk to Asterisk access voicemail - not working |
| 8:46PM |
2 |
outgoing sip calls work; incoming calls fail |
| 8:39PM |
2 |
Mp3 for IVR prompts |
| 4:13PM |
1 |
Grandstream GXP 2010 : multiple accounts not working |
| 2:32PM |
0 |
paging/intercom |
| 1:34PM |
0 |
Slightly OT: Astricon and Google Wave |
| 1:09PM |
1 |
delay to dial |
| 1:15AM |
2 |
Wifi GSM handover |
| |
| Friday October 9 2009 |
| Time | Replies | Subject |
|---|
| 11:20PM |
0 |
lawnmower man "attack" ?? |
| 11:12PM |
1 |
lawnmower man "attack" sip tag=Zerogij34 some one else notice this in 20th september or recently? |
| 9:48PM |
2 |
Incoming extension not working. |
| 8:45PM |
1 |
choppy sound |
| 8:27PM |
1 |
${REASON} not getting set. |
| 5:44PM |
3 |
Chanspy |
| 3:52PM |
0 |
calls ansowered for 1 second or less |
| 3:03PM |
1 |
Billing applications |
| 2:07PM |
1 |
VoiceMail and IMAP |
| 12:54PM |
1 |
wrond DTMF detection on Zap channel |
| 10:16AM |
0 |
Trunk and Pstn line |
| 9:09AM |
1 |
G.729 and Voicemail |
| 9:03AM |
0 |
Asterisk Queue & Agent |
| 8:00AM |
1 |
Today's problem: Inbound call routing |
| 7:41AM |
1 |
Digium G729 licence unattended install |
| 3:53AM |
2 |
SIP Hard Phone with SMS |
| |
| Thursday October 8 2009 |
| Time | Replies | Subject |
|---|
| 9:37PM |
1 |
Realtime static does not work in 1.6.1 or 1.6.2 |
| 8:40PM |
1 |
Help setting up IMAP_STORAGE on CentOS 5 |
| 8:03PM |
2 |
Best QoS for Linux |
| 7:43PM |
4 |
No sound on voicemail from analog line |
| 5:35PM |
0 |
asterisk 2bct/rlt calling |
| 5:10PM |
0 |
Fuori ufficio |
| 3:31PM |
0 |
Suggestions for low level RTP stream generator? |
| 3:22PM |
1 |
g729 free codec any idea |
| 2:55PM |
0 |
Friday Noon VUC with guest Alex Robar |
| 2:52PM |
2 |
Best afordable router with QOS for * |
| 2:29PM |
2 |
Server-side scripting when SIP phones register |
| 2:25PM |
1 |
Having trouble with "IF" and blanks |
| 1:43PM |
4 |
Dialplan problem |
| 1:20PM |
2 |
How to keep difference between 2 SIP-accounts/trunks from same server ?? |
| 12:40PM |
2 |
Asterisk and Sheeva "wall wart". |
| 12:22PM |
1 |
MeetMe option question |
| 8:57AM |
2 |
limiting number of channels to be accessed |
| 7:03AM |
0 |
MeetMe + SLA |
| 12:05AM |
1 |
Drop Call on ICMP Port Unreachable? |
| |
| Wednesday October 7 2009 |
| Time | Replies | Subject |
|---|
| 8:57PM |
2 |
Can dial long distance but not local? |
| 7:20PM |
1 |
VPS Server |
| 4:25PM |
1 |
DTMF Issues |
| 3:22PM |
1 |
Need provider recommendations for the UK |
| 10:21AM |
2 |
system cmd + fax line |
| 10:06AM |
0 |
Asterisk Integration with RDP Property Management Software? |
| |
| Tuesday October 6 2009 |
| Time | Replies | Subject |
|---|
| 10:40PM |
0 |
Problem sending a DTMF remotely. Please need help!!! |
| 8:22PM |
0 |
Asterisk 1.4.27-rc2, 1.6.0.16-rc2, 1.6.1.7-rc2, and 1.6.2.0-rc3 Now Available |
| 8:15PM |
1 |
Asterisk 1.6.1.6 suddently restarts ... |
| 8:01PM |
7 |
MPG123 Dying |
| 7:49PM |
2 |
Transfers from Queue Calls |
| 4:18PM |
2 |
adding modules |
| 3:55PM |
1 |
Is anyone doing real time updates to where asterisk registers? |
| 3:40PM |
4 |
Cent OS 5.3 All Updated Asterisk Installation Giving Error |
| 2:35PM |
1 |
Asterisk + Monitor() Poor quality |
| 2:05PM |
0 |
video support over iax |
| 1:51PM |
0 |
parse transparent ISUP parameters |
| 1:33PM |
0 |
What happened to MACRO_EXTEN in AEL macros since 1.6? |
| 12:07PM |
0 |
Lancom 1722 and Asterisk (i need HELP) |
| 9:24AM |
0 |
fsk callerid with DTAS start? |
| 5:02AM |
2 |
T38 REINVITe issue |
| |
| Monday October 5 2009 |
| Time | Replies | Subject |
|---|
| 11:14PM |
5 |
Networking Concept |
| 6:34PM |
6 |
Receptionist GUI? |
| 4:53PM |
2 |
Method to downgrade asterisk |
| 4:48PM |
0 |
web module for video calls |
| 3:45PM |
2 |
dahdi dies with "No more room in scheduler" |
| 3:26PM |
0 |
Asterisk and QSIG |
| 3:03PM |
1 |
DTMF problem during read() |
| 2:55PM |
3 |
OriginateResponse Event |
| 2:47PM |
0 |
What dahdi_dynamic and dahdi_transcode modules are for? |
| 2:17PM |
3 |
Questions about app_jack.c |
| 11:27AM |
1 |
Problem sending a DTMF remotely. Please need help!! |
| 9:14AM |
1 |
Grandstream GXW4024 experience |
| 7:47AM |
1 |
AEL problem: bug or feature? |
| 4:04AM |
1 |
Drop calls when using Flash Operator Panel |
| 1:05AM |
1 |
Peculiar error message when using Q-SIG |
| |
| Sunday October 4 2009 |
| Time | Replies | Subject |
|---|
| 2:41PM |
0 |
Hacking the network |
| 10:28AM |
9 |
Zaptel problems on SUSE 9.3 |
| 9:45AM |
1 |
x100p card |
| 12:05AM |
3 |
After call into console/dsp hangup hear ringing |
| |
| Saturday October 3 2009 |
| Time | Replies | Subject |
|---|
| 5:19PM |
0 |
Ideasip |
| 4:02PM |
0 |
ERROR[1499]: rtp.c:2482 ast_rtcp_write_sr: RTCP SR transmission error |
| 1:19PM |
1 |
Asterisk on NSLU2 : Grandstream not registering |
| 1:17PM |
1 |
Calls being dropped - Cisco 7940 with SIP 8.12 image |
| 9:36AM |
1 |
Asterisk and Jack |
| 12:40AM |
0 |
Problem sending a DTMF remotely. Please need help... |
| |
| Friday October 2 2009 |
| Time | Replies | Subject |
|---|
| 10:38PM |
1 |
Asterisk,click2talk, webphone |
| 6:42PM |
3 |
app_hackblock to prevent SIP/IAX reg trolling |
| 6:22PM |
3 |
Extra Sounds Missing on 1.6.1.6 install |
| 5:58PM |
2 |
Asterisk HA Current Thoughts (Centos 5.3 Platform) |
| 5:48PM |
0 |
Asterisk + Skype deployment |
| 2:51PM |
0 |
Sending a DTMF remotely with PlayDTMF problem. |
| 2:42PM |
1 |
One side SIP goes dead on length conversation |
| 1:20PM |
2 |
Followme |
| 12:52PM |
1 |
How to call extensions and add them to a conference room |
| 12:34PM |
1 |
Creating a clear channel on zaptel |
| 12:30PM |
1 |
Problem with inbound calls - asterisk 1.6.1.6 |
| 9:10AM |
0 |
srtp issue |
| 8:48AM |
0 |
Free version of softswitch with billing and routing released |
| 6:50AM |
0 |
Busy() returns immediately on IAX trunk |
| 12:14AM |
1 |
IAX2 Call rejected, CallToken Support required |
| |
| Thursday October 1 2009 |
| Time | Replies | Subject |
|---|
| 6:27PM |
1 |
QOS/DSCP for IAX? |
| 5:02PM |
1 |
DTMF problems during a message play |
| 2:10PM |
3 |
What are the reasons for VoIP echo? |
| 1:04PM |
1 |
Is there a way to get info who disconnected the call into CDR? |
| 11:13AM |
1 |
Busy app timeout |
| 11:09AM |
1 |
portech MV-378 SIP GSM Gateway |
| 10:23AM |
1 |
RTP Delayed during RTCP |
| 10:18AM |
0 |
Friday Oct 2: Digium's new Speech Recognition for Asterisk |
| 9:43AM |
2 |
INVITE Sending Local IP |
| 8:30AM |
2 |
help on ${RTPAUDIOQOS} |
| 7:32AM |
2 |
Softphone in Web |
| 4:56AM |
3 |
"got stuck at 150 calls, above that not working in stress test" |
| 2:56AM |
0 |
Issue with SIP & QSIG phones in MeetMe conf room |