| Tuesday May 31 2011 |
| Time | Replies | Subject |
|---|
| 10:24PM |
1 |
SIP Register DOS attack |
| 10:03PM |
0 |
Mitel PBX caller id format? |
| 5:38PM |
3 |
AMI buffering event output? |
| 4:48PM |
0 |
Dropping incompatible voice frame on DAHDI/i1/xxxxxxx of format slin since our native format has changed to 0x4 (ulaw) |
| 2:21PM |
1 |
queuemetrics with 1.8 queue_log |
| 12:32PM |
2 |
To know if the ISDN PRI E1 is UP? |
| 5:18AM |
1 |
BRI confiugration error |
| |
| Monday May 30 2011 |
| Time | Replies | Subject |
|---|
| 10:29PM |
1 |
Configuring ISDN PRI using DAHDI |
| 5:30PM |
3 |
please help |
| 12:32PM |
1 |
ControlPlayback's options |
| 9:23AM |
1 |
CLI command 'database deltree' doesn't remove family with space in its name |
| 8:03AM |
2 |
DAHDi installation problem |
| 6:20AM |
0 |
asterisk in some kind of loop |
| 1:10AM |
1 |
asterisk fails when DNS or internet fails |
| |
| Sunday May 29 2011 |
| Time | Replies | Subject |
|---|
| 11:47AM |
5 |
Free CNAM |
| 8:57AM |
3 |
Why PRI not BRI ? |
| |
| Saturday May 28 2011 |
| Time | Replies | Subject |
|---|
| 8:08PM |
8 |
Cisco registration problem with 1.8.3.3 |
| 11:34AM |
2 |
dtmf Caller-id detection before first ring |
| |
| Friday May 27 2011 |
| Time | Replies | Subject |
|---|
| 5:30PM |
2 |
More Cores or more CPU Speed |
| 5:12PM |
0 |
Asterisk on FreeBSD 8.2 |
| 4:33PM |
5 |
DAHDI span timeing source |
| 3:45PM |
2 |
disable sip registration |
| 3:10PM |
3 |
standalone PRI-to-SIP converter |
| 2:42PM |
4 |
DID for outbound PSTN call |
| 1:41PM |
0 |
Dahdi and function CHANNEL |
| 10:28AM |
1 |
About Redfone Configuration |
| 9:29AM |
7 |
how to specify the numbers to call with sip |
| 8:31AM |
2 |
Audio dropping |
| 6:44AM |
0 |
ldap questions |
| 12:50AM |
1 |
user asterisk does not exist - using root |
| |
| Thursday May 26 2011 |
| Time | Replies | Subject |
|---|
| 9:12PM |
0 |
Grandstream + IPv6 |
| 6:24PM |
0 |
Dahdi channel stuck in "ringing" state |
| 6:19PM |
3 |
DB driven voicemail |
| 4:55PM |
5 |
make calls from DID |
| 1:38PM |
0 |
DTMF digits received, but not completely forwareded |
| 1:09PM |
3 |
UK English sounds packs |
| 10:45AM |
1 |
Is this Asterisk issue of feature |
| |
| Wednesday May 25 2011 |
| Time | Replies | Subject |
|---|
| 9:33PM |
2 |
Tr : how to user SIP realtime option |
| 5:32PM |
6 |
Asterisk 1..8 multiple queue |
| 6:00AM |
0 |
How to put your call on hold with asterisk Dialplan |
| 5:26AM |
2 |
asterisk hint SIP presence |
| 4:45AM |
1 |
synway |
| |
| Tuesday May 24 2011 |
| Time | Replies | Subject |
|---|
| 11:20PM |
3 |
How to enable the addon in the Asterisk 1.8 compilation |
| 10:50PM |
0 |
Asterisk SIP Trunk with CUCM Express, Disable Comfort Noise? |
| 8:50PM |
1 |
Skype for Asterisk - RIP |
| 7:39PM |
1 |
Asterisk + Patton ISDN2e gateway (UK) |
| 4:20PM |
0 |
Asterisk 1.8.4.1 Now Available |
| 3:22PM |
0 |
Realtime dbase table mods |
| 2:07PM |
0 |
SMS callfile |
| 10:19AM |
1 |
SIP per-call heartbeat? |
| 10:12AM |
0 |
issue with asterisk 1.4 |
| 8:19AM |
0 |
Grandstream and setvar |
| |
| Monday May 23 2011 |
| Time | Replies | Subject |
|---|
| 9:30PM |
2 |
Sending call to specific IP address |
| 5:36PM |
2 |
chan_zap |
| 2:47PM |
0 |
Asterisk + USSD |
| 2:41PM |
1 |
AJAM XML output not valid xml |
| 12:43PM |
3 |
Latest DAHDI/libpri/Asterisk 1.8 & 1x BRI port HFC based ISDN card? |
| 12:41PM |
1 |
[Fwd: FW: extconfig.conf] |
| 12:29PM |
0 |
Asterisk 1.8 TLS with Softphone blink on Windows don´t work |
| 12:25PM |
0 |
asterisk 1.6.2.17.2 Warning on startup |
| 9:06AM |
1 |
Asterisk DTMF 'talkoff' issues |
| 7:26AM |
1 |
SIP-T to SIP Gateway |
| |
| Sunday May 22 2011 |
| Time | Replies | Subject |
|---|
| 11:05PM |
5 |
call files .vbs |
| |
| Saturday May 21 2011 |
| Time | Replies | Subject |
|---|
| 12:28PM |
1 |
how to user SIP realtime option |
| 12:19PM |
3 |
difference between SIP peer and SIP user ? |
| |
| Friday May 20 2011 |
| Time | Replies | Subject |
|---|
| 11:43PM |
0 |
looking for testers for app_meetme AMI patch |
| 10:04PM |
0 |
AstLinux 0.7.8 Release |
| 6:37PM |
0 |
Playback noanswer & SIP |
| 6:20PM |
2 |
Faxing with Asterisk 1.8.4 & T.38 |
| 6:10PM |
5 |
Restart asterisk destroy all registered SIP peers |
| 4:10PM |
0 |
Static agent in queue |
| 1:42PM |
1 |
SIP Diversion RDNIS - how to get reason parameter? |
| 1:37PM |
1 |
AstManProxy |
| 10:07AM |
1 |
Hints custom:abcdef |
| 9:59AM |
1 |
*8 pickup and CLI presentation |
| 9:57AM |
0 |
VUC: Sangoma NetBorder 4.0 and an Androis SIP client from Media5 Corp |
| 9:33AM |
0 |
first dtmf is not detected |
| 5:19AM |
0 |
Using a feature from AMI or CLI |
| |
| Thursday May 19 2011 |
| Time | Replies | Subject |
|---|
| 10:13PM |
2 |
[Fwd: FW: realtime mysql - p4] |
| 9:10PM |
2 |
Agent (Invalid) has taken no calls yet |
| 5:24PM |
1 |
Polycom IP335 3.3.1 Call Waiting |
| 4:44PM |
2 |
click to call with php |
| 4:41PM |
1 |
Static Vs Dynamic queue confusion |
| 4:23PM |
2 |
Dropping incompatible voice frame |
| 4:05PM |
3 |
Manager logged on/off messages |
| 3:10PM |
1 |
Getting 603 Declined after AGI execution |
| 1:39PM |
6 |
ConfBridge - Failed to find a bridge technology to satisfy capabilities |
| 6:05AM |
1 |
SIP 603 Declined after AGI execution |
| 2:05AM |
1 |
Pridialplan/ prilocaldialplan |
| 12:01AM |
1 |
v1.8.4: Extension Not found in Context? |
| |
| Wednesday May 18 2011 |
| Time | Replies | Subject |
|---|
| 8:40PM |
3 |
asterisk's zombie processes |
| 8:32PM |
1 |
asterisk18 - realtime/mysql - take 3 |
| 2:05PM |
2 |
Failover trunks |
| 12:07PM |
1 |
Using Asterisk/Digium repos => Astribank firmware not found |
| 9:22AM |
0 |
Make Multiple Calls using Chan_alsa module |
| 7:57AM |
0 |
Sending SRTP to Asterisk Gateway ends up in authentication failure |
| 5:21AM |
3 |
SRTP of Asterisk |
| 12:12AM |
0 |
Log off all agents from all queues... |
| |
| Tuesday May 17 2011 |
| Time | Replies | Subject |
|---|
| 10:25PM |
1 |
Queues.conf Language Agents |
| 8:21PM |
2 |
script to trim sip.conf |
| 6:50PM |
1 |
Question on AMI |
| 5:30PM |
5 |
Skype-like dialing from web page |
| 5:12PM |
1 |
OT, free software for SIP ladder diagrams? |
| 3:46PM |
1 |
Name or service not known |
| 2:36PM |
1 |
mysql call stored procedure |
| 2:16PM |
3 |
how to know how many calls are on hold |
| 1:45PM |
0 |
Type of number in outgoing SETUP frame |
| 1:17PM |
0 |
put multiple call on hold by dialplan in asterisk |
| 12:18PM |
1 |
OT - Which XMPP server for Jingle-enabled XMPP service ? |
| 8:43AM |
4 |
Automatic dialing + SMS |
| 1:27AM |
0 |
3. Re: ITSP Multi IPs (Alex Balashov) Asterisk-users Digest, Vol 82, Issue 33 |
| |
| Monday May 16 2011 |
| Time | Replies | Subject |
|---|
| 9:54PM |
2 |
Reporting Tool: To show who is login, queue, ... etc |
| 6:41PM |
3 |
dahdi command not available |
| 4:05PM |
1 |
Step by step guide |
| 4:03PM |
1 |
Missing Config Files under /etc/asterisk |
| 3:25PM |
1 |
AMI check if connection is alive |
| 1:36PM |
1 |
question on digium repo |
| 1:00PM |
2 |
Different box for SIP and RTP |
| 12:56PM |
1 |
AMD tweaking |
| 12:14PM |
2 |
AMI perl daemon |
| 10:50AM |
0 |
1.8.4 quitting console |
| 10:19AM |
0 |
1.8.4 keeps quitting console by itself |
| |
| Sunday May 15 2011 |
| Time | Replies | Subject |
|---|
| 11:26PM |
0 |
Asterisk 1.8 and 1.4 SlackBuilds for Slackware Linux |
| 7:59PM |
0 |
Alarms Sound files |
| 7:12PM |
0 |
asterisk-users Digest, Vol 82, Issue 52 |
| |
| Saturday May 14 2011 |
| Time | Replies | Subject |
|---|
| 11:51PM |
3 |
iptables for Asterisk - Any good guides out there? |
| 3:37PM |
1 |
Asterisk 1.41 - Warning and Notice about contact info and stale nonce |
| 3:31PM |
0 |
How to install the new cdr-stats? |
| 3:59AM |
10 |
Asterisk-cpu utilization > 60 % |
| |
| Friday May 13 2011 |
| Time | Replies | Subject |
|---|
| 6:58PM |
2 |
OPTIONS Keep alive - Reply: 481 No subscription |
| 6:37PM |
0 |
Unusual message |
| 5:32PM |
1 |
Asterisk 1.6: Custom Name for Recordings file |
| 3:28PM |
0 |
Asterisk 1.6 - voice quality becomes poor after several minutes. |
| 3:14PM |
1 |
res_timing_timerfd.so Vs res_timing_dahdi.so |
| 2:26PM |
1 |
outbound calls via google voice not answered by toll free numbers with ivrs |
| 2:24PM |
2 |
Backport of DEVICE_STATE to 1.4 |
| 2:19PM |
0 |
Unknown Agent Status on DAHDI |
| 1:02PM |
2 |
DAHDI Error |
| 9:46AM |
0 |
Asterisk 1.8 realtime tables. |
| 4:54AM |
0 |
Blocking multiple SIP registration |
| 4:38AM |
1 |
1.8.4 Core Dump after installing from source |
| 3:08AM |
1 |
asterisk 1.8 + google voice |
| 1:44AM |
1 |
undefined symbol: cap_set_proc on several modules after installation from source |
| |
| Thursday May 12 2011 |
| Time | Replies | Subject |
|---|
| 8:57PM |
0 |
asterisk 1.8 somehow dead |
| 8:27PM |
1 |
how to reload agents.conf ? |
| 7:40PM |
1 |
lead time for RPM's? |
| 7:29PM |
0 |
regarding core modules |
| 6:16PM |
1 |
Problem with PSTN calls (Asterisk as SIP client on embedded device) |
| 4:50PM |
2 |
Realtime - ara180 |
| 4:50PM |
8 |
Light indicator managed by Asterisk |
| 2:33PM |
3 |
ConfBridge for 1.8 ? |
| 1:11PM |
0 |
log full of Name or service not known |
| 10:07AM |
1 |
multiple calls into hold |
| 7:40AM |
0 |
Friday VUC: Discussion of Mobile SIP, Microsoft Lync |
| 7:10AM |
0 |
About minimum requirements to install PSTN GW+SIP Client |
| 5:57AM |
1 |
Different IP addresss for SIP and RTP |
| 1:54AM |
1 |
Disabling echo cancellation by software |
| 12:29AM |
1 |
Higher CPU usage on 1.6.1 than 1.4? |
| |
| Wednesday May 11 2011 |
| Time | Replies | Subject |
|---|
| 5:30PM |
2 |
Asterisk SIP Trunking with Cisco UC 560 |
| 5:21PM |
0 |
kernel: dahdi: Master changed to B4/0/x |
| 4:57PM |
4 |
concurrent call tracking |
| 3:48PM |
1 |
CLI - displaying all channel variables |
| 3:06PM |
1 |
With what options is asterisk compiled in rpm's |
| 12:04PM |
2 |
no audio with SIP:INFO in meetme |
| 7:16AM |
0 |
obd call drops after few seconds : only for mobile numbers |
| |
| Tuesday May 10 2011 |
| Time | Replies | Subject |
|---|
| 8:23PM |
14 |
When someone helps you, at least let them know if the problem is resolved or not |
| 7:27PM |
1 |
iax2 Max retries exceeded to host |
| 4:50PM |
2 |
About X100P and TDM400P analog card in China |
| 3:37PM |
1 |
Using MixMonitor() |
| 2:38PM |
2 |
Asterisk 1.8.4 Now Available |
| 12:21PM |
1 |
Plotting fxotune dump |
| 1:57AM |
2 |
1.8 and prematuremedia problem |
| 1:12AM |
1 |
ITSP Multi IPs |
| |
| Monday May 9 2011 |
| Time | Replies | Subject |
|---|
| 9:03PM |
0 |
Call ends when using SendDTMF(*) |
| 9:00PM |
1 |
Voicemail Configuration |
| 8:40PM |
3 |
Really, really loud ringers |
| 7:53PM |
1 |
Need help defining a stackexchange (i.e. stackoverflow) for telephony |
| 6:28PM |
2 |
Rates Importer Tool |
| 5:28PM |
0 |
high PDD |
| 5:25PM |
1 |
iax2 issue in asterisk |
| 3:48PM |
0 |
Free Alarms sound |
| 2:32PM |
3 |
asterisk syntax highlighting for gedit |
| 1:11PM |
4 |
Trying out a new version with sangoma card |
| 1:01PM |
0 |
Ustream feed as MOH |
| 12:26PM |
5 |
40sec between dial execution and sending SIP request |
| 12:10PM |
2 |
OT - Which Android handset with Wifi-only ? |
| 10:49AM |
0 |
conf syntax highlighting for gedit |
| 9:22AM |
0 |
RTP Path and t or T option |
| 8:03AM |
3 |
OUTBOUND CALLER ID |
| 7:47AM |
4 |
Slightly OT: Android phone as sip-gw? |
| 7:41AM |
3 |
how to play music when dial fail or time out |
| |
| Sunday May 8 2011 |
| Time | Replies | Subject |
|---|
| 9:19PM |
1 |
Cisco 7940 phone and tftpd provisioning - for ever? |
| 2:31PM |
0 |
txgain no effect |
| 11:43AM |
1 |
no ringback tone on outgoing call PRI line |
| 12:59AM |
3 |
Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS |
| |
| Saturday May 7 2011 |
| Time | Replies | Subject |
|---|
| 11:08PM |
0 |
asterisk-users Digest, Vol 82, Issue 27 |
| 5:04PM |
3 |
record call from iax to sip |
| 3:24PM |
3 |
[SOT] Virtualising Asterisk |
| 1:05AM |
1 |
Tricky: Progress, Delay, DTMF / background calling |
| |
| Friday May 6 2011 |
| Time | Replies | Subject |
|---|
| 8:51PM |
1 |
Blacklist with *30 |
| 6:48PM |
1 |
Supermicro X7SPE (Atom) as Asterisk server? |
| 6:14PM |
3 |
question on ways to activate voicemail light on polycom |
| 5:49PM |
0 |
polycom page custom ring |
| 5:03PM |
0 |
Gateway GSM x Comercio Indevido ? |
| 4:49PM |
3 |
Configuring Voicemail in Asterisk 1.8 |
| 4:11PM |
1 |
QueueCallerAbandon is not triggering after 1.8.3.3... |
| 3:58PM |
2 |
Cannot install dahdi-linux on (old) PAE kernel. |
| 3:32PM |
0 |
Cannot built kmod-dahdi-linux for PAE kvariant from SRPM |
| 3:30PM |
7 |
Background music during a call |
| 3:29PM |
0 |
Missed call when call is answered by other phone |
| 3:22PM |
1 |
Asterisk 1.6.2.18, Cisco 79XX not registering |
| 1:04PM |
3 |
TCP Trigger on incoming call request |
| 8:33AM |
1 |
is res_timing_timerfd module stable in 1.8? |
| 4:37AM |
0 |
Audiocodes MP-114 - modem dial not going through |
| 2:54AM |
0 |
how to let the call play audio when the dial fail |
| |
| Thursday May 5 2011 |
| Time | Replies | Subject |
|---|
| 10:51PM |
0 |
feedback mechanism |
| 9:14PM |
1 |
ael context ~~s~~ in macros broke Dial() U() option in 1.6.2.17.2 and newer |
| 7:11PM |
1 |
Why is PQMSTATUS empty? |
| 6:41PM |
1 |
estimated queue hold time |
| 4:36PM |
1 |
asterisk for g729 to g711 |
| 3:51PM |
0 |
Asterisk 10 / Trunk and RecieveFax "F" Option |
| 3:33PM |
1 |
Queues, pickup and transfers |
| 3:10PM |
0 |
Does IAX2 support call completion or callback ? |
| 2:13PM |
5 |
Asterisk 1.8 latest branch safe for production ? |
| 1:30PM |
1 |
Auto dialing Polycoms and other SIP phones |
| 12:46PM |
2 |
[Asterisk 1.8.3.2] Mixmonitor not working on member(calling part) channel of Queue. |
| 12:08PM |
4 |
SIP secruity: username and password |
| 10:10AM |
0 |
Could not place calls through IAX |
| 7:49AM |
0 |
Voicemail message storage in db w/o ODBC? |
| 3:37AM |
3 |
Issue with Asterisk & Aastra 57i at v3.2 |
| |
| Wednesday May 4 2011 |
| Time | Replies | Subject |
|---|
| 10:24PM |
0 |
Park a call when sip phone becomes unreachable? |
| 10:21PM |
3 |
Cordless VoIP Phones and Access Point hand-off? |
| 5:12PM |
3 |
asterisk-1.8 crash if no extension specified in Dial |
| 5:10PM |
2 |
Remove "name" part of SIP From header |
| 5:01PM |
2 |
Sangoma A400 background noise after a while |
| 4:01PM |
0 |
Compiling extra modules |
| 1:42PM |
1 |
pickup question |
| 9:56AM |
1 |
Invalid use of undefined type when make dahdi |
| 7:43AM |
2 |
asterisk HA for queue calls |
| 1:55AM |
1 |
asterisk 1.4.35 to 1.4.41 |
| |
| Tuesday May 3 2011 |
| Time | Replies | Subject |
|---|
| 9:31PM |
1 |
Having redundancy, so if first IP failed then send for the other |
| 7:20PM |
2 |
receive faxes |
| 6:13PM |
3 |
asterisk call forwarding |
| 5:57PM |
2 |
dial from voicemail |
| 4:43PM |
1 |
Asterisk 1.6 Questions |
| 3:52PM |
0 |
record call transfered from IAX softphone to SIP hardphone |
| 2:32PM |
2 |
Multiple cards using same IRQ - getting IRQ errors and hissing |
| 2:03PM |
1 |
audiohook.c: Failed to get 160 samples from write factory |
| 1:28PM |
0 |
asterisk 1.8 rpms and additional modules from source |
| 12:34PM |
1 |
Asterisk, bicolor BLF and DEVSTATE |
| 10:16AM |
2 |
Fading voice problem |
| 5:10AM |
1 |
How to debug MixMonitor misbehaviour |
| |
| Monday May 2 2011 |
| Time | Replies | Subject |
|---|
| 11:50PM |
7 |
ATA refuses to answer a call? |
| 8:56PM |
1 |
sip busy detect |
| 5:19PM |
4 |
asterisk call completion issue |
| 5:01PM |
0 |
music on hold skipping |
| 1:59PM |
2 |
Retrieving/Streaming audio/video files from DB using over AGI |
| 12:40PM |
1 |
Asterisk repository: asterisk14-addons-mysql |
| 11:33AM |
3 |
out of the blue one way audio |
| 10:09AM |
1 |
default context overrides context of peer |
| 7:20AM |
0 |
queue member invalid |
| 7:15AM |
1 |
Retrieving sound files from DB as opposed to filesystem |
| |
| Sunday May 1 2011 |
| Time | Replies | Subject |
|---|
| 6:46PM |
4 |
Odd error in libpri |
| 10:41AM |
1 |
Join and listen to conference call through web-interface |
| 10:20AM |
1 |
Queue Setup |