| Friday April 30 2010 |
| Time | Replies | Subject |
|---|
| 5:23PM |
1 |
Embedded IAX |
| 3:18PM |
2 |
B400P card crashes conncection |
| 2:02PM |
1 |
Fwd: Re: SpiderMux? |
| 1:16PM |
0 |
get hold event |
| 12:26PM |
1 |
GXW4024 |
| 11:20AM |
1 |
HDLC Receiver overrun on Wildcard TE410P |
| 10:51AM |
0 |
IAX trunks and audio codecs |
| 9:28AM |
0 |
Problems with t38modem and bitrate sent to t38-termination service |
| 8:57AM |
1 |
Call-Waiting, implementation ideas |
| 8:01AM |
5 |
Asterisk and Patton |
| 5:30AM |
0 |
Caller ID on Asterisk and Astribank |
| 5:10AM |
0 |
Friday 12 Noon EDT: Media5fone Mobile SIP Client Symbian S60 & iPhone |
| 3:55AM |
2 |
Continuing after a TIMEOUT(absolute) |
| |
| Thursday April 29 2010 |
| Time | Replies | Subject |
|---|
| 8:33PM |
3 |
Calls Dropping |
| 7:59PM |
2 |
Asterisk stopping for no reason |
| 7:36PM |
4 |
ATA shootout: PAP2T versus Grandstream Handytone 286 |
| 7:02PM |
2 |
Code in extensions.conf to leave a voice mail in another PBX ?! |
| 5:45PM |
0 |
Odd Issue With Polycom Phones] |
| 3:25PM |
1 |
Strange Invite issue |
| 1:51PM |
1 |
Dropping incompatible voice frame |
| 12:56PM |
1 |
incoming call should ring on several dahdi channels |
| 12:06PM |
0 |
Polycom 330 not connecting |
| 10:01AM |
1 |
Starting call recording using a dynamic feature to call a macro |
| 9:53AM |
0 |
mysql realtime schema |
| 9:02AM |
8 |
AGI <==> DeadAGI |
| 8:47AM |
1 |
Asterisk Query |
| 7:20AM |
1 |
SpiderMux? |
| 2:35AM |
1 |
Issue with (pattern) matching extension |
| 2:12AM |
2 |
No change in payload. (SDP) |
| 1:57AM |
1 |
Duplicated DTMF with bridged IAX channels maybe? |
| |
| Wednesday April 28 2010 |
| Time | Replies | Subject |
|---|
| 7:29PM |
1 |
Strange Error -- ASterisk 1.6 |
| 5:20PM |
0 |
sip jitter buffer |
| 5:14PM |
1 |
simple dialplan question |
| 3:59PM |
0 |
asterisk core dumps after cdr database writes using odbc |
| 2:37PM |
2 |
Gateway E1 <=> Asterisk ? |
| 1:44PM |
0 |
command-line dialplan "compiler" |
| 1:10PM |
0 |
Execute Macro when queue is answered |
| 12:49PM |
2 |
dialplan |
| 8:50AM |
6 |
Asterisk 1.4.30 is slow sending STDIN to AGI script |
| 5:12AM |
2 |
BN8S0, dahdi, wcb4xxp |
| 1:48AM |
6 |
Dial plan question. |
| 12:15AM |
2 |
Broadvoice inbound fails on Asterisk 1.6.1 |
| |
| Tuesday April 27 2010 |
| Time | Replies | Subject |
|---|
| 4:06PM |
0 |
callprogress issue |
| 2:53PM |
2 |
Record call without caller interference |
| 2:41PM |
5 |
E3 Card on Asterisk ? |
| 1:54PM |
2 |
Problems for Skype for Asterisk |
| 12:27PM |
2 |
Connect 2 asterisks servers |
| 11:31AM |
4 |
dialplan question |
| 10:00AM |
0 |
Message notification without MWI |
| 2:12AM |
0 |
Redone setup, bizare problems |
| |
| Monday April 26 2010 |
| Time | Replies | Subject |
|---|
| 9:40PM |
1 |
Building Asterisk-RPM for 1.4.24.1 |
| 8:23PM |
3 |
Inbound route question |
| 8:11PM |
0 |
DTMF from SIP phone to FXS/FXO |
| 6:21PM |
1 |
Dahdi will not compile on Unbuntu Studio Linux 9.10 (Karmic) 32bit |
| 5:32PM |
0 |
Taqua users out there? |
| 4:45PM |
0 |
problem of registration with Asterisk using exosip2 |
| 4:35PM |
0 |
SIP authentication |
| 3:51PM |
0 |
How to disable dialog-info based call pickups (Was: Re: 1.6.2 - Pickup and SIP Replaces header) |
| 3:16PM |
1 |
1.6.2 - Pickup and SIP Replaces header |
| 2:37PM |
0 |
misdn accountcode? |
| 2:21PM |
2 |
[PATCH] Make Queue announcements more consistent (1.4.26.2) |
| 12:22PM |
1 |
1.6.1.18 : app_voicemail is calling sendmail without any argument |
| 10:21AM |
1 |
play a sound from the callee before putting it in connection. |
| |
| Sunday April 25 2010 |
| Time | Replies | Subject |
|---|
| 7:19PM |
0 |
CONNECTEDLINE(), progressinband=no and 183 before 180 (with latest trunk) |
| 6:24PM |
0 |
SIP gain |
| 6:21PM |
2 |
hardware clock drift and CDR |
| 5:37PM |
1 |
DAHDI Congestion cause 34 |
| 3:59PM |
1 |
How to debug the problem of Asterisk using so much of CPU percentage...? |
| 2:15PM |
0 |
Asterisk 1.6, dialplans, and IVR |
| 2:06AM |
3 |
VoIP monitoring tools |
| 1:23AM |
1 |
VOIP Monitoring tools........ |
| |
| Saturday April 24 2010 |
| Time | Replies | Subject |
|---|
| 8:51PM |
0 |
PrivacyManager |
| 6:19PM |
2 |
Asterisk and Archlinux |
| 9:30AM |
2 |
Manager events & safety |
| 3:19AM |
0 |
automatic call with call files |
| 2:59AM |
0 |
Multiple Parking Lots in Asterisk 1.6.x |
| 1:36AM |
2 |
Asterisk not recognizing ACK from an OK message? Help debuging SIP retransmit problem |
| |
| Friday April 23 2010 |
| Time | Replies | Subject |
|---|
| 9:52PM |
1 |
What is needed to test issue 0013573: [Patch] Allow realtime_multi_ldap to behave like other realtime_multi functions |
| 7:21PM |
6 |
RTP over TCP |
| 5:11PM |
3 |
Playback all the sound files |
| 2:25PM |
0 |
VUC Friday: Bill Miller, former VP of Product Management |
| 2:36AM |
0 |
Hans Rauser |
| 1:55AM |
1 |
asterisk running @ 100% load doing nothing |
| |
| Thursday April 22 2010 |
| Time | Replies | Subject |
|---|
| 9:02PM |
4 |
More efficient dial plan for a list of selective inbound numbers |
| 7:18PM |
2 |
Swaping out phones. |
| 6:37PM |
0 |
Avaya UUI |
| 5:53PM |
1 |
Hangup after n seconds using originate ? |
| 5:31PM |
2 |
Follow-me to my answering machine :-( |
| 4:14PM |
3 |
How to do analog e&m on asterisk? |
| 8:41AM |
0 |
DAHDI User-User information "Message longer than it should be??" |
| 7:24AM |
1 |
Need to patch Asterisk for problem with FreePBX Call Confirmation |
| 1:25AM |
2 |
Security tests |
| 12:40AM |
5 |
${HANGUPCAUSE} is always 0 in the h extension |
| |
| Wednesday April 21 2010 |
| Time | Replies | Subject |
|---|
| 10:29PM |
1 |
Time difference in CSV CDR's and MySQL CDR's |
| 5:25PM |
3 |
Asterisk choking on voice messages announcements |
| 5:11PM |
1 |
Improving audio bitrate for all callers in a conference room for a podcast |
| 11:02AM |
1 |
Why app_fax.so there is no in asterisk16-1.6.2.6-1_centos5.x86_64.rpm? |
| 1:22AM |
2 |
Unable to load cdr_adaptive_odbc.so |
| |
| Tuesday April 20 2010 |
| Time | Replies | Subject |
|---|
| 9:57PM |
0 |
IBM X3650 with Asterisk??? |
| 9:31PM |
1 |
Manipulating audio in asterisk |
| 6:22PM |
0 |
SIP one-way audio |
| 6:19PM |
0 |
Initial audio dropping |
| 5:11PM |
1 |
Improving CLI Help - was [Re: 1.6.2 No "soft hangup"?] |
| 5:02PM |
4 |
How to record a call in a single file when transfered... |
| 4:18PM |
0 |
I figured it out!! |
| 4:05PM |
2 |
1.6.2 No "soft hangup"? |
| 2:59PM |
0 |
Feature Request - SoftHangup with delayed playback option |
| 2:46PM |
0 |
How to tell if a channel is on hold or not from diaplan ? |
| 1:34PM |
4 |
Voice mail "maxmessage " setting per mail box |
| 1:11PM |
2 |
Read Timeout |
| 9:49AM |
1 |
Portech MV-374 does not register |
| 8:55AM |
0 |
Dozens of SIP NOTIFY messages with unique call ID's, and the same mailbox repeated multiple times on 1.6.2.6 |
| 8:12AM |
0 |
Yesterday EC2, today Netnation Europe V.O.F. |
| 6:48AM |
1 |
Put a call on hold with Manager |
| 1:07AM |
6 |
Calls drop after 20 seconds |
| |
| Monday April 19 2010 |
| Time | Replies | Subject |
|---|
| 10:30PM |
0 |
new in Bridge(), How does it work? |
| 6:52PM |
1 |
Zap PRI failed with Cause 34 - Where to check for problems? |
| 5:46PM |
3 |
A matter of context |
| 5:13PM |
0 |
Xorcom Experience |
| 4:00PM |
1 |
Help with FastAGI server in Windows |
| 3:49PM |
0 |
Hangup after 1 second of ringing ? |
| 3:47PM |
0 |
RTP Timeouts not clearing calls |
| 3:41PM |
1 |
zapg723toslin did not update samples |
| 3:14PM |
1 |
G729 exhaustion conditions |
| 1:06PM |
5 |
Evaluating Asterisk |
| 9:22AM |
3 |
Detect if a Number is up or not |
| 8:48AM |
1 |
B400P and A1200P changes card order |
| 6:14AM |
2 |
OpenSIPS with Asterisk Backend |
| 2:38AM |
3 |
Extensions Reload | Asterisk Freezes ? 1.4 |
| |
| Sunday April 18 2010 |
| Time | Replies | Subject |
|---|
| 10:44PM |
2 |
kamailio |
| 5:59PM |
0 |
The best way to stop an ongoing call |
| 5:53PM |
1 |
meetme / upgrade to 1.6.2.6 |
| 5:43PM |
1 |
problems originating an outgoing IAX2 call |
| 12:25PM |
0 |
Asterisk-stat - Bugs |
| 12:18PM |
1 |
Slightly OT: OMA DM Solution |
| 10:10AM |
2 |
Amazon EC2 SIP floods - you can help |
| 9:10AM |
1 |
Bug or feature: cdr_odbc.conf.sample |
| 5:18AM |
0 |
VOIP at BerkeleyTIP-Global meeting on Sunday April 18 12N-3P, & April 27 |
| |
| Saturday April 17 2010 |
| Time | Replies | Subject |
|---|
| 9:20PM |
0 |
B410P and DTMF |
| 9:14PM |
2 |
Changing storm-prevention behaviour in logger.conf |
| 4:31PM |
1 |
QUICKLY - What is the command to change .wmv to Asterisk compatible format using ffmpeg? or any other tool? |
| 3:25PM |
1 |
X-lite direct sip call - Is it possible? |
| 1:38PM |
1 |
DIALSTATUS variable and qualify=no |
| 9:25AM |
1 |
Realtime changes not reflected realtime |
| |
| Friday April 16 2010 |
| Time | Replies | Subject |
|---|
| 9:39PM |
2 |
Testing a sip call through Asterisk? |
| 8:59PM |
7 |
AGI, FASTAGI or Windows Voice Server |
| 4:10PM |
1 |
On CLI SIP don't appear |
| 4:02PM |
1 |
incoming ghost call |
| 2:00PM |
2 |
Recording music in Queue |
| 1:19PM |
1 |
IAX connection slow between 1.4 and 1.6 dists |
| 10:15AM |
2 |
SS7 over an FXO interface |
| 10:09AM |
0 |
Friday April 16 @12 Noon EDT - Tim's Excellent Island Telephony Adventure, AstriEurop, and more EC2 rant |
| 8:12AM |
0 |
Spam and that recent 'attack' ... |
| 5:16AM |
2 |
How to set up Fax on Asterisk - Using analog Fax machines and HT502 (or FXS of a Digium TDM410P) |
| 5:04AM |
3 |
Delay the HungUp |
| |
| Thursday April 15 2010 |
| Time | Replies | Subject |
|---|
| 11:45PM |
0 |
moh files not playing in sort order |
| 7:25PM |
0 |
Asterisk-Addons 1.4.11, 1.6.0.5, 1.6.1.3, and 1.6.2.1 Now Available |
| 5:11PM |
1 |
SIP devide call-forward behaviour and CDRs |
| 3:45PM |
1 |
Avaya 9640 Convert to SIP (slightly off topic) |
| 2:11PM |
1 |
'o' option on Dial application |
| 1:09PM |
1 |
Asterisk/Polycom Dialed Party Name |
| 12:59PM |
1 |
Transfer_CONTEXT behaviour |
| 11:18AM |
1 |
shared lines (sla) with Asterisk 1.4.26, any hints? |
| 8:31AM |
0 |
Set CDR amaflags not work |
| 7:31AM |
0 |
say.conf implementation of Indian Languages to play numbers and dates |
| 1:55AM |
1 |
How can I record the conversations in a conference call? |
| 12:36AM |
0 |
Regarding remote registration of SIP user on zoiper |
| 12:26AM |
1 |
Queue call to specific queuemember |
| |
| Wednesday April 14 2010 |
| Time | Replies | Subject |
|---|
| 8:55PM |
2 |
[Conference] Audio/Video |
| 8:50PM |
1 |
Interpbx connection |
| 8:05PM |
0 |
Vestec vs Lumenvox |
| 5:13PM |
0 |
Sending SMS problems. |
| 5:04PM |
0 |
two FreePBX servers with load balancing |
| 4:37PM |
4 |
FastAGiin Windows Server |
| 3:44PM |
2 |
Conference Meetme |
| 2:52PM |
3 |
Converting GSM calls to SIP |
| 1:12PM |
0 |
1.6.2.6: can't upgrade from 1.6.1.18 |
| 6:47AM |
0 |
Shorewall rate limiting rules? |
| 1:06AM |
1 |
Ring Two Extensions Simultaneously with different caller ID values? |
| |
| Tuesday April 13 2010 |
| Time | Replies | Subject |
|---|
| 10:26PM |
1 |
Interesting One Way Audio |
| 10:25PM |
2 |
1.6.0 verses 1.6.2 |
| 9:02PM |
0 |
What's are the possible return values of AMI Originate when Async is set to 0? |
| 8:20PM |
0 |
SIP registration failure stops all SIP activity |
| 7:51PM |
1 |
DAHDI-Linux and DAHDI-Tools 2.3.0 Released |
| 7:19PM |
2 |
Possible AGI bug? |
| 6:51PM |
1 |
Do AMI Events have timestamps? |
| 6:17PM |
2 |
iptables miss up phone calls if not used properly |
| 6:07PM |
2 |
Time variables in system application |
| 4:56PM |
1 |
Merge .csv files |
| 4:48PM |
1 |
Using chan_lcr (and mISDN v2) ? |
| 4:46PM |
2 |
Merge master.csv files |
| 4:14PM |
0 |
Problem with Callfiles |
| 2:37PM |
1 |
SIP equivalent of zap "c" option |
| 2:22PM |
3 |
Is svn.asterisk.org down ? |
| 1:57PM |
1 |
Is restart of span a concern on PRI? |
| 1:50PM |
2 |
Full transfer details on inbound calls |
| 1:50PM |
0 |
[asterisk users] asterisk realtime - database driven dialplan |
| 12:53PM |
2 |
cat /proc/zaptel/* |
| 12:19PM |
1 |
dahdi_scan and OctoBRI. Bug or feature ? |
| 11:53AM |
0 |
Dial an extension with follow me |
| 10:46AM |
3 |
protocol used to connect Asterisk and GSM core network (MSC) |
| 9:42AM |
1 |
problem of "when memory become 50% or more then sound become noisy?" |
| 9:11AM |
2 |
SNOM M9 base station A to base station B |
| 5:54AM |
2 |
All incoming calls landing in [customers] context |
| 4:33AM |
0 |
ATA status intermittent |
| 2:49AM |
0 |
PRI Gurus ONLY - Too complex of an issue - SOLVED |
| 1:14AM |
1 |
PRI TBCT - Practical Experience, Anybody? |
| |
| Monday April 12 2010 |
| Time | Replies | Subject |
|---|
| 9:58PM |
2 |
Being attacked by an Amazon EC2 |
| 8:50PM |
1 |
Flood of REGISTERs - attack? |
| 7:22PM |
2 |
PRI Gurus ONLY - Too complex of an issue |
| 6:40PM |
1 |
Monitoring calls via sound card |
| 6:17PM |
1 |
cause 66 - Channel not implemented |
| 6:02PM |
0 |
Subscribe to a MWI when acting as a SIP client? |
| 5:19PM |
2 |
Asterisk room monitor |
| 4:16PM |
2 |
Dahdi, junghanns and qozap |
| 3:50PM |
1 |
Change in menuselect handling of sound files (in 1.6.1.X) |
| 9:55AM |
0 |
Outgoing routes with two PRI |
| |
| Sunday April 11 2010 |
| Time | Replies | Subject |
|---|
| 8:57PM |
2 |
mISDN installation via yum |
| 8:55PM |
0 |
Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCode |
| 5:45PM |
1 |
Asterisk in Debian/Lenny without Junghanns.net support? |
| 3:24AM |
1 |
over running my did's |
| |
| Saturday April 10 2010 |
| Time | Replies | Subject |
|---|
| 10:54PM |
1 |
Remote registering fails |
| 10:24PM |
0 |
How Cisco ATA 186 through SCCP with skinny.conf ?! |
| 9:34PM |
10 |
Being attacked by an Amazon EC2 ... |
| 8:14PM |
1 |
Asterisk script to repeat dial of a number |
| 7:46PM |
2 |
PRI - Native ZAP bridge fails - Is this my patch? |
| 4:50PM |
1 |
Asterisk + DRBD Performance |
| 3:50PM |
2 |
Sending RTP media to a different server than SIP Signaling |
| 1:35PM |
1 |
Repeated: Got SIP response 489 "Bad event" back from |
| 10:32AM |
2 |
t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED |
| 12:17AM |
1 |
Fax Over PRI connected to a Sangoma card - Fax machines connected to Sip Mediant AudioCodes |
| |
| Friday April 9 2010 |
| Time | Replies | Subject |
|---|
| 11:52PM |
3 |
Problems with Fax over TDM410P |
| 9:40PM |
2 |
res fax help |
| 7:27PM |
1 |
Callerid over IAX Trunks |
| 7:26PM |
2 |
Asterisk & Timezones |
| 12:34PM |
5 |
run script after completed |
| 11:10AM |
3 |
scratchy sound |
| 8:03AM |
0 |
Friday Apr 9th 2010 @ 12 Noon EDT: More Cloud Telephony |
| 6:21AM |
0 |
SNOM M9 : expand range |
| 2:33AM |
1 |
asterisk-users Digest, Vol 69, Issue 16 |
| 1:54AM |
2 |
tones detection |
| |
| Thursday April 8 2010 |
| Time | Replies | Subject |
|---|
| 11:48PM |
2 |
IVR menu sound processing for AMR and GSM + live test available |
| 8:30PM |
3 |
long return times from System() calls with 1.6.2.6? |
| 3:28PM |
3 |
jitterbuffer |
| 2:09PM |
1 |
Linksys/Sipura SPA-3201 FXO/FSA with Asterisk |
| 1:38PM |
0 |
MeetMe Options with S(10)L(100) |
| 12:37PM |
0 |
realtime jitter/latency measurements |
| 11:25AM |
1 |
Asterisk 1.4.26.2 died after 80 days uptime |
| 11:00AM |
3 |
dial extension and play sound file from shell on asterisk server? |
| 9:28AM |
0 |
OT - S450ip and R-key transfer |
| 8:28AM |
0 |
Opportunity to author Asterisk books- Packt Publishing. |
| 6:35AM |
2 |
Split E1 ISDN service for another device. |
| 2:44AM |
2 |
Need help with a pika warp asterisk appliance problem. |
| |
| Wednesday April 7 2010 |
| Time | Replies | Subject |
|---|
| 9:23PM |
1 |
Rebooting Polycom's - Could not create address for 'XXXX' |
| 6:30PM |
3 |
URGENT - How to exclude one ZAP channel for outgoin and incoming calls |
| 5:07PM |
1 |
Agent Callback methods? |
| 3:26PM |
3 |
D-Channel Span Up without Down |
| 3:07PM |
3 |
PSTN issues |
| 1:30PM |
2 |
AGI + Dial + stream file ? |
| |
| Tuesday April 6 2010 |
| Time | Replies | Subject |
|---|
| 8:50PM |
0 |
Busy(20) returns non-zero and exits immediately on IAX channel |
| 8:34PM |
0 |
dialplan checker |
| 3:56PM |
1 |
testexpr2 |
| 2:02PM |
1 |
IAX Problem |
| 1:07PM |
2 |
Limit Number Of Simultaneous Outbound SIP Calls |
| 12:37PM |
1 |
SIP Dialplan Failover Solution |
| 12:02PM |
2 |
polarity reverse |
| 8:26AM |
2 |
Anyone coming to Paris next week for AstriEurope? |
| 8:16AM |
1 |
Which rule for Asterisk to Asterisk-addons compatibility ? |
| 4:36AM |
2 |
Cache sound files for faster processing |
| 2:37AM |
1 |
OT: Wireless headset / phone combination |
| |
| Monday April 5 2010 |
| Time | Replies | Subject |
|---|
| 10:07PM |
1 |
Debug help |
| 9:51PM |
2 |
Access denied for user 'a2billinguser |
| 8:54PM |
1 |
Please sign Petition - Stop Child Labour |
| 7:30PM |
0 |
SIP Outdial Not Detecting Ringing Line |
| 6:55PM |
2 |
spool directories and filename |
| 5:45PM |
1 |
External Extension with Extension |
| 4:14PM |
2 |
call files in 1.6 |
| 2:28PM |
5 |
Continuous bothering message -- Remote UNIX connection disconnected |
| 1:21PM |
1 |
trying app_fax.c |
| |
| Sunday April 4 2010 |
| Time | Replies | Subject |
|---|
| 12:06PM |
1 |
[OT] phpagi help |
| |
| Saturday April 3 2010 |
| Time | Replies | Subject |
|---|
| 12:14PM |
2 |
asterisk 1.6.1/1.6.2 binary packages |
| |
| Friday April 2 2010 |
| Time | Replies | Subject |
|---|
| 3:53PM |
0 |
change IP address of AA 50? |
| 3:41PM |
2 |
asterisk start with php |
| 2:11PM |
2 |
How set debug file for RxFax application |
| 1:50PM |
1 |
Asterisk and spandsp fax problem |
| 1:02PM |
3 |
Asterisk send calls to SIP Trunks with Round Robin Call Distribution |
| 10:23AM |
1 |
RTCP How to stop |
| 7:26AM |
1 |
Gosub replacement within AEL2 dialplans |
| 6:50AM |
1 |
Strange Centos Problem with Dahdi installation |
| 5:20AM |
0 |
there have any one run asterisk on ubuntu enterprise cloud ? |
| |
| Thursday April 1 2010 |
| Time | Replies | Subject |
|---|
| 10:50PM |
1 |
SIP Connection Question |
| 9:05PM |
2 |
canary_thread |
| 9:01PM |
0 |
Question about MaxRetries in the Asterisk Outgoing folder |
| 7:18PM |
1 |
asterisk-gplonly dependency in asterisk-addons RPM |
| 6:36PM |
2 |
problem compiling asterisk with cdr_odbc |
| 5:50PM |
3 |
Exceptionally long voice queue length errors... |
| 2:36PM |
3 |
RPID on called party |
| 1:15PM |
2 |
Problem with Sangoma A104 and euroisdn pri |
| 12:28PM |
1 |
Asterisk Load Balancing with Redfone/TDMoE driver |
| 9:46AM |
0 |
OfficeSIP Communications Makes Its VoIP SIP Products Open Source |
| 5:53AM |
7 |
Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8 |