| Monday April 30 2007 |
| Time | Replies | Subject |
|---|
| 7:59PM |
0 |
Segfaults in 1.2.18 |
| 6:08PM |
1 |
Re: Voicemail on Different Server (MySQL Replication split thread) |
| 5:14PM |
4 |
Zaptel kernel module load order |
| 4:05PM |
5 |
Asterisk 1.4.4 VoiceMail ODBC Storage Help |
| 1:51PM |
2 |
Improving Asterisk's DNS support |
| 11:38AM |
2 |
CDR and Billing Issue |
| 11:33AM |
1 |
IVR dictionary dial-plan |
| 10:37AM |
1 |
automatically close a meetme |
| 10:30AM |
2 |
Confference function |
| 10:25AM |
3 |
ZAPTEL PROBLEM |
| 10:13AM |
2 |
Send Variable in Dial |
| 9:38AM |
1 |
Simple dial plan inquiry |
| 8:22AM |
0 |
voicemail + Dynamic mailbox |
| 6:52AM |
0 |
Remodified Asterisk brute force blockers.. |
| 5:29AM |
1 |
SugarCRM, NO!, Foxpro, SI? |
| 4:25AM |
1 |
TDM400P and Junghanns QuadBRI issue |
| 3:47AM |
2 |
don't want call to get answered |
| 2:56AM |
0 |
Priority in ACD |
| |
| Sunday April 29 2007 |
| Time | Replies | Subject |
|---|
| 7:17PM |
2 |
Polycom 650 |
| 12:57PM |
1 |
100 users - voip lan security and qos ? |
| 10:08AM |
0 |
asterisk 1.4 and zap channel flash |
| 9:26AM |
2 |
Early audio(progress) and MOH |
| 6:00AM |
2 |
Polycom 430 , 501 and 550 |
| 5:34AM |
0 |
Unable to find a codec translation path from ilbcto ulaw |
| 5:29AM |
4 |
Wildcard TDM11B & Wildcard TDM04B |
| 3:52AM |
1 |
Voicemail Creation |
| |
| Saturday April 28 2007 |
| Time | Replies | Subject |
|---|
| 10:08PM |
0 |
app_dictate problems |
| 2:22PM |
8 |
Poor man's High Availability solution |
| 1:48PM |
1 |
Viable using purchasing sip lines |
| 10:33AM |
4 |
Trixbox/FreePBX |
| 8:22AM |
2 |
ADSL routers with integrated SIP QoS for other devices |
| 1:05AM |
7 |
Two Connected Servers Sound Quailty |
| |
| Friday April 27 2007 |
| Time | Replies | Subject |
|---|
| 9:21PM |
2 |
Music on Hold issue with asterisk 1.4.2 |
| 5:22PM |
2 |
Call Pick Up |
| 3:56PM |
0 |
Asterisk 1.4.4 Released |
| 1:18PM |
0 |
chan_bluetooth as FXS? |
| 12:36PM |
1 |
execute commands after hangup |
| 11:55AM |
1 |
Free seating Agents and logged in / logged out indication |
| 11:00AM |
4 |
Fixed quantity calls per extension |
| 10:59AM |
1 |
New VICIDIAL astGUIclient Release: 2.0.3 |
| 10:34AM |
0 |
Problems with Digium TE110P |
| 10:28AM |
0 |
Re: Voicemail on Different Server, Voicemail with NFS |
| 9:59AM |
0 |
Live conference call on now |
| 9:58AM |
1 |
SIP<->H323 calls without proxying RTP |
| 9:37AM |
1 |
Problem of configuring musiconhold.conf file |
| 9:07AM |
4 |
Unable to find a codec translation path from ilbc to ulaw |
| 8:35AM |
0 |
zaptel/pri, early audio, dial() |
| 8:32AM |
2 |
CDR changes in 1.4.3? |
| 8:01AM |
0 |
Utilisation of multiple database tables in Asterisk |
| 7:24AM |
4 |
Best Wireless bridge for Polycoms |
| 6:36AM |
1 |
How to configure a stun server for a sip peer |
| 5:55AM |
1 |
Asterisk hosted Callwaiting??? |
| 1:17AM |
1 |
2 cards in a server |
| 1:09AM |
1 |
can´t anserd the call |
| 12:29AM |
0 |
Attended Transfer of a queue call fails |
| |
| Thursday April 26 2007 |
| Time | Replies | Subject |
|---|
| 7:02PM |
0 |
E1 Card recommendation |
| 6:52PM |
1 |
Asterisk 1.2.14 will not run without internet connection |
| 5:31PM |
1 |
Re: Voicemail on Different Server, Voicemail with NFS |
| 2:40PM |
0 |
FYI - PRS fraud |
| 2:10PM |
1 |
7970 sip success |
| 1:38PM |
1 |
How does Realtime read config files? |
| 1:20PM |
1 |
Call prority (QUEUE_PRO) in the queues |
| 1:10PM |
0 |
Static in Audio PRI, Got reject for frame 39, retransmitting frame 39 now, updating n_r! |
| 12:52PM |
0 |
problem with A400P01 OpenVox |
| 12:01PM |
0 |
Potential Consulting Opportunity |
| 11:48AM |
1 |
AsteriskNOW generation of zapata.conf file |
| 10:45AM |
1 |
IVR Testing |
| 10:44AM |
1 |
App Read() kills call when file doesn't exist |
| 10:07AM |
1 |
Asterisk cookbook |
| 9:18AM |
3 |
Two devices registrating same extension |
| 8:46AM |
2 |
MeetMe + IAX2 + Asterisk 1.2.18 = calls dropped |
| 8:30AM |
1 |
[OT] How to find out line that you are on from Bezeq |
| 8:20AM |
2 |
Changing Voice from Male to Female |
| 7:59AM |
1 |
Asterisk in support enviroment, some CTI maybe & RT integration? |
| 7:51AM |
1 |
Asterisk IVR and Call Center Agents |
| 7:13AM |
0 |
take-over && transfer question |
| 7:01AM |
1 |
Can asterisk record the duration of users putting on hold? |
| 6:47AM |
0 |
FW: ChanSpy and MeetMe |
| 6:38AM |
1 |
Asterisk Voice sound level |
| 6:37AM |
1 |
asterisk slows down when unplugging internet cable with VoIP lines |
| 5:55AM |
4 |
headsets for linksys/sipura phones? |
| 5:44AM |
1 |
Asterisk brute force watcher (was FYI) |
| 3:59AM |
0 |
Asterisk IAX Configuration Problem |
| 2:21AM |
1 |
Cisco 7920 sccp |
| 2:21AM |
1 |
analog line cards / adapter |
| 1:00AM |
0 |
Open Source Softphone |
| 12:21AM |
0 |
IAX channel unreliable with multiple hops |
| |
| Wednesday April 25 2007 |
| Time | Replies | Subject |
|---|
| 8:09PM |
1 |
Asterisk 1.4 Conference with G.722 |
| 6:58PM |
4 |
Too many open files, asterisk crash |
| 5:59PM |
1 |
asterisk answering machine |
| 3:13PM |
2 |
No Audio with SIP to only one provider when switching servers |
| 3:12PM |
3 |
call dispatching - legacy application |
| 1:36PM |
2 |
Polycom Provisioning Problems |
| 12:07PM |
1 |
prob with install on ubuntu linux |
| 11:51AM |
0 |
Error compiling Zaptel on CentOS 5 |
| 10:56AM |
0 |
Problems to transfer calls when it is ringing |
| 10:53AM |
3 |
How to check my voice mail from outside landline? |
| 10:34AM |
0 |
ZOOM 5806 ATA |
| 10:20AM |
0 |
Zaptel 1.4.2.1 Released |
| 10:19AM |
0 |
Zaptel 1.2.17.1 Released |
| 10:18AM |
1 |
Asterisk-addons 1.4.1 Released |
| 10:17AM |
0 |
Asterisk-addons 1.2.6 Released |
| 10:16AM |
0 |
Asterisk 1.4.3 Released |
| 10:15AM |
0 |
Asterisk 1.2.18 Released |
| 9:58AM |
4 |
Asterisk 1.4.3 segfaults on receiving calls. |
| 8:55AM |
3 |
FYI |
| 8:47AM |
3 |
SLA Appearance between 2 Cisco 7960's (SIP) |
| 7:23AM |
0 |
Call Parking is slow with park orbit on Snom 3xx / 360 |
| 7:02AM |
2 |
My Polycom IP 501 is formatted its file systemitself |
| 5:03AM |
1 |
agi and transfer |
| 4:21AM |
2 |
dialplan / problem with extension-length > 1 |
| 3:49AM |
0 |
Asterisk Users Conference Friday 12:30 PM EDT |
| 2:44AM |
5 |
Asterisk Business Edition Question |
| 2:40AM |
0 |
OriginateResponse 'reason' property. |
| 12:57AM |
1 |
Problem with SuSe 10.0 and zaptel 1.2.17 |
| 12:27AM |
1 |
Calllog |
| 12:20AM |
1 |
German voiceprompts for 1.4 available |
| |
| Tuesday April 24 2007 |
| Time | Replies | Subject |
|---|
| 10:52PM |
1 |
My Polycom IP 501 is formatted its file system itself |
| 9:44PM |
0 |
7970 Config Question |
| 8:45PM |
0 |
Re: agi timeout......clarification |
| 6:17PM |
2 |
Funky BIND/named errors |
| 5:28PM |
2 |
Voicemail on Different Server |
| 5:25PM |
2 |
Asterisk & Pix firewalls |
| 5:20PM |
1 |
E&M Wink start problem |
| 4:24PM |
0 |
ASA-2007-012: Remote Crash Vulnerability in Manager Interface |
| 4:24PM |
2 |
Random Asterisk deaths |
| 4:22PM |
0 |
Queue: SIP status not set to busy |
| 4:22PM |
0 |
ASA-2007-011: Multiple problems in SIP channel parser handling response codes |
| 4:21PM |
0 |
ASA-2007-010: Two stack buffer overflows in SIP channel's T.38 SDP parsing code |
| 4:09PM |
0 |
Asterisk Project Security Adivsory Process |
| 3:49PM |
0 |
app_dictate playback problems |
| 2:47PM |
0 |
agi timeout |
| 2:43PM |
1 |
SER/OpenSER, I Finally Get It.............General Observation |
| 2:14PM |
1 |
dundi problem * 1.4.2 |
| 1:47PM |
4 |
Marketing 101 |
| 12:13PM |
0 |
Free agent while are waiting calls |
| 10:29AM |
0 |
agentcallback login kicking agents out after call completion. |
| 9:35AM |
0 |
7960G + Asterisk auto attendant |
| 9:33AM |
2 |
Make an iso image or a kickstart-Really its too urgent |
| 9:24AM |
2 |
Call Connection Problem |
| 9:23AM |
1 |
TE412P (T1/E1+DSP) digium card cause server crash |
| 8:51AM |
1 |
Polycom SP 601 Reboot Issue- Help! |
| 8:33AM |
0 |
AstLinux 0.4.5 released |
| 7:50AM |
0 |
can't cancel call conference when invited by asterisk |
| 7:44AM |
0 |
Snom 360 Caller ID in missed / recieved calls |
| 7:41AM |
0 |
3 way calls and meetme problem |
| 6:50AM |
6 |
Digium card sale |
| 6:46AM |
5 |
tone generation |
| 4:50AM |
0 |
Hylafax EE and T.38 |
| 3:32AM |
3 |
auto dial out multiple destinations |
| 2:43AM |
2 |
Asterisk Problem |
| 2:00AM |
1 |
ast 1.2.x - cisco 7970 behind nat to external asterisk with no nat |
| 1:21AM |
6 |
help please |
| |
| Monday April 23 2007 |
| Time | Replies | Subject |
|---|
| 11:14PM |
1 |
Request for Configration details |
| 6:37PM |
1 |
problem when using Dial(Local/extension@context) |
| 6:32PM |
2 |
auto load error in asterisk cli |
| 5:43PM |
1 |
Linking asterisk servers |
| 4:57PM |
1 |
Trixbox 2 and MFC/R2 |
| 1:53PM |
0 |
Crackly Prompts but Voice OK |
| 1:47PM |
4 |
Missing "dialplan" commands in Asterisk 1.4.2 CLI |
| 1:37PM |
1 |
Purchasing a Sangoma A102 - should I get the hw echo cancellation or not? |
| 12:38PM |
0 |
End User guide |
| 10:40AM |
2 |
ztdummy |
| 9:01AM |
3 |
echo cancellation and ztdummy |
| 7:22AM |
0 |
Hardware Compatibility list |
| 7:11AM |
1 |
Make an iso image or a kickstart |
| 6:29AM |
1 |
Asterisk+mISDN drops calls after 3-4 secs |
| 6:25AM |
1 |
A400P01 from OpenVox |
| 6:20AM |
2 |
Billion ISDN problem |
| 6:09AM |
1 |
Asterisk codecs retranslation |
| 6:06AM |
4 |
SIP devices with packet loss tolerance |
| 6:02AM |
1 |
polycom boot server... |
| 5:28AM |
3 |
chan_zap not compiling. |
| 5:17AM |
0 |
Pass-thru |
| 4:57AM |
1 |
problem with 3-way conferenicing |
| 3:30AM |
3 |
Asterisk on Debian Etch |
| 3:00AM |
1 |
app_rxfax produces "RTP: Received packet with bad UDP checksum" |
| 2:16AM |
5 |
Asterisk dialing next extension only if first is busy? |
| 2:12AM |
1 |
Internet gateway problem |
| 2:05AM |
1 |
Microsoft Dynamics CRM 3.0 Integration with Asterisk |
| |
| Sunday April 22 2007 |
| Time | Replies | Subject |
|---|
| 8:16PM |
3 |
Asterisk & M$ SQL Server |
| 7:44PM |
0 |
Kvin's g.723-gcc4 and asterisk 1.4.1 |
| 4:49PM |
1 |
SLES? |
| 1:52PM |
1 |
Extension and language for users/registered ends |
| 1:49PM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 102 |
| 11:57AM |
2 |
Digium h/w serial numbers |
| 3:24AM |
1 |
Exten Length |
| 12:41AM |
0 |
Incoming SIP callerid |
| |
| Saturday April 21 2007 |
| Time | Replies | Subject |
|---|
| 10:54AM |
2 |
Apple IPhone mobile is released in India? |
| 10:50AM |
1 |
Asterisk config for Apple IPhone |
| 8:00AM |
1 |
UK zaptel and zapata.conf for TDM400P |
| 7:27AM |
1 |
Transer calls hitting # |
| 2:36AM |
3 |
FAX on PRI and TE205P |
| 12:52AM |
0 |
hints howto reset if wrong subscription Asterisk 1.2 |
| |
| Friday April 20 2007 |
| Time | Replies | Subject |
|---|
| 5:21PM |
1 |
Voicemail to Text Transcription(was: Re: [asterisk-dev] Voicemailto text translation) |
| 3:22PM |
3 |
why do I get this message |
| 2:43PM |
3 |
Developing Marketing materials ... |
| 2:33PM |
2 |
Queue problems |
| 2:07PM |
2 |
Big trouble with zap lines |
| 1:59PM |
0 |
FW: Asterisk & PiX devices |
| 1:58PM |
2 |
Asterisk & PiX devices |
| 1:14PM |
0 |
7921G running linux |
| 12:24PM |
1 |
VPM450: Not Present |
| 11:01AM |
6 |
How can I improve call quality? |
| 10:27AM |
1 |
Polycom Phones |
| 8:58AM |
0 |
Call queue problem |
| 8:48AM |
10 |
Softphone that supports central provisioning? |
| 8:26AM |
1 |
adding second TDM400P card causes echo cancellation to fail for all Zap channels |
| 8:25AM |
1 |
iaxComm problems |
| 8:10AM |
0 |
Pika boards - anyone are using it? |
| 7:50AM |
0 |
Polycom not picking up phone transferred phone call. |
| 7:26AM |
1 |
Why duoble digits must be so fast to activate features? |
| 7:24AM |
0 |
Agents.conf feature replication using addqueuemember |
| 6:57AM |
1 |
CallerID Auth |
| 6:28AM |
2 |
Asterisk stops responding to SIP/ZAP |
| 6:07AM |
3 |
Passive E1 Pri Tap for Voice Recording |
| 5:59AM |
1 |
G.729 & Voicemail |
| 5:37AM |
3 |
pci 2.2 - pci-e x16 |
| 3:50AM |
0 |
RAD IPmux8 |
| 1:41AM |
0 |
Friday April 20th Asterisk Users Conference at 12:30PM EDT |
| 12:54AM |
0 |
app_voicemail.c |
| |
| Thursday April 19 2007 |
| Time | Replies | Subject |
|---|
| 10:31PM |
1 |
Cell phone that can be connected to standad phone switch network |
| 9:52PM |
1 |
Newbie Question about E1 |
| 8:52PM |
0 |
SLA with SIP only configuration |
| 7:38PM |
1 |
Failed to authenticate on INVITE |
| 4:31PM |
1 |
users.conf SIP registration fails |
| 4:26PM |
1 |
MySQL Update from exten |
| 2:44PM |
2 |
3rd T1 of quad card won't change signaling |
| 2:23PM |
2 |
Polycom SIP Phones On LAN can't register without WAN (Internet) Access |
| 2:15PM |
1 |
Problem with TDM2400 and Polycom 501... Voice Quality Lost... |
| 1:38PM |
0 |
CLI Dialplan options... |
| 12:41PM |
2 |
CallerID masking |
| 12:36PM |
1 |
Asterisk 1.4.2 connection to Nortel CS1000M -followup with log |
| 12:07PM |
1 |
Asterisk 1.4.2 connection to Nortel CS1000M |
| 11:47AM |
1 |
aastra phones with asterisk 1.2.17 - hangup after 20 seconds |
| 10:40AM |
1 |
Setup Asterisk configuration |
| 10:02AM |
1 |
Asterisk 1.2 and mixmonitor stopping short |
| 9:29AM |
1 |
AudioCodes MP-104 MGCP? |
| 9:28AM |
3 |
ztdummy does not load properly at server startup |
| 9:12AM |
1 |
CDR(dst) != CALLERID(dnid) |
| 8:49AM |
1 |
Asterisk - Cisco Call Manager Express Trunk |
| 7:51AM |
2 |
Dial plans |
| 7:48AM |
3 |
Outgoing CallerID |
| 6:47AM |
1 |
Ser as IVR |
| 5:33AM |
1 |
Asterisk Queue Call Transfer |
| 5:03AM |
1 |
Hardware suggestions for 8-10 lines in the UK |
| 4:17AM |
5 |
Polycom IP 501 is displaying wrong time |
| 3:56AM |
2 |
extensions.conf #include behaviour |
| 3:29AM |
0 |
any format |
| 2:34AM |
1 |
Help Astertest - Asterisk stressing tool |
| 2:25AM |
0 |
DTMF issues |
| 2:08AM |
6 |
ZT_CHANCONFIG failed on channel 1: No such device or address |
| 1:47AM |
2 |
SIP kpml DTMF support in * |
| 12:28AM |
1 |
Improve voice quality on Asterisk + chan_capi + DIVA BRI |
| |
| Wednesday April 18 2007 |
| Time | Replies | Subject |
|---|
| 11:24PM |
1 |
Re: asterisk-users Digest, Vol 33, Issue 80 |
| 8:33PM |
2 |
Trigger for unavailable SIP peer |
| 3:38PM |
1 |
Timestamp in recorded calls filename |
| 3:08PM |
1 |
[OT] OMG Verizon is terrible |
| 3:02PM |
1 |
Monitor application inestability and high load |
| 2:57PM |
0 |
IM |
| 2:47PM |
2 |
MeetMe Error |
| 2:36PM |
1 |
Asterisk 1.4.2 + Cisco 7960G not registering |
| 2:21PM |
0 |
Audio playback problems with FC6 and Zaptel 1.2.16 |
| 2:11PM |
3 |
RE: OT (a little): IPV6 Ramifications Article |
| 2:01PM |
2 |
Segmentation Fault |
| 12:26PM |
1 |
gxp2000 expansion module blf leds not working |
| 10:18AM |
0 |
Dial out from AGI and then connect it to another dialled out call |
| 9:55AM |
0 |
Asterisk COLP (COnnected Line Presentation) |
| 9:14AM |
2 |
incoming SIP call |
| 7:57AM |
3 |
asterisk svn and zaptel |
| 5:55AM |
2 |
asterisk unable to create files, too many files open |
| 5:51AM |
0 |
QueueMetrics 1.3.4 released today |
| 5:47AM |
0 |
Phones working with 1.2.17, not with 1.4.2 |
| 5:30AM |
0 |
Reminder: HITBSecConf2007 - Malaysia: Call for Papers closing in 2 weeks |
| 3:28AM |
9 |
Feedback on Linksys SPA-921 and GrandStream GXP-2000 |
| 12:04AM |
2 |
SIP failover between Sip Providers |
| |
| Tuesday April 17 2007 |
| Time | Replies | Subject |
|---|
| 11:08PM |
1 |
Asterisk Billing |
| 8:52PM |
2 |
Can I add distinctive ring with asterisk and TDM400? |
| 8:37PM |
1 |
No Incoming Ring Tone (Even with "r" option) |
| 5:57PM |
0 |
Re: [Asterisk-java-users] asterisk-java.org up again with bonus article on Local/ channels |
| 5:55PM |
1 |
TM Malaysia E1 PRI signaling |
| 5:29PM |
0 |
Change to Sip in a phonecall, when the user registers meanwhile |
| 5:07PM |
0 |
Question regrading IAX |
| 4:53PM |
2 |
queues |
| 3:17PM |
1 |
Transfercapability DIGITAL |
| 2:46PM |
2 |
CDR datasets |
| 2:32PM |
2 |
Querying channel variables via the Manager API |
| 2:29PM |
1 |
internal sounds of asterisk / freePBX |
| 2:03PM |
1 |
Asterisk 1.2.16 - No Caller ID |
| 1:49PM |
1 |
Default lenguage |
| 1:25PM |
2 |
Connection between Asterisk - Cisco 2851 |
| 11:44AM |
1 |
What are service activation codes ? |
| 10:56AM |
0 |
TDM24 Cards |
| 10:17AM |
3 |
Transfer via CTI |
| 10:06AM |
2 |
Trigger a wake-up call from the shell? |
| 9:47AM |
2 |
Voicemail files permission |
| 8:39AM |
2 |
peers are using wrong contexts |
| 8:09AM |
1 |
PRAM |
| 7:10AM |
2 |
Whats this about! |
| 6:38AM |
0 |
Queue report statistics |
| 6:09AM |
4 |
Using meetme like call |
| 2:03AM |
0 |
GETVARIABLE and IAX |
| 12:54AM |
2 |
No of Calls |
| 12:40AM |
5 |
sending an SMS via Asterisk? |
| 12:17AM |
1 |
Make an iso image |
| |
| Monday April 16 2007 |
| Time | Replies | Subject |
|---|
| 11:50PM |
1 |
Asterisk VS Cisco,Avaya,Siemens,.... |
| 8:16PM |
1 |
Having trouble figuring this out... |
| 8:14PM |
2 |
[OT] Nokia E60 firmware update break SIP |
| 6:47PM |
1 |
Zaptel problems in Fedora 6 |
| 5:16PM |
1 |
Recommended hardware |
| 5:07PM |
0 |
Re: Dial outbound trunk numbers in a round-robin sequence? |
| 3:53PM |
2 |
Problem with queue |
| 3:18PM |
1 |
Problems with queue announcements under high call volumes |
| 2:36PM |
3 |
Audio Problems - Operating System?? |
| 1:43PM |
3 |
${CALLERIDNUM} DEPRECATED in 1.4.2 |
| 1:20PM |
1 |
Stuck on MySQL UPDATE |
| 12:10PM |
0 |
IAX implementation question |
| 11:27AM |
3 |
Redundant * servers |
| 10:53AM |
6 |
BSNL caller ID (India) |
| 10:50AM |
0 |
"jittershrinkrate" equivalent in current (new) iax jb implementation |
| 7:50AM |
1 |
Voicemail: How to send a notification even if Caller does not let any messages? |
| 7:40AM |
1 |
moving from asterisk1.2 to asterisk1.4 |
| 7:12AM |
1 |
Need some dialplan help for obscure user request |
| 7:00AM |
0 |
G.729 Pass-Thru & Voicemail |
| 6:43AM |
3 |
duration sec and billing sec in cdr |
| 5:51AM |
0 |
Dial n voicemaile |
| 5:24AM |
1 |
Difference between SCCP and Cisco Call Manager traffic? |
| 5:08AM |
1 |
Instability on Asterisk |
| 4:58AM |
2 |
Queue trouble |
| 3:03AM |
2 |
sip tcp support |
| 2:48AM |
2 |
injecting audio announcements into sip channel |
| 12:48AM |
2 |
Debian asterisk-bristuff |
| 12:12AM |
4 |
New T1 Asterisk installation |
| |
| Sunday April 15 2007 |
| Time | Replies | Subject |
|---|
| 6:53PM |
9 |
Loudspeaker |
| 6:39PM |
3 |
Digium TE205P and channelbank |
| 6:17PM |
1 |
G723 problems with TC400B |
| 2:33PM |
1 |
Optipoint 420std SIP Firmware |
| 2:17PM |
1 |
Hardware |
| 2:09PM |
2 |
Is STP wire decent for analog phones? |
| 12:09PM |
0 |
601 Rebooting/Crashing seems to be due to full directory |
| 12:00PM |
1 |
LEDs on Polycom Expansion Modules misbehave when paging |
| 11:11AM |
5 |
Fax with Asterisk + Hylafax |
| 10:59AM |
0 |
transfering calls |
| 10:43AM |
0 |
Call tranfer drops 1st. digit |
| 6:53AM |
0 |
features.conf and blind xfer |
| 5:50AM |
1 |
saydigits in another "language" |
| 2:02AM |
2 |
agents and music on hold with autoanswer.. |
| |
| Saturday April 14 2007 |
| Time | Replies | Subject |
|---|
| 9:06PM |
5 |
queue report problem |
| 7:32PM |
0 |
Trixbox abd broadvoice |
| 5:11PM |
1 |
Installing Applications |
| 3:11PM |
0 |
Presence on Polycom 301 partially broke? |
| 10:10AM |
2 |
a2billing |
| 9:37AM |
1 |
Fast busy on TDM400P |
| 9:12AM |
1 |
"HTTP Connection Timeout" Trouble with Cisco 7960 Phone |
| 3:27AM |
4 |
what version is running? |
| |
| Friday April 13 2007 |
| Time | Replies | Subject |
|---|
| 9:52PM |
1 |
no real ring back |
| 4:37PM |
5 |
Job listing on cisco.com for Asterisk...? |
| 4:28PM |
0 |
Asterisk, nat, gizmo and fwd |
| 3:09PM |
6 |
Hardware requirements question |
| 3:04PM |
4 |
openvz resources |
| 2:37PM |
1 |
SpanDSP (RxFax) |
| 2:25PM |
1 |
Dial outbount trunk numbers in a round-robin sequence? |
| 1:58PM |
3 |
LED does not glow on new Voicemail |
| 1:44PM |
4 |
E1 capacity |
| 1:13PM |
0 |
Parking calls and snom phones |
| 12:48PM |
1 |
Outgoing Calls on PRI ISDN |
| 12:47PM |
1 |
compile error on RHEL5 or CENTOS5 |
| 11:43AM |
2 |
MySQL query from extensions? |
| 10:10AM |
1 |
Bristuff and HPEC |
| 9:33AM |
5 |
SIP REGISTRATION TIME OUT |
| 9:01AM |
1 |
Call Recording Servers |
| 8:37AM |
2 |
FreePBX - Vicidial Integration |
| 6:18AM |
4 |
Polycom 501 sluggish keys: found the problem! |
| 4:52AM |
2 |
A question about an install i have been asked about... |
| 4:42AM |
1 |
PAP2T-NA Jitter Buffer |
| 2:13AM |
0 |
meet me monitor |
| 1:38AM |
1 |
How can i add multiple callerids to an inbound route? |
| 1:37AM |
2 |
How can i send voicemail to multiple email IDs? |
| 12:36AM |
2 |
voicemail - "digits/1F does not exist in any format" |
| 12:14AM |
0 |
Asterisks CDMA Cards |
| |
| Thursday April 12 2007 |
| Time | Replies | Subject |
|---|
| 11:58PM |
0 |
compilation error in CYGWIN |
| 10:46PM |
4 |
Zap failure: cause 66 - Channel not implemented |
| 10:02PM |
1 |
Recording and Conferencing |
| 8:08PM |
3 |
Huh? IP address ending with 611 |
| 6:22PM |
0 |
Outside Network PAP and also Outside Network eyeBeam Soft Phone |
| 6:06PM |
1 |
Re: Which SIP phones... |
| 5:26PM |
0 |
RAGI channel_status() never returnes |
| 4:26PM |
1 |
Spandsp-0.0.3 and asterisk 1.2 |
| 4:02PM |
3 |
zaptel/ssh interaction |
| 2:13PM |
3 |
Sharing trunks between asterisk machines |
| 12:43PM |
1 |
Asterisk and hard phone configuration |
| 11:24AM |
0 |
speex codec: Out of Buffer space |
| 10:54AM |
1 |
Asterisk-Java website |
| 10:54AM |
1 |
Destar web interface problem |
| 9:36AM |
0 |
Installing codec g729 on Asterisk 1.2.1 on FreeBSD 6.0 |
| 9:31AM |
1 |
Asterisk (1.4) and hints/presence/BLF |
| 9:22AM |
6 |
Fax Blast over IP? |
| 8:58AM |
2 |
Best External PRI Gateway? |
| 8:35AM |
1 |
Delay to start sip registration after asterisk restart |
| 7:54AM |
1 |
SIP: number to names |
| 7:49AM |
0 |
video phones and call files |
| 7:42AM |
2 |
DTMF problem with inbound calls on Toll-Free number |
| 7:17AM |
2 |
Asterisk 1.2.17 and Cisco 7940/SIP: bug or what? |
| 6:27AM |
1 |
(no subject) |
| 6:00AM |
1 |
CDR(disposition) |
| 4:25AM |
2 |
Measuring audio file legth |
| 4:07AM |
0 |
Asterisk 1.2.14 and zaptel 1.2.12 ivr hangs every 2 days |
| 4:06AM |
1 |
Automatic Hang |
| 4:06AM |
0 |
(no subject) |
| 3:23AM |
8 |
test |
| 2:25AM |
1 |
compile problem with wavelenght |
| 1:01AM |
1 |
hanguponpolarityswitch - where did it go?? |
| |
| Wednesday April 11 2007 |
| Time | Replies | Subject |
|---|
| 11:49PM |
0 |
How to set fromuser in sip.conf so each user gets it's own callerid? |
| 8:52PM |
0 |
Asterisk or Trixbox |
| 6:15PM |
2 |
Polycom 301 questions |
| 4:53PM |
5 |
What is your Backup Strategy? |
| 4:02PM |
0 |
ZAP does not disconnect |
| 2:53PM |
6 |
Which SIP phones to buy? |
| 1:09PM |
2 |
FW: Polycom 501 issue with latest firmware : sluggish keys |
| 1:03PM |
3 |
missing chan_zap.so |
| 12:52PM |
1 |
Polycom - Static IP |
| 10:43AM |
1 |
Mediatrix 1204 |
| 10:37AM |
2 |
Polycom 501 issue with latest firmware : sluggish keys |
| 10:15AM |
3 |
SIP Jitter Buffer Patch for 1.2.x branch? |
| 9:37AM |
2 |
IMAP Voicemail with MS Exchange |
| 9:28AM |
1 |
Call Pickup with more than one argument |
| 9:27AM |
1 |
outCALL- the open source Asterisk integration applicaiton for Microsoft Outlook |
| 9:00AM |
1 |
Cisco IP Phone services.xml sample? |
| 8:26AM |
2 |
SIP INFO message |
| 8:23AM |
0 |
GTalk and No Audio Problem |
| 7:53AM |
3 |
Execute EAGI script with params from extensions.conf |
| 6:44AM |
1 |
wideband codec support |
| 6:16AM |
0 |
Re: nadi: branch 1.4 r61342 - /branches/1.4/channels/chan_misdn.c |
| 4:16AM |
1 |
Purposely setting red alarm on PRI for testing purposes |
| 4:07AM |
1 |
calls bridging |
| 3:58AM |
0 |
Asterisk Pickup with more than one argument |
| 2:41AM |
0 |
IM on x-lite |
| 1:38AM |
10 |
Nagios asterisk monitoring |
| |
| Tuesday April 10 2007 |
| Time | Replies | Subject |
|---|
| 7:18PM |
3 |
Learn some terminalogy before mounting this task. |
| 6:15PM |
4 |
how to install asterisk on redhat ? |
| 12:31PM |
6 |
Help w/ Asterisk Cisco IP phone and SCCP |
| 11:28AM |
0 |
ISDN BRITE support? |
| 10:59AM |
1 |
XO Flex T-1 & Asterisk |
| 9:46AM |
1 |
Maximum retries exceeded on transmission |
| 9:32AM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 38 |
| 7:45AM |
0 |
Voicemail: How to send a notification if Caller hags up during announcement |
| 7:27AM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 37 |
| 7:01AM |
4 |
Asterisk without PSTN interface cards |
| 6:58AM |
2 |
Reverse-ATA : Using PSTN lines to connect to Asterisk |
| 6:52AM |
1 |
help with Sipura SPA 3000 |
| 5:14AM |
0 |
clarification about bridge the call |
| 4:50AM |
0 |
Dialplan help - MeetMe (or ChannelRedirect) and call monitoring |
| 4:26AM |
1 |
Dialplan help - MeetMe and call monitoring |
| 3:48AM |
1 |
Re: asterisk-users Digest, Vol 33, Issue 36 |
| 3:42AM |
0 |
checking credit by phone |
| 3:41AM |
0 |
Autoreply: [Posible Spam] asterisk-users Digest, Vol 33, Issue 36 |
| |
| Monday April 9 2007 |
| Time | Replies | Subject |
|---|
| 11:08PM |
2 |
Strange error, "logger.c: No more room in scheduler..." |
| 10:43PM |
1 |
zapata.conf |
| 10:23PM |
0 |
Call forwarding (from PHONE configuration) with PRI |
| 9:01PM |
2 |
Cisco GW, PRI & CallerID Name |
| 6:54PM |
1 |
Re: asterisk-users Digest, Vol 33, Issue 35 |
| 6:18PM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 35 |
| 6:13PM |
0 |
no reply to our critical packet |
| 3:50PM |
3 |
Too much silence, perceived delay |
| 3:08PM |
1 |
Date Wise Recordings |
| 2:20PM |
3 |
Play audio and continue to next priority before audio ends... |
| 1:37PM |
1 |
TellMe Voice Recognition in Asterisk working.. |
| 1:16PM |
1 |
${QUEUESTATUS} |
| 1:03PM |
0 |
Asterisk mini conference within IT360 in Toronto Apr30-May2nd |
| 12:50PM |
1 |
T100P -->>> TE120P |
| 12:06PM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 34 |
| 11:51AM |
0 |
OT: But telephony related and funny |
| 9:39AM |
2 |
trouble recording calls |
| 9:26AM |
1 |
Re: asterisk-users Digest, Vol 33, Issue 33 |
| 9:11AM |
1 |
Polycom 330/320 |
| 9:10AM |
1 |
Received mini frame before first full voice frame |
| 7:47AM |
2 |
Privacy Manager w/ No Recording |
| 7:07AM |
2 |
DTMF auto detection bug? |
| 6:56AM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 32 |
| 6:46AM |
4 |
incoming zaptel calls fail |
| 6:14AM |
3 |
Upgrade 4 to 8 Analog Lines Question |
| 5:26AM |
2 |
Asterisk installation issue - CLI showing 0 active channels |
| 2:43AM |
3 |
sip_header=value? |
| |
| Sunday April 8 2007 |
| Time | Replies | Subject |
|---|
| 7:43PM |
2 |
intermittent choppy sound over wifi link |
| 12:01PM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 31 |
| 7:11AM |
3 |
Is there a variable for SIP response codes? |
| 5:40AM |
1 |
Manager Originate and Var to long |
| 5:26AM |
1 |
Adding Noise or background noise |
| 2:37AM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 30 |
| |
| Saturday April 7 2007 |
| Time | Replies | Subject |
|---|
| 5:49PM |
0 |
Linux IAX client to zaptel voice quality issue |
| 2:13PM |
3 |
Prompt for a PIN number to make long distance call? |
| 12:15PM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 29 |
| 10:35AM |
1 |
Follow Me and Transferring Calls |
| 10:09AM |
2 |
Cannot compile 1.4.2 on Slackware 7 |
| 9:34AM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 28 |
| 8:45AM |
2 |
Verizon Vonage 101 |
| 4:11AM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 27 |
| 3:59AM |
2 |
Different devices for asterisk!!! |
| 3:36AM |
1 |
CRM integration with Asterisk |
| 1:00AM |
0 |
Wireless Bridge for SNOM360 |
| |
| Friday April 6 2007 |
| Time | Replies | Subject |
|---|
| 6:23PM |
1 |
Voicemail from GTalk says "from an unknown caller" |
| 5:22PM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 26 |
| 5:15PM |
0 |
Yellow alarm TE110P with latest release |
| 5:07PM |
1 |
How well does a celldock work with Asterisk? |
| 3:06PM |
12 |
Verizon-Vonage Lawsuit |
| 1:11PM |
5 |
Balancing the Hybrid |
| 12:14PM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 25 |
| 11:37AM |
1 |
Snom 320 voicemail key & MWI |
| 11:01AM |
1 |
hox to connecte two asterisk server |
| 7:43AM |
1 |
Is it possible to Voicemail menus (not just audio files) ? |
| 7:33AM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 24 |
| 7:14AM |
2 |
SIP Header fields? |
| 7:10AM |
0 |
sending the dialed no to the peer |
| 7:07AM |
1 |
Poor analog line quality, wireless "base station", FAX-ing |
| 7:03AM |
0 |
Meetme and on demand monitoring |
| 6:26AM |
1 |
pap2 - dtmf works when 'sip debug' is enabled |
| 5:42AM |
7 |
FAX thru TDM400p |
| 5:05AM |
0 |
SVN update |
| |
| Thursday April 5 2007 |
| Time | Replies | Subject |
|---|
| 11:06PM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 23 |
| 10:55PM |
9 |
HPEC audio clipping |
| 4:59PM |
5 |
Open Source VoIP client (on a webpage) |
| 4:18PM |
1 |
BeroNet HFC-4S card is now detected as only 2 ports |
| 1:32PM |
2 |
Analog phones, dial out |
| 1:06PM |
2 |
Queue call distribution |
| 1:03PM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 22 |
| 12:57PM |
0 |
IAX2 threads |
| 12:48PM |
1 |
Asterisknow or Trixbox? |
| 11:37AM |
1 |
Dialplan not reading MySQL table |
| 11:20AM |
0 |
detecting a "beep" |
| 10:56AM |
2 |
IAX Trunk Failover |
| 10:29AM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 21 |
| 8:34AM |
0 |
No response on extensions: TDM842 |
| 8:17AM |
1 |
What is this error message? (check_auth: stale nonce received from ...) |
| 8:02AM |
2 |
Polycom 601 message waiting indicator |
| 7:23AM |
2 |
PRI DCHAN Errors |
| 3:42AM |
0 |
Asterisk 1.2.17 and BRIstuff - SOLVED |
| 3:26AM |
1 |
Asterisk 1.2.17 and BRIstuff |
| 1:52AM |
0 |
SNOM and Got SUBSCRIBE for extensions without hint. Please add hint to *8 in context inbound_sip |
| 12:56AM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 20 |
| 12:06AM |
1 |
How to return dialstatus of second (sub) call |
| |
| Wednesday April 4 2007 |
| Time | Replies | Subject |
|---|
| 9:36PM |
1 |
polycom repair |
| 7:09PM |
1 |
Asterisk server hangs on after only few hours again. |
| 4:55PM |
0 |
Voicemail Playback Issue |
| 3:38PM |
0 |
"no gtalk capable clients to talk to." |
| 3:30PM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 19 |
| 2:51PM |
4 |
ZAP device reference in Zaptel 1.4 - SIMILAR |
| 2:28PM |
0 |
Bad Line Noise over T1 |
| 2:22PM |
1 |
Pound # key not being handled |
| 1:50PM |
1 |
Polycom |
| 1:12PM |
1 |
Tunnel Q.SIG through an IP network |
| 12:41PM |
1 |
Queue application strategy |
| 12:36PM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 18 |
| 12:24PM |
0 |
Speex codec in 1.4.2 |
| 12:21PM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 17 |
| 10:34AM |
2 |
make Zaptel 1.2.16 'struct inode' has no member named 'u'. |
| 10:06AM |
0 |
Console messages |
| 9:53AM |
0 |
Voicemail to Text Transcription(was: Re: [asterisk-dev] Voicemail to text translation) |
| 8:51AM |
1 |
Configuring sip.conf to allow guest access |
| 8:48AM |
0 |
Which GUI for call screening ? |
| 8:25AM |
1 |
call files |
| 8:14AM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 16 |
| 7:48AM |
0 |
Parked calls and Music on hold |
| 7:07AM |
1 |
disabling authentication |
| 6:50AM |
1 |
Using DUNDi in a failover environment |
| 6:32AM |
1 |
RE: asterisk-users Digest, Vol 33, Issue 15 |
| 6:18AM |
2 |
Ring file |
| 5:22AM |
0 |
make a call with IP address |
| 5:19AM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 15 |
| 4:41AM |
0 |
SIP - choppy sound on local LAN to T1 |
| 4:36AM |
5 |
"remote" SIP, no audio, or one way audio. |
| 3:56AM |
0 |
Asterisk Job in Saudi Arabian Companies? |
| 3:29AM |
0 |
what the cable to connect with digium TE110 and avaya s8300 |
| 3:14AM |
2 |
Digium B410P Need Help |
| 2:56AM |
0 |
Localise VM_DATE timestamp like the voicemessage envelope |
| 1:55AM |
1 |
extra field |
| 12:25AM |
2 |
Remastering asterisk |
| 12:17AM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 14 |
| |
| Tuesday April 3 2007 |
| Time | Replies | Subject |
|---|
| 9:18PM |
0 |
I can't use the 'Group', 'CallGroup' , 'PickupGroup' in SIP channel (asterisk1.4.2) |
| 6:21PM |
0 |
Faxing issues |
| 6:15PM |
0 |
Called Number Issue |
| 5:48PM |
1 |
ZAP device reference in Zaptel 1.4 |
| 5:14PM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 13 |
| 5:08PM |
0 |
DTMF via IAX ignored after a few seconds |
| 4:47PM |
2 |
stun |
| 2:18PM |
1 |
x100p not showing in core show channels |
| 1:37PM |
0 |
T400P 4 Port T1 Cards for Sale |
| 1:12PM |
1 |
Re: asterisk-users Digest, Vol 33, Issue 12 |
| 1:05PM |
3 |
asterisk and mplayer |
| 1:05PM |
2 |
Play "blank" sound while VM recording? |
| 12:56PM |
1 |
Re: asterisk-users Digest, Vol 33, Issue12 |
| 12:44PM |
6 |
Re: asterisk-users Digest, Vol 33, Issue 12 |
| 12:10PM |
1 |
Re: asterisk-users Digest, Vol 33, Issue 12 |
| 10:53AM |
1 |
Interconnecting Cisco 1760 routers with Asterisk |
| 9:26AM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 11 |
| 9:09AM |
1 |
RE: Asterisk-Addon-1.4.0 MySQL module |
| 9:02AM |
0 |
Problem with a Zap channel |
| 8:24AM |
1 |
Hints not working using SVN-branch-1.4-r59289 |
| 8:19AM |
1 |
master.csv interpretation |
| 7:51AM |
2 |
Require only GSM Codec |
| 7:45AM |
3 |
Adding DND to dialplan |
| 7:43AM |
1 |
realtime problem |
| 7:09AM |
0 |
CDR and RADIUS (cdr_radius) - working |
| 6:53AM |
1 |
kirk Wireless + Asterisk 1.2.7 Mysql Realtime problem |
| 6:35AM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 10 |
| 6:28AM |
1 |
Asterisk USER PORTAL |
| 5:30AM |
1 |
ipv6 patch |
| 3:58AM |
7 |
Zaptel 1.4.1 Install Modules CentOS |
| 2:53AM |
0 |
Dial Macros |
| 2:25AM |
0 |
Quad BRI cards |
| 1:04AM |
1 |
SDP bug |
| 12:20AM |
0 |
tdm400 problem |
| |
| Monday April 2 2007 |
| Time | Replies | Subject |
|---|
| 11:48PM |
1 |
understanding what h extension does |
| 11:41PM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 9 |
| 10:47PM |
1 |
adding chan_celliax |
| 8:45PM |
3 |
DTMF problem with 1.4.1 |
| 7:35PM |
1 |
TE120P and Unknown Signalling Method |
| 7:12PM |
2 |
Re: On Topic: Cheapest Asterisk USB Key? |
| 4:36PM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 8 |
| 2:50PM |
0 |
automonitor and CDR(userfiled) |
| 1:25PM |
1 |
Asterisk realtime |
| 1:21PM |
1 |
SIP 484 (Early Dial) and International Dialing |
| 12:45PM |
3 |
Replicating SIP Registrations Across Asterisk Servers |
| 12:32PM |
1 |
Quick question about time |
| 12:10PM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 7 |
| 12:07PM |
3 |
Fax detection (Sangoma) |
| 12:07PM |
5 |
Aastra 480 i |
| 10:31AM |
3 |
misdn and debian |
| 9:03AM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 6 |
| 8:24AM |
5 |
simplify |
| 8:16AM |
1 |
Mysql issue |
| 7:22AM |
0 |
Freeworddialup, no inbound calls |
| 7:14AM |
1 |
Yeastar Cards |
| 7:04AM |
1 |
chan-capi-HEAD and Asterisk 1.4.2 |
| 6:12AM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 5 |
| 6:12AM |
0 |
Queue |
| 4:48AM |
1 |
LDAP authentication in Asterisk |
| 4:11AM |
0 |
LDAPget in Asterisk |
| 2:04AM |
1 |
Number of calls |
| 2:03AM |
0 |
Interconnection of LDK to an Asterisk server |
| 1:46AM |
3 |
SIP - Automatic Redial on No Answer |
| 1:25AM |
0 |
hardware for 100-200 sip users |
| 1:16AM |
1 |
603 Error |
| |
| Sunday April 1 2007 |
| Time | Replies | Subject |
|---|
| 11:16PM |
5 |
Best Hardphone (Subjective?) |
| 9:31PM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 4 |
| 8:35PM |
2 |
Trigger and Email in Dial Plan |
| 7:35PM |
15 |
Problems with TE110P |
| 1:56PM |
5 |
[MACRO-SCREEN] and MACRO_RESULT |
| 12:02PM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 3 |
| 11:25AM |
0 |
Add/remove international prefix |
| 10:09AM |
1 |
Asterisk 1.2 and res_perl - lock that leads to weird behaviour |
| 9:10AM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 2 |
| 8:44AM |
3 |
Announcement: Asterisk Service Provider Edition v1.0 Beta |
| 7:57AM |
0 |
jiaxclient run error |
| 6:28AM |
2 |
Linking incoming calls |
| 6:08AM |
5 |
On Topic: Cheapest Asterisk USB Key? (was: Re: Off Topic: Open Source USB Softphone) |
| 6:00AM |
1 |
Anyone here have opinion on the Linksys SPA-400? |
| 3:16AM |
2 |
Weird extension behavior |
| 2:57AM |
0 |
Re: asterisk-users Digest, Vol 33, Issue 1 |
| 12:54AM |
1 |
No Audio with Gtalk |
| 12:02AM |
0 |
ISDN PRI DTMF problem |