| Saturday March 31 2007 |
| Time | Replies | Subject |
|---|
| 3:37PM |
1 |
Setting a call to be recorded before Xfer? |
| 2:14PM |
0 |
Understanding the dial flags |
| 12:22PM |
1 |
Re: asterisk-users Digest, Vol 32, Issue 129 |
| 9:34AM |
1 |
LUSYN patches |
| 8:43AM |
4 |
Sponsored development - Monodirectional audio handling |
| 5:58AM |
2 |
Meetme question |
| 3:08AM |
2 |
Question on Priorities |
| |
| Friday March 30 2007 |
| Time | Replies | Subject |
|---|
| 9:44PM |
1 |
Indicating agent status on the phone |
| 8:13PM |
0 |
Re: Lucent TNT - ring timer |
| 7:30PM |
0 |
Re: asterisk-users Digest, Vol 32, Issue 128 |
| 2:51PM |
2 |
switchtype and signalling query |
| 1:52PM |
0 |
Redirect failed, channel not up. |
| 1:49PM |
1 |
Paging |
| 11:45AM |
1 |
Realtime call-limit |
| 11:32AM |
3 |
Multi-Level Queue |
| 10:20AM |
0 |
forwarding loop not detected |
| 10:11AM |
0 |
unconditionally redirecting incoming calls by 302 Moved Temporarily messages doing right accounting |
| 9:48AM |
8 |
SNOM 360 |
| 9:38AM |
1 |
One way intermittent static to PSTN |
| 9:35AM |
1 |
call file vs. originate |
| 9:07AM |
0 |
SPA3102 PSTN fallback |
| 6:56AM |
1 |
xten web phone |
| 6:49AM |
0 |
pickupgroup / SIP / Cisc phones |
| 5:57AM |
1 |
Which IP Phones have buttons can be assigned to functions with Asterisk |
| 5:32AM |
1 |
Sipura SPA2000 Transfer Call |
| 5:18AM |
4 |
Speed Dial Application in * |
| 3:33AM |
1 |
bad case of buzzing |
| 2:08AM |
1 |
Asterisk 1.4 with Digium B410P - Timing problem |
| 2:05AM |
2 |
web based sip phone |
| 1:42AM |
0 |
Asterisk-Addon-1.4.0 MySQL |
| 12:40AM |
1 |
Security on long distance calls |
| |
| Thursday March 29 2007 |
| Time | Replies | Subject |
|---|
| 10:39PM |
1 |
Correct latency values in "sip show peers" |
| 9:04PM |
1 |
FW: REG : H.323 Configurations with Asterisk |
| 8:53PM |
2 |
Need help to strip variable |
| 5:56PM |
1 |
queue priority causes crash |
| 5:35PM |
1 |
Queue priority |
| 5:22PM |
5 |
SIP RTP Tunnel |
| 3:42PM |
2 |
Problem while using asterisk Realtime |
| 3:26PM |
4 |
Linksys SPA 3102 causing me problems |
| 2:39PM |
1 |
Setting rxgain per channel |
| 2:01PM |
3 |
CallerID + Name |
| 1:43PM |
1 |
Polycom Power |
| 12:49PM |
0 |
Re: [asterisk-dev] Find the name of queue |
| 12:22PM |
2 |
Call Waiting problems |
| 12:04PM |
1 |
DTMF Corruption Problem in 1.4.2 for SIP RFC2833 plz halp |
| 11:18AM |
1 |
UK PRI and outgoing CLI FYI |
| 10:43AM |
1 |
chan_misdn |
| 10:36AM |
0 |
Re: Problem converting a Cisco 7960 to SIP |
| 10:29AM |
1 |
bugetone 200's |
| 10:24AM |
2 |
Polycom 501 + Asterisk +Edit buttons |
| 10:23AM |
0 |
Hearing noise after 1min of calling |
| 10:20AM |
3 |
Asterisk hangs up SIP call after 6 200 retransmits |
| 9:44AM |
2 |
help - UNSUBSCRIBE |
| 9:18AM |
3 |
Re: Problem converting a Cisco 7960 to SIP |
| 8:51AM |
2 |
SIP & NAT |
| 8:21AM |
0 |
Scratchy Audio with Asterisk 1.2.4 over IAX on FreeBSD? |
| 8:18AM |
1 |
Is it possible to install CCM on a Linux platform ? |
| 8:16AM |
0 |
Asterisk does not reINVITE after 302Redirect & 401Unauthorized |
| 8:05AM |
2 |
maximum simultaneous calls |
| 7:51AM |
2 |
L options in Dial() dont seem to work.... |
| 7:49AM |
0 |
Where are Spandsp changelogs or bugs available ? |
| 7:33AM |
4 |
Off Topic: Open Source USB Softphone |
| 7:07AM |
0 |
SV: Set(CALLERID(all) not working with 'unknown'call? |
| 6:31AM |
0 |
Asterisk Feature attended transfer |
| 6:20AM |
1 |
Interconnexion d'un serveur Asterisk à des PABX LG ( IP LDK) |
| 5:46AM |
0 |
DISCONNECT 41 hangup problem on PRI |
| 5:38AM |
0 |
txfax and result |
| 5:04AM |
1 |
cisco 7902 |
| 3:56AM |
8 |
error in FreePBX |
| 3:09AM |
1 |
Set(CALLERID(all) not working with 'unknown' call? |
| 2:32AM |
1 |
Dell poweredge 860 acceptable forofficeenvironment ? |
| 1:56AM |
2 |
sip: failed the authenticate on INVITE |
| 12:00AM |
1 |
Voice mail |
| |
| Wednesday March 28 2007 |
| Time | Replies | Subject |
|---|
| 11:59PM |
0 |
vzaphfc installation? |
| 11:30PM |
1 |
Nice Transfer Feature |
| 10:00PM |
3 |
Call dies when I press * |
| 8:51PM |
1 |
Stepped deployment - T1 PRI passthru |
| 6:16PM |
1 |
How to place a call to a Google Talk user? |
| 5:17PM |
2 |
Polycom SoundPoint 501 |
| 3:52PM |
0 |
asterisk-addons-1.4 write wrong uniqueid |
| 2:45PM |
1 |
Unsetting Global Vars |
| 2:29PM |
3 |
Polycom and Asterisk |
| 2:06PM |
1 |
SIP OPTIONS dialog not understood |
| 1:57PM |
1 |
App_RXFax Problem. |
| 12:34PM |
3 |
Multi-line phones - Asterisk uses wrong callerid |
| 12:08PM |
2 |
Transfering not working - how to debug? |
| 10:56AM |
0 |
Wanted: German to English translator for Asterisk documenation |
| 9:45AM |
0 |
BRI Cards |
| 9:12AM |
1 |
Asterisk: recommended installation |
| 8:08AM |
3 |
PoE - IEEE 802.3af |
| 7:20AM |
1 |
Development of new features in Asterisk Manager |
| 6:30AM |
1 |
h323 |
| 6:29AM |
1 |
MOS Score |
| 6:27AM |
2 |
Meetme cant handle more than 5 users in a call?? hmmmm |
| 5:41AM |
0 |
* 1.4.1: connected to gtalk but no voice passing |
| 5:19AM |
7 |
wireless desktop phones |
| 5:03AM |
3 |
System from AMI |
| 4:40AM |
1 |
Friday asterisk users live conference/podcast at 12:30PM EDT |
| 3:55AM |
0 |
Voicemailmain not changing password? |
| 2:21AM |
0 |
REG : H.323 Configurations with Asterisk |
| 1:53AM |
2 |
Can I generate random SIP traffic? |
| 1:24AM |
1 |
Odd MeetMe bahaviour with MoH ... |
| 12:59AM |
2 |
just on my LAN |
| |
| Tuesday March 27 2007 |
| Time | Replies | Subject |
|---|
| 6:24PM |
0 |
Macro Dial - External DID |
| 4:45PM |
1 |
"Couldn't load variables.txt?aldope=xxxxx " |
| 4:29PM |
2 |
Inbound Voice Quality - Speed Change |
| 3:13PM |
0 |
ARI with * 1.4.2 won't display recordings |
| 1:59PM |
1 |
Erased log files |
| 12:34PM |
5 |
Park & No Announce? |
| 10:29AM |
1 |
P-Asserted-Identify or Remote-Party-ID, or both? |
| 8:19AM |
1 |
just call to user |
| 8:18AM |
5 |
TDM400p reliability |
| 6:57AM |
1 |
AOC billing |
| 6:56AM |
0 |
IAX Experiences [WAS: Question about DSP in Digium card] |
| 6:54AM |
1 |
Re: asterisk-users Digest, Vol 32, Issue 106 |
| 6:53AM |
2 |
cisco 7905 |
| 5:10AM |
1 |
UK BT PRI |
| 5:00AM |
0 |
Asterisk MSOutlook Dialer |
| 3:50AM |
0 |
AMI - delete voicemail |
| 3:46AM |
1 |
Using server side phonebook directory with SPA941 |
| 2:31AM |
3 |
ztdummy and MOH |
| 2:20AM |
1 |
vzaphfc installation... |
| |
| Monday March 26 2007 |
| Time | Replies | Subject |
|---|
| 11:28PM |
2 |
how to define a pilot number |
| 11:25PM |
1 |
SIP Video Camera |
| 7:43PM |
0 |
rx_fax and Asterisk 1.4.2 |
| 7:35PM |
1 |
Server Recomendation |
| 4:36PM |
0 |
SIP REFER |
| 2:32PM |
2 |
SRTP vs ZRTP in Asterisk |
| 2:17PM |
1 |
Emergency chan_sip issue |
| 2:05PM |
0 |
Device not registering after boot |
| 2:00PM |
1 |
SIP registration |
| 1:20PM |
4 |
Doorphone |
| 11:51AM |
0 |
Hang up detection time in FXS module |
| 11:18AM |
0 |
Getting an ASR Number |
| 9:14AM |
2 |
Polycom 601 loop |
| 9:05AM |
1 |
outbound call |
| 8:33AM |
9 |
Multi-registration ? |
| 7:58AM |
1 |
Asterisk and T38 ? |
| 7:46AM |
0 |
Registration timed out after a "sip reload" |
| 7:40AM |
1 |
1.4 - IAX2 - No registration for peer |
| 7:16AM |
2 |
How is this feature called ? |
| 6:59AM |
0 |
ARI with * 1.4.x |
| 6:31AM |
0 |
No Audio when integrating with openSER and Asterisk in the SAME LAN , |
| 5:23AM |
1 |
Asterisk incoming caller id problem |
| 3:51AM |
1 |
Counting callers |
| 3:17AM |
1 |
cutting hash in dial app |
| 2:12AM |
1 |
Moving from Bristuff to mISDN |
| 1:17AM |
2 |
Failure acknowledgement time |
| 12:17AM |
0 |
Asterisk 1.4 Realtime problems |
| 12:11AM |
7 |
Two or More Bri Cards |
| |
| Sunday March 25 2007 |
| Time | Replies | Subject |
|---|
| 7:31PM |
1 |
Chan_cellphone and CentOS 4.x |
| 7:15PM |
1 |
how to check and set D-channel status |
| 6:47PM |
1 |
ztdummy install in the new zaptel 1.4.1 |
| 11:25AM |
2 |
Anyone having trouble with claling US Domestic on Sellvoip? |
| 11:06AM |
1 |
voicemail is not playing messages |
| 8:51AM |
1 |
the age old telephone tree... why re-invent the wheel? |
| 5:25AM |
2 |
AOCD -> SendText()? |
| 3:44AM |
1 |
Answer Confirmation with SIP/AIX channels |
| |
| Saturday March 24 2007 |
| Time | Replies | Subject |
|---|
| 11:43PM |
1 |
Problem with ztdummy |
| 6:11PM |
1 |
asterisk: error while loading shared libraries: libiksemel.so.3: cannot open shared object file: No such file or directory |
| 3:47PM |
1 |
Timeout for conferences |
| 3:39PM |
2 |
Question about DSP in Digium card |
| 12:33PM |
1 |
Remote host can't match request NOTIFY to call |
| 10:57AM |
1 |
Asterisk with Dialplan or TrixBox for this case? |
| 9:24AM |
2 |
TDM analog cards, volume, echo, fxotune, ztmonitor and HPEC |
| 8:57AM |
1 |
Voicemail for an AT&T System 75 |
| 8:12AM |
3 |
Need feedback on vitelity |
| 7:34AM |
2 |
Can be called on FreeWorldDialup/IAX channel, but can't make calls |
| 7:16AM |
1 |
Shoretel integration into Salesforce.com |
| 6:21AM |
1 |
Asterisk Viruses? |
| 3:54AM |
1 |
Issue with Hamlet ISDN PCI card(Cologne Chipset) |
| 3:12AM |
2 |
freepbx -> DB Error messages... |
| |
| Friday March 23 2007 |
| Time | Replies | Subject |
|---|
| 3:45PM |
2 |
SER vs Asterisk? |
| 2:20PM |
0 |
switches |
| 1:12PM |
1 |
Problem with busy and unavailable |
| 12:13PM |
0 |
IAX2 certificate based (RSA) user auth in 1.4.x |
| 11:57AM |
7 |
Doorphone vs. Grandstream BT101 |
| 10:25AM |
2 |
Sendmail and exchange for voicemail integration |
| 9:47AM |
2 |
PC / Phone Combo |
| 9:45AM |
1 |
HUD Lite server on Debian |
| 9:36AM |
3 |
SIP/IAX peers UNREACHABLE and audio loss |
| 9:27AM |
1 |
Noob question regarding PCI 2.x & TDM400P Card |
| 8:58AM |
0 |
Debian Asterisk and MeetMe |
| 8:49AM |
3 |
Semi-OT: Use T.38 ATAs to Extend fax lines |
| 8:40AM |
2 |
cause 127 |
| 8:23AM |
0 |
no incoming dad with mISDN 1.1.1 and asterisk? |
| 8:12AM |
3 |
SRTP testers needed |
| 7:03AM |
2 |
AsteriskNow Beta 4 with T1 Cards? |
| 6:32AM |
0 |
No Audio when integrating openSER and Asterisk , in NAT |
| 5:10AM |
0 |
minimal asterisk for iax2 bridge |
| |
| Thursday March 22 2007 |
| Time | Replies | Subject |
|---|
| 11:23PM |
1 |
Outbound SIP call from asterisk extension |
| 9:51PM |
0 |
Zaptel 1.4.1 Released |
| 5:20PM |
1 |
About FAX and PAP2 |
| 5:12PM |
1 |
Problem in using Two BRi Cards in Asterisk |
| 3:29PM |
0 |
SIP REFER ans Dialplan |
| 2:46PM |
0 |
PASS CORRECT CALLER ID INFO TO OFFSITE TRANSFER |
| 2:41PM |
1 |
managers |
| 12:49PM |
2 |
Linksys/Sipura SPA-942 phones in larger deployments |
| 11:52AM |
1 |
chan_capi and only one B channel usable? |
| 10:39AM |
2 |
hardware spec |
| 10:02AM |
3 |
accepting a call, macros, and key presses. |
| 9:49AM |
0 |
Asterisk x Mera MVTS |
| 8:43AM |
1 |
using ael and extensions.conf togather? |
| 7:51AM |
1 |
Gizmo project answers every call - can I use it in hunt group? |
| 7:29AM |
3 |
ChanSpy and MeetMe |
| 7:16AM |
2 |
Asterisk 1.4.2 |
| 6:18AM |
0 |
Bridged ZAP calls do not release |
| 5:18AM |
0 |
Digium b410p and 2.6.17 kernel bug? |
| 4:40AM |
2 |
302 Moved temporarely |
| 4:28AM |
0 |
Test Message |
| 3:51AM |
0 |
fax and mISDN: a chimera? |
| 2:31AM |
0 |
beronet BN8S0 and isdn phone |
| 2:23AM |
1 |
strange ring |
| |
| Wednesday March 21 2007 |
| Time | Replies | Subject |
|---|
| 10:45PM |
2 |
A request for your input. |
| 10:30PM |
1 |
CN=Diarmaid O'Loughlin/O=QAD1 is out of the office. |
| 9:04PM |
0 |
SIP peer disappearing |
| 7:17PM |
3 |
Cisco 30VIP Phone |
| 5:23PM |
1 |
Too Many Open Files, Hung SIP Sessions, Can I Increase File Count? |
| 3:24PM |
1 |
G729 'disappears' randomly |
| 2:38PM |
1 |
Looking for a terminations provider (carrier grade) |
| 12:54PM |
3 |
Voicemail mailbox number passed in connection? |
| 12:49PM |
2 |
Asterisk 1.4.2 chan_zap |
| 11:17AM |
0 |
install and setup app_mp4 application |
| 10:25AM |
1 |
How to resolve CallerID from AudioCodes FXO |
| 10:03AM |
0 |
Asterisk 1.4.2 Requires Zaptel from 1.4 svn branch for zap_chan? |
| 9:48AM |
3 |
Cisco 7970 with skinny on * 1.4.1 |
| 9:34AM |
1 |
Dlink i2eye |
| 8:55AM |
1 |
PickUp a call with feature pickup (*8) from a IAX2 channel |
| 8:25AM |
0 |
res_musiconhold.c:1243 load_module: No music on hold classes configured |
| 8:11AM |
1 |
ses ActiveDirectory and also Ldap and Kerberos. |
| 8:00AM |
1 |
asterisk log analyzer |
| 7:34AM |
0 |
reducing the number of extensions for every user |
| 7:23AM |
0 |
Asterisk 1.4.2 Released |
| 7:22AM |
0 |
Asterisk 1.2.17 Released |
| 7:09AM |
6 |
How to get AEL2 |
| 7:07AM |
0 |
Asterisk with AudioCodes Mediant 2000 |
| 6:47AM |
1 |
wct4xxp problem |
| 6:15AM |
1 |
Metaswitch help needed |
| 5:48AM |
4 |
FWD outgoing problem |
| 5:45AM |
1 |
About Pickup Grandstream |
| 5:23AM |
5 |
automated dialout detect forward |
| 4:55AM |
2 |
Limit call duration |
| 3:39AM |
7 |
polycom random reboots |
| |
| Tuesday March 20 2007 |
| Time | Replies | Subject |
|---|
| 9:07PM |
3 |
wrong values in duration and billsec in CDR |
| 8:37PM |
1 |
SIP/Polycom Issue, Asterisk 1.2.16, calls dropped |
| 12:00PM |
1 |
codec_zap and Asterisk 1.4.1 |
| 10:26AM |
1 |
Can't Compile w/HPEC |
| 9:53AM |
1 |
High Pitched Noise |
| 9:31AM |
1 |
Asterisk Automated Outbound Messaging |
| 8:04AM |
0 |
Activating Incoming Demo |
| 7:25AM |
1 |
Zaptel 1.2.16 Released |
| 7:22AM |
0 |
GXP-2000 Phones with Cisco 3560 PoE Switch |
| 7:17AM |
1 |
modem passthru |
| 6:57AM |
9 |
asterisk on debian |
| 5:28AM |
0 |
which spandsp for asterisk 1.2.16 (eom) |
| 5:11AM |
2 |
Problem with "AT&T Maintenance" protocol in PRI connection, no B+D channels available |
| 5:07AM |
2 |
error, install freePbx |
| 4:33AM |
0 |
PROGRESS code |
| 3:32AM |
1 |
Zapateller not playing audio via SIP Trunk? |
| 1:49AM |
2 |
Which parameters of a live Asterisk server would you monitor ? |
| 1:40AM |
0 |
how to interconnection asterisk(sip) with mera |
| |
| Monday March 19 2007 |
| Time | Replies | Subject |
|---|
| 5:03PM |
0 |
SIP provider did not send BYE if callee is an queue |
| 4:57PM |
10 |
Microsoft launches first PABX |
| 4:06PM |
1 |
Festival works extension to extension, not on trunk |
| 3:52PM |
0 |
1.4.1 - T38 Pass Through - Seeing some odd errors but the fax works..... |
| 2:20PM |
0 |
H.323 with g729 |
| 2:07PM |
1 |
ExternalIVR() Dialplan function and Festival |
| 1:52PM |
4 |
Teliax problems, they say use SIP, more mature & better working than IAX |
| 12:40PM |
2 |
Zaptel Dummy Driver |
| 12:10PM |
1 |
Unicall Brazil |
| 11:47AM |
8 |
Configuring Faxs any help :) |
| 11:16AM |
0 |
One way dtmf tone on IAX |
| 10:23AM |
3 |
Cepstral and numbers |
| 8:55AM |
1 |
Dial(Local/${EXTEN}@longdistance)? |
| 7:56AM |
2 |
TDM400p, no CLI activity |
| 7:40AM |
4 |
Queue App - Free agent and waiting calls |
| 5:23AM |
1 |
no special context for sip peer |
| 4:26AM |
0 |
asterisk 1.4: choppy voicemail sound after upgrade from 1.2.9.1 |
| 3:06AM |
0 |
make calls with different phone numbers |
| 2:53AM |
0 |
Conference server (or how to make a call withmorethan 3 u |
| 1:32AM |
2 |
Conference server (or how to make a call withmore than 3 u |
| 1:14AM |
2 |
Conference server (or how to make a call with more than 3 u |
| |
| Sunday March 18 2007 |
| Time | Replies | Subject |
|---|
| 8:58PM |
2 |
zttool always reports "OK" on TDM400P |
| 4:41PM |
2 |
camp on off-line phone |
| 3:42PM |
6 |
T1 cable for Digium T1/E1 Cards |
| 11:30AM |
1 |
Choppy sound with chan_capi + Fritz Card USB |
| 11:07AM |
1 |
Conference server (or how to make a call with more than 3 users) |
| |
| Saturday March 17 2007 |
| Time | Replies | Subject |
|---|
| 12:41PM |
0 |
1.4 sample postgresql configs |
| 11:17AM |
2 |
Queues |
| 8:22AM |
0 |
detecting missing end-of-queue records |
| 4:21AM |
2 |
Call counter for sip misbehaving |
| 1:49AM |
2 |
SMS Integration and SMS commands |
| 1:33AM |
0 |
Re: asterisk-users Digest, Vol 32, Issue 67 |
| |
| Friday March 16 2007 |
| Time | Replies | Subject |
|---|
| 6:15PM |
0 |
Monitor queue calls with Local channel agents |
| 5:09PM |
0 |
Channel stuck problem.. |
| 5:08PM |
12 |
Follow me on multiple numbers.. |
| 5:03PM |
2 |
Jajah.com like script? |
| 2:15PM |
1 |
Pickup some else's call |
| 12:47PM |
1 |
Problems with MFCR2 and Meridian |
| 11:33AM |
0 |
DISA and repeating calls |
| 11:32AM |
2 |
Refund from SellVoip? |
| 11:17AM |
2 |
Error compiling zaptel 1.4.0 |
| 10:45AM |
1 |
FW: Microsoft buys Tellme |
| 10:43AM |
3 |
Asterisk 1.2.13 Caller ID problem |
| 9:35AM |
4 |
proposal: a new mailing list for asterisk 1.4, why not? |
| 8:43AM |
0 |
MAX TNT Question |
| 7:44AM |
1 |
Cisco + Asterisk list anyone? |
| 6:30AM |
2 |
Cepstral voices |
| 3:44AM |
1 |
transfer=mediaonly : can't hear nothing |
| 3:13AM |
0 |
Transfer feature not working on asterisk 1.4.0 |
| 3:12AM |
1 |
Warning LSP Low |
| 2:57AM |
2 |
SIP phone supporting more than 10 extension with a call transfer command |
| 2:53AM |
4 |
Dell poweredge 860 acceptable for office environment ? |
| 2:04AM |
1 |
Voicechanger update for asterisk 1.4 |
| |
| Thursday March 15 2007 |
| Time | Replies | Subject |
|---|
| 9:46PM |
0 |
Re: busy/hangup/answer detection in PRI E1 channels (Vidura Senadeera) |
| 8:34PM |
2 |
Help! Echo problem even at T1 PRI? |
| 6:32PM |
2 |
voip-info.org is back! |
| 3:38PM |
1 |
Re: zapata with Tiger3XX compilation error |
| 2:11PM |
1 |
sip_nat.conf - Asterisk with two Ethernet Interfaces |
| 1:52PM |
3 |
Incoming Caller ID |
| 1:39PM |
2 |
A200 card problem |
| 12:54PM |
1 |
shutdown |
| 12:12PM |
1 |
Dropped calls in Asterisk - A general question |
| 11:03AM |
1 |
Freepbx Incoming call's configuration |
| 9:13AM |
1 |
asterisk n-way call problem |
| 7:33AM |
1 |
snom led not working with asterisk 1.4.1 |
| 7:31AM |
1 |
qozap: t3 timer expired for span ... |
| 3:08AM |
2 |
busy/hangup/answer detection in PRI E1 channels |
| 2:55AM |
0 |
Meetme variables |
| 2:43AM |
0 |
MP3Player |
| 1:41AM |
0 |
Cost of Branded Equipment for Voip Provider Implementation |
| |
| Wednesday March 14 2007 |
| Time | Replies | Subject |
|---|
| 10:15PM |
3 |
DECT to SIP gateway experiences |
| 10:08PM |
0 |
AgentCallBackLogin Help! |
| 8:59PM |
3 |
DNIS/DNID |
| 7:20PM |
1 |
Voip-Wiki Site Information |
| 6:16PM |
0 |
Inbound PSTN CLID irratic with A200 |
| 5:15PM |
6 |
Cisco 7912 |
| 4:59PM |
7 |
voip-info.org status update |
| 2:51PM |
7 |
While the VoIP-Info.org site is down... |
| 1:31PM |
1 |
Packetization Rate |
| 12:50PM |
0 |
Sped up recordings with 1.4.1 |
| 12:26PM |
1 |
Which SIP method/option to display a short text message ? |
| 12:19PM |
1 |
IAX2 - Congestion |
| 11:05AM |
0 |
Can I use an Intertel IPPhone Plus 7704500 with Asterisk somehow |
| 10:36AM |
3 |
Call center manager for Asterisk (Release 0.3) |
| 9:56AM |
0 |
${EXTEN} is limited to 17 characters under IAX ? |
| 9:09AM |
0 |
IVR after hangup |
| 9:02AM |
1 |
[G.729] Input Gain |
| 7:45AM |
4 |
what happened to asterisk wiki??? |
| 7:18AM |
3 |
Zaptel version for asterisk 1.2.16 |
| 6:41AM |
2 |
Manager connection problems |
| 6:18AM |
0 |
Autoprovisioning ST2030S |
| 5:54AM |
2 |
What is the best phone to get when using a headset? |
| 4:54AM |
2 |
Earliest dial tone, after boot up. |
| 4:49AM |
3 |
What happend to voip-info? |
| 2:32AM |
0 |
ChanSpy with Record() : Doesnt seem to work for an unbridged SIP call |
| 2:13AM |
1 |
beronet BN4S0 |
| 1:16AM |
1 |
strange things on call transfer |
| |
| Tuesday March 13 2007 |
| Time | Replies | Subject |
|---|
| 11:35PM |
1 |
T1 Integrator Birch |
| 8:32PM |
1 |
Digium S101i - Adapter DTMF works perfeclty |
| 2:43PM |
0 |
Press quotes needed from ENUM users on Asterisk |
| 2:10PM |
6 |
Asterisknow with video and X-Lite not quite working |
| 1:10PM |
1 |
RE: In Asterisk 1.4.x, Why Digium has two H323 channels? |
| 12:54PM |
0 |
asterisk 1.2.15 fax |
| 12:41PM |
0 |
Re: asterisk-users Digest, Vol 32, Issue 48 |
| 12:00PM |
1 |
Getting 7970 to update |
| 11:39AM |
3 |
How to match wild card inside a GoToIf? |
| 10:18AM |
1 |
120 concurrent ZAP connections in asterisk open edition. Is that possible? |
| 9:54AM |
3 |
DST and VM timestamp |
| 8:44AM |
0 |
SIP hardphones with good jitter tolerance |
| 8:38AM |
1 |
IAX2 Question (Asterisk 1.4 tarball) |
| 8:01AM |
1 |
French PRI channel - exact signaling used |
| 7:41AM |
1 |
voicemail scenario |
| 7:19AM |
0 |
Queue - Call delivery problems - Asterisk 1.4 - autofill=yes |
| 6:27AM |
0 |
CDR and CallerID |
| 5:06AM |
0 |
MusicOnHold stops after upgrade from 1.4.0 to 1.4.1 |
| 1:53AM |
1 |
Update Asterisk 1.2.12 to 1.4.1 ? |
| |
| Monday March 12 2007 |
| Time | Replies | Subject |
|---|
| 9:33PM |
4 |
great problem with sounds and ztdummy |
| 5:38PM |
0 |
RE: Playback 0.5% Too Fast? |
| 5:32PM |
2 |
Playback 5% Too Fast? |
| 5:19PM |
2 |
TDM-400, Polycom SIP phones, and echo problems |
| 5:03PM |
1 |
Voicemails with occasional speeded up portions |
| 3:46PM |
2 |
Call Back |
| 12:00PM |
2 |
Polycom: warble on registration? |
| 11:12AM |
1 |
deprecated ALERT_INFO var andAMI's Originate command |
| 11:03AM |
0 |
Incase anyone wanted it - SNOM USA DST settings |
| 10:46AM |
0 |
GXV3000 & Speakphone |
| 10:46AM |
0 |
LIDB/CNAM STORAGE DATABASE NEEDED |
| 10:15AM |
1 |
OT: Sipura DST Rules |
| 10:14AM |
4 |
FW: Seamless Multi Office Asterisk Deployment |
| 9:55AM |
1 |
GXP-2000 DST Change |
| 9:52AM |
2 |
Create meetme conference rooms on the flight. |
| 9:05AM |
4 |
SIP unicode support ? |
| 8:50AM |
1 |
ACM question |
| 8:43AM |
1 |
Filter IDENT(Port 113) on Linksys router puts remote extensions to one way audio |
| 8:34AM |
0 |
DST 2007 Config for Cisco 7970 |
| 7:50AM |
3 |
Rebooting all Aastra phones |
| 7:50AM |
1 |
In Asterisk 1.4.x, Why Digium has two H323 Channels |
| 6:24AM |
2 |
New to Asterisk |
| 5:33AM |
2 |
Single sign on PC + phone? |
| 5:32AM |
0 |
Shoutcast music-on-hold |
| 5:18AM |
3 |
Rebooting ALL polycom phones |
| 5:17AM |
0 |
Citel Handset Gateway DST fix - FYI |
| 5:16AM |
1 |
AMI - DBPut |
| 4:16AM |
0 |
Problem with H323 |
| 3:58AM |
0 |
Re: Help: CallerID Name not being sent |
| 3:51AM |
3 |
_ALERT_INFO replacement in 1.4? |
| 2:49AM |
0 |
Coming events in Europe |
| 2:44AM |
0 |
Pickup group |
| 2:07AM |
4 |
How many outgoing phone line/voip account do I need? |
| 1:18AM |
1 |
Problems with Voice conferencing |
| |
| Sunday March 11 2007 |
| Time | Replies | Subject |
|---|
| 10:48PM |
4 |
Problem configuring voice conference |
| 10:15PM |
1 |
Asterisk and Databases |
| 9:12PM |
1 |
Re-parking (or transfer) a parked call |
| 6:53PM |
2 |
DST changes for the US |
| 4:40PM |
2 |
g711 -> iLBC garbled voice in 1.4? |
| 3:32PM |
2 |
Complicated callback solution |
| 8:37AM |
5 |
Fix for TZ values updates for DST |
| 3:37AM |
0 |
How to best manage my dial plans as the continueto grow, and grow, and grow.... |
| 12:30AM |
1 |
Follow Up on Cannot get back chan_zap.so module!?? |
| |
| Saturday March 10 2007 |
| Time | Replies | Subject |
|---|
| 5:56PM |
2 |
IAX2 audio issues |
| 3:41PM |
0 |
how to determine the remote IP address when dialing in ooh323 |
| 12:34PM |
0 |
Security and DTMF |
| 11:54AM |
0 |
Polycom call parking feature and Asteriskcallparking |
| 11:10AM |
5 |
asterisk on mini-itx |
| 9:00AM |
0 |
chan_misdn and Asterisk 1.4.1 - Early B3 not working any more |
| 7:33AM |
1 |
FastAGI Question |
| 6:03AM |
0 |
Connecting two asterisk server. |
| 1:19AM |
1 |
installation pb on debian etch |
| |
| Friday March 9 2007 |
| Time | Replies | Subject |
|---|
| 10:00PM |
0 |
DTMF issue with TDM404 |
| 6:29PM |
1 |
How to best manage my dial plans as the continue to grow, and grow, and grow.... |
| 4:56PM |
0 |
spandsp, app_rxfax: apps_Makefile.patch v1.2 > v1.4 = No Workie! |
| 3:27PM |
1 |
REALTIME and VIEW ENTIRE DIALPLAN |
| 2:17PM |
3 |
Polycom call parking feature and Asterisk call parking |
| 2:16PM |
5 |
Recorded file processing app wanted |
| 12:53PM |
2 |
play file and action only stop if one defined key has been pressed |
| 12:17PM |
0 |
OS X Frequent console disconnects 1.4.1 |
| 11:55AM |
1 |
Cdr_mysql compile question |
| 11:49AM |
2 |
AEL #include file |
| 10:49AM |
1 |
Which hylafax client ? |
| 10:27AM |
1 |
RE: Coaching in asterisk |
| 10:21AM |
1 |
RE: Coaching in asterisk |
| 9:22AM |
0 |
Boot order of 2 TE110P and 1 TDM400P in the same |
| 9:03AM |
1 |
Another Faxing Question |
| 8:42AM |
0 |
YAACID and manager.conf security |
| 8:24AM |
1 |
sip tunnel |
| 7:24AM |
2 |
disable client side hangup after dialing 911 |
| 5:34AM |
0 |
Re: Back to back E1 - asterisk <=> toshiba pbx - Call droping issue [ ref:00D36mPe.50032wycQ:ref ] |
| 5:19AM |
2 |
Is there any variable for Voicemail Password in Asterisk |
| 12:52AM |
3 |
How to enter bridge_native_loop??? |
| 12:41AM |
1 |
Can't hear any sound (This time in plain text) |
| 12:29AM |
2 |
When to use Echo Cancellation cards? |
| 12:13AM |
3 |
Zaptel problem after upgrading to 1.2.16 |
| 12:10AM |
0 |
Fwd: Can't hear any sound |
| 12:04AM |
1 |
Which VoIP router and switch to use for medium size business |
| |
| Thursday March 8 2007 |
| Time | Replies | Subject |
|---|
| 11:03PM |
1 |
Issues with a Linksys SPA 2102 and asterisk |
| 6:01PM |
3 |
Boot order of 2 TE110P and 1 TDM400P in the same machine |
| 4:37PM |
2 |
Is Allison going to be banned from foreign travel over polar bears? |
| 4:16PM |
2 |
Newbie Question |
| 3:56PM |
0 |
Re: Coaching in asterisk |
| 3:37PM |
3 |
1.4 compile issue |
| 2:56PM |
2 |
Call load balancing |
| 2:43PM |
2 |
Zap Channel Deadlocks |
| 2:41PM |
0 |
Asterisk SIP to MAX TNT Gateway, Sporadic Echo |
| 2:37PM |
1 |
No application 'Prefix' for extension in1.2x, what app I have to use instead? |
| 12:35PM |
1 |
outdial to phone for new VM notification |
| 12:27PM |
0 |
Number of groups? |
| 11:47AM |
0 |
transfers and CDR |
| 11:30AM |
3 |
Sender phone ringing while recipient talking |
| 10:29AM |
1 |
Packet2Packet Bridging Questions |
| 10:19AM |
2 |
Queue announcing hold sequence instead of hold time |
| 10:00AM |
4 |
Accessing Voicemail by dialing own number |
| 9:03AM |
4 |
Asterisk distributed deployment |
| 7:58AM |
0 |
cmd pickup Problem |
| 7:10AM |
0 |
Fwd: Back to back E1 - asterisk <=> toshiba pbx -Call droping |
| 6:54AM |
0 |
New Linksys SPA Daylight Saving Time Rule for US/Canada |
| 6:50AM |
1 |
Asterisk + Panasonic pbx |
| 6:36AM |
2 |
Hinting and Realtime |
| 3:44AM |
6 |
Empty Wildcard TDM400P as a MeetMe timer. |
| 3:39AM |
0 |
"pritimer" parameter in zapata.conf |
| 3:27AM |
2 |
Queue Announcements for Operators |
| 3:21AM |
1 |
How to handle SIP-Callerid? |
| 2:31AM |
0 |
Timeouts not working |
| 1:20AM |
1 |
Re: Pickup *8 with CallerID |
| 12:06AM |
1 |
Call recording and archiving |
| |
| Wednesday March 7 2007 |
| Time | Replies | Subject |
|---|
| 10:56PM |
1 |
Re: Back to back E1 - asterisk <=> toshiba pbx - Call droping |
| 9:58PM |
0 |
SIP remote crash bug |
| 9:32PM |
2 |
Number of SIP messages per minute |
| 8:15PM |
1 |
sip show channels |
| 6:39PM |
1 |
Problems with Zaptel Drivers |
| 1:51PM |
1 |
auto dialer |
| 12:47PM |
1 |
Asterisk Registering to other SIP servers. |
| 11:46AM |
0 |
gtalk2voip and Asteris |
| 10:56AM |
1 |
Problem HandyTone 488 does not call transfer |
| 10:27AM |
2 |
mobility with asterisk |
| 9:07AM |
1 |
AsteriskNow Beta 4 - zaptel.conf / zapata.conf problems |
| 9:00AM |
2 |
Asterisk Auto-dial out |
| 8:42AM |
2 |
Asterisk queue and agents |
| 8:42AM |
2 |
queue information in mySQL |
| 7:53AM |
4 |
OT Vonage V-Phone Adapter (Possible Hack) |
| 7:38AM |
3 |
asterisk and ssl |
| 7:29AM |
1 |
Realtime Extensions and "Include" |
| 6:20AM |
2 |
Background / Invalid Extension through cell phone |
| 5:45AM |
2 |
VoIP over Alvarion Wireless |
| 3:31AM |
1 |
capi installation problem |
| 3:17AM |
0 |
Asterisk 1.4.1 - Calling problem |
| 2:59AM |
0 |
Back to back E1 - asterisk <=> toshiba pbx - Calldroping issue |
| 12:14AM |
1 |
Back to back E1 - asterisk <=> toshiba pbx - Call droping issue |
| |
| Tuesday March 6 2007 |
| Time | Replies | Subject |
|---|
| 11:03PM |
0 |
Re: asterisk-users Digest, Vol 32, Issue 21 |
| 10:11PM |
0 |
Anybody having problems using sellvoip? |
| 9:22PM |
0 |
Long term voicemail archival and synchronisationbetween multiple storage locations? |
| 9:22PM |
2 |
Nomination for Coolest App in 2007 |
| 9:03PM |
3 |
GTalk/Jabber passing audio in 1.4.1! |
| 3:21PM |
1 |
Cancelling a digit in IVR |
| 3:21PM |
0 |
Add current caller to junk-callers-database |
| 2:57PM |
1 |
Compiling smsq in 1.2 |
| 1:44PM |
2 |
Manager.conf '127.0.0.1 unable to authenticate' |
| 1:11PM |
1 |
How many gsm channels |
| 12:02PM |
1 |
Building a new voicemail system... Testers needed! |
| 11:50AM |
0 |
Long term voicemail archival and synchronisation between multiple storage locations? |
| 7:21AM |
1 |
Linksys PAP2 and Caller ID |
| 7:11AM |
0 |
chan_cellphone won't pair with phone |
| 6:28AM |
0 |
Pickup failover |
| 6:22AM |
2 |
zaptel 1.4 on fedora core 6 with dell pe 2850 |
| 6:00AM |
0 |
Ringing does not terminate on mISDN after pickup |
| 5:22AM |
1 |
Asterisk 1.4.0 Installation error on Red Hat Linux 9.0-Urgent |
| 5:16AM |
2 |
Polycom 501 - Auto answer on one line appearance |
| 3:11AM |
1 |
visdn, misdn and the hell |
| 2:15AM |
0 |
[asterisk_voip] asterisk and ogg files |
| 1:52AM |
3 |
Micros-Fidelio - billing in hotel |
| 1:21AM |
0 |
web based sipphone |
| 12:57AM |
1 |
preventing voicemail pickup after SIP redirect ? |
| |
| Monday March 5 2007 |
| Time | Replies | Subject |
|---|
| 10:39PM |
0 |
app_queue not using exit context? |
| 10:31PM |
2 |
IAX2, DTMF and x86_64. |
| 9:41PM |
1 |
server generated outbound conference calls? |
| 7:21PM |
4 |
Polycom Questions |
| 6:10PM |
1 |
extra-sounds 1.4.5 timestapmed newer than 1.4.6 ??? |
| 5:42PM |
1 |
[Announce] Web-MeetMe V3.0.1 released |
| 5:04PM |
2 |
Using Asterisk as Voicemail Server on a dinosaur Meridian System |
| 4:37PM |
1 |
Voicemail question |
| 4:05PM |
1 |
Setting Sip Headers From Dial App? |
| 3:48PM |
1 |
g.729 on solaris10/x86 |
| 3:15PM |
6 |
A New Phone Service - www.virtualphoneline.com |
| 2:02PM |
1 |
Re: Asterisk Java w/ Threads |
| 11:53AM |
2 |
Rx+,Rx-,Tx+,Tx- of TE110P |
| 10:54AM |
2 |
TDM400P/FXS in a HP DL380 G5 |
| 10:11AM |
1 |
How to disable MOH completely? |
| 8:30AM |
4 |
TC400B |
| 4:59AM |
1 |
SMS ON ASTERISK |
| 4:44AM |
1 |
HITBSecConf2007 - Malaysia: Call for Papers now Open |
| 1:21AM |
1 |
new kernel and zaptel |
| 12:39AM |
2 |
Read() status? |
| 12:09AM |
1 |
Is the 1.0.X branch vulnerable to the SIP issue? |
| |
| Sunday March 4 2007 |
| Time | Replies | Subject |
|---|
| 4:41PM |
1 |
Configurations Files of TE110P |
| 2:20PM |
1 |
running error, unable to load *.conf files: load_modules: No 'modules.conf' found - svn version 1.4.1 |
| 1:45PM |
1 |
Real Time, sip.conf, [general] |
| 12:45PM |
0 |
x100p.com |
| 11:12AM |
2 |
When does local leg in call file start? |
| 6:32AM |
1 |
running error: load_modules: No 'modules.conf' found - vesrion 1.4.1 from svn |
| 4:03AM |
1 |
Voice quality issues |
| 2:01AM |
2 |
So does 1.4.1 show up in the /branches/1.4, or only in the tags/1.4.1 |
| |
| Saturday March 3 2007 |
| Time | Replies | Subject |
|---|
| 7:08PM |
3 |
dial question |
| 4:10PM |
0 |
creating new asterisk application |
| 1:17PM |
2 |
hanging an asterisk box off of a PBX analog extension |
| 3:43AM |
1 |
gtalk2voip and Asterisk |
| 12:29AM |
1 |
Asterisk - e164 (enum) lookup confused |
| |
| Friday March 2 2007 |
| Time | Replies | Subject |
|---|
| 10:28PM |
1 |
How to fail an AGI |
| 7:13PM |
2 |
Need comparison between PBXtra, Trixbox, Thirdlane, Druid, Aheeva etc. |
| 5:05PM |
4 |
Asterisk 1.4.1 Released |
| 5:05PM |
0 |
Asterisk 1.2.16 Released |
| 5:05PM |
0 |
Zaptel 1.2.15 Released |
| 4:48PM |
1 |
How to log VERBOSE statement to a file? |
| 2:21PM |
1 |
IP addresses |
| 2:20PM |
1 |
svn 1.4 - mp3 support and changing the installation directory |
| 1:46PM |
3 |
REMOTE CRASH FIX |
| 1:02PM |
4 |
rtsavesysname not working in 1.4 |
| 12:37PM |
2 |
PRI progress codes. |
| 11:00AM |
0 |
FW: Alec Saunders post about Mashable Telco's |
| 10:36AM |
3 |
DTMF from TDM400P and X100P |
| 10:17AM |
1 |
Double DTMF digits sent on IAX native bridge |
| 10:15AM |
1 |
Voicemail to SMS using asterisk |
| 9:56AM |
5 |
Help Voicemail to SMS using asterisk |
| 9:21AM |
0 |
WMI from Asterisk to Cisco Call Manager |
| 8:56AM |
1 |
DTMF detection problems on PRI channels? |
| 8:35AM |
3 |
Alec Saunders post about Mashable Telco's |
| 8:29AM |
7 |
Asterisk and Fax |
| 7:38AM |
1 |
T38 |
| 7:26AM |
1 |
cmd page crashes Asterisk SVN-branch-1.4-r57207 |
| 4:15AM |
1 |
Running Fax on E1 line |
| 3:46AM |
1 |
BLF not working with Asterisk 1.4.0 |
| 3:30AM |
0 |
Rif: Re: Multiple simultaneous calls |
| |
| Thursday March 1 2007 |
| Time | Replies | Subject |
|---|
| 8:28PM |
1 |
Test |
| 7:53PM |
2 |
How can I use the "GET VARIABLE variablename" in AGI |
| 5:07PM |
1 |
2 Call locations |
| 5:05PM |
2 |
Polycom reject button |
| 4:57PM |
1 |
gtalktovoip and Asteirsk |
| 4:46PM |
1 |
Digium S101i - pickupexten doesn't work |
| 4:05PM |
1 |
build rpm fails |
| 2:39PM |
2 |
Asterisk 1.4.1 |
| 11:51AM |
1 |
Tesco Internet Phone |
| 10:12AM |
5 |
Asterisk Realtime |
| 10:11AM |
0 |
About queues and multiple lines. |
| 9:47AM |
4 |
Multiple simultaneous calls |
| 9:31AM |
1 |
Extensions +International |
| 9:24AM |
0 |
Testing asterisk with sipp |
| 8:54AM |
2 |
blieve i my TE110P or My teleco provider ?? |
| 7:50AM |
7 |
IAX best practices |
| 6:41AM |
4 |
Cannot hear ringback music from telco |
| 5:46AM |
0 |
3 way calling independent of phone hw. |
| 4:57AM |
3 |
UK SIP Gateway |
| 4:27AM |
0 |
Revolution Call Accounting Desktop |
| 4:21AM |
1 |
TDM400p Loaded only once |
| 4:19AM |
0 |
Issue with Calling Name ID in SIP: Asterisk sets Caller ID Number as Name if NO Name |
| 1:40AM |
1 |
transfer function |
| 1:25AM |
0 |
Siemens HiPATH 3700 with Asterisk |
| 1:22AM |
2 |
DTMF not being detected with 1 provider. Works with the other provider... |