Thursday April 30 2009 |
Time | Replies | Subject |
10:46PM |
0 |
Voicemail Caller ID |
7:22PM |
1 |
Registration of 'cstore' rejected: 'Registration Refused' from: '62.213.196.38' |
7:12PM |
1 |
test |
4:37PM |
2 |
Proper install order for Asterisk and it's related packages.. |
4:17PM |
2 |
Asterisk and 4G |
4:10PM |
1 |
Wanpipe |
3:45PM |
1 |
${CALLERID(name)} question |
3:37PM |
2 |
Question with Asterisk and call waiting ${CALLERID(name/num)} |
3:31PM |
1 |
rtsp help |
3:06PM |
1 |
FW: Update: HD Communications Summit in NYC |
1:35PM |
0 |
Saturday May 2 - Asterisk @ Global FSW Conference via VOIP - BerkeleyTIP - 21 Videos - For forwarding |
12:44PM |
3 |
need help on asterisk call forwarding |
11:50AM |
1 |
Asterisk or Zaptel Issues |
11:31AM |
0 |
Asterisk and Shoretel integration |
3:34AM |
0 |
valetparking.c |
2:01AM |
0 |
automon *1 not working; asterisk-1.4.22.1 |
1:20AM |
1 |
ExtenSpy d option 1.6 |
|
Wednesday April 29 2009 |
Time | Replies | Subject |
9:39PM |
5 |
2nd Parking Lot |
9:29PM |
1 |
US Caller ID |
7:14PM |
5 |
What do I need to connect landline calls without telephony hardware? |
6:32PM |
1 |
Bounty for parking on <slot>@<context> |
6:25PM |
0 |
Verifone-Asterisk-AGI |
5:46PM |
3 |
Asterisk sudden crash |
5:41PM |
2 |
Something wrong with DAHDI signalling according to the CLI |
12:49PM |
1 |
problem in upgrading to 1.6.1.0 |
12:27PM |
1 |
Replacement of Macro() with Gosub() |
6:36AM |
2 |
Cisco SPA525G |
1:39AM |
3 |
I am looking for a good source of Monterrey DIDs |
|
Tuesday April 28 2009 |
Time | Replies | Subject |
11:39PM |
0 |
Asterisk 1.6.1.0 Now Available |
11:37PM |
0 |
Asterisk-Addons 1.6.1.0 Now Available |
11:36PM |
0 |
Asterisk-Addons 1.4.8 Now Available |
8:35PM |
1 |
Asterisk 1.6 and CDR/MySQL |
6:19PM |
1 |
asterisk -C option not honored 100% |
5:00PM |
1 |
Asterisk-Verifone-Agi |
3:49PM |
2 |
How to get PBX's clock with AMI? |
1:04PM |
0 |
Explain when DIALSTATUS is set to CANCEL |
10:15AM |
1 |
no source on calllogs |
9:35AM |
1 |
Call recording - posible to remove recorded file at the end of the call |
7:27AM |
1 |
finding the right amd.conf settings |
12:18AM |
3 |
POS modems |
|
Monday April 27 2009 |
Time | Replies | Subject |
9:33PM |
2 |
Who has the clever Polycom upgrade system? |
9:26PM |
1 |
Where I get free VoiP-in numbers? |
9:08PM |
1 |
IPv6 support? |
8:41PM |
1 |
Packet2packet bridging while in sip.conf canreinvite=no |
6:38PM |
0 |
SIP infrastructure |
5:29PM |
2 |
Change Termination of Read Command |
4:17PM |
1 |
No format for saving voicemail? |
3:09PM |
3 |
Diference between volume of mp3 and wav files |
2:20PM |
0 |
Going to AMOOCON? |
9:36AM |
1 |
music on hold using mms |
6:37AM |
4 |
[UK SPECIFIC] DAHDI and a OpenVox Card |
2:57AM |
3 |
Video Conference Software (Open Source) |
|
Sunday April 26 2009 |
Time | Replies | Subject |
11:51PM |
1 |
Error, Clue to what? |
7:03PM |
1 |
1.6.1: "DNS error" but ping works |
6:52PM |
1 |
sipgate doesn't work with sipgate anymore |
6:28PM |
1 |
file.c:655 ast_openstream_full: File /tmp/winkel-gesloten.alaw does not exist in any format |
2:26PM |
4 |
1.6.1: menuselect has problems with x86_64 ?? |
12:33PM |
3 |
Digium fax force T38? |
9:48AM |
0 |
FW: issue with sip 180 responses |
2:21AM |
5 |
Digium fax failing |
|
Saturday April 25 2009 |
Time | Replies | Subject |
2:37PM |
3 |
Outgoing Queues |
10:38AM |
0 |
Information required about ADM |
10:03AM |
4 |
64bit: any problems with asterisk? |
7:35AM |
0 |
sip.conf RTP settings |
4:43AM |
1 |
Callweaver/Asterisk 'outgoing' spool |
4:01AM |
1 |
Can't dial out until I dial in once |
|
Friday April 24 2009 |
Time | Replies | Subject |
10:30PM |
2 |
Asterisk & EC2 |
9:46PM |
1 |
Digium Fax for Asterisk |
8:19PM |
2 |
voicemail number of rings |
5:55PM |
0 |
dahdi_tool reports that dahdi_dummy is UNCONFIGURED |
5:03PM |
0 |
Dialtones as Progressinband |
4:06PM |
1 |
Asterisk 1.6.2 Beta |
2:33PM |
0 |
Duplicating existing PBX function |
1:59PM |
2 |
listen to prompt before bridging call. |
1:55PM |
1 |
FOP and UserEvent() |
10:09AM |
3 |
timing source problem |
7:39AM |
0 |
Friday Apr 24 @ 12 Noon: Wideband, or HD Voice as Polycom calls it |
7:35AM |
1 |
function originate |
7:07AM |
2 |
Feature request: manager show events |
4:03AM |
2 |
cheap CHEAP ata |
2:22AM |
1 |
Hangup Detection After Originate (Asterisk Manager API) |
|
Thursday April 23 2009 |
Time | Replies | Subject |
11:54PM |
1 |
Dahi-tools Compilation on Ubuntu/Xen |
10:01PM |
0 |
about Asterisk and AudioCodes FXO/H323 |
8:56PM |
1 |
BLINDTRANSFER and SIP hardphones |
8:43PM |
4 |
want to set up text based "adventure" for asterisk |
8:30PM |
0 |
dial and transfer while ringing |
8:27PM |
3 |
Record in mp3 |
8:08PM |
1 |
Convert file in GSM codec to G729 codec |
7:14PM |
2 |
Asterisk on Mac OS X |
6:53PM |
0 |
Libpri-1.4.10 Released |
5:49PM |
2 |
CDR issue |
4:26PM |
3 |
AGI PHP script |
3:00PM |
9 |
AMD Not Working |
2:38PM |
0 |
Fritz USB 2.1 on Asterisk 1.4.22 / trixbox |
2:24PM |
1 |
Dial-out via AMI |
2:20PM |
0 |
Do I need G729 codec for wholesale ? |
1:18PM |
2 |
Zaptel Not Releasing Channel (PRI) |
1:01PM |
1 |
Asterisk Double Invite |
12:49PM |
0 |
UserEvent doc : is Uniqueid mandatory in 1.6 |
11:49AM |
1 |
Cause 34 still there |
11:12AM |
2 |
Asterisk Capacity |
9:50AM |
3 |
Compact, fanless appliance? |
9:46AM |
0 |
Howto see the source ip address of SIP call in cli monitor |
9:29AM |
1 |
Asterisk and HUD server |
8:53AM |
3 |
Parked calls for multiple customers |
3:10AM |
1 |
voice quality |
|
Wednesday April 22 2009 |
Time | Replies | Subject |
11:42PM |
5 |
Step-by-Step Asterisk and Cisco 1760 Help |
9:18PM |
0 |
random hangups: how to debug? |
8:19PM |
0 |
how to know the channel from the iax phone side? |
6:34PM |
1 |
Upgrading from 1.4.21.2 to 1.6.0.5 breaks sql queries with backslashes? |
5:07PM |
0 |
HI |
3:56PM |
2 |
Conference problem |
2:00PM |
0 |
Zaptel tone debug |
1:31PM |
1 |
How to get to 10.000 open calls |
12:55PM |
0 |
[asterisk-dev] How to get to 10.000 open calls |
10:26AM |
1 |
Should you use UserEvents for monitoring calls ? |
10:09AM |
0 |
E1 not synchronized |
8:57AM |
0 |
mISDN DTMF endless tone |
8:12AM |
1 |
CDR feature not working properly for "failed call attempt" |
5:55AM |
1 |
Queue() Ignore Hangup Request |
5:14AM |
5 |
Asterisk routine maintenance activities |
1:58AM |
1 |
Faxing and TIFF files |
|
Tuesday April 21 2009 |
Time | Replies | Subject |
8:42PM |
0 |
Asterisk 1.6.1.0-rc5 Now Available |
5:48PM |
1 |
run dialplan when open line |
1:53PM |
4 |
Polycom wideband codecs? |
12:36PM |
1 |
Should I go for Asterisk 1.6 ? |
11:13AM |
4 |
Asterisk Database |
10:57AM |
0 |
Cleared Event Log |
10:02AM |
5 |
Asterisk process ended |
|
Monday April 20 2009 |
Time | Replies | Subject |
10:02PM |
6 |
Peer 'iaxfax' is now UNREACHABLE! Time: 3 |
7:13PM |
1 |
Asterisk PA system with cepstral |
5:33PM |
1 |
AstDB & MixMonitor queries |
2:45PM |
2 |
Voice mail does not contain a time? |
2:31PM |
2 |
Execute after hangup |
1:48PM |
0 |
1.4.24.1 freezing randomly - what files to use to downgrade |
12:24PM |
1 |
Zaptel to Dahdi |
9:34AM |
1 |
T38 fax failing |
7:47AM |
1 |
Asterisk 'outgoing' directory |
6:10AM |
2 |
About Asterisk 1.6 web GUI |
3:50AM |
2 |
Asterisk 1.4 to 1.6 extensions.conf |
3:06AM |
0 |
Note for all regarding Asterisk Addons |
12:29AM |
1 |
Asterisk Addons - part 2 |
12:01AM |
1 |
Asterisk addons - disable H323 |
|
Sunday April 19 2009 |
Time | Replies | Subject |
7:55AM |
1 |
issue with sip 180 responses |
12:04AM |
3 |
asterisk-1.6.0.9-x86_64: voicemail: Segmentation fault (core dumped) |
|
Saturday April 18 2009 |
Time | Replies | Subject |
11:01PM |
0 |
Astlinux 0.6.5 upgrade released |
7:32PM |
1 |
Insecure= |
5:21PM |
2 |
dialling multiple extensions in an internal context |
2:48PM |
2 |
Digium Fax for Asterisk questions |
11:26AM |
0 |
do i need to install libpri |
4:40AM |
0 |
Callweaver TXfax queuing |
1:23AM |
1 |
Origination and Termination |
|
Friday April 17 2009 |
Time | Replies | Subject |
11:57PM |
1 |
Sangoma A104d and Adtran 850 problems |
7:51PM |
0 |
Quick and Dirty HOW-TO for Enum on Asterisk 1.2? |
6:47PM |
15 |
Here is Step by Step Example of Asterisk PBX System Install and configuration |
4:51PM |
1 |
opening 2 and more channels on 1 SIP account |
4:50PM |
3 |
Getting Zaptel and Asterisk Link, also Dahdi |
4:20PM |
3 |
Alcatel OmniPCX Enterprise + Asterisk with E1 |
2:28PM |
2 |
Jabber and Presence |
1:37PM |
1 |
2BCT last mile... Hopefully |
12:22PM |
1 |
1.4.21.1 - weird freeze |
12:14PM |
3 |
Digium G.729 licenses |
11:52AM |
1 |
MOH always plays from the start |
6:30AM |
1 |
how to call forward on 1.6 |
3:12AM |
0 |
Canreinvite after media connection |
3:08AM |
0 |
Stun clients and canreinvite |
|
Thursday April 16 2009 |
Time | Replies | Subject |
7:57PM |
1 |
sending AT commands through the SIP channel to the end device?! |
7:12PM |
2 |
Sequential Ring Groups? |
6:50PM |
1 |
AMI IAXPeers |
6:44PM |
1 |
Set CDR(src) from dialplan |
3:06PM |
2 |
Simultaneous Calls at a time |
2:52PM |
1 |
Connection to non-human numbers |
2:08PM |
1 |
Problem transferring calls between Cisco 7940 with SIP firmware |
11:17AM |
1 |
Can Asterisk bridge between a SIP client and a Cisco Call |
10:59AM |
7 |
How to send "404 Not found" SIP reply? |
9:57AM |
0 |
mISDN ports and dstchannel CDR logging |
9:36AM |
1 |
AGI Programming |
9:22AM |
1 |
ISDN from Macau CTM |
8:37AM |
1 |
Remote BLF / hint on IAX2 trunk |
8:09AM |
0 |
reinvite problem |
6:50AM |
1 |
Friday Apr 17th @12 Noon ET: Digium's Open Source Asterisk Support |
2:41AM |
2 |
TDM2400P dial tone is not present on phones, but the phone ring with incoming calls |
|
Wednesday April 15 2009 |
Time | Replies | Subject |
3:59PM |
2 |
inbound filed |
2:01PM |
1 |
pickupexten *8 |
9:30AM |
0 |
chan_mISDN with asterisk version 1.4.22 + codec negotiation patch |
9:30AM |
1 |
What is "WARNING: Got 200 OK on REGISTER that isn't a register"? |
9:25AM |
1 |
astcanary not exiting in asterisk V1.6.1 |
|
Tuesday April 14 2009 |
Time | Replies | Subject |
11:22PM |
0 |
RTCP ports |
9:52PM |
6 |
2B Channel Transfer on XO-based T1 |
6:47PM |
0 |
Gxp 2000 softkey question |
6:30PM |
0 |
Asterisk Dial Pagers And Enter Callback Numbers |
6:19PM |
0 |
FW: Asterisk-beginner : cannot make phone calls using Asterisk |
5:14PM |
3 |
Ring All Queue |
4:55PM |
1 |
Vacation reply user nuked |
4:42PM |
1 |
OT - snom phone question |
4:05PM |
2 |
What means? Correct auth, but based on stale nonce received |
3:41PM |
0 |
How are you doing recently? |
2:39PM |
2 |
MOH |
12:30PM |
0 |
SRTP testers needed |
11:39AM |
5 |
.GSM -> .WAV (or ,MP3) Conversion |
11:00AM |
1 |
T38modem in loopback mode does not work on asterisk 1.4.20.1 |
10:04AM |
2 |
Exit Dial Application |
9:48AM |
0 |
OT - Define what substitution is ... |
8:37AM |
1 |
SIP and FW settings |
6:16AM |
4 |
Ignoring time spent waiting in queue in CDR |
1:29AM |
2 |
dynamic menus in dialplan |
12:28AM |
3 |
Changing menuselect values from CLI and not TUI |
|
Monday April 13 2009 |
Time | Replies | Subject |
10:19PM |
1 |
wanpipe 3.2.7.1 Compiling error |
9:32PM |
3 |
duration of rfc2833 generated dtmf |
8:49PM |
0 |
Asterisk-beginner : cannot make phonecalls using Asterisk (update) |
5:37PM |
0 |
MySQL queries |
4:18PM |
10 |
Asterisk-beginner : cannot make phonecalls using Asterisk |
2:30PM |
2 |
Agents on asterisk |
12:33PM |
0 |
Clock problem with TE122 |
11:05AM |
1 |
Send Re-invite from Dialplan application? |
11:02AM |
0 |
Sending Re-Invite with Dialplan application? |
10:11AM |
0 |
opensips and asterisk canreinvite |
4:19AM |
1 |
FAX reliability |
12:30AM |
2 |
retransmision error con asterisk 1.4.24.1 |
|
Sunday April 12 2009 |
Time | Replies | Subject |
2:38AM |
0 |
problem with asterisk 1.4.24.1 |
|
Saturday April 11 2009 |
Time | Replies | Subject |
8:44PM |
1 |
asterisknow 1.5 with X100P and TDM400P |
4:04PM |
2 |
[OFF TOPIC] wich virtualization solution to use? |
10:29AM |
1 |
Voicemail Greetings Will Not Save |
7:04AM |
2 |
[astersik-users] ss7 consultancy $1000 USD |
3:04AM |
1 |
Is there documentation explaining res_config_curl? |
2:55AM |
2 |
OT XEN asterisk and a digium board |
|
Friday April 10 2009 |
Time | Replies | Subject |
6:22PM |
0 |
one-button call parking/pickup on Asterisk with Polycom phones? |
4:04PM |
2 |
IVR Survey |
12:25PM |
0 |
Asterisk 1.4.24 and Gtalk audio failure |
11:15AM |
0 |
Friday April 10th: Google Voice, Asterisk Open-Source Support, etc. |
6:06AM |
3 |
Can Asterisk bridge between a SIP client and a Cisco Call Manager server? |
4:54AM |
0 |
IVR and DTMF |
|
Thursday April 9 2009 |
Time | Replies | Subject |
10:41PM |
2 |
DTMF |
9:00PM |
1 |
Looking for good IAX ATA |
8:59PM |
6 |
MeetMe not working - was before |
4:45PM |
1 |
Check sip availability |
4:36PM |
3 |
T.38 ATAs |
1:44PM |
4 |
asterisk command line problem |
1:42PM |
2 |
notifyringing=no does not work |
9:12AM |
0 |
AstManProxy and broadcast |
2:50AM |
2 |
Softphone question |
|
Wednesday April 8 2009 |
Time | Replies | Subject |
11:06PM |
3 |
Multi Frequency Cycle Timeout - E1-R2 METROTEL COLOMBIA |
9:27PM |
1 |
Perl AGI |
5:19PM |
5 |
Zopier Client |
3:58PM |
1 |
__ast_read: ast_read() called with no recorded file descriptor |
1:13PM |
2 |
Asterisk Trunk billing |
7:05AM |
4 |
Siemens Gigaset Phones get mute function. |
6:44AM |
0 |
Asterisk and Voice Recognition Sphinx |
4:56AM |
1 |
Call Pickup Works w/Linksys ATA, not with Cisco 7940G |
2:30AM |
0 |
chan_mobile sms compatible phone |
|
Tuesday April 7 2009 |
Time | Replies | Subject |
10:05PM |
1 |
dahdi_dummy: Unable to register DAHDI rtc driver |
7:53PM |
1 |
i have a probleme and my asterisk and ovh |
7:49PM |
1 |
Fwd: [asterisk-r2] MFCR2 With Unicall - can unicall affect zaptel commands? |
3:54PM |
3 |
Best Practice Advice? |
2:38PM |
1 |
AEL2, BASE64_DECODE and hexadecimal |
2:28PM |
2 |
immediate=yes is populated to all channels |
2:26PM |
2 |
Grandstream blind transfer issue |
1:51PM |
0 |
Zaptel connectivity issues |
1:00PM |
3 |
Logging Asterisk console |
9:41AM |
1 |
OT - SIP MESSAGE, newline chars and formatting |
8:16AM |
2 |
app_backticks and 1.6 |
5:23AM |
2 |
asterisk and patton |
12:41AM |
1 |
Australian NBN network announced |
|
Monday April 6 2009 |
Time | Replies | Subject |
10:11PM |
3 |
One way AUDIO |
8:55PM |
2 |
Zaptel Config |
7:55PM |
2 |
Hacked |
7:17PM |
0 |
Asterisk 1.6.0.9 Now Available |
7:14PM |
0 |
Asterisk 1.6.1.0-rc4 Now Available |
7:04PM |
3 |
Sangoma and BT single lines |
6:48PM |
1 |
IMAP Voicemail - can't get messages. Arrgh! |
4:13PM |
1 |
IOS Interface |
3:00PM |
1 |
Douds it |
2:40PM |
1 |
Relay ringing sip message 180 |
1:32PM |
0 |
OT - Call forwarding services for corporate users |
1:01PM |
2 |
IPkall |
12:45PM |
1 |
SIP Registration and INVITE question |
12:39PM |
2 |
25-50-100fxs |
10:00AM |
1 |
fail to retrieve the calling party information |
9:45AM |
0 |
app_queue.c: No one is answering queue |
8:42AM |
0 |
Mysql cache delay |
7:24AM |
1 |
Off-topic: SIP DTMF most supported method |
|
Sunday April 5 2009 |
Time | Replies | Subject |
7:43PM |
2 |
what can we do with lost voice packet on a congestioned VPN? |
5:07PM |
6 |
Inexpensive device for bandwidth management |
|
Saturday April 4 2009 |
Time | Replies | Subject |
10:07PM |
1 |
off topic - voip providers raided by FBI for unpaid telecom bills: |
5:51PM |
0 |
OT - Car Warranty Calls - OT |
12:31PM |
4 |
Advice |
8:02AM |
0 |
TODAY April 4 -Global FSW Voice Meeting BerkeleyTIP -Linus, Guido, Shuttleworth... |
|
Friday April 3 2009 |
Time | Replies | Subject |
11:27PM |
2 |
Live Support function? |
9:41PM |
3 |
Grandstream surveillance devices |
7:42PM |
1 |
conference calling |
7:39PM |
1 |
Eicon Diva 2.01 PCI Passive BRI ISDN card |
6:57PM |
1 |
Using multiple 'peer' identities on one phone with 1.4 |
5:45PM |
1 |
SIP Warnning Message |
5:43PM |
1 |
Radio interfaces for Asterisk - ISO image distro |
5:14PM |
1 |
ISDN Timer T309 |
4:02PM |
1 |
Seg Fault after upgrade to Asterisk 1.6.0.8 |
2:40PM |
1 |
Bridging Avaya IP systems and Cisco IP system |
2:30PM |
2 |
New ViciDial Call Center Suite Release: 2.0.5 |
1:24PM |
1 |
Unichan wtih Te201p alarms |
1:22PM |
1 |
agi no longer working with 1.4 svn 186229 |
8:14AM |
0 |
VoIP Farm |
6:23AM |
0 |
Asterisk and Call Manager |
|
Thursday April 2 2009 |
Time | Replies | Subject |
11:36PM |
2 |
Dahdi, TE220 Device, and Asterisk Problem |
11:22PM |
5 |
Ring group howto |
10:38PM |
1 |
Simple Queue question |
9:40PM |
1 |
Anyone actually built h323plus on Fedora? |
9:38PM |
1 |
FXS Line Voltage When Dahdi/Zaptel is off? |
8:57PM |
3 |
2-3 Calls at a time |
8:53PM |
3 |
problema con una x100p |
8:51PM |
2 |
cant get a x100p works |
6:59PM |
0 |
AST-2009-003: SIP responses expose valid usernames |
6:51PM |
0 |
Asterisk 1.2.32, 1.4.24.1, and 1.6.0.8 Now Available |
6:25PM |
3 |
Asterisk G729 codec... |
6:20PM |
0 |
VB6 to HUD Pro Integration |
5:22PM |
4 |
meetme dahdi and zaptel |
4:44PM |
1 |
T1/PRI ignore answer signal |
4:39PM |
0 |
Magic List: Thanks Shain Rufeel & Danny Nicholas. |
4:22PM |
2 |
opermode=? |
4:07PM |
1 |
Asterisk + Cisco Call Manager |
3:54PM |
3 |
Nothing at /proc/zaptel with new Digium TE201 |
3:45PM |
1 |
SIP vs RTP destination IP |
3:45PM |
1 |
fxotune and the bug |
3:37PM |
4 |
FXO Ignore ring |
3:17PM |
0 |
Asterisk SIP trunk to Cisco IAD2400 |
2:01PM |
2 |
activate telco redirection service from Asterisk |
11:39AM |
4 |
Mountain ahead of me! |
11:06AM |
2 |
Xorcom and Doorbell |
10:38AM |
1 |
Friday April 3rd Gizmo, OpenSky, Skype for Asterisk, SIP for Skype - where are they? |
7:30AM |
1 |
Trying to test my voicemail |
3:46AM |
4 |
400 calls at g711 how much cpu power |
3:27AM |
0 |
SIP topology hiding |
|
Wednesday April 1 2009 |
Time | Replies | Subject |
9:19PM |
0 |
Asterisk-Addons 1.6.2.0-beta1 Now Available |
8:55PM |
0 |
Asterisk 1.6.0.7 Now Available |
7:45PM |
1 |
SIP Context Confusion |
3:45PM |
1 |
Trunk SIP and configuration |
3:26PM |
1 |
login-logout asterisk |
3:14PM |
2 |
Extract a MOS value from Asterisk CDR |
1:06PM |
1 |
Remote host can't match request CANCEL to call |
12:18PM |
0 |
stress asterisk voicemail |
7:18AM |
10 |
FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVER FOR ASTERISK RELEASED TODAY |
4:59AM |
0 |
IAX2 transfer=force |