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Hi all, This plugin lets you specify which actions *must* be run under ssl (https). If a declared action is run without ssl, the user is redirect to https. Also, once you declare an action to "require_ssl", any links to that action are going to be https:// links. http://svn.ianwarshak.com/plugins/secure_actions Hopefully you all will find this useful. Ian

On Thu, 2004-01-15 at 19:18, dkwok@iware.com.au wrote: >> I do not have any zaptel hardware on the Asterisk box, I could not have >> meetme functioning. I did modify the Makefile in zaptel directory on >> line 168 by including ztdummy as one of the modules to compile in. try modprobe ztdummy This works. Should I...

I have just checked the Openbsd box on the if interface. fxp0: flags=8843<UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST> mtu 1500 address: 00:02:55:30:54:28 media: Ethernet autoselect (100baseTX full-duplex) status: active inet 192.168.1.1 netmask 0xffffff00 broadcast 192.168.1.255 inet6 fe80::202:55ff:fe30:5428%fxp0 prefixlen 64 scopeid 0x1 xl0:

Chris Albertson wrote: |What firmware version do you have? program version 1.0.4.39 -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url :

I do not have any zaptel hardware on the Asterisk box, I could not have meetme functioning. I did modify the Makefile in zaptel directory on line 168 by including ztdummy as one of the modules to compile in. The error message from the concole: -- Executing MeetMe("SIP/1002-e9ca", "4700") in new stack == Parsing '/etc/asterisk/meetme.conf': Found

For people who are using GS 101, what do you think the sidetone generated by the phone. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssionally. The volume of the sidetone can be turned down using the volume button but it also control the

Referring to my previous post about degradation of voice quality when having more than 2 connection. The actual route is: pc xlite -> local asterisk box -> iaxtel -> local asterisk I have tried out a different situation: pc xlite -> local asterisk box -> iaxtel and the second connection pc xlite -> local asterisk box -> iaxtel -> local asterisk The same degradation

In relation to voice degradation when having 2 or more connection to Asterisk. The comment on the network setup is quite possible. I am not too familiar with linux. How do I check whether the asterisk server's nic is running at full-duplex mode. Does Asterisk use the sound card on the box to do voice processing? I am running xlite on 2 pc and making calls through iax, FWD and back to my

This is ifconfig on openbsd box: fxp0: flags=8843<UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST> mtu 1500 I think this output shows that the fxp0 interface is on simplex mode. The voice degradation I referred was by using xlite soft phone. I open 2 line similtaneously and dial to FWD and back to my incoming extension. Xlite is runnning on a w2k box with realtek 100M nic in auto mode. I can

Any one has documented how-tos for making voicetronix openline 4 to work with Asterisk. I have been contacting Australian Digium resellers and Digium cards are not approved in Australia. So I suppose Australian users are interested into putting Voicetronix in use. Any expereience to share will be most appreciated. David Kwok -------------- next part -------------- A non-text attachment was

My ADSL connection speed is 512Kb up and 128Kb down. When making calls from Asterisk to IAX and back to the Asterisk, the sound is choppy and 20% of voice messages was lost. What is the production bandwidth requirement per internet call. I understand there is no guarantee of QoS but at least a benchmark to follow. -- David Kwok Iaxtel/FWD # 17001813482 -------------- next part

This may be of interest -- as a tool we can use to test our systems and as a weapon that may be used against us :) http://warvox.org/ A brief read-over looks like it uses iaxclient and ruby to war dial a range of numbers and record audio samples to be analyzed to identify if the call was answered by a modem, fax machine, human, etc. The calls are placed through a PSTN termination

I am running v2.0.7 and have set up a network drive for an accounting ledger system. The software is called MYOB and is quite popular in Australia. This is the first time I have to deal with multi user access to 1 data file. My setup is: Global oplocks = yes socket options = TCP_NODELAY socket options = IPTOS_LOWDELAY [MYOB] path=/home/office/MYOB force group = office directory mask = 0770

Just got GS 101 phone and plugged into the network. Got ip setup however, the following problems arise: 1. when dialing an extension, I cannot further send any key tone to Asterisk. 2. there is no sound coming from the other end. I have a sip.conf setup for GS: [General] disallow=all allow=ulaw allow=alaw [gs] canreinvite=no dtmfmode=info In the GS101 setting rtp port = 5004 sip port = 5060

...issue. Check that the ID and password values are correct. Use prserv, Ethereal, etc. to see if you are registering correctly. If so, and still no dial tone, this is probably also a hardware failure. --Stewart -----Original Message----- Date: Sat, 5 Jun 2004 20:13:23 +1000 (EST) From: <dkwok@iware.com.au> To: <asterisk-users@lists.digium.com> Subject: [Asterisk-Users] change cisco ata 186 dial behaviour Reply-To: asterisk-users@lists.digium.com I have ata-186 and grandstream connected to asterisk using sip. I have a voip account with ATP, in Australia. In order to ring HK, I need...

...; http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > --__--__-- > > Message: 10 > Date: Tue, 20 Jan 2004 17:20:46 +0100 > From: dkwok <dkwok@iware.com.au> > Organization: iware.com.au > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] echo cancellation > Reply-To: asterisk-users@lists.digium.com > > This is a cryptographically signed message in MIME format. > > --------------ms02080301030004060104080...

-----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of empire underground Sent: Friday, January 09, 2004 1:32 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Screen Pop & Remote Agents > can I put a .csv file in the sql DB and have it dial from there? and will I be able to set a > Dial Plan to

I have looked at matt's asterisk gui client at sourceforge. I am not a programmer by trade. The documentation there seems to be a bit lacking. Has anyone have the experience in installing the gui client and may perhaps have a how-to document available for sharing. -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 -------------- next part -------------- A non-text

Hi Everybody, Got a weird problem here I think. Got a setup with an asterisk (current from cvs as of a few hours ago) in a box with an el-cheapo ISDN BRI card connected to the PSTN network and two Snom phones internally (one Snom-100 and one Snom-200). Dialing between the snom phones or dialing out to PSTN from any of the snom phones works perfectly. But when I receive a call FROM the PSTN

Hi All Will cisco 7910 ip phone compatible with Asterisk? I know that 7960 are fine. David Kwok -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040112/e8023f35/smime.bin