[Asterisk-Users] Asterisk User Group in Winnipeg, CA
We are forming an Asterisk User Group in Winnipeg. Our first meeting will be sometime in the last half of February. If you are interested in participating please join our mailing list: http://www.muug.mb.ca/mailman/listinfo/asterisk Look forward to seeing you at our first meeting, Bill ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 3000 user guide is now available
The "SPA User Guide" now covers configuring the 3000. http://www.sipura.com/support/index.htm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users List Etiquette
For me the big issue of html is in the message digests. Since the html is mixed in with plain text browsers do not detect the html. For individual messages HTML is generally not a problem. Subject: RE: [Asterisk-Users] Asterisk-Users List Etiquette From: Steven Critchfield <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Date: Tue, 15 Jun 2004 10:10:12 -0500 Reply-To: [EMAIL PROTECTED] On Tue, 2004-06-15 at 08:35, Rich Adamson wrote: Isn't it odd as hell the same people that complain about html are also some of the same people that use "special" mail readers to emulate news readers? Both seem to want to influence the 8,000 list members their tools are the only one's in existence and we better all format our list postings to make their tool happy. NOT! Guess my 1996 reader must be a little odd; it handles top & bottom postings along with html without complaining a bit. Most all mail readers these days support HTML email. The difference though is when one uses a mail reader that renders the HTML and the HTML is written by someone who has no clue how the readers machine is set up. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: determining cause of dropped calls?
I am having a similar problem. It is not frequent, perhaps once in 80-100 calls. CVS-HEAD-05/08/04-21:57:50 using Cisco 7960 6.3 and X100P --__--__-- Date: Tue, 1 Jun 2004 21:04:14 -0700 (PDT) From: Bruce Komito <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: [Asterisk-Users] determining cause of dropped calls? I am trying to figure out why calls between SIP devices and the PSTN are being regularly dropped after anywhere from 2-15 minutes. I have turned on everything I can think of, but I don't see any obvious reasons for the drops. All I can see from turning on debug and verbosity is two messages advising of a destroyed call, followed by normal-looking SIP and ZAP termination messages. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk in the news
http://www.tmcnet.com/tmcnet/articles/2004/031704rt.htm Previous article by same author: http://www.tmcnet.com/it/0104/0104PO.htm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIPURA 2000 Problems (Senad Jordanovic)
I have had a similar problem upgrading to .24 . Sipura support suggested using tftp which worked successfully. On the tftp server you use the URL http://aaa.bbb.ccc.ddd/upgrade?/path_name/spa.bin where aaa.bbb.ccc.ddd is the IP address of the Sipura. Do not know why these instructions are not in the manual. -- Bill From: "Senad Jordanovic" <[EMAIL PROTECTED]> To: "'Stefan Meier'" <[EMAIL PROTECTED]> Cc: <[EMAIL PROTECTED]> Date: Tue, 16 Mar 2004 16:50:08 - Subject: [Asterisk-Users] SIPURA 2000 Problems Reply-To: [EMAIL PROTECTED] * I can not update device to latest .31 firmware. It just sits there waiting for SPA 2000 to "not to be in use". I waited and waited for many many minutes... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Voicemail Password Digit Timeout
FromJim Burwell, Dec 21,2003 __ I had the same problem with Grandsteam phones and *. No other hard or soft phones have the 'double digit' problem with *. I don't think Asterisk can do both RFC2833 and in-band DTMF at the same time. It does, however, do RFC2833 and SIP Info at the same time (SIP Info method seems to be on all the time, even when RFC2833 is selected in the sip.conf file). Switching the Grandsteam to SIP Info allowed it to talk to Asterisk and fixed the double digits problem. - Jim __ Date: Sat, 14 Feb 2004 10:56:39 -0600 From: Rob Fugina <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Voicemail Password Digit Timeout Reply-To: [EMAIL PROTECTED] On Thu, Feb 12, 2004 at 04:30:19PM -0600, Ryan R. Fligg wrote: I was wondering if there was any way to change the digit timeout or some setting of that sort on the voicemail password entry. Currently when our users enter their passwords they have to enter them very rapidly, otherwise asterisk will log the number twice. So if someone entered a voicemail password of 1234 slowly and deliberately on our system the asterisk receives it as the following number, 11223344 and thus returns the passcode invalid message. System: Asterisk CVS-02/10/04-13:27:57 built by [EMAIL PROTECTED] on a i686 running Linux 3 X100P cards 5 Snom200 phones I can't help you, but I can "me too". I have a TDM400, and accessing voice-mail from these extensions is always fine. I also have a Grandstream SIP phone, and it behaves exactly as you describe. It has to do with how long the number buttons are pressed. To make it work, you have to key your PIN like the buttons are too hot to touch... I'm running the latest (.46) Grandstream firmware. I'm using "dtmfmode=rfc2833" in sip.conf, and the matched setting on the phone. Rob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voip-info.org DNS seems broken
For the last few days I can not resolve voip-info.org from many DNS servers. It does resolve with some DNS servers but I suspect it may be related more to caching. Using the host command: host -a voip-info.org 130.179.16.23 Trying "voip-info.org" Using domain server: Name: 130.179.16.23 Address: 130.179.16.23#53 Aliases: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 33642 ;; flags: qr rd ra; QUERY: 1, ANSWER: 2, AUTHORITY: 2, ADDITIONAL: 2 ;; QUESTION SECTION: ;voip-info.org. IN ANY ;; ANSWER SECTION: voip-info.org. 86318 IN NS ns2.lj.net. voip-info.org. 86318 IN NS ns1.lj.net. ;; AUTHORITY SECTION: voip-info.org. 86318 IN NS ns2.lj.net. voip-info.org. 86318 IN NS ns1.lj.net. ;; ADDITIONAL SECTION: ns2.lj.net. 3518IN A 64.65.89.226 ns1.lj.net. 138603 IN A 64.65.89.226 Received 133 bytes from 130.179.16.23#53 in 95 ms ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
