similar to: SIP Authentication bug?

Hi, There is any possibility to dial a specific extension and then enter in your own mailbox (the one defined for that specific SIP phone) without asking for the exxtension number but only for the password? I want to be the same extension for all phones, not a specific one for each of them. It is possible to have a time stamp in the recorded message? I want to know when the message has been

I'm coming to Paris Dec 19. I was wondering if there was any interest in having an Asterisk get together in Paris sometime near there. Any one out there interested? Anyone in Paris who could help organize something like that? :) Mark

I've tried to figure out from the web site the minimum hardware cost to run a small office Asterix solution but I'm afraid I miss something: Let's say that I want to connect four/five analogic extension to the PBX. I have: - 1 computer as the server (with linux and Asterisk on it) - 1 dummy patch panel to connect all the analogic phones around the office What (and how many) cards

I have an X100P and when I place calls to the PSTN which are not answered, the Billsec field of the CDR still logs the seconds that the phone rang. Can someone please confirm that this has to do with the ringcadance of the indications.conf file? Is there anything else I need to check ? Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? Ta. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040421/3d91c7f6/attachment.htm

Can anyone recommend some phone sets that are adsi compliant and work well with asterisk?

Hi, Is Asterisk (or can it be set up as) a SIP proxy? Thanks -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

I too am having this problem reported by Frank Hoonhout. Asterisk runs fine for a few minutes and then stops accepting new calls. (I have a standalone server with SIP phones and I'm not doing any external registration). Asterisk CVS-04/07/03-09:28:50 0x420e0037 in poll () from /lib/i686/libc.so.6 (gdb) info threads 16 Thread 14351 (LWP 7258) 0x420e187e in select () from

A customer of mine has 3 TDM400P cards in a box running asterisk. On each card he has four FXO modules. I have set up the dialplan to dial via group 1 for an outgoing call. Channels 1-12 are in group 1. If he plugs a telephone cable into socket 2 or 3 etc, but not 1, when he dials out, it still tries to make the call via socket 1. Straight away the console says that it has dialed the

Dear all r-users, I am getting a big problem with matrix multiplication suppose I have, > weight Weight 1 1067640 2 8871500 3 42948778 4 127583735 5 22000000 6 44000000 7 56850000 8 23805662 and, > s a b c d e f g h a 402493.18 -133931.62 461483.3 -94042.86 674493.8

Hi Has anyone had any luck in putting two of these im a box? May I ask how.... I am starting to go bald.... I do have a suggestion is make another box and link them with iax2 Thanks Samantha r2d2:/home/s/samantha# ztcfg -vv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2

I've made a subtle but important SIP change as part of bug #155. According to Mediatrix, the URI in "Contact" should be used as the URI in the top part of the SIP request when sending follow up messages, and they're allowed to use the same IP for from and to. I made the change to support that but I want to be sure we didn't break anything else, so if it did, let me know, on

I believe I have repaired the SIP registration code for outbound SIP registrations (e.g. FWD). I would appreciate feedback from people on this. If you cvs update, please be sure to "make clean ; make install" I plan on releasing Asterisk 0.4.0 by the end of this week, so if you have any SIP or other issues you feel need to be addressed by then, please try to find me on IRC or via

Hi, I need to draw a line segment between two points on different plots in the same multigraph.I've tried looking at the zoominplot function in plotrix but havent understood much.any help is appreciated ~Aks [[alternative HTML version deleted]]

Skipped content of type multipart/alternative-------------- next part -------------- [root@LINUXVM root]# asterisk -vvvc == Parsing '/etc/asterisk/asterisk.conf': Found Asterisk 0.4.0, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= == Parsing

Hi List Is there a cron that I con do to replace this, as the fx0 card doesnt hang up properly phonegc:/home/samantha# asterisk -r Asterisk CVS-05/30/03-17:17:07, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= Connected to Asterisk CVS-05 currently running on

What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter, the last command in the history always defaults to 'stop now'. This is very bad, and it's caused accidental shutdowns more than once. Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399) Verbosity is at least 3 hera*CLI> A No such command 'A' (type 'help' for help)

I am getting this Error on my Asterisk. How to solve it ? "ERROR[2654]: chan_sip.c:11355 handle_request: Missing Cseq. Dropping this SIP message, it's incomplete." -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090115/cb953962/attachment.htm

Hi, Is it possible to pick the context of a call from chan_sip based on the domain of the To: header of the INVUTE? I've had a quick look throught he code and can't see anything, I want to use the voicemail virtual hosting with chan_sip. Can the sip domain be picked out with a global in extensions.conf? This woud also solve my problem. If not is there any specifc reason/restriction

On Jun 16, 2007, at 12:35 AM, Duncan Sands wrote: >> Result compiled with llvm-g++ 2.0: >> tan float: -2.18504 >> tan double: 0.309336 > > This may be due to bug 1505. It fails on x86 using x87 floating point, with the inliner not run, because of 1505, yes. Gonsolo, is that your situation? (What happens is, there is a wrapper in the header file for std::tan (float),