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Can T400P be configured for nfas (one d-channel providing signaling for more than one span)? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030522/f2b637a9/attachment.htm

Forwarded back to the list so others might get the benefit of the answers, and I get fact checked by others. -------- Forwarded Message -------- > From: Lee <leeb00@gmail.com> > Reply-To: Lee <leeb00@gmail.com> > To: Steven Critchfield <critch@basesys.com> > Subject: Re: [Asterisk-Users] udev or not? > Date: Fri, 10 Dec 2004 13:00:29 -0800 > On Fri, 10 Dec 2004 06:36:02 -0600, Steven Critchfield > <critch@basesys.com> wrote: > > On Thu, 2004-12-09 at 23:39 -0800, Lee wrote: > > > > >...

<P>hi everyone,</P> <P>I know this issue has been raised many times before, i think still the problem remains. When a call is made through a Zap channel, whether it is actually made or not (irrespective of whether, engaged, busy, or actually answered), asterisk logs it in CDRs as a call made. This makes it impossible to do an accurate billing. Has anybody found a way to overcome

Supposed scenario: one PC(2GHz CPU), one card 4E1, and one Internet link. There is somebody he know (has experienced) how many concurrent call (Classical Phone->Voip) can handle Asterisk ? Thanks ! -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

...s tied to the channel? If that is the case, then for my SIP phones, would the channel be (SIP/XXXX)? If it should be, then should extensions.conf look like: exten => 811NXXXXXX,1,MessageWaitOn(SIP/EXTEN:6) Again, thanks for the help. Jesse Date: Mon, 09 Jun 2003 22:37:01 -0500 From: Steven Critchfield <critch@basesys.com> Subject: Re: [Asterisk-Users] Adding an app To: asterisk-users@lists.digium.com Organization: Reply-To: asterisk-users@lists.digium.com On Mon, 2003-06-09 at 13:01, JKNUTSEN@UP.COM wrote: > I am in the process of testing out the Cisco ATA 186 to provide analog > ph...

Hi, i am recording a message, but it is always cut off at 40 secs. There are no time out configured. Gabriel -- The educated person is not the person who can answer the questions but the person who can question the answer.

I have set up a couple of test Asterisk servers and have never had a problem with sound. I've just done a fresh install on a dual 1GHZ PIII Asus box running Fedora Core3 with the Digium PCI Dev kit and following all the various Core 3 How-To's. I can make calls ok but when any sound is sent from the Asterisk box such as voice prompts and music on hold the sound is completely chopped up in

Anyone know what model and what support the new $100 Cisco has? http://biz.yahoo.com/djus/030428/1030001060_1.html -- Steven Critchfield <critch@basesys.com>

...p hardware, then leave a general list that does configuration other than VoIP related? The hope is that those asking SIP or H323 questions could get help from the various supporters while the main list can deal with transport neutral content like extension logic and voicemail configs. -- Steven Critchfield <critch@basesys.com>

Hello ml, i need some help on my zaptel configuration. My E100P only shows some YELLOW / RED alarm when I load the wct1xxp module and do a cat /proc/zaptel/1 Span 1: WCT1/0 "Digium Wildcard E100P E1/PRA Card 0" HDB3/CCS YELLOW RED ... .. . My /etc/zaptel.conf is: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=nl defaultzone=nl I tried zaptel-1.0RC2 and the latest CVS

hi guys, do you have any suggestions on how to connect 2 TE410P via E1? (for simulation and testing purposes) asterisk1 --> TE410P ----> ? ---------> ? ---->TE410P -->asterisk2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030907/698cd499/attachment.htm

We have 1000's of Remote Call Forward #'s across the USA / Canada, which forward into 1000's of 800 #'s in our call center. Is it possible to automate a solution where Asterisk could dial a given number, record the first 3 seconds of the call, save it to disk, and then go on to the next number, and just do this all day long ? We need to regularly check that the numbers work, for

Group Can I have 2 wildcard T1 cards in the same box? I was thinking the first card would have channels 1-24 and the second card would have 25-48 Does this sound correct?

Why does this person have my e-mail address ? ----- Original Message ----- From: <markogift@astriholics.org> To: <hackerwacker@cybermesa.com> Sent: Tuesday, November 23, 2004 1:13 PM Subject: Gift for Mark Spencer > Hello everyone! > > We have been thinking about something that we could do for Mark > Spencer as a holiday gift. We have decided to try to orgranize a

Has anyone thought about an ISO file that could be used to make a CD for a bootable install for a "basic" Linux/Asterisk system? Just re-boot and config. -- James Taylor jltaylor@metrotel.net 903-793-1953 --

If you are using Microsoft Outlook and you are reading this message you need to make 500% sure you are not propagating virii. I posted our support (at) nufone d0t net addy on this mailing list last night and have never posted it in an unprotected fashion like that anywhere else. So far today we have received over a hundred email virii to that address. I suggest you upgrade to a more secure

...8+extension-thats-ringing to have it go over to my extension. I am having a lot of trouble finding more information from the wiki and google. Thanks! -James -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Steven Critchfield Sent: Thursday, October 07, 2004 1:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dialplan to Pick up calls that are ringing onother extensions? On Thu, 2004-10-07 at 13:42 -0400, James Freire wrote: > Hi All, > > I have a qestion that I...

I think I have something misconfigured regarding voicemails. They work great, I have this setup: Sip.conf [ext1] Context=phones Mailbox=201 Voicemail.conf [home] 201,password,name,email@mail Voicemail delivery and all works great but when I check sip extension ext1 (analog phone using a Granstream ATA 286), the stutter tone signaling message waiting does not work. Anything wrong with

When setting include => daytime|9:00-21:00|mo-fri|*|* How does this determine what is different between 9 AM and 9 PM And after hours ??? I want different hours on Saturday and Sunday And a different welcome message after hours Any help appreciated Regards Mick