search for: freetext

Or even: <a class="moz-txt-link-freetext" href="http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&mid=4946">http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&amp;mid=4946</a> (same thing from the UK site:) <a class="moz-txt-link-freetext" href="http://www.blackb...

I have set up a * box with two hfc ISDN pci cards using mISDN both in TE mode with PmP mode. (using $MODPROBE hfcpci protocol=0x2,0x2 layermask=0xf,0xf) I have no problem dialing out by explicitly naming the mISDN port, ex: Dial(mISND/1/${EXTEN},60) or Dial(mISDN/2/${EXTEN},60) But it does NOT work when specifying the mISDN group: exten => _(outpattern),1,Dial(mISDN/g:TEmode/${EXTEN},60)

Does anyone test Asterisk 1.2.X + bristuff-0.3.X and TDM card? We can't get it to work. -David -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Julian J. M. Sent: Friday, March 31, 2006 1:44 AM To: Chris Earle; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: BRI

As usual, we'll be jawing about any and all asterisk-related subjects with the usual gang and any new people are always welcome, regardless of your level of expertise. You can even come and ask questions, it's guaranteed to be a more pleasant experience than it will be on IRC ;) http://VoipUsersConference.org/topics.php IRC; Freenode.net #voip-users-conference

...ailto:Gluster-users at gluster.org">&lt;mailto:Gluster-users at gluster.org&gt;</a><a class="moz-txt-link-rfc2396E" href="mailto:Gluster-users at gluster.org">&lt;mailto:Gluster-users at gluster.org&gt;</a> <a class="moz-txt-link-freetext" href="http://supercolony.gluster.org/mailman/listinfo/gluster-users">http://supercolony.gluster.org/mailman/listinfo/gluster-users</a> </pre> </blockquote> <pre wrap="">_______________________________________________ Gluster-u...

Actually it worked, but only after I defaulted all the settings on the phone and let it pick the config up fresh. Anyone know if there is any headset config options to default to headset/speaker? Thanks Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Lee Archer Sent: 03 January 2006 14:49 To:

hi everyone . Here I have a text where there are some integer and string variables.But I can not read them by readLines and scan the text is : weight ;30;130 food:2;1;12 color:white;black the first column is the names of the variables and others are the value of them. the column in different line are different. Can anyone help me ? -- TANG Jie Email: totangjie@gmail.com Tel: 0086-2154896104

Dear Mr,Ms; I am planing for a custom IVR, for example to act as a simple installer! I mean there is some choice via 0-9 and # as *Next* and * as *Back* button. is there any way for me to detect if the caller pressed # vs * on Dialplan ? -- Regards, Ali R. Taleghani 0936 322 4069 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Im using Asterisk 1.4.11 and I want to proceed some time and date operations in my dial plan. (for a time shifted callback). Should look like: CURRENT TIME + x minutes. Of course it should increase the hours for example in this case: 10.59 + 5 minutes = 11.04 I guess I've to use the math function in 1.4 but how can I manage easily the time operations? Kind Regards, Erik

Hi All; Is there an GUI for Asterisk that can help in showing the call flow (who is in progress, who is connected, called number, ...)? I was think in AsteriskNow does this? Any advise? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

The HAVE_MMAP_FILE macro comes from when you configured llvm. The configure tested if your system is giving access to a various resources required by llvm. LLVM needs the MMAP functionality by your system. Some times, if the required resources aren't present, we use other resources. However, in this case the building process is stopped, because the functionality can't be implemented

...gt; <li class="D_off">One Available 64-bit/100MHz PCI-X slot</li> </ul> The list has already answered what goes in what slot so I won't repeat that. It is a 1U unit and the above taken directly from the specs at :<br> <br> <a class="moz-txt-link-freetext" href="http://www.penguincomputing.com/index.php?option=com_content&task=view&id=386&Itemid=598">http://www.penguincomputing.com/index.php?option=com_content&amp;task=view&amp;id=386&amp;Itemid=598</a><br> <br> Oh, will repeat some...<b...

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=GB18030" http-equiv="Content-Type"> <title></title> </head> <body bgcolor="#ffffff" text="#000000"> Hello,<br>  I built llvm1.4 with gcc3.4.3 and glibc2.3.2 on a redhat 9.0 machine,

Hi, I'm testing/debuging the migration from mbox to maildir in a mailserver running Dovecot 1.0alpha5. I migrated all 472715 emails(in a test server) and later noted that dovecot could not get the same flags/status of the emails as in the mbox. I was using perfect_maildir.pl and I think its not dovecot fault. Seems to me that perfect_maildir.pl convert the emails in a way that dovecot could

Hello List, I'm having problems with registrating my Asterisk-server to the SIP-provider. Yesterday all worked fine, this evening I cannot call out. What can be wrong ? This is my registration in sip.conf : register => 092779077:XXXX at 85.119.188.3 This the output of SIP show peers : asterisk*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status

Docs? Polycom has docs? Where would one find this fabled land of... err I mean Polycom does stating what ftp servers are supported? Doug. -----Original Message----- From: Ken D'Ambrosio [mailto:ken@jots.org] Sent: Tuesday, March 07, 2006 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

HI, I'm not able to find some good doc or manual regarding Integration of Asterisk with SER. Bacially, I want to forward my calls from SER to asterisk. If some one already done this please guide me. thanks in advance arun -------------- next part -------------- An HTML attachment was scrubbed... URL:

...; </pre> <blockquote type="cite"> <pre wrap="">-----Original Message----- From: <a class="moz-txt-link-abbreviated" href="mailto:llvmdev-bounces@cs.uiuc.edu">llvmdev-bounces@cs.uiuc.edu</a> [<a class="moz-txt-link-freetext" href="mailto:llvmdev-bounces@cs.uiuc.edu">mailto:llvmdev-bounces@cs.uiuc.edu</a>] On Behalf Of Chris Lattner Sent: Friday, March 18, 2011 1:15 PM To: <a class="moz-txt-link-abbreviated" href="mailto:Anton.Lokhmotov@arm.com">Anton.Lokhmotov@arm.com&...