similar to: FWD SIP phone format=2, FWD call format=4, why?

Hi! The university where I work just bought four Cisco 7960G IP phones (they didn't ask, just came across the door and gave me a box and told me: "Can you make this work with the Asterisk PBX we have?"). According to what I read, there is no much hope, because I have not the SIP firmware (too bad). Has anybody succesfully got an answer from cisco?, or does anybody happend to

Hi! I'm trying to use Directory Administrator from http://diradmin.open-it.org/files.php, but It only work with the old sambaAccount schema, so, My question would be: What is *really* new on samba 3 with the use of sambaSamAccount?, do I lose something if I use the compat mode? Thanks in advance for any answer, sincerely, Ildefonso Camargo icamargo@merkurio.com.ve icamargo@unet.edu.ve

Hi! I think it is a great idea. The DS80C400 needs external memory, and/or flash. It have the Ethernet integrated, but it is really slow (it is 8051 architecture), and yes, I know it can go up ti 75Mhz, but only gives 18MIPS max. I would use ATmega128 from atmel (16MIPS at only 16Mhz), take a look at: http://www.ethernut.de (project using mega128 with Ethernet, includes schematics). It

Hi! I decided to apply Chris's patch for the rtp problem, it is working just fine now. Thanks Chris!. I think that Mark should submit it to the CVS. Ildefonso. icamarg@unet.edu.ve >Pete, > >Try this patch below... I noticed that eStara's softphone has the same >problem as xten's softphone when it comes to DTMF. Seems as though = >Asterisk >is not looking for

Hi! I want to automatically delete old messages from one user's Maildir, and I was thinking on running something like this: find /home/user/Maildir/cur/ /home/user/Maildir/new/ -daystart -mtime +15 -delete but I'm not sure if by deleting messages "manually" I will break the dovecot.index* files. What do you think? Thanks! Ildefonso.

Hi! I have this configuration: SIP client A <-> NAT box A (real external IP) <-> Asterisk server (real IP) <-> (real external IP) NAT box B <-> SIP client B The echo test form any of the clients to the asterisk server is working just fine, even without canreinvite=no. When I try to call from SIP client A to B, wihtout the canreinvite=no in the sip.conf, the call

Not sure, I'm reopening an OLD thread here (sorry). I need some answers, looking somewhere I found this: http://msdn.microsoft.com/library/default.asp?url=/library/en-us/dnkerb/html/MSDN_PAC.asp I'm not sure, I just gave it a brieft read. Can't this be used to include PAC data on a kerberos ticket in order to use the kerberos autentication on win2k/xp? I know that it also

Hi! I have been reading for about two weeks (maybe I'm reading on the wrong places). I have found as many documents as one could expect describind how to build a LDAPv3 server, or how to build samba with ldap. This far, I have failed, and have a BIG confution in the order in wich the things should go: In one document, they recommend this: samba -> ldap -> sasl -> kerberos

Hi! >Message: 5 >Date: Thu, 08 Jul 2004 17:00:21 +0530 >From: Sudheer Divakaran <sudheer@svw.com> >To: lartc@mailman.ds9a.nl >Subject: [LARTC] Is Linux based Router feasible > >Hi, > >I''ve a local LAN consisting of about 150 machines. I''m using a Linux >machine as the gateway machine which inturn connects to two different >ISPs. My

I concur with Jose. The Atmel AVR series packs a lot of bang for the buck. They also come in a 3.3v low power version for use in battery powered systems. Gene -----Original Message----- From: Leo Ann Boon [mailto:leo@innovax.com.sg] Sent: Tuesday, August 19, 2003 7:21 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: Open source IP phone, maybe? Ubicom's Scenix IP2K.

I have some stuck channels that I think I'm going to have to bounce Asterisk to get rid of, but am curious to know what they are and how they've managed to accumulate. The show up with a channel identifier of '(None)' as in the output below, and do not show up in the soft hangup list, and so can't be cleared by that method. Here is the output from iax2 show channels:

I have installed Centos 7 Minimal in a 1TB KVM. Used XFS file system. I did not use LVM. I need to install Directadmin which requires quotas. I have this in fstab: UUID=b482396d-d2fc-49ed-b9df-c49e9387405b / xfs defaults 0 0 UUID=e24a16e0-57ab-42b2-af0b-9edf789376e5 /boot xfs defaults 0 0 UUID=b08b8243-6b46-444c-a7bd-02934246d884 swap swap defaults 0 0 I get this: #

hi, i want to call PC2PC between to IAX client without authentication i want to allow every user to use PC2PC no any password required. Please let me know what i have need to do in IAX.conf or any other file to allow any user to call Pc2Pc. My IAX.conf [guest] type=user context=default callerid="Guest IAX User" My extensions.conf [default] exten=>_.,1,Dial(IAX2/${EXTEN})

Hi all, Below is what I did to run Asterisk in pass-thru mode: sip.conf: [general] disallow=all allow=ulaw canreinvite=yes For each channel, canreinvite=yes is enabled. No dial command has 't' option. However, it seems that Asterisk still stay in the media path and bridge the 2 end points. Am I missing something??? sip*CLI> show channels Channel (Context Extension

hi, i want to call PC2PC between to IAX client without authentication i want to allow every user to use PC2PC no any password required. Please let me know what i have need to do in IAX.conf or any other file to allow any user to call Pc2Pc. My IAX.conf [guest] type=user context=default callerid="Guest IAX User" My extensions.conf [default]

http://bugzilla.mindrot.org/show_bug.cgi?id=1198 Summary: ssh associates the host identification to just IP/hostname , and it should also include remote port. Product: Portable OpenSSH Version: 4.3p2 Platform: All OS/Version: Linux Status: NEW Severity: enhancement Priority: P2

Hi! Sure, just look for: Wonder Shaper. It's a HTB based shaper configuration wich have some very good features, I use a variation of that here at my College. http://lartc.org/wondershaper/ It is the page (a simple google search). Also make sure to uncomment the line tos=lowdelay in every config file of asterisk that have it. Hope it is usefull, sincerely, Ildefonso Camargo

Hi! I've been trying to use the Voicemail (and Voicemail2) applications with an SIP Phone (X-Lite, for those who cares), when I use g.711(a/u) codec, it works perfectly with inband (it detects the whole mailbox (in my case 10007)), but not with rfc2833 (in this case, it only detects 107 as the mailbox number). With gsm codec, the inband doesn't work, I guess that's due to the

Hi! I was trying to use gnophone with asterisk, but I can't make a call (It just get the a answer of "REJET"), but I can register an everything. Anyway, I decided to move to the cvs version of gnophone, so I checked out EVERYTHING from cvs.digium.com (yes, a cvs -z7 co .). I installed libiax2, gsm (the one that was inside gnophone), and got gnophone to start compiling. But

Hi! I have been reading the Samba 3 by example avaible on the web site (samba-guide.pdf), I think it is very good, but have a question: - In section 6.3.5 (page 150, numerated), there is a note wich says that the computers account must be inside the People container due to an error in samba. Is this true?, or can it be due to the config of the nss-ldap and the pam-ldap modules wich is on