similar to: Re: Asterisk-Users digest, Vol 1 #882 - 11 msgs

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'm trying an asterisk-FWD connection. I'm using X-Lite OR SIPPS as the IP phone. I configured the X-Lite and SIPPS to use GSM codec. Whe I call FWD, I get this info on the channels when the call has not been stablished yet: sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.246.69.223 613 1770bf3430d 00102/00000

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

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

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.

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 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

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

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

Are there any other codecs that can be used with the 7960 and the ATA-186? I have been using the default gsm codec and wanted to see if I could make use of something a little less bandwidth intensive. Kim Callis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030723/34e950e0/attachment.htm

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 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! I answer to two messages from this thread (I use digest). lartc-request@mailman.ds9a.nl wrote: > > Message: 3 > Date: Fri, 16 Jul 2004 10:51:37 -0700 (PDT) > From: ibro tj <ibb_linux@yahoo.com> > Subject: Re: [LARTC] QoS for Voip. > To: lartc@mailman.ds9a.nl, alessandro.ren@opservices.com.br > > Hi, > > the hint from Martin A Brown which I am

Hi All, Any clues as to what is causing this? I have seen similar questions asked before in regards to joining ADS domain but have not been able to find a solution to my problem. The domain is a Native mode ADS on win2k3 with signing required. Please let me know if additional info or logs are required to diagnose the problem. I did a 'net ads join ADSDOM -U

Dear Pals One customer has a Panasonic PBX KX-T336 with 60 ext. and a E1 (R2) for Trunks working perfectly now, This customer has 10 wireless links to his branches, wireless working great now, no voice at the present MY IDEA : T1Card into the Panasonic (additional to the E1) connected to a T1 Digium Card into Asterisk as far as I know the T1 Card can be configured as E&M to act as

I noticed that there is a soft button for transfer when you initiate a call. I pressed it, and it actually put the call on hold, although I was able to call another extension. Is that soft button functional? And if so, how do you make use of it? And if not, how does one transfer a call? Kim C. Callis -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is it just me or do others have a problem with the ATA-186 de-registering? Every couple of hours, if I don't make use of the ATA connected line, I find that I have to unplug and let the ATA reboot. After that it is good to go for awhile, but eventually I have to repeat the process. My ATA sits behind a NATd firewall, any ideas what might cause the de-registration? Kim C. Callis

As I was working on my extensions.conf file, I started to segment calling privileges. For the everyday workers, I don't free reign to LD access unless it is business related. So I was wondering if there was a way to implement some type of accounting code to be entered before accessing LD, which of course would be noted in the CDR (however it is implemented, either comma delimited or MySQL).

Yesterday, I started to experiment with Cisco to Cisco SIP calls using the g729 codec. According to the documentation, both the ATA-186 and 7960 are able to make use of the g729. >From an earlier e-mail, I made a change to the configuration of the ATA, changing the values: LBRCodec:3 RxCodec: 3 TxCodec: 3 The first thing I noticed was that when I did a sip show channels, the format had