similar to: SIP client<->NAT<->Asterisk<->NAT<->SIP client. only works with canreinvite=no.

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! 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! 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 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

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

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

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 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

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 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 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! 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! >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 have a samba server set so that each user in a windows 2003 active directory can have their own personal, private share. I would like to know how to set up Samba so that their directory is created automatically, rather than me creating 1300 directories on the linux server. I cannot use pam_mkhomedir.so because 'security = ads' in smb.conf and 'encryption = yes' must be

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

Hello, I need to test canreinvite=yes with 2softphones and 1 asterisk. I want to have that : http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb ridge.png But I have that http://www.zimagez.com/zimage/canreinvite.php Canreinvite=yes work for all phones or just asterisk?... Can you help me? Thank you -------------- next part -------------- An HTML

Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients

Hi all I am using Asterisk 1.2.2 on frdora core 4. i have two sip UA. if i put canreinvite=yes voice Ok on both sides. and if i change canreinvite=no there is no voice (media through asterisk) one thing more if i try to use playback application for playing some sound file it is also working (like exten => 500,1,Playback(demo-abouttotry) this is working). here is sip.conf

I've gotten canreinvite=yes to work with a sip device behind NAT!! You *MUST* port forward the SIPPort to in your gateway router to your phone. This is a MUST. Okay, now on to my problem.. I have people who will be using ulaw, and I have people who will be using g729.. I want to set it up so that canreinivte will work.. I have a single cisco gateway.. Asterisks isn't handling the