similar to: phonecore, gnophone from CVS.

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

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! 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

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! 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

Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT.

Hi all, I installed all needed RPMs by GnoPhone to be installed without problems but when attempting to install GnoPhone itself I get this message: # rpm -Uvh gnophone-0.2.4-1.i386.rpm error: Failed dependencies: mozilla >= 0.9.2 is needed by gnophone-0.2.4-1 libgtkembedmoz.so is needed by gnophone-0.2.4-1 libgtksuperwin.so is needed by gnophone-0.2.4-1 I'm using

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

I have just finished installing all the pieces of Redhat Linux 9 (2.4.20-8), Asterisk-0.9.1, Gnophone-0.2.4 on my IBM T30. Audio card is SoundMAX Integrated Digital Audio. Not sure what the chips are. But everything works on both W/XP and RH9. (Machine is obviously dual boot) Everything starts, Asterisk is up, sound/audio is great for CD player, Volume controls, Voice recorder, RealPlayer, etc.

I am having a really hard time getting gnophone working with asterisk. Gnophone tries to register with my server but there is no response. I can direct incoming calls to gnophone but if gnophone answers them, asterisk does not recognize it. Here is my configuration: iax.conf [jambo] type=user host=dynamic defaultip=136.159.99.100 permit=136.159.99.100 username=jambo secret=fubar

How do I get Gnophone to register to my Asterisk server? I have set up iax. conf as follows: [tim] type=friend ;username=tstornes host=dynamic ;defaultip=207.194.60.56 secret=1111 context=from-iax callerid => "Tim" <5000> auth=plaintext qualify=10 permit=0.0.0.0/0.0.0.0 and extensions.conf includes a section in the context from-iax: exten => 5000,1,Dial(IAX/tim/s|100|r)

Hello, trying to get gnophone to fully function under RH9 and finally have the browser working from the rpm from 2001. Turns out I had to get an old Mozilla 1.2 version and alter the startup gnophone script to point to it (MOZILLA_FIVE_HOME and LD_LIBRARY_PATH). How much better is the CVS stuff? I cvs checked out the gnophone and the Changelog only shows a comment about 0.2.5 but no real changes.

Greetings everyone, I'm still a newbie, so please indulge me. I have set up an asterisk server on my RH9 boxes for testing. Instead of immediately launching into the hardware aspect of it I decided to go with 2 IAX clients (gnophone) which I placed on 2 other strategic machines within my LAN. Here's my problem, on one of the IAX clients (gnophone) I am able to do the asterisk demo

hello everybody well...while trying to make gnophone to gnophone call using IAXtel's PBX server......i am not being able to establish a connetion.... possibly whats happening is, we are not being able to transmit our message packets properly and result is its not being able to establish connection n is resending frmaes again and gain here is message we are getting Tx-Frame Retry[000]

I hope this isn't something newbiesque and Steve will denigrate me. . . I just built gnophone, and I'm having trouble figuring out just how to sync the "Telephony Preferences" in gnophone and iax.conf on my asterisk server. I am heading on a trip and I'm pretty sure I'm going to be behind a NAT gateway, and my asterisk server is *separately* behind a NAT gateway. I