similar to: More beginner questions

Hi, Samba guru; I need your help. I am using samba 2.0.6. security = server and password server = NT PDC; nt acl support = yes. I am trying to change permission using the NT security dialogs but failed. In the log file I found that apparently when I connect to the samba directory, I get authenticate correctly. But when try to change permission, samba does not try to authenticate through password

Hi all, New user to asterisk having just got it compiled and installed. Running with no digium hardware (yet) and no soundcard in asterisk box. Problem is using the sample configs with a sip phone added as follows [2203] type=friend username=2203 secret=2203 host=dynamic defaultip=192.168.0.2 dtmfmode=inband canreinvite=yes the console on * when running with -vvvvc says :- (whenb trying to

Hello. I have some problem, with a new configuration on a new PC. I want to setup a SAMBA PDC using an HOWTO. This howto was working on OPENSUSE 10.1 with a X86 processor and I have used it a lot of time. Now I use OPENSUSE 10.3. The new PC run a X64 processor. After the fresh install and following : http://samba.org/samba/docs/man/Samba-HOWTO-Collection/diagnosis.html I could not make

hi new user here cant seem to get fwd running, got asterisk from download site as tarball, did the readln and openssl start. Also configured the sip.conf and extensions.conf but an error with the chan_sip.c shows up? any ideas...somebody...anybody! thanx jai

Hi. I have a x100p card installed and also asterisk, but I just dont get asterisk to register with my sip provider (FWD)... when I start asterisk using the following command I get the following messages (first, a lot of messages show up immediatly after starting up: I'read this is normal, then the CLI console comes out and this messages appear): NOTICE[229390]: chan_sip.c:3922

Hi :) I've had all this working before, but I'm revisiting it, and in short, I currently have huge problems receiving incoming calls. I've been trying with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel and libpri as of yesterday afternoon. Would someone mind helping? :) My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set as the 'DMZ

Hi, If there is no blocker (James?), we should be able to move on, so I filed as a starter: https://reviews.llvm.org/D25710 <https://reviews.llvm.org/D25710>: [Doc] Drop MSVC2013 support — Mehdi > On Oct 12, 2016, at 4:03 PM, Reid Kleckner <rnk at google.com> wrote: > > I migrated the sanitizer-windows builder to VS 2015, and it went green here: >

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

I have a small HUGE problem with *. I have installed * but I have 2 problems. 1 - Making call to FWD. 2 - Receiving call from FWD More info of the problem at the end. Here is the sip.conf file. ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ;default Default for incoming calls register =>

Hi..I'm having trouble registering my asterisk box with FWD....It worked the other day. I also have an individual Grandstream phone which registers fine right now. I looked at the archives and saw the thing about the maximum retries limit to 5...but since my Grandstream phone seems to register on the first try, I'm thinking the problem lies elsewhere. Any ideas? sip show peers

Hi! I am new to Asterisk too, I got the similar problem and I would like to know how to get * to work behind NAT. When I have the SIP Debug turn on, I got the error 479 from FWD when * try to register with FWD, it looks like * is using the local IP (192.168.x.x) in the Contact field. I have put the nat=yes in the [FWD.Pulver.com] content, but it does not seems to make Asterisk aware the

Hi. I have installed a x100p (THE x100p for those who have seen my former post). Now I just want to connect a "normal" phone (not an IP phone) to the card and use it as a sip extension (I have a FWD account)... more clearly: I want to be able to pick up the phone and call any FWD user using my FWD account... receive the FWD calls in that phone, and also to be able to make normal

I've tried to register * at FWD but * segfaults, so i guess my register-line in sip.conf isn't correct. It looks like: register => fwd#:pwd@192.246.69.223 should it be different? Chris

I'v a sip client and a sip trunk to FWD: [general] port=5060 context=default tos=reliability disallow=all allow=ulaw careinvite=no [freeworlddialup] context=default type=friend username=MYUSERNAME secret=MYPASSWORD host=fwd.pulver.com [igor] type=friend callerid="Me" host=dynamic dtmfmode=rfc2833 careinvite=no When i try to call a FWD number from SIP client i obtain a lot of

Hello, I'm having a problem with SIP. More specifically, I can't make any calls using SIP. I have had an iConnectHere account and Free World Dialup account working for quite some time, and now all of a sudden I can't make any SIP outgoing calls. PBX*CLI> sip show registry Host Username Refresh State 192.246.69.223:5060 XXXXX 120 Registered

Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",

I can't get my Asterisk to register/place calls with FWD. Here's what I have in my SIP.CONF: register => 11111@fwd.pulver.com/11111 [fwd] type=friend secret=somesecret host=fwd.pulver.com username=11111 fromuser=11111 fromdomain=fwd.pulver.com I'm using CVS version of Asterisk, checked it out last week. I get authenticate error when registering with fwd, and all my calls to

Hi ;) I'm a french newbie and i installed asterisk 1 day ago. I've got an ATA186 and a computer with Sjphone installed. If i want to call the sjphone from the ata or call the ata from de sjphone everything is ok. My problem is ,that i can't call the voicemail or any other phone number ..as 600 for exemple from the ata or the jphone. I don't know why but i looked after a long

hi My stations are behinds a firewall, the system is windows 2000 & 98, i use sjphone asterisk is on the internet gateway where is the firewall Shorewall the system is linux debian (sid) kernel 2.4.20 j do whaton http://www.automated.it/guidetoasterisk.htm (grateful Andy) to write my sip.conf but i can't call an external sip user. (an external user can call me) i try without asterisk with

We have been running Asterisk for about a month now and one of the things I miss the most is the ability to se who's online and available and who's not. Surely, there's the manager interface, but unless you'd want to install extra software on each client computer, this is not a good option. Then there's the presence / IM function in SIP. Since we're only using SIP