similar to: Issues after upgrading from 1.2 to 1.4: hangup immediately

Is there a way I can have "voice mail check" calls coming from my internal users automatically get to the right extension, without having the user enter their extension? I'm thinking that I could have the local SPA boxes translate, or have each user live in a context where the extension in question exists uniquely per user, but both of these seem kludgey. Thanks in advance for

Hello I currently run a few samba servers one being used as a PDC. Today I added a user to the domain and for some reason I can not get it to connect to any of the shares but "home" on the file server. % smbclient -U gregi //server3/public Password: Domain=[HYDRIX-MALVERN] OS=[Unix] Server=[Samba 3.0.28] tree connect failed: NT_STATUS_ACCESS_DENIED However I can connect with : $

Hello I am running the following setup in order to make VoIP calls at home. Home Phone <-> SPA3000 <-> Asterisk Home <- IAX2 over Internet <-> Asterisk Office The voice mail for Home Phone is hosted on the Asterisk Office machine. I wanted to have a way to check the status of my voicemail on my home phone directly. In order to do so, I've patched both Asterisk Home and

Hi there. Feel like I'm getting nowhere. Code: FreeBSD samba 3.6.6 ; connected to a Windows 2008 R2 AD wbinfo -u and wbinfo -g will properly show all the users and groups ; but for some reasons, I can get getent with winbind to show any users... I've must have tried over 100 different configuration, read all the tutorials I could find ; I never got it to work :( smb.conf: [global]

Thanks Peter, Yes, indeed the problem seems to be exactly what you describe. It's overhere the same. If I dial a mobile number it disconnects immediately when I hangup the mobile. But for analog numbers it takes around 10 seconds or so... Well, at least now I know how to debug pri :-) Walter. On Thu, 29 Jul 2004, Walter Klomp wrote: > However, if I dial-in from the SIP phone to my

Hi All After lots of try I was successfull in connecting to PSTN to make and recevice calls , I used AMP for this purpose , now I wanted to try out this Asterisk server answers the call , ask for the extensions and then after the extension entered the call is forwarded /transfered to the extension no , I use Asterisk 1.2.4, configured using AMP , on RHEL3 I did some configuration for my

Hello I run the latest kernel from centosplus and use a JFS partition. Even though the jfs tools are installed, when the server crashes, it doesn't run fsck.jfs automatically when at boot time it detects that the partition is corrupted I have to run fsck.jfs manually and mount the drive. Is there a way to make fsck.jfs run automatically if required ? (I recently moved from Fedora to

Hi there. Very particular issues here.. A few months ago, we migrated our authentication system to a MacOS 10.6 server mac mini. It acts as a KDC and Windows primary domain controller. No major problem there. Except that once a week our main file server (samba client, domain authentication) would deny any incoming connections. I found that issuing a net join command ; entering the admin

Hi there. I installed OpenSolaris 10 on a test machine, and I'm trying to make it join my samba domain. Unfortunately, the only references I can find about joining a domain on Solaris are related to joining an AD domain. I just want to join a samba domain (Windows NT/2000 style) smbadm -u name DOMAIN always fail with: failed to find any domain controllers for DOMAIN I wanted to experiment

Dear all. I'm experiencing a weird crash with one of our desktop running CentOS 5.1 We have 5 machines identical, onle one has this problem. Right after it starts, it will kernel panic. Unfortunately, from the backtrace this is all I've managed to get : serial port isn't working. So it's a manual copy of what the screen show. [<c04721f3>] sync_buffer+0x0/0x33

Hi there. Our IT moved all the user accounts to a new domain controller. It wasn't much of a migration, more so a complete setup on a new machine, new OS, new domain ; it just happens that the username and group names remained the same. I have been asked to look after the migration of the existing unix servers (linux and freebsd running samba 3.4). All the unix machine use winbind for

We have officially made the first release candidate of Zaptel, Libpri, Asterisk and Gastman available. While there are still open major bugs, they are relatively limited, and it was time to go ahead and get the 1.0 ball rolling in earnest. ftp://ftp.digium.com/pub/asterisk Enjoy the code. Special thanks to all the bug marshals and contributers and to everyone who has supported Asterisk

Hi Last month, people reported a crash with Asterisk 1.2.13 and spandsp-0.0.3 when receiving a fax using fax detection. Today I've tried 1.2.14 with the latest spandsp 0.0.3pre27 and with the snapshots for app_rxfax.c and app_txfax.c. The problem still happens. Has anyone found how to resolve this issue? I tried emailing Steve Underwood (with crash backtrace) but he hasn't

Hi there. I have a FreeBSD server running Samba 3.3, connected to a domain who's PDC is a MacOS 10.6 server running Samba 3.0.28 (ancient I know). Working all fine, except for one thing I find annoying. MacOS server has a concept of username alias. You can have as many aliases as you want, using any of those aliases are the same as using the primary one. It's rather well implemented in

Sorry for the crosspost (this was also posted to asterisk-at-uc-dot-org) but I haven't got a response. I have a cell phone added to a queue as a local extension (member => Local/299). I want the cell phone to be able to reject calls to the queue without the person sitting in the queue being hung up on, etc. The way my dialplan is set up, the person hits 1 to answer the call and any other

Hi In order to add several phones to a single extension number, I have replaced entries like: exten => 100,1,Dial(SIP/sipphone100,20,Tr) into: exten => 100,1,Dial(Local/100 at phones,20,Tr) [phones] exten => XXX,1,Dial(SIP.sipphone${EXTEN}) etc This works great. However in the CDR, than seeing one entry for each call, I see several entries in the CDR Worse, if I do something like:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello I'm the one who posted a message about the fact that nobody answer anymore to questions asked. I posted two days ago a problem I was facing that calls made over the Internet to my Asterisk gateway would hang-up after just 5s (no NAT were involved) Answer I got was: it's your config Well, it wasn't (as I was expecting). I

Dear all. I recently subscribed to a VoIP provider through IAX. The require to connect with either G729 or GSM, I chose G729 based on their recommendation. The service works very well, however ... people mentions how distorted our voice sounds. We have plenty of bandwidth available so I don't think it comes from our side. What it means is that it comes from two things: 1-G729 gives bad

hello, yesterday the uniden uip200 phone was recommended to someone. i am looking for an alternative to grandstream bt-100 because i can not do a supervised tranfer with it. here my questions: 1) does the uip200 support supervised transfers? 2) can i buy the phones in europe, especially in germany? thanks in advance, jan goericke

Hello I used to have an extension like this which worked fine with asterisk 1.0.7 I first dial to see if an IAX phone is present, if not I would try on SIP instead exten=s,1,Dial(IAX2/iax${ARG3},20,tr) ; 20sec timeout exten=s,2,Goto(s-${DIALSTATUS},1) ; Default action exten=s,200,DBget(temp=CFBS/${ARG1}) ; Get CFBS key, if not existing, goto 301