similar to: Question about outgoing callerid

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:

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 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

Hi, Has anybody managed to get callerid properly set on a call from local to asterisk SIP endpoint through h323-pstn gateway to a regular phone. I'm using ata186 as SIP endpoint. It has 12125551234 assigned to it. When I place a call to pstn I'm not receiving 12125551234 as the clid, but a number assigned to PRI channel by phone company. It worked with chan_oh323, but there were other

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]

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. 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

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Adams, Gavin Is there any additional information I could provide to start tracking this down? I was thinking about looking into the various applications source to see how they access the data elements for callerid. I know where the values are pulled

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

Hey all, We've upgraded our PRI trunk to support what BellSouth calls "enhanced caller id name delivery". The weird part is, I'm only capable of seeing the name portion on incoming calls within voicemail2's email delivery. For example, on an incoming call, asterisk is reporting this: Context from extensions.conf (BS delivers 7-digit DIDs): exten => 9133727,1,Answer

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

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

* --> SIP --> CISCO --> PRI --> PSTN The PSTN sees no callerid. *---> PRI[zaptel]--> PSTN Callerid is there... which makes me think it's the cisco, not the PRI/PSTN/telco. CISCO PRI--> * PRI [zaptel] Callerid IS there... which makes me shake my head in disbelief, because * can see clid from the cisco pri, but pstn doesn't... but when * sends info on that pri,

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

Hi all, i use asterisk 1.2.7 and i have a problem with callerid. i use sangoma a200 cards. one fxo one fxs module i have these scenario - bob calls adam, where bob calls into my asterisk and adam picks up "from" my asterisk - bob and adam are speaking to each other - meanwhile eve calls adam, adam hears a beep, and knows there is an other caller on line. - bob and adam stop seaking

-----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

Hello, A question on channel originating (call files and AMI Originate): How can I change the CALLERID(num) var (because of the E1 provider needs), but having another n?mber (the original one) stored on the "clid" CDR field on the database? A channel agnostic solution would be the best one, without having to deal with the problem based on what type of Tech used for the outgoing

Hello I've been having the following problem today : I have a quite simple dialplan made to receive a fax: [answer-extension] exten => 1,1,Answer exten => 1,2,Macro(setcallerid) exten => 1,3,Ringing exten => 1,4,Wait(3) exten => 1,5,Macro(stdfwd3iax-notransfer,${EXTENSION},${EXTENSION},$ {EXTENSION}) exten => fax,1,Goto(faxreceive,1,1) The Wait(3) is there simply to let

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

Hello, I have a TDM400P with 2x2 configuration of FXOs and FXSs. I set a test extension of '444' to dial out a specific zap trunk and call a local #. First time I call out to '444' everything works fine. If I hang up the call, and within 10 seconds dial the same number again, I get a fast busy. Seems it isn't letting go of the trunk or something, and I don't have a