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Hi, I need to replace digital handsets in offices where there cabling is appareantly not Ethernet-compliant. Today's usage is to press a key to toggle between private ou public line before issuing an outgoing call. Are you aware of a DECT handset (to overcome cabling limitations) that mimic this line-key behaviour ? For instance, acceptable behaviours would be to dial number string and press

Would it be possible to execute some kind of script when for example Asterisk restarts... or stops... ? How can one read the status of Asterisk so that when the service is stopped I could be notified by mail, by text message,... ? I don't know how to read the status of Asterisk (or the change of status) in a bash-script. Thanks for the reply ! Kind regards, Jonas. -------------- next part

Hello I have few questions : - what's the difference between a subscribe request et a register request ? - in asterisk 1.6 allowguest=yes or no param does it work ? if yes, please someone could explain how doest it work because I think i'm a little bit confuse. - if I configure a sip terminal in sip.conf like this [john] type=friend username=JOHN secret=mypassword host=dynamic

...suffix = ou=Groups ldap user suffix = ou=Users ldap admin dn = cn=admin,dc=asta,dc=lan ldap passwd sync = yes ldap idmap suffix = ou=Idmap ldap ssl = no ldap delete dn = no ldap passwd sync = yes # Die IDMAP-Einstellungen sollten mit denen in Krefeld ?bereinstimmen, so dass SAMBA funktioniert. # Zweck der IDMAP-Einstellungen ist die Darstellung der Windows-SIDs als UNIX-ID...

Hello, Let me explain a scenario There are different Asterisk Servers at different Remote locations. Recording in different formats for FIVE seconds reveals that Format : Size wav : 84 KB gsm : 8.3 KB sln : 84 KB It can be recorded in any format. This is size for five seconds only. We need to transfer these files from different remote servers to a centralized server. We need to play these

Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register =>

Hi, My customer has a outdated firewall that is also presenting a NAT nightmare for getting the Asterisk server reachable from the internet. What firewalls work good with VOIP? I really want to steer away from any ALG supported firewall. I just want a good firewall that works well with Asterisk. Thanks, David Wathen -------------- next part -------------- An HTML attachment was scrubbed...

Hi everyone, I have an issue with Samba agains Active Directory. The authentication works just fine but when it comes to shares I've ran into some problems. If I use any group mapping from the AD it won't let me access it so I figure that is where the problem lays. If I comment out "valid users", "force user" and "force group" then I have no problems and it

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to connect to ekiga.net through a client connected to my Asterisk server. For it I am being based on this [1] document. Next I put the configurations that I am using. /etc/asterisk/sip.conf: ; Outgoing to ekiga.net [ekiga] type=friend username=MyUser secret=MyPass host=ekiga.net canreinvite=no qualify=300 nat = yes stunaddr =