similar to: How to send announcement after called has picked up the phone?

Next strange problem... W2k3 ADS. Sambe as ADS Member. pam_krb5 nss_ldap winbindd all seam to working correctls. Windows Users can access the shares on the Samba Server and can login using pam. smbclient works for all users... except from the Domain Administrator. smbclient //server/user -U user => is fine.... smbclient //server/Administrator -U Administrator [2005/03/23 17:33:30, 0]

Hi all We have the following environement: Microsoft ADS for Windows Users, NIS for Un*x Users. Samba 3.x Fileservers. Win2k/XP Clients which use CIFS to connect to the Fileserver. FreeBSD/Linux Clients which use NFS to connect to the Fileserver. For the moment, Windows User authenticate against the ADS and Un*x users authenticate against a NIS Server. Everything runs fine. But we would like

Hi all I have a very strange problem here... I use a hfc-s card with mISDN in NT mode with an ISDN Phone connected. When I make a call, the phone rings two or three times and then misdn runs into a timeout... I don't know where to set that timeout, but it's way to short for the called to pick up the phone. If the destination phone is picked up, then everything is allright and the

Hi Soeren, We encounter here a deadlock with a quite new ATA 120GB disk. The disk worked good for about 3 weeks, but now we have a strange problem. There seems to be one defective file on the disk. fsck doesn't find it, and if I do a cat file > /dev/null The machine locks completly. Serial console is dead, no remote DDB via ALTBREAK possible anymore, no panic message, just freezed. The

Folks, I'd like to change the value of ${CALLERIDNAME} for incoming PSTN calls from certain numbers, but haven't found a way that works. The goal is to provide more informative names on my phones' caller ID displays--e.g., I would prefer to display "ROB CELL" instead of "CELLULAR CALL" when I call home from my cell phone. This is what I tried in the context

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Hi Gang To increase security against phished passwords and similar attacks, we consider offering customers to define IP ranges (or GeoIP locations) from which their dynamic registrations are being accepted. I can already look at the source IP in the dial plan, so no issue with validate an INVITE against a source IP. But I would also like to prevent registrations from outside of this

I have working with a polycom IP500 phone. I like the idea of having each line button on the phone as a separate sip device. If I understand it right, each phone could have three extensions (one for each line.) This would be great since I could then use the dialplan to forward calls to the desired extension. I envision something like this: Extenson 101 - Company-A Extension 102 - Company B

Dear fellow list readers This is the situation: ISDN Devices => Patton ISDN to SIP GW => Asterisk PJSIP The Patton GW resides on a dynamic IP address, so I cannot really use match=ip in the identify section. The Patton does not send a line parameter. The ISDN Devices behind the patton have different MSN and should be able to send them in the From: Header, so the default endpoint

Hi Joshua Thank you for your reply. Indeed, Ubuntu only ships with this old version. Upgraded to 16.2. via PPA. Problem persisted. Well, I already mentioned that this is a machine with two physical interfaces with different routes which on the 'external' side handles SIP customer registrations and has an 'internal' IC Trunk to a commercial Voice Switch via private IP Range. I

Hi Gang Server, two interfaces, routing to two different networks. Two transports defined, each bound to the corresponding ip assigned to the interface. But still, especially when an 183 message is sent, the Contact header does contain the wrong IP Address. Is this a known issue 13.18.3? Or is there a way to make absolutely sure the IP addresses within the Contact header is corresponding to

I've got a need to do something like the following: [foo-context] exten => _.,1,SetCIDNum(123) exten => _.,2,SetCIDName(XYZ) include => local include => tollfree But of course, this example won't work. The goal here is this: if a call ends up being handled by the "local" or "tollfree" contexts, I want those SetCID*** commands executed. Otherwise, I

Is there a bug in Firefly (3rdparty) wherein it does not show caller ID? I am using SetCIDNum(12345) before I dial my firefly (IAX2) phone... no caller ID. CallerID is passed properly to other clients. -A.

Hi all I noticed that most caller are quite confused by the standard voicemail announcement text. Especialy as the number read is the 'internal' number. Callers often hang up because they think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all Winbind supports kerberos. Fine! Now I've set up a Samba 3 member server to a W2k ADS runing in Native Mode, but did not specify and LDAP or Kerberos stuff in smb.conf. Does winbind/samba do kerberos all by itself if hitting a ADS in Native Mode or do you have to configure it explicitly? How can I check if kerberos is being used or if

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-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all There's this situation: W2k ADS (no changes are allowed to the schema, so no Posix Data to be saved there) All users are managed via ADS and are only to be managed there (no separate manualy managed Database for ID Mapping) 2 Un*x servers runing samba 3.x with winbind being used as Fileservers. With the filebased winbind idmap the

Hi all Situation: Samba 3.0.11 FreeBSD 5 nss_ldap pam_krb5 Connecting to W2k3 ADS with installed MSSFU. (LDAP Posix Schema) pw user show -a pw group show -a both work. Authentication via Kerberos works fine. Users have access via samba to the files and directories that belong to them. But not to the Files belonging to their group. The 'Security' Tab under Windows shows the groups as

Hi, I've been trying to use the set_callerid function in the AGI. It sets the CallerIDname properly but I can't figure out how to set the CallerIDnumber. Is it at at possible ? Cheers. SL

Hello List I work at an SIP Provider and we have added and SBC in front of our Voice Switch to protect it. This requires all our SIP Trunk customers to register via a 'proxy'. I struggle with Asterisk to work over a proxy. This is what I have done so far. register => username at sip.example.com:password at sbc.example.com This works fine, asterisk is sending registrations via the