similar to: Winbindd as NIS replacement in heterogen environement

Hi I would like to send a text to the called person when he picks up the phone before the call gets connected through. Is there a way to do this? Example: I'm registered to multiple SIP providers. They come in to a context each and then get through to my phone. Now I would like to send myself an announcement about from which SIP provider this call came from. -- Beno?t Panizzon,

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 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

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

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

-----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

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

Hi Richard > That could be possible and would be a bug in chan_sip. Ok, so I switched to PJSIP to see if this behaves differently So ip do a Transfer(PJSIP/${DESTNUMBER}@trunk) And this results in: Failed to parse destination URI '[destnumber scrubber]' for channel PJSIP/trunk-00000011 Do I have to specify the destination number differently when using Transfer with pjsip that I

Hi List One more Problem I stumbled upon. Using Asterisk in a TSP environement. Incomming IC Calls are e164 and have a NPRN (Routing Number) prefixed. Example: +4198055615995555 +41 country prefix 98055 Routing Prefix 615995555 effective phone number Calls routed to Customers need to be put in the 'local' format. 0615995555 This is also the format of the From / To / Invite header

Hello List Next question where google did not spit out an unsable answer. When redirecting a call with Transfer, I would like to correctly indicate the reason. I did try this: exten => XX,1,NoOp(Call to ${EXTEN} from ${CALLERID(all)}) exten => XX,n,Dial(SIP/ZZ) exten => XX,n,set(REDIRECTING(reason)=cfb) exten => XX,n,Transfer(SIP/YY) I did try with 'reason'

Hi Richard Thank you > You need to set more redirecting information [1]. > > In sip.conf send_diversion=yes needs to be in effect. You also need > to setup > the from party id information (at least the from number) to indicate > where you > are redirecting from. You should also increment the redirecting > count. > > Richard > > [1] >

Dear List I fear I stumbled over a bug in asterisk 13.14.1. My 'phones' are roaming around, sometimes some are connecting from ipv6 enabled networks, another time they are not. If a connection is ipv6 I would prefer to use ipv6 to avoid ipv4-nat problems. I have not specified a transport in the endpoint section, so that the appropriate transport which corresponds to the registration

Hi Joshua > The "rtp_keepalive" option can be used to have the RTP stack send an > RTP packet out. Try that and see what happens. Once again 'bullseye' that fixed the problem. Thank you! Mit freundlichen Gr?ssen -Beno?t Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29

Hi all I have a problem that starts driving me crazy... Win2k3 ADS, added some attributes like loginshell, gid, uid etc. Unix clients use NSS_LDAP to get 'passwd' data and kerberos to authenticate users. Authentication does not happen via LDAP. winbindd is used to autocreate sid => uid/gid mappings. This worked very fine with samba 3.0.14a. Upgraded to samba 3.0.23 Now the owner

Hello List Asterisk 13.14.1 in use with pjsip stack. On the remote side is a SBC which performs some 'nat' detection. I suppose this means the SBC listens from where it is getting RTP data and then replies to that ip. As long as the asterisk is initiating the call this is fine, the asterisk start sending RTP to the media IP of the SBC and the SBC is sending media back. Now I want to do

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

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