similar to: INVITE Privacy Information

hello all...I have been trying to get a handle on CallerPres with an analog handset. I have usecallingpres=yes in my chan_dahdi.conf file and when I dial *67 on my analog handset I see Disabling Caller*ID on DAHDI/4-1 but when the call is then forwarded to my outbound SIP provider the RPID header is not correct privacy=off;screen=no instead of full and yes how can I correct this?

Hello, on my ISDN phone I can configure that on the next outgoing call, my telephone number should not be transmitted, instead it should be UNKNOWN. How can I configure Asterisk to do the same? Is this a feature/parameter of the driver (chan_capi) that I'm using? BTW: I'm using ISDN and Deutsche Telekom, if the provider makes any difference. Thanks for your help, Stefan --

I have a Centos 7.6 server with samba 4.8.3  configured as a member of an AD domain using "ads' security and the "nss" idmap backend. Clients are unable to access the shares on the server - they repeatedly get asked for their credentials. The smbd log shows the user authenticating and a mapping from the user's SID to their unix uid is found. However, it seems that

Hello, I want to enable call forwarding for asterisk 1.6.0.6 I couldnt seen any config or option on gui or extensions.conf about it. I found some dialing plans to enable it on web as follows: [apps] ; Unconditional Call Forward exten => _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4}) exten => _*21*X.,2,Hangup exten => #21#,1,DBdel(CFIM/${CALLERIDNUM}) exten => #21#,2,Hangup ;

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant

Hi, We are running a samba fileserver, access controlled using posix acl (right 770, with users/groups on the filesystem level. Therefore samba shares look like this: [share] path = /srv/academic read only = no writable = yes create mask = 0770 directory mask = 0770 Now certain users complain that they cannot access certain folders, but looking at the folders from the linux fileystem, their

Hi; How can a user block his caller-id in Astersik? Regards Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041229/07ecf20f/attachment.htm

Hi, how can I setup asterisk to use the number presentation bits on the isdn side to suppress the number presentation? We need to transmit the subscriber number for billing purposes via ISDN whether or not the user wants to hide his/her number. Is there any way to do this? Deti

I've just upgraded to the latest CVS head, and my outbound calls stopped working. I traced it back to the line exten => s,9,CallingPres(${ARG2}) It seems as if this application is now missing. I tracked back the changes and found in 1.415 of chan_zap.c the code was removed because it was "duplicated". However, it does not exist anywhere ! Am I being stupid, missed

Hi, Using * 1.2.5 with a euro_isdn PRI I need to hide the callerID on certain extensions. I have usecallingpres=yes in zapata.conf, and am using SetCallerPres(prohib) in my dialplan prior to the Dial command. No matter what I set SetCallerPres to the CID is still displayed. Is there something else I need to make this work? I can't just set the CallerIDNUM to null, as it is needed for

Hola, compañeros. Pido ayuda con algo que sé que tiene que ser simple, pero la presión de tener que sacarmelo de encima me simplifica a mí demasiado y no me doy cuenta. Tengo una matriz de datos en la que tengo características tipo factor, necesito trasponer esa información a una matriz de datos binarios, en función de algunas combinaciones de esas variables. Unas combinaciones tienen que pasar a

Hi, I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now having a problem with Originate and chan_local. I'm using the following Manager API action to originate a call: Action: originate Priority: 1 Context: trunk Callerid: 100 Channel: Local/100 at callback/n Exten: 123456789 Variable: USERFIELD=127.0.0.1|USEREXT=123456789 WaitTime: 30 This is intended to first call

Hi, Try this: lines1<- readLines(textConnection("gene1 or1|1234 or3|56 or4|793 gene4 or2|347 gene5 or3|23 or7|123456789")) lines2<-readLines(textConnection(">or1|1234 ATCGGATTCAGG >or2|347 GAACCTATCGGGGGGGGAATTTATATATTTTA >or3|56 ATCGGAGATATAACCAATC >or3|23 AAAATTAACAAGAGAATAGACAAAAAAA >or4|793 ATCTCTCTCCTCTCTCTCTAAAAA >or7|123456789

Dear all, I'm in basic setup of my network: I try to do a call from a softphone to an other one but I got the error 603 Declined. Below the sip.conf: *[gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* extension.conf: *[tutorial] exten => 1234,1,Dial(SIP,gianca)* *exten

I'm trying to get Samba up and running and having a terrible time. It says that I should be able to run nmap and see that 137 and 139 are open - which they are not. I have not added any restrictions in smb.conf, do not have a firewall running and I have increased the log level to 5 to see all of the messages. It says that it is talking on 137 but it kind of looks like it's not talking

Hi My head hurts... Can anyone help out here, my remote IAX can see my local IAX and visa versa, conversation starts, I can dial my remote (POTS) landline number, remote end answers, trys to route to local iax2, I see it start the conversation here, the extension (SIP) rings once and then it dies... Both ends are defined with accept IPADDRESS to keep it in the family and simple.. Debug info

Hi, I've migrated from an old samba installation (Samba as PDC) that used TDB backend for password. I've setup a box with ubuntu and samba 3 + ldap and I imported the old users. Old users works fine. I have problems with new users and machines. Old users works but they don't show up with smbldap-usershow command and I've problem in changing their passwords. If I check the ldap

Not sure what's up with the mailing list here. For some reason mails are not coming through. Try again... I am trying to link an asterisk box to my provider's asterisk server via SIP. (I know I could use IAX, but the provider does not allow that, so I can't). When an inbound call happens I get this: Jan 31 13:09:14 NOTICE[3716] chan_sip.c: Failed to authenticate user

Hello, I have an internal extension, e.g. 1005 which is being called from an external/public number like 123456789. Now when it comes to the spoken voicemail information it says something like "number 1000 not available", however it should say "number 123456789 not available". How can I configure this? I already googled and I guess this is really easy, but I just couldn't