Re: [Asterisk-Users] Asterisk C source code documentation
Paulo <[EMAIL PROTECTED]> writes: > Hi all, Hi Paulo, > I was wondering if there is any documentation of the Asterisk C > source code. A good point to start is the file apps/app_skel.c which shows the minumum you need in an application. Regards, Fabian Müller ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and nat
"Ashling O'Driscoll" <[EMAIL PROTECTED]> writes: > I first set up asterisk and two clients on the same network and it > worked fine. I now have asterisk set up which is acting as a sip > registrar. It is behind nat. I also have two clients which are behind > nat on two separate networks. I can no longer register the clients. I > have set 'nat=yes' in the client config but is there something else I > must do for the asterisk sevrer itself?... I am only able to give you a link which might help you. Benjamin on Asterisk always writes good articles about nat (and other things as well). Here is one of them: http://lists.digium.com/pipermail/asterisk-users/2004-October/068275.html You probably should read the whole thread. Click the link "thread" on that page and search in the new page for postings with the the subject "Almost there--Remote connection". -- Fabian Müller ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] accept DTMF while beeing in a queue
Hello, I would like to know, if it is possible to accept DTMF signals from a caller while he is in a queue. I would like to accomplish something like this: 1) The caller is in the queue. 2) The caller dials 123. 3) The caller is sent to extension 123. just for your information: When the caller is in the queue and sends a DTMF signal I see this message: DEBUG[327698]: chan_zap.c:3955 zt_read: DTMF digit: 5 on Zap/4-1 Regards, Fabian Müller ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] accept DTMF while beeing in a queue
"Oleg A. Arkhangelsky" <[EMAIL PROTECTED]> writes: > A context may be specified, in which if the user types a SINGLE > digit extension while they are in the queue, they will be taken out > of the queue and sent to that extension in this context. > > See queues.conf. Wow, thanks a lot Oleg. I overlooked that :-( Regards, Fabian Müller ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip to pstn gateway
Hello, here is what I am able to do: - I am able to register a SIP Phone on an Asterisk server. - I am able to call an extension on the remote Asterisk server with my SIP phone and hear the congratulation message. Informations about the configuration: - There is no phonecard (no digium card, no isdn card ...) in the Asterisk box but only a network interface card which connects the Asterisk server to a softswitch. This softswitch is the gateway to PSTN. Unfortunately I do not know anything about the softswitch. Is there something important that I should know about it? Here is what I would like to have: - I would like to be able to call a PSTN/ISDN phone with my SIP phone. - That means when I take my SIP Phone and dial a telephone number that belongs to the PSTN Asterisk must route the SIP packages to the softswitch which in turn routes the call to the PSTN. - When I dial another registered SIP phone Asterisk should connect the two sip phones so that they can speak to each other. - -- | SIP phone | ---> | Asterisk | ---> | Softswitch | ---> | PSTN | - -- | | - |> | Sip phone | - I have no idea how to configure Asterisk to accomplish this task. I started reading documents like ftp://ftp.isi.edu/in-notes/rfc3372.txt and the sip RFC and Mailing List articles and so on but they did not make me able to configure Asterisk in that way. Does anybody know where I can find documents that describe how I can do what I would like to have? Do I have to configure a SIP Proxy (SER for example) on the Asterisk box or does it work without a SIP proxy as well? Do I have to register the Asterisk box on the softswitch? (Should this be possible at all?) Thanks very much in advance for any kind of help. Regards, Fabian Müller ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Mode supported by VoiceMail Application
"Jean-Marc Salsa" <[EMAIL PROTECTED]> writes: > Which mode should I force into sip.conf ( general, only for peer ? ) > so that the Voicemail application is understanding password from users ... This depends on what your users are using. If you are using a Grandstream device you can configure in its administration interface which dtmf mode the telefone should use. If your IP phone is configured to use rfc2833 for example then you would write dtmfmode=rfc2833 in your sip.conf. If all users use the same dtmfmode it should be ok to write this to the general section. Fabian Müller ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unknown RTP codec 100 received
"Nedi" <[EMAIL PROTECTED]> writes: > I am new to asterisk. My system is ASTLINUX > if receive a Fax on my sipura spa2000 > > i get this: Feb 25 07:41:00 NOTICE[1708]: rtp.c:564 ast_rtp_read: Unknown RTP > codec 100 received Probably point 6 on http://www.asteriskguru.com/tutorials/unknown_codec_received.html can help you. Fabian Müller ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sendtext or sip message - where in RFC
> I was looking in apps/sendtext.c hoping to find a reference > to the RFC number and section etc where this is talked about. Because sendtext.c is not SIP specific you will not find a reference to SIP related information there. chan_sip.c has a reference to RFC 3428 (http://www.rfc-editor.org/rfc/rfc3428.txt). Have a look at the comment of the function receive_message() in chan_sip.c. Fabian Müller ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do you recompile individual source modules?
Bart Fisher <[EMAIL PROTECTED]> wrote: > If I understand, I cd to asterisk source folder and run make - it take > card of rest? > > Also, when/why should you use astxs? I repeat in other words what Russell said: When you have a clean source tree and type make a lot of source files are compiled. When you type make again, nothing gets recompiled. Now when you change an individual source file and type make again only the modfied source file gets recompiled. If you start with a clean source tree and want to compile only one file you can use the perl script contrib/scripts/astxs. For example if you want to compile only apps/app_skel.c you do this by typing contrib/scripts/astxs apps/app_skel.c You will have to make astxs executable for this to work: chmod +x contrib/scripts/astxs (For these two commands you have to be in the root directory of your Asterisk sources of course.) Fabian Müller ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.4 download
<[EMAIL PROTECTED]> wrote: > How do I download the development branch of asterisk 1.4. I am > eagerly waiting for it. svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk See http://www.asterisk.org/download for further information. Fabian Müller ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What I can use with ASTERISK to call clients to remind them about their appointments
dmitri smirnoff <[EMAIL PROTECTED]> wrote: > What I can use with ASTERISK to call clients to remind them about their > appointments? You can use Callfiles. Look how the wake-up script works: http://www.voip-info.org/wiki/index.php?page=Asterisk+tips+Wake-Up+Call+PHP Also have a look at http://www.voip-info.org/tiki-print.php?page=Asterisk+auto-dial+out Fabian Müller ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] placing a call with the Manager interface
Eric <[EMAIL PROTECTED]> wrote > Action: Originate > Channel: SIP/dualphone > Exten: SIP/snom > Context: default > Priority: 1 > Timeout: 500 > CallerID: Dualphone <9> > Async: true > > [...] > > Any ideas what is causing this hangup? Can you please try to increase the 'Timeout'? What happens when you set it to this: Timeout: 5000 Fabian Müller ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
