-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 > Hello Daniel,
Hi Dana. > You will find the information at http://www.voip-info.org/ and > http://oreilly.com/catalog/9780596510480/ (.PDF downloadable from the > "Online Book" link) very useful. I have the second edition that covers Asterisk 1.4 and it seems interesting. You made me remember that I had downloaded it the last year, although just now I have more time to dedicate to Asterisk. The fact of to have already installed it is an important step :-) > The asterisk package by itself should be adequate for SIP/IAX calls. > I don't think you need libpri unless you are planning on connecting asterisk > to a digital connection such as ISDN or a PRI. > You will need Zaptel (for Asterisk versions 1.2,1.4) or DAHDI (Asterisk > versions >=1.6) if you choose to install an internal card (OpenVOX, Digium, > Sangoma, etc.) I do not know if or how well this will work with a VM. Thanks for the indication. According to I saw in the site of Asterisk[1], only make reference to DAHDI for Asterisk 1.4, but according to which you say to me, both can be used. My idea is to buy an ATA to connect a conventional telephone and make tests of communication between it and softphone. The idea by which I thought about using an ATA is because I am not sure with my version of KVM (KVM-62) can make PCI pass through. But with the ATA must not have problem. Having this in mind, I installed the packages dahdi-linux-2.1.0.4.tar.gz and dahdi-tools-2.1.0.2.tar.gz having loaded only the module dahdi_dummy and so far commenting all that appear in /etc/dahdi/modules. > I suggest testing your SIP softphone with the Echo() and/or Playback() > dialplan applications before attempting to call another > softphone/hardphone/etc. This will allow you to confirm that the one > endpoint functions properly before adding more complexity by calling another > endpoint. I was testing and sometimes with Echo() and MusicOnHold the sound is broken. Is there some form to solve this? > some things that allow you to call a conventional telephone: > an ATA with an FXS port > an internal card (such as OpenVOX, Digium, Sangoma) with an FXS port > call a conventional phone number through the PSTN (below) > > To connect to the PSTN you can use any of: > an ATA with an FXO port (plug an analog phone line into it) > internal card with an FXO port (also to plug an analog phone line in) > account with an ITSP (there is occasionally discussion on the list about > advantages/issues/opinions/and flames with various ITSPs - google > "site:lists.digium.com ITSP") > [...] I believe that with the example I understood a little better how it works. As it mentioned above, I am thinking about buying a Linksys SPA3102 to make both internals and with PSTN tests. > Hope that gets you going in the right direction. > > http://www.voipsupply.com/ is a good source to see what equipment is > generally available to end users. Thanks for your reply and by all the references and examples that you provided to me. Regards, Daniel [1] http://www.asterisk.org/downloads -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkoHNXIACgkQZpa/GxTmHTccUACfbHY+st10rhsqrsZnE9SJLZrV hFQAnR5Y85XQQr7Jm1wWzD106qxNkd4g =EYud -----END PGP SIGNATURE----- _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users