I have installed Asterisk@Home on a PC here and need to have it forward calls to the PSTN. We have Cisco CallManager 3.3.4. However I found out that this version doesn't support configuring SIP Trunks. Is there an alternative solution. Thanks Walid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050110/99d6d45f/attachment.htm
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type"> <title></title> </head> <body bgcolor="#ffffff" text="#000000"> -----BEGIN PGP SIGNED MESSAGE-----<br> Hash: SHA1<br> <br> Hello<br> <br> You can use H323 to connect to Cisco CallManager.<br> Add asterisk as an h323 gateway on cisco callmanager.<br> Then you can send & receive call from asterisk.<br> <br> TIP: Use OH323 instead off asterisk h323 native driver.<br> <br> Regards<br> <br> João Amaro<br> <br> <br> <br> Walid Azab wrote:<br> <br> | I have installed Asterisk@Home <a class="moz-txt-link-rfc2396E" href="mailto:Asterisk@Home"><mailto:Asterisk@Home></a> on a PC here<br> | and need to have it forward calls to the PSTN. We have Cisco<br> | CallManager 3.3.4. However I found out that this version doesn't<br> | support configuring SIP Trunks. Is there an alternative solution.<br> | Thanks<br> |<br> | Walid<br> |<br> |<br> | ----------------------------------------------------------------------<br> |<br> |<br> | _______________________________________________ Asterisk-Users<br> | mailing list <a class="moz-txt-link-abbreviated" href="mailto:Asterisk-Users@lists.digium.com">Asterisk-Users@lists.digium.com</a><br> | <a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a> To<br> | UNSUBSCRIBE or update options visit:<br> | <a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br> -----BEGIN PGP SIGNATURE-----<br> Version: GnuPG v1.2.4 (GNU/Linux)<br> <br> iD8DBQFB4plaJUm/Bor63CERAgXMAKDGJA+KXiC0FRnW7yjhJo3+YA3EMQCdEV+A<br> c5tmH6UTgCRW2kDr4mqNoQ4=<br> =gH7x<br> -----END PGP SIGNATURE-----<br> <br> </body> </html>
Sir/Mam, Good PM to all! I'm new to asterisk but I was able to setup a asterisk server using softphones. I have some questions in mind, I have a working asterisk server and I want to add digium cards w/ a telephone line. Will it be able to forward a call from the a person who is in the U.S. using a PC connected to a broadband dsl connection to my residence phone? <<< If you don't want to receive spam,>>> <<< don't connect to the Internet, >>> <<< or don't have an e-mail address. >>> --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term' -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050126/229aab30/attachment.htm
hi folks. i'm planning to connect * to 120 POTS line. i've done some research on FXO cards but unfortunately most manufacturers only make 4 ports/card. the most i've found is 12 ports. so do i have to get 10 of these cards and setup 3 Asterisk servers (assuming each have 4 free PCI slots) link them together with some insane dialplan? or is there an easier way? any suggestions? comments? remarks? parameters? thx. -- Edwin Lam <edwin@officegeneral.com> Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
Edwin Lam wrote:> hi folks. > > i'm planning to connect * to 120 POTS line. i've done some research > on FXO cards but unfortunately most manufacturers only make 4 > ports/card. the most i've found is 12 ports. so do i have to get 10 > of these cards and setup 3 Asterisk servers (assuming each have 4 > free PCI slots) link them together with some insane dialplan? or is > there an easier way? > > any suggestions? comments? remarks? parameters? > thx.You'd want to look into getting a channel bank and connect to it via a T1 card. http://www.voip-info.org/tiki-index.php?page=Asterisk+Channel+Bank