[Asterisk-Users] unsubscribe
Jeremy Melanson wrote: Hello all. I'm trying to see if anyone knows of an alternative solution, commercial or non-commercial, to SpanDSP. I'm specifically looking for another software-based, DSP fax that doesn't require me to add a tie up a bunch of extensions on my PBX. Has anyone ever seen such an animal, or gotten such it to play nice with Asterisk? ___ 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 -- - Mike S. Lacanilao, [EMAIL PROTECTED], [EMAIL PROTECTED] Hyperion System Group (Globalink Corp.) Registered Linux User: #242093 GPG public key:0xDE01A745 -- Read The Manual Before Asking!! Not by might, nor by power.. but by Spirit, says the Lord ___ 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
[Asterisk-Users] voice output
hi list, im using xlite softphone clients in the windows box, and i can make a call (connection) in each other, but im wondering why is that i cannot hear any voice from client's end (vice-versa), i already checkd the volume of the speaker, is there something missing in the configuration file of asterisk or in the xlite config?? Thanks in advanced -mike -- - Mike S. Lacanilao, [EMAIL PROTECTED], [EMAIL PROTECTED] Hyperion System Group (Globalink Corp.) Registered Linux User: #242093 GPG public key:0xDE01A745 -- Not by might, nor by power.. but by Spirit, says the Lord ___ 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] voice output
hi, thanks for the reply. i just want to clarify that my asterisk box and client softphone are in the same network as follows: 1. 192.168.17.57 = asterisk 2. 192.168.17.59 = client1 3. 192.168.17.60 = client2 so you mean that the even the same network, the firewall will affect the channeling of data accross the asterisk? Wilson Pickett wrote: i cannot hear any voice from client's end This is a common problem if you are using NAT (behind router) Google for asterisk one way audio and take a loog here http://www.voip-info.org/wiki-Asterisk+SIP+NAT ___ 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 -- - Mike S. Lacanilao, [EMAIL PROTECTED], [EMAIL PROTECTED] Hyperion System Group (Globalink Corp.) Registered Linux User: #242093 GPG public key:0xDE01A745 -- Read The Manual Before Asking!! Not by might, nor by power.. but by Spirit, says the Lord ___ 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
[Asterisk-Users] newbie question
hi all, its my first time to post here, im in the process of building asterisk based telephone system (just small). i already installed asterisk server, i just wanted to test 2 sip softphones to get working before i move on, is it possible to have 2 softphones talk to each other without any cards?? (i.e. digium, just for testing purposes) im using kphone, and i follow the instruction on how to configure it for asterisk, but there is a problem when i register the kphone, is there any configuration must follow before i make a calls using asterisk?? pls help. THanks, mike ___ 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
