Re: [Asterisk-Users] Re: asterisk-grandstream call

2004-02-11 Thread Michael Koehler
Asterisk is ignoring the codec offer of the caller.  Asterisk is always sending the whole codec list inside 200 OK (on invites), which should be just a subset of that what is received before within the dialog initiating invite. Workaround:  Try "disallow=gsm" regards, Michael Bill Michaelso

Re: [Asterisk-Users] Re: asterisk-grandstream call

2004-02-10 Thread Andres
Bill Michaelson wrote: I am trying to muddle my way tthrough getting something - actually anything to work - with Asterisk. I've acquired a Grandstream phone and I've got * on a Red Hat 9 box. I've gotten to a point where I can see (via ethereal) that the phone REGISTER's successfully with

[Asterisk-Users] Re: asterisk-grandstream call

2004-02-10 Thread Bill Michaelson
>I am trying to muddle my way tthrough getting something - actually >anything to work - with Asterisk. I've acquired a Grandstream phone and >I've got * on a Red Hat 9 box. I've gotten to a point where I can see >(via ethereal) that the phone REGISTER's successfully

[Asterisk-Users] Re: asterisk-grandstream call

2004-02-10 Thread Doug Meredith
Bill Michaelson <[EMAIL PROTECTED]> wrote: >I am trying to muddle my way tthrough getting something - actually >anything to work - with Asterisk. I've acquired a Grandstream phone and >I've got * on a Red Hat 9 box. I've gotten to a point where I can see >(via ethereal) that the phone REGIST

[Asterisk-Users] Re: asterisk-grandstream call

2004-02-09 Thread Bill Michaelson
Right - OK - sans comments for brevity: sip.conf: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls [248379] username=billdesk type=friend host=dynamic canreinvite=no mailbox=1234 context=demo extensions.conf: [general