Re: [Asterisk-Users] Gsm + snom phones
Matteo Brancaleoni wrote: Hi. About a month ago I made a test with snom200b. At least then it worked ok with *. At the moment I'm using mainly g711a. So, there is always a possibility something but you also tested gsm ? It works for me with gsm :-) 6 snom 200 and one snom 105 with gsm over german t-dsl (128 Kbit) sound is really good :-) Greets,Matteo. -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ 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 -- Best regards Detlef Wengorz [EMAIL PROTECTED] Detlef Wengorz [EMAIL PROTECTED] ___ 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] ISDN CAPI and anonymous callers
Walter Doerr wrote: Hello, I am trying to use * to handle anonymous ISDN callers. Something like exten = 5150/0,1,Congestion use: exten = 5150/,1,Congestion should work but doesn't. Apparently because the ISDN CAPI doesn't use 0 for callers who don't send their number. Is there a way to make * identify ISDN callers who use CLIR? -Walter -- Walter Doerr =*= [EMAIL PROTECTED] =*= FAX: +49 2421 962001 The poor folks who only have 100MBytes of RAM five years from now may not be able to buffer a 16MB packet, but that's their tough luck. (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT) ___ 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 -- Best regards Detlef Wengorz [EMAIL PROTECTED] Detlef Wengorz [EMAIL PROTECTED] ___ 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] Strange code in rtp.c / disconnect - maybe reinvite problems
Daniel Chabrol wrote:
Hi List!
I get WARNING[14351]: File rtp.c, Line 1202 (ast_rtp_bridge): codec0 =
524300 is not codec1 = 524300, can't do reinvite at my asterisk console.
The code there looks realy strange:
codec0 = pr0-get_codec(c0);
codec1 = pr1-get_codec(c1);
ast_log(LOG_WARNING, codec0 = %d is not codec1 = %d, can't do
reinvite\n,codec0,codec1);
/* Hey, we can't do reinvite if both parties speak diffrent codecs */
if (codec0 != codec1)
return -2;
I think the message should only occur *after* checking equality:
if (codec0 != codec1) {
ast_log(LOG_WARNING, codec0 = %d is not codec1 = %d, can't do
reinvite\n,codec0,codec1);
return -2;
}
I hoped this can't do reinvite would explain my disconnects from the
nikotel.com sip server after 60 seconds. But this little bug seems only
to be display-specific and not affect funtion. But maybe i oversight
That's correct :-(
but change the code like this
if (codec0 != codec1) {
ast_log(LOG_WARNING,
codec0 = %d is not codec1 = %d, can't do
reinvite\n,codec0,codec1);
ast_mutex_unlock(c0-lock); // unlock before return
ast_mutex_unlock(c1-lock); // unlock before return
return -2;
}
and try again.
maybe it helps.
something which still disables the reinvite even if i use
canreinvite=yes in my sip.conf:
[nikotel]
type=friend
username=USERID
fromuser=USERID
secret=PASSWORD
host=calamar0.nikotel.com
canreinvite=yes
context=internal
; no nat entry because im not using nat!
Is there someone which is able to use Nikotel.com with the current
CVS-Version (in my case CVS-11/24/03-19:24:22). BTW: 0.5.0 don't work
too in my case (at least not longer than 60 seconds). Pulver.com calls
and so on are no problem. Any suggestions?
Best regards,
Daniel
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Best regards
Detlef Wengorz [EMAIL PROTECTED]
Detlef Wengorz [EMAIL PROTECTED]
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