search for: ast_state_down

...tch (see below) - EuroISDN with Point-to-Point (ptp) mode (Austria) - With Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k from debian sarge But am having some problems: 1) I needed to patch chan_capi.c from debian sarge (see below) to give the new channel to asterisk in state AST_STATE_RING instead of AST_STATE_DOWN. I just did the same for PTP as is done for PTM (point to multipoint) in the code and it works (sort of) for me. If I leave it in AST_STATE_DOWN, the call is never seen by asterisk and the caller gets a failure indication. Is this patch really necessary or am I missing something obvious...

Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's.

Hi. I was wondering if there's a parameter to limit the number of concurrent sessions in IAX, globally or on a per-user basis. That could be needed for security purposes (to prevent dos attacks), to limit bandwidth / cpu usage, or to not allow more than N guest connections, for example. Any other VoIP channel support that? (like SIP, MGCP) Matteo. -- Brancaleoni Matteo

Hi there, Besides the problem of Voicetronix dialing too early before the carrier gives a dial tone, there also appears to be issues with the audio quality when more than 1 channel is utilized.

...3:30:19.000000000 +0 @@ -2141,7 +2175,8 @@ ast_pthread_mutex_init(&(p->lock),NULL); i->mypipe = p; if (i->isdnmode) { - p->c = capi_new(i,AST_STATE_DOWN); + //RSC + p->c = capi_new(i,AST_STATE_RING); i->state = CAPI_STATE_DID; } else { p->c = capi_ne...