From: Michael V. D. <mv...@va...> - 2004-02-13 01:37:27
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Steve Sokol and I have been trying to track down this problem for quite = some time. I think that the problem had nothing to do with scheduling, but with = native transfers. (I think that iax.conf needs a canreinvite parameter, like sip.conf) I changed the debug code to print to a log file (since wxWindows appears = to eat stderr) and found that the problems seemed to be occurring when asterisk = sent a TXREQ and the softphone sent a TXCNT. And another TXCNT. And another = TXCNT. And ..... the asterisk sent several TXREJs and then audio stopped = flowing. I tried to follow the chan-iax2 code in the asterisk code, but then = looked at the iaxlib code on the digium ftp site. Then I looked at the CVS *libiax2* code. Aha, we're out of date. I = noticed that we weren't applying the apparent_address when we were sending our = TXCNT, so I just copied the iax.c from digium's libiax2 CVS into the iaxclient's libiax2/src and recompiled. IT WORKS (mostly) iaxComm -> asterisk -> iaxComm works fine if there's no NAT anywhere in = the picture. Steve Sokol tells me that=20 iax Phone ->| |linksys router -> asterisk iax Phone ->| doesn't work. I've found that iaxComm -> asterisk -> NAT VPN -> iaxComm doesn't work. Steve Kann: I'd like to update the iaxclient CVS to have iax.c from = digium, but with the addition of Steve Sokol's blind_transfer function, OK? Michael (aka Steve V) |