On Sun May 20 2007 21:45:43 you wrote: > Mark: > > There is likely a problem in your configuration file. The solution is > quite simple, but the information you have provided is not enough to solve > your issue. The debug messages you provided merely shows the > registration has failed. We need to look at 2 things: > > 1. Your SIP.CONF file ... not the entire file, but the segment that has > your 6010 settings. I suspect you have it under your [6010] context. > > 2. Your Aastra configuration and model number and the exact location where > you entered your credentials. That is "Global SIP" and/or settings under > "Line 1" of your phone. We need to see the exact entries except the > password. > > Its clear from the message that your Aastra is connecting to your Asterisk > server and for some reasons it is not accepting your registration. > Additionally please check the firmware version of your phone. > > Last Friday we were at a clients who purchased close to a dozen phones > (brand new) but had firmware dated May 2005. We upgraded it to firmware > date 2007 version. 1.4.something... and it resolved their issues without > us having to touch any configurations on their phones! > > Waiting for your settings to be able to properly analyze. > > Cheers! > Reza.
Hi Reza, would you have a thought as to why rtp doesn't seem to pass thru a vpn tunnel? Still with this same config; it registers and I can dial the remote, it rings then hangs up immediate. I have reinvite=no and also nat=yes so reinvite should be disabled anyways. it looks to me like the rtp route is not succeeding - is it a property of vpn not to pass udp? some items from the sip debug log - "peer audio RTp is at port 192.168.0.226:3000" - so * sees the port... "capabilities ...." both have ulaw/alaw avail; sip show peers shows monitored status at 135ms - i see no reason that rtp would not pass transparent thru vpn tunnel. in your opinion should i stick a second * at the remote site & use iax instead of trying to make sip/rtp work? thanks for your help.
