I installed * on a virtual server in our hosting environment. (The hosting people were complaining about latency, I figured there was no better test of latency than *.) I was using X-Lite to make calls. The calls would come up, then when I click hang up on X-Lite it would sit there for a while then drop. Looking at show channels, the call was still in place and ethereal was showing RTP packets still coming in. * was also throwing an error about max retries on sending a packet?
Looking at the SDP packets, the server was telling the client to send the RTP packets back to 127.0.0.1. The virtual server has a weird network setup..
venet0 Link encap:UNSPEC HWaddr 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
inet addr:127.0.0.1 P-t-P:127.0.0.1 Bcast:0.0.0.0 Mask:255.255.255.0
UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1
RX packets:140238 errors:0 dropped:0 overruns:0 frame:0
TX packets:167187 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:30016231 (28.6 Mb) TX bytes:35546749 (33.9 Mb)
venet0:0 Link encap:UNSPEC HWaddr 00-00-00-00-00-00-00-00-00-00-00-00-00-00-00-00
inet addr:209.43.121.215 P-t-P:209.43.121.215 Bcast:209.43.121.215 Mask:255.255.255.255
UP BROADCAST POINTOPOINT RUNNING NOARP MTU:1500 Metric:1
RX packets:140238 errors:0 dropped:0 overruns:0 frame:0
TX packets:167187 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:30016231 (28.6 Mb) TX bytes:35546749 (33.9 Mb)
From the looks of it, * was sending the IP from "venet0" back to the client for the return of the RTP stream. In the sip.conf file, I put "bindaddr=209.43.121.215". After that, calls come up normal and end completely normally. Like I said, this is probably not the problem in your situation.. but hopefully it'll lead you in the right direction?
Tom
On Jul 23, 2004, at 5:43 AM, David Wilson wrote:
Hi there,
I'm having problems with the Grandstream Budgetone 101 on hangup - "show channels"/"show channels concise" output is still showing the call's channels as active.
The problem does not exist when I use SJPhone, so I'm assuming it isn't an Asterisk configuration issue. Has anyone seen this, or better, does anyone have a fix? :)
Thanks,
David.
--
Before you judge a man, walk a mile in his shoes. After that, who cares?
He's a mile away and you've got his shoes.
-- Billy Connely
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