I have been running Asterisk for years on a machine with a public
IP. Most recently, I have been running 1.2.17, from the day it
came out, with no (noticeable) problems.
Yesterday, I switched over to a new server that is on the same
public subnet, one higher than the original server.
I built 1.2.17 from source on that machine (as I did on the old
server). My firewall on the new machine is configured identically
to the old one as well.
All of my IAX connections just worked. All but one of my SIP
connections just worked as well (which is why I can't believe it's
a firewall issue).
StanaPhone, which I use for 2 incoming DIDs, registers correctly,
and rings my phones correctly when a call comes in. However, once
answered, there is dead silence in both directions, on 100% of the
calls.
There isn't any problem on StanaPhone's side (which has provided a
_fantastic_ service ever since I signed up!), because I can
connect to them with X-Lite and receive calls with audio. More
importantly, if I fire up Asterisk on the old server, it still
works!!! I can connect with X-Lite to the new server, so the new
server definitely accepts SIP connections, and audio works.
It's _not_ a codec problem. I verified that on both the working
and non-working servers the connection is established with ulaw on
both sides.
I have dumped the "peer" and the "channel" on both, while the call
was active, and they look identical to me, except for the random
bits associated with a particular connection. Here are the ones
from the machine that fails:
*CLI> sip show peer XXXXXXXXXX
* Name : XXXXXXXXXX
Secret : <Set>
MD5Secret : <Not set>
Context : default
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : No
Callerid : "" <>
Expire : -1
Insecure : port,invite
Nat : RFC3581
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Trust RPID : No
Send RPID : No
DTMFmode : rfc2833
LastMsg : 0
ToHost : sip.stanaphone.com
Addr->IP : 204.147.183.18 Port 5060
Defaddr->IP : 0.0.0.0 Port 0
Def. Username: 12345678
SIP Options : (none)
Codecs : 0x4 (ulaw)
Codec Order : (ulaw)
Status : OK (20 ms)
Useragent :
Reg. Contact :
new*CLI> sip show channel
[EMAIL PROTECTED]
* SIP Call
Direction: Outgoing
Call-ID: [EMAIL PROTECTED]
Our Codec Capability: 4
Non-Codec Capability: 1
Their Codec Capability: 4
Joint Codec Capability: 4
Format ulaw
Theoretical Address: 204.147.183.18:5060
Received Address: 204.147.183.18:5060
NAT Support: RFC3581
Audio IP: AAA.BBB.CCC.DDD (local)
Our Tag: as360c7ca5
Their Tag: 0bd46ffd48e4fbffb3a68f13f8ad2599
SIP User agent:
Username: 87654321
Peername: 12345678
Original uri: sip:204.147.183.55:1024
Need Destroy: 0
Last Message: Tx: ACK
Promiscuous Redir: No
Route: sip:204.147.183.18;ftag=as360c7ca5;lr=on
DTMF Mode: rfc2833
SIP Options: (none)
Finally, I built 1.2.18 from source today, and everything is
working perfectly _except_ for StanaPhone, which continued to
connect with no problems, but deliver no audio in either direction.
I have no idea what else to try, and would appreciate _any_ guidance.
Thanks in advance!
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