I snipped all of the previous data, as I'm trying to "boil down" this problem to its essence...

I turned off the firewall for a few seconds, and still got no audio. For those that will be suspicious, the commands were:

shorewall stop
shorewall clear

tested connection, no audio

shorewall start

I also have a SIPPhone number, which (obviously), connects via SIP. I called that number from the outside, using one of their "Access Numbers", and my phone rang and I heard audio in both directions (this with the firewall back on), so SIP definitely works, just not with StanaPhone.

Then I connected from another server that I run, which is behind a NAT router. That server is running 1.2.18 (as is the one that isn't working, but is on a public IP). Audio works perfectly with this one.

To my knowledge the only difference between them is that the two servers that work are both Red Hat 9, with Asterisk 1.2.18 built from source. The one that fails is CentOS 5.0, with Asterisk 1.2.18 built from source. Here is a dump of the active channel from the NAT'ed server, which _works_:

  * SIP Call
  Direction:              Incoming
Call-ID: [EMAIL PROTECTED]
  Our Codec Capability:   1822
  Non-Codec Capability:   1
  Their Codec Capability:   262
  Joint Codec Capability:   262
  Format                  ulaw
  Theoretical Address:    204.147.183.18:5060
  Received Address:       204.147.183.18:5060
  NAT Support:            RFC3581
  Audio IP:               XX.XX.XX.XX (local)
  Our Tag:                as78cfb201
  Their Tag:              da6aae9eb017f29b6c9de270fb85c352
  SIP User agent:         Sippy
  Original uri:           sip:204.147.183.55:1024
  Caller-ID:              XXXXXXXXXX
  Need Destroy:           0
  Last Message:           Rx: ACK
  Promiscuous Redir:      No
Route: sip:204.147.183.18;ftag=da6aae9eb017f29b6c9de270fb85c352;lr=on
  DTMF Mode:              rfc2833
  SIP Options:            (none)

The only things edited above are the Audio IP, which is my correct "local" (before NAT) server address, and my Caller-ID. Everything else is unchanged.

Here is the channel with dead audio:

  * SIP Call
  Direction:              Incoming
Call-ID: [EMAIL PROTECTED]
  Our Codec Capability:   1542
  Non-Codec Capability:   1
  Their Codec Capability:   262
  Joint Codec Capability:   6
  Format                  ulaw
  Theoretical Address:    204.147.183.18:5060
  Received Address:       204.147.183.18:5060
  NAT Support:            RFC3581
  Audio IP:               XX.XX.XX.XX (local)
  Our Tag:                as45dbcfef
  Their Tag:              420bab62c5da9eae42686897ae65a385
  SIP User agent:         Sippy
  Original uri:           sip:204.147.183.55:1024
  Caller-ID:              XXXXXXXXXX
  Need Destroy:           0
  Last Message:           Rx: ACK
  Promiscuous Redir:      No
Route: sip:204.147.183.18;ftag=420bab62c5da9eae42686897ae65a385;lr=on
  DTMF Mode:              rfc2833
  SIP Options:            (none)


The same two fields are edited above, and both were correct.

To my eye, these are identical. Both are selecting ulaw, correctly. I'm stumped. I guess that I didn't do any packet tracing, but I'm not sure what the value of that would be given that it's not a firewall problem...

Suggestions welcome!
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to