[Asterisk-Users] Seperate RTP server, voice not being received.

[Ray Van Dolson]

Our setup:

              SIP                 IAX2                 SIP
<SIP Phone> <-----> <Asterisk1> <------> <Asterisk2> <-----> <PacWest SIP>
                                          |    RTP
                                          \-----------> <PacWest RTP>

I can make or receive calls from my SIP Phone, however voice only works in one
direction. 

I see the following when a call happens on Asterisk2:

Sip read: 
INVITE sip:XXXXXXXXXXXXXX4@<Asterisk2_IP>:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP <PacWest SIP>:5060;branch=z9hG4bKad7c753a29b-a9fbbac5
Via: SIP/2.0/UDP <PacWest RTP>;branch=z9hG4bKac483280508
To: <sip:9167243434@<PacWest SIP>;user=phone>
From: sip:7079953217@<PacWest RTP>;tag=1c483274524
Call-ID: 483273899483274014@<PacWest RTP>
CSeq: 1 INVITE
Max-Forwards: 69
Contact: sip:7079953217@<PacWest RTP>
Record-Route: <sip:<PacWest SIP>:5060;lr>
Supported: em, 100rel, timer, replaces, path
Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER,
INFO, SUBSCRIBE, UPDATE
User-Agent: Audiocodes-Sip-Gateway-TrunkPack 1610/v.4.40.211.387 
Content-Type: application/sdp
Content-Length: 331

<More log sinppets ...>

15 headers, 15 lines
Using latest request as basis request
Sending to <PacWest SIP> : 5060 (non-NAT)
Found peer 'pacwest-peer'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 106
Found RTP audio format 101
Peer audio RTP is at port <PacWest RTP>:9610

So I see the second Via line wanting me to send RTP data to a seperate server.
I can confirm with tcpdump that there is traffic outbound to this server once
the call is up and running.

However, I do not hear any audio on the other end of my call (just a plain old
POTS phone line, also have tried to a cell phone with the same result).

Here are the pertinent entries from my sip.conf file:

;
; User Account for PacWest
;

[pacwest-user]
  type=user
  host=<PacWest SIP>
  context=incoming
  insecure=very

;
; PacWest Peer
;


[pacwest-peer]
  type=peer
  host=<PacWest SIP>
  context=incoming
  insecure=very


Is there any additional setup I need to do for the PacWest RTP server in
either my sip.conf file or rtp.conf file?  From what I understand this should
_just_ work, but my upstream provider (PacWest) is telling me I need to add
something for their RTP server, but it appears to me as if the traffic is
already going outbound although I have no way to know if it's valid or being
accepted, etc.

Just hoping someone can verify that I'm doing the correct setup.  Let me know
if there's any additional info I can provide.

Ray

-- 
Ray Van Dolson
Linux/Unix Systems Administrator
Digital Path, Inc.