<div>Hi Joe,</div>
<div> </div>
<div>The debug trace you've enclosed is a NOTIFY message sent from * for the message waiting feature - and is not related to the call.</div>
<div>You can however tell that something is wrong since the message is being retransmitted since the server didn't receive 200 OK in reply - while it could be due to the client being offline or not supporting this feature It could imply a NAT issue so try to recheck your NAT configs.
</div>
<div> </div>
<div>can you post a full trace (starting with the INVITE message)? also you can try to run a sniffer trace on the client side to see if it receives/sends the messages correctly.<br> </div>
<div>Joss.<br> </div>
<div><span class="gmail_quote">On 4/9/07, <b class="gmail_sendername">Joe Acquisto</b> <<a href="mailto:joea@j4computers.com">joea@j4computers.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">I never get this far, apparently. While the connection seems to be made, and calls can be "completed" (rings, answers) there is no audio. On CLI, I can see what appears to be call being made and connected. These are x-lite phones (for testing, one hopes) there appears to be no codec selection available.
<br><br>I see no CODEC dialog. What I see is six iterations of the below:<br><br>. . . .<br>---<br><br>Retransmitting #6 (NAT) to xx.xx.xx.xx:64909:<br>NOTIFY <a href="mailto:sip:3306@xx.xx.xx.xx">sip:3306@xx.xx.xx.xx</a>
SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://192.168.0.202:5060">192.168.0.202:5060</a>;branch=z9hG4bK363305c9;rport<br>From: "nnnnn"<<a href="mailto:sip:3306@xx.xx.xx.xx">sip:3306@xx.xx.xx.xx</a>;tag=as67e5c857
<br>To: "nnnnn"<<a href="mailto:sip:3306@xx.xx.xx.xx">sip:3306@xx.xx.xx.xx</a>>;tag=9c58a77e<br>Contact: <<a href="mailto:sip:3306@192.168.0.xxx">sip:3306@192.168.0.xxx</a>><br>Call-ID: MjY1NTgyYzQ0YTdjNmM3NTJkODE2ODM1ZmNhMWE3OGE.
<br>CSeq: 102 NOTIFY<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Event: message-summary<br>Content-Type: application/simple-message-summary<br>Subscription-State: terminated;reason=timeout<br>Content-Length: 0<br>-----
<br><br>Does this imply anyting to anyone?<br><br>Call can be made, after this.<br><br>joe a.<br><br>******<br>dave cantera <<a href="mailto:david.cantera@iacnet.net">david.cantera@iacnet.net</a>> Wrote: 4/7/2007 3:53 PM:
<br>> joe,<br>> when I have problems with audio and other connections seem to work, I<br>> always look for a codec incompatibility... use 'sip set debug peer<br>> <extension>' and look for the codec handshaking... make sure both
<br>> extensions have a compatible codec choice...<br>> daveC<br>><br>> Using INVITE request as basis request - <a href="mailto:58867de69e90aa51@192.168.15.100">58867de69e90aa51@192.168.15.100</a><br>> Found user '401'
<br>> Found RTP audio format 0<br>> Found RTP audio format 8<br>> Found RTP audio format 3<br>> Found RTP video format 99<br>> Peer audio RTP is at port <a href="http://192.168.15.100:5004">192.168.15.100:5004
</a><br>><br>> *Found description format PCMU for ID 0<br>> Found description format PCMA for ID 8<br>> Found description format GSM for ID 3<br>> Found description format H264 for ID 99<br>><br>> *Capabilities: us - 0x20000e (gsm|ulaw|alaw|h264), peer -
<br>> audio=0x20000e<br>> (gsm|ulaw|alaw|h264)/video=0x200000 (h264), combined - 0x20000e<br>> (gsm|ulaw|alaw|h264)<br>><br>> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0<br>> (nothing), combined - 0x0 (nothing)
<br>> Peer audio RTP is at port <a href="http://192.168.15.100:5004">192.168.15.100:5004</a><br>> Peer video RTP is at port <a href="http://192.168.15.100:5006">192.168.15.100:5006</a><br>> Looking for 404 in inbound-video (domain
<a href="http://sip3701.ibsonecall.com">sip3701.ibsonecall.com</a>)<br>> list_route: hop: <sip:401@192.168.15.100:5060;user=phone><br>><br>><br>><br>> Joe Acquisto wrote:<br>>> Steve Totaro <
<a href="mailto:stevetotaro@hotmail.com">stevetotaro@hotmail.com</a>> Wrote: 4/4/2007 8:44 PM:<br>>><br>>>> Joe Acquisto wrote:<br>>>><br>>>>> Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite
<br>>>>> softphones, for eval/testing. They do get registered, and can call each<br>>>>> other, but mostly get no audio, sometimes one way audio.<br>>>>><br>>>>> Suggestions/fixes?
<br>>>>><br>>>>> joe a.<br>>>>><br>>>>><br>>>> Is there NAT on both sides? Are you using qualify? Paint a clearer<br>>>> picture.<br>>>><br>>>>
<br>>><br>>><br>>> Sorry, I missed your reply, till now.<br>>><br>>> ------------------switch<br>>> | | |----phones<br>>> | |---------asterisk box<br>>>
<br>>> |---------------IPcop------------|---internet-----|-----home/remote-office--<br>>> --|----sip phone<br>>><br>>> |-----ditto<br>>><br>>> Hope that is intelligible.<br>>><br>
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