Hi Joe, The debug trace you've enclosed is a NOTIFY message sent from * for the message waiting feature - and is not related to the call. 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.
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. Joss. On 4/9/07, Joe Acquisto <[EMAIL PROTECTED]> wrote:
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. I see no CODEC dialog. What I see is six iterations of the below: . . . . --- Retransmitting #6 (NAT) to xx.xx.xx.xx:64909: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK363305c9;rport From: "nnnnn"<sip:[EMAIL PROTECTED];tag=as67e5c857 To: "nnnnn"<sip:[EMAIL PROTECTED]>;tag=9c58a77e Contact: <sip:[EMAIL PROTECTED]> Call-ID: MjY1NTgyYzQ0YTdjNmM3NTJkODE2ODM1ZmNhMWE3OGE. CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: terminated;reason=timeout Content-Length: 0 ----- Does this imply anyting to anyone? Call can be made, after this. joe a. ****** dave cantera <[EMAIL PROTECTED]> Wrote: 4/7/2007 3:53 PM: > joe, > when I have problems with audio and other connections seem to work, I > always look for a codec incompatibility... use 'sip set debug peer > <extension>' and look for the codec handshaking... make sure both > extensions have a compatible codec choice... > daveC > > Using INVITE request as basis request - [EMAIL PROTECTED] > Found user '401' > Found RTP audio format 0 > Found RTP audio format 8 > Found RTP audio format 3 > Found RTP video format 99 > Peer audio RTP is at port 192.168.15.100:5004 > > *Found description format PCMU for ID 0 > Found description format PCMA for ID 8 > Found description format GSM for ID 3 > Found description format H264 for ID 99 > > *Capabilities: us - 0x20000e (gsm|ulaw|alaw|h264), peer - > audio=0x20000e > (gsm|ulaw|alaw|h264)/video=0x200000 (h264), combined - 0x20000e > (gsm|ulaw|alaw|h264) > > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 > (nothing), combined - 0x0 (nothing) > Peer audio RTP is at port 192.168.15.100:5004 > Peer video RTP is at port 192.168.15.100:5006 > Looking for 404 in inbound-video (domain sip3701.ibsonecall.com) > list_route: hop: <sip:[EMAIL PROTECTED]:5060;user=phone> > > > > Joe Acquisto wrote: >> Steve Totaro <[EMAIL PROTECTED]> Wrote: 4/4/2007 8:44 PM: >> >>> Joe Acquisto wrote: >>> >>>> Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite >>>> softphones, for eval/testing. They do get registered, and can call each >>>> other, but mostly get no audio, sometimes one way audio. >>>> >>>> Suggestions/fixes? >>>> >>>> joe a. >>>> >>>> >>> Is there NAT on both sides? Are you using qualify? Paint a clearer >>> picture. >>> >>> >> >> >> Sorry, I missed your reply, till now. >> >> ------------------switch >> | | |----phones >> | |---------asterisk box >> >> |---------------IPcop------------|---internet-----|-----home/remote-office-- >> --|----sip phone >> >> |-----ditto >> >> Hope that is intelligible. >> >> joe a >> >> _______________________________________________ >> --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 >> >> >> >> > > -- > Building Strong Relationships w/ Intelligent Customer Service > -- > > Interlocking Business Solutions, LLC > 856-380-0894 x5000 > > > _______________________________________________ > --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 _______________________________________________ --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
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