Hi. Is there a way to isolate what shows on CLI to just the conversation with that extension? There appears to be a lot of stuff unrelated to this extension.
Packet traces are not out of the question, but cannot be done today. joe a. "Yossi Ben Hagai" <[EMAIL PROTECTED]> Wrote: 4/9/2007 12:56 PM: > 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 >> _______________________________________________ --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
