[asterisk-users] "remote" SIP, no audio, or one way audio.

[Yossi Ben Hagai]

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 <joea at j4computers.com> 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:3306 at xx.xx.xx.xx SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK363305c9;rport
> From: "nnnnn"<sip:3306 at xx.xx.xx.xx;tag=as67e5c857
> To: "nnnnn"<sip:3306 at xx.xx.xx.xx>;tag=9c58a77e
> Contact: <sip:3306 at 192.168.0.xxx>
> 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 <david.cantera at iacnet.net> 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 - 58867de69e90aa51 at 192.168.15.100
> > 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:401 at 192.168.15.100:5060;user=phone>
> >
> >
> >
> > Joe Acquisto wrote:
> >> Steve Totaro <stevetotaro at hotmail.com> 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
> >>
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