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

[Joe Acquisto]

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" <yossibh at gmail.com> 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 <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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>> >>
>> >>
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