[Asterisk-Dev] HELP: BYE-request not sent to SIP-peer
Josh Roberson
twisted at indigent-networks.com
Fri Aug 13 21:13:13 MST 2004
Roland, It would have been nice to post a followup :P
A few of us took a crack at this on IRC, and have decided that the real
problem here are the contact headers being set by the provider, and * is
not at fault at all, since we honored the contact headers. Many thanks
to pfn, as he pointed out the providers headers were wrong, and asked
Roland to set nat=yes, which solved the issue temproarily.
-Josh (twisted)
Roland Zagler wrote:
>Hello,
>
>When i have a "Hangup" in my dialplan (extensions.conf) the RFC says to
>terminate the session is to send a BYE request to UA. After tracing the
>traffic on port 5060 UDP i recognized that my asterisk is NOT sending a
>BYE request to it's peer, so the peer doen't know to end the session and
>continues to send RTP packages to me. Does anyone know how to fix this?
>
>Here's the complete trace from ngrep(make call, speak for 5 seconds,
>hangup): UDP port 5060 in all directions
>
>
>U [myIP]:5060 -> [peerIP]:5060
> INVITE sip:011423663900828 at sip.provider.com SIP/2.0..Via: SIP/2.0/UDP
>[myIP]:5060;branch=z9hG4bK4246930c..From: "423663098668" <sip:
> user at sip.provider.com>;tag=as10b2c259..To:
><sip:011423663900828 at sip.provider.com>..Contact: <sip:user@[myIP]>..Call
> -ID: 6dccb2ab1469a1c87fff22d2076fd449@[myIP]..CSeq: 102
>INVITE..User-Agent: Grandstream..Date: Fri, 13 Aug 2004 21:57:57
>GMT..Allow: INVI
> TE, ACK, CANCEL, OPTIONS, BYE, REFER..Content-Type:
>application/sdp..Content-Length: 184....v=0..o=root 3608 3608 IN IP4
>[myIP]..s=sessio
> n..c=IN IP4 [myIP]..t=0 0..m=audio 19430 RTP/AVP 8 0..a=rtpmap:8
>PCMA/8000..a=rtpmap:0 PCMU/8000..a=silenceSupp:off - - - -..
>#
>U [peerIP]:5060 -> [myIP]:5060
> SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP
>[myIP]:5060;branch=z9hG4bK4246930c..From:
><sip:user at sip.provider.com>;tag=as10b2c25
> 9..To: <sip:011423663900828 at sip.provider.com>..CSeq: 102
>INVITE..Call-ID: 6dccb2ab1469a1c87fff22d2076fd449@[myIP]..Contact:
>alfredko
> hl@[myIP]..WWW-Authenticate: Digest
>realm="sip.provider.com",algorithm="MD5",qop="auth",nonce="573FEFAFEF25C
>B48",opaque="901158A266D
> 481F7"..Max-Forwards: 70..Content-Length: 0....
>
>#
>U [myIP]:5060 -> [peerIP]:5060
> ACK sip:011423663900828 at sip.provider.com SIP/2.0..Via: SIP/2.0/UDP
>[myIP]:5060;branch=z9hG4bK4246930c..From: "423663098668" <sip:alf
> redkohl at sip.provider.com>;tag=as10b2c259..To:
><sip:011423663900828 at sip.provider.com>..Contact:
><sip:user@[myIP]>..Call-ID
> : 6dccb2ab1469a1c87fff22d2076fd449@[myIP]..CSeq: 102 ACK..User-Agent:
>Grandstream..Content-Length: 0....
>#
>U [myIP]:5060 -> [peerIP]:5060
> INVITE sip:011423663900828 at sip.provider.com SIP/2.0..Via: SIP/2.0/UDP
>[myIP]:5060;branch=z9hG4bK069df2d9..From: "423663098668" <sip:
> user at sip.provider.com>;tag=as10b2c259..To:
><sip:011423663900828 at sip.provider.com>..Contact: <sip:user@[myIP]>..Call
> -ID: 6dccb2ab1469a1c87fff22d2076fd449@[myIP]..CSeq: 103
>INVITE..User-Agent: Grandstream..Authorization: Digest
>username="user at s1.do
> uglastelecom.com", realm="sip.provider.com", algorithm=MD5,
>uri="user@[myIP]", nonce="573FEFAFEF25CB48", response="00e118ce8d2
> 72181311a762c91ea6cdc", opaque="901158A266D481F7", qop="auth",
>cnonce="7de950c3", nc=00000001..Date: Fri, 13 Aug 2004 21:57:57
>GMT..Allow: INVIT
> E, ACK, CANCEL, OPTIONS, BYE, REFER..Content-Type:
>application/sdp..Content-Length: 184....v=0..o=root 3608 3609 IN IP4
>[myIP]..s=session
> ..c=IN IP4 [myIP]..t=0 0..m=audio 19430 RTP/AVP 8 0..a=rtpmap:8
>PCMA/8000..a=rtpmap:0 PCMU/8000..a=silenceSupp:off - - - -..
>
>#
>U [peerIP]:5060 -> [myIP]:5060
> SIP/2.0 100 trying..Via: SIP/2.0/UDP
>[myIP]:5060;branch=z9hG4bK069df2d9..To:
><sip:011423663900828 at sip.provider.com>..From: <sip:alfr
> edkohl at sip.provider.com>;tag=as10b2c259..CSeq: 103 INVITE..Call-ID:
>6dccb2ab1469a1c87fff22d2076fd449@[myIP]..Contact: user at 62.
> 99.190.238..Max-Forwards: 70..Content-Length: 0....
>
>#
>U [peerIP]:5060 -> [myIP]:5060
> SIP/2.0 180 ringing..Via: SIP/2.0/UDP
>[myIP]:5060;branch=z9hG4bK069df2d9..To:
><sip:011423663900828 at sip.provider.com>..From: <sip:alf
> redkohl at sip.provider.com>;tag=as10b2c259..CSeq: 103 INVITE..Call-ID:
>6dccb2ab1469a1c87fff22d2076fd449@[myIP]..Contact: user at 62
> .99.190.238..Max-Forwards: 70..Content-Length: 0....
>
>#
>U [peerIP]:5060 -> [myIP]:5060
> SIP/2.0 200 OK..Via: SIP/2.0/UDP
>[myIP]:5060;branch=z9hG4bK069df2d9..To:
><sip:011423663900828 at sip.provider.com>..From: <sip:alfredko
> hl at sip.provider.com>;tag=as10b2c259..CSeq: 103 INVITE..Call-ID:
>6dccb2ab1469a1c87fff22d2076fd449@[myIP]..Contact: user at 62.99.1
> 90.238..Content-type: application/sdp..Max-Forwards:
>70..Content-Length: 133....v=0..o=none 0 0 IN IP4 [peerIP]..s=-..c=IN
>IP4 198.31.231.1
> 7..t=0 0..m=audio 18691 RTP/AVP 8..a=rtpmap:8 PCMA/8000..a=ptime:30..
>
>
>
>Thanxxxx
>
>Roland Zagler
>mailto:laureen at laureen.at
>@fog smart partners
>
>
More information about the asterisk-dev
mailing list