[Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection,
SIP Timing Issue??
Jason p
voiceoveripguru at gmail.com
Wed Nov 23 19:24:53 MST 2005
I had the same problem when we were setting up these boxes after katrina.
What i found is that they will only do one G729 session at a time. so that
mesg that your showing is that its trying to register two chans as 729.
what i did to get around this was to turn off fource prefered codec on one
line. This threw me for a loop also but trust me this is the fix, and yes
you can only make one 729 call at a time.
Jason Price
On 11/23/05, Aaron Clauson <aza at azaclauson.com> wrote:
>
> Hi,
>
> I have a very strange Asterisk SIP call signalling problem that is proving
> extremely difficult to track down. The problem is that any SIP INVITE
> request that is coming into Asterisk over a satellite connection from a
> Linksys Router or PAP2 is getting a "Not Acceptable Here (codec error)"
> from
> Asterisk. I've done all the normal checks on the allowed codecs in
> sip.conf
> but to no avail.
>
> I've even gone as far as writing a basic SIP stack to authenticate and
> send
> the INVITE request to Asterisk with exactly the same SDP payload to let me
> brute force different options in the SDP request to try an narrow it down
> that way. The preplexing thing from that length exercise is that if
> exactly
> the same INVITE request comes in from my app across the same satellite
> connection to Asterisk it gets 200 Ok'ed but coming from the Linksys PAP2
> or
> WRT54GP2 it gets 488 Codec Not Acceptable Here'ed.
>
> The first time this happened we went through all the usual checks and got
> nowhere and the person drifted off and it was put down to something
> speicifc
> to that set up/connection. But now it's cropped up again with a different
> person who also just happens to be on a satellite connection but from a
> different provider, although it is possible both providers use the same
> infrastructure. In both cases incoming calls to the Linksys devices worked
> correctly it's just the outgoing calls from the devices to Asterisk that
> are
> getting the rejection. In the second case we can't put it down to
> something
> to do with the connection because the person has a Vonage service working
> no
> problems across the same satellite link we are getting the rejection on.
>
> The SIP trace is below and I'm wondering if anybody has ever seen
> something
> similar. The only thing I can think of is that it's somehow a timing issue
> I
> can't see how it can be a codec issue since the exactly the same SDP
> payload
> will get OK'ed if coming from my app. Is the Asterisk SIP stack sensitive
> to
> the any timings in the INVITE request? It seems highly unlikely but I just
> can't think of anything else.
>
> INVITE sip:018XXX at sip.XXX SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-3b91173f
> From: XXX <sip:XXX at sip.XXX>;tag=831f2cca367c3ddfo1
> To: <sip:018XXX at sip.xxx>
> Call-ID: c71dab66-43f06ff3 at 192.168.1.248
> CSeq: 103 INVITE
> Max-Forwards: 70
> Proxy-Authorization: Digest
> username="XXX",realm="asterisk",nonce="489bfe04",uri="sip:018XXX at sip.XXX
> ",al
> gorithm=MD5,response="22f566e03a225047469d73bec5ab640c"
> Contact: XXX <sip:XXX at 192.168.1.248:5061>
> Expires: 240
> User-Agent: Linksys/PAP2-3.1.3(LS)
> Content-Length: 424
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura
> Content-Type: application/sdp
>
> v=0
> o=- 418210 418210 IN IP4 192.168.1.248
> s=-
> c=IN IP4 192.168.1.248
> t=0 0
> m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 100 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:2 G726-32/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729a/8000
> a=rtpmap:96 G726-40/8000
> a=rtpmap:97 G726-24/8000
> a=rtpmap:98 G726-16/8000
> a=rtpmap:100 NSE/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:30
> a=sendrecv
>
>
> ----------------------------------------------------------------------------
> ----
>
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP
> 192.168.1.248:5061;branch=z9hG4bK-3b91173f;received=xxx;rport=5061
> From: xxx <sip:XXX at sip.xxx>;tag=831f2cca367c3ddfo1
> To: <sip:018xxx at sip.xxx>;tag=as17d663fb
> Call-ID: c71dab66-43f06ff3 at 192.168.1.248
> CSeq: 103 INVITE
> User-Agent: asterisk
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Contact: <sip:018xxx at xxx>
> Proxy-Authenticate: Digest realm="asterisk", nonce="48554be3"
> Content-Length: 0
>
>
>
> ----------------------------------------------------------------------------
> ----
>
> ACK sip:018xxx at sip.xxx SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-c341696b
> From: xxx <sip:xxx at sip.xxx>;tag=831f2cca367c3ddfo1
> To: <sip:018xxx at sip.xxx>;tag=as50c8f92d
> Call-ID: c71dab66-43f06ff3 at 192.168.1.248
> CSeq: 102 ACK
> Max-Forwards: 70
> Proxy-Authorization: Digest
> username="xxx",realm="asterisk",nonce="3cb4e5eb",uri="sip:018xxx at sip.xxx
> ",al
> gorithm=MD5,response="d4438aec627cefa82b6388a3b0c2cb1f"
> Contact: xxx <sip:xxx at 192.168.1.248:5061>
> User-Agent: Linksys/PAP2-3.1.3(LS)
> Content-Length: 0
>
>
>
> ----------------------------------------------------------------------------
> ----
>
> INVITE sip:018xxx at sip.xxx SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.248:5061;branch=z9hG4bK-3b91173f
> From: xxx <sip:xxx at sip.xxx>;tag=831f2cca367c3ddfo1
> To: <sip:018xxx at sip.xxx>
> Call-ID: c71dab66-43f06ff3 at 192.168.1.248
> CSeq: 103 INVITE
> Max-Forwards: 70
> Proxy-Authorization: Digest
> username="xxx",realm="asterisk",nonce="489bfe04",uri="sip:018xxx at sip.xxx
> ",al
> gorithm=MD5,response="22f566e03a225047469d73bec5ab640c"
> Contact: xxx <sip:xxx at 192.168.1.248:5061>
> Expires: 240
> User-Agent: Linksys/PAP2-3.1.3(LS)
> Content-Length: 424
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura
> Content-Type: application/sdp
>
> v=0
> o=- 418210 418210 IN IP4 192.168.1.248
> s=-
> c=IN IP4 192.168.1.248
> t=0 0
> m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 100 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:2 G726-32/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729a/8000
> a=rtpmap:96 G726-40/8000
> a=rtpmap:97 G726-24/8000
> a=rtpmap:98 G726-16/8000
> a=rtpmap:100 NSE/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:30
> a=sendrecv
>
>
> ----------------------------------------------------------------------------
> ----
>
> SIP/2.0 488 Not Acceptable Here (codec error)
> Via: SIP/2.0/UDP
> 192.168.1.248:5061;branch=z9hG4bK-3b91173f;received=xxx;rport=5061
> From: xxx <sip:xxx at sip.xxx>;tag=831f2cca367c3ddfo1
> To: <sip:018xxx at sip.xxx>;tag=as17d663fb
> Call-ID: c71dab66-43f06ff3 at 192.168.1.248
> CSeq: 103 INVITE
> User-Agent: asterisk
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Contact: <sip:018xxx at xxx>
> Content-Length: 0
>
> Thanks,
>
> Aaron
>
>
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