[Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection,
SIP Timing Issue??
Aaron Clauson
aza at azaclauson.com
Wed Nov 23 20:06:55 MST 2005
Hi,
Thanks for the tip I'll try it out. That would explain some situations where
one of the peeople concerned was mucking around with the codec settings on
the PAP2 and managed to get some calls out.
It's a bit baffling how the Linksys devices will get INVITES through without
G.729 being set across non-satellite links and yet can't get the very same
INVITE through across a satellite link. Fair enough if it was the Linksys
generating the 488 during the INVITE negotiation but how does Asterisk even
know the difference??
Aaron
> -----Original Message-----
> From: Jason p [mailto:voiceoveripguru at gmail.com]
> Sent: 24 November 2005 02:25
> To: aza at azaclauson.com; Asterisk Users Mailing List -
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Satellite, SIP Invite 488 Codec
> Rejection, SIP Timing Issue??
>
> 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:018X
> XX 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 <mailto:sip:xxx at sip.xxx>
> >;tag=831f2cca367c3ddfo1
> To: <sip:018xxx at sip.xxx>;tag=as50c8f92d
> Call-ID: c71dab66-43f06ff3 at 192.168.1.248
> <mailto:c71dab66-43f06ff3 at 192.168.1.248>
> CSeq: 102 ACK
> Max-Forwards: 70
> Proxy-Authorization: Digest
>
> username="xxx",realm="asterisk",nonce="3cb4e5eb",uri="sip:018x
> xx 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 <mailto: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:018x
> xx 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 <mailto:sip:xxx at sip.xxx>
> >;tag=831f2cca367c3ddfo1
> To: <sip:018xxx at sip.xxx>;tag=as17d663fb
> Call-ID: c71dab66-43f06ff3 at 192.168.1.248
> <mailto: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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