Indeed. 200 OK has no contact as well. So you now have a double whammy.
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 176.34.136.92 <http://176.34.136.92>:
> 5080;branch=z9hG4bK158560057fd8d521ed786e9511dbfcf0343836
> To: <sip:[email protected]
<mailto:sip%[email protected]>>;tag=rpz2agt27fobzvvd.o
> From: <sip:[email protected]
<mailto:sip%[email protected]>>;tag=55886164
> Call-ID: [email protected]
<mailto:[email protected]>
> CSeq: 1 INVITE
> Server: Sippy
> Content-Length: 0
On 06/07/2012 06:37 PM, Sven Evensen wrote:
I totally agree! Just that often the answer is "Find another ITSP". In
this case I want to put in some extra effort to resolve the issue. Not
sure yet which side the issue originates. Sometimes it i easier to
make a fix on our side than to get ITSP to change things.
Sven
On Thu, Jun 7, 2012 at 11:31 AM, <[email protected]
<mailto:[email protected]>> wrote:
Sven Evensen <[email protected]
<mailto:[email protected]>> wrote on 07-06-2012 12:24:11:
> I saw that and I believe we made a patch once to add SDP to the
> reInvite. Will have to check that.
>
> But the BYE from sipx says "sipXbridge;cause=205;text=\"Protocol
> Error - 200 OK with no contact\"". I am trying to understand if sipX
> or Sippy is causing this. The original INVITE from Sippy has
> "Anonymous" as contact. Also their ACK has no Contact (which is
> legal they say)
>
> I remember Ranga mentioned a property in sipxBridge about relaxing
> strict protocol. Is that still there? Is it worth trying?
>
> I really want to try to do something rather than saying the SIP
> trunk is not compliant. Because they are not just a ITSP, but also a
> potential customer.
Even ITSP's that are potential customers should at least be made
aware that they don't adhere to standards.
Always try to improve instead of engineer around issues. If it
can't be fixed then that's another thing.
Paul Scheepens
>
> Regards,
> Sven
> On Thu, Jun 7, 2012 at 11:01 AM, Joegen Baclor
<[email protected] <mailto:[email protected]>> wrote:
> sipX bridge disconnects because when it sent an INVITE without SDP
> to Sippy B2BUA, that entity also responded with a 200 OK without
> SDP. That is uncompliance in Sippy part.
>
>
>
> INVITE sip:194.6.238.85:5061 <http://194.6.238.85:5061> SIP/2.0
> CSeq: 1 INVITE
> To: <sip:[email protected]
<mailto:sip%[email protected]>>;tag=rpz2agt27fobzvvd.o
> From: <sip:[email protected]
<mailto:sip%[email protected]>>;tag=55886164
> Call-ID: [email protected]
<mailto:[email protected]>
> Max-Forwards: 70
> Route: <sip:194.6.238.85;lr>
> User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
> P-Asserted-Identity: <sip:[email protected]
<mailto:sip%[email protected]>>
> References: [email protected].
<mailto:[email protected].>
> 101-0;rel=chain;sipxecs-tag=request-refer-z9hg4bk-xx-4ab4jjiscj
> `j9ig27d8fw74puw
> Allow: INVITE,BYE,ACK,CANCEL,OPTIONS
> Supported: timer
> Via: SIP/2.0/UDP 176.34.136.92 <http://176.34.136.92>:
> 5080;branch=z9hG4bK158560057fd8d521ed786e9511dbfcf0343836
> Contact: <sip:176.34.136.92:5080 <http://176.34.136.92:5080>>
> Accept: application/sdp
> Content-Length: 0
>
>
-------------------------------------------------------------------------------------------------
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 176.34.136.92 <http://176.34.136.92>:
> 5080;branch=z9hG4bK158560057fd8d521ed786e9511dbfcf0343836
> To: <sip:[email protected]
<mailto:sip%[email protected]>>;tag=rpz2agt27fobzvvd.o
> From: <sip:[email protected]
<mailto:sip%[email protected]>>;tag=55886164
> Call-ID: [email protected]
<mailto:[email protected]>
> CSeq: 1 INVITE
> Server: Sippy
> Content-Length: 0
>
>
>
> On 06/07/2012 03:47 PM, Sven Evensen wrote:
> Anyone?
> On Tue, Jun 5, 2012 at 1:05 PM, Sven Evensen
<[email protected] <mailto:[email protected]>> wrote:
> I have a new installation with a new SIP trunk provider. It seems
> everything is working except transfer. sipx 4.4.
>
> Incoming external come through SIP trunk to attendant
> Hit 1 to transfer to conference bridge
> When transfer starts, call is dropped.
>
> This works find when a-party is internal.
>
> The problem might be in the INVITE coming from the REFER :
> "SIP","Request: INVITE sip:194.6.238.85:5061
<http://194.6.238.85:5061>, in-dialog"
> causing them to answer with a 200 OK with no Contact header which
> sipX does not like.
>
> Can anyone see from the attached pcap why this is happening
> --
> Sven Evensen, Operations Consultant
> OnRelay
> Elizabeth House ? 39 York Road, London SE1 7NQ, UK ? +44 (0) 207
902 8123 <tel:%2B44%20%280%29%20207%20902%208123> ?
> mailto:[email protected]? www.onrelay.com
<http://www.onrelay.com>
>
> This electronic message transmission contains information from
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> --
> Sven Evensen, Operations Consultant
> OnRelay
> Elizabeth House ? 39 York Road, London SE1 7NQ, UK ? +44 (0) 207
902 8123 <tel:%2B44%20%280%29%20207%20902%208123> ?
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> --
> Sven Evensen, Operations Consultant
> OnRelay
> Elizabeth House ? 39 York Road, London SE1 7NQ, UK ? +44 (0) 207
902 8123 <tel:%2B44%20%280%29%20207%20902%208123> ?
> mailto:[email protected]? www.onrelay.com
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> This electronic message transmission contains information from
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--
*Sven Evensen, Operations Consultant*
*OnRelay*
Elizabeth House ? 39 York Road, London SE1 7NQ, UK ? +44 (0) 207 902
8123 ? mailto:[email protected] ? www.onrelay.com
<http://www.onrelay.com/>
This electronic message transmission contains information from
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disclosure, copying, distribution or use of the contents of this
information or any attachment, is prohibited. If you have received
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delete this message, along with any attachments, from your computer.
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