[OpenSIPS-Users] No record-route on reply's

Mon Jan 6 16:14:11 CET 2014

As i receive the 200 ok without the Record-Route by the gateway, is it
possible to the gateway stablish the signalling directly with the user and
in this case i didnt receive the bye ?

i think that is what happen with this gateway

2014/1/5 Mike Tesliuk <mike at ultra.net.br>

> Hello Razvan (and everybody),
>
> I try this, the dialog seems to be ok because the dialog is beeing
> deleted, but i got this messages on syslog
>
> Jan  5 22:54:55 gcl-ss01a /usr/sbin/opensips[23924]:
> ERROR:rr:get_remote_target: Invalid routing type - 0
> Jan  5 22:54:55 gcl-ss01a /usr/sbin/opensips[23924]:
> ERROR:dialog:dlg_validate_dialog: failed fetching remote target from msg
> Jan  5 22:54:55 gcl-ss01a /usr/sbin/opensips[23924]: In-Dialog BYE from
> XX.XX.XX.XX (callid=1b1c3705-f0ff-1231-c396-001a4bd5a0b4) is not valid
> according to dialog
>
>
>
> in this case, the user send the bye
>
> U __IP__CUSTOMER__:30664 -> __IP__OPENSIPS__:5069
> BYE sip:255755813256 at __IP__OPENSIPS__:5069;transport=udp SIP/2.0.
> Via: SIP/2.0/UDP
> __IP__CUSTOMER__:30664;branch=z9hG4bK-d8754z-2d4c2058767ff73a-1---d8754z-;rport.
> Max-Forwards: 70.
> Contact: <sip:200214 at __IP__CUSTOMER__:30664>.
> To: <sip:255755813256 at __IP__OPENSIPS__:5069>;tag=ZSag78XKZQ6yD.
> From: <sip:200214 at 209.160.78.42:5069>;tag=ea263e5b.
> Call-ID: OGFiYzliMGIzZmYwOTczY2E2YTg1OWYwNzZhMDVlMTA..
> CSeq: 3 BYE.
> Proxy-Authorization: Digest
> username="200214",realm="__IP__OPENSIPS__",nonce="8ef536ce-d39b-4036-90d1-04c54fe9e133",uri="sip:255755813256@
> __IP__OPENSIPS__:5069;transport=udp",response="72ef0971dda591f6b7684b375a5a3ac1",cnonce="8db354c7dc934c13b3577ff5db18297c",nc=00000002,qop=auth,algorithm=MD5.
> User-Agent: Bria Professional release 2.4 stamp 49381.
> Reason: SIP;description="User Hung Up".
> Content-Length: 0.
>
>
> And on opensips this is what i have.
>
> #########
> U __IP__GATEWAY__:39040 -> __IP_OPENSIPS:5079
> BYE sip:255755813256 at __IP_OPENSIPS:5021;transport=udp SIP/2.0.
> Via: SIP/2.0/UDP 10.255.2.21:5021
> ;branch=z9hG4bKervg1296366635;received=10.255.2.21.
> From: <sip:255755813256 at __IP_OPENSIPS:5079>;tag=12ab34cd.
> To: "200214" <sip:200214 at __IP_OPENSIPS>;tag=023883eQv0vHS.
> Call-ID: 1b1c3705-f0ff-1231-c396-001a4bd5a0b4.
> CSeq: 23 BYE.
> Max-Forwards: 70.
> Content-Length: 0.
> .
>
> #
> U __IP_OPENSIPS:5079 -> __IP__GATEWAY__:39040
> SIP/2.0 481 Call Does Not Exist.
> Via: SIP/2.0/UDP 10.255.2.21:5021
> ;branch=z9hG4bKervg1296366635;rport=39040;received=__IP__GATEWAY__.
> From: <sip:255755813256 at __IP_OPENSIPS:5079>;tag=12ab34cd.
> To: "200214" <sip:200214 at __IP_OPENSIPS>;tag=023883eQv0vHS.
> Call-ID: 1b1c3705-f0ff-1231-c396-001a4bd5a0b4.
> CSeq: 23 BYE.
>
> User-Agent: vBilling.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> NOTIFY.
> Supported: precondition, path, replaces.
> Content-Length: 0.
>
>
>
> The call-ID on sip is ok, it is the same of the invite, session progress
> etc..
>
>
> if you guys have any tip I will apreciate, this is a new situation for me,
> happen just with this gateway (I dont remember the brand now , it is a
> friend enviroment and Im trying to help)
>
>
> 2013/12/25 Mike Tesliuk <mike at ultra.net.br>
>
>> Hello Razvan,
>>
>> thank you for your help, i check about this function before, i will try
>> that and i let you know if solve , thank you and happy hollidays
>>
>>
>> 2013/12/24 Răzvan Crainea <razvan at opensips.org>
>>
>>> Hi, Mike!
>>>
>>> Have you tried matching the dialogs using the match_dialog()
>>> function[1]? Also, for sequential requests, you should try using the
>>> fix_route_dialog() function[2].
>>>
>>> [1] http://www.opensips.org/html/docs/modules/1.10.x/dialog.
>>> html#id295144
>>> [2] http://www.opensips.org/html/docs/modules/1.10.x/dialog.
>>> html#id295287
>>>
>>> Best regards,
>>>
>>> Răzvan Crainea
>>> OpenSIPS Core Developer
>>> http://www.opensips-solutions.com
>>>
>>>
>>> On 12/21/2013 06:25 PM, Mike Tesliuk wrote:
>>>
>>>> Hello Guys,
>>>>
>>>>
>>>> Im getting a strange situation here that i dont know how to deal
>>>>
>>>> i have an enviroment where i have freeswitch receiving a call to billing
>>>> and opensips doing the load_balance to the gateways.
>>>>
>>>> When i send the call to the gateway i receive the reply without the
>>>> record-route header, i try to put  an asterisk server as gateway and
>>>> this not happen in this scenario .
>>>>
>>>> Below the invite that i send to the gateway
>>>>
>>>> U 10.1.69.1:5079 <http://10.1.69.1:5079> -> 10.255.2.31:5031
>>>> <http://10.255.2.31:5031>
>>>> INVITE sip:255755813256 at 10.1.69.1:5079
>>>> <http://sip:255755813256@10.1.69.1:5079> SIP/2.0.
>>>>
>>>> Record-Route: <sip:10.1.69.1:5079;lr;ftag=HgcSt10Xa854e;did=9d2.
>>>> 723c6252>.
>>>> Via: SIP/2.0/UDP 10.1.69.1:5079;branch=z9hG4bKe98.72455346.0.
>>>> Via: SIP/2.0/UDP
>>>> 10.1.69.1:5069;received=10.1.69.1;rport=5069;branch=
>>>> z9hG4bKK5N8yU10cgage.
>>>> Max-Forwards: 68.
>>>> From: "200214" <sip:200214 at 10.1.69.1
>>>> <mailto:sip%3A200214 at 10.1.69.1>>;tag=HgcSt10Xa854e.
>>>> To: <sip:255755813256 at 10.1.69.1:5079
>>>> <http://sip:255755813256@10.1.69.1:5079>>.
>>>>
>>>> Call-ID: 4c6591da-e483-1231-6cb4-001a4bd5a0b4.
>>>> CSeq: 53458861 INVITE.
>>>> Contact: <sip:gw+os at 10.1.69.1:5069;transport=udp;gw=os>.
>>>> User-Agent: vBilling.
>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>> REGISTER, NOTIFY.
>>>> Supported: precondition, path, replaces.
>>>> Allow-Events: talk, hold, conference, refer.
>>>> Content-Type: application/sdp.
>>>> Content-Disposition: session.
>>>> Content-Length: 195.
>>>> X-FS-Support: update_display,send_info.
>>>> Remote-Party-ID: "200214" <sip:200214 at 10.1.69.1
>>>> <mailto:sip%3A200214 at 10.1.69.1>>;party=calling;screen=yes;privacy=off.
>>>>
>>>>
>>>>
>>>> and below the 200 ok that i receive
>>>>
>>>> U 10.255.2.31:5031 <http://10.255.2.31:5031> -> 10.1.69.1:5079
>>>> <http://10.1.69.1:5079>
>>>>
>>>> SIP/2.0 200 OK.
>>>> Via: SIP/2.0/UDP
>>>> 10.1.69.1:5079;branch=z9hG4bKe98.72455346.0;received=10.1.69.1.
>>>> Via: SIP/2.0/UDP
>>>> 10.1.69.1:5069;received=10.1.69.1;rport=5069;branch=
>>>> z9hG4bKK5N8yU10cgage.
>>>> To: <sip:255755813256 at 10.1.69.1:5079
>>>> <http://sip:255755813256@10.1.69.1:5079>>;tag=12ab34cd.
>>>> From: "200214" <sip:200214 at 10.1.69.1
>>>> <mailto:sip%3A200214 at 10.1.69.1>>;tag=HgcSt10Xa854e.
>>>>
>>>> CSeq: 53458861 INVITE.
>>>> Call-ID: 4c6591da-e483-1231-6cb4-001a4bd5a0b4.
>>>> Allow: INVITE, BYE, CANCEL, ACK, INFO, REGISTER.
>>>> Supported:.
>>>> Allow-Events: telephone-event.
>>>> Contact: <sip:255755813256 at 10.1.69.1:5031;transport=udp>.
>>>> Content-Type: application/sdp.
>>>> Content-Length: 196.
>>>>
>>>> when i send the call to this gateway the loose route did not execute,
>>>> and i get error's on dialog because the dialog is not matched
>>>>
>>>>
>>>> how should i deal with a situation like this ?
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
>
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