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

Mike Tesliuk mike at ultra.net.br
Mon Jan 6 16:59:46 CET 2014


ok, i try to use this and i got diferent results now

                topology_hiding("U");


i didnt see the dialog errors anymore , i got a problem with the bye now
but i will made some changes on the network to check about this.

i will post the result later




2014/1/6 Mike Tesliuk <mike at ultra.net.br>

> 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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