[OpenSIPS-Users] load_balance not releasing resources
Schneur Rosenberg
rosenberg11219 at gmail.com
Tue Nov 15 02:16:02 CET 2011
Here is the code sending the 404 not here, I dont understand why
if(loose_route()) does not return true, is this the way its supposed
to be?
if (has_totag())
{
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route())
{
if (is_method("BYE"))
{
setflag(1); # do accounting ...
setflag(3); # ... even if the transaction fails
}
else if (is_method("INVITE"))
{
record_route();
}
route(1);
}
else
{
if ( is_method("ACK") )
{
if ( t_check_trans() )
{
t_relay();
exit;
}
else
{
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
On Tue, Nov 15, 2011 at 3:00 AM, Schneur Rosenberg
<rosenberg11219 at gmail.com> wrote:
> I see asterisk is sending the BYE to the phone, but opensips sends a
> not here, bellow is the sip strace
>
> U 93.172.0.116:1047 -> opensipsip:5060INVITE
> sip:1917398XXXX at opensipsip SIP/2.0.Via: SIP/2.0/UDP
> 192.168.1.8:5060;branch=z9hG4bK-b5ec4068.From:
> <sip:solhome3 at opensipsip>;tag=9c059eac8018b3c8o0.To:
> <sip:19173985000 at opensipsip>.Remote-Party-ID:
> <sip:solhome3 at opensipsip>;screen=yes;party=calling.Call-ID:
> 82537c-a80f0538 at 192.168.1.8.CSeq: 101 INVITE.Max-Forwards: 70.Contact:
> <sip:solhome3 at 192.168.1.8:5060>.Expires: 240.User-Agent:
> Linksys/SPA2102-5.2.12.Content-Length: 444.Allow: ACK, BYE, CANCEL,
> INFO, INVITE, NOTIFY, OPTIONS, REFER.Supported: x-sipura,
> replaces.Content-Type: application/sdp.
>
>
> U opensipsip:5060 -> 93.172.0.116:1047
> SIP/2.0 407 Proxy Authentication Required.
> Via: SIP/2.0/UDP
> 192.168.1.8:5060;branch=z9hG4bK-b5ec4068;rport=1047;received=93.172.0.116.
> From: <sip:solhome3 at opensipsip>;tag=9c059eac8018b3c8o0.
> To: <sip:1917398XXXX at sopensipsip>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.ef95.
> Call-ID: 82537c-a80f0538 at 192.168.1.8.
> CSeq: 101 INVITE.
> Proxy-Authenticate: Digest realm="opensipsip",
> nonce="4ec215f200000583a544fd65eef0e3a85cb377369e5b8eee".
> Server: OpenSIPS (1.6.4-2-notls (x86_64/linux)).
> Content-Length: 0.
>
>
> U 93.172.0.116:1047 -> opensipsIP:5060
> INVITE sip:1917398XXXX at opensipsIP SIP/2.0.
> Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK-ec946528.
> From: <sip:solhome3 at opensipsIP>;tag=9c059eac8018b3c8o0.
> To: <sip:1917398XXXX at opensipsIP>.
> Remote-Party-ID: <sip:solhome3 at opensipsIP>;screen=yes;party=calling.
> Call-ID: 82537c-a80f0538 at 192.168.1.8.
> CSeq: 102 INVITE.
> Max-Forwards: 70.
> Proxy-Authorization: Digest
> username="solhome3",realm="opensipsIP",nonce="4ec215f200000583a544fd65eef0e3a85cb377369e5b8eee",uri="sip:1917398XXXX at opensipsIP",algorithm=MD5,response="db2640507b2e9824235649f51629ceee".
> Contact: <sip:solhome3 at 192.168.1.8:5060>.
> Expires: 240.
> User-Agent: Linksys/SPA2102-5.2.12.
> Content-Length: 444.
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.
> Supported: x-sipura, replaces.
> Content-Type: application/sdp.
>
>
> U opensipsIP:5060 -> 93.172.0.116:1047
> SIP/2.0 100 Giving a try.
> Via: SIP/2.0/UDP
> 192.168.1.8:5060;branch=z9hG4bK-ec946528;rport=1047;received=93.172.0.116.
> From: <sip:solhome3 at opensipsIP>;tag=9c059eac8018b3c8o0.
> To: <sip:1917398xxxx at opensipsIP>.
> Call-ID: 82537c-a80f0538 at 192.168.1.8.
> CSeq: 102 INVITE.
> Server: OpenSIPS (1.6.4-2-notls (x86_64/linux)).
> Content-Length: 0.
>
> U opensipsIP:5060 -> asteriskIP:5060
> INVITE sip:1917398XXXX at opensipsIP SIP/2.0.
> Record-Route: <sip:opensipsIP;lr=on;did=935.e9420777>.
> Via: SIP/2.0/UDP opensipsIP;branch=z9hG4bK9049.19290602.0.
> Via: SIP/2.0/UDP
> 192.168.1.8:5060;rport=1047;received=93.172.0.116;branch=z9hG4bK-ec946528.
> From: <sip:solhome3 at opensipsIP>;tag=9c059eac8018b3c8o0.
> To: <sip:19173985000 at opensipsIP>.
> Remote-Party-ID: <sip:solhome3 at opensipsIP>;screen=yes;party=calling.
> Call-ID: 82537c-a80f0538 at 192.168.1.8.
> CSeq: 102 INVITE.
> Max-Forwards: 69.
> Contact: <sip:solhome3 at 93.172.0.116:1047>.
> Expires: 240.
> User-Agent: Linksys/SPA2102-5.2.12.
> Content-Length: 444.
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER.
> Supported: x-sipura, replaces.
> Content-Type: application/sdp.
>
> U asteriskIP:5060 -> opensipsIP:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDPopensipsIP;branch=z9hG4bK9049.19290602.0;received=opensipsIP;rport=5060.
> Via: SIP/2.0/UDP
> 192.168.1.8:5060;rport=1047;received=93.172.0.116;branch=z9hG4bK-ec946528.
> Record-Route: <sip:opensipsIP;lr=on;did=935.e9420777>.
> From: <sip:solhome3 at opensipsIP>;tag=9c059eac8018b3c8o0.
> To: <sip:1917398xxxx at opensipsIP>.
> Call-ID: 82537c-a80f0538 at 192.168.1.8.
> CSeq: 102 INVITE.
> Server: Asterisk PBX 1.8.7.1.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH.
> Supported: replaces, timer.
> Contact: <sip:19173985000 at 64.69.47.109:5060>.
> Content-Length: 0.
>
> U DIDProviderIP:5060 -> opensipsIP:5060
> INVITE sip:917398xxxx at opensipsIP SIP/2.0.
> Via: SIP/2.0/UDP DIDProviderIP:5060;branch=z9hG4bK0b523109;rport.
> Max-Forwards: 70.
> From: "ROSENBERG S" <sip:9173985xxxx at DIDproviderIP>;tag=as09899a91.
> To: <sip:917398xxxx at opensipsIP>.
> Contact: <sip:917398xxxx at DIDProviderip>.
> Call-ID: 66d0ba94185dba0430f45f195772e31a at DIDProvidorIP.
> CSeq: 102 INVITE.
> User-Agent: Linksys/SPA2100-3.3.6(0911s).
> Remote-Party-ID: "ROSENBERG S"
> <sip:917398xxxx at DIDProviderIP>;privacy=off;screen=no.
> Date: Mon, 14 Nov 2011 23:35:28 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Supported: replaces, timer.
> Content-Type: application/sdp.
> Content-Length: 340.
>
> U opensipsIP:5060 -> asterisk2ip:5060
> INVITE sip:did917398xxxx at opensipsIP SIP/2.0.
> Via: SIP/2.0/UDP opensipsIP;branch=z9hG4bKf77f.f5d40393.0.
> Via: SIP/2.0/UDPDIDProviderIP:5060;received=DIDProviderIP;branch=z9hG4bK0b523109;rport=5060.
> Max-Forwards: 69.
> From: "ROSENBERG S" <sip:917398xxxx at DIDProviderIP>;tag=as09899a91.
> To: <sip:9173985000 at opensipsIP>.
> Contact: <sip:917398xxxx at DIDProviderIP>.
> Call-ID: 66d0ba94185dba0430f45f195772e31a at DIDProviderIP.
> CSeq: 102 INVITE.
> User-Agent: Linksys/SPA2100-3.3.6(0911s).
> Remote-Party-ID: "ROSENBERG S"
> <sip:917398xxxx at DIDProviderIP>;privacy=off;screen=no.
> Date: Mon, 14 Nov 2011 23:35:28 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Supported: replaces, timer.
> Content-Type: application/sdp.
> Content-Length: 340.
> P-hint: Unathenticated from outside ie did.
>
> U asterisk2IP:5060 -> opensipsIP:5060
> SIP/2.0 100 Trying
> Truncated because of length
>
> U asterisk2IP:5060 -> opensipsIP:5060
> INVITE sip:solhome7 at opensipsIP SIP/2.0.
> Via: SIP/2.0/UDP asterisk2IP:5060;branch=z9hG4bK39459435;rport.
> Max-Forwards: 70.
> From: "ROSENBERG S" <sip:917398xxxx at asterisk2IP>;tag=as5ec8d074.
> To: <sip:solhome5 at opensipsIP>.
> Contact: <sip:917398xxxx at asterisk2IP:5060>.
> Call-ID: 73f977bc448143a26b68be5d38de196e at asterisk2IP:5060.
> CSeq: 102 INVITE.
> User-Agent: Asterisk PBX 1.8.7.1.
> Date: Mon, 14 Nov 2011 23:35:19 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH.
> Supported: replaces, timer.
> P-Asserted-Identity: "ROSENBERG S" <sip:917398xxxx at asterisk2IP>.
> Content-Type: application/sdp.
> Content-Length: 282.
>
> RINGING
>
> U 93.172.0.116:5060 -> opensipsIP:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP opensipsIP;branch=z9hG4bKa96f.8afc2a77.0.
> Via: SIP/2.0/UDP
> asterisk2IP:5060;received=asterisk2IP;branch=z9hG4bK727d493c;rport=5060.
> From: "ROSENBERG S" <sip:917398xxxx at asterisk2IP>;tag=as605029e0.
> To: <sip:solhome7 at sopensipsIP>;tag=6A174081-8FE8464C.
> CSeq: 102 INVITE.
> Call-ID: 09fdaad65a393c1751acd56e150d50a9 at asterisk2IP:5060.
> Contact: <sip:solhome7 at 192.168.1.2>.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
> NOTIFY, PRACK, UPDATE, REFER.
> User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.7.0134.
> Accept-Language: en.
> Content-Type: application/sdp.
> Content-Length: 197.
>
> U opensipsIP:5060 -> asterisk2IP:5060
> SIP/2.0 200 OK.
>
> U asterisk2IP:5060 -> opensipsIP:5060
> ACK sip:solhome7 at 93.172.0.116:5060;nat=yes SIP/2.0.
>
> U 93.172.0.116:1047 -> opensipsIP:5060
> BYE sip:1917398xxxx at asteriskIP:5060;nat=yes SIP/2.0.
> Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK-5f187bca.
> From: <sip:solhome3 at opensipsIP>;tag=9c059eac8018b3c8o0.
> To: <sip:1917398xxxx at opensipsIP>;tag=as5852d19d.
> Call-ID: 82537c-a80f0538 at 192.168.1.8.
> CSeq: 103 BYE.
> Max-Forwards: 70.
> Route: <sip:opensipsIP;lr=on;did=935.e9420777>.
> Proxy-Authorization: Digest
> username="solhome3",realm="opensipsIP",nonce="4ec215f200000583a544fd65eef0e3a85cb377369e5b8eee",uri="sip:1917398xxxx at asteriskIP:5060",algorithm=MD5,response="3bc688c27090bca344187bef1a5e4eee".
> User-Agent: Linksys/SPA2102-5.2.12.
> Content-Length: 0.
> .
>
>
> U opensipsIP:5060 -> asteriskIP:5060
> BYE sip:1917398xxxx at asteriskIP:5060 SIP/2.0.
> Via: SIP/2.0/UDP opensipsIP;branch=z9hG4bKa049.76464162.0.
> Via: SIP/2.0/UDP
> 192.168.1.8:5060;rport=1047;received=93.172.0.116;branch=z9hG4bK-5f187bca.
> From: <sip:solhome3 at opensikpsIP>;tag=9c059eac8018b3c8o0.
> To: <sip:1917398xxxx at opensipsIP>;tag=as5852d19d.
> Call-ID: 82537c-a80f0538 at 192.168.1.8.
> CSeq: 103 BYE.
> Max-Forwards: 69.
> Proxy-Authorization: Digest
> username="solhome3",realm="opensipsIP",nonce="4ec215f200000583a544fd65eef0e3a85cb377369e5b8eee",uri="sip:1917398xxxx at asteriskIP:5060",algorithm=MD5,response="3bc688c27090bca344187bef1a5e49d8".
> User-Agent: Linksys/SPA2102-5.2.12.
> Content-Length: 0.
>
> U asteriskIP:5060 -> opensipsIP:5060
> SIP/2.0 200 OK.
>
> U opensipsIP:5060 -> 93.172.0.116:1047
> SIP/2.0 200 OK.
>
> U asterisk2IP:5060 -> opensipsIP:5060
> BYE sip:solhome7 at 93.172.0.116:5060;nat=yes SIP/2.0.
>
> .
> U opensipsIP:5060 -> asteriskIP:5060
> SIP/2.0 404 Not here.
>
>
>
>
> On Tue, Nov 15, 2011 at 2:19 AM, <duane.larson at gmail.com> wrote:
>> Could you provide a sip trace of a call from INVITE to BYE? Also in your
>> opensips config look and see where you have "404 Not here" configured.
>>
>>
>>
>> On , Schneur Rosenberg <rosenberg11219 at gmail.com> wrote:
>>> In my case this is not relevant, because I'm calling the other phone
>>>
>>> through a DID and the did needs to go to asterisk to decide what to do
>>>
>>> with it, it can send it to a IVR which can later send it to Opensips
>>>
>>> etc. in any case I need to know why asterisk is not sending the BYE to
>>>
>>> the phone, and why opensips sends a not here when the BYE comes from a
>>>
>>> phone not on the system, in that case asterisk sends the BYE to
>>>
>>> opensips which sends a not here instead of sending it to the phone
>>>
>>>
>>>
>>> On Tue, Nov 15, 2011 at 2:06 AM, duane.larson at gmail.com> wrote:
>>>
>>> > If you want VM then you send to Asterks when the call times out (AKA the
>>>
>>> > callee doesn't pick up). We weren't talking about VM here. If you want
>>> > MOH
>>>
>>> > then that is a totally different beast. You would always have to send
>>> > the
>>>
>>> > calls to Asterisk and Asterisk would stay in the flow of the call. From
>>> > what
>>>
>>> > I read above it sounded like the following
>>>
>>> >
>>>
>>> > When I call from one phone on the system to another phone on the
>>>
>>> > same opensips, the phone sends a BYE to opensips which sends it to the
>>>
>>> > asterisk but the BYE never gets sent to the called phone.
>>>
>>> >
>>>
>>> > Sounds like Asterisk is not sending the BYE back to OpenSIPS because its
>>>
>>> > stated " opensips which sends it to the asterisk but the BYE never gets
>>> > sent
>>>
>>> > to the called phone."
>>>
>>> >
>>>
>>> >
>>>
>>> >
>>>
>>> >
>>>
>>> > On , Nick Khamis symack at gmail.com> wrote:
>>>
>>> >> On Mon, Nov 14, 2011 at 6:50 PM, duane.larson at gmail.com> wrote:
>>>
>>> >>
>>>
>>> >> > If two phones are registered with OpenSIPS and they call each other
>>> >> > why
>>>
>>> >>
>>>
>>> >> > would you send the SIP messages to Asterisk?
>>>
>>> >>
>>>
>>> >>
>>>
>>> >>
>>>
>>> >> Because "http://www.opensips.org/Resources/DocsTutAsterisk" said so! ;)
>>>
>>> >>
>>>
>>> >>
>>>
>>> >>
>>>
>>> >> > You need to set up route logic so that if two local users call each
>>>
>>> >> > other then
>>>
>>> >>
>>>
>>> >> > the asterisk boxes are kept out of the equation.
>>>
>>> >>
>>>
>>> >>
>>>
>>> >>
>>>
>>> >> Amazing idea! But what would happen to MOH, and VM?
>>>
>>> >>
>>>
>>> >>
>>>
>>> >>
>>>
>>> >> Nick.
>>>
>>> >>
>>>
>>> >>
>>>
>>> >>
>>>
>>> >> _______________________________________________
>>>
>>> >>
>>>
>>> >> 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
>>>
>>> >
>>>
>>> >
>>>
>>>
>>>
>>> _______________________________________________
>>>
>>> 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
>>
>>
>
More information about the Users
mailing list