[OpenSIPS-Users] load_balance not releasing resources

Schneur Rosenberg rosenberg11219 at gmail.com
Tue Nov 15 02:00:21 CET 2011 

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
>
>

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