[OpenSIPS-Users] load_balance not releasing resources
Duane Larson
duane.larson at gmail.com
Tue Nov 15 02:30:47 CET 2011
Nice.
On Mon, Nov 14, 2011 at 7:29 PM, Schneur Rosenberg <rosenberg11219 at gmail.com
> wrote:
> I added record_route() before calling route(20) (incoming did route)
> and now the BYE does hit the phone, I think I might of fixed the
> problem, I will test it tomorrow I need to get moving, thanks Duane
> and Khamis.
>
>
>
> On Tue, Nov 15, 2011 at 3:25 AM, Duane Larson <duane.larson at gmail.com>
> wrote:
> > The OpenSIPS setup I usually work with doesn't proxy that much with
> Asterisk
> > doing all the work so take what I say sparingly.
> >
> > 404 Not Here means that OpenSIPS is saying no user account exists. So in
> > your Asterisk BYE the user is
> >
> > U asterisk2IP:5060 -> opensipsIP:5060
> > BYE sip:solhome7 at 93.172.0.116:5060;nat=yes SIP/2.0.
> >
> > Does OpenSIPS know of a user named solhome7 at 93.172.0.116? Since that
> is all
> > that is in the SIP message that is all I have to go by. I also see that
> > there are devices called solhome7, solhome3 and solhome5
> >
> >
> > On Mon, Nov 14, 2011 at 7:00 PM, 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<http://3.1.7.92/>
> .
> >> 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
> >> >
> >> >
> >>
> >> _______________________________________________
> >> Users mailing list
> >> Users at lists.opensips.org
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> >
> > --
> > --
> > *--*--*--*--*--*
> > Duane
> > *--*--*--*--*--*
> > --
> >
> > _______________________________________________
> > 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
>
--
--
*--*--*--*--*--*
Duane
*--*--*--*--*--*
--
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