[OpenSIPS-Users] load_balance not releasing resources
Schneur Rosenberg
rosenberg11219 at gmail.com
Tue Nov 15 02:42:09 CET 2011
I just did a test, and sure enough the it fixed the dialog problem too.
On Tue, Nov 15, 2011 at 3:31 AM, Schneur Rosenberg
<rosenberg11219 at gmail.com> wrote:
> Yes opensips knows of all the users that asterisk knows, they share
> the same database, I think my problems was a missing record_route(),
> thanks good night
>
> 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.
>>> 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
>>
>>
>
More information about the Users
mailing list