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:[email protected]:5060;nat=yes SIP/2.0. Does OpenSIPS know of a user named [email protected]? 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 <[email protected] > 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@opensipsip SIP/2.0.Via: SIP/2.0/UDP > 192.168.1.8:5060;branch=z9hG4bK-b5ec4068.From: > <sip:solhome3@opensipsip>;tag=9c059eac8018b3c8o0.To: > <sip:19173985000@opensipsip>.Remote-Party-ID: > <sip:solhome3@opensipsip>;screen=yes;party=calling.Call-ID: > [email protected]: 101 INVITE.Max-Forwards: 70.Contact: > <sip:[email protected]: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@opensipsip>;tag=9c059eac8018b3c8o0. > To: <sip:1917398XXXX@sopensipsip > >;tag=c97b4d1cb1f3d0da549e06a8d482ef63.ef95. > Call-ID: [email protected]. > 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@opensipsIP SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK-ec946528. > From: <sip:solhome3@opensipsIP>;tag=9c059eac8018b3c8o0. > To: <sip:1917398XXXX@opensipsIP>. > Remote-Party-ID: <sip:solhome3@opensipsIP>;screen=yes;party=calling. > Call-ID: [email protected]. > CSeq: 102 INVITE. > Max-Forwards: 70. > Proxy-Authorization: Digest > > username="solhome3",realm="opensipsIP",nonce="4ec215f200000583a544fd65eef0e3a85cb377369e5b8eee",uri="sip:1917398XXXX@opensipsIP > ",algorithm=MD5,response="db2640507b2e9824235649f51629ceee". > Contact: <sip:[email protected]: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@opensipsIP>;tag=9c059eac8018b3c8o0. > To: <sip:1917398xxxx@opensipsIP>. > Call-ID: [email protected]. > CSeq: 102 INVITE. > Server: OpenSIPS (1.6.4-2-notls (x86_64/linux)). > Content-Length: 0. > > U opensipsIP:5060 -> asteriskIP:5060 > INVITE sip:1917398XXXX@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@opensipsIP>;tag=9c059eac8018b3c8o0. > To: <sip:19173985000@opensipsIP>. > Remote-Party-ID: <sip:solhome3@opensipsIP>;screen=yes;party=calling. > Call-ID: [email protected]. > CSeq: 102 INVITE. > Max-Forwards: 69. > Contact: <sip:[email protected]: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@opensipsIP>;tag=9c059eac8018b3c8o0. > To: <sip:1917398xxxx@opensipsIP>. > Call-ID: [email protected]. > 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:[email protected]:5060>. > Content-Length: 0. > > U DIDProviderIP:5060 -> opensipsIP:5060 > INVITE sip:917398xxxx@opensipsIP SIP/2.0. > Via: SIP/2.0/UDP DIDProviderIP:5060;branch=z9hG4bK0b523109;rport. > Max-Forwards: 70. > From: "ROSENBERG S" <sip:9173985xxxx@DIDproviderIP>;tag=as09899a91. > To: <sip:917398xxxx@opensipsIP>. > Contact: <sip:917398xxxx@DIDProviderip>. > Call-ID: 66d0ba94185dba0430f45f195772e31a@DIDProvidorIP. > CSeq: 102 INVITE. > User-Agent: Linksys/SPA2100-3.3.6(0911s). > Remote-Party-ID: "ROSENBERG S" > <sip:917398xxxx@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@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@DIDProviderIP>;tag=as09899a91. > To: <sip:9173985000@opensipsIP>. > Contact: <sip:917398xxxx@DIDProviderIP>. > Call-ID: 66d0ba94185dba0430f45f195772e31a@DIDProviderIP. > CSeq: 102 INVITE. > User-Agent: Linksys/SPA2100-3.3.6(0911s). > Remote-Party-ID: "ROSENBERG S" > <sip:917398xxxx@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@opensipsIP SIP/2.0. > Via: SIP/2.0/UDP asterisk2IP:5060;branch=z9hG4bK39459435;rport. > Max-Forwards: 70. > From: "ROSENBERG S" <sip:917398xxxx@asterisk2IP>;tag=as5ec8d074. > To: <sip:solhome5@opensipsIP>. > Contact: <sip:917398xxxx@asterisk2IP:5060>. > Call-ID: 73f977bc448143a26b68be5d38de196e@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@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@asterisk2IP>;tag=as605029e0. > To: <sip:solhome7@sopensipsIP>;tag=6A174081-8FE8464C. > CSeq: 102 INVITE. > Call-ID: 09fdaad65a393c1751acd56e150d50a9@asterisk2IP:5060. > Contact: <sip:[email protected]>. > 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:[email protected]:5060;nat=yes SIP/2.0. > > U 93.172.0.116:1047 -> opensipsIP:5060 > BYE sip:1917398xxxx@asteriskIP:5060;nat=yes SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK-5f187bca. > From: <sip:solhome3@opensipsIP>;tag=9c059eac8018b3c8o0. > To: <sip:1917398xxxx@opensipsIP>;tag=as5852d19d. > Call-ID: [email protected]. > 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@asteriskIP > :5060",algorithm=MD5,response="3bc688c27090bca344187bef1a5e4eee". > User-Agent: Linksys/SPA2102-5.2.12. > Content-Length: 0. > . > > > U opensipsIP:5060 -> asteriskIP:5060 > BYE sip:1917398xxxx@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@opensikpsIP>;tag=9c059eac8018b3c8o0. > To: <sip:1917398xxxx@opensipsIP>;tag=as5852d19d. > Call-ID: [email protected]. > CSeq: 103 BYE. > Max-Forwards: 69. > Proxy-Authorization: Digest > > username="solhome3",realm="opensipsIP",nonce="4ec215f200000583a544fd65eef0e3a85cb377369e5b8eee",uri="sip:1917398xxxx@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:[email protected]: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, <[email protected]> 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 <[email protected]> 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, [email protected]> 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 [email protected]> wrote: > >> > >> >> On Mon, Nov 14, 2011 at 6:50 PM, [email protected]> 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 > >> > >> >> > >> > >> >> [email protected] > >> > >> >> > >> > >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >> > >> >> > >> > >> >> > >> > >> > _______________________________________________ > >> > >> > Users mailing list > >> > >> > [email protected] > >> > >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >> > >> > > >> > >> > > >> > >> > >> > >> _______________________________________________ > >> > >> Users mailing list > >> > >> [email protected] > >> > >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >> > >> > > _______________________________________________ > > Users mailing list > > [email protected] > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- -- *--*--*--*--*--* Duane *--*--*--*--*--* --
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