Nice. On Mon, Nov 14, 2011 at 7:29 PM, Schneur Rosenberg <[email protected] > 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 <[email protected]> > 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:[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 > > *--*--*--*--*--* > > -- > > > > _______________________________________________ > > 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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