The packet does not reach asterisk, I did a ngrep on the asterisk server and not a single packet arrives from the opensips when using the Aastra phone, therefore its not sending back anything, the asterisk CLI is also quiet nothing whatsoever :-(
On Thu, Sep 22, 2011 at 1:03 AM, Vallimamod ABDULLAH <[email protected]> wrote: > Hi Schneur, > > What do you mean precisely by never hitting the asterisk server ? > As your ngrep trace shows, both packets are sent over the wire to the exact > same address (68.233.222.9:5060) so they should both reach Asterisk. But it's > possible that the latter doesn't treat them the same way, depending on nat > issues most of the time (Asterisk send replies to the contact header URI by > default if I recall correctly...) > > Try to make a ngrep trace on the iface attached to 68.233.222.9 and check if > Asterisk does not send the answer to a different IP. Also enable the debug > log on the asterisk console to spot any error / warning messages or sip > retransmissions. > > Hope this would help. > > Regards, > -vma > . > > On Sep 21, 2011, at 11:43 PM, Schneur Rosenberg wrote: > >> NO These are the invites going from the opensips to the asterisk NOT >> the ones from the phone, I did a ngrep on the asterisk box and the >> packet never reaches it, both opensips and asterisk are open no NAT, >> the phones are behind a nat as you can see in the sip packets >> >> >> On Thu, Sep 22, 2011 at 12:37 AM, Duane Larson <[email protected]> >> wrote: >>> These are the INVITES that are coming from your Phones correct? These won't >>> help to troubleshoot I don't think. You will need to show the INVITES that >>> are leaving OpenSIPS and heading towards your Asterisk server. >>> >>> Honestly if your opensips.cfg does the exact same thing for linksys and >>> aastra phones I can't see it being an opensips issue. That's just a guess >>> since I don't have anything to go on. >>> >>> On Wed, Sep 21, 2011 at 4:25 PM, Schneur Rosenberg >>> <[email protected]> wrote: >>>> >>>> I'm pretty new to opensips, I'm having a interesting problem, I use my >>>> opensips for loadbalancing purposes I'm trying to place a call, and >>>> from My linksys phone everything works fine, call comes into opensips >>>> and opensips sends it to my asterisk system and call goes through >>>> properly, from other phone (Aastra) Opensips accept the call, it even >>>> sends it to the Asterisk but in never hits the asterisk server, can >>>> anyone please review the 2 invites and let me know why second invite >>>> gets lost, and how I can fix it >>>> >>>> Here is the invite from the Linksys that worked >>>> >>>> U 64.69.40.120:5060 -> 68.233.222.9:5060 >>>> INVITE sip:[email protected]:5060 SIP/2.0. >>>> Record-Route: <sip:64.69.40.120;lr=on>. >>>> Via: SIP/2.0/UDP 64.69.40.120;branch=z9hG4bK9857.800b0681.0. >>>> Via: SIP/2.0/UDP >>>> >>>> 192.168.1.104:5060;rport=5060;received=173.220.6.65;branch=z9hG4bK-4194567e. >>>> From: solhome5 >>>> <sip:[email protected]>;tag=833ac73613f3482o0. >>>> To: <sip:[email protected]>. >>>> Remote-Party-ID: solhome5 >>>> <sip:[email protected]>;screen=yes;party=calling. >>>> Call-ID: [email protected]. >>>> CSeq: 102 INVITE. >>>> Max-Forwards: 69. >>>> Contact: solhome5 <sip:[email protected]:5060;nat=yes>. >>>> Expires: 240. >>>> User-Agent: Linksys/SPA2102-5.2.12. >>>> Content-Length: 446. >>>> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. >>>> Supported: x-sipura, replaces. >>>> Content-Type: application/sdp. >>>> >>>> Here is the invite of the Aastra that did not work >>>> >>>> U 64.69.40.120:5060 -> 68.233.222.9:5060 >>>> INVITE sip:[email protected]:5060;user=phone SIP/2.0. >>>> Record-Route: <sip:64.69.40.120;lr=on>. >>>> Via: SIP/2.0/UDP 64.69.40.120;branch=z9hG4bK847d.93e9e112.0. >>>> Via: SIP/2.0/UDP >>>> >>>> 192.168.1.102;rport=32857;received=173.220.6.65;branch=z9hG4bK7d9e53a542fc6ebe6.6e57a6889920eccf1. >>>> Max-Forwards: 69. >>>> From: "test2" <sip:[email protected]:5060>;tag=ef646132b8. >>>> To: <sip:[email protected]:5060;user=phone>. >>>> Call-ID: f12b5324f31c0d30. >>>> CSeq: 20777 INVITE. >>>> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, >>>> PRACK, SUBSCRIBE, INFO. >>>> Allow-Events: talk, hold, conference, LocalModeStatus. >>>> Contact: "test2" >>>> >>>> <sip:[email protected]:32857;transport=udp;nat=yes>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D2A6C9E>". >>>> Supported: path, 100rel, replaces. >>>> User-Agent: Aastra 57iCT/3.2.2.56. >>>> Content-Type: application/sdp. >>>> Content-Length: 630. >>>> >>>> _______________________________________________ >>>> 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 > > > _______________________________________________ > 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
