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Re: Sip invite sent, not reaching dest from certain phones > (Schneur Rosenberg) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 22 Sep 2011 00:03:01 +0200 >From: Vallimamod ABDULLAH <[email protected]> >Subject: Re: [OpenSIPS-Users] Sip invite sent, not reaching dest from > certain phones >To: OpenSIPS users mailling list <[email protected]> >Message-ID: <[email protected]> >Content-Type: text/plain; charset=iso-8859-1 > >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 > > > > >------------------------------ > >Message: 2 >Date: Thu, 22 Sep 2011 01:07:01 +0300 >From: Schneur Rosenberg <[email protected]> >Subject: Re: [OpenSIPS-Users] Sip invite sent, not reaching dest from > certain phones >To: OpenSIPS users mailling list <[email protected]> >Message-ID: > <CANvjR0U66tgJeeQb+tO+AHF9=k_rtmu2gfdovt2t93jjp9u...@mail.gmail.com> >Content-Type: text/plain; charset=ISO-8859-1 > >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 >> > > > >------------------------------ > >Message: 3 >Date: Wed, 21 Sep 2011 17:09:07 -0500 >From: Brett Nemeroff <[email protected]> >Subject: Re: [OpenSIPS-Users] Sip invite sent, not reaching dest from > certain phones >To: OpenSIPS users mailling list <[email protected]> >Message-ID: > <CAPwC5ww8n11k+ucme14KkDWA2m-mODfPjAEG=ys8yosk4kh...@mail.gmail.com> >Content-Type: text/plain; charset="iso-8859-1" > >On Wed, Sep 21, 2011 at 5:03 PM, 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...) >> > >I think asterisk does reply to the contact header and they are obviously >different in the two traces. You'll see one is port 5060 and the other is >based on some NAT translation. Need to find out why those are different.. > >-Brett >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: ><http://lists.opensips.org/pipermail/users/attachments/20110921/b9232d06/attachment-0001.htm> > >------------------------------ > >Message: 4 >Date: Thu, 22 Sep 2011 01:12:42 +0300 >From: Schneur Rosenberg <[email protected]> >Subject: Re: [OpenSIPS-Users] Sip invite sent, not reaching dest from > certain phones >To: OpenSIPS users mailling list <[email protected]> >Message-ID: > <canvjr0xwxur1vrgxbwxwulri6g1zgzb_sxkwu2z5sjaqx4g...@mail.gmail.com> >Content-Type: text/plain; charset=ISO-8859-1 > >If the packet would of reached asterisk then you might of been right, >problem is a ngrep trace does not show a single packet reaching it. > >On Thu, Sep 22, 2011 at 1:09 AM, Brett Nemeroff <[email protected]> wrote: >> On Wed, Sep 21, 2011 at 5:03 PM, 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...) >> >> I think asterisk does reply to the contact header and they are obviously >> different in the two traces. You'll see one is port 5060 and the other is >> based on some NAT translation. Need to find out why those are different.. >> >> -Brett >> >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > >------------------------------ > >Message: 5 >Date: Thu, 22 Sep 2011 00:24:15 +0200 >From: Vallimamod ABDULLAH <[email protected]> >Subject: Re: [OpenSIPS-Users] Sip invite sent, not reaching dest from > certain phones >To: OpenSIPS users mailling list <[email protected]> >Message-ID: <[email protected]> >Content-Type: text/plain; charset=windows-1252 > >Then you have any intermediate device (known or unknown) that does filtering >or mangling in some way? >Try to trace the sip packet on every hop between the 2 servers to see how far >it goes. > >Regards, >- vma >. > >On Sep 22, 2011, at 12:07 AM, Schneur Rosenberg wrote: > >> 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. > > > > >------------------------------ > >Message: 6 >Date: Thu, 22 Sep 2011 01:40:18 +0300 >From: Schneur Rosenberg <[email protected]> >Subject: Re: [OpenSIPS-Users] Sip invite sent, not reaching dest from > certain phones >To: OpenSIPS users mailling list <[email protected]> >Message-ID: > <canvjr0uqcnuqjoyavawdn4ahzvzmaf9pxivz793s7oyarqc...@mail.gmail.com> >Content-Type: text/plain; charset=windows-1252 > >both systems are on the open internet, I have no firewalls etc on any >of the systems, I will try another 2 systems with same configurations >and see what happens. > >On Thu, Sep 22, 2011 at 1:24 AM, Vallimamod ABDULLAH ><[email protected]> wrote: >> Then you have any intermediate device (known or unknown) that does filtering >> or mangling in some way? >> Try to trace the sip packet on every hop between the 2 servers to see how >> far it goes. >> >> Regards, >> - vma >> . >> >> On Sep 22, 2011, at 12:07 AM, Schneur Rosenberg wrote: >> >>> 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. >> >> >> _______________________________________________ >> 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 > > >End of Users Digest, Vol 38, Issue 65 >************************************* _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
