Hi Khan, The 401 is for a REGISTER (look at the Cseq header).
anyhow, the lack of an ACK from the caller means the caller didn't received the reply (200 OK). If the caller is behind a nat, be sure you do force_rport() in script (at INVITE time) - this will correctly route back the replies via the NAT. Regards, Bogdan Khan wrote: > Ok, > > I guess I sort of see the problem but dont know how to fix it... i > capture the trafic from the SjPhone UAC which transmit OPTIONS after > 200 OK, it seems like its getting 401 from server on authentication, > wonder why? > > Here is a link http://pastebin.com/m298ec8c6 > please let me know if i am on the right track!!!! > > Thanks for all your time and efforts... > > Khan > > On Thu, Apr 9, 2009 at 11:00 AM, Khan <[email protected]> wrote: > >> On Thu, Apr 9, 2009 at 2:13 AM, Uwe Kastens <[email protected]> wrote: >> >>> Khan, >>> >>> Would it be possible to add a tcpdump/wirshark on the opensips and on >>> the client in the external network? That make it much easier to debug. >>> >> I haven't done this before so, let me try to get the tcpdump for you, >> I will install wireshark today (like i said im rookie) >> I will post the tcpdump today :) >> >> >>> One question: If you use xlite internaly, is the call dropped after >>> 35secs or not? >>> >> No, it only happens outside the network, I believe my NAT traversal >> works fine, for some reasons my voice reaches them but theirs is lost >> somewhere in clouds :) >> >> >>> BR >>> >>> Uwe >>> >>> Khan schrieb: >>> >>>> Uwe, >>>> >>>> I am using xlite within my network which works fine the problem is >>>> outside the network, Xlite sends SUBSCRIBE and SJphone Sends OPTIONS >>>> request... >>>> >>>> An example of debug is as follows, >>>> >>>> >>>> Xlite registered fine the dump during the call process is as follows, >>>> the call last for 35 seconds in which other party could hear me but i >>>> can see a message on my Sjphone "ACK message awaiting" and then it >>>> disconnects with the message "Network failure" please review the >>>> following link... >>>> http://pastebin.com/dca5bbb0 >>>> >>>> Another example is this SJphone which registers fine but after >>>> registration constantly sends the OPTIONS requsts. The link is as >>>> follows: >>>> http://pastebin.com/d3a4fb379 >>>> >>>> My opensips.cfg is at this link: >>>> http://pastebin.com/d6ce3e43d >>>> >>>> Thanks for all your help ... >>>> >>>> >>>> Khan >>>> >>>> >>>> >>>> On Wed, Apr 8, 2009 at 1:46 PM, Uwe Kastens <[email protected]> wrote: >>>> >>>>> Hi Khan, >>>>> >>>>> A easy way to debug this problem is to use a kind of network sniffer on >>>>> your opensips and directly after your UA. Try to debug this issue with a >>>>> softphone like xlite, so you can start your network dump on the client. >>>>> >>>>> BR >>>>> >>>>> Uwe >>>>> >>>>> Khan schrieb: >>>>> >>>>>> Hi everyone, >>>>>> >>>>>> I'm rookie in SIP technology, strugling with several issues. I am >>>>>> having problem with UAC's outside network. I have 3 UAC registered >>>>>> within the network (SJ Phone, Xlite) they are working fine, I can talk >>>>>> within the network but the problem arrise when I use the UAC outside >>>>>> my network. I am seeing two different things from two different UAC's. >>>>>> >>>>>> 1. Xlite on a network behind NAT try to register, it registers >>>>>> successfully after receiving 200 OK it starts senting SUBSCRIBE >>>>>> requests, which results in 483 Erro (set up in my config) and when >>>>>> call is placed on this it gives ACK time out, person on the other side >>>>>> can hear me but i cant hear him. >>>>>> >>>>>> 2. SjPhone is on another network behind NAT, it regiesters fine, and >>>>>> after registration it starts sending OPTIONS request, which I have >>>>>> configured to respond as 200 OK. It continiously keep sending the >>>>>> requst and my config respond to 200 OK. >>>>>> >>>>>> My question is several parts, what am I doing wrong, >>>>>> a) why don't I get ACK after 200 OK, >>>>>> b) how do i handle SUBSCRIBE requests >>>>>> c) how do i handle OPTIONS request >>>>>> >>>>>> The sever is simply being used as SIP server for calls, no IM, Video, >>>>>> or other applications are implemented yet. There are OpenSIPS, MySQL >>>>>> server, and RTPProxy is running on the box. >>>>>> >>>>>> Please respond to my request considering my skills in the SIP as >>>>>> rookie, guide me on how to resolve problem... >>>>>> >>>>>> Thanks, >>>>>> >>>>>> >>>>>> Khan.... >>>>>> >>>>>> Sorry for such a long email, I am frustrated :( >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> [email protected] >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>>> >>>>> -- >>>>> >>>>> kiste lat: 54.322684, lon: 10.13586 >>>>> >>>>> >>> -- >>> >>> kiste lat: 54.322684, lon: 10.13586 >>> >>> > > _______________________________________________ > 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
