[Spce-user] can't answer calls- log shows Double attempt to record-route and Failure route for local call

Miguel Rios miguelrios35 at yahoo.com
Mon Dec 2 10:43:06 EST 2013


Guess not. I guess no one has seen these errors before, and/or I've become a nuisance with my questions.

Well I'm going to give it one last shot before I move on.
I'm going to try to reinstall spce from scratch.
Is there a script to remove the existing install or does everything have to be removed manually? 
Any links or docs explaining what needs to be removed would be much appreciated.

Thanks



On Monday, December 2, 2013 1:29 PM, Miguel Rios <miguelrios35 at yahoo.com> wrote:
 
Anyone have any advice?

Thanks




On Sunday, December 1, 2013 8:12 PM, Miguel Rios <miguelrios35 at yahoo.com> wrote:
 
Hi,

I've been struggling with getting spce to work reliably with me. I've had this strange issue where I can't sometimes answer incoming calls for some time and initially I thought the problem was with the CSipSimple client I'm using on my Android smartphone (it's based on PJSIP).
However, this weekend I recompiled the client with the latest PJSIP and made a bunch of tests, which has allowed me to rule out client issues.
That leaves me with spce...

I'm running spce v3.1 (recently updated) on debian 7.

I also notice that I always get in the kamailio proxy logs the following error messages
ERROR: rr [rr_mod.c:272]: w_record_route(): Double attempt to record-route
and a little later:
NOTICE: <script>: Failure route for local call - R=sip:1000 at mydomain.com:5071 ID=pigEFgptrsh1-xnNksc46zOU1KGybkD.


Debugging on the client side, I see that indeed the client tries to answer the call but there's no response from spce, so the RTP never flows:

18:54:50.663   pjsua_core.c  ..TX 637 bytes Response msg 100/INVITE/cseq=10 (tdta0x5929c860) to TLS 79.xxx.xx.xx:5071:
SIP/2.0 100 Trying
v: SIP/2.0/TLS 79.xxx.xx.xx:5071;received=79.xxx.xx.xx;branch=z9hG4bK7331.877ee9068205a39d27688668eec6804e.0
v: SIP/2.0/UDP 127.0.0.1:5080;rport=5080;branch=z9hG4bK6NHO6aw2
Record-Route: <sip:79.xxx.xx.xx:5071;transport=tls;lr;r2=on;ftag=35D9F3F7-529B85F900060CCA-7E5C0700;ngcplb=yes;socket=sip:79.xxx.xx.xx:5071>
Record-Route: <sip:127.0.0.1;lr;r2=on;ftag=35D9F3F7-529B85F900060CCA-7E5C0700;ngcplb=yes;socket=sip:79.xxx.xx.xx:5071>
i: pigEFgptrsh1-xnNksc46zOU1KGybkD._b2b-1
f: <sip:351967181511 at fortknox.is>;tag=35D9F3F7-529B85F900060CCA-7E5C0700
t: <sip:1000 at fortknox.is> CSeq: 10 INVITE
l:  0
18:54:50.676 stuntp0x592ca4  .STUN mapped address found/changed: 95.xxx.xxx.xx:4027
18:54:50.744        icetp00  .Comp 1: Binding discovery complete, srflx address is 95.xxx.xxx.xx:4027
18:54:50.751 stuntp0x57acd7  .STUN mapped address found/changed: 95.xxx.xxx.xx:4033
18:54:50.815        icetp00  .Comp 2: Binding discovery complete, srflx address is 95.xxx.xxx.xx:4033
18:54:56.536   pjsua_call.c !Answering call 0: code=200
18:54:56.536   pjsua_call.c  .Pending answering call 0 upon completion of media transport
18:55:00.404   pjsua_call.c  Answering call 0: code=200
18:55:00.404   pjsua_call.c  .Pending answering call 0 upon completion of media transport
18:55:03.471   pjsua_call.c  Answering call 0: code=200
18:55:03.471   pjsua_call.c  .Pending answering call 0 upon completion of media transport
18:55:08.356   pjsua_call.c  Answering call 0: code=200
18:55:08.356   pjsua_call.c  .Pending answering call 0 upon completion of media transport
18:55:11.070   pjsua_call.c  Answering call 0: code=200
Any suggestions as to what may be wrong? I remember the first time I installed spce 3.0 a few weeks ago the install never went too smoothly.... Should I scratch this installation and start fresh with 3.1? If so, is there any documentation on removing spce cleanly and starting over?

Thanks,
Miguel
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