Caller ID for incoming calls displays "MOD_SOFIA" instead of extension number when connected to freeswitch about sipek2 HOT 12 OPEN

Also tested OK with jain-sip communicator and x-lite

Problem doesn't show up with Yate

@roughraf: same setup (and error) here.
I fixed it by patching last line of function on_incoming_call in pjsipDll.cpp:

function: static on_incoming_call(pjsua_acc_id acc_id,
pjsua_call_id call_id,                                                   pjsip_rx_data *rdata)

original line:
if (cb_callincoming != 0) cb_callincoming(call_id, 
call_info.remote_contact.ptr);

patched line:
if (cb_callincoming != 0) cb_callincoming(call_id, call_info.remote_info.ptr);

It works for me, waiting for Sasa confirmation...

@claudio.nicora
Thanks, I've seen your comment only tonight. Do you know if this change is 
packaged in 
a release? Or can you share a precompiled pjsip.dll with this fix?

@roughraf
I don't think any official release contains this fix because when I last update 
from
SVN it was not included yet.
Still waiting for Sasa...

I attached my pjsip.dll, statically linked for x86.
Hope it works for you.

...too bad it doesn't. FreeSWITCH cannot forward any call to the registered 
sipekphone reporting the following error:

2009-08-01 00:08:40 [CRIT] mod_local_stream.c:237 read_stream_thread() Leaking 
stream 
handle! [switch_ivr_originate() ..\..\src\switch_ivr_originate.c:1640]

I've also tried a pjsip dll from here 
http://code.google.com/p/sipeksdk/source/browse/#svn/trunk/pjsipdll/Lib
but it shows similar behaviour

Thanks for having tried - sasa, we need your help!

Ciao,
   Raffaele

@claudio.nicora

maybe SipekSdk.dll is required to be updated as well? Does it make any sense? 
Can you 
try to attach it, too?

@claudio.nicora

do you know what sip variable "remote_info" and "remote_contact" represent? 
maybe I can 
try to patch it in FS dialplan playing with variables

found it! The variable is "sip_contact_user". Assigning it a value from the 
dialplan 
(or in any other way) corrects the problem. 

You can try it from FreeSWITCH command line:

   originate user/1001 1000 
   --> shows an incoming call from mod_sofia
   originate {sip_contact_user=1000}user/1001 1000
   --> shows an incoming call from 1000

This behaviour is probably a bit different from the standard - but in FS or in 
Sipek? 
Easy to correct, anyhow, but I'll point this out in the FS mailing list

@roughraf: great, you sorted it out! I'll try ASAP in my FS.
Please update this thread if get an answer from FS mailing list.

Anyway, this is my patched SipekSdk.dll

Dear all,

I am unable to figure out a workaround for this problem. I tried @roughraf's 
method from freeswitch console, still my application using SIPEK SDK shows 
"mod_sofia" as calling number, and "NULL" as calling name.

BTW I used an instance of IStateMachine as _call, and I am using 
CallManager.getCall(sessiodId) to receive an incoming call.

The pjsip DLL here (attached above) does not work and shows up an error about 
not being accessible or invalid COM component, when I try to add it as 
reference in my VB.NET (VS2008) project. 

Please help me out if some one has found a solution. Anything, even workarounds 
like roughraf suggested about modifying some variables in dialplan or directory 
that are successful would be highly appreciated!