[asterisk-users] Still dropped calls :(

[Matt Riddell]

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Doug wrote:
> At 07:55 2/10/2008, mccoy silva wrote:
>> Hello All!
>>
>> I have a problem with my calls, that drops after 20 - 30 seconds. I 
>> got a piece of PAP2-NA log and Asterisk log and there's an error 603 
>> - call declived, as showed.
>> Thanks for any help.
>>
>> McCoy
> 
> What is your version of Asterisk?
> 
> http://lists.digium.com/pipermail/asterisk-users/2007-May/187951.html

Interestingly this is still 2 in SVN-branch-1.4-r92815M.

svn diff
Index: chan_sip.c
===================================================================
- --- chan_sip.c  (revision 92817)
+++ chan_sip.c  (working copy)
@@ -3020,8 +3020,8 @@
        struct sip_pvt *cur, *prev = NULL;
        struct sip_pkt *cp;

- -       if (sip_debug_test_pvt(p) || option_debug > 2)
- -               ast_verbose("Really destroying SIP dialog '%s' Method:
%s\n", p->callid, sip_methods[p->method].text);
+//     if (sip_debug_test_pvt(p) || option_debug > 2)
+//             ast_verbose("Really destroying SIP dialog '%s' Method:
%s\n", p->callid, sip_methods[p->method].text);

        if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) ||
ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
                update_call_counter(p, DEC_CALL_LIMIT);
@@ -3620,7 +3620,7 @@
                                ast_log(LOG_DEBUG,"T38State change to %d
on channel %s\n", p->t38.state, ast->name);
                        res = transmit_response_with_t38_sdp(p, "200
OK", &p->initreq, XMIT_CRITICAL);
                } else
- -                       res = transmit_response_with_sdp(p, "200 OK",
&p->initreq, XMIT_CRITICAL);
+                       res = transmit_response_with_sdp(p, "200 OK",
&p->initreq, XMIT_RELIABLE);
        }
        ast_mutex_unlock(&p->lock);
        return res;

- --
Kind Regards,

Matt Riddell
Director
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