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Doug at NaTel.net
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 Posted: Sun Feb 10, 2008 5:12 pm Post subject: [asterisk-users] Still dropped calls :( |
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At 07:55 2/10/2008, mccoy silva wrote:
| Quote: | 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
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What is your version of Asterisk?
http://lists.digium.com/pipermail/asterisk-users/2007-May/187951.html |
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matt at venturevoip.com
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| Posted: Sun Feb 10, 2008 6:07 pm Post subject: [asterisk-users] Still dropped calls :( |
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Hash: SHA1
Doug wrote:
| Quote: | At 07:55 2/10/2008, mccoy silva wrote:
| Quote: | 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
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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