search for: handle_incoming

...UG[1654]: chan_sip.c:3482 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 192.168.2.172:5060 [Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:7694 sip_alloc: Allocating new SIP dialog for hDVA1Kyx-1327766611250 at lucidesktop.lan - INVITE (No RTP) [Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:24907 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE [Jan 28 23:03:32] DEBUG[1654]: netsock2.c:134 ast_sockaddr_split_hostport: Splitting '192.168.2.159:5062' into... [Jan 28 23:03:32] DEBUG[1654]: netsock2.c:188 ast_sockaddr_split_hostport: ...host '192.168.2.159' and port '5062...

Hi A question, i have upgraded a beta serveur from Asterisk 1.6.1 to 1.6.2 and now all SIP Relatime user are rejeted : [Oct 5 05:39:22] DEBUG[15081]: chan_sip.c:21639 handle_incoming: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 5 05:39:22] DEBUG[15081]: chan_sip.c:21658 handle_incoming: Ignoring SIP message because of retransmit (REGISTER Seqno 44199, ours 44199) [Oct 5 05:39:22] DEBUG[15081]: res_config_mysql.c:1602 mysql_reconnect: MySQL RealTime: Connecti...

I am wanting to perform a regression test on an application but am not certain of the starting point I want to test. Test results for 0.9.53 indicated simply that it worked OK, without details of all aspects of the application, some of which are being reported as broken now. So, I want to confirm that it really did work OK then, as when I test with 0.9.60 through to 1.0rc3 it fails on some

...;register_verify" #10 0x0000000809066110 in handle_request_register (p=0x802817028, req=0x7fffff892a40, sin=0x7fffff892a30, e=0x7fffff892e79 "sip:10.3.8.1") at chan_sip.c:18996 res = 8 __PRETTY_FUNCTION__ = "handle_request_register" #11 0x00000008090670ec in handle_incoming (p=0x802817028, req=0x7fffff892a40, sin=0x7fffff892a30, recount=0x7fffff8929bc, nounlock=0x7fffff8929b8) at chan_sip.c:19210 cmd = 0x7fffff892e70 "REGISTER" cseq = 0x7fffff892eb4 "2 REGISTER" useragent = 0x7fffff892f27 "Ekiga/3.2.0" seqn...

...dlg0x7f5f18019fc8 ...Received Response msg 100/INVITE/cseq=24420 (rdata0x7f5f18052b08) [2017-06-15 07:43:57] DEBUG[25171]: pjproject:0 <?>: dlg0x7f5f18019fc8 ...Transaction tsx0x7f5f18095998 state changed to Proceeding [2017-06-15 07:43:57] DEBUG[25171]: res_pjsip_session.c:2485 handle_incoming: Received response [2017-06-15 07:43:57] DEBUG[25171]: res_pjsip_session.c:2469 handle_incoming_response: Response is 100 Trying [2017-06-15 07:43:57] DEBUG[25171]: res_pjsip/pjsip_distributor.c:785 distribute: rdata clone remove distributed: Response msg 100/INVITE/cseq=24420 (rdata0x7f5f18052b08)...

On Mon, Jun 5, 2017, at 04:26 PM, Michael Maier wrote: > On 06/05/2017 at 06:29 PM, Joshua Colp wrote: > > On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote: > >> > >> Do you have any idea where to start to look at? Adding additional output > >> in the source code? Which functions could be interesting? I may add own > >> debug code to see why things

On 06/14/2017 at 05:53 PM Joshua Colp wrote: > On Wed, Jun 14, 2017, at 12:47 PM, Michael Maier wrote: > > <snip> > >> >> I added this patch to see, if really all packages are are freed after >> they have been processed: >> >> --- b/res/res_pjsip/pjsip_distributor.c 2017-05-30 19:44:16.000000000 >> +0200 >> +++

Hi I have noticed that asterisk returns 'SIP 603' when the called party does not answer. My test setup is simple: two SIP phones (extensions: 100 and 111) registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds. When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request to 111 (expected) and a '603 Decline' response to 100 (unexpected &