search for: find_call

...uting Dial("SIP/xlite1-7a03", "H323/120/smallbox") in new stack --- h323_request - data 120/smallbox format 0x4 (ulaw) --- find_peer +++ find_peer +++ h323_request --- h323_call- 120/smallbox +++ h323_call -- Called 120/smallbox --- onNewCallCreated ooh323c_1 --- find_call +++ find_call Outgoing call smallbox(ooh323c_1) - Codec prefs - (gsm|alaw|ulaw) Adding capabilities to call(outgoing, ooh323c_1) Adding gsm capability to call(outgoing, ooh323c_1) Adding g711 alaw capability to call(outgoing, ooh323c_1) Adding g711 ulaw capability to call(out...

...oad_config +++ ooh323_do_reload -- Executing Dial("SIP/yyy-2965", "OOH323/203@xxx") in new stack --- ooh323_request - data 203@xxx format 0x4 (ulaw) --- find_peer +++ find_peer +++ ooh323_request --- ooh323_call- 203@xxx --- onNewCallCreated ooh323c_o_22 --- find_call +++ find_call setting callid number 203 Outgoing call xxx(ooh323c_o_22) - Codec prefs - (gsm|ulaw|g723) Adding capabilities to call(outgoing, ooh323c_o_22) Adding gsm capability to call(outgoing, ooh323c_o_22) Adding g711 ulaw capability to call(outgoing, ooh323c_o_22)...

...ted ooh323c_7 --- ooh323_onReceivedSetup ooh323c_7 --- find_user +++ find_user Adding capabilities to call(incoming, ooh323c_7) --- configure_local_rtp +++ configure_local_rtp +++ ooh323_onReceivedSetup - Determined context default, extension 6384 --- onAlerting ooh323c_7 --- find_call +++ find_call +++ onAlerting ooh323c_7 -- Executing Dial("OOH323/mms mms-fa6a", "SIP/6384|40|owWtT") in new stack -- Called 6384 -- SIP/6384-d9f2 is ringing ----- ooh323_indicate 3 on call ooh323c_7 ++++ ooh323_indicate 3 on ooh323c_7 -- SIP/6384-d9f2 answ...

...ring. If I answer the phone at this point, it attempts to answer, but does not succeed. Looking at the asterisk debug, it appears to destroy the SIP dialog for the call. It also destroys the RTP instance. When I answer, I receive messages... [Oct 1 15:35:34] DEBUG[3129]: chan_sip.c:6206 find_call: That's odd... Got a response on a call we dont know about. Callid 2c47c6e4740289d90a0d1337261fd704 at 192.168.9.241 [Oct 1 15:35:34] DEBUG[3129]: chan_sip.c:21256 handle_request_do: Invalid SIP message - rejected , no callid, len 715 [Oct 1 15:35:35] DEBUG[3129]: chan_sip.c:6206 find_call:...

...Unavailable'. But it cannot be considered to belong to the same dialog because the tags are different, although the call-id is the same. We have used pedantic checking. Could it be considered as a bug? Looking at the code of chan_sip.c (version 1.4.23.1), we have observed that in function 'find_call' line 4667, asterisk is considering the call as FOUND because of this test: !ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED). Commenting out this comparison, the call proceeds correctly. Sure, there is some reason for this checking and we would like to know which is and in what...

...o the main problem is when using combination Asterisk 1.8.x (I have tried the last 1.8.9.0 also), and using lib java peers client, it is fail. This is the Full DEBUG log. I dont know what else to do, googling found no one with the similar problem. [Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:8014 find_call: = Looking for Call ID: hDVA1Kyx-1327766611250 at lucidesktop.lan (Checking From) --From tag grUqFtoE --To-tag [Jan 28 23:03:32] DEBUG[1654]: acl.c:728 ast_ouraddrfor: For destination '192.168.2.159', our source address is '192.168.2.172'. [Jan 28 23:03:32] DEBUG[1654]: chan_sip.c:...

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 &

...dlock for channel '0xa29dd20' [May 9 08:42:42] DEBUG[20530]: pbx.c:1795 pbx_extension_helper: Launching 'Wait' [May 9 08:42:42] DEBUG[18512]: devicestate.c:287 do_state_change: Changing state for SIP/128.91.56.38 - state 2 (In use) [May 9 08:42:42] DEBUG[18518]: chan_sip.c:4393 find_call: = Found Their Call ID: 9CB723DE-FD6111DB-9FBF9C45-B28951D9@128.91.56.38 Their Tag 479EE6C-1A45 Our tag: as33bbca55 [May 9 08:42:42] DEBUG[18518]: chan_sip.c:14725 handle_request: **** Received ACK (6) - Command in SIP ACK [May 9 08:42:42] DEBUG[18518]: chan_sip.c:2107 __sip_ack: Stopping ret...

I am having a problem registering my cisco phones which is exactly like that described in http://lists.digium.com/pipermail/asterisk-users/2011-May/262306.html except that I am on Asterisk 1.8.3.3 and using sip level POS3-07-4-00 The symptoms are: o 7960 lines show [X] o Outbound calls can be made from the phone, including call pickup of inbound calls, but not to it. o Trace shows REGISTER

...er, when a call comes in from the phone company over a T1 with em_w trunks, the phone on the Sipura will ring but I can not answer it. With Sip debug set I will see a bunch of messages but the following messages seems important: 7 headers, 18 lines Jun 14 11:35:08 WARNING[4101]: chan_sip.c:2012 find_call: Call missing call ID from '192.168.50.119' (192.168.50.119 is the IP of my Sipura device) Calls coming in from the phone company though em_w trunks work fine when terminated to analog phones (fxo_ks via a channel bank) or to another pbx analog trunks (fxs_ks via a channel bank) or to...

...CK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64) Jan 25 22:26:09 DEBUG[41042]: chan_sip.c:3318 parse_request: Header 11: Content-Length: 0 (17) Jan 25 22:26:09 DEBUG[41042]: chan_sip.c:3318 parse_request: Header 12: (0) --- (12 headers 0 lines)--- Jan 25 22:26:09 DEBUG[41042]: chan_sip.c:3150 find_call: = Found Their Call ID: e62dffffcd4dffff@194.183.145.211 Their Tag f8e70000a6870000 Our tag: as29b31203 Jan 25 22:26:09 DEBUG[41042]: chan_sip.c:10933 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Jan 25 22:26:09 DEBUG[41042]: chan_sip.c:10945 handle_request: Ignoring SIP mes...

Hello, How to turn off native bridge in Asterisk. Is it possible ?? Bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031023/91dbe38d/attachment.htm

...risk-users/2005-February/ 087929.html I'm pretty much running the script unmodified, just with my info plugged in. The trouble is I can't seem to get the boxes on the remote ends to accept the NOTIFY messages. They just spit back this: Mar 25 16:51:39 WARNING[5648]: chan_sip.c:2416 find_call: Call missing call ID from 'xxx.xxx.xxx.xxx' I have Call ID specified in the SIP messages I'm sending like this: Call-ID: 1234567890@!IP! (this is unmodified from the script) I've tried all sorts of combinations here, but I haven't come up with one that my * servers w...

...1 07:51:11 DEBUG[24601]: chan_sip.c:3322 parse_request: Header 10: Authorization: Digest realm="asterisk",nonce="6a62f137" (55) Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:3322 parse_request: Header 11: (0) --- (11 headers 0 lines)--- Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:3154 find_call: = Found Their Call ID: 3deb0da4bada8c3@192.168.1.2 Their Tag 632743770590650000 Our tag: as738d9ccd Feb 1 07:51:11 DEBUG[24601]: chan_sip.c:10945 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to yyy.yyy.yyy.yyy : 5060 (...

Hi, I'm testing Asterisk with a new provider. On calls to US toll-free numbers, there is no audio (calls to normal numbers are ok). In response to a valid INVITE from Asterisk, something like this is received: SIP/2.0 183 Session Progress v:SIP/2.0/UDP [my public IP]:5060;branch=z9hG4bK62d91cea CSeq:103 INVITE i:7a1791cf52d6f3dc2d12b208051d0a21@[provider].com f:"Test User"

...[11050]: chan_sip.c:3442 parse_request: Header 6: CSeq: 101 ACK (13) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 7: Content-Length: 0 (17) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse_request: Header 8: (0) 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3239 find_call: = Found Their Call ID: 0013c367-7fdf000e-454e4cdb-2b294b26@10.0.10.136 Their Tag 0013c3677fdf00ae6752cb07-7fbc304d Our tag: as1ae4df20 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:11320 handle_request: **** Received ACK (6) - Command in SIP ACK 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:1403...

...ul 21 15:28:21] DEBUG[2028] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 21 15:28:21] DEBUG[2028] chan_sip.c: SIP message could not be handled, bad request: ZjFhZjZlNmZmZjM3OWFlYzE0MGYzZDYwYzJmODAwNDg. Console output: *CLI> [Jul 21 15:40:47] DEBUG[2105]: chan_sip.c:4562 find_call: = Found Their Call ID: 1832465624 at 192.168.0.25 Their Tag Our tag: as60d9fbbb [Jul 21 15:40:47] DEBUG[2105]: chan_sip.c:15154 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 21 15:40:47] NOTICE[2105]: chan_sip.c:15049 handle_request_register: Registration from '&qu...