Summary: | ASTERISK-04809: [request] Make it possible to retrieve SIP headers on a REFER (SIP transfer) | ||
Reporter: | Brett Nemeroff (brettnem) | Labels: | |
Date Opened: | 2005-08-10 17:11:34 | Date Closed: | 2011-06-07 14:10:29 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_sip/Interoperability |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | I'm trying to redirect a call with a REFER with some custom headers. SIP_HEADER returns these fine on INVITE, but does not on REFERs. Both the INVITE and the REFER hit this macro. ****** ADDITIONAL INFORMATION ****** REFER sip:15125551003@10.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1:5061;branch=z9hG4bK448f.36bbb762.0 To: <sip:15125551003@localhost>;tag=as75d0c411 From: <sip:controller@foo.bar>;tag=112366460222004 CSeq: 2 REFER Call-ID: 112366460222004.fifouacctd Content-Length: 0 User-Agent: OpenSer (0.10.0-dev5 (i386/linux)) Contact: <sip:caller@10.0.0.1:5061> Referred-By: <sip:controller@foo.bar> Refer-To: sip:7135552000@localhost X-CallLeg: Terminate Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: REFER sip:15125551003@10.0.0.1 SIP/2.0 (43) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: Via: SIP/2.0/UDP 10.0.0.1:5061;branch=z9hG4bK448f.36bbb762.0 (65) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: To: <sip:15125551003@localhost>;tag=as75d0c411 (46) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: From: <sip:controller@foo.bar>;tag=112366460222004 (50) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: CSeq: 2 REFER (13) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: Call-ID: 112366460222004.fifouacctd (35) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: Content-Length: 0 (17) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: User-Agent: OpenSer (0.10.0-dev5 (i386/linux)) (46) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: Contact: <sip:caller@10.0.0.1:5061> (40) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: Referred-By: <sip:controller@foo.bar> (37) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: Refer-To: sip:7135552000@localhost (34) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: X-CallLeg: Terminate (20) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: (0) Aug 10 05:03:27 VERBOSE[19347] logger.c: --- (12 headers 0 lines)Aug 10 05:03:27 VERBOSE[19347] logger.c: --- (12 headers 0 lines)--- Aug 10 05:03:27 DEBUG[19347] chan_sip.c: **** Received REFER (9) - Command in SIP REFER Aug 10 05:03:27 DEBUG[19347] chan_sip.c: SIP call transfer received for call 112366460222004.fifouacctd (REFER)! Aug 10 05:03:27 VERBOSE[19347] logger.c: Transfer to 7135552000 in ctd_extensions Aug 10 05:03:27 VERBOSE[19347] logger.c: Transfer from controller in ctd_extensions Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Unsupervised transfer to (Refer-To): 7135552000 Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Transferred by (Referred-by: ) controller Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Transfer Contact Info <sip:caller@10.0.0.1:5061> (REFER_CONTACT) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: 202 Accepted (blind) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Got SIP blind transfer, applying to 'SIP/test-0730' Aug 10 05:03:27 DEBUG[19347] channel.c: Planning to masquerade SIP/test-0730 into the structure of AsyncGoto/SIP/test-0730 Aug 10 05:03:27 DEBUG[19347] channel.c: Done planning to masquerade AsyncGoto/SIP/test-0730 into the structure of SIP/test-0730 Aug 10 05:03:27 DEBUG[19347] channel.c: Actually Masquerading SIP/test-0730(6) into the structure of AsyncGoto/SIP/test-0730(6) Aug 10 05:03:27 DEBUG[19347] channel.c: Got clone lock for masquerade on 'SIP/test-0730' at 0x81792fc Aug 10 05:03:27 DEBUG[19347] channel.c: Set channel SIP/test-0730 to write format ulaw Aug 10 05:03:27 DEBUG[19347] channel.c: Set channel SIP/test-0730 to read format ulaw Aug 10 05:03:27 DEBUG[19347] channel.c: Putting channel SIP/test-0730 in 4/4 formats Aug 10 05:03:27 DEBUG[19347] channel.c: Released clone lock on 'AsyncGoto/SIP/test-0730<ZOMBIE>' Aug 10 05:03:27 DEBUG[19347] channel.c: Done Masquerading SIP/test-0730 (6) Aug 10 05:03:27 DEBUG[19341] devicestate.c: Changing state for AsyncGoto/SIP/test - state 4 (Invalid) Aug 10 05:03:27 VERBOSE[19347] logger.c: Transmitting (no NAT) to 10.0.0.1:5061: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.0.0.1:5061;branch=z9hG4bK448f.36bbb762.0 From: <sip:controller@foo.bar>;tag=112366460222004 To: <sip:15125551003@localhost>;tag=as75d0c411 Call-ID: 112366460222004.fifouacctd CSeq: 2 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: <sip:15125551003@10.0.0.1> Content-Length: 0 X-Asterisk-HangupCause:: Normal Clearing --- Aug 10 05:03:27 VERBOSE[19347] logger.c: set_destination: Parsing <sip:caller@10.0.0.1:5061> for address/port to send to Aug 10 05:03:27 VERBOSE[19347] logger.c: set_destination: set destination to 10.0.0.1, port 5061 Aug 10 05:03:27 VERBOSE[19347] logger.c: Reliably Transmitting (no NAT) to 10.0.0.1:5061: NOTIFY sip:caller@10.0.0.1:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK166dfd7a From: <sip:15125551003@localhost>;tag=as75d0c411 To: <sip:controller@foo.bar>;tag=112366460222004 Contact: <sip:15125551003@10.0.0.1> Call-ID: 112366460222004.fifouacctd CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- Aug 10 05:03:27 VERBOSE[19347] logger.c: set_destination: Parsing <sip:caller@10.0.0.1:5061> for address/port to send to Aug 10 05:03:27 VERBOSE[19347] logger.c: set_destination: set destination to 10.0.0.1, port 5061 Aug 10 05:03:27 VERBOSE[19347] logger.c: Reliably Transmitting (no NAT) to 10.0.0.1:5061: BYE sip:caller@10.0.0.1:5061 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK6b5bc330 From: <sip:15125551003@localhost>;tag=as75d0c411 To: <sip:controller@foo.bar>;tag=112366460222004 Contact: <sip:15125551003@10.0.0.1> Call-ID: 112366460222004.fifouacctd CSeq: 103 BYE User-Agent: Asterisk PBX X-Asterisk-HangupCause:: Normal Clearing Content-Length: 0 --- Aug 10 05:03:27 VERBOSE[19347] logger.c: <-- SIP read from 10.0.0.1:5061: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK166dfd7a From: <sip:15125551003@localhost>;tag=as75d0c411 To: <sip:controller@foo.bar>;tag=112366460222004 Call-ID: 112366460222004.fifouacctd CSeq: 102 NOTIFY Server: OpenSer (0.10.0-dev5 (i386/linux)) Content-Length: 0 Warning: 392 10.0.0.1:5061 "Noisy feedback tells: pid=18431 req_src_ip=10.0.0.1 req_src_port=5060 in_uri=sip:caller@10.0.0.1:5061 out_uri=sip:caller@10.0.0.1:5061 via_cnt==1" Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: SIP/2.0 404 Not Found (21) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK166dfd7a (58) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: From: <sip:15125551003@localhost>;tag=as75d0c411 (48) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: To: <sip:controller@foo.bar>;tag=112366460222004 (48) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: Call-ID: 112366460222004.fifouacctd (35) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: CSeq: 102 NOTIFY (16) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: Server: OpenSer (0.10.0-dev5 (i386/linux)) (42) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: Content-Length: 0 (17) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: Warning: 392 10.0.0.1:5061 "Noisy feedback tells: pid=18431 req_src_ip=10.0.0.1 req_src_port=5060 in_uri=sip:caller@10.0.0.1:5061 out_uri=sip:caller@10.0.0.1:5061 via_cnt==1" (195) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: (0) Aug 10 05:03:27 VERBOSE[19347] logger.c: --- (9 headers 0 lines)Aug 10 05:03:27 VERBOSE[19347] logger.c: --- (9 headers 0 lines)--- Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Stopping retransmission on '112366460222004.fifouacctd' of Request 102: Found Aug 10 05:03:27 VERBOSE[19347] logger.c: Response message NOTIFY arrived Aug 10 05:03:27 VERBOSE[19347] logger.c: <-- SIP read from 10.0.0.1:5061: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK6b5bc330 From: <sip:15125551003@localhost>;tag=as75d0c411 To: <sip:controller@foo.bar>;tag=112366460222004 Call-ID: 112366460222004.fifouacctd CSeq: 103 BYE Server: OpenSer (0.10.0-dev5 (i386/linux)) Content-Length: 0 Warning: 392 10.0.0.1:5061 "Noisy feedback tells: pid=18431 req_src_ip=10.0.0.1 req_src_port=5060 in_uri=sip:caller@10.0.0.1:5061 out_uri=sip:caller@10.0.0.1:5061 via_cnt==1" Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: SIP/2.0 404 Not Found (21) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK6b5bc330 (58) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: From: <sip:15125551003@localhost>;tag=as75d0c411 (48) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: To: <sip:controller@foo.bar>;tag=112366460222004 (48) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: Call-ID: 112366460222004.fifouacctd (35) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: CSeq: 103 BYE (13) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: Server: OpenSer (0.10.0-dev5 (i386/linux)) (42) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: Content-Length: 0 (17) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: Warning: 392 10.0.0.1:5061 "Noisy feedback tells: pid=18431 req_src_ip=10.0.0.1 req_src_port=5060 in_uri=sip:caller@10.0.0.1:5061 out_uri=sip:caller@10.0.0.1:5061 via_cnt==1" (195) Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Header: (0) Aug 10 05:03:27 VERBOSE[19347] logger.c: --- (9 headers 0 lines)Aug 10 05:03:27 VERBOSE[19347] logger.c: --- (9 headers 0 lines)--- Aug 10 05:03:27 DEBUG[19347] chan_sip.c: Stopping retransmission on '112366460222004.fifouacctd' of Request 103: Found Aug 10 05:03:27 VERBOSE[19347] logger.c: Response message BYE arrived Aug 10 05:03:27 DEBUG[22013] channel.c: Didn't get a frame from channel: AsyncGoto/SIP/test-0730<ZOMBIE> Aug 10 05:03:27 DEBUG[22013] channel.c: Bridge stops bridging channels SIP/CTD-7b81 and AsyncGoto/SIP/test-0730<ZOMBIE> Aug 10 05:03:27 DEBUG[22013] channel.c: Hanging up zombie 'AsyncGoto/SIP/test-0730<ZOMBIE>' Aug 10 05:03:27 DEBUG[22013] app_dial.c: Exiting with DIALSTATUS=ANSWER. Aug 10 05:03:27 DEBUG[22013] app_macro.c: Spawn extension (macro-CallToSIPProvider,s,9) exited non-zero on 'SIP/CTD-7b81' in macro 'CallToSIPProvider' Aug 10 05:03:27 DEBUG[22013] pbx.c: Spawn extension (ctd_extensions,15125551003,1) exited non-zero on 'SIP/CTD-7b81' Aug 10 05:03:27 DEBUG[22013] pbx.c: Function result is 'controller' Aug 10 05:03:27 DEBUG[22013] pbx.c: Function result is 'controller' Aug 10 05:03:27 DEBUG[22013] pbx.c: Function result is '15125551003' Aug 10 05:03:27 DEBUG[22013] pbx.c: Function result is 'ctd_extensions' Aug 10 05:03:27 DEBUG[22013] pbx.c: Function result is 'SIP/CTD-7b81' Aug 10 05:03:27 DEBUG[22013] pbx.c: Function result is 'SIP/test-0730' Aug 10 05:03:27 DEBUG[22013] pbx.c: Function result is 'Dial' Aug 10 05:03:27 DEBUG[22013] pbx.c: Function result is 'SIP/test/15125551003||d(1234)' Aug 10 05:03:27 DEBUG[22013] pbx.c: Function result is '2005-08-10 05:03:23' Aug 10 05:03:27 DEBUG[22013] pbx.c: Function result is '2005-08-10 05:03:27' Aug 10 05:03:27 DEBUG[22013] pbx.c: Function result is '2005-08-10 05:03:27' Aug 10 05:03:27 DEBUG[22013] pbx.c: Function result is '4' Aug 10 05:03:27 DEBUG[22013] pbx.c: Function result is '0' Aug 10 05:03:27 DEBUG[22013] pbx.c: Function result is 'ANSWERED' Aug 10 05:03:27 DEBUG[22013] pbx.c: Function result is 'DOCUMENTATION' Aug 10 05:03:27 DEBUG[22013] pbx.c: Function result is '(null)' Aug 10 05:03:27 DEBUG[22013] pbx.c: Function result is '1123664603.39' Aug 10 05:03:27 DEBUG[22013] pbx.c: Function result is '(null)' Aug 10 05:03:27 DEBUG[22013] channel.c: Hanging up channel 'SIP/CTD-7b81' Aug 10 05:03:27 DEBUG[22013] chan_sip.c: Hangup call SIP/CTD-7b81, SIP callid 112366460222004.fifouacctd) Aug 10 05:03:27 DEBUG[22013] chan_sip.c: update_user_counter(CTD) - decrement inUse counter Aug 10 05:03:27 DEBUG[22013] res_monitor.c: monitor executing ( nice -n 19 soxmix "/var/spool/asterisk/monitor/15125551003-20050810-050323-in.gsm" "/var/spool/asterisk/monitor/15125551003-20050810-050323-out.gsm" "/var/spool/asterisk/monitor/15125551003-20050810-050323.gsm" && rm -f "/var/spool/asterisk/monitor/15125551003-20050810-050323-"* ) & Aug 10 05:03:27 NOTICE[19343] res_musiconhold.c: Request to schedule in the past?!?! Aug 10 05:03:27 DEBUG[22022] pbx.c: Launching 'Macro' Aug 10 05:03:27 VERBOSE[22022] logger.c: -- Executing Macro("SIP/test-0730", "CallToSIPProvider|7135552000") in new stack Aug 10 05:03:27 DEBUG[22022] pbx.c: Launching 'NoOp' Aug 10 05:03:27 VERBOSE[22022] logger.c: -- Executing NoOp("SIP/test-0730", "Placing Call to SIPProvider") in new stack Aug 10 05:03:27 DEBUG[22022] pbx.c: Function result is '(null)' Aug 10 05:03:27 DEBUG[22022] pbx.c: Launching 'Set' Aug 10 05:03:27 VERBOSE[22022] logger.c: -- Executing Set("SIP/test-0730", "CallLeg=") in new stack Aug 10 05:03:27 DEBUG[22022] pbx.c: Function result is '(null)' Aug 10 05:03:27 DEBUG[22022] pbx.c: Launching 'Set' Aug 10 05:03:27 VERBOSE[22022] logger.c: -- Executing Set("SIP/test-0730", "RecordFlag=") in new stack Aug 10 05:03:27 DEBUG[22022] pbx.c: Function result is '(null)' Aug 10 05:03:27 DEBUG[22022] pbx.c: Launching 'Set' Aug 10 05:03:27 VERBOSE[22022] logger.c: -- Executing Set("SIP/test-0730", "PBXExten=") in new stack Aug 10 05:03:27 DEBUG[22022] pbx.c: Launching 'NoOp' Aug 10 05:03:27 VERBOSE[22022] logger.c: -- Executing NoOp("SIP/test-0730", "Record Flag is set to ") in new stack Aug 10 05:03:27 DEBUG[22022] pbx.c: Launching 'NoOp' Aug 10 05:03:27 VERBOSE[22022] logger.c: -- Executing NoOp("SIP/test-0730", "Call Leg is ") in new stack Aug 10 05:03:27 DEBUG[22022] pbx.c: Launching 'Macro' Aug 10 05:03:27 VERBOSE[22022] logger.c: -- Executing Macro("SIP/test-0730", "CheckRecord||7135552000") in new stack Aug 10 05:03:27 WARNING[22022] ast_expr.y: ast_yyerror(): syntax error: syntax error; Input: = 1 ^ Aug 10 05:03:27 DEBUG[22022] pbx.c: Expression is '0' Aug 10 05:03:27 DEBUG[22022] pbx.c: Launching 'GotoIf' Aug 10 05:03:27 VERBOSE[22022] logger.c: -- Executing GotoIf("SIP/test-0730", "0?100:200") in new stack Aug 10 05:03:27 VERBOSE[22022] logger.c: -- Goto (macro-CheckRecord,s,200) Aug 10 05:03:27 DEBUG[22022] pbx.c: Launching 'NoOp' Aug 10 05:03:27 VERBOSE[22022] logger.c: -- Executing NoOp("SIP/test-0730", "Recording is not enabled. Skipping") in new stack Aug 10 05:03:27 DEBUG[22022] pbx.c: Launching 'MacroExit' Aug 10 05:03:27 VERBOSE[22022] logger.c: -- Executing MacroExit("SIP/test-0730", "") in new stack Aug 10 05:03:27 DEBUG[22022] pbx.c: Launching 'SetMusicOnHold' Aug 10 05:03:27 VERBOSE[22022] logger.c: -- Executing SetMusicOnHold("SIP/test-0730", "default)") in new stack Aug 10 05:03:27 DEBUG[22022] pbx.c: Launching 'Dial' Aug 10 05:03:27 VERBOSE[22022] logger.c: -- Executing Dial("SIP/test-0730", "SIP/test/7135552000||d()") in new stack Aug 10 05:03:27 DEBUG[22022] app_dial.c: SIMPLE DIAL (NO URL) Aug 10 05:03:27 DEBUG[22028] app_queue.c: Device 'AsyncGoto/SIP/test' changed to state '4' (Invalid) Aug 10 05:03:27 DEBUG[19341] devicestate.c: Changing state for AsyncGoto/SIP/test - state 4 (Invalid) Aug 10 05:03:27 DEBUG[22022] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Aug 10 05:03:27 DEBUG[22022] chan_sip.c: Setting NAT on RTP to 0 Aug 10 05:03:27 VERBOSE[19347] logger.c: Destroying call '112366460222004.fifouacctd' Aug 10 05:03:27 DEBUG[22022] channel.c: Not copying variable STACK-macro-CallToSIPProvider-s-9. Aug 10 05:03:27 DEBUG[22022] channel.c: Not copying variable STACK-macro-CallToSIPProvider-s-8. Aug 10 05:03:27 DEBUG[22022] channel.c: Not copying variable MACRO_PRIORITY. Aug 10 05:03:27 DEBUG[22022] channel.c: Not copying variable MACRO_CONTEXT. Aug 10 05:03:27 DEBUG[22022] channel.c: Not copying variable MACRO_EXTEN. Aug 10 05:03:27 DEBUG[22022] channel.c: Not copying variable ARG1. Aug 10 05:03:27 DEBUG[22022] channel.c: Not copying variable STACK-macro-CheckRecord-s-201. Aug 10 05:03:27 DEBUG[22022] channel.c: Not copying variable STACK-macro-CheckRecord-s-200. Aug 10 05:03:27 DEBUG[22022] channel.c: Not copying variable STACK-macro-CheckRecord-s-1. Aug 10 05:03:27 DEBUG[22022] channel.c: Not copying variable STACK-macro-CallToSIPProvider-s-7. Aug 10 05:03:27 DEBUG[22022] channel.c: Not copying variable STACK-macro-CallToSIPProvider-s-6. Aug 10 05:03:27 DEBUG[22022] channel.c: Not copying variable STACK-macro-CallToSIPProvider-s-5. Aug 10 05:03:27 DEBUG[22022] channel.c: Not copying variable PBXExten. Aug 10 05:03:27 DEBUG[22022] channel.c: Not copying variable STACK-macro-CallToSIPProvider-s-4. Aug 10 05:03:27 DEBUG[22022] channel.c: Not copying variable RecordFlag. Aug 10 05:03:27 DEBUG[22022] channel.c: Not copying variable STACK-macro-CallToSIPProvider-s-3. A Here's my macro.. [macro-CallToSIPProvider] exten => s,1,NoOp(Placing Call to SIPProvider) exten => s,n,Set(CallLeg=${SIP_HEADER(X-CallLeg)}) exten => s,n,Set(RecordFlag=${SIP_HEADER(X-Record-Flag)}) exten => s,n,Set(PBXExten=${SIP_HEADER(X-PBX-Exten)}) exten => s,n,NoOp(Record Flag is set to ${RecordFlag}) exten => s,n,NoOp(Call Leg is ${CallLeg}) exten => s,n,Macro(CheckRecord,${RecordFlag},${MACRO_EXTEN}) exten => s,n,SetMusicOnHold(default)) exten => s,n,Dial(SIP/test/${ARG1},,d(${PBXExten})) exten => s,n,Hangup | ||
Comments: | By: Mark Spencer (markster) 2005-08-11 01:42:21 SIP_HEADER is only designed to get the headers from an invite, not from any other message. By: Brett Nemeroff (brettnem) 2005-08-11 01:57:17 Is there a reason that this cannot also work for REFERs? The specific REFER I am doing causes asterisk to process a call in the dialplan, like any other call. I pass arguments into Asterisk from other Proxies with headers. By: Olle Johansson (oej) 2005-08-11 03:21:22 I understand what you mean, however, it's not a simple task. You might still want to be able to retrieve headers from the original INVITE. By: Brett Nemeroff (brettnem) 2005-08-11 09:11:29 so, does this turn to a feature request? For an Asterisk only system, I can see how this is not necessary. However if we are to interoperate with other SIP devices out there, this just might be necessary. What I don't understand, is that the debug log already shows chan_sip.c parsing the headers. Why can't it be stored into a variable at that point? I'm not much of a coder, so I'm sure I'm overlooking the complexity of it. FWIW. I'm using this for a CTD (Click-To-Dial) application. The Refer sends the CTD originator to their destination. If I were to send my call variables in the original Invite, the REFER becomes a different channel and doesn't inherit the variables from the originate side; which is why I think I need this.. Am I just missing something here? By: Olle Johansson (oej) 2005-08-11 09:36:47 This is a feature request, because we do not currently support getting a header from any other SIP request than INVITE. As I said, it will require a bit of work to accomplish this. By: Michael Jerris (mikej) 2005-08-18 07:06:52 Suspending this request due to no response. I suggest that you try placing a bounty on the wiki, or contracting this work to be completed. |