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</o:shapelayout></xml><![endif]--></head><body bgcolor=white lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='color:#1F497D'>Thank you Mark for the tips.<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>Is the code below close to what you were thinking?<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>I ran some initial tests and it seems to be working. I can override the default Referred-By value by setting the SIPREFERREDBYHDR variable.<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>static void transfer_refer(struct ast_sip_session *session, const char *target)<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>{<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> pjsip_evsub *sub;<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> enum ast_control_transfer message = AST_TRANSFER_SUCCESS;<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> pj_str_t tmp;<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> pjsip_tx_data *packet;<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> if (pjsip_xfer_create_uac(session->inv_session->dlg, NULL, &sub) != PJ_SUCCESS) {<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> message = AST_TRANSFER_FAILED;<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> return;<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> }<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> if (pjsip_xfer_initiate(sub, pj_cstr(&tmp, target), &packet) != PJ_SUCCESS) {<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> message = AST_TRANSFER_FAILED;<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> pjsip_evsub_terminate(sub, PJ_FALSE);<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> return;<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> }<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> /**** Start of changes ****/<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> pjsip_hdr *hdr;<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> const pj_str_t str_ref_by = { "Referred-By", 11 };<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> const char *ref_by_val = pbx_builtin_getvar_helper(session->channel, "SIPREFERREDBYHDR");<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> pj_str_t tmp2;<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> if (ref_by_val && !ast_strlen_zero(ref_by_val))<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> {<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> hdr = (pjsip_hdr*)pjsip_generic_string_hdr_create(packet->pool, &str_ref_by, pj_cstr(&tmp2, ref_by_val));<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> }<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> else<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> {<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> /* Add Referred-By header */<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> hdr = (pjsip_hdr*)pjsip_generic_string_hdr_create(packet->pool, &str_ref_by, &session->inv_session->dlg->local.info_str);<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> }<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> pjsip_msg_add_hdr(packet->msg, hdr);<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> /**** End of changes ****/<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> pjsip_xfer_send_request(sub, packet);<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'> ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'>}<o:p></o:p></span></p><p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'> asterisk-dev-bounces@lists.digium.com [mailto:asterisk-dev-bounces@lists.digium.com] <b>On Behalf Of </b>Mark Michelson<br><b>Sent:</b> Tuesday, August 25, 2015 1:17 PM<br><b>To:</b> Asterisk Developers Mailing List<br><b>Subject:</b> Re: [asterisk-dev] Transfer cmd (PJSIP not sending Referred-By but chan_sip does)<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>That code gets you about 95% of the way there.<br><br>The biggest difference would be that the pjsip_msg_add_hdr() call may need to be broken up based on whether the SIPREFERREDBYHDR channel variable is set or not. To determine that, you can use the pbx_builtin_getvar_helper() function call declared in include/asterisk/pbx.h to retrieve the variable value, and the ast_strlen_zero() function to determine if the channel variable value is zero-length or not.<br><br>Other than that, you should be able to omit the call to pjsua_process_msg_data() since Asterisk doesn't use pjsua.<br><br>On 08/25/2015 12:35 PM, Dan Cropp wrote:<o:p></o:p></p></div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><p class=MsoPlainText>In doing a little research, it seems the Referred-By header could be added after the pjsip_xfer_initiate.<o:p></o:p></p><p class=MsoPlainText>This is the approach PJSIP did for some code as far back as PJSIP 1.6.<o:p></o:p></p><p class=MsoPlainText> <o:p></o:p></p><p class=MsoPlainText> /*<o:p></o:p></p><p class=MsoPlainText> * Create REFER request.<o:p></o:p></p><p class=MsoPlainText> */<o:p></o:p></p><p class=MsoPlainText> status = pjsip_xfer_initiate(sub, dest, &tdata);<o:p></o:p></p><p class=MsoPlainText> if (status != PJ_SUCCESS) {<o:p></o:p></p><p class=MsoPlainText> pjsua_perror(THIS_FILE, "Unable to create REFER request", status);<o:p></o:p></p><p class=MsoPlainText> pjsip_dlg_dec_lock(dlg);<o:p></o:p></p><p class=MsoPlainText> return status;<o:p></o:p></p><p class=MsoPlainText> }<o:p></o:p></p><p class=MsoPlainText> <o:p></o:p></p><p class=MsoPlainText> /* Add Referred-By header */<o:p></o:p></p><p class=MsoPlainText> gs_hdr = pjsip_generic_string_hdr_create(tdata->pool, &str_ref_by,<o:p></o:p></p><p class=MsoPlainText> &dlg->local.info_str);<o:p></o:p></p><p class=MsoPlainText> pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr*)gs_hdr);<o:p></o:p></p><p class=MsoPlainText> <o:p></o:p></p><p class=MsoPlainText> <o:p></o:p></p><p class=MsoPlainText> /* Add additional headers etc */<o:p></o:p></p><p class=MsoPlainText> pjsua_process_msg_data( tdata, msg_data);<o:p></o:p></p><p class=MsoPlainText> <o:p></o:p></p><p class=MsoPlainText> /* Send. */<o:p></o:p></p><p class=MsoPlainText> status = pjsip_xfer_send_request(sub, tdata);<o:p></o:p></p><p class=MsoPlainText> if (status != PJ_SUCCESS) {<o:p></o:p></p><p class=MsoPlainText> pjsua_perror(THIS_FILE, "Unable to send REFER request", status);<o:p></o:p></p><p class=MsoPlainText> pjsip_dlg_dec_lock(dlg);<o:p></o:p></p><p class=MsoPlainText> return status;<o:p></o:p></p><p class=MsoPlainText> }<o:p></o:p></p><p class=MsoPlainText> <o:p></o:p></p><p class=MsoPlainText>Could anyone provider some insight into how difficult this might be for me to add and submit for approval? Depending on the answer, my manager may be willing to let me work on this.<o:p></o:p></p><p class=MsoPlainText>I've developed in C/C++ for over 25 years so I'm plenty familiar with the language.<o:p></o:p></p><p class=MsoPlainText>I'm less familiar with the syntax and coding standards of Asterisk. I know the group is very good at letting people know about their mistakes and how to fix them.<o:p></o:p></p><p class=MsoPlainText> <o:p></o:p></p><p class=MsoPlainText>Have a great day!<o:p></o:p></p><p class=MsoNormal>Dan<span style='font-size:9.0pt;color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal><span style='font-size:9.0pt;color:#1F497D'> </span><o:p></o:p></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'> <a href="mailto:asterisk-dev-bounces@lists.digium.com">asterisk-dev-bounces@lists.digium.com</a> [<a href="mailto:asterisk-dev-bounces@lists.digium.com">mailto:asterisk-dev-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Dan Cropp<br><b>Sent:</b> Tuesday, August 25, 2015 10:50 AM<br><b>To:</b> Asterisk Developers Mailing List<br><b>Subject:</b> Re: [asterisk-dev] Transfer cmd (PJSIP not sending Referred-By but chan_sip does)</span><o:p></o:p></p></div></div><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal><span style='color:#1F497D'>Thank you Mark</span><o:p></o:p></p><p class=MsoNormal><span style='color:#1F497D'> </span><o:p></o:p></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'> <a href="mailto:asterisk-dev-bounces@lists.digium.com">asterisk-dev-bounces@lists.digium.com</a> [<a href="mailto:asterisk-dev-bounces@lists.digium.com">mailto:asterisk-dev-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Mark Michelson<br><b>Sent:</b> Tuesday, August 25, 2015 10:30 AM<br><b>To:</b> Asterisk Developers Mailing List<br><b>Subject:</b> Re: [asterisk-dev] Transfer cmd (PJSIP not sending Referred-By but chan_sip does)</span><o:p></o:p></p></div></div><p class=MsoNormal> <o:p></o:p></p><div><p class=MsoNormal>The answer to this is actually pretty simple: adding Referred-By in outgoing SIP REFERs is simply not implemented in chan_pjsip's chan_pjsip_transfer() function.<br><br>As far as the syntax required for the Transfer() application, that's probably a case where that needs to be clarified in documentation. There are lots of places in PJSIP configuration where we require full SIP URIs rather than just IP addresses or bare URIs (user@domain).<br><br>On 08/25/2015 10:00 AM, Dan Cropp wrote:<o:p></o:p></p></div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><p class=MsoNormal>I asked the question on asterisk–users but did not receive a response, so I am sending the question here.<o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal>I am running Asterisk 13.5.0.<o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal>A call comes in, Asterisk answers it. After some actions, the call needs to be Transferred (SIP REFER) to another number. The other switch is responsible for accepting the Transfer and tromboning the lines internally. It will also send a BYE so Asterisk no longer has the call.<o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal>The behavior works when I have the endpoint configured at chan_sip. It does not work when the endpoint is configured as PJSIP. I worked with the other switch vendor and he determined chan_sip includes the Referred-By header. PJSIP does not include the Referred-By header. The other switch requires the Referred-By header to be present.<o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal>I tried setting the channel’s SIPREFERREDBYHDR variable before the Transfer command and that still did not force the Referred-By header to be part of the REFER packet.<o:p></o:p></p><p class=MsoNormal>I tried the PJSIP_HEADER add and it still did not add the Referred-By header to the REFER packet.<o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal>Is there a PJSIP setting to force the Referred-By to be part of the REFER packet?<o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal>chan_sip (succeeds)<o:p></o:p></p><p class=MsoNormal>19:27:32.512123 IP (tos 0x0, ttl 64, id 11492, offset 0, flags [none], proto UDP (17), length 630)<o:p></o:p></p><p class=MsoNormal> 192.168.xxx.xxx.sip > 192.168.yyy.yyy.sip: SIP, length: 602<o:p></o:p></p><p class=MsoNormal> REFER <a href="sip:3400@192.168.yyy.yyy:5060">sip:3400@192.168.yyy.yyy:5060</a> SIP/2.0<o:p></o:p></p><p class=MsoNormal> Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;branch=z9hG4bK58f4bd1d<o:p></o:p></p><p class=MsoNormal> Max-Forwards: 70<o:p></o:p></p><p class=MsoNormal> From: <<a href="sip:3344@192.168.xxx.xxx">sip:3344@192.168.xxx.xxx</a>>;tag=as44000cf4<o:p></o:p></p><p class=MsoNormal> To: <<a href="sip:3400@192.168.yyy.yyy">sip:3400@192.168.yyy.yyy</a>>;tag=7Iy0JkwDC<o:p></o:p></p><p class=MsoNormal> Contact: <<a href="sip:3344@192.168.xxx.xxx:5060">sip:3344@192.168.xxx.xxx:5060</a>><o:p></o:p></p><p class=MsoNormal> Call-ID: <a href="mailto:jdEuqpAK-0002-@192.168.yyy.yyy">jdEuqpAK-0002-@192.168.yyy.yyy</a><o:p></o:p></p><p class=MsoNormal> CSeq: 102 REFER<o:p></o:p></p><p class=MsoNormal> User-Agent: Asterisk PBX 13.5.0<o:p></o:p></p><p class=MsoNormal> Date: Thu, 20 Aug 2015 19:27:32 GMT<o:p></o:p></p><p class=MsoNormal> Refer-To: <<a href="sip:370@192.168.yyy.yyy">sip:370@192.168.yyy.yyy</a>><o:p></o:p></p><p class=MsoNormal> Referred-By: <<a href="sip:3344@192.168.xxx.xxx:5060">sip:3344@192.168.xxx.xxx:5060</a>><o:p></o:p></p><p class=MsoNormal> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE<o:p></o:p></p><p class=MsoNormal> Supported: replaces, timer<o:p></o:p></p><p class=MsoNormal> Content-Length: 0<o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal>Pjsip<o:p></o:p></p><p class=MsoNormal>18:46:58.386372 IP (tos 0x0, ttl 64, id 38690, offset 0, flags [DF], proto UDP (17), length 654)<o:p></o:p></p><p class=MsoNormal> 192.168.xxx.xxx.sip > 192.168.yyy.yyy.sip: SIP, length: 626<o:p></o:p></p><p class=MsoNormal> REFER <a href="sip:3400@192.168.yyy.yyy:5060">sip:3400@192.168.yyy.yyy:5060</a> SIP/2.0<o:p></o:p></p><p class=MsoNormal> Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;rport;branch=z9hG4bKPjec41c3b9-d734-482d-82c1-2a6f8d9452a3<o:p></o:p></p><p class=MsoNormal> From: <<a href="sip:3344@192.168.xxx.xxx">sip:3344@192.168.xxx.xxx</a>>;tag=3c10f423-e468-42ea-87a1-658ae106581c<o:p></o:p></p><p class=MsoNormal> To: <<a href="sip:3400@192.168.yyy.yyy">sip:3400@192.168.yyy.yyy</a>>;tag=WITKDakt<o:p></o:p></p><p class=MsoNormal> Contact: <<a href="sip:192.168.xxx.xxx:5060">sip:192.168.xxx.xxx:5060</a>><o:p></o:p></p><p class=MsoNormal> Call-ID: <a href="mailto:s6Wk6l6Q-0001-@192.168.yyy.yyy">s6Wk6l6Q-0001-@192.168.yyy.yyy</a><o:p></o:p></p><p class=MsoNormal> CSeq: 981 REFER<o:p></o:p></p><p class=MsoNormal> Event: refer<o:p></o:p></p><p class=MsoNormal> Expires: 600<o:p></o:p></p><p class=MsoNormal> Supported: 100rel, timer, replaces, norefersub<o:p></o:p></p><p class=MsoNormal> Accept: message/sipfrag;version=2.0<o:p></o:p></p><p class=MsoNormal> Allow-Events: message-summary, presence, dialog, refer<o:p></o:p></p><p class=MsoNormal> Refer-To: <<a href="sip:370@192.168.yyy.yyy">sip:370@192.168.yyy.yyy</a>><o:p></o:p></p><p class=MsoNormal> Max-Forwards: 70<o:p></o:p></p><p class=MsoNormal> User-Agent: Asterisk PBX 13.5.0<o:p></o:p></p><p class=MsoNormal> Content-Length: 0<o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal>One other slight oddity.<o:p></o:p></p><p class=MsoNormal>To get chan_sip to Transfer<o:p></o:p></p><p class=MsoNormal><span style='color:#1F497D'><a href="mailto:370@192.168.yyy.yyy">370@192.168.yyy.yyy</a></span><o:p></o:p></p><p class=MsoNormal><span style='color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal><span style='color:#1F497D'>To get PJSIP to Transfer with the correct Refer-To header, I had to include the <> and sip:</span><o:p></o:p></p><p class=MsoNormal><span style='color:#4F81BD'><<u><a href="sip:370@192.168.yyy.">sip:370@192.168.yyy.</a></u></span><u><span style='color:#1F497D'>yyy</span></u><span style='color:#4F81BD'>></span><o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal>Have a great day!<o:p></o:p></p><p class=MsoNormal>Dan<o:p></o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:12.0pt;font-family:"Times New Roman , serif","serif"'> </span><o:p></o:p></p></blockquote><p class=MsoNormal><span style='font-size:12.0pt;font-family:"Times New Roman , serif","serif"'> </span><o:p></o:p></p><p class=MsoNormal><span style='font-size:12.0pt;font-family:"Times New Roman","serif"'><br><br><o:p></o:p></span></p></blockquote><p class=MsoNormal><span style='font-size:12.0pt;font-family:"Times New Roman","serif"'><o:p> </o:p></span></p></div></body></html>