API tools faq
paste
Guest User

asterisk-1.6.2.10-called-rpid.patch

a guest
Jun 20th, 2011
235
0
Never
Add comment
Not a member of Pastebin yet? Sign Up, it unlocks many cool features!
text 6.62 KB | None | 0 0
raw download clone embed print report
  1. --- asterisk-1.6.2.10/channels/chan_sip.c.orig 2010-07-24 19:24:31.000000000 -0400
  2. +++ asterisk-1.6.2.10/channels/chan_sip.c 2010-07-24 19:24:20.000000000 -0400
  3. @@ -331,6 +331,25 @@
  4. <para>Always returns <literal>0</literal>.</para>
  5. </description>
  6. </application>
  7. + <application name="SIPCalledRPID" language="en_US">
  8. + <synopsis>
  9. + Set the RPID header for Called Party Identification
  10. + </synopsis>
  11. + <syntax>
  12. + <parameter name="Name" required="true" />
  13. + <parameter name="Number" required="true" />
  14. + </syntax>
  15. + <description>
  16. + <para>Sets the Remote-Party-ID header information to be sent to a call
  17. + originator on further progress through this extension. Many SIP
  18. + phones use this header to display the name and number of the party
  19. + which has been called from the telephone.</para>
  20. + <para>This progress may be updated throughout the call progress whenever
  21. + Asterisk sends a new 180 Ringing, 183 Call Progress, or 200 OK response
  22. + to the calling channel.</para>
  23. + <para>Always returns <literal>0</literal>.</para>
  24. + </description>
  25. + </application>
  26. <function name="SIP_HEADER" language="en_US">
  27. <synopsis>
  28. Gets the specified SIP header.
  29. @@ -1552,6 +1571,7 @@
  30. int method; /*!< SIP method that opened this dialog */
  31. AST_DECLARE_STRING_FIELDS(
  32. AST_STRING_FIELD(callid); /*!< Global CallID */
  33. + AST_STRING_FIELD(calledid); /*!< Called Party RPID Header String */
  34. AST_STRING_FIELD(randdata); /*!< Random data */
  35. AST_STRING_FIELD(accountcode); /*!< Account code */
  36. AST_STRING_FIELD(realm); /*!< Authorization realm */
  37. @@ -2279,6 +2299,7 @@
  38. static int transmit_response_using_temp(ast_string_field callid, struct sockaddr_in *sin, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg);
  39. static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req);
  40. static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req);
  41. +static int transmit_response_with_calledrpid(struct sip_pvt *p, char *msg, struct sip_request *req);
  42. static int transmit_response_with_date(struct sip_pvt *p, const char *msg, const struct sip_request *req);
  43. static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable, int oldsdp);
  44. static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported);
  45. @@ -6531,8 +6552,12 @@
  46. p->invitestate = INV_EARLY_MEDIA;
  47. if (!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) ||
  48. (ast_test_flag(&p->flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER)) {
  49. - /* Send 180 ringing if out-of-band seems reasonable */
  50. - transmit_provisional_response(p, "180 Ringing", &p->initreq, 0);
  51. + /* Send 180 ringing if out-of-band seems reasonable */
  52. + if (!ast_strlen_zero(p->calledid)) {
  53. + transmit_response_with_calledrpid(p, "180 Ringing", &p->initreq);
  54. + } else {
  55. + transmit_provisional_response(p, "180 Ringing", &p->initreq, 0);
  56. + }
  57. ast_set_flag(&p->flags[0], SIP_RINGING);
  58. if (ast_test_flag(&p->flags[0], SIP_PROG_INBAND) != SIP_PROG_INBAND_YES)
  59. break;
  60. @@ -9795,6 +9820,15 @@
  61. add_header(req, "Date", tmpdat);
  62. }
  63.  
  64. +/*! \brief transmit_response_with_calledrpid: Transmit response, no retransmits */
  65. +static int transmit_response_with_calledrpid(struct sip_pvt *p, char *msg, struct sip_request *req)
  66. +{
  67. + struct sip_request resp;
  68. + respprep(&resp, p, msg, req);
  69. + add_header(&resp, "Remote-Party-ID", p->calledid);
  70. + return send_response(p, &resp, XMIT_UNRELIABLE, 0);
  71. +}
  72. +
  73. /*! \brief Append date and content length before transmitting response */
  74. static int transmit_response_with_date(struct sip_pvt *p, const char *msg, const struct sip_request *req)
  75. {
  76. @@ -10498,6 +10532,9 @@
  77. return -1;
  78. }
  79. respprep(&resp, p, msg, req);
  80. + if (!ast_strlen_zero(p->calledid)) {
  81. + add_header(&resp, "Remote-Party-ID", p->calledid);
  82. + }
  83. if (p->rtp) {
  84. if (!p->autoframing && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
  85. ast_debug(1, "Setting framing from config on incoming call\n");
  86. @@ -25532,6 +25569,7 @@
  87. static char *app_dtmfmode = "SIPDtmfMode";
  88. static char *app_sipaddheader = "SIPAddHeader";
  89. static char *app_sipremoveheader = "SIPRemoveHeader";
  90. +static char *app_sipcalledrpid = "SIPCalledRPID";
  91.  
  92. /*! \brief Set the DTMFmode for an outbound SIP call (application) */
  93. static int sip_dtmfmode(struct ast_channel *chan, void *data)
  94. @@ -25662,6 +25700,46 @@
  95. return 0;
  96. }
  97.  
  98. +/*! \brief Set the value of the RPID header for called party identification */
  99. +static int sip_calledrpid(struct ast_channel *chan, void *data)
  100. +{
  101. + struct sip_pvt *p;
  102. + AST_DECLARE_APP_ARGS(args,
  103. + AST_APP_ARG(name);
  104. + AST_APP_ARG(number);
  105. + );
  106. + const char *fromdomain;
  107. + char rpid[256];
  108. +
  109. + if (ast_strlen_zero(data)) {
  110. + ast_log(LOG_WARNING, "This application requires the arguments: Name, Number\n");
  111. + return 0;
  112. + }
  113. +
  114. + AST_STANDARD_APP_ARGS(args, data);
  115. +
  116. + if (ast_strlen_zero(args.name) || ast_strlen_zero(args.number)) {
  117. + ast_log(LOG_WARNING, "One or more arguments are invalid: Name, Number\n");
  118. + return 0;
  119. + }
  120. +
  121. + ast_channel_lock(chan);
  122. +
  123. + if (chan->tech != &sip_tech) {
  124. + ast_log(LOG_WARNING, "This application is only useful for calls originated from SIP UA's\n");
  125. + ast_channel_unlock(chan);
  126. + return 0;
  127. + }
  128. +
  129. + p = chan->tech_pvt;
  130. + fromdomain = S_OR(p->fromdomain, ast_inet_ntoa(p->ourip.sin_addr));
  131. + sprintf(rpid, "\"%s\" <sip:%s@%s>;party=called;id-type=subscriber;screen=yes;privacy=off", args.name, args.number, fromdomain);
  132. + ast_string_field_set(p, calledid, rpid);
  133. +
  134. + ast_channel_unlock(chan);
  135. + return 0;
  136. +}
  137. +
  138. /*! \brief Transfer call before connect with a 302 redirect
  139. \note Called by the transfer() dialplan application through the sip_transfer()
  140. pbx interface function if the call is in ringing state
  141. @@ -25947,6 +26025,7 @@
  142. ast_register_application_xml(app_dtmfmode, sip_dtmfmode);
  143. ast_register_application_xml(app_sipaddheader, sip_addheader);
  144. ast_register_application_xml(app_sipremoveheader, sip_removeheader);
  145. + ast_register_application_xml(app_sipcalledrpid, sip_calledrpid);
  146.  
  147. /* Register dialplan functions */
  148. ast_custom_function_register(&sip_header_function);
  149. @@ -26010,6 +26089,7 @@
  150. ast_unregister_application(app_dtmfmode);
  151. ast_unregister_application(app_sipaddheader);
  152. ast_unregister_application(app_sipremoveheader);
  153. + ast_unregister_application(app_sipcalledrpid);
  154.  
  155. /* Unregister CLI commands */
  156. ast_cli_unregister_multiple(cli_sip, ARRAY_LEN(cli_sip));
Advertisement
Add Comment
Please, Sign In to add comment
Public Pastes
We use cookies for various purposes including analytics. By continuing to use Pastebin, you agree to our use of cookies as described in the Cookies Policy.  OK, I Understand
Not a member of Pastebin yet?
Sign Up, it unlocks many cool features!