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res_pjsip: Backport missing commits for user_eq_phone
This backports the following from trunk, which were missed: r427257 | file | 2014-11-04 16:31:16 -0600 (Tue, 04 Nov 2014) | 2 lines res_pjsip: Allow + at the beginning of a phone number when user_eq_phone is enabled. r427259 | file | 2014-11-04 16:51:32 -0600 (Tue, 04 Nov 2014) | 2 lines res_pjsip: Apply the 'user_eq_phone' setting to the To header as well. It also adds the Alembic script for the option. ASTERISK-24643 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -2155,6 +2155,10 @@ void ast_sip_add_usereqphone(const struct ast_sip_endpoint *endpoint, pj_pool_t
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return;
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}
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if (pj_strbuf(&sip_uri->user)[0] == '+') {
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i = 1;
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}
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/* Test URI user against allowed characters in AST_DIGIT_ANY */
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for (; i < pj_strlen(&sip_uri->user); i++) {
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if (!strchr(AST_DIGIT_ANYNUM, pj_strbuf(&sip_uri->user)[i])) {
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@@ -2221,6 +2225,7 @@ pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint,
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/* Add the user=phone parameter if applicable */
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ast_sip_add_usereqphone(endpoint, dlg->pool, dlg->target);
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ast_sip_add_usereqphone(endpoint, dlg->pool, dlg->remote.info->uri);
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/* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
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dlg->sess_count++;
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