<p> Attention is currently required from: Michael Bradeen. </p>
<p>Patch set 2:<span style="border-radius: 3px; display: inline-block; margin: 0 2px; padding: 4px;background-color: #ffd4d4; color: #000000;">Code-Review -1</span></p><p><a href="https://gerrit.asterisk.org/c/asterisk/+/19611">View Change</a></p><p>1 comment:</p><ul style="list-style: none; padding: 0;"><li style="margin: 0; padding: 0;"><p><a href="null">Patchset:</a></p><ul style="list-style: none; padding: 0;"><li style="margin: 0; padding: 0 0 0 16px;"><p style="margin-bottom: 4px;"><a href="https://gerrit.asterisk.org/c/asterisk/+/19611?tab=comments">Patch Set #2:</a> </p><p style="white-space: pre-wrap; word-wrap: break-word;">A few more things that can be cleaned up, mainly sip.conf related stuff:</p><p style="white-space: pre-wrap; word-wrap: break-word;">res/ari/resource_endpoints.h: /*! Technology of the endpoints (sip,iax2,...) */<br>res/res_phoneprov.c:/*! \li \ref res_phoneprov.c uses the configuration file \ref phoneprov.conf and \ref users.conf and \ref sip.conf<br>res/res_phoneprov.c:#define SIPUSERS_PROVIDER_NAME "sipusers"<br>res/res_phoneprov.c: if ((cfg = ast_config_load("sip.conf", config_flags)) && cfg != CONFIG_STATUS_FILEINVALID) {<br>res/res_phoneprov.c: /* This cleans up the sip.conf/users.conf provider (called specifically for clarity) */<br>res/res_phoneprov.c: /* Register ourselves as the provider for sip.conf/users.conf */</p><p style="white-space: pre-wrap; word-wrap: break-word;">include/asterisk/config_options.h: const char *matchvalue; /*!< The value of the option to require for matching (i.e. 'peer' for type= in sip.conf) */</p><p style="white-space: pre-wrap; word-wrap: break-word;">main/config.c: if (!strcasecmp(v->name, "sipfriends")) {<br>main/config.c: ast_log(LOG_WARNING, "The 'sipfriends' table is obsolete, update your config to use sippeers instead.\n");<br>main/config.c: ast_realtime_append_mapping("sippeers", driver, database, table ? table : "sipfriends", pri);</p><p style="white-space: pre-wrap; word-wrap: break-word;">main/udptl.c: ast_log(LOG_WARNING, "t38faxudpec in udptl.conf is no longer supported; use the t38pt_udptl configuration option in sip.conf instead.\n");</p><p style="white-space: pre-wrap; word-wrap: break-word;">configs/samples/res_config_mysql.conf.sample:; sippeers => mysql,readhost.asterisk/writehost.asterisk,sippeers</p><p style="white-space: pre-wrap; word-wrap: break-word;"><br>configs/samples/extensions.ael.sample:// Note that you must have a [sipprovider] section in sip.conf whereas<br>configs/samples/extensions.ael.sample://_41X. => Dial(SIP/${EXTEN:2}@sipprovider,,r);<br>configs/samples/cli_aliases.conf.sample:sip debug=sip set debug<br>configs/samples/cli_aliases.conf.sample:sip no debug=sip set debug off<br>configs/samples/cli_aliases.conf.sample:sip history=sip set history on<br>configs/samples/cli.conf.sample:;sip set debug on = yes<br>configs/samples/extconfig.conf.sample:;sippeers => odbc,asterisk<br>configs/samples/extconfig.conf.sample:;sipregs => odbc,asterisk ; (avoid sipregs if possible, e.g. by using a view)<br>configs/samples/res_config_mysql.conf.sample:; sippeers => mysql,readhost.asterisk/writehost.asterisk,sippeers</p><p style="white-space: pre-wrap; word-wrap: break-word;"><br>configs/samples/users.conf.sample:; files (e.g. sip.conf, iax.conf, etc) but is intended to accelerate the<br>configs/samples/users.conf.sample:hassip = yes<br>configs/samples/users.conf.sample:;hassip = yes<br>configs/samples/sip_notify.conf.sample:Content=>Message-Account: sip:asterisk@127.0.0.1<br>configs/samples/sip_notify.conf.sample:[sipura-check-cfg]<br>configs/samples/sip_notify.conf.sample:[sipura-get-report]</p><p style="white-space: pre-wrap; word-wrap: break-word;"><br>configs/samples/res_ldap.conf.sample:; sip.conf = ldap,"dc=myDomain,dc=myDomainExt",config</p><p style="white-space: pre-wrap; word-wrap: break-word;">configs/samples/extensions.lua.sample: -- not have allowguest=no in sip.conf, everybody out there may use your</p><p style="white-space: pre-wrap; word-wrap: break-word;"><br>configs/samples/extensions.conf.sample:;exten => _41X.,1,Dial(SIP/${FILTER(0-9,${EXTEN:2})}@sipprovider,,r)</p><p style="white-space: pre-wrap; word-wrap: break-word;"><br>configs/samples/cli_permissions.conf.sample:; sip show peer<br>configs/samples/cli_permissions.conf.sample:; allowing the user to use 'channel originate' on a sip channel and with the 'extension'<br>configs/samples/cli_permissions.conf.sample:; permit=sip show ; all commands starting with 'sip show' will be allowed<br>configs/samples/cli_permissions.conf.sample:;permit=sip<br>configs/samples/cli_permissions.conf.sample:;permit=sip show peer<br>configs/samples/cli_permissions.conf.sample:;deny=sip show peers<br>configs/samples/cli_permissions.conf.sample:; permit=sip show</p><p style="white-space: pre-wrap; word-wrap: break-word;"><br>pbx/pbx_config.c: if (ast_true(ast_config_option(cfg, cat, "hassip"))) {</p><p style="white-space: pre-wrap; word-wrap: break-word;"><br>rest-api/api-docs/endpoints.json: "description": "Technology of the endpoints (sip,iax2,...)",<br></p></li></ul></li></ul><p>To view, visit <a href="https://gerrit.asterisk.org/c/asterisk/+/19611">change 19611</a>. 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<div style="display:none"> Gerrit-Project: asterisk </div>
<div style="display:none"> Gerrit-Branch: master </div>
<div style="display:none"> Gerrit-Change-Id: Ic700168c80b68879d9cee8bb07afe2712fb17996 </div>
<div style="display:none"> Gerrit-Change-Number: 19611 </div>
<div style="display:none"> Gerrit-PatchSet: 2 </div>
<div style="display:none"> Gerrit-Owner: Michael Bradeen <mbradeen@sangoma.com> </div>
<div style="display:none"> Gerrit-Reviewer: Friendly Automation </div>
<div style="display:none"> Gerrit-Reviewer: Joshua Colp <jcolp@sangoma.com> </div>
<div style="display:none"> Gerrit-Reviewer: N A <asterisk@phreaknet.org> </div>
<div style="display:none"> Gerrit-Attention: Michael Bradeen <mbradeen@sangoma.com> </div>
<div style="display:none"> Gerrit-Comment-Date: Tue, 13 Dec 2022 19:18:45 +0000 </div>
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