diff options
Diffstat (limited to 'res')
-rw-r--r-- | res/res_pjsip.c | 28 | ||||
-rw-r--r-- | res/res_pjsip_mwi.c | 2 |
2 files changed, 15 insertions, 15 deletions
diff --git a/res/res_pjsip.c b/res/res_pjsip.c index bf859fe88..310ff20bc 100644 --- a/res/res_pjsip.c +++ b/res/res_pjsip.c @@ -66,7 +66,7 @@ It contains the core SIP related options only, endpoints are <emphasis>NOT</emphasis> dialable entries of their own. Communication with another SIP device is accomplished via Addresses of Record (AoRs) which have one or more - contacts assicated with them. Endpoints <emphasis>NOT</emphasis> configured to + contacts associated with them. Endpoints <emphasis>NOT</emphasis> configured to use a <literal>transport</literal> will default to first transport found in <filename>pjsip.conf</filename> that matches its type. </para> @@ -449,7 +449,7 @@ <configOption name="timers_min_se" default="90"> <synopsis>Minimum session timers expiration period</synopsis> <description><para> - Minimium session timer expiration period. Time in seconds. + Minimum session timer expiration period. Time in seconds. </para></description> </configOption> <configOption name="timers" default="yes"> @@ -467,7 +467,7 @@ <configOption name="timers_sess_expires" default="1800"> <synopsis>Maximum session timer expiration period</synopsis> <description><para> - Maximium session timer expiration period. Time in seconds. + Maximum session timer expiration period. Time in seconds. </para></description> </configOption> <configOption name="transport"> @@ -509,7 +509,7 @@ <synopsis>Must be of type 'endpoint'.</synopsis> </configOption> <configOption name="use_ptime" default="no"> - <synopsis>Use Endpoint's requested packetisation interval</synopsis> + <synopsis>Use Endpoint's requested packetization interval</synopsis> </configOption> <configOption name="use_avpf" default="no"> <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this @@ -647,7 +647,7 @@ Forward error correction should be used. </para></enum> <enum name="redundancy"><para> - Redundacy error correction should be used. + Redundancy error correction should be used. </para></enum> </enumlist> </description> @@ -1111,7 +1111,7 @@ <description><para>Only used when auth_type is <literal>md5</literal>.</para></description> </configOption> <configOption name="password"> - <synopsis>PlainText password used for authentication.</synopsis> + <synopsis>Plain text password used for authentication.</synopsis> <description><para>Only used when auth_type is <literal>userpass</literal>.</para></description> </configOption> <configOption name="realm"> @@ -1316,7 +1316,7 @@ </description> </configOption> <configOption name="symmetric_transport" default="no"> - <synopsis>Use the same transport for outgoing reqests as incoming ones.</synopsis> + <synopsis>Use the same transport for outgoing requests as incoming ones.</synopsis> <description> <para>When a request from a dynamic contact comes in on a transport with this option set to 'yes', @@ -1361,7 +1361,7 @@ <configOption name="qualify_timeout" default="3.0"> <synopsis>Timeout for qualify</synopsis> <description><para> - If the contact doesn't repond to the OPTIONS request before the timeout, + If the contact doesn't respond to the OPTIONS request before the timeout, the contact is marked unavailable. If <literal>0</literal> no timeout. Time in fractional seconds. </para></description> @@ -1445,8 +1445,8 @@ <literal>endpoint</literal> for calls. </para><para> This can be used as another way of grouping a list of contacts to dial - rather than specifing them each directly when dialing via the dialplan. - This must be used in conjuction with the <literal>PJSIP_DIAL_CONTACTS</literal>. + rather than specifying them each directly when dialing via the dialplan. + This must be used in conjunction with the <literal>PJSIP_DIAL_CONTACTS</literal>. </para><para> Registrations: For Asterisk to match an inbound registration to an endpoint, the AoR object name must match the user portion of the SIP URI in the "To:" @@ -1486,7 +1486,7 @@ <configOption name="maximum_expiration" default="7200"> <synopsis>Maximum time to keep an AoR</synopsis> <description><para> - Maximium time to keep a peer with explicit expiration. Time in seconds. + Maximum time to keep a peer with explicit expiration. Time in seconds. </para></description> </configOption> <configOption name="max_contacts" default="0"> @@ -1560,7 +1560,7 @@ <configOption name="qualify_timeout" default="3.0"> <synopsis>Timeout for qualify</synopsis> <description><para> - If the contact doesn't repond to the OPTIONS request before the timeout, + If the contact doesn't respond to the OPTIONS request before the timeout, the contact is marked unavailable. If <literal>0</literal> no timeout. Time in fractional seconds. </para></description> @@ -1659,7 +1659,7 @@ <configOption name="disable_multi_domain" default="no"> <synopsis>Disable Multi Domain support</synopsis> <description><para> - If disabled it can improve realtime performace by reducing number of database requsts. + If disabled it can improve realtime performance by reducing the number of database requests. </para></description> </configOption> <configOption name="max_initial_qualify_time" default="0"> @@ -1785,7 +1785,7 @@ in the user field of a SIP URI then the field is truncated at the first semicolon. This effectively makes the semicolon a non-usable character for PJSIP endpoint names, extensions, - and AORs. This can be useful for improving compatability with + and AORs. This can be useful for improving compatibility with an ITSP that likes to use user options for whatever reason. </para> <example title="Sample SIP URI"> diff --git a/res/res_pjsip_mwi.c b/res/res_pjsip_mwi.c index 90fdc098c..4cd892c05 100644 --- a/res/res_pjsip_mwi.c +++ b/res/res_pjsip_mwi.c @@ -650,11 +650,11 @@ static void send_mwi_notify(struct mwi_subscription *sub) .body_type = AST_SIP_MESSAGE_ACCUMULATOR, .body_data = &counter, }; - const char *resource = ast_sip_subscription_get_resource_name(sub->sip_sub); ao2_callback(sub->stasis_subs, OBJ_NODATA, get_message_count, &counter); if (sub->is_solicited) { + const char *resource = ast_sip_subscription_get_resource_name(sub->sip_sub); struct ast_sip_endpoint *endpoint = ast_sip_subscription_get_endpoint(sub->sip_sub); struct ast_sip_aor *aor = find_aor_for_resource(endpoint, resource); pjsip_dialog *dlg = ast_sip_subscription_get_dialog(sub->sip_sub); |