[asterisk-commits] oej: branch oej/codename-pineapple r48284 - in
/team/oej/codename-pineapple: ...
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Dec 5 14:40:59 MST 2006
Author: oej
Date: Tue Dec 5 15:40:59 2006
New Revision: 48284
URL: http://svn.digium.com/view/asterisk?view=rev&rev=48284
Log:
- stringfieldify devices (still compiles with errors)
- added globals to domains (just a start, need to figure out configuration syntax)
Modified:
team/oej/codename-pineapple/channels/chan_sip3.c
team/oej/codename-pineapple/channels/sip3/sip3.h
team/oej/codename-pineapple/channels/sip3/sip3_compose.c
team/oej/codename-pineapple/channels/sip3/sip3_config.c
team/oej/codename-pineapple/channels/sip3/sip3_services.c
team/oej/codename-pineapple/configs/sip3.conf.sample
Modified: team/oej/codename-pineapple/channels/chan_sip3.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/chan_sip3.c?view=diff&rev=48284&r1=48283&r2=48284
==============================================================================
--- team/oej/codename-pineapple/channels/chan_sip3.c (original)
+++ team/oej/codename-pineapple/channels/chan_sip3.c Tue Dec 5 15:40:59 2006
@@ -417,7 +417,7 @@
/* Default setttings are used as a channel setting and as a default when
configuring devices */
-/* Global settings only apply to the channel */
+/*! Global settings only apply to the channel */
struct sip_globals global;
/* Object counters */
@@ -835,6 +835,9 @@
device->registry->peer = NULL;
ASTOBJ_UNREF(device->registry,sip_registry_destroy);
}
+
+ /* Free the stringfield pool */
+ ast_string_field_free_pools(device);
free(device);
}
@@ -2682,7 +2685,8 @@
return;
if (username)
- ast_copy_string(peer->defaultuser, username, sizeof(peer->defaultuser));
+ ast_string_field_set(peer, defaultuser, username);
+ //ast_copy_string(peer->defaultuser, username, sizeof(peer->defaultuser));
if (contact)
ast_copy_string(peer->fullcontact, contact, sizeof(peer->fullcontact));
@@ -2904,9 +2908,11 @@
peer->sipoptions = pvt->sipoptions;
if (curi) { /* Overwrite the default username from config at registration */
- ast_copy_string(peer->defaultuser, curi, sizeof(peer->defaultuser));
+ ast_string_field_set(peer, defaultuser, curi);
+ //ast_copy_string(peer->defaultuser, curi, sizeof(peer->defaultuser));
} else
- peer->defaultuser[0] = '\0';
+ ast_string_field_set(peer, defaultuser, "");
+ //peer->defaultuser[0] = '\0';
if (peer->expire > -1)
ast_sched_del(sched, peer->expire);
@@ -6361,7 +6367,7 @@
}
/* Send MWI */
ast_set_flag(&p->flags[0], SIP_OUTGOING);
- transmit_notify_with_mwi(p, newmsgs, oldmsgs, peer->vmexten);
+ transmit_notify_with_mwi(p, newmsgs, oldmsgs, (char *) peer->vmexten);
return 0;
}
Modified: team/oej/codename-pineapple/channels/sip3/sip3.h
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3.h?view=diff&rev=48284&r1=48283&r2=48284
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3.h (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3.h Tue Dec 5 15:40:59 2006
@@ -45,8 +45,8 @@
#endif
#define DEFAULT_DEFAULT_EXPIRY 120
-#define DEFAULT_MIN_EXPIRY 60
-#define DEFAULT_MAX_EXPIRY 3600
+#define DEFAULT_MIN_EXPIRY 60 /*! Minimum expiry time for subscription and registration requests */
+#define DEFAULT_MAX_EXPIRY 3600 /*! Maximum expiry time for regs and subs */
#define DEFAULT_REGISTRATION_TIMEOUT 20
#define DEFAULT_MAX_FORWARDS "70"
@@ -497,6 +497,62 @@
char hop[0];
};
+/*! Global settings only apply to the channel */
+struct sip_globals {
+ struct ast_jb_conf jbconf; /*!< Jitterbuffer configuration */
+ int maxforwards; /*!< Max forwards */
+ int rtautoclear; /*!< Realtime caching options */
+ int notifyringing; /*!< Send notifications on ringing */
+ int alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */
+ int srvlookup; /*!< SRV Lookup on or off. Default is off, RFC behavior is on */
+ int autocreatepeer; /*!< Auto creation of peers at registration? Default off. */
+ int relaxdtmf; /*!< Relax DTMF */
+ int rtptimeout; /*!< Time out call if no RTP */
+ int rtpholdtimeout;
+ int rtpkeepalive; /*!< Send RTP keepalives */
+ int reg_timeout;
+ int regattempts_max; /*!< Registration attempts before giving up */
+ int allowguest; /*!< allow unauthenticated users/peers to connect? */
+ int allowsubscribe; /*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE
+ the global setting is in globals_flags[1] */
+ int mwitime; /*!< Time between MWI checks for peers */
+ unsigned int tos_sip; /*!< IP type of service for SIP packets */
+ unsigned int tos_audio; /*!< IP type of service for audio RTP packets */
+ unsigned int tos_video; /*!< IP type of service for video RTP packets */
+ unsigned int tos_presence; /*!< IP type of service for SIP presence packets */
+ int compactheaders; /*!< send compact sip headers */
+ int recordhistory; /*!< Record SIP history. Off by default */
+ int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
+ char realm[MAXHOSTNAMELEN]; /*!< Default realm */
+ char regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */
+ char useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */
+ int allow_external_domains; /*!< Accept calls to external SIP domains? */
+ int callevents; /*!< Whether we send manager events or not */
+ int t1min; /*!< T1 roundtrip time minimum */
+ enum transfermodes allowtransfer; /*!< SIP Refer restriction scheme */
+ int autoframing;
+ struct ast_flags flags[2]; /* Flags for various default settings */
+ /* Default values */
+ int default_qualifycheck_ok; /*!< Default qualify time when status is ok */
+ int default_qualifycheck_notok; /*!< Default qualify time when statusis not ok */
+ int default_qualify; /*!< Default Qualify= setting */
+ int capability; /*!< Codec support */
+ int dtmf_capability; /*!< DTMF support (2833) */
+ int t38_capability; /*!< T38 Capability */
+ char default_context[AST_MAX_CONTEXT];
+ char default_subscribecontext[AST_MAX_CONTEXT];
+ char default_language[MAX_LANGUAGE];
+ char default_callerid[AST_MAX_EXTENSION];
+ char default_fromdomain[AST_MAX_EXTENSION];
+ char default_notifymime[AST_MAX_EXTENSION];
+ char default_vmexten[AST_MAX_EXTENSION];
+ char default_mohinterpret[MAX_MUSICCLASS]; /*!< Global setting for moh class to use when put on hold */
+ char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
+ * a bridged channel on hold */
+ int default_maxcallbitrate; /*!< Maximum bitrate for call */
+ struct ast_codec_pref default_prefs; /*!< Default codec prefs */
+};
+
/*! \brief Domain data structure.
\note In the future, we will connect this to a configuration tree specific
@@ -506,6 +562,7 @@
char domain[MAXHOSTNAMELEN]; /*!< SIP domain we are responsible for */
char context[AST_MAX_EXTENSION]; /*!< Incoming context for this domain */
enum domain_mode mode; /*!< How did we find this domain? */
+ struct sip_globals globals; /*!< Globals for this domain */
AST_LIST_ENTRY(domain) list; /*!< List mechanics */
};
@@ -771,6 +828,7 @@
int expiry; /*!< How long we take to expire */
char tag[11]; /*!< Dialog ID: Our tag for this session */
int callingpres; /*!< Caller ID presentation settings */
+ int maxforwards; /*!< Max forwards for this dialog */
struct sip_route *route; /*!< Head of linked list of routing steps (fm Record-Route) */
int route_persistant; /*!< Is this the "real" route? */
char lastmsg[256]; /*!< Last Message sent/received */
@@ -860,34 +918,50 @@
a domain only
For type=phone, the domain needs to be a locally hosted domain */
enum objecttype type; /*!< SIP_PEER */
- char secret[80]; /*!< Password */
- char md5secret[80]; /*!< Password in MD5 */
+ AST_DECLARE_STRING_FIELDS(
+ AST_STRING_FIELD(secret); /*!< Secret (password) */
+ AST_STRING_FIELD(md5secret); /*!< Secret (password in md5 digest with realm and username) */
+ AST_STRING_FIELD(context); /*!< Default context for incoming calls */
+ AST_STRING_FIELD(subscribecontext); /*!< Default context for subscriptions */
+ AST_STRING_FIELD(defaultuser); /*!< Temporary username until registration */
+ AST_STRING_FIELD(authuser); /*!< Authentication user name */
+ AST_STRING_FIELD(regexten); /*!< Extension to register (if regcontext is used) */
+ AST_STRING_FIELD(fromuser); /*!< From: user when calling this peer */
+ AST_STRING_FIELD(fromdomain); /*!< From: domain when calling this peer */
+ AST_STRING_FIELD(vmexten); /*!< Dialplan extension for MWI notify message*/
+ AST_STRING_FIELD(mailbox); /*!< Mailbox setting for MWI checks */
+ );
+ //char secret[80]; /*!< Password */
+ //char md5secret[80]; /*!< Password in MD5 */
struct sip_auth *auth; /*!< Realm authentication list */
- char context[AST_MAX_CONTEXT]; /*!< Default context for incoming calls */
- char subscribecontext[AST_MAX_CONTEXT]; /*!< Default context for subscriptions */
- char defaultuser[80]; /*!< Temporary username until registration */
- char authuser[80]; /*!< Authentication user name */
+ //char context[AST_MAX_CONTEXT]; /*!< Default context for incoming calls */
+ //char subscribecontext[AST_MAX_CONTEXT]; /*!< Default context for subscriptions */
+ //char defaultuser[80]; /*!< Temporary username until registration */
+ //char authuser[80]; /*!< Authentication user name */
char accountcode[AST_MAX_ACCOUNT_CODE]; /*!< Account code */
int amaflags; /*!< AMA Flags (for billing) */
char tohost[MAXHOSTNAMELEN]; /*!< If not dynamic, IP address */
- char regexten[AST_MAX_EXTENSION]; /*!< Extension to register (if regcontext is used) */
- char fromuser[80]; /*!< From: user when calling this peer */
- char fromdomain[MAXHOSTNAMELEN]; /*!< From: domain when calling this peer */
- char fullcontact[256]; /*!< Contact registered with us (not in sip.conf) */
+ //char regexten[AST_MAX_EXTENSION]; /*!< Extension to register (if regcontext is used) */
+ //char fromuser[80]; /*!< From: user when calling this peer */
+ //char fromdomain[MAXHOSTNAMELEN]; /*!< From: domain when calling this peer */
+ char fullcontact[256]; /*!< Contact registered with us (not in sip.conf)
+ * \note XXX Needs to move to stringfields!!! */
char cid_num[80]; /*!< Caller ID num */
char cid_name[80]; /*!< Caller ID name */
int callingpres; /*!< Calling id presentation */
int inUse; /*!< Number of calls in use */
int inRinging; /*!< Number of calls ringing */
+ int maxforwards; /*!< Max forwards */
int onHold; /*!< Peer has someone on hold */
int call_limit; /*!< Limit of concurrent calls */
enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
- char vmexten[AST_MAX_EXTENSION]; /*!< Dialplan extension for MWI notify message*/
- char mailbox[AST_MAX_EXTENSION]; /*!< Mailbox setting for MWI checks */
+ //char vmexten[AST_MAX_EXTENSION]; /*!< Dialplan extension for MWI notify message*/
+ //char mailbox[AST_MAX_EXTENSION]; /*!< Mailbox setting for MWI checks */
char language[MAX_LANGUAGE]; /*!< Default language for prompts */
char mohinterpret[MAX_MUSICCLASS];/*!< Music on Hold class */
char mohsuggest[MAX_MUSICCLASS];/*!< Music on Hold class */
- char useragent[256]; /*!< User agent in SIP request (saved from registration) */
+ char useragent[256]; /*!< User agent in SIP request (saved from registration)
+ *! \note XXX Needs to move to stringfields!!! */
struct ast_codec_pref prefs; /*!< codec prefs */
int lastmsgssent;
time_t lastmsgcheck; /*!< Last time we checked for MWI */
@@ -957,60 +1031,6 @@
struct sip_peer *peer; /*!< If we have a known peer for this registry entry, use it for incoming calls */
};
-/* Global settings only apply to the channel */
-struct sip_globals {
- struct ast_jb_conf jbconf; /*!< Jitterbuffer configuration */
- int rtautoclear; /*!< Realtime caching options */
- int notifyringing; /*!< Send notifications on ringing */
- int alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */
- int srvlookup; /*!< SRV Lookup on or off. Default is off, RFC behavior is on */
- int autocreatepeer; /*!< Auto creation of peers at registration? Default off. */
- int relaxdtmf; /*!< Relax DTMF */
- int rtptimeout; /*!< Time out call if no RTP */
- int rtpholdtimeout;
- int rtpkeepalive; /*!< Send RTP keepalives */
- int reg_timeout;
- int regattempts_max; /*!< Registration attempts before giving up */
- int allowguest; /*!< allow unauthenticated users/peers to connect? */
- int allowsubscribe; /*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE
- the global setting is in globals_flags[1] */
- int mwitime; /*!< Time between MWI checks for peers */
- unsigned int tos_sip; /*!< IP type of service for SIP packets */
- unsigned int tos_audio; /*!< IP type of service for audio RTP packets */
- unsigned int tos_video; /*!< IP type of service for video RTP packets */
- unsigned int tos_presence; /*!< IP type of service for SIP presence packets */
- int compactheaders; /*!< send compact sip headers */
- int recordhistory; /*!< Record SIP history. Off by default */
- int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
- char realm[MAXHOSTNAMELEN]; /*!< Default realm */
- char regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */
- char useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */
- int allow_external_domains; /*!< Accept calls to external SIP domains? */
- int callevents; /*!< Whether we send manager events or not */
- int t1min; /*!< T1 roundtrip time minimum */
- enum transfermodes allowtransfer; /*!< SIP Refer restriction scheme */
- int autoframing;
- struct ast_flags flags[2]; /* Flags for various default settings */
- /* Default values */
- int default_qualifycheck_ok; /*!< Default qualify time when status is ok */
- int default_qualifycheck_notok; /*!< Default qualify time when statusis not ok */
- int default_qualify; /*!< Default Qualify= setting */
- int capability; /*!< Codec support */
- int dtmf_capability; /*!< DTMF support (2833) */
- int t38_capability; /*!< T38 Capability */
- char default_context[AST_MAX_CONTEXT];
- char default_subscribecontext[AST_MAX_CONTEXT];
- char default_language[MAX_LANGUAGE];
- char default_callerid[AST_MAX_EXTENSION];
- char default_fromdomain[AST_MAX_EXTENSION];
- char default_notifymime[AST_MAX_EXTENSION];
- char default_vmexten[AST_MAX_EXTENSION];
- char default_mohinterpret[MAX_MUSICCLASS]; /*!< Global setting for moh class to use when put on hold */
- char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
- * a bridged channel on hold */
- int default_maxcallbitrate; /*!< Maximum bitrate for call */
- struct ast_codec_pref default_prefs; /*!< Default codec prefs */
-};
/* Network interface settings */
struct sip_network {
Modified: team/oej/codename-pineapple/channels/sip3/sip3_compose.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3_compose.c?view=diff&rev=48284&r1=48283&r2=48284
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_compose.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_compose.c Tue Dec 5 15:40:59 2006
@@ -507,7 +507,7 @@
snprintf(tmp, sizeof(tmp), "%d %s", ++p->ocseq, sip_method2txt(sipmethod));
add_header(req, "Via", p->via);
- add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
+ add_header(req, "Max-Forwards", global.maxforwards);
/* SLD: FIXME?: do Route: here too? I think not cos this is the first request.
* OTOH, then we won't have anything in p->route anyway */
/* Build Remote Party-ID and From */
@@ -587,7 +587,7 @@
set_destination(p, p->route->hop);
add_route(req, is_strict ? p->route->next : p->route);
}
- add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
+ add_header(req, "Max-Forwards", global.maxforwards);
ot = get_header(orig, "To");
of = get_header(orig, "From");
Modified: team/oej/codename-pineapple/channels/sip3/sip3_config.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3_config.c?view=diff&rev=48284&r1=48283&r2=48284
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_config.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_config.c Tue Dec 5 15:40:59 2006
@@ -279,8 +279,10 @@
}
ast_copy_flags(&device->flags[0], &global.flags[0], SIP_FLAGS_TO_COPY);
ast_copy_flags(&device->flags[1], &global.flags[1], SIP_PAGE2_FLAGS_TO_COPY);
- strcpy(device->context, global.default_context);
- strcpy(device->subscribecontext, global.default_subscribecontext);
+ ast_string_field_set(device, context, global.default_context);
+ ast_string_field_set(device, subscribecontext, global.default_subscribecontext);
+ //strcpy(device->context, global.default_context);
+ //strcpy(device->subscribecontext, global.default_subscribecontext);
strcpy(device->language, global.default_language);
strcpy(device->mohinterpret, global.default_mohinterpret);
strcpy(device->mohsuggest, global.default_mohsuggest);
@@ -292,14 +294,20 @@
device->rtpkeepalive = global.rtpkeepalive;
device->maxcallbitrate = global.default_maxcallbitrate;
strcpy(device->vmexten, global.default_vmexten);
- device->secret[0] = '\0';
- device->md5secret[0] = '\0';
+ ast_string_field_set(device, secret, "");
+ ast_string_field_set(device, md5secret, "");
+ ast_string_field_set(device, fromdomain, "");
+ ast_string_field_set(device, fromuser, "");
+ ast_string_field_set(device, regexten, "");
+ ast_string_field_set(device, mailbox, "");
+ //device->fromdomain[0] = '\0';
+ //device->fromuser[0] = '\0';
+ //device->regexten[0] = '\0';
+ //device->mailbox[0] = '\0';
+ //device->secret[0] = '\0';
+ //device->md5secret[0] = '\0';
device->cid_num[0] = '\0';
device->cid_name[0] = '\0';
- device->fromdomain[0] = '\0';
- device->fromuser[0] = '\0';
- device->regexten[0] = '\0';
- device->mailbox[0] = '\0';
device->callgroup = 0;
device->pickupgroup = 0;
device->allowtransfer = global.allowtransfer;
@@ -337,6 +345,11 @@
} else {
if (!(device = ast_calloc(1, sizeof(*device))))
return NULL;
+
+ if (ast_string_field_init(device, 512)) { /* Initialize string field buffer */
+ free(device);
+ return NULL;
+ }
if (realtime)
sipcounters.realtime_peers++;
@@ -378,11 +391,13 @@
ast_copy_string(device->fullcontact, v->value, sizeof(device->fullcontact));
ast_set_flag(&device->flags[1], SIP_PAGE2_RT_FROMCONTACT);
} else if (!strcasecmp(v->name, "secret"))
- ast_copy_string(device->secret, v->value, sizeof(device->secret));
+ ast_string_field_set(device, secret, v->value);
+ //ast_copy_string(device->secret, v->value, sizeof(device->secret));
else if (!strcasecmp(v->name, "authuser"))
ast_copy_string(device->authuser, v->value, sizeof(device->authuser));
else if (!strcasecmp(v->name, "md5secret"))
- ast_copy_string(device->md5secret, v->value, sizeof(device->md5secret));
+ ast_string_field_set(device, md5secret, v->value);
+ //ast_copy_string(device->md5secret, v->value, sizeof(device->md5secret));
else if (!strcasecmp(v->name, "auth"))
device->auth = add_realm_authentication(device->auth, v->value, v->lineno);
else if (!strcasecmp(v->name, "callerid")) {
@@ -392,15 +407,19 @@
} else if (!strcasecmp(v->name, "cid_number")) {
ast_copy_string(device->cid_num, v->value, sizeof(device->cid_num));
} else if (!strcasecmp(v->name, "context")) {
- ast_copy_string(device->context, v->value, sizeof(device->context));
+ ast_string_field_set(device, context, v->value);
+ //ast_copy_string(device->context, v->value, sizeof(device->context));
} else if (!strcasecmp(v->name, "subscribecontext")) {
- ast_copy_string(device->subscribecontext, v->value, sizeof(device->subscribecontext));
- } else if (!strcasecmp(v->name, "fromdomain")) {
- ast_copy_string(device->fromdomain, v->value, sizeof(device->fromdomain));
+ ast_string_field_set(device, subscribecontext, v->value);
+ //ast_copy_string(device->subscribecontext, v->value, sizeof(device->subscribecontext));
} else if (!strcasecmp(v->name, "usereqphone")) {
ast_set2_flag(&device->flags[0], ast_true(v->value), SIP_USEREQPHONE);
+ } else if (!strcasecmp(v->name, "fromdomain")) {
+ ast_string_field_set(device, fromdomain, v->value);
+ //ast_copy_string(device->fromdomain, v->value, sizeof(device->fromdomain));
} else if (!strcasecmp(v->name, "fromuser")) {
- ast_copy_string(device->fromuser, v->value, sizeof(device->fromuser));
+ ast_string_field_set(device, fromuser, v->value);
+ //ast_copy_string(device->fromuser, v->value, sizeof(device->fromuser));
} else if (!strcasecmp(v->name, "host") || !strcasecmp(v->name, "outboundproxy")) {
if (!strcasecmp(v->value, "dynamic")) {
if (!strcasecmp(v->name, "outboundproxy") || obproxyfound) {
@@ -456,11 +475,13 @@
if (device->callingpres == -1)
device->callingpres = atoi(v->value);
} else if (!strcasecmp(v->name, "defaultuser")) {
- ast_copy_string(device->defaultuser, v->value, sizeof(device->defaultuser));
+ ast_string_field_set(device, subscribecontext, v->value);
+ //ast_copy_string(device->defaultuser, v->value, sizeof(device->defaultuser));
} else if (!strcasecmp(v->name, "language")) {
ast_copy_string(device->language, v->value, sizeof(device->language));
} else if (!strcasecmp(v->name, "regexten")) {
- ast_copy_string(device->regexten, v->value, sizeof(device->regexten));
+ ast_string_field_set(device, subscribecontext, v->value);
+ //ast_copy_string(device->regexten, v->value, sizeof(device->regexten));
} else if (!strcasecmp(v->name, "call-limit")) {
device->call_limit = atoi(v->value);
if (device->call_limit < 0)
@@ -480,12 +501,14 @@
} else if (!strcasecmp(v->name, "mohsuggest")) {
ast_copy_string(device->mohsuggest, v->value, sizeof(device->mohsuggest));
} else if (!strcasecmp(v->name, "mailbox")) {
- ast_copy_string(device->mailbox, v->value, sizeof(device->mailbox));
+ ast_string_field_set(device, mailbox, v->value);
+ //ast_copy_string(device->mailbox, v->value, sizeof(device->mailbox));
sipcounters.peers_with_mwi++;
} else if (!strcasecmp(v->name, "subscribemwi")) {
ast_set2_flag(&device->flags[1], ast_true(v->value), SIP_PAGE2_SUBSCRIBEMWIONLY);
} else if (!strcasecmp(v->name, "vmexten")) {
- ast_copy_string(device->vmexten, v->value, sizeof(device->vmexten));
+ ast_string_field_set(device, vmexten, v->value);
+ //ast_copy_string(device->vmexten, v->value, sizeof(device->vmexten));
} else if (!strcasecmp(v->name, "callgroup")) {
device->callgroup = ast_get_group(v->value);
} else if (!strcasecmp(v->name, "allowtransfer")) {
@@ -654,6 +677,7 @@
/*!< This is default: NO MMR and JBIG trancoding, NO fill bit removal, transferredTCF TCF, UDP FEC, Version 0 and 9600 max fax rate */
global->t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600;
+ global->maxforwards = DEFAULT_MAX_FORWARDS;
global->tos_sip = DEFAULT_TOS_SIP;
global->tos_audio = DEFAULT_TOS_AUDIO;
global->tos_video = DEFAULT_TOS_VIDEO;
@@ -788,6 +812,10 @@
ast_set2_flag(&global.flags[1], ast_true(v->value), SIP_PAGE2_RTCACHEFRIENDS);
} else if (!strcasecmp(v->name, "rtsavesysname")) {
ast_set2_flag(&global.flags[1], ast_true(v->value), SIP_PAGE2_RTSAVE_SYSNAME);
+ } else if (!strcasecmp(v->name, "maxforwards")) {
+ global.maxforwards = atoi(v->value);
+ if (global.maxforwards < 1)
+ global.maxforwards = DEFAULT_MAX_FORWARDS;
} else if (!strcasecmp(v->name, "rtupdate")) {
ast_set2_flag(&global.flags[1], ast_true(v->value), SIP_PAGE2_RTUPDATE);
} else if (!strcasecmp(v->name, "ignoreregexpire")) {
Modified: team/oej/codename-pineapple/channels/sip3/sip3_services.c
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/channels/sip3/sip3_services.c?view=diff&rev=48284&r1=48283&r2=48284
==============================================================================
--- team/oej/codename-pineapple/channels/sip3/sip3_services.c (original)
+++ team/oej/codename-pineapple/channels/sip3/sip3_services.c Tue Dec 5 15:40:59 2006
@@ -511,7 +511,7 @@
add_header(&req, "Call-ID", p->callid);
add_header(&req, "CSeq", tmp);
add_header(&req, "User-Agent", global.useragent);
- add_header(&req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
+ add_header(&req, "Max-Forwards", global.maxforwards);
if (auth) /* Add auth header */
Modified: team/oej/codename-pineapple/configs/sip3.conf.sample
URL: http://svn.digium.com/view/asterisk/team/oej/codename-pineapple/configs/sip3.conf.sample?view=diff&rev=48284&r1=48283&r2=48284
==============================================================================
--- team/oej/codename-pineapple/configs/sip3.conf.sample (original)
+++ team/oej/codename-pineapple/configs/sip3.conf.sample Tue Dec 5 15:40:59 2006
@@ -110,8 +110,6 @@
; and subscriptions (seconds)
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outgoing registration
-;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
- ; Defaults to 100 ms
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
@@ -177,10 +175,6 @@
; contrary to the RFC3551 specification, the peer _should_
; be negotiating AAL2-G726-32 instead :-(
-;default-qualify-ok = 30 ; Set default time between OPTIONS packets when qualify=yes
- ; Default is 60. Recommended for NAT keepalives is 30 secs
-;default-qualify-notok = 10 ; How often to check when NOT ok
-
;
; If regcontext is specified, Asterisk will dynamically create and destroy a
; NoOp priority 1 extension for a given peer who registers or unregisters with
@@ -427,6 +421,15 @@
; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
+;------------------------------ ADVANCED CONFIGURATION -----------------------------
+;maxforwards = 70 ; Set the Max forards (TTL) header for outbound SIP requests
+ ; Default= 70
+;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
+ ; Defaults to 100 ms
+;default-qualify-ok = 30 ; Set default time between OPTIONS packets when qualify=yes
+ ; Default is 60. Recommended for NAT keepalives is 30 secs
+;default-qualify-notok = 10 ; How often to check when NOT ok
+
;-----------------------------------------------------------------------------------
[authentication]
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