Update sip_request_call SIP dial string documentation.
This was missed when merging review r1859. ........ Merged revisions 375074 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375078 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375079 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
parent
c4df9778cb
commit
2142fc3bc7
|
@ -29429,13 +29429,18 @@ static int sip_devicestate(const char *data)
|
|||
/*! \brief PBX interface function -build SIP pvt structure
|
||||
* SIP calls initiated by the PBX arrive here.
|
||||
*
|
||||
* \verbatim
|
||||
* SIP Dial string syntax
|
||||
* SIP/exten@host!dnid
|
||||
* or SIP/host/exten!dnid
|
||||
* or SIP/host!dnid
|
||||
* \verbatim
|
||||
* SIP Dial string syntax:
|
||||
* SIP/devicename
|
||||
* or SIP/username@domain (SIP uri)
|
||||
* or SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
|
||||
* or SIP/devicename/extension
|
||||
* or SIP/devicename/extension/IPorHost
|
||||
* or SIP/username@domain//IPorHost
|
||||
* and there is an optional [!dnid] argument you can append to alter the
|
||||
* To: header.
|
||||
* \endverbatim
|
||||
*/
|
||||
*/
|
||||
static struct ast_channel *sip_request_call(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *dest, int *cause)
|
||||
{
|
||||
struct sip_pvt *p;
|
||||
|
|
Loading…
Reference in New Issue