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Update sip_request_call SIP dial string documentation.

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This was missed when merging review r1859.
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Merged revisions 375074 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 375078 from http://svn.asterisk.org/svn/asterisk/branches/10
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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:
Walter Doekes
2012-10-16 19:25:11 +00:00
parent c4df9778cb
commit 2142fc3bc7
1 changed files with 11 additions and 6 deletions
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17
channels/chan_sip.c
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@@ -29429,13 +29429,18 @@ static int sip_devicestate(const char *data)
/*! \brief PBX interface function -build SIP pvt structure /*! \brief PBX interface function -build SIP pvt structure
* SIP calls initiated by the PBX arrive here. * SIP calls initiated by the PBX arrive here.
* *
* \verbatim * \verbatim
* SIP Dial string syntax * SIP Dial string syntax:
* SIP/exten@host!dnid * SIP/devicename
* or SIP/host/exten!dnid * or SIP/username@domain (SIP uri)
* or SIP/host!dnid * 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 * \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) 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; struct sip_pvt *p;