Author: oej
Date: Tue Oct 21 04:58:59 2014
New Revision: 426077
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=426077
Log:
- Add Allow header that was missing
- Remove extra ;
- Add documentation in sip.conf
Modified:
team/oej/gin-register-bnc-1.8/channels/chan_sip.c
team/oej/gin-register-bnc-1.8/configs/sip.conf.sample
Modified: team/oej/gin-register-bnc-1.8/channels/chan_sip.c
URL:
http://svnview.digium.com/svn/asterisk/team/oej/gin-register-bnc-1.8/channels/chan_sip.c?view=diff&rev=426077&r1=426076&r2=426077
==============================================================================
--- team/oej/gin-register-bnc-1.8/channels/chan_sip.c (original)
+++ team/oej/gin-register-bnc-1.8/channels/chan_sip.c Tue Oct 21 04:58:59 2014
@@ -12349,14 +12349,13 @@
if (p->socket.type != SIP_TRANSPORT_UDP) {
snprintf(params, sizeof(params), ";transport=%s%s",
get_transport(p->socket.type), param);
} else {
- snprintf(params, sizeof(params), ";%s", param);
+ snprintf(params, sizeof(params), "%s", param);
}
}
ast_string_field_build(p, our_contact, "<sip:%s%s%s%s>", user,
ast_strlen_zero(user) ? "" : "@",
ast_sockaddr_stringify_remote(&p->ourip),
params);
- }
}
/*! \brief Initiate new SIP request to peer/user */
@@ -13917,6 +13916,8 @@
add_header(&req, "To", to);
add_header(&req, "Call-ID", p->callid);
add_header(&req, "CSeq", tmp);
+ add_header(&req, "Allow", ALLOWED_METHODS);
+
if (!ast_strlen_zero(global_useragent))
add_header(&req, "User-Agent", global_useragent);
Modified: team/oej/gin-register-bnc-1.8/configs/sip.conf.sample
URL:
http://svnview.digium.com/svn/asterisk/team/oej/gin-register-bnc-1.8/configs/sip.conf.sample?view=diff&rev=426077&r1=426076&r2=426077
==============================================================================
--- team/oej/gin-register-bnc-1.8/configs/sip.conf.sample (original)
+++ team/oej/gin-register-bnc-1.8/configs/sip.conf.sample Tue Oct 21 04:58:59
2014
@@ -692,6 +692,12 @@
;
; and more readable because you don't have to write the parameters in two
places
; (note that the "port" is ignored - this is a bug that should be fixed).
+;
+; If you use the extension "-bnc-" Asterisk will activate RFC 6140 Bulk Number
Contact
+; support used in SIP Connect trunks. Asterisk will register without a
username part
+; in the Contact: header and calls will be addressed to the DID you have with
the
+; server. Your dialplan needs to support these numbers. This is a SIP extension
+; that has to be supported by the server for it to work.
;
; Note that a register= line doesn't mean that we will match the incoming call
in any
; other way than described above. If you want to control where the call enters
your
--
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