diff options
| author | Maxim Sobolev <sobomax@FreeBSD.org> | 2008-05-16 09:24:29 +0000 |
|---|---|---|
| committer | Maxim Sobolev <sobomax@FreeBSD.org> | 2008-05-16 09:24:29 +0000 |
| commit | 285e2c278ab94c27d91796679099a77878524cdf (patch) | |
| tree | fdb5db734e5d21b486bc025816887cd269754368 /net/asterisk14/files/patch-channels::chan_sip.c | |
| parent | Kradview is a GPLed viewer of images obtained for some different (diff) | |
o Update to 1.4.19.2;
o move all additional functionality into separate patches and make it
opt in. It has been concern of asterisks devs that the FreeBSD pacakage
adds functionalty not present in the original version, whch could be
confusing.
Notes
Notes:
svn path=/head/; revision=213118
Diffstat (limited to 'net/asterisk14/files/patch-channels::chan_sip.c')
| -rw-r--r-- | net/asterisk14/files/patch-channels::chan_sip.c | 25 |
1 files changed, 8 insertions, 17 deletions
diff --git a/net/asterisk14/files/patch-channels::chan_sip.c b/net/asterisk14/files/patch-channels::chan_sip.c index 916cc7ffa0ac..e54df0eb1bf1 100644 --- a/net/asterisk14/files/patch-channels::chan_sip.c +++ b/net/asterisk14/files/patch-channels::chan_sip.c @@ -1,6 +1,6 @@ ---- channels/chan_sip.c.orig Mon Dec 24 11:59:46 2007 -+++ channels/chan_sip.c Mon Dec 24 11:58:47 2007 -@@ -493,7 +493,7 @@ +--- channels/chan_sip.c.orig 2008-03-18 16:42:59.000000000 +0200 ++++ channels/chan_sip.c 2008-03-18 17:08:34.000000000 +0200 +@@ -495,7 +495,7 @@ #define DEFAULT_MOHINTERPRET "default" #define DEFAULT_MOHSUGGEST "" #define DEFAULT_VMEXTEN "asterisk" @@ -9,16 +9,7 @@ #define DEFAULT_NOTIFYMIME "application/simple-message-summary" #define DEFAULT_MWITIME 10 #define DEFAULT_ALLOWGUEST TRUE -@@ -3985,6 +3985,8 @@ - ast_codec_pref_remove2(&tmp->nativeformats, ~i->usercapability); - fmt = ast_codec_pref_index_audio(&tmp->nativeformats, 0); - -+ pbx_builtin_setvar_helper(tmp, "SIP_CODEC_USED", ast_getformatname(fmt)); -+ - /* If we have a prefcodec setting, we have an inbound channel that set a - preferred format for this call. Otherwise, we check the jointcapability - We also check for vrtp. If it's not there, we are not allowed do any video anyway. -@@ -15845,6 +15847,9 @@ +@@ -15873,6 +15881,9 @@ char *ext, *host; char tmp[256]; char *dest = data; @@ -26,9 +17,9 @@ + char *md5secret = NULL; + char *authname = NULL; - if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE))) { - ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory or socket error)\n", (char *)data); -@@ -15866,6 +15871,17 @@ + oldformat = format; + if (!(format &= ((AST_FORMAT_MAX_AUDIO << 1) - 1))) { +@@ -15903,6 +15914,17 @@ if (host) { *host++ = '\0'; ext = tmp; @@ -46,7 +37,7 @@ } else { ext = strchr(tmp, '/'); if (ext) -@@ -15898,6 +15914,14 @@ +@@ -15933,6 +15955,14 @@ ast_string_field_set(p, username, ext); ast_string_field_free(p, fullcontact); } |