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| author | Matthew Jordan <mjordan@digium.com> | 2014-12-24 15:26:53 +0000 |
|---|---|---|
| committer | Matthew Jordan <mjordan@digium.com> | 2014-12-24 15:26:53 +0000 |
| commit | b521c612fc2116cc381412c8e72bf8a719922481 (patch) | |
| tree | 95971d186e3e3e0fbd0e901fbbe263ea0ed2f662 /res | |
| parent | 915bb88d3e973f647eb9d9e560688d6a02af2c2a (diff) | |
res_pjsip: Backport missing commits for user_eq_phone
This backports the following from trunk, which were missed:
r427257 | file | 2014-11-04 16:31:16 -0600 (Tue, 04 Nov 2014) | 2 lines
res_pjsip: Allow + at the beginning of a phone number when user_eq_phone is enabled.
r427259 | file | 2014-11-04 16:51:32 -0600 (Tue, 04 Nov 2014) | 2 lines
res_pjsip: Apply the 'user_eq_phone' setting to the To header as well.
It also adds the Alembic script for the option.
ASTERISK-24643
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res')
| -rw-r--r-- | res/res_pjsip.c | 5 |
1 files changed, 5 insertions, 0 deletions
diff --git a/res/res_pjsip.c b/res/res_pjsip.c index 223e58429..6f5826997 100644 --- a/res/res_pjsip.c +++ b/res/res_pjsip.c @@ -2155,6 +2155,10 @@ void ast_sip_add_usereqphone(const struct ast_sip_endpoint *endpoint, pj_pool_t return; } + if (pj_strbuf(&sip_uri->user)[0] == '+') { + i = 1; + } + /* Test URI user against allowed characters in AST_DIGIT_ANY */ for (; i < pj_strlen(&sip_uri->user); i++) { if (!strchr(AST_DIGIT_ANYNUM, pj_strbuf(&sip_uri->user)[i])) { @@ -2221,6 +2225,7 @@ pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, /* Add the user=phone parameter if applicable */ ast_sip_add_usereqphone(endpoint, dlg->pool, dlg->target); + ast_sip_add_usereqphone(endpoint, dlg->pool, dlg->remote.info->uri); /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */ dlg->sess_count++; |