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res_pjsip_outbound_registration: generate correct Contact URI for TLS

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There are two types of SIP URIs indicating a secure transport:
* sips:user@example.org
* sip:user@example.org;transport=tls

When using a sips URI, Asterisk checks incoming INVITEs and answers from
the other side for sips URIs, and rejects the packet if there are only
sip URIs. So Asterisk should only generate a sips Contact URI if the
other side supports it.

This patch makes Asterisk generate either a sip or sips Contact URI
depending on the format of the server URI.

If you want a sip URI, use:
server_uri=sip:example.org\;transport=tls

If you want a sips URI, use:
server_uri=sips:example.org

ASTERISK-25990 #close
Reported-by: Sebastian Damm

Change-Id: I5ae57d6531ce940b5fc64d5cd2673e60db0f9ba2
This commit is contained in:
Sebastian Damm
2016-05-10 17:19:48 +02:00
committed by Joshua Colp
parent a01ce2b889
commit a94a12bbf7
1 changed files with 1 additions and 1 deletions
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2
res/res_pjsip_outbound_registration.c
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@@ -1096,7 +1096,7 @@ static int sip_dialog_create_contact(pj_pool_t *pool, pj_str_t *contact, const c
contact->ptr = pj_pool_alloc(pool, PJSIP_MAX_URL_SIZE);
contact->slen = pj_ansi_snprintf(contact->ptr, PJSIP_MAX_URL_SIZE,
"<%s:%s@%s%.*s%s:%d%s%s%s%s>",
(pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) ? "sips" : "sip",
((pjsip_transport_get_flag_from_type(type) & PJSIP_TRANSPORT_SECURE) && PJSIP_URI_SCHEME_IS_SIPS(uri)) ? "sips" : "sip",
user,
(type & PJSIP_TRANSPORT_IPV6) ? "[" : "",
(int)local_addr.slen,