[asterisk-bugs] [JIRA] (ASTERISK-28513) Should To: be rewritten when forwarding to a phone

Brian J. Murrell (JIRA) noreply at issues.asterisk.org
Fri Oct 11 06:53:48 CDT 2019


    [ https://issues.asterisk.org/jira/browse/ASTERISK-28513?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=248376#comment-248376 ] 

Brian J. Murrell commented on ASTERISK-28513:
---------------------------------------------

I tried this patch (instead of my hard-coded system name):

{noformat}
diff --git a/res/res_pjsip_messaging.c b/res/res_pjsip_messaging.c
--- a/res/res_pjsip_messaging.c
+++ b/res/res_pjsip_messaging.c
@@ -43,6 +43,7 @@
 #include "asterisk/res_pjsip.h"
 #include "asterisk/res_pjsip_session.h"
 #include "asterisk/taskprocessor.h"
+#include "asterisk/paths.h"
 
 const pjsip_method pjsip_message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
 
@@ -230,10 +230,14 @@
 	parsed_name_addr = (pjsip_name_addr *) pjsip_parse_uri(tdata->pool, to, strlen(to),
 		PJSIP_PARSE_URI_AS_NAMEADDR);
 	if (parsed_name_addr) {
-		if (pj_strlen(&parsed_name_addr->display)) {
-			pjsip_name_addr *name_addr =
-				(pjsip_name_addr *) PJSIP_MSG_TO_HDR(tdata->msg)->uri;
+		pjsip_sip_uri *uri;
+		pjsip_name_addr *name_addr =
+			(pjsip_name_addr *) PJSIP_MSG_TO_HDR(tdata->msg)->uri;
+		uri = pjsip_uri_get_uri(name_addr);
+		pj_str_t my_host = pj_str(ast_config_AST_SYSTEM_NAME);
+		pj_strdup(tdata->pool, &uri->host, &my_host);
 
+		if (pj_strlen(&parsed_name_addr->display)) {
 			pj_strdup(tdata->pool, &name_addr->display, &parsed_name_addr->display);
 		}
 	}
{noformat}

but that resulted in a {{To:}} with the domainname part empty.

Can I not use {{ast_config_AST_SYSTEM_NAME}} here like this?

> Should To: be rewritten when forwarding to a phone
> --------------------------------------------------
>
>                 Key: ASTERISK-28513
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28513
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip_messaging
>    Affects Versions: 13.28.0
>            Reporter: Brian J. Murrell
>            Assignee: Unassigned
>            Severity: Minor
>              Labels: pjsip
>
> I have run into a problem with {{MESSAGE}} s and the [Linphone Android client|https://github.com/BelledonneCommunications/linphone-android].
> The [issue|https://github.com/BelledonneCommunications/linphone-android/issues/605] as described in their tracker is that when {{MESSAGE}} s come to the linphone client from the same sender, they can be "filed" into many different threads, rather than all in one chat/thread.  This is because linphone separates chats based on both the From: and To: headers.
> As I am sure you know, the {{To:}} header of a client can vary wildly based on the IP address it's connecting from.  This means that every time the IP address of the SIP client changes, a new chat for the same sender is created.
> But the problem is that Asterisk is setting the {{To:}} header of the {{MESSAGE}} to the {{user at ip_address}} of the remote SIP client and so this means that every time the IP address of the remote SIP client changes, a new To: header is created, and so is a new chat in the SIP client.
> Linphone defends this behaviour by insisting that the {{To:}} header value is a logical value of the recipient for a given domain and should always remain it's logical value no matter whether it's being forwarded on to a SIP client or not.
> So for example, if my Asterisk server is at pbx.example.com and somebody (my VOIP provider for example) send a {{MESSAGE}} to 555-555-1212 at ip-address-of-my-asterisk, when my Asterisk server receives that message and then wants to forward it on to a SIP client, the To: should be {{To:  _recipient_]@pbx.example.com}}, not {{To: _recipient_ at ip-address-of-SIP-client}}.
> They quote [RFC 3261 section 8.1.1.2|https://tools.ietf.org/html/rfc3261#section-8.1.1.2] further in defending this behaviour.  My reading of it doesn't leave me with much to argue against their defence.
> I don't see any way to make Asterisk (with PJSIP) to follow this behaviour.



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