Summary:	ASTERISK-19302: Messagesend
Reporter:	benasse (benasse)	Labels:
Date Opened:	2012-02-06 09:14:00.000-0600	Date Closed:	2012-02-06 12:18:52.000-0600
Priority:	Minor	Regression?
Status:	Closed/Complete	Components:	Channels/chan_sip/General
Versions:	10.1.0	Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	SIP messaging Try to chat with sip messaging but on the answer the messages are send to asterisk not at the user. sip.conf : [100] type=friend callerid="bebe" <100> username=100 host=dynamic secret=toto dtmfmode=rfc2833 qualify=yes context=default disallow=all allow=alaw allow=h263 allow=h263p allow=h264 [101] type=friend callerid="titi" <101> username=101 host=dynamic secret=toto dtmfmode=rfc2833 qualify=yes context=default disallow=all allow=alaw allow=h263 allow=h263p allow=h264 extension.conf : [messages] exten => _[a-z0-9].,1,NoOp(Send message) exten => _[a-z0-9].,n,Set(MESSAGE(from)=${MESSAGE(from)}) exten => _[a-z0-9].,n,NoOp(${MESSAGE(to)}) exten => _[a-z0-9].,n,MessageSend(${MESSAGE(to):0:-14}) Am I using it good ? sip debug : SIP Debugging enabled *CLI> <--- SIP read from UDP:10.128.19.201:45024 ---> MESSAGE sip:101@10.128.15.201 SIP/2.0 Via: SIP/2.0/UDP 10.128.19.201:45024;branch=z9hG4bK-d8754z-56b65d195534b2fd-1---d8754z-;rport Max-Forwards: 70 To: "101"<sip:101@10.128.15.201> From: "100"<sip:100@10.128.15.201>;tag=84f8fcf5 Call-ID: MGIyM2ZkOTVlYzhiNzJmYzY2Yzk3YWNjMzk2ZjMxZDM. CSeq: 15 MESSAGE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/im-iscomposing+xml User-Agent: X-Lite 4 release 4.1 stamp 63214 Authorization: Digest username="100",realm="asterisk",nonce="0ec22756",uri="sip:101@10.128.15.201",response="51fee875756120228a8ba4f57cf0bbd9",algorithm=MD5 Content-Length: 312 <?xml version='1.0' encoding='UTF-8'?> <isComposing xmlns='urn:ietf:params:xml:ns:im-iscomposing' xmlns:xsi='http://www.w3.org/2001/XMLSchema-instance'> <state>active</state> <lastactive>2012-02-06T15:08:35Z</lastactive> <contenttype>text/html</contenttype> <refresh>60</refresh> </isComposing> <-------------> --- (12 headers 3 lines) --- Receiving message! <--- Transmitting (NAT) to 10.128.19.201:45024 ---> SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP 10.128.19.201:45024;branch=z9hG4bK-d8754z-56b65d195534b2fd-1---d8754z-;received=10.128.19.201;rport=45024 From: "100"<sip:100@10.128.15.201>;tag=84f8fcf5 To: "101"<sip:101@10.128.15.201>;tag=as73a45a25 Call-ID: MGIyM2ZkOTVlYzhiNzJmYzY2Yzk3YWNjMzk2ZjMxZDM. CSeq: 15 MESSAGE Server: Asterisk PBX 10.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'MGIyM2ZkOTVlYzhiNzJmYzY2Yzk3YWNjMzk2ZjMxZDM.' in 32000 ms (Method: MESSAGE) <--- SIP read from UDP:10.128.17.123:5060 ---> jaK <-------------> <--- SIP read from UDP:10.128.19.201:45024 ---> MESSAGE sip:101@10.128.15.201 SIP/2.0 Via: SIP/2.0/UDP 10.128.19.201:45024;branch=z9hG4bK-d8754z-9bf927330b3b7ccf-1---d8754z-;rport Max-Forwards: 70 To: "101"<sip:101@10.128.15.201> From: "100"<sip:100@10.128.15.201>;tag=84f8fcf5 Call-ID: MGIyM2ZkOTVlYzhiNzJmYzY2Yzk3YWNjMzk2ZjMxZDM. CSeq: 16 MESSAGE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: text/plain User-Agent: X-Lite 4 release 4.1 stamp 63214 Authorization: Digest username="100",realm="asterisk",nonce="0ec22756",uri="sip:101@10.128.15.201",response="51fee875756120228a8ba4f57cf0bbd9",algorithm=MD5 Content-Length: 4 toto <-------------> --- (12 headers 1 lines) --- Receiving message! Looking for 101 in messages (domain 10.128.15.201) <--- Transmitting (NAT) to 10.128.19.201:45024 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.128.19.201:45024;branch=z9hG4bK-d8754z-9bf927330b3b7ccf-1---d8754z-;received=10.128.19.201;rport=45024 From: "100"<sip:100@10.128.15.201>;tag=84f8fcf5 To: "101"<sip:101@10.128.15.201>;tag=as06ec8b7e Call-ID: MGIyM2ZkOTVlYzhiNzJmYzY2Yzk3YWNjMzk2ZjMxZDM. CSeq: 16 MESSAGE Server: Asterisk PBX 10.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'MGIyM2ZkOTVlYzhiNzJmYzY2Yzk3YWNjMzk2ZjMxZDM.' in 32000 ms (Method: MESSAGE)    -- Executing [101@messages:1] NoOp("Message/ast_msg_queue", "Send message") in new stack    -- Executing [101@messages:2] Set("Message/ast_msg_queue", "MESSAGE(from)=sip:100@10.128.15.201") in new stack    -- Executing [101@messages:3] NoOp("Message/ast_msg_queue", "sip:101@10.128.15.201") in new stack    -- Executing [101@messages:4] MessageSend("Message/ast_msg_queue", "sip:101") in new stack Reliably Transmitting (NAT) to 10.128.17.123:5060: MESSAGE sip:101@10.128.17.123;line=ee515fb4035ac82 SIP/2.0 Via: SIP/2.0/UDP 10.128.15.201:5060;branch=z9hG4bK5d72134c;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@10.128.15.201>;tag=as209f3717 To: <sip:101@10.128.17.123;line=ee515fb4035ac82> Contact: <sip:asterisk@10.128.15.201:5060> Call-ID: 3103338d0232e9a352f8acbb5fe30749@127.0.1.1:0 CSeq: 102 MESSAGE User-Agent: Asterisk PBX 10.1.0 Content-Type: text/plain;charset=UTF-8 Content-Length: 4 toto --- Scheduling destruction of SIP dialog '3103338d0232e9a352f8acbb5fe30749@127.0.1.1:0' in 6400 ms (Method: MESSAGE)    -- Auto fallthrough, channel 'Message/ast_msg_queue' status is 'UNKNOWN' <--- SIP read from UDP:10.128.17.123:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.128.15.201:5060;branch=z9hG4bK5d72134c;rport=5060 From: "asterisk" <sip:asterisk@10.128.15.201>;tag=as209f3717 To: <sip:101@10.128.17.123;line=ee515fb4035ac82>;tag=1454076274 Call-ID: 3103338d0232e9a352f8acbb5fe30749@127.0.1.1:0 CSeq: 102 MESSAGE User-Agent: Linphone/3.5.0 (eXosip2/3.3.0) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '3103338d0232e9a352f8acbb5fe30749@127.0.1.1:0' Method: MESSAGE <--- SIP read from UDP:10.128.17.123:5060 ---> MESSAGE sip:asterisk@10.128.15.201 SIP/2.0 Via: SIP/2.0/UDP 10.128.17.123:5060;rport;branch=z9hG4bK1860991468 From: <sip:101@10.128.15.201>;tag=8878175 To: "asterisk" <sip:asterisk@10.128.15.201> Call-ID: 112815453 CSeq: 20 MESSAGE Content-Type: text/plain Max-Forwards: 70 User-Agent: Linphone/3.5.0 (eXosip2/3.3.0) Content-Length: 3 fgf <-------------> --- (10 headers 1 lines) --- Receiving message! Looking for asterisk in messages (domain 10.128.15.201)    -- Executing [asterisk@messages:1] NoOp("Message/ast_msg_queue", "Send message") in new stack <--- Transmitting (NAT) to 10.128.17.123:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 10.128.17.123:5060;branch=z9hG4bK1860991468;received=10.128.17.123;rport=5060 From: <sip:101@10.128.15.201>;tag=8878175 To: "asterisk" <sip:asterisk@10.128.15.201>;tag=as21ad434a Call-ID: 112815453 CSeq: 20 MESSAGE Server: Asterisk PBX 10.1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------>    -- Executing [asterisk@messages:2] Set("Message/ast_msg_queue", "MESSAGE(from)=sip:101@10.128.15.201") in new stack Scheduling destruction of SIP dialog '112815453' in 32000 ms (Method: MESSAGE)    -- Executing [asterisk@messages:3] NoOp("Message/ast_msg_queue", "sip:asterisk@10.128.15.201") in new stack    -- Executing [asterisk@messages:4] MessageSend("Message/ast_msg_queue", "sip:asterisk") in new stack [Feb  6 16:09:13] ERROR[23615]: netsock2.c:263 ast_sockaddr_resolve: getaddrinfo("asterisk", "(null)", ...): Temporary failure in name resolution [Feb  6 16:09:13] WARNING[23615]: chan_sip.c:5469 create_addr: No such host: asterisk Really destroying SIP dialog '12d15cbf1fdd69e17de47526728c4026@127.0.1.1:0' Method: MESSAGE    -- Auto fallthrough, channel 'Message/ast_msg_queue' status is 'UNKNOWN' To make the answer working i do this in the extension.conf exten => _[a-z0-9].,1,NoOp(Send message) exten => _[a-z0-9].,n,NoOp(${SIPFROMUSER}) exten => _[a-z0-9].,n,Set(SIPFROMUSER=${MESSAGE(from):4:-14}) exten => _[a-z0-9].,n,MessageSend(${MESSAGE(to):0:-14},${SIPFROMUSER})
Comments:	By: Richard Mudgett (rmudgett) 2012-02-06 12:18:31.445-0600 What you are showing is duplicated by ASTERISK-18992 and ASTERISK-18917.  Those fixes will be in Asterisk v10.2.0. How to use questions are best discussed on the asterisk-users mailing list or IRC #asterisk. The MessageSend(from) parameter accepts the following: known peer name or "display-name" <uri> Currently Asterisk only accepts incoming MESSAGE's of Content-Type text/plain.