Eloi Bail wrote: > > I am trying to enable SIP SIMPLE communication in my test environment. > > I have the following env : > > - one server (192.168.50.126) with Asterisk 11 > - 2 clients using pidgin : demo-bob and demo-alice on my 192.168.50.143 > > I successfully had a phone call between clients. > > I used the following link to enable SIMPLE messaging between my clients : > http://highsecurity.blogspot.ca/2012/03/asterisk-10-110-sms-messaging-or-sip.html > > Both users managed to register. > > Adding verbose on the server, I have the following traces when I send the > message "MESSAGE FROM ALICE TO BOB" from "demo-alice" to "demo-bob" > > http://paste.fedoraproject.org/19489/37158861/ > > As you can see I succeed to have the message sent from alice to Asterisk. > > When the server is trying to transmitting, I see a 401 error message. > According to this post ( http://forums.digium.com/viewtopic.php?f=1&t=72814 ) > the first 401 should be normal as authentication is requested. > > Afterwards the server emit 202 message. > > But "demo-bob" never receives a message. > I ran wireshark on server and client. It confirms that no message is sent from > Asterisk to "demo-bob". > > Could you please give me advice ? > > Here are my extensions.conf and sip.conf according to the link I mentioned. > http://paste.fedoraproject.org/19626/16493741/ > > http://paste.fedoraproject.org/19627/49423137/
Eloi, The trace shows that the initial MESSAGE from Alice does not include an Authorization header so Asterisk responds with a 401 Unauthorized. Alice then replies with a MESSAGE with an Authorization header, but reuses the same CSeq header (CSeq: 6 MESSAGE) which causes Asterisk to ignore it as a retransmit: > [Jun 18 16:49:35] DEBUG[16266] chan_sip.c: Ignoring SIP message because of > retransmit (MESSAGE Seqno 6, ours 6) I believe this is a bug in Pidgin because RFC 3261 [1] states: CSeq or Command Sequence contains an integer and a method name. The CSeq number is incremented for each new request within a dialog and is a traditional sequence number. ... Requests within a dialog MUST contain strictly monotonically increasing and contiguous CSeq sequence numbers (increasing-by-one) in each direction (excepting ACK and CANCEL of course, whose numbers equal the requests being acknowledged or cancelled). However, there is also a similar issue [2] that can be worked around by setting "pedantic=no" in sip.conf. If that doesn't work, you can give the following (untested) patch to chan_sip.c a try: ================================================================================ --- chan_sip.c.orig 2013-06-19 11:44:38.000000000 -0400 +++ chan_sip.c 2013-06-19 11:47:22.000000000 -0400 @@ -28078,6 +28078,7 @@ } else if (p->icseq && p->icseq == seqno && req->method != SIP_ACK && + p->method != SIP_MESSAGE && (p->method != SIP_CANCEL || p->alreadygone)) { /* ignore means "don't do anything with it" but still have to respond appropriately. We do this if we receive a repeat of ================================================================================ Good luck and please let the list know how this works out. [1] http://www.ietf.org/rfc/rfc3261.txt [2] https://issues.asterisk.org/jira/browse/ASTERISK-19139 Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users