Hi, Thanks a lot for this detailed answer :
- I managed to have it working disabling auth message request : auth_message_requests = no in sip.conf - pedantic=no does not resolve the issue - reenabling auth_message_requests = yes and removing pedantic option, your patch in chan_sip resolves the issues ! As it looks like pidgin has an issue, I guess that we can use it as a workaround. I would like know to enable presence notification between each users. To fulfill it, I am using http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceStates_id265377.html Am I doing it in a good way ? Thanks ! Eloi On Wed, Jun 19, 2013 at 12:11 PM, Matthew J. Roth <mr...@imminc.com> wrote: > 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 >
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