Are you sure that NAT is set correctly everywhere? I sometimes forget to set the phone to be NAT aware.
That is weird that 'sip show peers/users' doesn't show the phone both times. Have you stopped/started asterisk since these changes? Do it again just to make sure. The only thing I can say is that this works in our office. Asterisk is on public IP while phones are all inside private network, NAT'd to outside. -Matthew > From: Ronald Wiplinger <[EMAIL PROTECTED]> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Date: Mon, 14 Mar 2005 00:42:07 +0800 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Subject: Re: [Asterisk-Users] Realtime does not work yet, ... > > Matthew Boehm wrote: > >> You may not have most recent CVS. You should have this in your sip.conf: >> >> >> > You are right, ... but the sip.conf will not be updated anyway, if I do > not want to loose all my settings. > >> rtcachefriends=yes >> ; Cache realtime friends by adding them to the internal list >> ; just like friends added from the config file only on a >> ; as-needed basis. >> >> rtnoupdate=yes >> ; do not send the update request over realtime. >> >> rtautoclear=yes >> ; Auto-Expire friends created on the fly on the same schedule >> ; as if it had just registered when the registration expires >> ; the friend will vanish from the configuration until requested >> ; again. If set to an integer, friends expire >> ; within this number of seconds instead of the >> ; same as the registration interval >> >> NAT should be VARCHAR(5) >> >> >> > I have added the three variables and changed the table to varchar(5) > >> If everything works fine when UA's are defined in sip.conf then there is >> most likely a db data issue. Try changing NAT as above. Be sure to use "yes" >> or "no". >> >> >> > > Now I cannot dial in neither direction. CLI shows: > > Connected to Asterisk CVS-HEAD-03/13/05-23:38:12 currently running on > vpbx (pid = 29502) > Verbosity is at least 3 > -- Executing Dial("SIP/601-6540", "SIP/621|60|Ttrm") in new stack > Mar 14 00:24:45 NOTICE[29502]: app_dial.c:936 dial_exec_full: Unable to > create channel of type 'SIP' (cause 3) > == Everyone is busy/congested at this time (1:0/1/0) > -- Executing VoiceMail("SIP/601-6540", "u621") in new stack > -- Playing 'vm-theperson' (language 'en') > -- Playing 'digits/6' (language 'en') > -- Playing 'digits/2' (language 'en') > -- Playing 'digits/1' (language 'en') > -- Playing 'vm-isunavail' (language 'en') > == Spawn extension (default, 621, 2) exited non-zero on 'SIP/601-6540' > -- Executing Hangup("SIP/601-6540", "") in new stack > == Spawn extension (default, h, 1) exited non-zero on 'SIP/601-6540' > vpbx*CLI> > vpbx*CLI> > vpbx*CLI> > vpbx*CLI> > Mar 14 00:25:05 NOTICE[29502]: chan_sip.c:2917 process_sdp: No > compatible codecs! > > > First case is 601 dials to 621, second case 621 dials to 601 > mysql> select * from sip_buddies; > +----+------+-------------+----------+-----------+--------------+------------- > +---------+-----------+----------+----------+------------+---------+---------- > -----+---------------+----------+----------+-----------+-----------+-----+---- > ----+------+------+-------------+------+---------+-------------+------------+- > ---------------+-----------+--------+----------+-----------+----------+------- > ------+------------+--------+----------------+ > | id | name | accountcode | amaflags | callgroup | callerid | > canreinvite | context | defaultip | dtmfmode | fromuser | fromdomain | > host | incominglimit | outgoinglimit | insecure | language | > mailbox | md5secret | nat | permit | deny | mask | pickupgroup | port > | qualify | restrictcid | rtptimeout | rtpholdtimeout | secret | > type | username | allow | disallow | musiconhold | regseconds | > ipaddr | cancallforward | > +----+------+-------------+----------+-----------+--------------+------------- > +---------+-----------+----------+----------+------------+---------+---------- > -----+---------------+----------+----------+-----------+-----------+-----+---- > ----+------+------+-------------+------+---------+-------------+------------+- > ---------------+-----------+--------+----------+-----------+----------+------- > ------+------------+--------+----------------+ > | 1 | 621 | NULL | NULL | NULL | "Demo",<621> | > yes | inhouse | NULL | rfc2833 | NULL | NULL | > dynamic | NULL | NULL | NULL | NULL | > [EMAIL PROTECTED] | NULL | yes | NULL | NULL | NULL | 1 | > | 999 | NULL | NULL | NULL | Password | > friend | 621 | ulaw;alaw | all | NULL | 0 > | | yes | > +----+------+-------------+----------+-----------+--------------+------------- > +---------+-----------+----------+----------+------------+---------+---------- > -----+---------------+----------+----------+-----------+-----------+-----+---- > ----+------+------+-------------+------+---------+-------------+------------+- > ---------------+-----------+--------+----------+-----------+----------+------- > ------+------------+--------+----------------+ > 1 row in set (0.00 sec) > > > The first case has in debug: > Mar 14 00:23:38 DEBUG[29502]: build_route: Contact hop: > <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> > Mar 14 00:23:38 DEBUG[29502]: MySQL RealTime: Retrieve SQL: SELECT * > FROM sip_buddies WHERE name = '621' > Mar 14 00:23:38 DEBUG[29502]: MySQL RealTime: Everything is fine. > Mar 14 00:23:38 DEBUG[29502]: Unable to find key '621' in family > 'SIP/Registry' > Mar 14 00:23:38 DEBUG[29502]: Setting NAT on RTP to 524288 > Mar 14 00:23:38 DEBUG[29502]: Exiting with DIALSTATUS=CONGESTION. > > Can somebody explain what it means "Unabble to find key '621' in family > 'SIP/Registry' ? > > The second case has in debug: > > Mar 14 00:24:45 DEBUG[29502]: Check for res for 601 > Mar 14 00:24:45 DEBUG[29502]: build_route: Contact hop: > <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> > Mar 14 00:24:45 DEBUG[29502]: Setting NAT on RTP to 524288 > Mar 14 00:24:45 DEBUG[29502]: Exiting with DIALSTATUS=CONGESTION. > > > sip show users shows 621/621 while sip show peers does not show 621 > > > > bye > > Ronald > > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
