[Asterisk-Users] Realtime does not work yet, ...

[Matthew Boehm]

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 <ronald at elmit.com>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Date: Mon, 14 Mar 2005 00:42:07 +0800
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> 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     |
> 621 at other | 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:601 at 61.220.121.190:5060;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:601 at 61.220.121.190:5060;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
> 
> 
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