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

Matthew Boehm mboehm at cytelcom.com
Sun Mar 13 19:10:28 MST 2005


On the "no compatible codecs" error, do a "sip show peer 621" and see what
codecs it has listed.

For the changes: when you do a "make update" there should be new copies of
sample configs inside asterisk/configs/ that you can read through.

-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 09:27:52 +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:
> 
>> 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
>>  
>> 
> Matthew,
> 
> I came a big step further!
> I rebooted the Grandstream and now I get both: sip show users/peers
> However, it is still not working ;-(
> 
> Calling to 621 (Grandstream) is working, but from 621 will give me in
> *CLI a:
> Mar 14 09:03:30 NOTICE[29502]: chan_sip.c:2917 process_sdp: No
> compatible codecs!
> 
> I used these working settings of the the sip.conf to create the database
> record:
> 
> ; Test phone set 621 (Grandstream BudgeTone 101)
> [621]  
> type=friend
> username=621
> secret=Password
> nat=yes
> host=dynamic
> context=inhouse  
> canreinvite=yes
> disallow=all
> allow=ulaw
> allow=alaw
> dtmfmode=rfc2833
> qualify=1000
> mailbox=621 at other
> pickupgroup=1
> 
> After changing the nat field from int(1) to varchar(5) I used the
> following script to create this table with one record again:
> DROP TABLE sip_buddies;
> CREATE TABLE sip_buddies (
>   id int(11) NOT NULL auto_increment,
>   name varchar(80) NOT NULL default '',
>   accountcode varchar(20) default NULL,
>   amaflags varchar(7) default NULL,
>   callgroup varchar(10) default NULL,
>   callerid varchar(80) default NULL,
>   canreinvite char(3) default 'yes',
>   context varchar(80) default NULL,
>   defaultip varchar(15) default NULL,
>   dtmfmode varchar(7) default NULL,
>   fromuser varchar(80) default NULL,
>   fromdomain varchar(80) default NULL,
>   host varchar(31) NOT NULL default '',
>   incominglimit int(2) default NULL,
>   outgoinglimit int(2) default NULL,
>   insecure varchar(4) default NULL,
>   language char(2) default NULL,
>   mailbox varchar(50) default NULL,
>   md5secret varchar(80) default NULL,
>   nat varchar(5) NOT NULL default 'yes',
>   permit varchar(95) default NULL,
>   deny varchar(95) default NULL,
>   mask varchar(95) default NULL,
>   pickupgroup varchar(10) default NULL,
>   port varchar(5) NOT NULL default '',
>   qualify char(3) default NULL,
>   restrictcid char(1) default NULL,
>   rtptimeout char(3) default NULL,
>   rtpholdtimeout char(3) default NULL,
>   secret varchar(80) default NULL,
>   type varchar(6) NOT NULL default 'friend',
>   username varchar(80) NOT NULL default '',
>   allow varchar(100) default 'g729;ilbc;gsm;ulaw;alaw',
>   disallow varchar(100) default 'all',
>   musiconhold varchar(100) default NULL,
>   regseconds int(11) NOT NULL default '0',
>   ipaddr varchar(15) NOT NULL default '',
>   cancallforward char(3) default 'yes',
>   PRIMARY KEY  (id),
>   UNIQUE KEY name (name),
>   KEY name_2 (name)
> ) TYPE=MyISAM ROW_FORMAT=DYNAMIC;
> 
> --
> -- Dumping data for table `sip_buddies`
> --
> 
> INSERT INTO sip_buddies VALUES
> (1,'621',NULL,NULL,NULL,'\"Demo\",<621>','yes','inhouse',NULL,'rfc2833',NULL,N
> ULL,'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,'','y
> es');
> 
> since the qualify field is only 3 characters, I changed it from 1000 to
> 999. Could I change the field length to 4 characters, to get 1000 in
> again, without breaking it on another place?
> 
> and:
> 
>>> 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
>>>> 
>>>>      
>>>> 
> BTW, is there an easy way to find out what has changed for the config files?
> 
> 
> 
> bye
> 
> Ronald
> 
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