[Asterisk-Users] Realtime does not work yet, ...
Ronald Wiplinger
ronald at elmit.com
Sun Mar 13 20:35:18 MST 2005
Matthew Boehm wrote:
>On the "no compatible codecs" error, do a "sip show peer 621" and see what
>codecs it has listed.
>
>
>
vpbx*CLI>
* Name : 621
Secret : <Set>
MD5Secret : <Not set>
Context : inhouse
Language :
AMA flags : Unknown
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup : 1, 33
Mailbox : 621 at other
LastMsgsSent : 2
Inc. limit : 0
Outg. limit : 0
Dynamic : Yes
Callerid : "Demo" <621>
Expire : 22361
Expiry : 900
Insecure : no
Nat : Always
ACL : No
CanReinvite : Yes
PromiscRedir : No
User=Phone : No
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : 192.168.250.114 Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 621
Codecs : 0x0 (nothing)
Codec Order : (none)
Status : OK (5 ms)
Useragent : Grandstream BT100 1.0.5.18
Full Contact : sip:621 at 203.70.36.26:65397
Indeed it does not have a Codecs and no Codec Order.
The table fields from the mysqldump is still below. Can you see what is
wrong?
>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.
>
>
>
thanks, I did not notice that!
bye
Ronald
>-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
>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>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
>Asterisk-Users at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
>
--
Ronald Wiplinger (CEO of ELMIT)
http://www.elmit.com +886 (0) 939--77-55-16 or FWD 511208
- I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org
PS: Spam prevention!
Our system is protected with a spam prevention program.
If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please.
After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again.
More information about the asterisk-users
mailing list