SIP trunking without fromuser and register

stanimir
Posts: 44
Member Since:
2007-01-31

Hello,

I'm using Asterisk Realtime SIP with PstgreSQL database. I have SQL table sip_conf with the following format:

    CREATE TABLE sip_conf
    (
    id serial NOT NULL,
    "name" character varying(80) NOT NULL DEFAULT ''::character varying,
    host character varying(31) NOT NULL DEFAULT ''::character varying,
    nat character varying(5) NOT NULL DEFAULT 'no'::character varying,
    "type" character varying(255) NOT NULL DEFAULT 'friend'::character varying,
    accountcode character varying(20),
    amaflags character varying(13),
    callgroup character varying(10),
    callerid character varying(80),
    cancallforward character(3) DEFAULT 'yes'::bpchar,
    canreinvite character(6) DEFAULT 'yes'::bpchar,
    context character varying(80),
    defaultip character varying(15),
    dtmfmode character varying(7),
    fromuser character varying(80),
    fromdomain character varying(80),
    insecure character varying(10),
    "language" character(2),
    mailbox character varying(50),
    md5secret character varying(80),
    deny character varying(95),
    permit character varying(95),
    mask character varying(95),
    musicclass character varying(100),
    pickupgroup character varying(10),
    qualify character(3),
    regexten character varying(80),
    restrictcid character(3),
    rtptimeout character(3),
    rtpholdtimeout character(3),
    secret character varying(80),
    setvar character varying(100),
    disallow character varying(100) DEFAULT 'all'::character varying,
    allow character varying(100) DEFAULT 'alaw;ulaw;gsm;g729'::character varying,
    fullcontact character varying(80) NOT NULL DEFAULT ''::character varying,
    ipaddr character varying(15) NOT NULL DEFAULT ''::character varying,
    port character varying(5) NOT NULL DEFAULT 5060,
    regserver character varying(100),
    username character varying(80) NOT NULL DEFAULT ''::character varying,
    regseconds character varying(10),
    lastms character varying(10),
    siptype smallint DEFAULT 0,
    CONSTRAINT sip_conf_pkey PRIMARY KEY (id),
    CONSTRAINT sip_conf_type_check CHECK ("type"::text = 'user'::text OR "type"::text = 'peer'::text OR "type"::text = 'friend'::text)
    )
    WITHOUT OIDS;
    ALTER TABLE sip_conf OWNER TO asterisk;

I want to create separate trunks for incoming an outgoing calls from/to other asterisk servers. Each trunk has its own user and password. I don't want to use the register and fromuser options because:
register can't be used because I want to use only the SQL table
fromuser can't be used because I don't want to override the callerid

Here is an example of what I want to have (I'm not writing the exact syntax, only for illustration purpose):
------Server A------
[server_a_incoming_trunk]
username=serverA_user1
password=serverA_password1
context=incoming_context

[server_a_outgoing_trunk]
username=serverB_user1
password=serverB_password1
----------------------

------Server B------
[server_b_incoming_trunk]
username=serverB_user1
password=serverB_password1
context=incoming_context

[server_b_outgoing_trunk]
username=serverA_user1
password=serverA_password1
----------------------

So I want:
333@ServerB calls 123@ServerA through server_b_outgoing_trunk (e.g. _9X.,1,Dial(SIP/server_b_outgoing_trunk/${EXTEN:1},,Tt) )
extension 123 sees on the display '333' NOT 'serverA_user1' (which will be displayed if using fromuser in [server_b_outgoing_trunk] )

Regards
Stan



stanimir
Posts: 44
Member Since:
2007-01-31
Found solution for caller id

I found a way to send the original Caller ID.

For outgoing connection(peer) there should be senrtp=yes
For incoming connection(user) there should be trustrtp=yes

However I have other problem. I have qualify=yes and rtcachefriends=on but when I do "sip show peers" for status on my peers I see Unknown, they are working but somehow the qualify=yes isn't and it so only for trunks. SIP extensions are OK.



SkykingOH
Posts: 9680
Member Since:
2007-12-17
For those watching those

For those watching those variables are sendrpid and trust rpid. RTP is a streaming media protocol.

--

Scott

aka "Skyking"



stanimir
Posts: 44
Member Since:
2007-01-31
my mistake

Yes, SkykingOH is right. I was in hurry so I guess this is the reason why I did this stupid mistake

sendrpid=yes/no
trustrpid=yes/no



Comment viewing options

Select your preferred way to display the comments and click "Save settings" to activate your changes.