[SR-Users] help to fix error in carrierroute module.
marius zbihlei
marius.zbihlei at 1and1.ro
Mon Aug 23 12:33:46 CEST 2010
truong ngoc THANH wrote:
> hi all,
> please help to correct configure in carrierroute module.
> i have configure carrierroute in kamailio as bellow.
> =========
Hello,
First of all let me tell you that the e-mail is fairly long and hard to
follow. Try to break the problem into smaller steps and better pin-point
the issue to a particular step. After reading your mail I am not certain
that this is a carrierroute problem
> kamailio.cfg
> ===
> route[1] {
> # route calls based on hash over callid
> # choose route domain 0 of the default carrier
> if(!cr_route("default", "default", "$rU", "$rU", "call_id")){
> sl_send_reply("403", "Not allowed");
> } else {
> # In case of failure, re-route the request
> t_on_failure("1");
> # Relay the request to the gateway
> t_relay();
> }
> }
>
The block above looke ok
> failure_route[1] {
> # In case of failure, send it to an alternative route:
> if (t_check_status("408|5[0-9][0-9]")) {
>
> #choose route domain 1 of the default carrier
> if(!cr_route("default", "default", "$rU", "$rU", "call_id")){
> } else {
> xlog("failure 2");
> t_on_failure("2");
> t_relay();
> }
> }
> }
>
>
Here some problems seem to appear (in failure_route[1]). The
carrierroute module is used for balanceing, least cost routing etc. If
you have only one route (as I can see in the table dump) then the
purpose of the module is gone. If you look at the cr_route statement in
the failure_route[1] you will see that it is identical to the one in
route[1] so the R-URI will be rewritten with the same SIP URI. You
should have another entry in the carrierroute table with domain 2
carrier default pointing to another sip uri and then call
cr_route("default", "2", "$rU", "$rU", "$call_id"). And check again the
docs as the revert_uri() call is important
(http://www.kamailio.org/docs/modules/1.5.x/carrierroute.html#id2976981)
Please make this changes and try again. Also keep in mind 2 things
1. the ngrep you provided look broken (try ngrep -W byline -d any sip
port 5060)
2. the carrierroute module only rewrites R-URI, witch according to the
dump you provided it did.(INVITE sip:1000 at 192.168.1.42 SIP/2.0.) Maybe
the problem is somewhere else. Please use an incremental approach on
solving this problem
Cheers
Marius
> =========\
> in database:
> mysql> select * from carrierroute;
> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
> | id | carrier | domain | scan_prefix | flags | mask | prob | strip |
> rewrite_host | rewrite_prefix | rewrite_suffix | description |
> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
> | 1 | 1 | 1 | 1000 | 0 | 0 | 1 | 0 |
> 192.168.1.42 | | | FRANCE |
> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
> 1 row in set (0.00 sec)
> mysql> select * from carrier_name;
> +----+---------+
> | id | carrier |
> +----+---------+
> | 1 | default |
> +----+---------+
> mysql> select * from domain_name;
> +----+---------+
> | id | domain |
> +----+---------+
> | 1 | default |
> +----+---------+
> ================
> in server 192.168.1.40 : subscriber table:
> mysql> select * from subscriber;
> +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+
> | id | username | domain | password | email_address |
> ha1 | ha1b |
> rpid |
> +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+
> | 1 | 1 | 192.168.1.40 | 1 | |
> b8ee4a7080092d7fd27357de1799b111 | d1343e4f0259199ed6b06d6338d39903 |
> NULL |
> | 2 | 2 | 192.168.1.40 | 2 | |
> 25f61f90a19afe508629127838c11509 | 7f4ec87f39f48d0d8f8a537cfc34c5c9 |
> NULL |
> | 3 | 3 | kamailio.com <http://kamailio.com/> | 3
> | | 63a131bf02b4123073970731a4525470 |
> 64e49a4f89317c5ee757ba0366ec9307 | NULL |
> | 4 | 1000 | 192.168.1.40 | 1000 | |
> 8e9f00da8e49cc8b844b6cee1752b949 | 2bff4783b0ce3412bf33e7400fbbedce |
> NULL |
> | 5 | 4 | 192.168.1.40 | 4 | |
> 037cf4277ce31c104e4b1bc7bbcf8ad4 | cb2321d9ac8bfe749cedc1d66cad5d73 |
> NULL |
> | 6 | 1 | kamailio.com | 1 | |
> 31fb6891a2fcb9b13a952ce3f12359c9 | 84add6cffd100f8ca3c8660c3c9617b5 |
> NULL |
> | 7 | 1 | 192.168.1.42 | 1 | |
> ad57b2abdf288984bc4a897a9e7f2ba9 | 60b9ab690724e5a535df794131e2345f |
> NULL |
> | 8 | 1000 | 192.168.1.42 | 1000 | |
> 0a9fb3a1de135a32b9090798a0dcb240 | bf5a95934a59b480dbf85abe386144b6 |
> NULL |
> +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+
>
> I try to make call from 1 at 192.168.1.40 to 1000 at 192.168.1.42
> but the call failed.
> sip client 1 at 192.168.1.40 show that : server failure
> and sip client 1000 at 192.168.1.42 ringing.
> ============
> sip trace for 192.168.1.40:
>
> U 192.168.1.40:5060 -> 192.168.1.42:5060
> INVITE sip:1000 at 192.168.1.42 SIP/2.0.
> Record-Route: <sip:192.168.1.40;lr=on>.
> Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKf92a.649f39e3.0.
> Via: SIP/2.0/UDP
> 192.168.1.10:50495;branch=z9hG4bK-d8754z-ec188605061deb73-1---d8754z-;rport=50495.
> Max-Forwards: 69.
> Contact: <sip:1 at 192.168.1.10:50495;rinstance=9a88ebaf62e2a935>.
> To: <sip:1000 at 192.168.1.40:5060>.
> From: <sip:1 at 192.168.1.40:5060>;tag=fd3fe92e.
> Call-ID: MmNjYTZjMjNmOWZkN2RkZWE3NWUxOWZlNWUyOTY3ZmM..
> CSeq: 2 INVITE.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
> REFER, INFO.
> Content-Type: application/sdp.
> Supported: replaces.
> User-Agent: 3CXPhone 4.0.10858.0.
> Content-Length: 279.
> .
> v=0.
> o=3cxVCE 96031755 384028260 IN IP4 192.168.1.10.
> s=3cxVCE Audio Call.
> c=IN IP4 192.168.1.10.
> t=0 0.
> m=audio 40002 RTP/AVP 0 8 3 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
>
>
> U 192.168.1.40:5060 -> 192.168.1.10:50495
> SIP/2.0 500 I'm terribly sorry, server error occurred (6/SL).
> Via: SIP/2.0/UDP
> 192.168.1.10:50495;branch=z9hG4bK-d8754z-ec188605061deb73-1---d8754z-;rport=50495.
> To:
> <sip:1000 at 192.168.1.40:5060>;tag=ae07808f44ebda0182a54cf170e36dbe.ba60.
> From: <sip:1 at 192.168.1.40:5060>;tag=fd3fe92e.
> Call-ID: MmNjYTZjMjNmOWZkN2RkZWE3NWUxOWZlNWUyOTY3ZmM..
> CSeq: 2 INVITE.
> Server: kamailio (3.0.1 (i386/linux)).
> Content-Length: 0.
> .
>
>
> U 192.168.1.10:50495 -> 192.168.1.40:5060
> ACK sip:1000 at 192.168.1.40:5060 SIP/2.0.
> Via: SIP/2.0/UDP
> 192.168.1.10:50495;branch=z9hG4bK-d8754z-ec188605061deb73-1---d8754z-;rport.
> Max-Forwards: 70.
> To:
> <sip:1000 at 192.168.1.40:5060>;tag=ae07808f44ebda0182a54cf170e36dbe.ba60.
> From: <sip:1 at 192.168.1.40:5060>;tag=fd3fe92e.
> Call-ID: MmNjYTZjMjNmOWZkN2RkZWE3NWUxOWZlNWUyOTY3ZmM..
> CSeq: 2 ACK.
> Content-Length: 0.
>
>
> and sip trace in 192.168.1.42:
>
> ====
> U 192.168.1.40:5060 -> 192.168.1.42:5060
> INVITE sip:1000 at 192.168.1.42 SIP/2.0.
> Record-Route: <sip:192.168.1.40;lr=on>.
> Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKc4e.5ee44227.0.
> Via: SIP/2.0/UDP
> 192.168.1.10:50495;branch=z9hG4bK-d8754z-7e347d55637cf134-1---d8754z-;rport=50495.
> Max-Forwards: 69.
> Contact: <sip:1 at 192.168.1.10:50495;rinstance=9a88ebaf62e2a935>.
> To: <sip:1000 at 192.168.1.40:5060>.
> From: <sip:1 at 192.168.1.40:5060>;tag=722dc40c.
> Call-ID: ZGY0YTY5YmYzM2E2MzEzNDgzODhkMTllN2RiY2ExOGM..
> CSeq: 2 INVITE.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
> REFER, INFO.
> Content-Type: application/sdp.
> Supported: replaces.
> User-Agent: 3CXPhone 4.0.10858.0.
> Content-Length: 280.
> .
> v=0.
> o=3cxVCE 227296140 367695825 IN IP4 192.168.1.10.
> s=3cxVCE Audio Call.
> c=IN IP4 192.168.1.10.
> t=0 0.
> m=audio 40000 RTP/AVP 0 8 3 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
>
>
> U 192.168.1.42:5060 -> 192.168.1.40:5060
> SIP/2.0 100 trying -- your call is important to us.
> Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKc4e.5ee44227.0.
> Via: SIP/2.0/UDP
> 192.168.1.10:50495;branch=z9hG4bK-d8754z-7e347d55637cf134-1---d8754z-;rport=50495.
> To: <sip:1000 at 192.168.1.40:5060>.
> From: <sip:1 at 192.168.1.40:5060>;tag=722dc40c.
> Call-ID: ZGY0YTY5YmYzM2E2MzEzNDgzODhkMTllN2RiY2ExOGM..
> CSeq: 2 INVITE.
> Server: kamailio (3.0.1 (i386/linux)).
> Content-Length: 0.
> .
>
>
> U 192.168.1.42:5060 -> 192.168.1.10:6872
> INVITE sip:1000 at 192.168.1.10:6872;rinstance=c308344d9524d1a4 SIP/2.0.
> Record-Route: <sip:192.168.1.42;lr=on>.
> Record-Route: <sip:192.168.1.40;lr=on>.
> Via: SIP/2.0/UDP 192.168.1.42;branch=z9hG4bKc4e.f9fad397.0.
> Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKc4e.5ee44227.0.
> Via: SIP/2.0/UDP
> 192.168.1.10:50495;branch=z9hG4bK-d8754z-7e347d55637cf134-1---d8754z-;rport=50495.
> Max-Forwards: 68.
> Contact: <sip:1 at 192.168.1.10:50495;rinstance=9a88ebaf62e2a935>.
> To: <sip:1000 at 192.168.1.40:5060>.
> From: <sip:1 at 192.168.1.40:5060>;tag=722dc40c.
> Call-ID: ZGY0YTY5YmYzM2E2MzEzNDgzODhkMTllN2RiY2ExOGM..
> CSeq: 2 INVITE.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY,
> REFER, INFO.
> Content-Type: application/sdp.
> Supported: replaces.
> User-Agent: 3CXPhone 4.0.10858.0.
> Content-Length: 280.
> .
> v=0.
> o=3cxVCE 227296140 367695825 IN IP4 192.168.1.10.
> s=3cxVCE Audio Call.
> c=IN IP4 192.168.1.10.
> t=0 0.
> m=audio 40000 RTP/AVP 0 8 3 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
>
>
> U 192.168.1.10:6872 -> 192.168.1.42:5060
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP 192.168.1.42;branch=z9hG4bKc4e.f9fad397.0.
> Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKc4e.5ee44227.0.
> Via: SIP/2.0/UDP
> 192.168.1.10:50495;branch=z9hG4bK-d8754z-7e347d55637cf134-1---d8754z-;rport=50495.
> Record-Route: <sip:192.168.1.42;lr>.
> Record-Route: <sip:192.168.1.40;lr=on>.
> Contact: <sip:1000 at 192.168.1.10:6872;rinstance=c308344d9524d1a4>.
> To: <sip:1000 at 192.168.1.40:5060>;tag=660be63a.
> From: <sip:1 at 192.168.1.40:5060>;tag=722dc40c.
> Call-ID: ZGY0YTY5YmYzM2E2MzEzNDgzODhkMTllN2RiY2ExOGM..
> CSeq: 2 INVITE.
> User-Agent: PortGo v6.0, Build 07282010.
> Content-Length: 0.
>
>
>
>
> ========================
> and in server 192.168.1.42:subscriber table:
> mysql> select * from subscriber;
> +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+
> | id | username | domain | password | email_address |
> ha1 | ha1b |
> rpid |
> +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+
> | 1 | 1000 | 192.168.1.42 | 1000 | |
> 0a9fb3a1de135a32b9090798a0dcb240 | bf5a95934a59b480dbf85abe386144b6 |
> NULL |
> | 2 | 1 | 192.168.1.42 | 1 | |
> ad57b2abdf288984bc4a897a9e7f2ba9 | 60b9ab690724e5a535df794131e2345f |
> NULL |
> | 3 | 1 | 192.168.1.40 | 1 | |
> b8ee4a7080092d7fd27357de1799b111 | d1343e4f0259199ed6b06d6338d39903 |
> NULL |
>
>
> please suggest to get correct configue.
> thanks for looking.
>
> TRUONG NGOC THANH
> Telecommunications Engineer
> Tel: 0984 480 646 begin_of_the_skype_highlighting 0984
> 480 646 end_of_the_skype_highlighting
> Y!M: ngoc217thanh
>
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