[SR-Users] Kamailio Does NOT Forward Registration Requests To Asterisk.
Mahmoud Ramadan Ali
cisco.and.more.blog at gmail.com
Tue Nov 18 03:20:08 CET 2014
Hi Mohamed,
Thank you for your interest in helping me,I've configured the the auth_db
module with the Asterisk DB URL and the SIP username and password table
name and verified the MYSQL remote connection from Kamailio to the Asterisk
DB and get connected as predicted.
I tried to register a phone after applying the changes and Kamailio
forwarded the register request to Asterisk only once and without successful
authentication ! now i didn't change anything in the configuration file and
can NOT get any registration requests forwarded from Kamailio to Asterisk
and get only events on Kamailio that it can NOT register the incoming
registration request like this.
root at debian:/usr/local/etc/kamailio# ngrep -W byline -d eth1 port 5060
U 192.168.50.2:50886 -> 192.168.50.1:5060
REGISTER sip:192.168.50.1 SIP/2.0.
Via: SIP/2.0/UDP 192.168.50.2:50886
;branch=z9hG4bK-d8754z-cb65023b979d0a36-1---d8754z-;rport.
Max-Forwards: 70.
Contact: <sip:1001 at 192.168.50.2:50886;rinstance=8000799665fa4b54>.
To: "Mahmoud Ramadan Ali"<sip:1001 at 192.168.50.1>.
From: "Mahmoud Ramadan Ali"<sip:1001 at 192.168.50.1>;tag=9f381b5f.
Call-ID: MzcxNzYwMmUyN2E0M2FkMWRmOTI0ZjNkMjJmNWNhYTc.
CSeq: 2 REGISTER.
Expires: 3600.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO.
User-Agent: X-Lite 4.7.1 74247--W6.1.
Authorization: Digest
username="1001",realm="192.168.50.1",nonce="VGqbxVRqmpngschsiE6AuMiOfCS/MIp7",uri="sip:192.168.50.1",response="1788f6b9cfc322b863a93c91f3b623dc",algorithm=MD5.
Content-Length: 0.
#
U 192.168.50.1:5060 -> 192.168.50.2:50886
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 192.168.50.2:50886
;branch=z9hG4bK-d8754z-cb65023b979d0a36-1---d8754z-;rport=50886.
To: "Mahmoud Ramadan Ali"<sip:1001 at 192.168.50.1
>;tag=b27e1a1d33761e85846fc98f5f3a7e58.0bcb.
From: "Mahmoud Ramadan Ali"<sip:1001 at 192.168.50.1>;tag=9f381b5f.
Call-ID: MzcxNzYwMmUyN2E0M2FkMWRmOTI0ZjNkMjJmNWNhYTc.
CSeq: 2 REGISTER.
WWW-Authenticate: Digest realm="192.168.50.1",
nonce="VGqbxVRqmpngschsiE6AuMiOfCS/MIp7".
Server: kamailio (4.1.6 (i386/linux)).
Content-Length: 0.
But when using the Ngrep command on Asterisk to capture traffic on port
5050 or even 5060 i get no thing ! other troubleshooting steps i followed
including :
1.Verfiying the Mysql connection from Kamailio and the account tabe name
and SIP username / password column.
root at debian:/usr/local/etc/kamailio# mysql -u sipuser -h 192.168.100.10 -p
Enter password:
Welcome to the MySQL monitor. Commands end with ; or \g.
Your MySQL connection id is 149
Server version: 5.1.73 Source distribution
Copyright (c) 2000, 2014, Oracle and/or its affiliates. All rights reserved.
Oracle is a registered trademark of Oracle Corporation and/or its
affiliates. Other names may be trademarks of their respective
owners.
Type 'help;' or '\h' for help. Type '\c' to clear the current input
statement.
mysql> use asterisk;
Reading table information for completion of table and column names
You can turn off this feature to get a quicker startup with -A
Database changed
mysql> SELECT * FROM sip;
+------+------------------+---------------------------------+-------+
| id | keyword | data | flags |
+------+------------------+---------------------------------+-------+
| 1001 | pickupgroup | | 22 |
| 1001 | callgroup | | 21 |
| 1001 | encryption | no | 20 |
| 1001 | icesupport | no | 19 |
| 1001 | force_avp | no | 18 |
| 1001 | avpf | no | 17 |
| 1001 | transport | udp,tcp,tls | 16 |
| 1001 | qualifyfreq | 60 | 15 |
| 1001 | qualify | yes | 14 |
| 1001 | port | 5050 | 13 |
| 1001 | nat | no | 12 |
| 1001 | type | friend | 11 |
| 1001 | sendrpid | no | 10 |
| 1001 | trustrpid | yes | 9 |
| 1001 | host | dynamic | 8 |
| 1001 | context | from-internal | 7 |
| 1001 | canreinvite | no | 6 |
| 1001 | dtmfmode | rfc2833 | 5 |
| 1001 | secret | 1001secret | 4 |
| 1001 | secret_origional | 1001secret | 3 |
| 1001 | sipdriver | chan_sip | 2 |
| 1001 | dial | SIP/1001 | 25 |
| 1002 | pickupgroup | | 22 |
| 1002 | callgroup | | 21 |
| 1002 | encryption | no | 20 |
| 1002 | icesupport | no | 19 |
| 1002 | force_avp | no | 18 |
| 1002 | avpf | no | 17 |
| 1002 | transport | udp,tcp,tls | 16 |
| 1002 | qualifyfreq | 60 | 15 |
| 1002 | qualify | yes | 14 |
| 1002 | port | 5060 | 13 |
| 1002 | nat | no | 12 |
| 1002 | type | friend | 11 |
| 1002 | sendrpid | no | 10 |
| 1002 | trustrpid | yes | 9 |
| 1002 | host | dynamic | 8 |
| 1002 | context | from-internal | 7 |
| 1002 | canreinvite | no | 6 |
| 1002 | dtmfmode | rfc2833 | 5 |
| 1002 | secret | 1002secret | 4 |
| 1002 | secret_origional | 1002secret | 3 |
| 1002 | sipdriver | chan_sip | 2 |
| 1002 | dial | SIP/1002 | 25 |
| 1002 | disallow | | 23 |
| 1002 | allow | | 24 |
| 1002 | accountcode | | 26 |
| 1002 | mailbox | 1002 at device | 27 |
| 1002 | deny | 0.0.0.0/0.0.0.0 | 28 |
| 1002 | permit | 0.0.0.0/0.0.0.0 | 29 |
| 1002 | account | 1002 | 30 |
| 1002 | callerid | Ahmed Ramadan's Device <1002> | 31 |
| 1001 | disallow | | 23 |
| 1001 | allow | | 24 |
| 1001 | accountcode | | 26 |
| 1001 | mailbox | 1001 at device | 27 |
| 1001 | deny | 0.0.0.0/0.0.0.0 | 28 |
| 1001 | permit | 0.0.0.0/0.0.0.0 | 29 |
| 1001 | account | 1001 | 30 |
| 1001 | callerid | Mahmoud Ramadan's Device <1001> | 31 |
+------+------------------+---------------------------------+-------+
60 rows in set (0.00 sec)
2.Verifying that Asterisk can listen at 5050 which is the same Asterisk
port configured on Kamailio.
[root at Asterisk VM 01 ~]# asterisk -r
Asterisk 11.13.1, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 11.13.1 currently running on Asterisk VM 01 (pid =
2456)
Asterisk VM 01*CLI> sip show settings
Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5050
I know it is a long message but i wanted to give you all the INFO you might
need also I've attached my configuration file so you can check it.Thank you
Mohamed for your assistance.
On Sun, Nov 16, 2014 at 8:25 PM, Muhammad Shahzad <shaheryarkh at gmail.com>
wrote:
> Because both kamailio and asterisk use the same db table for
> authentication, see the auth_db module parameters in kamailio config.
>
> The REGISTER request from sip user is authenticated by kamailio using
> auth_db module and upon success kamailio generates REGISTER request back to
> asterisk (using the credentials sent by sip user for authentication with
> kamailio), this request is now authenticated by asterisk using realtime sip
> users interface.
>
> Thank you.
>
>
>
> On Sun, Nov 16, 2014 at 2:53 PM, Mahmoud Ramadan Ali <
> cisco.and.more.blog at gmail.com> wrote:
>
>> Hi Muhammad,
>> If the users MUST authenticate to Kamailio first,This means that Kamailio
>> should be aware of the SIP users exist in the Asterisk DB to be able to
>> authenticate them and NOT receive 401 Unauthorized error message from
>> Kamailio.
>> My question now might be simple but it a point of confusion to me and it
>> is how to tell Kamailio about the SIP users in the Asterisk DB ?!
>>
>> Best Regards,
>>
>>
>> On Sun, Nov 16, 2014 at 3:01 PM, Muhammad Shahzad <shaheryarkh at gmail.com>
>> wrote:
>>
>>> This seems to be fine. The user MUST authenticate to Kamailio, only then
>>> Kamailio will create REGISTER request that is send to asterisk. That's the
>>> key security feature behind the idea.
>>>
>>> Look at the register architecture diagram,
>>>
>>>
>>> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb#registration
>>>
>>> Thank you.
>>>
>>>
>>>
>>> On Sat, Nov 15, 2014 at 10:31 PM, Mahmoud Ramadan Ali <
>>> cisco.and.more.blog at gmail.com> wrote:
>>>
>>>> Hi Dears,
>>>> I'm trying to configure Kamailio as SBC in multi home mode for Asterisk
>>>> by authenticating the inbound SIP registration requests,i'm following this
>>>> tutorial
>>>> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
>>>> to achieve this goal. i have modified the necessary changes like the
>>>> Asterisk DB URL and the SIP table name and Username and password column and
>>>> verified the connection.
>>>>
>>>> My topology like this *Asterisk (192.168.100.10)
>>>> <----Internal:192.168.100.1---->Kamailio<---External:192.168.50.1-----> SIP
>>>> Phone (192.168.50.2)*
>>>> But when trying to register a SIP phone Kamailio does NOT forward the
>>>> authentication request to Asterisk and sends 401 Unauthorized error
>>>> message.I've attached my config file if any one wants to check it and
>>>> thanks in advance.
>>>> Best Regards
>>>>
>>>>
>>>> U 192.168.50.2:37297 -> 192.168.50.1:5060
>>>> REGISTER sip:192.168.50.1;transport=UDP SIP/2.0.
>>>> Via: SIP/2.0/UDP 192.168.50.2:37297
>>>> ;branch=z9hG4bK-d8754z-a46e0c7c9d98fe52-1---d8754z-;rport;transport=UDP.
>>>> Max-Forwards: 70.
>>>> Contact: <sip:1001 at 192.168.50.2:37297
>>>> ;rinstance=1d7c44dbcb8a7a2f;transport=UDP>.
>>>> To: <sip:1001 at 192.168.50.1;transport=UDP>.
>>>> From: <sip:1001 at 192.168.50.1;transport=UDP>;tag=1d222e19.
>>>> Call-ID: NTc2NDBjMGQ2YWFmZjdmNWI0MzVmN2Y4NzYyODJlMTc..
>>>> CSeq: 2 REGISTER.
>>>> Expires: 70.
>>>> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
>>>> SUBSCRIBE.
>>>> Supported: replaces, norefersub, extended-refer, timer,
>>>> X-cisco-serviceuri.
>>>> User-Agent: Z 3.2.21357 r21367.
>>>> Authorization: Digest
>>>> username="1001",realm="192.168.50.1",nonce="VGfAuFRnv4wMvoTG7wA9tqYD9fgZDe3D",uri="sip:192.168.50.1;transport=UDP",response="8bbd01d879250585eafee4f510689f73",algorithm=MD5.
>>>> Allow-Events: presence, kpml.
>>>> Content-Length: 0.
>>>> #
>>>> U 192.168.50.1:5060 -> 192.168.50.2:37297
>>>> SIP/2.0 401 Unauthorized.
>>>> Via: SIP/2.0/UDP 192.168.50.2:37297
>>>> ;branch=z9hG4bK-d8754z-a46e0c7c9d98fe52-1---d8754z-;rport=37297;transport=UDP.
>>>> To: <sip:1001 at 192.168.50.1
>>>> ;transport=UDP>;tag=b27e1a1d33761e85846fc98f5f3a7e58.fe8b.
>>>> From: <sip:1001 at 192.168.50.1;transport=UDP>;tag=1d222e19.
>>>> Call-ID: NTc2NDBjMGQ2YWFmZjdmNWI0MzVmN2Y4NzYyODJlMTc..
>>>> CSeq: 2 REGISTER.
>>>> WWW-Authenticate: Digest realm="192.168.50.1",
>>>> nonce="VGfAuFRnv4wMvoTG7wA9tqYD9fgZDe3D".
>>>> Server: kamailio (4.1.6 (i386/linux)).
>>>> Content-Length: 0.
>>>>
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>> sr-users at lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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#!KAMAILIO
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_USRLOCDB
#!define WITH_ASTERISK
#
# Kamailio (OpenSER) SIP Server v4.0 - default configuration script
# - web: http://www.kamailio.org
# - git: http://sip-router.org
#
# Direct your questions about this file to: <sr-users at lists.sip-router.org>
#
# Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php
# for an explanation of possible statements, functions and parameters.
#
# Several features can be enabled using '#!define WITH_FEATURE' directives:
#
# *** To run in debug mode:
# - define WITH_DEBUG
#
# *** To enable mysql:
# - define WITH_MYSQL
#
# *** To enable authentication execute:
# - enable mysql
# - define WITH_AUTH
# - add users using 'kamctl'
#
# *** To enable IP authentication execute:
# - enable mysql
# - enable authentication
# - define WITH_IPAUTH
# - add IP addresses with group id '1' to 'address' table
#
# *** To enable persistent user location execute:
# - enable mysql
# - define WITH_USRLOCDB
#
# *** To enable presence server execute:
# - enable mysql
# - define WITH_PRESENCE
#
# *** To enable nat traversal execute:
# - define WITH_NAT
# - install RTPProxy: http://www.rtpproxy.org
# - start RTPProxy:
# rtpproxy -l _your_public_ip_ -s udp:localhost:7722
#
# *** To enable PSTN gateway routing execute:
# - define WITH_PSTN
# - set the value of pstn.gw_ip
# - check route[PSTN] for regexp routing condition
#
# *** To enable database aliases lookup execute:
# - enable mysql
# - define WITH_ALIASDB
#
# *** To enable speed dial lookup execute:
# - enable mysql
# - define WITH_SPEEDDIAL
#
# *** To enable multi-domain support execute:
# - enable mysql
# - define WITH_MULTIDOMAIN
#
# *** To enable TLS support execute:
# - adjust CFGDIR/tls.cfg as needed
# - define WITH_TLS
#
# *** To enable XMLRPC support execute:
# - define WITH_XMLRPC
# - adjust route[XMLRPC] for access policy
#
# *** To enable anti-flood detection execute:
# - adjust pike and htable=>ipban settings as needed (default is
# block if more than 16 requests in 2 seconds and ban for 300 seconds)
# - define WITH_ANTIFLOOD
#
# *** To block 3XX redirect replies execute:
# - define WITH_BLOCK3XX
#
# *** To enable VoiceMail routing execute:
# - define WITH_VOICEMAIL
# - set the value of voicemail.srv_ip
# - adjust the value of voicemail.srv_port
#
# *** To enhance accounting execute:
# - enable mysql
# - define WITH_ACCDB
# - add following columns to database
#!ifdef ACCDB_COMMENT
ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default '';
ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default '';
ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
#!endif
####### Defined Values #########
# *** Value defines - IDs used later in config
#!ifdef WITH_MYSQL
# - database URL - used to connect to database server by modules such
# as: auth_db, acc, usrloc, a.s.o.
#!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio"
#!ifdef WITH_ASTERISK
#!define DBASTURL "mysql://sipuser:sippassword@192.168.100.10/asterisk"
#!endif
#!endif
#!ifdef WITH_MULTIDOMAIN
# - the value for 'use_domain' parameters
#!define MULTIDOMAIN 1
#!else
#!define MULTIDOMAIN 0
#!endif
# - flags
# FLT_ - per transaction (message) flags
# FLB_ - per branch flags
#!define FLT_ACC 1
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
#!define FLT_NATS 5
#!define FLB_NATB 6
#!define FLB_NATSIPPING 7
####### Global Parameters #########
#!ifdef WITH_DEBUG
debug=4
log_stderror=yes
#!else
debug=2
log_stderror=no
#!endif
memdbg=5
memlog=5
log_facility=LOG_LOCAL0
fork=yes
children=4
/* uncomment the next line to disable TCP (default on) */
#disable_tcp=yes
/* uncomment the next line to disable the auto discovery of local aliases
based on reverse DNS on IPs (default on) */
#auto_aliases=no
/* add local domain aliases */
#alias="sip.mydomain.com"
/* uncomment and configure the following line if you want Kamailio to
bind on a specific interface/port/proto (default bind on all available) */
#listen=udp:10.0.0.10:5060
/* port to listen to
* - can be specified more than once if needed to listen on many ports */
port=5060
#!ifdef WITH_TLS
enable_tls=yes
#!endif
# life time of TCP connection when there is no traffic
# - a bit higher than registration expires to cope with UA behind NAT
tcp_connection_lifetime=3605
mhomed=1
####### Custom Parameters #########
# These parameters can be modified runtime via RPC interface
# - see the documentation of 'cfg_rpc' module.
#
# Format: group.id = value 'desc' description
# Access: $sel(cfg_get.group.id) or @cfg_get.group.id
#
#!ifdef WITH_PSTN
# PSTN GW Routing
#
# - pstn.gw_ip: valid IP or hostname as string value, example:
# pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address"
#
# - by default is empty to avoid misrouting
pstn.gw_ip = "" desc "PSTN GW Address"
#!endif
#!ifdef WITH_VOICEMAIL
# VoiceMail Routing on offline, busy or no answer
#
# - by default Voicemail server IP is empty to avoid misrouting
voicemail.srv_ip = "192.168.100.10" desc "VoiceMail IP Address"
voicemail.srv_port = "5050" desc "VoiceMail Port"
#!endif
#!ifdef WITH_ASTERISK
asterisk.bindip = "192.168.100.10" desc "Asterisk IP Address"
asterisk.bindport = "5050" desc "Asterisk Port"
kamailio.bindip = "192.168.100.1" desc "Kamailio IP Address"
kamailio.bindport = "5060" desc "Kamailio Port"
#!endif
####### Modules Section ########
# set paths to location of modules (to sources or installation folders)
#!ifdef WITH_SRCPATH
mpath="modules_k:modules"
#!else
mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/"
#!endif
#!ifdef WITH_MYSQL
loadmodule "db_mysql.so"
#!endif
loadmodule "mi_fifo.so"
loadmodule "kex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "mi_rpc.so"
loadmodule "acc.so"
#!ifdef WITH_AUTH
loadmodule "auth.so"
loadmodule "auth_db.so"
#!ifdef WITH_IPAUTH
loadmodule "permissions.so"
#!endif
#!endif
#!ifdef WITH_ALIASDB
loadmodule "alias_db.so"
#!endif
#!ifdef WITH_SPEEDDIAL
loadmodule "speeddial.so"
#!endif
#!ifdef WITH_MULTIDOMAIN
loadmodule "domain.so"
#!endif
#!ifdef WITH_PRESENCE
loadmodule "presence.so"
loadmodule "presence_xml.so"
#!endif
#!ifdef WITH_NAT
loadmodule "nathelper.so"
loadmodule "rtpproxy.so"
#!endif
#!ifdef WITH_TLS
loadmodule "tls.so"
#!endif
#!ifdef WITH_ANTIFLOOD
loadmodule "htable.so"
loadmodule "pike.so"
#!endif
#!ifdef WITH_XMLRPC
loadmodule "xmlrpc.so"
#!endif
#!ifdef WITH_DEBUG
loadmodule "debugger.so"
#!endif
#!ifdef WITH_ASTERISK
loadmodule "uac.so"
#!endif
# ----------------- setting module-specific parameters ---------------
# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
# ----- tm params -----
# auto-discard branches from previous serial forking leg
modparam("tm", "failure_reply_mode", 3)
# default retransmission timeout: 30sec
modparam("tm", "fr_timer", 30000)
# default invite retransmission timeout after 1xx: 120sec
modparam("tm", "fr_inv_timer", 120000)
# ----- rr params -----
# add value to ;lr param to cope with most of the UAs
modparam("rr", "enable_full_lr", 1)
# do not append from tag to the RR (no need for this script)
#!ifdef WITH_ASTERISK
modparam("rr", "append_fromtag", 1)
#!else
modparam("rr", "append_fromtag", 0)
#!endif
# ----- registrar params -----
modparam("registrar", "method_filtering", 1)
/* uncomment the next line to disable parallel forking via location */
# modparam("registrar", "append_branches", 0)
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
# max value for expires of registrations
modparam("registrar", "max_expires", 3600)
# set it to 1 to enable GRUU
modparam("registrar", "gruu_enabled", 0)
# ----- acc params -----
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_ack", 0)
modparam("acc", "report_cancels", 0)
/* by default ww do not adjust the direct of the sequential requests.
if you enable this parameter, be sure the enable "append_fromtag"
in "rr" module */
modparam("acc", "detect_direction", 0)
/* account triggers (flags) */
modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "log_missed_flag", FLT_ACCMISSED)
modparam("acc", "log_extra",
"src_user=$fU;src_domain=$fd;src_ip=$si;"
"dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
/* enhanced DB accounting */
#!ifdef WITH_ACCDB
modparam("acc", "db_flag", FLT_ACC)
modparam("acc", "db_missed_flag", FLT_ACCMISSED)
modparam("acc", "db_url", DBURL)
modparam("acc", "db_extra",
"src_user=$fU;src_domain=$fd;src_ip=$si;"
"dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
#!endif
# ----- usrloc params -----
/* enable DB persistency for location entries */
#!ifdef WITH_USRLOCDB
modparam("usrloc", "db_url", DBURL)
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "use_domain", MULTIDOMAIN)
#!endif
# ----- auth_db params -----
#!ifdef WITH_AUTH
modparam("auth_db", "user_column", "account")
modparam("auth_db", "password_column", "secret")
modparam("auth_db", "db_url", "mysql://sipuser:sippassword@192.168.100.10/asterisk")
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "load_credentials", "")
#!ifdef WITH_ASTERISK
modparam("auth_db", "user_column", "account")
modparam("auth_db", "password_column", "secret")
modparam("auth_db", "db_url", "mysql://sipuser:sippassword@192.168.100.10/asterisk")
modparam("auth_db", "version_table", 0)
#!else
modparam("auth_db", "db_url", "db_url","mysql://sipuser:sippassword@192.168.100.10/asterisk")
modparam("auth_db", "password_column", "secret")
modparam("auth_db", "user_column", "account")
modparam("auth_db", "use_domain", MULTIDOMAIN)
#!endif
# ----- permissions params -----
#!ifdef WITH_IPAUTH
modparam("permissions", "db_url", DBURL)
modparam("permissions", "db_mode", 1)
#!endif
#!endif
# ----- alias_db params -----
#!ifdef WITH_ALIASDB
modparam("alias_db", "db_url", DBURL)
modparam("alias_db", "use_domain", MULTIDOMAIN)
#!endif
# ----- speedial params -----
#!ifdef WITH_SPEEDDIAL
modparam("speeddial", "db_url", DBURL)
modparam("speeddial", "use_domain", MULTIDOMAIN)
#!endif
# ----- domain params -----
#!ifdef WITH_MULTIDOMAIN
modparam("domain", "db_url", DBURL)
# register callback to match myself condition with domains list
modparam("domain", "register_myself", 1)
#!endif
#!ifdef WITH_PRESENCE
# ----- presence params -----
modparam("presence", "db_url", DBURL)
# ----- presence_xml params -----
modparam("presence_xml", "db_url", DBURL)
modparam("presence_xml", "force_active", 1)
#!endif
#!ifdef WITH_NAT
# ----- rtpproxy params -----
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:pinger at kamailio.org")
# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif
#!ifdef WITH_TLS
# ----- tls params -----
modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg")
#!endif
#!ifdef WITH_ANTIFLOOD
# ----- pike params -----
modparam("pike", "sampling_time_unit", 2)
modparam("pike", "reqs_density_per_unit", 16)
modparam("pike", "remove_latency", 4)
# ----- htable params -----
# ip ban htable with autoexpire after 5 minutes
modparam("htable", "htable", "ipban=>size=8;autoexpire=300;")
#!endif
#!ifdef WITH_XMLRPC
# ----- xmlrpc params -----
modparam("xmlrpc", "route", "XMLRPC");
modparam("xmlrpc", "url_match", "^/RPC")
#!endif
#!ifdef WITH_DEBUG
# ----- debugger params -----
modparam("debugger", "cfgtrace", 1)
#!endif
####### Routing Logic ########
# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
# - note: this is the same as route { ... }
request_route {
# per request initial checks
route(REQINIT);
# NAT detection
route(NATDETECT);
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
# authentication
route(AUTH);
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE"))
record_route();
# account only INVITEs
if (is_method("INVITE"))
{
setflag(FLT_ACC); # do accounting
}
# dispatch requests to foreign domains
route(SIPOUT);
### requests for my local domains
# handle presence related requests
route(PRESENCE);
# handle registrations
route(REGISTRAR);
if ($rU==$null)
{
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# dispatch destinations to PSTN
route(PSTN);
# user location service
route(LOCATION);
route(RELAY);
}
route[RELAY] {
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|SUBSCRIBE")) {
t_on_branch("MANAGE_BRANCH");
t_on_reply("MANAGE_REPLY");
}
if (is_method("INVITE")) {
t_on_failure("MANAGE_FAILURE");
}
if (!t_relay()) {
sl_reply_error();
}
exit;
}
# Per SIP request initial checks
route[REQINIT] {
#!ifdef WITH_ANTIFLOOD
# flood dection from same IP and traffic ban for a while
# be sure you exclude checking trusted peers, such as pstn gateways
# - local host excluded (e.g., loop to self)
if(src_ip!=myself)
{
if($sht(ipban=>$si)!=$null)
{
# ip is already blocked
xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n");
exit;
}
if (!pike_check_req())
{
xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n");
$sht(ipban=>$si) = 1;
exit;
}
}
#!endif
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}
if(!sanity_check("1511", "7"))
{
xlog("Malformed SIP message from $si:$sp\n");
exit;
}
}
# Handle requests within SIP dialogs
route[WITHINDLG] {
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("BYE")) {
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the transaction fails
}
if ( is_method("ACK") ) {
# ACK is forwarded statelessy
route(NATMANAGE);
}
route(RELAY);
} else {
if (is_method("SUBSCRIBE") && uri == myself) {
# in-dialog subscribe requests
route(PRESENCE);
exit;
}
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# no loose-route, but stateful ACK;
# must be an ACK after a 487
# or e.g. 404 from upstream server
t_relay();
exit;
} else {
# ACK without matching transaction ... ignore and discard
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
}
# Handle SIP registrations
route[REGISTRAR] {
if (is_method("REGISTER"))
{
if(isflagset(FLT_NATS))
{
setbflag(FLB_NATB);
# uncomment next line to do SIP NAT pinging
## setbflag(FLB_NATSIPPING);
}
if (!save("location"))
sl_reply_error();
#!ifdef WITH_ASTERISK
route(REGFWD);
#!endif
exit;
}
}
# USER location service
route[LOCATION] {
#!ifdef WITH_SPEEDIAL
# search for short dialing - 2-digit extension
if($rU=~"^[0-9][0-9]$")
if(sd_lookup("speed_dial"))
route(SIPOUT);
#!endif
#!ifdef WITH_ALIASDB
# search in DB-based aliases
if(alias_db_lookup("dbaliases"))
route(SIPOUT);
#!endif
#!ifdef WITH_ASTERISK
if(is_method("INVITE") && (!route(FROMASTERISK))) {
# if new call from out there - send to Asterisk
# - non-INVITE request are routed directly by Kamailio
# - traffic from Asterisk is routed also directy by Kamailio
route(TOASTERISK);
exit;
}
#!endif
$avp(oexten) = $rU;
if (!lookup("location")) {
$var(rc) = $rc;
route(TOVOICEMAIL);
t_newtran();
switch ($var(rc)) {
case -1:
case -3:
send_reply("404", "Not Found");
exit;
case -2:
send_reply("405", "Method Not Allowed");
exit;
}
}
# when routing via usrloc, log the missed calls also
if (is_method("INVITE"))
{
setflag(FLT_ACCMISSED);
}
}
# Presence server route
route[PRESENCE] {
if(!is_method("PUBLISH|SUBSCRIBE"))
return;
#!ifdef WITH_PRESENCE
if (!t_newtran())
{
sl_reply_error();
exit;
};
if(is_method("PUBLISH"))
{
handle_publish();
t_release();
}
else
if( is_method("SUBSCRIBE"))
{
handle_subscribe();
t_release();
}
exit;
#!endif
# if presence enabled, this part will not be executed
if (is_method("PUBLISH") || $rU==$null)
{
sl_send_reply("404", "Not here");
exit;
}
return;
}
# Authentication route
route[AUTH] {
# if caller is not local subscriber, then check if it calls
# a local destination, otherwise deny, not an open relay here
if (from_uri!=myself && uri!=myself)
{
sl_send_reply("403","Not relaying");
exit;
}
#!ifdef WITH_AUTH
#!ifdef WITH_ASTERISK
# do not auth traffic from Asterisk - trusted!
if(route(FROMASTERISK))
return;
#!endif
#!ifdef WITH_IPAUTH
if((!is_method("REGISTER")) && allow_source_address())
{
# source IP allowed
return;
}
#!endif
if (is_method("REGISTER") || from_uri==myself)
{
# authenticate requests
#!ifdef WITH_ASTERISK
if (!auth_check("$fd", "sip", "1")) {
#!else
if (!auth_check("$fd", "subscriber", "1")) {
#!endif
auth_challenge("$fd", "0");
exit;
}
# user authenticated - remove auth header
if(!is_method("REGISTER|PUBLISH"))
consume_credentials();
}
#!endif
return;
}
# Caller NAT detection route
route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
fix_nated_contact();
}
setflag(FLT_NATS);
}
#!endif
return;
}
# RTPProxy control
route[NATMANAGE] {
#!ifdef WITH_NAT
if (is_request()) {
if(has_totag()) {
if(check_route_param("nat=yes")) {
setbflag(FLB_NATB);
}
}
}
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
rtpproxy_manage();
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
# Routing to foreign domains
route[SIPOUT] {
if (!uri==myself)
{
append_hf("P-hint: outbound\r\n");
route(RELAY);
}
}
# PSTN GW routing
route[PSTN] {
#!ifdef WITH_PSTN
# check if PSTN GW IP is defined
if (strempty($sel(cfg_get.pstn.gw_ip))) {
xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n");
return;
}
# route to PSTN dialed numbers starting with '+' or '00'
# (international format)
# - update the condition to match your dialing rules for PSTN routing
if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$"))
return;
# only local users allowed to call
if(from_uri!=myself) {
sl_send_reply("403", "Not Allowed");
exit;
}
$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
route(RELAY);
exit;
#!endif
return;
}
# XMLRPC routing
#!ifdef WITH_XMLRPC
route[XMLRPC] {
# allow XMLRPC from localhost
if ((method=="POST" || method=="GET")
&& (src_ip==127.0.0.1)) {
# close connection only for xmlrpclib user agents (there is a bug in
# xmlrpclib: it waits for EOF before interpreting the response).
if ($hdr(User-Agent) =~ "xmlrpclib")
set_reply_close();
set_reply_no_connect();
dispatch_rpc();
exit;
}
send_reply("403", "Forbidden");
exit;
}
#!endif
# route to voicemail server
route[TOVOICEMAIL] {
#!ifdef WITH_VOICEMAIL
if(!is_method("INVITE"))
return;
# check if VoiceMail server IP is defined
if (strempty($sel(cfg_get.voicemail.srv_ip))) {
xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n");
return;
}
if($avp(oexten)==$null)
return;
$ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip)
+ ":" + $sel(cfg_get.voicemail.srv_port);
route(RELAY);
exit;
#!endif
return;
}
# manage outgoing branches
branch_route[MANAGE_BRANCH] {
xdbg("new branch [$T_branch_idx] to $ru\n");
route(NATMANAGE);
}
# manage incoming replies
onreply_route[MANAGE_REPLY] {
xdbg("incoming reply\n");
if(status=~"[12][0-9][0-9]")
route(NATMANAGE);
}
# manage failure routing cases
failure_route[MANAGE_FAILURE] {
route(NATMANAGE);
if (t_is_canceled()) {
exit;
}
#!ifdef WITH_BLOCK3XX
# block call redirect based on 3xx replies.
if (t_check_status("3[0-9][0-9]")) {
t_reply("404","Not found");
exit;
}
#!endif
#!ifdef WITH_VOICEMAIL
# serial forking
# - route to voicemail on busy or no answer (timeout)
if (t_check_status("486|408")) {
route(TOVOICEMAIL);
exit;
}
#!endif
}
#!ifdef WITH_ASTERISK
# Test if coming from Asterisk
route[FROMASTERISK] {
if($si==$sel(cfg_get.asterisk.bindip)
&& $sp==$sel(cfg_get.asterisk.bindport))
return 1;
return -1;
}
# Send to Asterisk
route[TOASTERISK] {
$du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":"
+ $sel(cfg_get.asterisk.bindport);
route(RELAY);
exit;
}
# Forward REGISTER to Asterisk
route[REGFWD] {
if(!is_method("REGISTER"))
{
return;
}
$var(rip) = $sel(cfg_get.asterisk.bindip);
$uac_req(method)="REGISTER";
$uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport);
$uac_req(furi)="sip:" + $au + "@" + $var(rip);
$uac_req(turi)="sip:" + $au + "@" + $var(rip);
$uac_req(hdrs)="Contact: <sip:" + $au + "@"
+ $sel(cfg_get.kamailio.bindip)
+ ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n";
if($sel(contact.expires) != $null)
$uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n";
else
$uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n";
uac_req_send();
}
#!endif
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