Re: [OpenSIPS-Users] no ringback
Hi, Bogdan:
Yes, I tried that . The phone rings once , and then keeps silent. I tried
to put more
180 there, but the phone still only rings once.
If there is a solution that sends 180 periodically at 2 to 3 seconds
interval until the
callee answers, probably then it will work, but is there anyway to get this
done ?
Jinsong
- Original Message -
From: "Bogdan-Andrei Iancu"
To: "Jinsong Hu"
Cc:
Sent: Thursday, August 13, 2009 2:31 AM
Subject: Re: [OpenSIPS-Users] no ringback
> Hi Jimmy ,
>
> There is a simple thing you can do:
>
> - just before relaying the INVITE the Asterisk, from OpenSIPS cfg, to a
> sl_send_reply("180","ringing"); to fire a local 180 - of course this is a
> bit bogus from logical perspective (as the end party does not actually
> ring, so you force some information that you cannot check).
>
> Regards,
> Bogdan
>
> Jinsong Hu wrote:
>> Hi, There:
>> I am using opensips/kamailio in front of asterisk pool. my user
>> register on the opensips, and pstn call are routed out via asterisk.
>> what I find out is that when the caller calls callee, some of the UA
>> doesn't generate ring back. for example, if I use xlite, the ring back
>> works fine. but if I use sipura 3000,
>> I don't hear anything until the callee picks up phone.
>> I did a debug and found that after INVITE, I get 200 back, and then the
>> UA sends out ACK. the callee never sends 180 or 183 back to the caller
>> UA. so before the callee pick up phone, all the caller can hear is just
>> silence.
>>
>> if my user registers directly on the asterisk, he can hear the ringback
>> because the Dial() command by default
>> will send ring back to the UA.
>>
>> How do I solve this problem in this case ? I searched all over internet
>> and don't see any body having any solution.
>>
>> Jimmy
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
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[OpenSIPS-Users] opensips before asterisk, but there is no ring back when calling. how to solve this ?
Hi, There: I am using opensips/kamailio in front of asterisk pool. my user register on the opensips, and pstn call are routed out via asterisk. what I find out is that when the caller calls callee, some of the UA doesn't generate ring back. for example, if I use xlite, the ring back works fine. but if I use sipura 3000, I don't hear anything until the callee picks up phone. I did a debug and found that after INVITE, I get 200 back, and then the UA sends out ACK. the callee never sends 180 or 183 back to the caller UA. so before the callee pick up phone, all the caller can hear is just silence. if my user registers directly on the asterisk, he can hear the ringback because the Dial() command by default will send ring back to the UA. How do I solve this problem in this case ? I searched all over internet and don't see any body having any solution. Jimmy ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] no ringback
Hi, There: I am using opensips/kamailio in front of asterisk pool. my user register on the opensips, and pstn call are routed out via asterisk. what I find out is that when the caller calls callee, some of the UA doesn't generate ring back. for example, if I use xlite, the ring back works fine. but if I use sipura 3000, I don't hear anything until the callee picks up phone. I did a debug and found that after INVITE, I get 200 back, and then the UA sends out ACK. the callee never sends 180 or 183 back to the caller UA. so before the callee pick up phone, all the caller can hear is just silence. if my user registers directly on the asterisk, he can hear the ringback because the Dial() command by default will send ring back to the UA. How do I solve this problem in this case ? I searched all over internet and don't see any body having any solution. Jimmy ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] opensips sending huge amount of accounting records for invite
does anybody know why opensips sends out huge amount of invite accounting records ? I set radius flag and missed flag all to 1. and used opensips 1.5.1. radius 1.1.7. from radius log, when I do invite, I found more than 70 invite UDP packets sent to radius. and this created a big burden on my database for radius. I changed the time out value to 300 milliseconds, and it seems it doesn't help. Jimmy ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] help needed for auth-radius
The freeradius I got come from http://cdrtool.ag-projects.com/wiki/Install I installed it via apt-get install freeradius-xs freeradius-xs-mysql command as listed there in debian lenny. I checked the version of the installed freeradius, it is 1.1.7 . according to the instruction , it is a patched version. attached is the result of "freeradius -X". the client I used is libradiusclient-ng2 in debian lenny. I suspect that client.conf or radiusd.conf some configuration is wrong, but I just don't know what is wrong. Jimmy - Original Message - From: "Uwe Kastens" To: "Jinsong Hu" Cc: Sent: Thursday, May 21, 2009 5:36 AM Subject: Re: [OpenSIPS-Users] help needed for auth-radius Jimmy, could you repeat the request after starting freeradius with -X. I am using freeradius-2 with libradiusclient-ng. BR Uwe Uwe Kastens schrieb: Jimmy, Have you installed all dictionaries which are required? What kind of radius server are you using? BR Uwe Jinsong Hu schrieb: Hi, There : I am trying to use auth-radius module in opensips 1.5, and it doesn't work. I turned on the debug mode for freeradius, and got this log. indeed, I noticed that the message doesn't contain a row for password like this: User-Password = "1234567890" does any body know what I am missing here ? I am running freeradius 1.1.7 version, with freeradius-ng client. I suspect that the client config file needs change, but don't know how. can anybody help ? Jimmy rad_recv: Access-Request packet from host 127.0.0.1:58938, id=47, length=297 User-Name = "1234567...@voip.mydomain.com <mailto:1234567...@voip.mydomain.com>" Digest-Attributes = 0x0a0d3137373735353530313031 Digest-Attributes = 0x011a6f7370317030312e736572766963656f6e7765622e636f6d Digest-Attributes = 0x023234613134653863343030303030303164646330623064313731643864616165613465363164663264616362623938 Digest-Attributes = 0x04137369703a3139322e3136382e312e313533 Digest-Attributes = 0x030a5245474953544552 Digest-Response = "dcc035349020529884e8cfbece126a48" Service-Type = Sip-Session Sip-Uri-User = "1234567890" Acct-Session-Id = "OTllMmE4YTA3NGU1MzY3OGVhYzM2MWFhNDk2MDdjYjM." NAS-Port = 5060 NAS-IP-Address = 127.0.0.1 auth: No authenticate method (Auth-Type) configuration found for the request: Rejecting the user auth: Failed to validate the user. Login incorrect: [1234567...@voip.mydomain.com/] (from client localhost port 5060) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- kiste lat: 54.322684, lon: 10.13586 Starting - reading configuration files ... reread_config: reading radiusd.conf Config: including file: /etc/freeradius/proxy.conf Config: including file: /etc/freeradius/clients.conf Config: including file: /etc/freeradius/snmp.conf Config: including file: /etc/freeradius/sql.conf main: prefix = "/usr" main: localstatedir = "/var" main: logdir = "/var/log/freeradius" main: libdir = "/usr/lib/freeradius" main: radacctdir = "/var/log/freeradius/radacct" main: hostname_lookups = no main: snmp = no main: max_request_time = 6 main: cleanup_delay = 5 main: max_requests = 1024 main: delete_blocked_requests = 0 main: port = 0 main: allow_core_dumps = no main: log_stripped_names = no main: log_file = "/var/log/freeradius/radius.log" main: log_auth = yes main: log_auth_badpass = yes main: log_auth_goodpass = yes main: pidfile = "/var/run/freeradius/freeradius.pid" main: user = "freerad" main: group = "freerad" main: usercollide = no main: lower_user = "no" main: lower_pass = "no" main: nospace_user = "no" main: nospace_pass = "no" main: checkrad = "/usr/sbin/checkrad" main: proxy_requests = yes proxy: retry_delay = 1 proxy: retry_count = 5 proxy: synchronous = no proxy: default_fallback = yes proxy: dead_time = 120 proxy: post_proxy_authorize = yes proxy: wake_all_if_all_dead = no security: max_attributes = 200 security: reject_delay = 1 security: status_server = no main: debug_level = 0 read_config_files: reading dictionary read_config_files: reading naslist Using deprecated naslist file. Support for this will go away soon. read_config_files: reading clients read_config_files: reading realms radiusd: entering modules setup Module: Library search path is /usr/lib/freeradius Module: Loaded preprocess preprocess: huntgroups = "/etc/freeradius/huntgroups" preprocess: hints
[OpenSIPS-Users] help needed for auth-radius
Hi, There : I am trying to use auth-radius module in opensips 1.5, and it doesn't work. I turned on the debug mode for freeradius, and got this log. indeed, I noticed that the message doesn't contain a row for password like this: User-Password = "1234567890" does any body know what I am missing here ? I am running freeradius 1.1.7 version, with freeradius-ng client. I suspect that the client config file needs change, but don't know how. can anybody help ? Jimmy rad_recv: Access-Request packet from host 127.0.0.1:58938, id=47, length=297 User-Name = "1234567...@voip.mydomain.com" Digest-Attributes = 0x0a0d3137373735353530313031 Digest-Attributes = 0x011a6f7370317030312e736572766963656f6e7765622e636f6d Digest-Attributes = 0x023234613134653863343030303030303164646330623064313731643864616165613465363164663264616362623938 Digest-Attributes = 0x04137369703a3139322e3136382e312e313533 Digest-Attributes = 0x030a5245474953544552 Digest-Response = "dcc035349020529884e8cfbece126a48" Service-Type = Sip-Session Sip-Uri-User = "1234567890" Acct-Session-Id = "OTllMmE4YTA3NGU1MzY3OGVhYzM2MWFhNDk2MDdjYjM." NAS-Port = 5060 NAS-IP-Address = 127.0.0.1 auth: No authenticate method (Auth-Type) configuration found for the request: Rejecting the user auth: Failed to validate the user. Login incorrect: [1234567...@voip.mydomain.com/] (from client localhost port 5060) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] get_profile_size() fails with profile not definited
I solved this problem by myself:
define
modparam("dialog", "profiles_with_value", "caller ; callee")
solved it.
- Original Message -
From: Jinsong Hu
To: users@lists.opensips.org
Sent: Thursday, May 14, 2009 9:47 PM
Subject: Re: [OpenSIPS-Users] get_profile_size() fails with profile
not definited
Does anybody how to define profile ?
Jimmy
Hi,
I tried the sample code in
http://www.opensips.org/Resources/DocsTutConcurrentCalls
there it has:
# get current calls for uuid
get_profile_size("caller","$avp(s:caller_uuid)","$var(calls)");
and when dialog is created, it has caller profile defined:
create_dialog();
set_dlg_profile("caller","$avp(s:caller_uuid)");
but when I run this, I got the following exception:
May 14 05:45:56 [2633] CRITICAL:dialog:fixup_profile: profile not
definited
May 14 05:45:56 [2633] ERROR:core:fix_actions: fixing failed (code=-6) at
cfg line 753
May 14 05:45:56 [2633] ERROR:core:main: failed to fix configuration with
err code -6
does anybody know how to define the profile ?
I read the dialog doc
http://www.opensips.org/html/docs/modules/1.5.x/dialog.html
and it looks set_dlg_profile("caller","$avp(s:caller_uuid)"); is the method
to define the dialog,
if that is the case, why the sample doesn't work ?
Jimmy
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Re: [OpenSIPS-Users] get_profile_size() fails with profile not definited
Does anybody how to define profile ?
Jimmy
Hi,
I tried the sample code in
http://www.opensips.org/Resources/DocsTutConcurrentCalls
there it has:
# get current calls for uuid
get_profile_size("caller","$avp(s:caller_uuid)","$var(calls)");
and when dialog is created, it has caller profile defined:
create_dialog();
set_dlg_profile("caller","$avp(s:caller_uuid)");
but when I run this, I got the following exception:
May 14 05:45:56 [2633] CRITICAL:dialog:fixup_profile: profile not
definited
May 14 05:45:56 [2633] ERROR:core:fix_actions: fixing failed (code=-6) at cfg
line 753
May 14 05:45:56 [2633] ERROR:core:main: failed to fix configuration with err
code -6
does anybody know how to define the profile ?
I read the dialog doc
http://www.opensips.org/html/docs/modules/1.5.x/dialog.html
and it looks set_dlg_profile("caller","$avp(s:caller_uuid)"); is the method
to define the dialog,
if that is the case, why the sample doesn't work ?
Jimmy
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[OpenSIPS-Users] get_profile_size("caller","$avp(s:caller_uuid)","$var(calls)"); fails with profile not definited
Hi,
I tried the sample code in
http://www.opensips.org/Resources/DocsTutConcurrentCalls
there it has:
# get current calls for uuid
get_profile_size("caller","$avp(s:caller_uuid)","$var(calls)");
and when dialog is created, it has caller profile defined:
create_dialog();
set_dlg_profile("caller","$avp(s:caller_uuid)");
but when I run this, I got the following exception:
May 14 05:45:56 [2633] CRITICAL:dialog:fixup_profile: profile not
definited
May 14 05:45:56 [2633] ERROR:core:fix_actions: fixing failed (code=-6) at cfg
line 753
May 14 05:45:56 [2633] ERROR:core:main: failed to fix configuration with err
code -6
does anybody know how to define the profile ?
I read the dialog doc
http://www.opensips.org/html/docs/modules/1.5.x/dialog.html
and it looks set_dlg_profile("caller","$avp(s:caller_uuid)"); is the method to
define the dialog,
if that is the case, why the sample doesn't work ?
Jimmy
___
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[OpenSIPS-Users] sst.so load hangs
Hi, I used opensips 1.5.1 and the sst.so module hangs when initializing. when I comment it out, then the whole server starts fine. here is the log. I am running opensips in xen , debian lenny distribution. does any body know what is going on ? Jimmy May 13 16:52:24 [12189] INFO:usrloc:ul_init_locks: locks array size 512 May 13 16:52:24 [12189] NOTICE:signaling:mod_init: initializing module .. May 13 16:52:24 [12189] INFO:registrar:mod_init: initializing... May 13 16:52:24 [12189] INFO:textops:mod_init: initializing... May 13 16:52:24 [12189] INFO:xlog:mod_init: initializing... May 13 16:52:24 [12189] INFO:acc:mod_init: initializing... May 13 16:52:24 [12189] INFO:auth:mod_init: initializing... May 13 16:52:24 [12189] INFO:auth_db:mod_init: initializing... May 13 16:52:24 [12189] INFO:alias_db:mod_init: initializing... May 13 16:52:24 [12189] INFO:siptrace:mod_init: initializing... May 13 16:52:24 [12189] INFO:pike:pike_init: initializing... May 13 16:52:24 [12189] INFO:pike:init_lock_set: probing 256 set size May 13 16:52:24 [12189] INFO:avpops:avpops_init: initializing... May 13 16:52:24 [12189] INFO:auth_radius:mod_init: initializing... May 13 16:52:24 [12189] INFO:avp_radius:mod_init: initializing... May 13 16:52:24 [12189] INFO:dialog:mod_init: Dialog module - initializing May 13 16:52:24 [12189] INFO:drouting:dr_init: Dynamic-Routing - initializing May 13 16:52:24 [12189] INFO:sst:mod_init: SIP Session Timer module - initializing ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] fix_nated_register replacement
Hi, I noticed that in nathelper module, there is a fix_nated_register() method to fix the nat for register . however, in nat_traversal module, there is no such method. so if I migrate from nathelper to nat_traversal, what do I do to fix the nat and save the registration to usrloc ? Jimmy ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [OpenSIPS] sample script that works with cdrtools, freeradius, nat, drouting
Go ahead. I am not sure how good it is. if you find problems , please let me know. I am sure there must be problems there. But since I am also new, so I can't spot them all. that is why I posted it. it will be nice that there are people spot the problem and tell me. I can continue to improve it. when it is good enough, maybe this can be put in opensips repository for everybody to share. Jinsong - Original Message - From: "Khan" To: "Jinsong Hu" Cc: Sent: Monday, May 11, 2009 11:55 AM Subject: Re: [OpenSIPS-Users] [OpenSIPS] sample script that works with cdrtools, freeradius, nat, drouting Thanks Jimmy, I have been looking for some sample script for ages at every possible place. So, far no luck but its really nice of you to post your script here. I always wondered with open source software what is the deal of posting your functional script (hiding your confidential info) but never had an answer. Finally I see a brave person posting script... Would you mind if i try your script and see if it works for me??? Thanks, On Mon, May 11, 2009 at 1:31 AM, Jinsong Hu wrote: > Hi, There: > It looks ag-projects is maintaining the cdrtools, media proxy. but I > searched around and didn't find anywhere there is a script that supports > all > the needed feature: cdrtools, mediaproxy, nat_traversal, and drouting. so > now I'm trying to be a little brave and post my script that includes all > above. this script doesn't handle instance message, but only voice calls. > can any body spot problems with this script ? > The goal of the script is to let locally registered user to use gateway to > make outgoing call, and receive incoming call. the numbering plan is for > US. > free radius should have good authenticaing and accounting for different > messages, and some special DID are mapped to several numbers and routed to > asterisk. Hopefully this script will be useful for a general VOIP carrier. > I try to paste the document to be comment. Hopefully, by going through > this exercise, we can get a good starting script for people to use as a > model starting script. > > > Jimmy > > > > ### > # > # $Id: opensips.cfg,v 1.13 2009/05/11 06:06:00 jinsong Exp $ > # > # OpenSIPS basic configuration script > # by Anca Vamanu > # > # Please refer to the Core CookBook at > http://www.opensips.org/dokuwiki/doku.php > # for a explanation of possible statements, functions and parameters. > # > #INVITE :Invites a user to a call > #ACK : Acknowledgement is used to facilitate reliable message exchange for > INVITEs. > #BYE :Terminates a connection between users > #CANCEL :Terminates a request, or search, for a user. It is used if a > client > sends an INVITE and then changes its decision to call the recipient. > #OPTIONS :Solicits information about a server's capabilities. > #REGISTER :Registers a user's current location > #INFO :Used for mid-session signaling > #MESSAGE : IMS send message > #SUBSCRIBE : IMS presence subscribe message > #PUBLISH: IMS publish message > > #1xx: Provisional -- request received, continuing to process the request; > #2xx: Success -- the action was successfully received, understood, and > accepted; > #3xx: Redirection -- further action needs to be taken in order to complete > the request; > #4xx: Client Error -- the request contains bad syntax or cannot be > fulfilled > at this server; > #5xx: Server Error -- the server failed to fulfill an apparently valid > request; > #6xx: Global Failure -- the request cannot be fulfilled at any server. > > #This function sets the value of the flag given as parameter to 1 (true). > The value of the parameter must be an integer between 0 and 31. > > > > > > ### Global Parameters # > > debug=3 > log_stderror=no > log_facility=LOG_LOCAL0 > > fork=yes > children=4 > > /* uncomment the following lines to enable debugging */ > #debug=6 > #fork=no > #log_stderror=yes > > /* uncomment the next line to disable TCP (default on) */ > #disable_tcp=yes > > /* uncomment the next line to enable the auto temporary blacklisting of > not available destinations (default disabled) */ > #disable_dns_blacklist=no > > /* uncomment the next line to enable IPv6 lookup after IPv4 dns > lookup failures (default disabled) */ > #dns_try_ipv6=yes > > #disable dns to scale > dns=no > rev_dns=no > > /* uncomment the next line to disable the auto discovery of local aliases > based on revers DNS on IPs (default on) */ > #auto_aliases=no > alias=machinename.somedomain.com > > > > /* uncomment the following lines to enable TLS support (default o
[OpenSIPS-Users] [OpenSIPS] sample script that works with cdrtools, freeradius, nat, drouting
Hi, There: It looks ag-projects is maintaining the cdrtools, media proxy. but I searched around and didn't find anywhere there is a script that supports all the needed feature: cdrtools, mediaproxy, nat_traversal, and drouting. so now I'm trying to be a little brave and post my script that includes all above. this script doesn't handle instance message, but only voice calls. can any body spot problems with this script ? The goal of the script is to let locally registered user to use gateway to make outgoing call, and receive incoming call. the numbering plan is for US. free radius should have good authenticaing and accounting for different messages, and some special DID are mapped to several numbers and routed to asterisk. Hopefully this script will be useful for a general VOIP carrier. I try to paste the document to be comment. Hopefully, by going through this exercise, we can get a good starting script for people to use as a model starting script. Jimmy ### # # $Id: opensips.cfg,v 1.13 2009/05/11 06:06:00 jinsong Exp $ # # OpenSIPS basic configuration script # by Anca Vamanu # # Please refer to the Core CookBook at http://www.opensips.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # #INVITE :Invites a user to a call #ACK : Acknowledgement is used to facilitate reliable message exchange for INVITEs. #BYE :Terminates a connection between users #CANCEL :Terminates a request, or search, for a user. It is used if a client sends an INVITE and then changes its decision to call the recipient. #OPTIONS :Solicits information about a server's capabilities. #REGISTER :Registers a user's current location #INFO :Used for mid-session signaling #MESSAGE : IMS send message #SUBSCRIBE : IMS presence subscribe message #PUBLISH: IMS publish message #1xx: Provisional -- request received, continuing to process the request; #2xx: Success -- the action was successfully received, understood, and accepted; #3xx: Redirection -- further action needs to be taken in order to complete the request; #4xx: Client Error -- the request contains bad syntax or cannot be fulfilled at this server; #5xx: Server Error -- the server failed to fulfill an apparently valid request; #6xx: Global Failure -- the request cannot be fulfilled at any server. #This function sets the value of the flag given as parameter to 1 (true). The value of the parameter must be an integer between 0 and 31. ### Global Parameters # debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes #disable dns to scale dns=no rev_dns=no /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no alias=machinename.somedomain.com /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/etc/opensips/tls/user/user-cert.pem" #tls_private_key = "/etc/opensips/tls/user/user-privkey.pem" #tls_ca_list = "/etc/opensips/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want opensips to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 ### Modules Section #set module path mpath="/usr/lib/opensips/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "signaling.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded
