[Users] Re: OpenSER + Asterisk - Music On Hold
Edoardo Serra
edoardo.serra at webrainstorm.it
Fri Mar 16 11:05:40 CET 2007
Hi all,
I don't want to bother you about this problem again but I'm a little
bit stuck.
Someone can help ?
Tnx in advance
Regards
Edoardo
Edoardo Serra ha scritto:
> Tnx Carsten for attention.
>
> My openser.cfg is following
> I probably contains some misconfiguration as I'm still new to OpenSER
>
> Tnx in advance
>
> Regards
>
>
>
> # ----------- global configuration parameters ------------------------
> check_via=yes # (cmd. line: -v)
> dns=no # (cmd. line: -r)
> rev_dns=no # (cmd. line: -R)
> port=5060
> fifo="/tmp/ser_fifo"
>
> # ------------------ module loading ----------------------------------
>
> loadmodule "/usr/lib/openser/modules/sl.so"
> loadmodule "/usr/lib/openser/modules/tm.so"
> loadmodule "/usr/lib/openser/modules/rr.so"
> loadmodule "/usr/lib/openser/modules/maxfwd.so"
> loadmodule "/usr/lib/openser/modules/usrloc.so"
> loadmodule "/usr/lib/openser/modules/registrar.so"
> loadmodule "/usr/lib/openser/modules/nathelper.so"
> loadmodule "/usr/lib/openser/modules/textops.so"
> loadmodule "/usr/lib/openser/modules/exec.so"
> loadmodule "/usr/lib/openser/modules/uri.so"
> loadmodule "/usr/lib/openser/modules/uri_db.so"
> loadmodule "/usr/lib/openser/modules/dispatcher.so"
> loadmodule "/usr/lib/openser/modules/mysql.so"
> loadmodule "/usr/lib/openser/modules/auth.so"
> loadmodule "/usr/lib/openser/modules/auth_db.so"
>
> modparam("usrloc", "db_mode", 2)
> modparam("usrloc", "db_url",
> "mysql://XXX:XXX@192.168.252.5/openser")
> modparam("usrloc", "timer_interval", 120)
>
> modparam("auth_db", "calculate_ha1", 0)
> modparam("auth_db", "db_url",
> "mysql://XXX:XXX@192.168.252.5/userdb")
>
> modparam("uri_db", "db_url",
> "mysql://XXX:XXX@192.168.252.5/openser")
>
> modparam("rr", "enable_full_lr", 1)
>
> modparam("registrar", "nat_flag", 6)
> modparam("registrar", "max_expires", 3600)
> modparam("registrar", "min_expires", 60)
> modparam("registrar", "append_branches", 0)
> modparam("registrar", "desc_time_order", 1)
>
> modparam("nathelper", "natping_interval", 20) # Ping interval 20 s
> modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind
> NAT
>
> modparam("dispatcher", "force_dst", 1)
>
> # ------------------------- request routing logic -------------------
>
> # main routing logic
>
> route{
> # initial sanity checks -- messages with
> # max_forwards==0, or excessively long requests
> if (!mf_process_maxfwd_header("10")) {
> sl_send_reply("483","Too Many Hops");
> exit;
> };
> if ( msg:len > max_len ) {
> sl_send_reply("513", "Message too big");
> exit;
> };
>
> if ( (method=="OPTIONS") || (method=="SUBSCRIBE") ||
> (method=="NOTIFY") ) {
> sl_send_reply("405", "Method Not Allowed");
> exit;
> }
>
> if (!method=="REGISTER") {
> record_route();
> };
>
>
> if ((src_ip==111.222.333.444) || (src_ip==555.666.777.888)) {
> # IPs of our PSTN gateways
> if (!lookup("location")) {
> sl_send_reply("486", "Busy here");
> exit;
> };
> # forward to current uri now; use stateful forwarding; that
> # works reliably even if we forward from TCP to UDP
> if (!t_relay()) {
> sl_reply_error();
> };
> exit;
> };
>
> if (nat_uac_test("3")) {
> if ((method=="REGISTER") || (method=="INVITE") ||
> (method=="OPTIONS")) {
> fix_nated_contact();
> force_rport();
> setflag(6); # Mark as NATed
> }
> }
> # if the request is for other domain use UsrLoc
> # (in case, it does not work, use the following command
> # with proper names and addresses in it)
> if (method=="REGISTER") {
> if (!proxy_authorize("exorsa", "openser_view")) {
> proxy_challenge("exorsa", "0");
> exit;
> }
> if (!check_to()) {
> sl_send_reply("403", "Digest username and URI
> username do NOT match! Stay away!");
> exit;
> }
>
> save("location");
>
> exit;
> };
>
>
> if (method=="INVITE") {
> if (!proxy_authorize("exorsa", "openser_view")) {
> proxy_challenge("exorsa", "0");
> exit;
> }
>
> if (!check_from()) {
> sl_send_reply("403", "Digest username and URI username do
> NOT match! Stay away!");
> exit;
> }
> }
>
> # loose-route processing
> if (loose_route()) {
> # mark routing logic in request
> append_hf("P-hint: rr-enforced\r\n");
> route(1);
> exit;
> };
>
> if (!uri==myself) {
> # mark routing logic in request
> append_hf("P-hint: outbound\r\n");
> route(1);
> exit;
> };
>
> append_hf("P-hint: usrloc applied\r\n");
> route(1);
> }
>
> route[1]
> {
> # !! Nathelper
> if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)"
> && !search("^Route:")){
> sl_send_reply("479", "We don't forward to private IP
> addresses");
> exit;
> };
>
> # NAT processing of replies; apply to all transactions (for
> example,
> # re-INVITEs from public to private UA are hard to identify as
> # NATed at the moment of request processing); look at replies
> t_on_reply("1");
>
> # send it out now; use stateful forwarding as it works reliably
> # even for UDP2TCP
> if ((src_ip!=111.222.333.444) && (src_ip!=555.666.777.888)) {
> #IPs of our PSTN Gateways
> ds_select_dst("1", "0");
> }
>
> if (!t_relay()) {
> sl_reply_error();
> };
> }
>
> # !! Nathelper
> onreply_route[1] {
> # NATed transaction ?
> if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
> fix_nated_contact();
> # otherwise, is it a transaction behind a NAT and we did not
> # know at time of request processing ? (RFC1918 contacts)
> } else if (nat_uac_test("1")) {
> fix_nated_contact();
> };
> }
> Carsten Bock ha scritto:
>> Hi Edoardo,
>>
>> Normally this would be handled by an Record-Route/Loose-Route construct.
>> When doing record-routing/loose-routing, the in-dialog request
>> "Re-INVITE" (for Music-On-Hold) should take the same route as the
>> initial request (following the route headers) and you should no longer
>> need to query the dispatcher-module for these in-dialog requests.
>> Maybe you could post your config, i guess then we could help a little
>> more.
>>
>> Carsten
>>
>>
>> Am Montag, den 12.03.2007, 13:10 +0100 schrieb Edoardo Serra:
>>> Daniel,
>>> thanks for your interest in the problem.
>>>
>>> I better analyzed the problem and found the point in it.
>>> I try to describe where I guess the problem is
>>>
>>> When one of our users receive a call from the PSTN, the PSTN Gateway
>>> (Asterisk) sends an INVITE at username at openser, the INVITE is
>>> correctly forwarded to the user and the call is set up without problems.
>>> (RTP from PSTN gw to USER and SIP through OpenSER)
>>>
>>> When the user wants to put the caller OnHold it sends an INVITE to
>>> OpenSER but OpenSER forwards the INVITE to one of the PSTN GW using
>>> dispatcher module.
>>> This way, if the INVITE is not forwarded to the PSTN GW which is
>>> handling the call a second call is generated.
>>>
>>> Do you have any suggestion ?
>>> Every kind of help is appreciated.
>>>
>>> Sorry for not having sent a network capture, but is quite difficult to
>>> prepare such a capture on our system because it's always very busy
>>>
>>> Hoping to hear from you soon
>>>
>>> Regards
>>>
>>> Edoardo
>>>
>>>
>>> Daniel-Constantin Mierla ha scritto:
>>>> Hello,
>>>>
>>>> a network trace (ngrep or wireshark) will help to spot what might be
>>>> the problem, otherwise is hard to guess.
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>>
>>>> On 03/04/07 17:32, Edoardo Serra wrote:
>>>>> Hi all,
>>>>> I have an OpenSER server in front of serveral Asterisk acting
>>>>> as a load balancer and registrar server.
>>>>>
>>>>> We're offering both, inbound and outbound call services.
>>>>>
>>>>> When an outbound call is made, OpenSER, through the dispatcher
>>>>> module, choose an Asterisk server to handle the media of the call.
>>>>>
>>>>> When an inbound call is received (by a PSTN GW interconnected with
>>>>> one of the Asterisks), Asterisk calls SIP/username at openser.
>>>>>
>>>>> Media flows directly from user to Asterisks without using RTPProxy
>>>>> as every Asterisk server has got a public IP Address..
>>>>>
>>>>> I have the following problem with MOH.
>>>>>
>>>>> If a user tries to put on hold an outbound call (placed by him)
>>>>> everything is OK, Asterisk start playing MOH and stops when the
>>>>> user wants to stop it.
>>>>>
>>>>> But, if a user wants to put on hold an inbound call (a call just
>>>>> answered), as soon as it press the hold button another call to the
>>>>> caller is originated and the first call is not put on hold by the
>>>>> Asterisk
>>>>>
>>>>> I guess the problem is that, in this case, the asterisk doesn't
>>>>> recognise the INVITE as a re-INVITE and originate a new call
>>>>> instead of putting the other on hold.
>>>>>
>>>>> Do you have any idea on how to solve the problem ?
>>>>> Every suggestion is appreciated.
>>>>>
>>>>> Regards
>>>>>
>>>>> Edoardo Serra
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at openser.org
>>>>> http://openser.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at openser.org
>>> http://openser.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
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