Re: [OpenSIPS-Users] RTPProxy + OpenSIPS 1.7 Integration
After resolving the incorrect rtp proxy -l flag, i'm still having a
hard time getting two way audio behind a router.
What I have in my config is the following:
./rtpproxy -s udp:127.0.0.1:7789 -l 192.168.2.102 -m 1 -M 2 -u
root root -F -d INFO LOG_LOCAL0
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7789")
route{
if (is_method("INVITE") && has_totag())
engage_rtp_proxy("ie","192.168.2.102");
if (is_method("ACK") && has_body("application/sdp")) rtpproxy_answer();
}
route[1] {
if (is_method("INVITE")) {
xlog("Start Call for route [ fu=$fu/ tu=$tu /ru=$ru/ ci=$ci]");
if (has_body("application/sdp")) {
if (rtpproxy_offer("ie","192.168.2.102"))
t_on_reply("1");
else t_on_reply("2");
}
t_on_branch("2");
t_on_failure("1");
}
...
}
onreply_route[1] {
xlog("incoming reply\n");
if (has_body("application/sdp")) rtpproxy_answer();
exit;
}
onreply_route[2] {
xlog("incoming reply\n");
if (has_body("application/sdp")) rtpproxy_offer();
exit;
}
I do have two way audio once every 30-50 calls, but cannot pin any
combination that would keep things
working. Outgoing audio always works. I really appreciate any help on this.
Kind Regards,
Nick.
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Re: [OpenSIPS-Users] exec_avp in timer route
Hmmm thats what I am doing using startup and timer routes except I don't use avp query to get data I use exec_avp. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/exec-avp-in-timer-route-tp7103327p7113390.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] exec_avp in timer route
On Tue, Dec 20, 2011 at 1:47 PM, ddgiants wrote: > Hey Brett - thanks for the communication. So you are using avp_db_queries? > That's what i was using but the issue is if I change anything in startup I > have to change it in timer as well Using the script I change it once. > What I normally do is use startup route to push objects into memcache for something like 10 minutes. Then run a timer route every 5 minutes and update the memcache. This keeps the cache fresh and lets objects expire as they disappear. Your requirements may require something else :) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] exec_avp in timer route
Hey Brett - thanks for the communication. So you are using avp_db_queries? That's what i was using but the issue is if I change anything in startup I have to change it in timer as well Using the script I change it once. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/exec-avp-in-timer-route-tp7103327p7113056.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] exec_avp in timer route
On Sat, Dec 17, 2011 at 6:53 AM, ddgiants wrote: > FYI - I changed lines 69 & 71 in modules/exec/exec_mod.c and it now loads > with exec_avp in time_route. I have absolutely no idea what ramifications > this has though. Just an FYI. > > Old line 69 & 71 > REQUEST_ROUTE|FAILURE_ROUTE|LOCAL_ROUTE|STARTUP_ROUTE}, > > New line > REQUEST_ROUTE|FAILURE_ROUTE|LOCAL_ROUTE|STARTUP_ROUTE|TIMER_ROUTE}, For what it's worth, I've done the exact same thing in an older version to make exec_avp work in startup_route. I also get the URI failures on startup, but I'm sure that's because the exec_avp method is expecting to parse a message. I get the error, but no obvious ramifications. Doesn't mean there arn't any. :) Been doing it for a while. My new code, however I'm using startup and timer routes more and doing my logic right in the code instead of external scripts (like loading accounts and such) ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] makeann problem
Hi guys, I'm trying to use rtpproxy_stream2uac, but everytime that I try to make the announcement file using the "makeann", the voice change, and the file turn into something slow. I'm just using, #makeann ann_.wav then two files appears, ann_.wav.0 and ann_.wav.8 There's anything else that I have to do? Eng.º Rodrigo Ferreira Supervisor de Telecomunicações Fone: (13) 4010-1037 Cel: (13) 9615-7774 VIPWay Serviços de Telecomunicações LTDA___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] RTPProxy + OpenSIPS 1.7 Integration
Hello Bogdan, Thank you so much for your response and time. I've been really struggling trying to get two way audio going. I think at some point I broke down in tears! This is a test virtual machine setup before migrating to the host servers. What I have is the following Router--->OpenSIPS/RTPProxy> Asterisk 1..n---> WAN ITSP Router 192.168.2.1 UC Polycom: 192.168.2.11 OpenSIPS: 192.168.2.102 Asterisk 1: 192.168.2.10 Asterisk 2: 192.168.2.11 My frist question is, do I need NAT and/or RTP Proxy since everything but the ITSP is on the same subnet? If not, I guess there is something wrong with my configuration. Eventually I will need NAT, and RTP, so I might as well try to get a grasp, and correct configuration using them. Can RTP proxy work when behind a router? Kind Regards, Nick. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] About msilo module config
Hi Kevin, The idea is simple - in script, each time you figured out you cannot deliver the MESSAGE (like callee is not registered, callee rejected or timeout in failure route), you can use the m_store() function : http://www.opensips.org/html/docs/modules/1.7.x/msilo.html#id293116 Regards, Bogdan On 12/20/2011 03:31 AM, Kevin wrote: Hi Friends I'm so sorry to trouble you. Who can tell me how to configure the msilo module to store all of the off line messages? Or give me a configure example? Thank you very much Regards,Kevin ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer OpenSIPS solutions and "know-how" ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] RTPProxy + OpenSIPS 1.7 Integration
Hi Nick,
are you sure the public and private IPs are local to the machine where
you run rtpproxy ? as I see that rtpproxy complain on not being able to
use the ips you configured via the "-l" param.
Regards,
Bogdan
On 12/20/2011 04:45 AM, Nick Khamis wrote:
Hello Everyone,
I am having trouble getting the RTPProxy 1.2.0 to work with OpenISP 1.7.
Starting the proxy using:
./rtpproxy -s udp:127.0.0.1:12221 -l public/private -p
/var/run/rtpproxy.pid -u root root -F -d INFO LOG_LOCAL0
opensips.cf relevant pieces
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:12221")
modparam("rtpproxy", "rtpproxy_autobridge", 1)
modparam("rtpproxy", "rtpproxy_timeout", "0.5")
modparam("rtpproxy", "rtpproxy_retr", 3)
if (has_totag()&& is_method("INVITE")) {
engage_rtp_proxy("ie");
}
if (is_method("ACK")&& has_body("application/sdp")) {
rtpproxy_answer();
}
route[1] {
xlog("Enter route 1");
if (has_body("application/sdp")) rtpproxy_answer();
if (is_method("INVITE")) {
xlog("New Call for route [ fu=$fu/ tu=$tu /ru=$ru/ ci=$ci]");
if (rtpproxy_offer()) {
t_on_reply("1");
}
else {
t_on_reply("2");
}
t_on_branch("2");
t_on_failure("1");
}
if (!t_relay()) {
sl_reply_error();
}
exit;
}
onreply_route[1] {
xlog("incoming reply\n");
if (has_body("application/sdp")) rtpproxy_answer();
exit;
}
onreply_route[2] {
xlog("incoming reply\n");
if (has_body("application/sdp")) rtpproxy_offer();
exit;
}
When starting OpenSIPS everything looks fine:
Dec 19 21:38:58 [3397] INFO:rtpproxy:rtpp_test: rtp proxy
found, support for it enabled
Dec 19 21:38:58 [3400] INFO:rtpproxy:rtpp_test: rtp proxy
found, support for it enabled
Dec 19 21:38:58 [3398] INFO:rtpproxy:rtpp_test: rtp proxy
found, support for it enabled
Dec 19 21:38:58 [3395] INFO:rtpproxy:rtpp_test: rtp proxy
found, support for it enabled
Dec 19 21:38:58 [3401] INFO:rtpproxy:rtpp_test: rtp proxy
found, support for it enabled
Dec 19 21:38:58 [3402] INFO:rtpproxy:rtpp_test: rtp proxy
found, support for it enabled
Making a call howver, yields the following error (watch it work now ;):
ERROR:rtpproxy:force_rtp_proxy_body: no available proxies
/var/log/syslog
Dec 19 21:28:18 opensips1 rtpproxy[3348]: INFO:main: rtpproxy started, pid 3348
Dec 19 21:28:31 opensips1 rtpproxy[3348]: INFO:handle_command: new
session f679c215-bae17257-a0a8a5b8@192.168.2.11, tag C0847B09-9D90A2;1
requested, type strong
Dec 19 21:28:31 opensips1 rtpproxy[3348]: ERR:create_twinlistener:
can't bind to the IPv4 port 50570: Cannot assign requested address
Dec 19 21:28:31 opensips1 rtpproxy[3348]: ERR:handle_command: can't
create listener
Dec 19 21:38:49 opensips1 rtpproxy[3379]: INFO:main: rtpproxy started, pid 3379
Dec 19 21:39:36 opensips1 rtpproxy[3379]: INFO:handle_command: new
session 5537fa8e-69de2248-78eef465@192.168.2.11, tag
A2311CB2-BB413627;1 requested, type strong
Dec 19 21:39:36 opensips1 rtpproxy[3379]: ERR:create_twinlistener:
can't bind to the IPv4 port 57636: Cannot assign requested address
Dec 19 21:39:36 opensips1 rtpproxy[3379]: ERR:handle_command: can't
create listener
Your help is greatly appreciated,
Nick.
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Re: [OpenSIPS-Users] Blank domain field in subscriber table
Hi Schneur, Have you tried to disable the use_domain param in auth_db module: http://www.opensips.org/html/docs/modules/1.7.x/auth_db.html#id250091 and to force a fix value for the "realm" param for both challenge and authorize functions ? Regards, Bogdan On 12/19/2011 07:14 PM, Schneur Rosenberg wrote: here is my problem, my server has multiple domain names, different customers register with a different domain name (we combined a few servers into one, therefore some register with different names) now as long as I dont have the correct domain set in the subscriber table, the system will keep on replying with a 401 Unauthorized. I assume it has to do with the way ha1 passwords are handled, I set the system to use plain text passwords, I also tried setting static challenge for www_authorize, I thought this will eliminate the need for the domain in the challenge, but still no registration. in my debug I get the following 2 lines Dec 19 09:11:06 opensipsmaster /sbin/opensips[4937]: DBG:db_mysql:db_mysql_convert_rows: no rows returned from the query Dec 19 09:11:06 opensipsmaster /sbin/opensips[4937]: DBG:auth_db:get_ha1: no result for user 'gra...@sipsvr1.mydomain.com' Schneur Rosenberg ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer OpenSIPS solutions and "know-how" ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to change P-Preferred-Identity
Hi Nick,
You can simply remove the header and add a new one preserving the URI
part, like :
remove_hf("P-Preferred-Identity");
append_hf('P-Preferred-Identity: "my_display_name"
<$(hdr(P-Preferred-Identity){nameaddr.uri})>\r\n');
Regards,
Bogdan
On 12/20/2011 11:26 AM, Nick wrote:
Hello
I want to change P-Preferred-Identity with header.
Now, My P-Preferred-Identity is
I want to change display name.
Can everyone give me a suggestion?
Thanks
Nick
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OpenSIPS solutions and "know-how"
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[OpenSIPS-Users] How to change P-Preferred-Identity
Hello I want to change P-Preferred-Identity with header. Now, My P-Preferred-Identity is I want to change display name. Can everyone give me a suggestion? Thanks Nick ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Blank domain field in subscriber table
Hi,
if i understand this correctly, this should be easy to solve if all your
domains point to one ip. The www_challenge function has a parameter
"realm" if you set this to your ip (where all your domains / subdomains
point to) and also add this IP as domain for every user in subscriber
table it should work out of the box?
www_authorize("yourIP", "subscriber")) {
Best Regards
Max M.
Am 19.12.2011 23:04, schrieb Andreas Sikkema:
I don't quite remember how we did it, but at a previous employer we migrated
all our users from one domain to another by 'rewriting' all incoming messages
to the new domain. All the accounts were converted to the be domain using a
script just before going live
We didn't quite rewrite, by substituted the domain when authenticating and
perhaps when routing calls but it's been at least 7 years...
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