[OpenSIPS-Users] OpenSIPS scalability with TLS connections.
Dear all, OpenSIPS can achieve extremely high performance according to the tests provided by the official site. However, it seems that the results were based on UDP. I am trying to use OpenSIPS to set up a scalable SIP server which aims to support tens of thousands SIP clients using TLS. Such a large number of SIP clients would send REGISTER requests in different time slots, but each SIP client keeps its TLS connection alive. Due to my own limitation, I can establish about 2000 TLS connections between SIP clients and OpenSIPS server. I am curious that has anyone employed OpenSIPS to build a scalable SIP server with so many lasting TLS connections (e.g., 5 TLS connections)? Any comment is greatly appreciated. Best wishes, Chen-Che -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-scalability-with-TLS-connections-tp7595800.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] Changing Address of Opensips Server
Hi David,
Are you sure it is not a port issue ? maybe the src port does not match
the port your have in the address table.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 26.02.2015 15:38, David Crow wrote:
I’ve done some more troubleshooting on this and it doesn’t appear to
have anything to do with the NAT. The issue seems to be related to
one host sending calls to Opensips. From all other hosts it works
fine, but this one host doesn’t work at all. I’ve double checked the
ip address, port, etc. I”ve also tried removing the address from the
table and adding it back, still nothing. Very strange.
-David
*From:* users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] *On Behalf Of *David Crow
*Sent:* Tuesday, February 24, 2015 5:10 PM
*To:* 'users@lists.opensips.org'
*Subject:* [OpenSIPS-Users] Changing Address of Opensips Server
I’m moving our one of our opensips server to a new datacenter with new
ip addresses and I’m running into an issue.
It seems to be failing to find the group properly based on the 503
message I’m getting back.
$avp(s:group) = get_source_group();
# Reject if no group found for this call
if ($avp(s:group) == -1) {
sl_send_reply(503,Service Unavailable - NOPERM);
exit;
}
This server is behind a firewall doing NAT and I’ve changed all
references in the opensips.cfg to the new external address.
Opensipsctl address dump shows the address table and the server that’s
sending the calls from properly.
I’m not really sure where to look next, I’m pretty green when it comes
to opensips. I only have to tinker with it when I change something
like this. Any assistance would be greatly appreciated.
*David Crow**| **Senior Systems Architect*
*1301 Gervais Street, Suite 1800 | Columbia, SC 29201*
*(d) 803.978.2727 | (f) 803.733.5888*
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Re: [OpenSIPS-Users] Destination IP in the Branch and Reply Routes
You can use nat_uac_test() Take a look at http://www.opensips.org/html/docs/modules/1.11.x/nathelper.html Il 08/Mar/2015 02:25 Terrance Devor ter.de...@gmail.com ha scritto: Hello Everyone, Our opensips is in a nat'ed environment , and we need a way to test if the destination ip (ie, where the next message is going to be sent) is public or private. Is there a variable, or even better a function that can test for this. Kind Regards, Terrance ___ 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
Re: [OpenSIPS-Users] Planning OpenSIPS 2.1 release - heads up
Fixed ! Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 09.03.2015 02:28, Satish Patel wrote: Bogdan, I am running 2.1.x and so far great, I had issue with sipteace with homer which I already reported. So please look into it before release. -- Sent from my iPhone On Mar 8, 2015, at 7:02 PM, Terrance Devor ter.de...@gmail.com wrote: Good news, What is rtpengine support. Will the proxy manage RTP directly? Or will we still have to use RTP/Media Proxy Terrance ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] TLS handling
Does someone have any idea how to handle it? 05.03.2015, 16:22, "Чалков Артём" achal...@ya.ru:debug=4Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:tcp_read_req: We're releasing the connection in state 3Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:io_watch_del: io_watch_del op on index -1 36 (0x77dee0, 36, -1, 0x10,0x1) fd_no=2 calledMar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:release_tcpconn: releasing con 0x7f2be91663a8, state 0, fd=36, id=1Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:release_tcpconn: extra_data (nil)Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_msg: SIP Request:Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_msg: method: MESSAGEMar 5 16:07:46 cs17792 /usr/sbin/opensips[19028]: DBG:core:handle_tcp_child: reader response= 7f2be91663a8, 0 from 0Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_msg: uri: sip:achalkov1@x.x.x.x:3631;transport=TCPMar 5 16:07:46 cs17792 /usr/sbin/opensips[19028]: DBG:core:io_watch_add: io_watch_add op on 52 (0x77dd80, 52, 2, 0x7f2be91663a8,1), fd_no=38Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_msg: version: SIP/2.0Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_headers: flags=2Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_via_param: found param type 232, branch = z9hG4bK-d8754z-668ef50b1a4c0a31-1---d8754z-; state=6Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_via_param: found param type 235, rport = n/a; state=17Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_via: end of header reached, state=5Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_headers: via found, flags=2Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_headers: this is the first viaMar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:receive_msg: After parse_msg...Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:receive_msg: preparing to run routing scripts...Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_headers: flags=800Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_to: end of header reached, state=10Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_to: display={}, ruri={sip:achalkov1@x.x.x.x:3631;transport=TCP}Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:get_hdr_field: To [51]; uri=[sip:achalkov1@x.x.x.x:3631;transport=TCP]Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:get_hdr_field: to body [sip:achalkov1@x.x.x.x:3631;transport=TCP#015#012]Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:get_hdr_field: cseq CSeq: 3 MESSAGEMar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:maxfwd:is_maxfwd_present: value = 70Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_headers: flags=Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:get_hdr_field: content_length=3Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:get_hdr_field: found end of headerMar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_headers: flags=Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:decode_mime_type: Decoding MIME type for:[text/plain]Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_to_param: tag=b2b91161Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_to: end of header reached, state=29Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_to: display={"achalkov"}, ruri={sip:achalkov@x.x.x.x:3631;transport=TCP}Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_headers: flags=Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_methods: methods 0x173FMar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:uri:has_totag: no totagMar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_headers: flags=78Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:tm:t_lookup_request: start searching: hash=32018, isACK=0Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:tm:matching_3261: RFC3261 transaction matching failedMar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:tm:t_lookup_request: no transaction foundMar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:core:parse_headers: flags=200Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:rr:find_first_route: No Route headers foundMar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:rr:loose_route: There is no Route HFMar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:auth:check_nonce: comparing [54f85536be0ae02858c2001d58774c222d958f86] and [54f85536be0ae02858c2001d58774c222d958f86]Mar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:db_mysql:has_stmt_ctx: ctx found for subscriberMar 5 16:07:46 cs17792 /usr/sbin/opensips[19012]: DBG:db_mysql:db_mysql_do_prepared_query: conn=0x7f2bef4e28b0
Re: [OpenSIPS-Users] ERROR:siptrace:pipport2su: bad protocol
Just to let people know, this was fixed in 2.1 (it was triggered by adding the new websocket protocol). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 10.03.2015 18:42, Satish Patel wrote: I set debug=6 and here is the logs most of time error pop'ed up near tm module DBG:tm:run_trans_callbacks: trans=0x7f9570f06710, callback type 1024, id 1 entered DBG:siptrace:trace_onreq_out: trace on req out DBG:core:parse_headers: flags=40 DBG:siptrace:trace_msg_out: trace msg out ERROR:siptrace:pipport2su: bad protocol On Tue, Mar 10, 2015 at 12:12 PM, Satish Patel satish@gmail.com mailto:satish@gmail.com wrote: I am configuring siptrace with homer server and getting following error, ERROR:siptrace:pipport2su: bad protocol ERROR:siptrace:pipport2su: bad protocol ERROR:siptrace:pipport2su: bad protocol Razvan fix following patch but still getting above error in log https://github.com/OpenSIPS/opensips/commit/178b0cc26b05a81947de150fe1c2df36d600ccaa Am i missing something? ___ 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
[OpenSIPS-Users] Opensips-CP
To whom it may concern, I am looking to implement the Opensips-cp system on top of our current Opensips servers/dbs. Do you offer a consulting contract to help guide us through this to help prevent us from having downtime? If so what is the cost and how long would it take? Can we schedule a call to discuss? Thank you, John ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] PUA_USRLOC
Hello,
maybe I don't understood completely how pua_usrloc module work. I'm
trying using this module without success.
I understood I have to put the pua_set_publish function on the REGISTER
block, like:
if (is_method(REGISTER)) {
pua_set_publish();
Only when the user is registered and his registration is present on the
location table, OpenSIPs generate a PUBLISH SIP MESSAGE that save on the
presentity table?; so if another user subscribe for status of this user,
he see the user online.
I'm trying with X-Lite and 3CX without success.
I think en both case, the problem is here:
Mar 12 10:28:06 li926-75 /sbin/opensips[24300]: DBG:pua:publ_cback_func:
Record not found
Any hint?
Regards
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Re: [OpenSIPS-Users] OpenSIPS and Apple Push Notification
Hello Bogdan, Wow!!! It took me so much to understand this... hahaha I just need the last help: - considering that the UAC has an unlimited expiry (so the lookup will return 1) - my actual configuration is trying to send the INVITE to the contact in the DB but it will timeout (because the UAC is not connected anymore but it has unlimited expiry). - after this time out i should send the PN and wait for the UAC to re-register and then establish a new INVITE with this new info. Could you give me a couple of ideas on how to implement this? Thanks a lot! Leo -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-and-Apple-Push-Notification-tp7591783p7595564.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] Planning OpenSIPS 2.1 release - heads up
The D day for releasing 2.1 is 18th of March ! Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 24.02.2015 16:22, Bogdan-Andrei Iancu wrote: Hello all, Everybody is looking forward for the next OpenSIPS release, the major 2.1, which is planned for end of February - at least this was the plan so far. I our desire to make of 2.1 a radical improvement, we committed to several major changes/redesigns and new valuable functionalities. And they do burn time to get them done, especially as we want them : in the best possible way. In oder to finish all we committed for, the release date is now estimated for first half of March - some important parts of code still need to be settled down and we do not want to make any kind of compromise in quality. Just to give you a heads up on what OpenSIPS 2.1 will bring: - internal re-design around async reactors for load-balancing tasks inside OpenSIPS - async I/O support in scripts for db queries, exec, rest client - refactoring of the networking and protocol related code (protos are now modules) - webSockets support - Quality Based Routing new module - Compression new module - Fraud Detection new module - Emergency Call handling new module - rtpengine support - partitioning support in Dynamic Routing, Diaplan and Dispatcher and many other - when the coding taks is done, we will proceed withe bothersome task of updating docs, new and migration. Whoever gives a try to 2.1 version, please report any problems or crashes asap to us ! better have them done now rather than later :P Best regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Changing Address of Opensips Server
I have 0 in as the port, I've also tried putting 5060 to see if that makes a
difference.
-David
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: Friday, February 27, 2015 2:20 PM
To: OpenSIPS users mailling list; David Crow
Subject: Re: [OpenSIPS-Users] Changing Address of Opensips Server
Hi David,
Are you sure it is not a port issue ? maybe the src port does not match the
port your have in the address table.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 26.02.2015 15:38, David Crow wrote:
I've done some more troubleshooting on this and it doesn't appear to have
anything to do with the NAT. The issue seems to be related to one host sending
calls to Opensips. From all other hosts it works fine, but this one host
doesn't work at all. I've double checked the ip address, port, etc. Ive also
tried removing the address from the table and adding it back, still nothing.
Very strange.
-David
From: users-boun...@lists.opensips.orgmailto:users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of David Crow
Sent: Tuesday, February 24, 2015 5:10 PM
To: 'users@lists.opensips.orgmailto:users@lists.opensips.org'
Subject: [OpenSIPS-Users] Changing Address of Opensips Server
I'm moving our one of our opensips server to a new datacenter with new ip
addresses and I'm running into an issue.
It seems to be failing to find the group properly based on the 503 message I'm
getting back.
$avp(s:group) = get_source_group();
# Reject if no group found for this call
if ($avp(s:group) == -1) {
sl_send_reply(503,Service Unavailable - NOPERM);
exit;
}
This server is behind a firewall doing NAT and I've changed all references in
the opensips.cfg to the new external address.
Opensipsctl address dump shows the address table and the server that's sending
the calls from properly.
I'm not really sure where to look next, I'm pretty green when it comes to
opensips. I only have to tinker with it when I change something like this.
Any assistance would be greatly appreciated.
David Crow | Senior Systems Architect
1301 Gervais Street, Suite 1800 | Columbia, SC 29201
(d) 803.978.2727 | (f) 803.733.5888
david.c...@vc3.commailto:david.c...@vc3.com| www.VC3.comhttp://www.vc3.com/
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Re: [OpenSIPS-Users] TLS handling
Yes, i sure. I have only 1 branch because i have only one instance to receive
sent MESSAGE and i dont add branches manually. Actually, that is all MESSAGE
routing:
...
if (is_method(MESSAGE)) route(MESSAGE);
...
route[MESSAGE] {
...
if (!lookup(location)) {
...
} else {
t_on_reply(ON_REPLY);
if ($si != 127.0.0.1) t_on_failure(ON_FAIL_MESSAGE);
t_relay(0x01);
$var(retcode) = $retcode;
xlog(L_INFO, [!!!MESSAGE_DEBUG!!!] t_relay returns
$var(retcode) LOGHDR);
ifdef(`LOGS', `xlog(L_INFO, [MESSAGE] Request is leaving
server LOGHDR);')
exit;
}
}
and that is debug=4 log from moment, when i killed softphone (destination of
MESSAGE) without sip unregistering, but with removing tcp/tls connection:
Mar 6 13:40:57 cs17792 /usr/sbin/opensips[6695]: DBG:stun:receive: Sending:
from [9 x.x.x.x 3631] to [y.y.y.y 50693]
Mar 6 13:40:57 cs17792 /usr/sbin/opensips[6695]: DBG:stun:receive: #012#012
Mar 6 13:40:57 cs17792 /usr/sbin/opensips[6695]: DBG:stun:freeStunMsg: freeing
Mar 6 13:40:57 cs17792 /usr/sbin/opensips[6695]: DBG:stun:freeStunMsg: freeing
Mar 6 13:40:57 cs17792 /usr/sbin/opensips[6695]: DBG:stun:stun_loop: READING
Mar 6 13:40:57 cs17792 /usr/sbin/opensips[6713]: DBG:tm:timer_routine: timer
routine:2,tl=0x7ff79cd256b0 next=(nil), timeout=12
Mar 6 13:40:57 cs17792 /usr/sbin/opensips[6713]: DBG:tm:wait_handler: removing
0x7ff79cd25630 from table
Mar 6 13:40:57 cs17792 /usr/sbin/opensips[6713]: DBG:tm:delete_cell: delete
transaction 0x7ff79cd25630
Mar 6 13:40:57 cs17792 /usr/sbin/opensips[6713]: DBG:tm:wait_handler: done
Mar 6 13:41:02 cs17792 /usr/sbin/opensips[6698]: DBG:core:udp_rcv_loop:
probing packet received len = 2
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6734]: DBG:core:handle_tcpconn_ev:
data available on 0x7ff79cd2c578 53
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6734]: DBG:core:io_watch_del:
io_watch_del op on index -1 53 (0x77dd80, 53, -1, 0x0,0x1) fd_no=40 called
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6734]: DBG:core:send2child: to tcp
child 0 0(6717), 0x7ff79cd2c578 rw 1
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6717]: DBG:core:handle_io: We have
received conn 0x7ff79cd2c578 with rw 1
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6717]: DBG:core:io_watch_add:
io_watch_add op on 36 (0x77dee0, 36, 2, 0x7ff79cd2c578,1), fd_no=1
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6717]: DBG:core:tcp_read_req: Using
the global ( per process ) buff
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6717]: DBG:core:tls_update_fd: New
fd is 36
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6717]: ERROR:core:_tls_read: SYSCALL
error - (104) Connection reset by peer
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6717]: ERROR:core:_tls_read: TLS
connection to y.y.y.y:58858 read failed
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6717]: ERROR:core:_tls_read: TLS
read error: 5
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6717]: DBG:core:tcp_read_req: read=
-1 bytes, parsed=0, state=0, error=1
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6717]: DBG:core:tcp_read_req: last
char=0x00, parsed msg=#012
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6717]: ERROR:core:tcp_read_req:
failed to read
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6717]: DBG:core:io_watch_del:
io_watch_del op on index -1 36 (0x77dee0, 36, -1, 0x10,0x3) fd_no=2 called
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6717]: DBG:core:release_tcpconn:
releasing con 0x7ff79cd2c578, state -2, fd=36, id=2
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6717]: DBG:core:release_tcpconn:
extra_data 0x7ff79cd2c6f8
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6734]: DBG:core:handle_tcp_child:
reader response= 7ff79cd2c578, -2 from 0
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6734]: DBG:core:tcpconn_destroy:
destroying connection 0x7ff79cd2c578, flags 0002
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6734]: DBG:core:tls_close: closing
TLS connection
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6734]: DBG:core:tls_update_fd: New
fd is 53
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6734]: ERROR:core:tls_shutdown:
something wrong in SSL:
Mar 6 13:41:03 cs17792 /usr/sbin/opensips[6734]: DBG:core:tls_tcpconn_clean:
entered
Mar 6 13:41:05 cs17792 /usr/sbin/opensips[6705]: DBG:msilo:m_clean_silo:
cleaning stored messages - 20
Mar 6 13:41:07 cs17792 /usr/sbin/opensips[6715]: DBG:avpops:ops_dbquery_avps:
query [SELECT username FROM silo GROUP BY username HAVING count(*) 0]
Mar 6 13:41:07 cs17792 /usr/sbin/opensips[6715]:
DBG:db_mysql:mysql_raise_event: MySQL status has not changed: connected
Mar 6 13:41:07 cs17792 /usr/sbin/opensips[6715]: DBG:core:db_new_result:
allocate 48 bytes for result set at 0x7ff7a39ac9c8
Mar 6 13:41:07 cs17792 /usr/sbin/opensips[6715]:
DBG:db_mysql:db_mysql_get_columns: 1 columns returned from the query
Mar 6 13:41:07 cs17792 /usr/sbin/opensips[6715]: DBG:core:db_allocate_columns:
allocate 28 bytes for
Re: [OpenSIPS-Users] Planning OpenSIPS 2.1 release - heads up
Hi Terrance, RTPengine is an alternative to MediaProxy or RTPproxy. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 09.03.2015 01:02, Terrance Devor wrote: Good news, What is rtpengine support. Will the proxy manage RTP directly? Or will we still have to use RTP/Media Proxy Terrance ___ 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
Re: [OpenSIPS-Users] Announcing rtpproxy v2.0.0
Hi, Maxim, Is there any plans for rtp header compression in future. I can't see anything in the change log for 2.0.0 On Tuesday, 10 March 2015, Maxim Sobolev sobo...@sippysoft.com wrote: Hi All, I'm happy to announce that we have released rtpproxy v2.0.0. You can review the release notes here: https://github.com/sippy/rtpproxy/releases/tag/v2.0.0 -sobomax -- Sent from iPhone 6 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Announcing rtpproxy v2.0.0
Good news. Can install it into my OpenSIPS 1.8 server and click play? Or are there modifications needed in opensips.cfg ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Destination IP in the Branch and Reply Routes
Hello Danilo, Thank you for your response. We use nat_uac_test() in the config file for originating clients, unfortunately this is focused towards the destination (ie, the IP address of which the next signal is going to be sent to) T ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Problem with opensips core dumping on high load
Hello,
I recognized opensips coredumping in the latest version when being under high
load. The problem is an invalid pointer when iterating over the user location
in memory table (DB mode is 0). Below the backtrace of where this happen. Note:
we added an extra NULL check in the line to make sure that _r is not null. But
as u can see the one is still crashing so we assume that the location table got
corrupted – pointer not null but pointing to nirwana.
We are load testing only register cycles (register/401/register/200). After the
trace the current config … The important part is the “route[register_request]”.
Tests have shown that if we strip off the cachedb calls or using cache local
instead of cache_mysql – this not happening. Could it be a conflict in the
timer procedure used by both modules? The cache needs to be distributed –
that’s why cache_mysql was chosen.
Any thoughts?
Core was generated by `/opt/opensips/sbin/opensips -P /var/run/opensips.pid -w
/opt/opensips -m 8192 -'.
Program terminated with signal SIGSEGV, Segmentation fault.
#0 nodb_timer (_r=0x7f8f) at urecord.c:223
223 if (!_r-contacts) return 0;
(gdb) #0 nodb_timer (_r=0x7f8f) at urecord.c:223
ptr = optimized out
t = optimized out
#1 timer_urecord (_r=_r@entry=0x7f8f,
ins_list=ins_list@entry=0x7f8fed2ec480) at urecord.c:367
No locals.
#2 0x7f91eed428ac in mem_timer_udomain (_d=0x7f8fed2ec478) at udomain.c:791
ptr = 0x7f8f
dest = optimized out
i = 61
ret = optimized out
flush = 0
it = {node = 0x7f8ff14136c8, map = 0x7f8fed2f0238}
__FUNCTION__ = mem_timer_udomain
#3 0x7f91eed356fe in synchronize_all_udomains () at dlist.c:694
res = 0
ptr = 0x7f8fed2ec418
#4 0x7f91eed47ca7 in timer (ticks=optimized out, param=optimized out)
at ul_mod.c:439
__FUNCTION__ = timer
#5 0x004df9c8 in timer_ticker (drift=synthetic pointer,
timer_list=optimized out) at timer.c:384
t = 0x7f91f1186da0
j = 300
ij = 30005
ij_marker = 30005
#6 run_timer_process (tpl=0x7f91f11862e8, tpl=0x7f91f11862e8) at timer.c:506
multiple = optimized out
cnt = optimized out
o_tv = optimized out
tv = {tv_sec = 0, tv_usec = 0}
drift = 0
uinterval = 10
wait = optimized out
#7 start_timer_processes () at timer.c:616
tpl = 0x7f91f11862e8
pid = optimized out
__FUNCTION__ = start_timer_processes
#8 0x004170f9 in main_loop () at main.c:1016
i = optimized out
pid = optimized out
si = 0x0
startup_done = 0x0
chd_rank = 64
rc = optimized out
load_p = 0x7f8fed2f67c8
#9 main (argc=optimized out, argv=optimized out) at main.c:1634
cfg_log_stderr = optimized out
cfg_stream = optimized out
c = optimized out
r = optimized out
tmp = 0x7fff9dd62f17
tmp_len = optimized out
port = optimized out
proto = optimized out
options = 0x5c59a8 f:cCm:M:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:o:
ret = -1
seed = 3178038181
__FUNCTION__ = main
Configuration:
### Global Parameters #
tcp_children=256
tcp_max_connections=50
tcp_keepalive=1
tcp_keepcount=3
tcp_keepidle=300
tcp_keepinterval=300
tcp_connection_lifetime=1200
tcp_max_msg_chunks=20
tcp_max_msg_time=15
tcp_connect_timeout=1
tcp_send_timeout=1
#tcp_async=1
disable_tls=yes
tls_verify_server = 1
# There won't be a client presenting a real valid cert. If a client would
# presente a cert - even if not required (see below) this option would make
# the client connecting fail (see opensips docu).
tls_verify_client = 0
tls_require_client_certificate = 0
tls_method = TLSv1
disable_core_dump=no
debug=0
server_header=Server: Jibe Mobile SIP Proxy 1.10.1
log_stderror=no
log_facility=LOG_LOCAL0
fork=yes
children=64
auto_aliases=no
listen=...
### Modules Section
#set module path
mpath=/opt/opensips/lib64/opensips/modules/
# helper and util modules
loadmodule maxfwd.so
loadmodule db_mysql.so
loadmodule textops.so
loadmodule sipmsgops.so
# - Management interface -
loadmodule mi_fifo.so
modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)
modparam(mi_fifo, fifo_mode, 0666)
# core modules for sip routing and checking
# - Stateless signalling
loadmodule sl.so
# - Transaction Management -
loadmodule tm.so
# NOTE: signaling.so requires sl/tm loaded before itself
loadmodule signaling.so
# set final reply timer for SIP messages in seconds
modparam(tm, fr_timer, 32)
modparam(tm, fr_inv_timer, 150)
# - Record Routing -
loadmodule rr.so
modparam(rr, append_fromtag, 1)
modparam(rr, enable_double_rr, 1)
# Dialog module to track RTP proxy in use for the RTP session
loadmodule dialog.so
# - user location -
loadmodule usrloc.so
modparam(usrloc,
[OpenSIPS-Users] REGISTER request to other servers
Hi team,
I am looking for a solution of my problem. i have 1 opensip server and 2
other asterisk servers. i want to get my user registered on asterisk
servers. so user will send the regsiter request to opensip and opensip will
redirect the regsiter request to one of the asteirsk servers. So user will
actually register on asterisk server but all the request would come directly
to opensip.
i tried it many ways but no luck yet.
this is what i am doing?
#if (method==REGISTER) {
xlog(REGISTER from $fu);
# rewrite current URI, which is always part of destination ser
rewritehostport(asterisk1.com:5060);
append_branch(sip:redir...@asterisk2.com);
sl_send_reply(300, Redirect);
return;
}
its not working i see on the phone says Redirect but the REGISTER request is
not going to anywhere.
any help please
thanks
--
View this message in context:
http://opensips-open-sip-server.1449251.n2.nabble.com/REGISTER-request-to-other-servers-tp7595776.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
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[OpenSIPS-Users] Announcing rtpproxy v2.0.0
Hi All, I'm happy to announce that we have released rtpproxy v2.0.0. You can review the release notes here: https://github.com/sippy/rtpproxy/releases/tag/v2.0.0 -sobomax ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Planning OpenSIPS 2.1 release - heads up
Fetch MASTER 2.1.x and after compile i try to run and got this error, [root@opensips ]# /usr/local/opensips-2-head/sbin/opensips -c -f opensips.cfg Mar 13 01:03:43 [22204] CRITICAL:core:yyerror: parse error in config file opensips.cfg, line 60, column 1-12: syntax error Mar 13 01:03:43 [22204] CRITICAL:core:yyerror: parse error in config file opensips.cfg, line 60, column 1-12: Mar 13 01:03:43 [22204] ERROR:core:main: bad config file (2 errors) It is not supporting following option, after removing them it parse file successfully. disable_tcp=yes disable_tls=yes On Thu, Mar 12, 2015 at 3:16 PM, Satish Patel satish@gmail.com wrote: On download page, http://www.opensips.org/Downloads/Downloads, currently we have following. GIT clone of development stable version 2.1.1 (MASTER): Does on march 18th release will be 2.1.2 ? On Thu, Mar 12, 2015 at 11:26 AM, Bogdan-Andrei Iancu bog...@opensips.org wrote: The D day for releasing 2.1 is 18th of March ! Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 24.02.2015 16:22, Bogdan-Andrei Iancu wrote: Hello all, Everybody is looking forward for the next OpenSIPS release, the major 2.1, which is planned for end of February - at least this was the plan so far. I our desire to make of 2.1 a radical improvement, we committed to several major changes/redesigns and new valuable functionalities. And they do burn time to get them done, especially as we want them : in the best possible way. In oder to finish all we committed for, the release date is now estimated for first half of March - some important parts of code still need to be settled down and we do not want to make any kind of compromise in quality. Just to give you a heads up on what OpenSIPS 2.1 will bring: - internal re-design around async reactors for load-balancing tasks inside OpenSIPS - async I/O support in scripts for db queries, exec, rest client - refactoring of the networking and protocol related code (protos are now modules) - webSockets support - Quality Based Routing new module - Compression new module - Fraud Detection new module - Emergency Call handling new module - rtpengine support - partitioning support in Dynamic Routing, Diaplan and Dispatcher and many other - when the coding taks is done, we will proceed withe bothersome task of updating docs, new and migration. Whoever gives a try to 2.1 version, please report any problems or crashes asap to us ! better have them done now rather than later :P Best regards, ___ 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
Re: [OpenSIPS-Users] Planning OpenSIPS 2.1 release - heads up
Does rtp engine support ICE for both video and audio in its current version ? Does rtpengine have a page with info aside from : http://www.opensips.org/html/docs/modules/devel/rtpengine.html Satish, Are you streaming sip trace to another opensips 2.1 instance for homer purposes? The reason I ask is because I was having issues with hep headers when using versions of 1.1X Thanks, Tito On Thu, Mar 12, 2015 at 3:16 PM, Satish Patel satish@gmail.com wrote: On download page, http://www.opensips.org/Downloads/Downloads, currently we have following. GIT clone of development stable version 2.1.1 (MASTER): Does on march 18th release will be 2.1.2 ? On Thu, Mar 12, 2015 at 11:26 AM, Bogdan-Andrei Iancu bog...@opensips.org wrote: The D day for releasing 2.1 is 18th of March ! Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 24.02.2015 16:22, Bogdan-Andrei Iancu wrote: Hello all, Everybody is looking forward for the next OpenSIPS release, the major 2.1, which is planned for end of February - at least this was the plan so far. I our desire to make of 2.1 a radical improvement, we committed to several major changes/redesigns and new valuable functionalities. And they do burn time to get them done, especially as we want them : in the best possible way. In oder to finish all we committed for, the release date is now estimated for first half of March - some important parts of code still need to be settled down and we do not want to make any kind of compromise in quality. Just to give you a heads up on what OpenSIPS 2.1 will bring: - internal re-design around async reactors for load-balancing tasks inside OpenSIPS - async I/O support in scripts for db queries, exec, rest client - refactoring of the networking and protocol related code (protos are now modules) - webSockets support - Quality Based Routing new module - Compression new module - Fraud Detection new module - Emergency Call handling new module - rtpengine support - partitioning support in Dynamic Routing, Diaplan and Dispatcher and many other - when the coding taks is done, we will proceed withe bothersome task of updating docs, new and migration. Whoever gives a try to 2.1 version, please report any problems or crashes asap to us ! better have them done now rather than later :P Best regards, ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Planning OpenSIPS 2.1 release - heads up
Im not sure what you mean by supports ICE. RTPEngine can inject itself as ice relay, or host, candidate for all streams, regardless of the number of streams in the SDP. I am working on a tutorial on how to use rtpengine with webrtc and non-webrtc clients. In the mean time you can check my github wiki for info https://github.com/etamme/federated-sip/wiki Or look at the rtpengine github https://github.com/sipwise/rtpengine for documentation. -Eric ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] orchestration.
Group, I am curious about what the best approach towards orchestrating proxies based on incremental demand is. What is the best way to implement environmental variables such as ip addresses used in listening to interfaces and variables used to raise events. Is there a way for opensips to reach out and provision itself? Thanks, Tito ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Planning OpenSIPS 2.1 release - heads up
On download page, http://www.opensips.org/Downloads/Downloads, currently we have following. GIT clone of development stable version 2.1.1 (MASTER): Does on march 18th release will be 2.1.2 ? On Thu, Mar 12, 2015 at 11:26 AM, Bogdan-Andrei Iancu bog...@opensips.org wrote: The D day for releasing 2.1 is 18th of March ! Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 24.02.2015 16:22, Bogdan-Andrei Iancu wrote: Hello all, Everybody is looking forward for the next OpenSIPS release, the major 2.1, which is planned for end of February - at least this was the plan so far. I our desire to make of 2.1 a radical improvement, we committed to several major changes/redesigns and new valuable functionalities. And they do burn time to get them done, especially as we want them : in the best possible way. In oder to finish all we committed for, the release date is now estimated for first half of March - some important parts of code still need to be settled down and we do not want to make any kind of compromise in quality. Just to give you a heads up on what OpenSIPS 2.1 will bring: - internal re-design around async reactors for load-balancing tasks inside OpenSIPS - async I/O support in scripts for db queries, exec, rest client - refactoring of the networking and protocol related code (protos are now modules) - webSockets support - Quality Based Routing new module - Compression new module - Fraud Detection new module - Emergency Call handling new module - rtpengine support - partitioning support in Dynamic Routing, Diaplan and Dispatcher and many other - when the coding taks is done, we will proceed withe bothersome task of updating docs, new and migration. Whoever gives a try to 2.1 version, please report any problems or crashes asap to us ! better have them done now rather than later :P Best regards, ___ 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
[OpenSIPS-Users] rr not changed using set_advertised_address
Hello Everyone, I sent this email earlier however did not seem to reach the list. Sorry for the redundancy. We are running opensips in a NAT environment. When trying to change the record_route using set_advertised_address in the on_reply route there is no change. I am testing using some test headers and know that the location is getting triggered however, cannot change the RR using set_advertised_address Is this not possible? Kind Regards, Terrance. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
