[OpenSIPS-Users] Tuning for maximum number of TCP connections
We're trying to load up opensips with as many TCP connections as we possibly can. So far we've got it to about 82K, but failures start occurring at that point. We have 8GBs of RAM allocated to the server as a whole (is that enough? we don't appear to be exhausting it). We've set the following parameters for OpenSIPS: tcp_children=32 tcp_max_connections=25 tcp_connection_lifetime=610 tcp_keepalive=1 tcp_keepcount=3 tcp_keepidle=300 tcp_keepinterval=300 We have also set ulimit -n 1024000 and ulimit -s 768. The scenario is that our load driver establishes client connections to OpenSIPS via TCP, and sends REGISTERs over those connections. While the REGISTERs come in over TCP, they are sent out to our registrar via UDP. Around the point where we get to the 40K connection mark we start seeing the following in the logs: Apr 25 12:28:19 blackmamba rcsuser-opensips[27540]: ERROR:core:tcp_blocking_connect: poll error: flags 1c Apr 25 12:28:19 blackmamba rcsuser-opensips[27540]: ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (111) Connection refused Apr 25 12:28:19 blackmamba rcsuser-opensips[27540]: ERROR:core:tcpconn_connect: tcp_blocking_connect failed Apr 25 12:28:19 blackmamba rcsuser-opensips[27540]: ERROR:core:tcp_send: connect failed Apr 25 12:28:19 blackmamba rcsuser-opensips[27540]: ERROR:tm:msg_send: tcp_send failed It almost appears as though opensips is trying to establish a connection somewhere and is being refused. Except that it shouldn't be trying to establish any, unless it's for internal purposes. Unfortunately the logs aren't clear on that point (in terms of what connection is trying to be established). One other thing that appears puzzling: it seems that one of the opensips processes is bearing most of the brunt. I am assuming that it's the instance that is actually accepting the connections, and that the subsequent (low) amount of traffic is then handed off to the children. But if that's the case, it also means that it's handling a lot of the workload, and I was hoping that it would be more evenly distributed. Here is a snapshot of the opensips processes in top: 27577 rcsuser 20 0 6516m 2.5g 2.5g R 76 31.9 8:15.26 opensips 27542 rcsuser 20 0 6516m 181m 180m S 16 2.3 0:54.60 opensips 27541 rcsuser 20 0 6516m 182m 180m S 14 2.3 0:54.47 opensips 27539 rcsuser 20 0 6516m 182m 180m S 13 2.3 0:53.75 opensips 27540 rcsuser 20 0 6516m 182m 180m S 11 2.3 0:53.64 opensips 27545 rcsuser 20 0 6516m 37m 29m S0 0.5 0:01.03 opensips 27551 rcsuser 20 0 6516m 35m 27m S0 0.4 0:00.94 opensips 27553 rcsuser 20 0 6516m 36m 28m S0 0.5 0:00.95 opensips 27555 rcsuser 20 0 6516m 37m 29m S0 0.5 0:00.99 opensips 27557 rcsuser 20 0 6516m 35m 27m S0 0.4 0:00.92 opensips 27558 rcsuser 20 0 6516m 35m 27m S0 0.4 0:00.90 opensips 27560 rcsuser 20 0 6516m 36m 28m S0 0.5 0:00.98 opensips 27563 rcsuser 20 0 6516m 36m 28m S0 0.5 0:00.94 opensips 27564 rcsuser 20 0 6516m 36m 27m S0 0.5 0:00.93 opensips 27565 rcsuser 20 0 6516m 36m 28m S0 0.5 0:00.93 opensips 27567 rcsuser 20 0 6516m 36m 28m S0 0.5 0:00.95 opensips 27575 rcsuser 20 0 6516m 36m 28m S0 0.5 0:00.95 opensips 27576 rcsuser 20 0 6516m 36m 28m S0 0.5 0:00.98 opensips So basically what I'm looking for is some help on getting the operating system and opensips tuned to the point where we can get substantially more than 80K connections. Or am I asking for too much? Thanks, Gavin ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Media-Proxy Segfault on startup [error 4 in libpthread-2.15.so]
How to resolve this problem？ I have a same mistake like this. OS is ubuntu 12.04 Apr 26 11:32:48 ubuntu kernel: [63144.432110] media-relay[23052]: segfault at c ip b77b8cb7 sp bf8bcb90 error 4 in libpthread-2.15.so[b77b+17000] Apr 26 11:33:00 ubuntu kernel: [63156.141836] media-dispatche[23060]: segfault at c ip b7700cb7 sp bfeb00a0 error 4 in libpthread-2.15.so[b76f8000+17000] -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Media-Proxy-Segfault-on-startup-error-4-in-libpthread-2-15-so-tp7580614p7586027.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] Media-Proxy Segfault on startup [error 4 in libpthread-2.15.so]
On Apr 26, 2013, at 5:50 AM, arra wrote: How to resolve this problem？ I have a same mistake like this. OS is ubuntu 12.04 Apr 26 11:32:48 ubuntu kernel: [63144.432110] media-relay[23052]: segfault at c ip b77b8cb7 sp bf8bcb90 error 4 in libpthread-2.15.so[b77b+17000] Apr 26 11:33:00 ubuntu kernel: [63156.141836] media-dispatche[23060]: segfault at c ip b7700cb7 sp bfeb00a0 error 4 in libpthread-2.15.so[b76f8000+17000] We provide a libgcrypt11 package in our ubuntu repository, install it from there and you shouldn't run into this anymore. Regards, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] lookup() not working as documented
Hi list,
I have in my config:
if (!lookup(location,m)) {
switch ($retcode) {
case -1: # Not used
case -3: # User not found, forward
xlog(DESCISION:
$protoCallID:$ciCSeq:$csMethod:$rm$si:$spto$Ri:$Rp To: $tu not
found. Route upstream);
route(upstream);
exit;
case -2: # Not used
sl_send_reply(405, Method Not Allowed);
exit;
}
}
When I'm getting an incomming call, I see in my log:
/usr/sbin/opensips[11393]: DESCISION:
udpCallID:x@x.x.x.xCSeq:11061Method:INVITEx.x.x.x:5060toy.y.y.y:5060
To: sip:0790@y.y.y.y:5060 not found. Route upstream
Ergo, OpenSIPS fails to lookup the subscriber. However, when using
opensipsctl the subscriber is clearly registered on the server:
AOR:: 0...@domain.no
The only difference is that the AOR is registered with the domain and the
INVITE is referencing the server IP. According to the documentation [1],
this should not be a problem because The functions extracts username from
Request-URI and tries to find all contacts for the username in usrloc..
As I'm no SIP expert I thought that I might have misunderstood what
actually was meant by the username part of the R-URI, I assumed it was
just the actual phone number, exclusive of host-name. Section 19.1.1 of RFC
3261 (SIP) confirms this sip:user:password@host
:port;uri-parameters?headers.
[1]. http://www.opensips.org/html/docs/modules/devel/registrar.html#id293055
version: opensips 1.9.0-notls (x86_64/linux)
So I can only conclude that either code is broken or documentation is
broken, or both.
Brgds,
Stian Øvrevåge
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[OpenSIPS-Users] refer scenario - record-route header
Hello, Just tried to play with the b2b refer scenario with opensips. The config is pretty much the default LB config from opensips.org, so nothing sexy in the conf, it works fine without the b2b stuff. LB destinations are reachable through the private ip of the server. if I use the b2b topology hiding scenario, it also work fine. when I use the refer scenario, things goes mad :) Sending this to the caller party (default LB config): U 2013/04/30 02:19:05.920262 opensips_public_ip:5060 - caller_ip:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP caller_ip:5060;received=caller_ip;branch=z9hG4bK25704719;rport=5060. Record-Route: sip:username@opensips_private_ip ;r2=on;lr;ftag=as38e68f66;did=a3c.83df6ae1. Record-Route:sip:username@opensips_public_ip ;r2=on;lr;ftag=as38e68f66;did=a3c.83df6ae1. New call with the refer settings applied: U 2013/04/30 02:13:50.130487 opensips_public_ip:5060 - caller_ip:5060 SIP/2.0 183 Session Progress. Record-Route: sip:6288808754418@;r2=on;lr;ftag=as439eda7d;did=35.b69971a2. Record-Route: sip:6288808754418@opensips_public_ip ;r2=on;lr;ftag=as439eda7d;did=35.b69971a2. In the first record-route header, the host part is missing. The relevant lines from the syslog in this case: Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:lumps_len: lumps_len called with null send_sock Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:lump_check_opt: null send socket Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:lump_check_opt: null send socket Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:lump_check_opt: null send socket Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:lump_check_opt: null send socket Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:process_lumps: null bind_address Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:lump_check_opt: null send socket Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:lump_check_opt: null send socket Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:lump_check_opt: null send socket Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:lump_check_opt: null send socket What can cause this? It is a multihomed enviroment with opensips 1.9. # opensips -V version: opensips 1.9.0-notls (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: 2:9980 @(#) $Id: main.c 9790 2013-02-15 10:14:34Z bogdan_iancu $ main.c compiled on 00:52:55 Apr 30 2013 with gcc 4.4.6 -Laszlo ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Too many RFCs ????
Trying to access : http://tools.ietf.org/html/rfc5626 You get: % args) IOError: [Errno 28] No space left on device Maybe IETF has too many RFCs and DRAFTs.sorting them out by quality would be an option ;) Regards, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Too many RFCs ????
:) nice On Mon, Apr 29, 2013 at 2:55 PM, Bogdan-Andrei Iancu bog...@opensips.orgwrote: Trying to access : http://tools.ietf.org/html/**rfc5626http://tools.ietf.org/html/rfc5626 You get: % args) IOError: [Errno 28] No space left on device Maybe IETF has too many RFCs and DRAFTs.sorting them out by quality would be an option ;) Regards, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.**com http://www.opensips-solutions.com __**_ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-**bin/mailman/listinfo/usershttp://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] memory consumed by t_relay
Hello Chen-Che, A SIP transaction is a requests plus all its replied (provisional and final) - the TM module automatically frees all transactions when they complete (got a final reply or they timeout). Via statistics you can see how many ongoing transactions you have in memory opensipsctl fifo get_statistics tm: - the in_used value. The memory may be exhausted because you have too many transactions in mem (processing rate is slower than the receiving rate). In regards to memory troubleshooting, also take a look at http://www.opensips.org/Documentation/TroubleShooting-OutOfMem Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 04/25/2013 09:26 AM, microx wrote: Hi all, In my environment, I have an outbound proxy and two internal SIP proxy servers. The outbound proxy listens on two interfaces where one is (61.60.x.x) for communication with outside UAs and the other is (192.168.x.x) for communication with internal SIP server. When the outbound proxy receives a SIP message from outside UA, it uses force_send_socket(udp:192.168.x.x:5060) and t_relay() to forward the SIP message to some internal SIP proxy server. On the other hand, when the outbound proxy gets a SIP message from internal SIP server, it uses force_send_socket(udp:61.60.x.x) and t_relay() to forward the message to the destined UA. After processing about 10,000 INVITEs, the outbound proxy runs out of its allocated memory. The reason seems to be the use of t_rely() which generates transactions. If so, when the transaction data will be cleared from the memory at the outbound proxy (the outbound proxy forwards BYEs as well)? Or whether some functions can be used in the configuration file to remove such transaction data? I had tried to use send() instead of t_relay() since send() is stateless. However, force_send_socket does not take effect for send(). Specifically, when the outbound proxy uses force_send_socket(udp:192.168.x.x:5060) and send() to forward a SIP message to some internal SIP proxy server, the SIP message is actually sent with 61.60.x.x rather than 192.168.x.x. Can anyone instruct me how to solve this issue? Deeply thanks for any comment. Best regards, Chen-Che -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/memory-consumed-by-t-relay-tp7586016.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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:registrar:update_contacts: invalid cseq for aor
Hello Duane, The Invalid Cseq is actually a side effect of the REGISTER retransmissions - you have 2 REGISTER (original + retransmission) with CSeq: 756 REGISTER . First is executed in one process, gets stored and a 200 reply ; the retransmission gets executed in a different process, and because it has same cseq the storing generated the error. You need to filter out the retransmissions (use a t_newtran() before the save(location) ) - it should solve the problem. But the question is why does it take for OpenSIPS more than 500ms to generate the answer (and make the sender to do retransmission) ?? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 04/26/2013 03:23 AM, Duane Larson wrote: I originally posted this via Nabble but I am not sure if it went to the Opensips User mailing list so please excuse me if this shows up as multiple posts. I am starting to see this issue a lot lately. My Snom phones will so as not registered on their display screen and when I look in the syslogs I see the following error ERROR:registrar:update_contacts: invalid cseq for aor 9*1**@all.com http://all.com I am not sure if this started happening because of updated Snom code or because of updated OpenSIPS code. Luckily I was able to capture a SIP trace from one of the Snom phones today. Here is a SIP trace of REGISTERs without the issue http://pastebin.com/RyaZQUBa Here is a SIP trace of REGISTER showing the issue http://pastebin.com/YC1AyTJ6--- The last message in this paste is a 400 Bad Request So is the Snom phone doing something wrong or might it be on the OpenSIPS side? ___ 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] Tuning for maximum number of TCP connections
Hello Gavin, The errors you get indicates that OpenSIPS is trying to open a TCP connection to a destination which does not accept it. Based on your description, I would say there is not need for OpenSIPS to open TCP connections - they will be open by the clients when registering. Ruling out the scenario of a misrouting , the only explanation will be that the TCP connections expires (timeout without traffic) long before the corresponding registration - so you end up with a registration (in usrloc) which has no TCP conn towards the actual device. Are you using the tcp_persistent_flag ? http://www.opensips.org/html/docs/modules/1.9.x/registrar.html#id250105 About the load on the processes, you can do opensipsctl fifo ps to get the listing of the processes and their description - you could correlate with the TOP info to see what's the process burning CPU Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 04/26/2013 05:44 PM, Gavin Murphy wrote: We're trying to load up opensips with as many TCP connections as we possibly can. So far we've got it to about 82K, but failures start occurring at that point. We have 8GBs of RAM allocated to the server as a whole (is that enough? we don't appear to be exhausting it). We've set the following parameters for OpenSIPS: tcp_children=32 tcp_max_connections=25 tcp_connection_lifetime=610 tcp_keepalive=1 tcp_keepcount=3 tcp_keepidle=300 tcp_keepinterval=300 We have also set ulimit -n 1024000 and ulimit -s 768. The scenario is that our load driver establishes client connections to OpenSIPS via TCP, and sends REGISTERs over those connections. While the REGISTERs come in over TCP, they are sent out to our registrar via UDP. Around the point where we get to the 40K connection mark we start seeing the following in the logs: Apr 25 12:28:19 blackmamba rcsuser-opensips[27540]: ERROR:core:tcp_blocking_connect: poll error: flags 1c Apr 25 12:28:19 blackmamba rcsuser-opensips[27540]: ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (111) Connection refused Apr 25 12:28:19 blackmamba rcsuser-opensips[27540]: ERROR:core:tcpconn_connect: tcp_blocking_connect failed Apr 25 12:28:19 blackmamba rcsuser-opensips[27540]: ERROR:core:tcp_send: connect failed Apr 25 12:28:19 blackmamba rcsuser-opensips[27540]: ERROR:tm:msg_send: tcp_send failed It almost appears as though opensips is trying to establish a connection somewhere and is being refused. Except that it shouldn't be trying to establish any, unless it's for internal purposes. Unfortunately the logs aren't clear on that point (in terms of what connection is trying to be established). One other thing that appears puzzling: it seems that one of the opensips processes is bearing most of the brunt. I am assuming that it's the instance that is actually accepting the connections, and that the subsequent (low) amount of traffic is then handed off to the children. But if that's the case, it also means that it's handling a lot of the workload, and I was hoping that it would be more evenly distributed. Here is a snapshot of the opensips processes in top: 27577 rcsuser 20 0 6516m 2.5g 2.5g R 76 31.9 8:15.26 opensips 27542 rcsuser 20 0 6516m 181m 180m S 16 2.3 0:54.60 opensips 27541 rcsuser 20 0 6516m 182m 180m S 14 2.3 0:54.47 opensips 27539 rcsuser 20 0 6516m 182m 180m S 13 2.3 0:53.75 opensips 27540 rcsuser 20 0 6516m 182m 180m S 11 2.3 0:53.64 opensips 27545 rcsuser 20 0 6516m 37m 29m S0 0.5 0:01.03 opensips 27551 rcsuser 20 0 6516m 35m 27m S0 0.4 0:00.94 opensips 27553 rcsuser 20 0 6516m 36m 28m S0 0.5 0:00.95 opensips 27555 rcsuser 20 0 6516m 37m 29m S0 0.5 0:00.99 opensips 27557 rcsuser 20 0 6516m 35m 27m S0 0.4 0:00.92 opensips 27558 rcsuser 20 0 6516m 35m 27m S0 0.4 0:00.90 opensips 27560 rcsuser 20 0 6516m 36m 28m S0 0.5 0:00.98 opensips 27563 rcsuser 20 0 6516m 36m 28m S0 0.5 0:00.94 opensips 27564 rcsuser 20 0 6516m 36m 27m S0 0.5 0:00.93 opensips 27565 rcsuser 20 0 6516m 36m 28m S0 0.5 0:00.93 opensips 27567 rcsuser 20 0 6516m 36m 28m S0 0.5 0:00.95 opensips 27575 rcsuser 20 0 6516m 36m 28m S0 0.5 0:00.95 opensips 27576 rcsuser 20 0 6516m 36m 28m S0 0.5 0:00.98 opensips So basically what I'm looking for is some help on getting the operating system and opensips tuned to the point where we can get substantially more than 80K connections. Or am I asking for too much? Thanks, Gavin ___ 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] Caller Name and P-Assterted
Hello Nick, As the P-Asserted-Id hdr is an name-addr like header (like FROM, TO, Contact), you should do: append_hf(P-Asserted-Identity: \Test User\ sip:15453387...@test.server.com mailto:sip%3a15453387...@test.server.com;user=phone\r\n, Call-ID); where Test User is the display name - but depends on the end devices if they do display it or not. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 04/29/2013 02:19 AM, Nick Khamis wrote: Hello Everyone, Is it possible to pass a meaningful caller name along with the caller id given the carrier supports CLI? Before asking the carrier, I was doing some tests. I know that append_hf(P-Asserted-Identity: sip:15453387...@test.server.com mailto:sip%3a15453387...@test.server.com; user=phone\r\n, Call-ID) works fine for the phone number alone (i.e., 15453387463) however, append_hf(P-Asserted-Identity: From: \ Test User \ sip:15453387...@test.server.com mailto:sip%3a15453387...@test.server.com; user=phone\r\n, Call-ID) generates a meaningless CID. The SIP Trace: P-Asserted-Identity: From: Mike Peer sip:15453387...@test.server.com mailto:sip%3a15453387...@test.server.com; user=phone. Once I know what is accepted for PAI, I plan on assigning subscriber.rpid that value. To load rpid for a specific caller and INVITE, I have added the following code: modparam(auth_db, load_credentials, rpid) modparam(auth, rpid_avp, $avp(rpid)) Testing from branch route, $avp(rpid) is NULL. We do not allow users to REGISTER. Any way we can get subscriber.rpid for INVITES? Finally, is branch_route and failure_routes, the safest place to append the PAI? Thanks in Advance, Nick ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] lookup() not working as documented
Hello Stian,
AOR 0790@y.y.y.y and 0...@domain.no
mailto:0...@domain.no are completly different - according to SIP,
they do not match or so. So you have to use the same AOR when
registering the user (AOR will be in To hdr) and when calling the user
(AOR will be in RURI).
Maybe the doc can be misleading a bit when using the terminology of
username - but this has to be interpreted in the context of
use_domain parameter from usrloc:
http://www.opensips.org/html/docs/modules/1.9.x/usrloc.html#id292895
So, docs are not clear, we will take care of that!
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 04/29/2013 01:28 PM, Stian Øvrevåge wrote:
Hi list,
I have in my config:
if (!lookup(location,m)) {
switch ($retcode) {
case -1: # Not used
case -3: # User not found, forward
xlog(DESCISION:
$protoCallID:$ciCSeq:$csMethod:$rm$si:$spto$Ri:$Rp To: $tu
not found. Route upstream);
route(upstream);
exit;
case -2: # Not used
sl_send_reply(405, Method Not Allowed);
exit;
}
}
When I'm getting an incomming call, I see in my log:
/usr/sbin/opensips[11393]: DESCISION:
udpCallID:x@x.x.x.xCSeq:11061Method:INVITEx.x.x.x:5060toy.y.y.y:5060
To: sip:0790@y.y.y.y:5060 not found. Route upstream
Ergo, OpenSIPS fails to lookup the subscriber. However, when using
opensipsctl the subscriber is clearly registered on the server:
AOR:: 0...@domain.no mailto:0...@domain.no
The only difference is that the AOR is registered with the domain and
the INVITE is referencing the server IP. According to the
documentation [1], this should not be a problem because The functions
extracts username from Request-URI and tries to find all contacts for
the username in usrloc..
As I'm no SIP expert I thought that I might have misunderstood what
actually was meant by the username part of the R-URI, I assumed it
was just the actual phone number, exclusive of host-name. Section
19.1.1 of RFC 3261 (SIP) confirms this
sip:user:password@host:port;uri-parameters?headers.
[1].
http://www.opensips.org/html/docs/modules/devel/registrar.html#id293055
version: opensips 1.9.0-notls (x86_64/linux)
So I can only conclude that either code is broken or documentation is
broken, or both.
Brgds,
Stian Øvrevåge
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Re: [OpenSIPS-Users] refer scenario - record-route header
Hi Lazlo, Could you post somewhere the SIP capture of the call ? Just to be sure I correctly understand your scenario. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 04/29/2013 02:38 PM, Laszlo wrote: Hello, Just tried to play with the b2b refer scenario with opensips. The config is pretty much the default LB config from opensips.org http://opensips.org, so nothing sexy in the conf, it works fine without the b2b stuff. LB destinations are reachable through the private ip of the server. if I use the b2b topology hiding scenario, it also work fine. when I use the refer scenario, things goes mad :) Sending this to the caller party (default LB config): U 2013/04/30 02:19:05.920262 opensips_public_ip:5060 - caller_ip:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP caller_ip:5060;received=caller_ip;branch=z9hG4bK25704719;rport=5060. Record-Route: sip:username@opensips_private_ip;r2=on;lr;ftag=as38e68f66;did=a3c.83df6ae1. Record-Route:sip:username@opensips_public_ip;r2=on;lr;ftag=as38e68f66;did=a3c.83df6ae1. New call with the refer settings applied: U 2013/04/30 02:13:50.130487 opensips_public_ip:5060 - caller_ip:5060 SIP/2.0 183 Session Progress. Record-Route: sip:6288808754418@;r2=on;lr;ftag=as439eda7d;did=35.b69971a2. Record-Route: sip:6288808754418@opensips_public_ip;r2=on;lr;ftag=as439eda7d;did=35.b69971a2. In the first record-route header, the host part is missing. The relevant lines from the syslog in this case: Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:lumps_len: lumps_len called with null send_sock Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:lump_check_opt: null send socket Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:lump_check_opt: null send socket Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:lump_check_opt: null send socket Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:lump_check_opt: null send socket Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:process_lumps: null bind_address Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:lump_check_opt: null send socket Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:lump_check_opt: null send socket Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:lump_check_opt: null send socket Apr 30 02:13:47 svr2 /usr/local/sbin/opensips[30081]: CRITICAL:core:lump_check_opt: null send socket What can cause this? It is a multihomed enviroment with opensips 1.9. # opensips -V version: opensips 1.9.0-notls (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: 2:9980 @(#) $Id: main.c 9790 2013-02-15 10:14:34Z bogdan_iancu $ main.c compiled on 00:52:55 Apr 30 2013 with gcc 4.4.6 -Laszlo ___ 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] Caller Name and P-Assterted
Hello Bogdan, As usual. Thank you so much! In the past year I learned so much about SIP then I did in the past five using the different media servers. The gateway I sent the PAI hdr seems to accept it, but I the CallID Name part does not seem to be supported by the carrier. Ce la vie... On the question about rpid, I have: Test User sip:15453387...@test.server.com;user=phone\r\n stored in subscriber.rpid. In the branch and failure routes, I would like to load this value into append_hf. What is the best way of doing this since we do not perform any authentication for registration or invites on our system, and thus no consume_credentials Kind Regards, Nick ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPs Radius Accounting.
Hi, I wanted to confirm if radius accounting requests are generated on a successful transaction or it can be generated on a received BYE only. To elaborate my question you can look at 2 diagrams below. Is first scenario correct based on RFC's? My next question is that if scenario A is correct then how can we account the call if say user B has gone offline and we do not receive 200 OK of the BYE sent? Can we send a manual accounting request to Radius with acc_aaa_request in accounting module? *Scenario A:* User AOpenSIPsRadius User B |---BYE---| | | |-BYE| | |---acct-BYE---| *Scenario B:* User AOpenSIPsRadius User B |---BYE---| | | | |-BYE| | |---200 OK -| |200 OK -| | |---acct-BYE---| Regards, Qasim Ayyaz Khan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Caller Name and P-Assterted
Hey team, Do you know if 1.6.2 release of opensip has issues of hanging often, will upgrading to 1.8 issue can fix it ? Warm regards, Sukanya R -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Monday, April 29, 2013 7:27 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Caller Name and P-Assterted Use avp_db_load() or avp_db_query() from the avpops module. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 04/29/2013 04:21 PM, Nick Khamis wrote: Hello Bogdan, As usual. Thank you so much! In the past year I learned so much about SIP then I did in the past five using the different media servers. The gateway I sent the PAI hdr seems to accept it, but I the CallID Name part does not seem to be supported by the carrier. Ce la vie... On the question about rpid, I have: Test Usersip:15453387...@test.server.com;user=phone\r\n stored in subscriber.rpid. In the branch and failure routes, I would like to load this value into append_hf. What is the best way of doing this since we do not perform any authentication for registration or invites on our system, and thus no consume_credentials Kind Regards, Nick ___ 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] Caller Name and P-Assterted
Get me a position at Ericsson Canada, and I will tell you ;). N. On 4/29/13, Sukanya R sukany...@ericsson.com wrote: Hey team, Do you know if 1.6.2 release of opensip has issues of hanging often, will upgrading to 1.8 issue can fix it ? Warm regards, Sukanya R -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Monday, April 29, 2013 7:27 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Caller Name and P-Assterted Use avp_db_load() or avp_db_query() from the avpops module. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 04/29/2013 04:21 PM, Nick Khamis wrote: Hello Bogdan, As usual. Thank you so much! In the past year I learned so much about SIP then I did in the past five using the different media servers. The gateway I sent the PAI hdr seems to accept it, but I the CallID Name part does not seem to be supported by the carrier. Ce la vie... On the question about rpid, I have: Test Usersip:15453387...@test.server.com;user=phone\r\n stored in subscriber.rpid. In the branch and failure routes, I would like to load this value into append_hf. What is the best way of doing this since we do not perform any authentication for registration or invites on our system, and thus no consume_credentials Kind Regards, Nick ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Caller Name and P-Assterted
Thanks Again Bogdan,
To take advantage of fast db querying, and not have to worry about
making sure I am immune to DoS attacks, I opted to go with db_load:
AVPOPS: http://www.opensips.org/html/docs/modules/1.7.x/avpops.html#id250324
modparam(avpops, avp_table, subscriber)
modparam(avpops,use_domain,0)
modparam(avpops,db_scheme,
scheme0:table=subscriber;uuid_col=username;value_col=rpid)
modparam(avpops, db_url, mysql://user:sec...@db.server.com/sipproxysdb)
branch_route[1] {
xlog(L_INFO,New Branch For: $ru at IP: $si\n);
if(is_present_hf(P-Asserted-Identity))
remove_hf(P-Asserted-Identity);
if(is_present_hf(Remote-Party-ID)) remove_hf(Remote-Party-ID);
if(is_present_hf(Privacy)) remove_hf(Privacy);
avp_db_load($fU,$avp(rpid)/$scheme0);
if(is_avp_set($avp(rpid))) append_hf(P-Asserted-Identity:
$rpid; user=phone\r\n, Call-ID);
}
In the subscriber table, I have the following for rpid:
Test User sip:1555...@test.server.com
Enriching Search Engines,
Nick.
On 4/29/13, Nick Khamis sym...@gmail.com wrote:
Get me a position at Ericsson Canada, and I will tell you ;).
N.
On 4/29/13, Sukanya R sukany...@ericsson.com wrote:
Hey team,
Do you know if 1.6.2 release of opensip has issues of hanging often, will
upgrading to 1.8 issue can fix it ?
Warm regards,
Sukanya R
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei
Iancu
Sent: Monday, April 29, 2013 7:27 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Caller Name and P-Assterted
Use avp_db_load() or avp_db_query() from the avpops module.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 04/29/2013 04:21 PM, Nick Khamis wrote:
Hello Bogdan,
As usual. Thank you so much! In the past year I learned so much about
SIP then I did in the past five using the different media servers. The
gateway I sent the PAI hdr seems to accept it, but I the CallID Name
part does not seem to be supported by the carrier. Ce la vie...
On the question about rpid, I have:
Test Usersip:15453387...@test.server.com;user=phone\r\n stored in
subscriber.rpid. In the branch and failure routes, I would like to
load this value into append_hf. What is the best way of doing this
since we do not perform any authentication for registration or invites
on our system, and thus no consume_credentials
Kind Regards,
Nick
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[OpenSIPS-Users] migrate db with MariaDB
Hello, As my distro switched to MariaDB it seems there is a problem when trying to migrate the database from 1.8 to 1.9. Everything goes like # opensipsdbctl migrate opensips opensips-new Enter character set name: latin1 INFO: creating database opensips-new ... ERROR 1064 (42000) at line 1: You have an error in your SQL syntax; check the manual that corresponds to your MariaDB server version for the right syntax to use near '-new character set latin1' at line 1 Any hints where I should look further in order to solve this? Many thanks. -- Arthur Titeica signature.asc Description: This is a digitally signed message part. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OpenSIPS FreeRadius - Not all fields get inserted in the DB
Hello Everyone, We have setup opensips to work with freeradius using the CDTool documentation however, some of the fields that are filled in the the radius log file, for example: Service-Type = SIP Sip-Response-Code = 200 Sip-Method = Bye Event-Timestamp = Apr 29 2013 01:54:11 EDT Sip-From-Tag = as3469a78c Sip-To-Tag = as50c4af01 is not showing up in the DB: +-+-+---++--+ | servicetype | sipresponsecode | sipmethod | sipfromtag | siptotag | +-+-+---++--+ | SIP | 0 | || | It's not everything, things like servicetype, Acct-Session-Id, and User-Name, are consistent with the raidus log files. The dictionaries, and sql.conf are taken from CDRTool project. On a slightly unrelated, I had to add the following field to radacct schema included with CDTTool, to stop some sql errors: -- Added: -- `FramedProtocol` varchar(15) NOT NULL default '', -- `XAScendSessionSVRKey` varchar(15) NOT NULL default '', PS Added the modification to the sql procedures as well... Thanks in Advance, Nick. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Caller Name and P-Assterted
Hello Sukanya, Hard to tell - 1.6.2 is a real ancient release, do not know what kind of hanging you experience ( could be I/Os, not code issues). Anyhow I do advice to upgrade to 1.8 and 1.9. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 04/29/2013 05:55 PM, Sukanya R wrote: Hey team, Do you know if 1.6.2 release of opensip has issues of hanging often, will upgrading to 1.8 issue can fix it ? Warm regards, Sukanya R -Original Message- From: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Monday, April 29, 2013 7:27 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Caller Name and P-Assterted Use avp_db_load() or avp_db_query() from the avpops module. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 04/29/2013 04:21 PM, Nick Khamis wrote: Hello Bogdan, As usual. Thank you so much! In the past year I learned so much about SIP then I did in the past five using the different media servers. The gateway I sent the PAI hdr seems to accept it, but I the CallID Name part does not seem to be supported by the carrier. Ce la vie... On the question about rpid, I have: Test Usersip:15453387...@test.server.com;user=phone\r\n stored in subscriber.rpid. In the branch and failure routes, I would like to load this value into append_hf. What is the best way of doing this since we do not perform any authentication for registration or invites on our system, and thus no consume_credentials Kind Regards, Nick ___ 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:registrar:update_contacts: invalid cseq for aor
I will try the t_newtran(). I can only guess that the 500ms is coming from my MySQL database lookup. On Mon, Apr 29, 2013 at 7:40 AM, Bogdan-Andrei Iancu bog...@opensips.orgwrote: ** Hello Duane, The Invalid Cseq is actually a side effect of the REGISTER retransmissions - you have 2 REGISTER (original + retransmission) with CSeq: 756 REGISTER . First is executed in one process, gets stored and a 200 reply ; the retransmission gets executed in a different process, and because it has same cseq the storing generated the error. You need to filter out the retransmissions (use a t_newtran() before the save(location) ) - it should solve the problem. But the question is why does it take for OpenSIPS more than 500ms to generate the answer (and make the sender to do retransmission) ?? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developerhttp://www.opensips-solutions.com On 04/26/2013 03:23 AM, Duane Larson wrote: I originally posted this via Nabble but I am not sure if it went to the Opensips User mailing list so please excuse me if this shows up as multiple posts. I am starting to see this issue a lot lately. My Snom phones will so as not registered on their display screen and when I look in the syslogs I see the following error ERROR:registrar:update_contacts: invalid cseq for aor 9*1**@all.com I am not sure if this started happening because of updated Snom code or because of updated OpenSIPS code. Luckily I was able to capture a SIP trace from one of the Snom phones today. Here is a SIP trace of REGISTERs without the issue http://pastebin.com/RyaZQUBa Here is a SIP trace of REGISTER showing the issue http://pastebin.com/YC1AyTJ6 --- The last message in this paste is a 400 Bad Request So is the Snom phone doing something wrong or might it be on the OpenSIPS side? ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] ERROR:registrar:update_contacts: invalid cseq for aor
Wow those thresholds give you a good amount of info. I'll have to see how
I can make my MySQL service quicker.
Apr 29 15:31:37 SIPProxy02 /usr/local/sbin/opensips[3104]:
WARNING:db_mysql:log_expiry: threshold exceeded : mysql query took too long
- 512355 us.Source : insert into location (username,contact,expires,q,ca
llid,cseq,flags,cflags,user_agent,received,path,socket,methods,last_modified,sip_instance,domain
) values
('9*12*1$%$%','sip:9*12*1$%$%@172.*.*.33:57369;line=mtj8kl6c','2013-04-29
16:31:37',1.00 ,'26
00cbed-7lfki4ifz7hz',1260,0,0,'snom720/8.7.3.19
','sip:172.*.*.33:57369',NULL,'udp:50.57.54.156:5060',7999,'2013-04-29
15:31:37',NULL,'all.com') on duplicate key update username='9*12*1$%$%',co
ntact='sip:9*12*1$%$%@172.*.*.33:57369;line=mtj8kl6c',expires='2013-04-29
16:31:37',q=1.00
,callid='2600cbed-7lfki4ifz7hz',cseq=1260,flags=0,cflags=0,user_agent='snom720/
8.7.3.19',received='sip:1
72.12.199.33:57369',path=NULL,socket='udp:50.57.54.156:5060',methods=7999,last_modified='2013-04-29
15:31:37',sip_instance=NULL,domain='all.com'
Apr 29 15:31:37 SIPProxy02 /usr/local/sbin/opensips[3104]:
WARNING:core:log_expiry: threshold exceeded : msg processing took too long
- 519804 us.Source : REGISTER sip:all.com SIP/2.0#015#012Via: SIP
/2.0/UDP 172.*.*.33:57369;branch=z9hG4bK-clxrqnw5iv48;rport#015#012From:
901-201-5656 sip:9*12*1$%$%@all.com;tag=bs72h6ifw5#015#012To:
901-201-5656 sip:9*12*1$%$%@all.com#015#012Ca
ll-ID: 2600cbed-7lfki4ifz7hz#015#012CSeq: 1260
REGISTER#015#012Max-Forwards: 69#015#012Contact:
sip:9*12*1$%$%@172.*.*.33:57369;line=mtj8kl6c;reg-id=1;q=1.0;audio;mobility=fixed;duplex=full;desc
ription=snom720;actor=principal;events=dialog;methods=INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO#015#012User-Agent:
snom720/8.7.3.19#015#012Allow-Events: dialog#015#012X-
Real-IP: 192.168.1.71#015#012Supported: path#015#012Authorization: Digest
username=9*12*1$%$%,realm=all.com
,nonce=517ed8c7f0973e1dd14a403ead3ffb092efd04c3,uri=sip:all.com
,qop=auth,nc=
0001,cnonce=649029e8,response=c1cdca29ccd998e22b8caed1454719d2,algorithm=MD5#015#012Expires:
3600#015#012Content-Length: 0#015#012#015#012
Apr 29 15:31:37 SIPProxy02 /usr/local/sbin/opensips[3104]:
WARNING:core:log_expiry: #1 is a core action : 6 - 519586us - line 553
Apr 29 15:31:37 SIPProxy02 /usr/local/sbin/opensips[3104]:
WARNING:core:log_expiry: #2 is a core action : 14 - 519573us - line 595
Apr 29 15:31:37 SIPProxy02 /usr/local/sbin/opensips[3104]:
WARNING:core:log_expiry: #3 is a core action : 6 - 519567us - line 594
Apr 29 15:31:37 SIPProxy02 /usr/local/sbin/opensips[3104]:
WARNING:core:log_expiry: #4 is a core action : 14 - 519564us - line 967
Apr 29 15:31:37 SIPProxy02 /usr/local/sbin/opensips[3104]:
WARNING:core:log_expiry: #5 is a module action : save - 516033us - line 956
Apr 29 15:31:38 SIPProxy02 /usr/local/sbin/opensips[3106]:
WARNING:db_mysql:log_expiry: threshold exceeded : mysql query took too long
- 508532 us.Source : insert into location (username,contact,expires,q,ca
llid,cseq,flags,cflags,user_agent,received,path,socket,methods,last_modified,sip_instance,domain
) values
('9*12*1$%$%','sip:9*12*1$%$%@172.*.*.33:57369;line=mtj8kl6c','2013-04-29
16:31:37',1.00 ,'26
00cbed-7lfki4ifz7hz',1260,0,0,'snom720/8.7.3.19
','sip:172.*.*.33:57369',NULL,'udp:50.57.54.156:5060',7999,'2013-04-29
15:31:37',NULL,'all.com') on duplicate key update username='9*12*1$%$%',co
ntact='sip:9*12*1$%$%@172.*.*.33:57369;line=mtj8kl6c',expires='2013-04-29
16:31:37',q=1.00
,callid='2600cbed-7lfki4ifz7hz',cseq=1260,flags=0,cflags=0,user_agent='snom720/
8.7.3.19',received='sip:1
72.12.199.33:57369',path=NULL,socket='udp:50.57.54.156:5060',methods=7999,last_modified='2013-04-29
15:31:37',sip_instance=NULL,domain='all.com'
Apr 29 15:31:38 SIPProxy02 /usr/local/sbin/opensips[3106]:
WARNING:core:log_expiry: threshold exceeded : msg processing took too long
- 513747 us.Source : REGISTER sip:all.com SIP/2.0#015#012Via: SIP
/2.0/UDP 172.*.*.33:57369;branch=z9hG4bK-clxrqnw5iv48;rport#015#012From:
901-201-5656 sip:9*12*1$%$%@all.com;tag=bs72h6ifw5#015#012To:
901-201-5656 sip:9*12*1$%$%@all.com#015#012Ca
ll-ID: 2600cbed-7lfki4ifz7hz#015#012CSeq: 1260
REGISTER#015#012Max-Forwards: 69#015#012Contact:
sip:9*12*1$%$%@172.*.*.33:57369;line=mtj8kl6c;reg-id=1;q=1.0;audio;mobility=fixed;duplex=full;desc
ription=snom720;actor=principal;events=dialog;methods=INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO#015#012User-Agent:
snom720/8.7.3.19#015#012Allow-Events: dialog#015#012X-
Real-IP: 192.168.1.71#015#012Supported: path#015#012Authorization: Digest
username=9*12*1$%$%,realm=all.com
,nonce=517ed8c7f0973e1dd14a403ead3ffb092efd04c3,uri=sip:all.com
,qop=auth,nc=
0001,cnonce=649029e8,response=c1cdca29ccd998e22b8caed1454719d2,algorithm=MD5#015#012Expires:
3600#015#012Content-Length: 0#015#012#015#012
Apr 29 15:31:38 SIPProxy02 /usr/local/sbin/opensips[3106]:
WARNING:core:log_expiry: #1 is a
Re: [OpenSIPS-Users] OpenSIPs Radius Accounting.
I have tried this scenario. Still if the User B is behind a NAT or is unreachable the opensips generates the BYE with retransmitted BYE's and the dialog is closed but there is no response to BYE received from that user hence no radius acct request. Regards, Qasim On Mon, Apr 29, 2013 at 8:36 PM, Muhammad Shahzad shaherya...@gmail.comwrote: Per my understanding, accounting event is sent when BYE completes, whether if destination replies with 200 OK or BYE re-transmission times out and opensips responds with 408 Request timeout. In each case SIP response code is set appropriately and you should use stop time as accounting end time rather then the time your receive account stop request on radius (they both may differ, e.g. under high load scenarios). Thank you. On Mon, Apr 29, 2013 at 3:27 PM, qasimak...@gmail.com qasimak...@gmail.com wrote: Hi, I wanted to confirm if radius accounting requests are generated on a successful transaction or it can be generated on a received BYE only. To elaborate my question you can look at 2 diagrams below. Is first scenario correct based on RFC's? My next question is that if scenario A is correct then how can we account the call if say user B has gone offline and we do not receive 200 OK of the BYE sent? Can we send a manual accounting request to Radius with acc_aaa_request in accounting module? *Scenario A:* User AOpenSIPsRadius User B |---BYE---| | | |-BYE| | |---acct-BYE---| *Scenario B:* User AOpenSIPsRadius User B |---BYE---| | | | |-BYE| | |---200 OK -| |200 OK -| | |---acct-BYE---| Regards, Qasim Ayyaz Khan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Mit freundlichen Grüßen Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ 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
