Re: [OpenSIPS-Users] Music On Hold Opensips
Hi, osiris123d wrote: So say RTPProxy can successfully do MOH, if your current VoIP infrastructure uses Mediaproxy then you would need to set up a RTPProxy server and have the customers that wish to have MOH use RTPProxy instead of Mediaproxy correct? sort of I guess the only caveat is that RTPProxy doesn't work with CDRTool as far as the BYE message is concerned. you can try to make it - RTPproxy is also able to report the timeouts at media level and to trigger and external script (where you can push a radius/acc/etc event) Regards, Bogdan -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Private IP in registered AOR causing failure
Hi Brian, opensipsl...@encambio.com wrote: 1) the BLA part is sending the SUBSCRIBEs based on user location and it is using the received field if present, so it should be ok. Unless you set in the bla module the outbound param - this will override the received info. I know that at least some part of the BLA logic or presence logic is indeed using the 'received' field, because OpenSIPS is sending to the correct UAS 1/2 the time. It is equially clear that some other part of BLA or presence is ignoring the 'received' field. For each UAS which registers tcpdump shows two SUBSCRIBE messages. One with the 'received' IP and one with the private IP. My module parameters for presence and BLA are: modparam(presence, server_address, sip:p...@name.host.tld) modparam(presence_xml, force_active, 1) modparam(pua_bla, server_address, sip:p...@name.host.tld) modparam(pua_bla, default_domain, name.host.tld) modparam(pua_bla, header_name, Sender) ...so I'm not setting the 'outbound' parameter it seems, right? I'm using: Solaris 11 x86 (nv-b91) OpenSIPS 1.6.0 with TLS In such a case, check in usrloc if the registrations for the user the SUBSCRIBE belongs to, do has the received field - maybe the error is when saving the contacts and not when using them. 2) the subscribe you posted - as you captured it, I supposed it exists on network. But I see that the transport in TLS (so TCP based), but how come you see the message if opensips is not able to open the TCP conn to the private IP.. Good question, but I showed you just how I was capturing traffic to the private IP. You probably missed it, so here it is again: After running a socket listener on 192.168.0.31 on the OpenSIPS host: $ socat TCP4-LISTEN:2310,bind=192.168.0.31,reuseaddr - SUBSCRIBE sip:mylogin-os...@192.168.0.31:2310;transport=tls;line=2acy67zm SIP/2.0 Via: SIP/2.0/TCP 86.90.39.44;branch=G4z9hb82dK8.f144.0 To: sip:mylogin-os...@name.host.tld;tag=ty6sjh9iz9 From: sip:mylogin-os...@name.host.tld;tag=6c9d4319c74d756e6b696-baa1 CSeq: 11 SUBSCRIBE Call-ID: b1c04118-8...@86.90.39.44 Content-Length: 0 User-Agent: OpenSIPS (1.6.1-tls) Max-Forwards: 70 Event: dialog;sla Contact: sip:prese...@name.host.tld Expires: 610 It's plain old TCP/IP networking. First do a ifconfig to create a virtual interface with IP 192.168.0.31, then set a static route, and finally run software to listen on the TCP socket on the address. This works as long as I do it on the same machine that OpenSIPS (the presence modules) are running on. Any ideas? If you could try to get a capture of the whole flow - starting with REGISTER, etc.plus the logs... But first check the usrloc (see 1) ) Regards, Bogdan Thanks, Brian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] What is protocol/port mismatch?
Hi Brian, Do you use force_send_socket() functions in your script, before the t_relay ? Regards, Bogdan opensipsl...@encambio.com wrote: Hello list, I'm using: Solaris 11 x86 (nv-b91) OpenSIPS 1.6.0 with TLS ...and I see this in the log: warning opensips[2717]: WARNING:core:get_send_socket: protocol/port mismatch some hundreds of times per day (about once every 20 minutes per registered UAC.) I have this in the route script: listen = tls:name.host.tld:5061 [...] t_relay(name.host.tld:5080); There is a voicemail server (not OpenSIPS) listening on TCP (not TLS) port 5080 on the same host. OpenSIPS and the other server exchange traffic such as SUBSCRIBE and NOTIFY for MWI and INVITEs to voicemail. The voicemail server sends SIP messages to OpenSIPS port 5061 over TLS (I think.) Question: What exactly is the meaning of these warnings in the log, and does it seem that I'm doing something wrong in the route script? Thanks, Brian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Query regarding Rtp Proxy opensips
Hi, I doubt you can change that as RTPproxy is not decoding the RTP stream - as the name says, the tool is only RTP aware, so cannot interpret the content. But I guess you can google for some other audio tools to help mixing the 2 streams. Regards, Bogdan Indiver wrote: Hi Bodgan, Yes. These files are raw rtp files. When ever call is ended rtp proxy storing the 2 raw rtp files in to specified destination folder. One for callee and other for caller. The problem i faced is i have to merge these 2 raw rtp files of each call and convert into wav file to hear the conversation. Is there any other solution that to record call directly as a wav file using rtpproxy? Thanks in advance! -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Private IP in registered AOR causing failure
Hello Bogdan, An ven., janv 29, 2010, Bogdan-Andrei Iancu schrieb: opensipsl...@encambio.com wrote: 1) the BLA part is sending the SUBSCRIBEs based on user location and it is using the received field if present, so it should be ok. Unless you set in the bla module the outbound param - this will override the received info. My module parameters for presence and BLA are: modparam(presence, server_address, sip:p...@name.host.tld) modparam(presence_xml, force_active, 1) modparam(pua_bla, server_address, sip:p...@name.host.tld) modparam(pua_bla, default_domain, name.host.tld) modparam(pua_bla, header_name, Sender) ...so I'm not setting the 'outbound' parameter it seems, right? In such a case, check in usrloc if the registrations for the user the SUBSCRIBE belongs to, do has the received field - maybe the error is when saving the contacts and not when using them. That would be nice, but I don't think so: # /pfx/sbin/opensipsctl ul show AOR:: user1 Contact:: sip:us...@192.168.0.31:3618;transport=tls;line=9hm7f0ua Q=1 Expires:: 352 Callid:: 703c26357076-zmjcebpdyaey Cseq:: 1058 User-agent:: Unimportant Received:: sip:82.90.12.232:2108;transport=TLS State:: CS_SYNC Flags:: 0 Cflag:: 64 Socket:: tls:62.124.111.222:5061 Methods:: 7999 After running a socket listener on 192.168.0.31 on the OpenSIPS host: $ socat TCP4-LISTEN:2310,bind=192.168.0.31,reuseaddr - SUBSCRIBE sip:mylogin-os...@192.168.0.31:2310;transport=tls;line=2acy67zm SIP/2.0 Via: SIP/2.0/TCP 86.90.39.44;branch=G4z9hb82dK8.f144.0 To: sip:mylogin-os...@name.host.tld;tag=ty6sjh9iz9 From: sip:mylogin-os...@name.host.tld;tag=6c9d4319c74d756e6b696-baa1 CSeq: 11 SUBSCRIBE Call-ID: b1c04118-8...@86.90.39.44 Content-Length: 0 User-Agent: OpenSIPS (1.6.1-tls) Max-Forwards: 70 Event: dialog;sla Contact: sip:prese...@name.host.tld Expires: 610 Any ideas? If you could try to get a capture of the whole flow - starting with REGISTER, etc.plus the logs... But first check the usrloc (see 1)) I'll do that next, when I have time over the weekend. Capturing is actually quite difficult because everything is TLS, and there are more than one phone, as well as a voicemail server... Thanks for helping so far, and if you think of anything based on your suggestion 1) and my answer, then please advise. Thanks, Brian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] What is protocol/port mismatch?
Hello Bogdan, An ven., janv 29, 2010, Bogdan-Andrei Iancu schrieb: opensipsl...@encambio.com wrote: ...and I see this in the log: warning opensips[2717]: WARNING:core:get_send_socket: protocol/port mismatch some hundreds of times per day (about once every 20 minutes per registered UAC.) I have this in the route script: listen = tls:name.host.tld:5061 [...] t_relay(name.host.tld:5080); Do you use force_send_socket() functions in your script, before the t_relay ? No, I have 'force_rport()' in the script, but neither 'force_send_socket()' nor 'force_tcp_alias()' is there at all. Are you suggesting that I use it before t_relay to the server with a different transport? Is the basic idea of this warning message that OpenSIPS exchanges a SIP message with a UA over a certain transport (TLS in this case) and port number (5061 in this case), but a t_relay in the route script forwards the message over a different transport or to a different port number? If so, what is being compared and how is it compared? Port1 == Port2 Transport1 == Transport2 Transport1 Port1 == Transport2 Port2 ...? Thanks, Brian ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] B2Bua : send 2 INVITE without caring about the 200 OK
hello, I have a question concerning the B2Bua : I'd like to send 2 INVITE without 200 OK. Indeed I'd like to send the second INVITE even if I haven't received the 200 OK of the first one. But I find that the B2Bua is too much sequetial Is there a solution ? thank you Une messagerie gratuite, garantie à vie et des services en plus, ça vous tente ? Je crée ma boîte mail www.laposte.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] B2BUA - Opensips crash on b2b_init_request(top hiding);
Hi Max, I am investigating the reported crash and I will find the problem. I will let you know when it is done to update your code. Thanks and regards, -- Anca Vamanu www.voice-system.ro Max Mühlbronner wrote: Hello everyone, I have a problem with opensips 1.6.1-notls, everything else worked fine but at this point i can not get the b2bua module (topology hiding scenario) to work. Lately i have added the b2bua module and while testing Opensips crashes whenever a request hits b2b_init. I thought it could be an misconfiguration on my side, but could not find anything wrong. I tried many things but could not find any solution, now i see there is also an open bug which describes the same problem: Does anyone else have these issues or similar crashes with b2bua, or any ideas to verify if this could be a valid bug? - *opensips crashes on reply recieved to b2bua - ID: 2937441 Looking forward to any ideas. Best Regards Max M. * ___ 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] B2BUA not passing ACKs
Hi Brian,
opensipsl...@encambio.com wrote:
Hello Anca,
Sorry for the delay.
An lun., janv 04, 2010, Anca Vamanu schrieb:
There is a misunderstanding from your side in what the b2b scenario
documents are concerned ( please read carefully the documentation -
http://www.opensips.org/Resources/B2buaTutorial ).
It's hard to figure out which document to read, as the documents are
so unclear that they need documentation themselves. What I mean is:
Why are there two documents listed on the website for the same
thing. One called 'B2buaTutorial' and the other 'B2buaTutorial16'?
Is the second a older document only useful for OpenSIPS 1.6.0, or
is it a newer version of the document B2buaTutorial?
The documentation versions, as it is normal, refer to the version of
code, so the newest version is for the devel branch. Anyhow, the only
addition in devel is support for REFER scenario.
There are also no links to plain text config files, and everything
is HTML. A complete working route script is not available.
You are right about the scripts, I will add links to text files.
There is a complete working opensips script here:
http://www.opensips.org/Resources/B2bConfigExample
The important thing is that there should only be rules in the
scenario for requests that need a special handling. In the prepaid
scenario - when the BYE from the media server is received the
caller must be connected to a human operator, so we have a rule for
this. All the other requests need only simple pass forward - so if
an ACK is received from one side it only need to be forwarded to
the other. 'pass forward' is the implicit action and it will be
applied to all requests that don't match a rule.
Thanks for clearing that up (about the implicit action.) I think
I understand better now, but still I would like to start from the
beginning and use the supplied prepaid.xml (which I assume is
correctly written.)
Yes, it is correctly written.
I see that you say that the prepaid scenario does not work for you.
What version are you testing with?
Solaris 11 x86 (nv-b91)
OpenSIPS 1.6.0 with TLS
I've copied the example 'prepaid.xml' word for word from the URL:
http://www.opensips.org/Resources/B2buaTutorial16
Here are the relevant parts of the route script:
listen = udp:name.host.tld:5060
listen = tls:name.host.tld:5061
modparam(tm, pass_provisional_replies, 1)
modparam(b2b_entities, server_address, sip:b2...@name.host.tld)
modparam(b2b_logic, script_scenario,
/pfx/etc/opensips/b2bua/prepaid.xml)
if (has_totag()) {
if (loose_route()) {
# code here
}
}
if (!is_method(REGISTER|MESSAGE)) {
record_route();
}
if (is_method(INVITE) src_ip != myself) { # Start of B2BUA
if (!t_newtran()) { # logic block, do
sl_reply_error(); # media announcements
exit; # to users
}
b2b_init_request(prepaid, sip:playso...@123.123.123.123:5080,
sip:playso...@123.123.123.123:5080);
exit;
}
if (src_ip != myself) {
if ($hdr(P-hint) != outbound) {
append_hf(P-hint: outbound\r\n);
}
}
Does that look like it should work? What about the parameters
'123.123.123.123'? Is 't_newtran' necessary?
t_newtran is necessary because b2b should not handle retransmissions.
And yes, the configuration file seems correct and should work. If it
doesn't, try to find the exact problem. Check if there are errors in
opensips log and watch the network traffic.
If you see something not working as in the schema from the
documentation, send a detailed report.
Regards,
--
Anca Vamanu
www.voice-system.ro
Regards,
Brian
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Re: [OpenSIPS-Users] Experimenting with B2b modules
Hi Jayesh,
FIrst of all, the error that those parameters do not exist is normal
since they do not exist :). costum_headers is a parameter that was added
in devel branch, and the correct name for the other parameter is
clean_period. But I have checked now, and in the documentation it is in
fact written cleanup_period. I will fix this in the code to really
have cleanup_period.
Can you please investigate the core file with gdb and print here the
backtrace? There is also another crash report and I am investigating it now.
Regards,
--
Anca Vamanu
www.voice-system.ro
Jayesh Nambiar wrote:
Hi All,
I have been trying to do some experiments with B2b modules in
Opensips. What I am trying to do is create another OpenSips instance
which will only act as Topology Hiding Server in front of my proxy.
So calls processed from my proxy will go to the B2b Opensips instance,
the B2b instance will extract a header which will contain the
destination domain and route the call to that domain in B2b mode (Is
this doable?).
First issue:
I get these errors on loading the parameters:
parameter cleanup_period not found in module b2b_logic
parameter custom_headers not found in module b2b_logic
I have compiled Opensips 1.6.1 from source in Debian.
Second Issue:
I commented these parameters and tried running opensips but ran into
Segfault.
Snippet of my cfg file:
loadmodule b2b_entities.so
modparam(b2b_entities, server_address, sip:b2...@opensips.org
mailto:sip%3ab2...@opensips.org)
loadmodule b2b_logic.so
#modparam(b2b_logic, cleanup_period, 60)
#modparam(b2b_logic, custom_headers, Status)
route {
if(method==INVITE) {
$rd = $hdr(Dest);
b2b_init_request(top hiding);
exit;
}
}
Can i find few more examples somewhere of using the B2B modules in
opensips so that i can start thinking of how do I integrate these
features into my current setup !!
Any help is very much appreciated as always.
Thanks,
--- Jayesh
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Re: [OpenSIPS-Users] B2BUA - Opensips crash on b2b_init_request(top hiding);
Hi again,
i have made a recompile/ new setup and still receive a segmentation
fault, it was fine for one call and so i thought it was finally working
(looked at the trace on another machine and the contact header was
modified correctly by B2bua) and then on the next Call it crashed again.
I hope the coredump may help in any way.
Program terminated with signal 11, Segmentation fault.
[New process 31135]
#0 parse_headers (msg=0x2e343331, flags=18446744073709551615, next=0)
at parser/msg_parser.c:298
298 end=msg-buf+msg-len;
(gdb) bt
#0 parse_headers (msg=0x2e343331, flags=18446744073709551615, next=0)
at parser/msg_parser.c:298
#1 0xb7b13cf4 in b2b_send_reply (et=B2B_SERVER, b2b_key=0xa79cfdb8,
code=200, text=0x81bd994, body=0xbf8f220c, extra_headers=0xbf8f2204) at
dlg.c:765
#2 0xb7b04f23 in b2b_logic_notify (src=1, msg=0x81bd978,
key=0xa79ca46c, type=1, param=0xa79cdedc) at logic.c:444
#3 0xb7b06343 in b2b_client_notify (msg=0x81bd978, key=0xa79ca46c,
type=1, param=0xa79cdedc) at logic.c:938
#4 0xb7b14a14 in b2b_tm_cback (htable=0xa7961638, ps=0xb7b66e54) at
dlg.c:1542
#5 0xb7b0cf1b in b2b_client_tm_cback (t=0xa79cee44, type=512,
ps=0xb7b66e54) at client.c:44
#6 0xb7b4250b in run_trans_callbacks (type=512, trans=0xa79cee44,
req=0x0, rpl=0x81bd978, code=200) at t_hooks.c:208
#7 0xb7b58cae in local_reply (t=0xa79cee44, p_msg=0x81bd978, branch=0,
msg_status=200, cancel_bitmap=0xbf8f2540) at t_reply.c:1339
#8 0xb7b59ff1 in reply_received (p_msg=0x81bd978) at t_reply.c:1484
#9 0x08067172 in forward_reply (msg=0x81bd978) at forward.c:559
#10 0x080978db in receive_msg (
buf=0x8174380 SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP
62.134.184.16;rport=5060;received=62.134.184.16;branch=z9hG4bKe2c8.d65d92e.0\r\nFrom:
sip:49302318910...@62.134.184.11;tag=44c604a5e7912308af351193a53d7a0e-833e\r\nTo:
s..., len=950, rcv_info=0xbf8f2664) at receive.c:200
#11 0x080d8a14 in udp_rcv_loop () at udp_server.c:492
#12 0x0806e329 in main (argc=9, argv=0xbf8f2804) at main.c:818
also more detailed :
(gdb) bt full
#0 parse_headers (msg=0x2e343331, flags=18446744073709551615, next=0)
at parser/msg_parser.c:298
hf = value optimized out
itr = value optimized out
tmp = 0x0
rest = value optimized out
end = 0xa79cfde3
.0.1264773824sip:1234493055...@62.134.184.16sip:49302318910...@62.134.184.11
orig_flag = value optimized out
__FUNCTION__ = parse_headers
#1 0xb7b13cf4 in b2b_send_reply (et=B2B_SERVER, b2b_key=0xa79cfdb8,
code=200, text=0x81bd994, body=0xbf8f220c, extra_headers=0xbf8f2204) at
dlg.c:765
hash_index = 274
local_index = 0
dlg = (b2b_dlg_t *) 0xa79cdaec
to_tag = value optimized out
tm_tran = (struct cell *) 0xa79c1b88
msg = (struct sip_msg *) 0x2e343331
buffer = \206òõ·h\000\000\000È÷d\b\b \000\000\b
\000\000Xöd\b`±÷·t.é·0ød\bF[é·È÷d\bh\000\000\000Xöd\b
\037\217¿Ûúè·Ð÷d\b`±÷·ô\237÷·0ød\b¸\037\217¿0ød\bÐ÷\000\000`±÷·Xöd\bÔ\037\217¿8é·`±÷·Xöd\bPöd\bô\237÷·¼/\032\b\000\000\000\000è/\032\b\030!\217¿\214t\n\b¼/\032\bxÙ\033\b\000\000\000\000È÷d\bô\237÷·\025\000\000\000Xöd\b°
\217¿\226øï·Ð÷d\bÐ÷d\b\\\000\000\000\...@\000\000 \224÷·X \217¿...
p = value optimized out
ehdr = {s = 0x2 Address 0x2 out of bounds, len = -1482891545}
table = (b2b_table) 0xa796062c
pto = value optimized out
TO = {error = -1081139060, body = {s = 0xbf8f20a8 ¾, len =
134783674}, uri = {s = 0x81bd978 ., len = 135930932}, display = {
s = 0xbf8f208c ÆD\027\b\001, len = 9}, tag_value = {s = 0x0, len =
0}, parsed_uri = {user = {s = 0x10058 Address 0x10058 out of bounds,
len = 136042872}, passwd = {s = 0x81a2e00 \017, len =
135933548}, host = {s = 0x7e Address 0x7e out of bounds, len =
135742662}, port = {
s = 0x1 Address 0x1 out of bounds, len = -1081138960}, params =
{s = 0xbf8f2108 =í\033\b, len = 135204128}, headers = {
s = 0xbf8f20f0 \\\031\bv, len = 136044600}, port_no = 190,
proto = 0, type = 3080147028, transport = {s = 0xbf8f20c8 , len =
135931420}, ttl = {
s = 0x6 Address 0x6 out of bounds, len = 136042872}, user_param
= {s = 0xbf8f2130 \001, len = 136044600}, maddr = {s = 0x0, len = 1},
method = {
s = 0x81a432c \017, len = 0}, lr = {s = 0x81bece0 Ø, len =
-1213161172}, r2 = {s = 0x8195c20 , len = -1081138684}, transport_val = {
s = 0xbf8f2148 8\\217¿#O°·, len = -1213204822}, ttl_val = {s =
0x8195c20 , len = 118}, user_param_val = {s = 0x0, len = 136044652},
maddr_val = {
s = 0x16 Address 0x16 out of bounds, len = 10}, method_val = {s
= 0x Address 0x out of bounds, len = 136047933},
lr_val = {
s = 0x0, len = 0}, r2_val = {s = 0xb7b09b85 Require, len =
-1213162611}}, param_lst = 0x0, last_param = 0x0}
__FUNCTION__ = b2b_send_reply
#2 0xb7b04f23 in b2b_logic_notify (src=1, msg=0x81bd978,
key=0xa79ca46c, type=1, param=0xa79cdedc) at logic.c:444
hash_index =
[OpenSIPS-Users] Problem with miss call logging in radius (radacct) database
Hi all,
I have upgraded my CDRTool-6.9.x to CDRTool-7.0.0 . I could see the cdrs
of miss calls when i was using the CDRTool-6.9.x but after upgrading i do
not see the missed call logs for CDRTool-7.0.0 nor in the radacct table. I
have followed all the instructions for the upgrade.
Below is the radius log.
*rad_recv: Accounting-Request packet from host 127.0.0.1 port 60042, id=222,
length=352
Acct-Status-Type = Failed
Service-Type = Sip-Session
Sip-Response-Code = 487
Sip-Method = Invite
Event-Timestamp = Jan 29 2010 10:03:24 EST
Sip-From-Tag = 1b1bad63
Sip-To-Tag = 177306757
Acct-Session-Id =
3e19711acb6a541dYzE5ODJiZGU5NTg5ZWQ5MmE5NmE4MGFlOGMzYTI1NjA.
User-Name = sip:5...@aaa.aaa.com sip%3a5...@aaa.aaa.com
Calling-Station-Id = sip:5...@aaa.aaa.com sip%3a5...@aaa.aaa.com
Called-Station-Id = sip:1...@aaa.aaa.com sip%3a1...@aaa.aaa.com
Sip-Translated-Request-URI = sip:1...@202.63.111.2:5060
Source-IP = 59.92.156.18
Source-Port = 50682
Canonical-URI = sip:1...@aaa.aaa.com sip%3a1...@aaa.aaa.com
Contact = sip:5...@59.92.156.18:50682
SIP-Proxy-IP = aaa.aaa.com
NAS-Port = 5060
Acct-Delay-Time = 0
NAS-IP-Address = 127.0.0.1
+- entering group preacct {...}
++[preprocess] returns ok
[acct_unique] Hashing 'NAS-Port = 5060,Client-IP-Address =
127.0.0.1,NAS-IP-Address = 127.0.0.1,Acct-Session-Id =
3e19711acb6a541dYzE5ODJiZGU5NTg5ZWQ5MmE5NmE4MGFlOGMzYTI1NjA.,User-Name =
sip:5...@aaa.aaa.com sip%3a5...@aaa.aaa.com'
[acct_unique] Acct-Unique-Session-ID = 0ef7a78049ee134a.
++[acct_unique] returns ok
++[files] returns noop
+- entering group accounting {...}
[detail] expand:
/usr/local/var/log/radius/radacct/%{Client-IP-Address}/detail-%Y%m%d -
/usr/local/var/log/radius/radacct/127.0.0.1/detail-20100129
[detail]
/usr/local/var/log/radius/radacct/%{Client-IP-Address}/detail-%Y%m%d expands
to /usr/local/var/log/radius/radacct/127.0.0.1/detail-20100129
[detail] expand: %t - Fri Jan 29 10:03:24 2010
++[detail] returns ok
++[unix] returns noop
[radutmp] expand: /usr/local/var/log/radius/radutmp -
/usr/local/var/log/radius/radutmp
rlm_radutmp: NAS localhost port 5060 unknown packet type 15)
++[radutmp] returns noop
[sql] Unsupported Acct-Status-Type = 15
++[sql] returns noop
[attr_filter.accounting_response] expand: %{User-Name} -
sip:5...@aaa.aaa.com sip%3a5...@aaa.aaa.com
attr_filter: Matched entry DEFAULT at line 12
++[attr_filter.accounting_response] returns updated
Sending Accounting-Response of id 222 to 127.0.0.1 port 60042
Finished request 1.
Cleaning up request 1 ID 222 with timestamp +48
Going to the next request
Ready to process requests.*
--
Thanking You,
Ashwini BR Naidu
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[OpenSIPS-Users] t_relay() not relaying payload
We currently have opensips setup to route through another carrier for
certain calls. All signaling and media works well except for reinvites.
if (has_totag()) {
# sequential request within a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method(BYE)) {
setflag(1); # do accounting ...
setflag(3); # ... even if the
transaction fails
} else if (is_method(INVITE)) {
# even if in most of the cases is
useless, do RR for
# re-INVITEs alos, as some buggy
clients do change route set
# during the dialog.
record_route_preset(hidden);
}
fix_nated_contact();
t_relay();
...
The first reinvite passes through fine with the nated contact fixed
for the contact field. The second reinvite does not get relayed
correctly. Instead a udp packet with no SIP payload at all is sent to
the UA. We can't find any particular error in the debug log.
Does anyone have thoughts on this?
--
Thamer
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[OpenSIPS-Users] carrierroute rule_fixup log entry
Hello All, Quick question on a log file entry with regards to carrierroute 1.4.5-notls. opensips[32717]: INFO:carrierroute:rule_fixup: fixing tree 0 opensips[32717]: INFO:carrierroute:rule_fixup_recursor: hashless rule with host x.x.x.x hash hash_index 1 The second line repeats for entry in the table. I have 6 carriers defined in the route_tree; 4 NANP PSTN providers, 1 International for LCR weighting via prob, and default. I use cr_user_carrier to load the avp of the preferred NANP PSTN provider before calling cr_route. Is there need for concern with the INFO log entries at startup and route reloads? Cheers, Kyle -- Kyle Romulas DH Platform-Services kyle.romu...@gmail.com attachment: winmail.dat___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Problem with miss call logging in radius (radacct) database
Hi, El 29/01/10 16:28, ASHWINI NAIDU escribió: Hi all, I have upgraded my CDRTool-6.9.x to CDRTool-7.0.0 . I could see the cdrs of miss calls when i was using the CDRTool-6.9.x but after upgrading i do not see the missed call logs for CDRTool-7.0.0 nor in the radacct table. I have followed all the instructions for the upgrade. Did you also upgrade the MySQL stored procedures? Regards, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] lookup b flag - one registration at a time
Bogdan,
Over the last day or so I've noticed some other weirdness too, particularly
with engage_media_proxy() not working reliably. I'm thinking I might have
pieces-parts of different builds laying around. I'm not always the best at
removing /usr/local/lib/opensips before I install a new build.
I started fresh with 6547 tonight being sure to remove the old lib directory
before installing. Both this issue and the Mediaproxy one seem to have
resolved themselves.
How much manual housekeeping should be required after an update from svn? I'd
prefer not to believe I'm the only one who has encountered issues like this. :)
- Jeff
On Jan 27, 2010, at 6:54 AM, Bogdan-Andrei Iancu wrote:
Hi Jeff,
Got your email - I used the SB you had and from script check_address() -
to be able to check with your IPs . It worked
Also I tested using some IPs from our network versus
check_source_address() and again, It worked.
Please apply the attached patched (just debugs) and use debug=4 to
capture the logs to see what is going wrong in your case.
Regards,
Bogdan
Bogdan-Andrei Iancu wrote:
I do not think this is related to the registrar update, but rather to
the permissions fixes from today (see rev 6539).
I did some tests and seams to work ok - if you want, you can privately
send me the dump of the address table and the IP which does not match,
just to check what is going on.
Regards,
Bogdan
Jeff Pyle wrote:
Bogdan,
I didn't catch it before, but it appears the update seems to have broken
the following:
if (check_source_address(1,$avp(s:peer_uuid))) {
.. do stuff ..
}
The check_source_address returns false. As a temporary workaround I've
added lines the following lines:
if ($si == '2.3.4.5') {
.. do stuff ..
} else if ($si == '2.3.4.6') {
.. do the same stuff .. {
} else if ...
- Jeff
On Jan 25, 2010, at 11:49 PM, Jeff Pyle wrote:
Bogdan,
Early results have this update working perfectly.
- Jeff
On Jan 25, 2010, at 10:29 AM, Jeff Pyle wrote:
Bogdan,
Will do. Thanks.
- Jeff
On Jan 25, 2010, at 9:35 AM, Bogdan-Andrei Iancu wrote:
Hi Jeff,
See revision #6527 on trunk - if you could run some more tests on it and
report if works ok, it will be great.
Regards,
Bogdan
Jeff Pyle wrote:
The f flag sounds fantastic. Thanks.
- Jeff
On Jan 18, 2010, at 9:24 AM, Bogdan-Andrei Iancu wrote:
Hi Jeff,
Jeff Pyle wrote:
Iñaki,
On Jan 9, 2010, at 5:00 PM, Iñaki Baz Castillo wrote:
El Sábado, 9 de Enero de 2010, Jeff Pyle escribió:
Hello,
The docs say that when using the b flag with lookup() when
multiple
records are present, it will load only the one with the highest q.
What
if the q is the same for all? How does it decide which to use?
I've not tested it with multiple users sharing same q. however it
should
fetch all the users with highest q, not just one of them.
Perhaps I'm asking the wrong question. I'm looking to allow only one
registration per user in the sense that if a second successful
registration comes in it will replace tne existing one. My approach
so far is to use a max_contacts=2 and the lookup() function with the
b flag to retrieve only one.
maybe without the b flag as the b flag will return you all the
registered contacts.
max_contacts=1 returns a 503 to the new replacement registration
request, so that's out.
Perhaps the hot ticket is to run an all-DB mode running a manual
mysql query with avp_db_query after successful REGISTER
authentication but before the save() so we can remove any existing
registrations before the new one is saved. Thoughts?
No way - the SIP contact matching is much to complicated to do it at
DB
level.
As I found that kind of behaviour was more and more asked by people, I
will add a new flag f to force at save() time the override of the
existing contacts if the max_contacts() was exceeded.
Regards,
Bogdan
- Jeff
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Bogdan-Andrei Iancu
www.voice-system.ro
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Regards,
Jeff Pyle
Director, Voice Engineering
Fidelity Voice Data | 23250 Chagrin Blvd, Suite 250 | Beachwood, Ohio
44122
P: 216-245-4106
F: 216-595-0706
E: jp...@fidelityvoice.com
Visit us at http://www.fidelityvoice.com
2008 2009 Inductee to the prestigious Weatherhead 100
--
Bogdan-Andrei Iancu
Re: [OpenSIPS-Users] Users Digest, Vol 18, Issue 97
Hi Anca,
Thanks for your reply.
I was checking the module documentation of 1.6 version for b2b_logic where
both the parameters custom_headers as well as cleanup_period is documented.
Later I realized that it points to the devel folder.
I will check with the devel version anyways. But just for my understanding;
is the code snippet pasted kind of correct. I mean if the $rd is specified
to the script and b2b_init is called will rest of the work like initiating
dialogs on both sides and bridging calls will be handled automatically.
Nothin else needs to be done??
Thanks for all the help !!
--- Jayesh
Hi Jayesh,
FIrst of all, the error that those parameters do not exist is normal
since they do not exist :). costum_headers is a parameter that was added
in devel branch, and the correct name for the other parameter is
clean_period. But I have checked now, and in the documentation it is in
fact written cleanup_period. I will fix this in the code to really
have cleanup_period.
Can you please investigate the core file with gdb and print here the
backtrace? There is also another crash report and I am investigating it
now.
Regards,
--
Anca Vamanu
www.voice-system.ro
Jayesh Nambiar wrote:
Hi All,
I have been trying to do some experiments with B2b modules in
Opensips. What I am trying to do is create another OpenSips instance
which will only act as Topology Hiding Server in front of my proxy.
So calls processed from my proxy will go to the B2b Opensips instance,
the B2b instance will extract a header which will contain the
destination domain and route the call to that domain in B2b mode (Is
this doable?).
First issue:
I get these errors on loading the parameters:
parameter cleanup_period not found in module b2b_logic
parameter custom_headers not found in module b2b_logic
I have compiled Opensips 1.6.1 from source in Debian.
Second Issue:
I commented these parameters and tried running opensips but ran into
Segfault.
Snippet of my cfg file:
loadmodule b2b_entities.so
modparam(b2b_entities, server_address,
sip:b2...@opensips.orgsip%3ab2...@opensips.org
mailto:sip%3ab2...@opensips.org sip%253ab2...@opensips.org)
loadmodule b2b_logic.so
#modparam(b2b_logic, cleanup_period, 60)
#modparam(b2b_logic, custom_headers, Status)
route {
if(method==INVITE) {
$rd = $hdr(Dest);
b2b_init_request(top hiding);
exit;
}
}
Can i find few more examples somewhere of using the B2B modules in
opensips so that i can start thinking of how do I integrate these
features into my current setup !!
Any help is very much appreciated as always.
Thanks,
--- Jayesh
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