[OpenSIPS-Users] format of the CDR in OpenCP 1.4 using OpenSIPS 1.6.2
Hi, I installed and OpenSIPS 1.6.2 and openCP 1.4. I configured the cdrviewer part. I am puzzled of the format of the CDRs basically the acc/cdrs, cron and display mechanism in the openSIPS CP is working fine. I got all the information stored in the DB, in the CDRS table (coming from acc processing). but in the OpenCP by default I have fields I do not have in the acc table (caller, Callee, Leg Type). Do we have to manually define extra accounting fields in acc to get the correct display in the CP? If so, shall the tables (acc cdrs) be modified accordingly? by default the cdrs table in 1.6 is limited to the following fields - cdr_id - call_start_time - duration - sip_call_id - sip_from_tag - sip_to_tag - created thus no caller and callee here..or did I miss something? regards Morgan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] avp_db_load() fetch multiple attribute at one time
Hello List, Please let me know how to load multiple attribute at one time. let say. my usr_preferences table has 3 attribute for single user (e.g fwdoffline,fwdbusy,callfwd). Now I load these 3 attributes at 3 times. I want to reduce DB queries :) I did this. --8 avp_db_load($ruri/username, $avp(s:)); xlog(TEST: OFF LINE FORWARD TO: $avp(s:fwdoffline)\n); xlog(TEST: BUSY FORWARD TO to: $avp(s:fwdbusy)\n); xlog(TEST: CALL FORWARD TO : $avp(s:callfwd)\n); --8 and the result was TEST : OFF LINE FORWARD TO: null TEST : BUSY FORWARD TO to: null TEST : CALL FORWARD TO : null P.S I followed this http://www.mail-archive.com/us...@openser.org/msg00589.html Thank you, Sujeev ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips ACK routing problem
Hi, using Opensips 1.6.2. We are using Opensips as outbound proxy using TLS just for final hop between UAC and Opensips. Other legs of the call will be udp. We have a test set up with Sipgate (but same occurs with other providers) where an incoming INVITE to the UAC via opensips results in 200ok generated by UAC and then the ACK coming back from sipgate server is routed over udp connection as opposed to tls, opensips seems to be ignoring the Route header in the ACK If the contact header has a sips uri opensips will route it correctly but some voip providers do not like sips uri in contact header and we are only using tls for the final hop so are not using a sips uri for contact header. Do we have our understanding of the spec wrong or should opensips be routing according to the route header or should it use the contact header. Opensips ip 172.230.135.190 UAC ip 81.13.94.206 U 217.10.79.23:5060 - 172.230.135.190:5060 ACK sip:9082...@81.13.94.206:59053 SIP/2.0. Via: SIP/2.0/UDP 217.10.79.23:5060;branch=z9hG4bK59c7.0bbb2877.2. Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK59c7.0bbb2877.2. Via: SIP/2.0/UDP 217.10.79.23:5060;received=217.10.68.226;branch=z9hG4bK6a423428. Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK6a423428;rport=5060. Route: sip:172.230.135.190;r2=on;lr=on;ftag=as126bf37e,sip:172.230.135.190:5061;transport=tls;r2=on;lr=on;ftag=as126bf37e. From: anonymous sip:anonym...@sipgate.co.uk;tag=as126bf37e. To: sip:00441519082...@sipgate.co.uk;tag=94gc3N42r9X5D. Contact: sip:anonym...@217.10.66.71. Call-ID: 181c48541a5ad00d04322ccc0752c...@sipgate.co.uk. CSeq: 102 ACK. Max-Forwards: 67. Content-Length: 0. X-hint: rr-enforced. . U 172.230.135.190:5060 - 81.13.94.206:59053 ACK sip:9082...@81.13.94.206:59053 SIP/2.0. Via: SIP/2.0/UDP 172.230.135.190;branch=z9hG4bK59c7.6f77cf2.3. Via: SIP/2.0/UDP 217.10.79.23:5060;rport=5060;received=217.10.79.23;branch=z9hG4bK59c7.0bbb2877.2. Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK59c7.0bbb2877.2. Via: SIP/2.0/UDP 217.10.79.23:5060;received=217.10.68.226;branch=z9hG4bK6a423428. Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK6a423428;rport=5060. From: anonymous sip:anonym...@sipgate.co.uk;tag=as126bf37e. To: sip:00441519082...@sipgate.co.uk;tag=94gc3N42r9X5D. Contact: sip:anonym...@217.10.66.71. Call-ID: 181c48541a5ad00d04322ccc0752c...@sipgate.co.uk. CSeq: 102 ACK. Max-Forwards: 66. Content-Length: 0. X-hint: rr-enforced. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] LB under a 64bit Debian Lenny
Hi Bogdan, Any luck regarding this problem? No need to hurry, I'm just asking :-) Dimitri 2010/8/10 Bogdan-Andrei Iancu bog...@voice-system.ro Hi Dimitri, Hard to tellI will try to do a setup on 64b machine with more than 32. Let me come back to you... Regards, Bogdan Dmitri G. wrote: Hi, I have compiled Opensips 1.6.2 (downloaded from opensips website) and I have used the LB opensips.cfg file on opensips.org http://opensips.org. I have also added 80 destinations to the load_balancer table. For some reason, destinations above Id 30 aren't considered, always gettibg back 500 Service full. In the same box, I have used this source and built a 32bit Opensips with pbuilder, it works fine for all destinations. With using Opensips 1.6.3 (downloaded from opensips.org http://opensips.org), the errors are the same (with 64bit, destinations above ID 30 doesn1t work at all). when I run lb_list, all destinations are listed, and marked as active (true for both opensips 1.6.2 and 1.6.3, both 32bit and 64bit) In syslog, there is no errors. What can cause this? Thanks, Dimitri ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 20 - 24 September 2010, Frankfurt, Germany www.voice-system.ro ___ 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] Next Gateway with Load balance module
Hi, Is it possible to use the next gateway from within a failure route when using the load balance module? Thanks, Ross ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] avp_db_load() fetch multiple attribute at one time
http://www.opensips.org/html/docs/modules/1.6.x/avpops.html#id228513 second example. I believe the main problem is the : (colon) I'm also not sure about your var $ruri. This should work: avp_db_load($ru/username, s); On Mon, Aug 23, 2010 at 2:55 AM, Sujeev suppo...@meewadaya.com wrote: Hello List, Please let me know how to load multiple attribute at one time. let say. my usr_preferences table has 3 attribute for single user (e.g fwdoffline,fwdbusy,callfwd). Now I load these 3 attributes at 3 times. I want to reduce DB queries :) I did this. --8 avp_db_load($ruri/username, $avp(s:)); xlog(TEST: OFF LINE FORWARD TO: $avp(s:fwdoffline)\n); xlog(TEST: BUSY FORWARD TO to: $avp(s:fwdbusy)\n); xlog(TEST: CALL FORWARD TO : $avp(s:callfwd)\n); --8 and the result was TEST : OFF LINE FORWARD TO: null TEST : BUSY FORWARD TO to: null TEST : CALL FORWARD TO : null P.S I followed this http://www.mail-archive.com/us...@openser.org/msg00589.html Thank you, Sujeev ___ 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 process dying (dialplan module?)
Hi list,
my opensips process is dying constantly.
According the dump core, it seems (for me) to be a function in
dialplan that is causing it.
Follows the dump from gdb.
Is someone facing this kind of problem?
opensips version is: opensips 1.6.1-notls (i386/linux)
Thanks,
Core was generated by `/usr/local/sbin/opensips -P
/var/run/opensips.pid -m 256'.
Program terminated with signal 11, Segmentation fault.
#0 0x00e450de in rule_translate (msg=0x81dbeac, string=...,
rule=0xa821fda0, result=0xbfc34c4c) at dp_repl.c:192
192 memcpy(result-s + result-len,
match.begin, match.len);
(gdb) bt full
#0 0x00e450de in rule_translate (msg=0x81dbeac, string=...,
rule=0xa821fda0, result=0xbfc34c4c) at dp_repl.c:192
repl_nb = 0
offset = 0
token = {offset = 0, size = 2, type = REPLACE_NMATCH, u =
{nmatch = 2, c = 2 '\002', spec = {type = PVT_NULL, getf = 0,
setf = 0, pvp = {pvn = {type = 1, u = {isname = {type =
0, name = {n = 9534827, s = {s = 0x917d6b \201É\302\v,
len = -1}}}, dname = 0x0}}, pvi = {type =
53, u = {ival = 10197728, dval = 0x9b9ae0}}, pvv = {
s = 0xbfc362ec
\017\352\n\b\230\353\033\b\254\276\035\b, len = 9531122}}, pvc =
0xa821f964, trans = 0xe4721c}}}
subst_comp = 0xa821fc8c
repl_comp = 0xa821fc14
match = {begin = 0x830457a 1150212...@200.225.81.91:5060, len = -3366}
sv = {rs = {s = 0x4a98 Address 0x4a98 out of bounds, len =
136167084}, ri = 137377864, flags = 12}
uri = 0x0
__FUNCTION__ = rule_translate
#1 0x00e463eb in translate (msg=0x81dbeac, input=...,
output=0xbfc34c4c, idp=0xa821497c, attrs=0x0) at dp_repl.c:346
rulep = 0xa821fda0
indexp = value optimized out
rez = value optimized out
__FUNCTION__ = translate
#2 0x00e3d2f7 in dp_translate_f (msg=0x81dbeac, str1=0x81d6dd4
\001, str2=0x81d6e50 \001) at dialplan.c:368
dpid = 15
input = {s = 0x8304578 551150212...@200.225.81.91:5060, len = 12}
output = {s = 0xe4ec20 1150212...@200.225.81.91:5060, len = 0}
idp = 0xa821497c
repl_par = 0x81d6e50
attrs = {s = 0x81c7ee7 post_ruri_id), len = 12}
attrs_par = 0x0
__FUNCTION__ = dp_translate_f
#3 0x080546fd in do_action (a=0x81c845c, msg=0x81dbeac) at action.c:967
val_s = {s = 0x81dbeac \316#\001, len = 136120380}
aux = {s = 0x3bf816 \205\300\017\205b\001, len = 136035128}
ret = value optimized out
v = value optimized out
to = value optimized out
p = value optimized out
tmp = value optimized out
new_uri = value optimized out
end = value optimized out
crt = value optimized out
len = value optimized out
user = value optimized out
uri = {user = {s = 0x3ce110 \200\001, len = 136035128},
passwd = {s = 0x81c2155 post_ruri_id), len = -1077719320},
host = {s = 0x81c7fdc \001, len = 136167084}, port = {s =
0x4 Address 0x4 out of bounds, len = -1077719208}, params = {
s = 0x80aea0f
\211\306\...@\377\377\377\213m\020\211l$\b\213u\f\211t$\004\213s\f\211\064$\350\071\376\377\377\211ƅ\300\017---type
return to continue, or q return to quit---
\205\035\377\377\377\213u\020\211t$\b\213E\f\211D$\004\213{\030\211$\350\026\376\377\377\211\306\351\375\376\377\377\377$\275\254c\025\b\213}\020\211|$\b\213M\f\211L$\004\213S\f\211\024$\350\357\375\377\377\211ƃ,
incomplete sequence \370, len = 136085424},
headers = {s = 0x81dbeac \316#\001, len = 0}, port_no = 0,
proto = 0, type = 3217249364, transport = {s = 0x0, len = 256},
ttl = {s = 0x81dbeac \316#\001, len = -1077719352},
user_param = {s = 0x81d4384 \004, len = -1077719320}, maddr = {
s = 0x808f1a2
\205\300\017\205+\377\377\377\366E\354\001t\022\213\025\\9\025\b\213\r`9\025\b\211U\340\211M\344\213u\344\211u\274\213E\310\001\360\213}\024;\a\017\215W\001,
len = 15}, method = {s = 0x7 Address 0x7 out of bounds, len =
-1077719352}, lr = {
s = 0x6 Address 0x6 out of bounds, len = 1}, r2 = {s =
0x816752e , len = -1474188324}, transport_val = {s = 0x0,
len = 2}, ttl_val = {s = 0x9f3269b 0.0.0.0, len = 14},
user_param_val = {
s = 0x Address 0x out of bounds, len =
4}, maddr_val = {
s = 0xfff8 Address 0xfff8 out of bounds, len =
136120612}, method_val = {s = 0xe4e870 4G\001,
len = -1077717836}, lr_val = {s = 0x81c2155
post_ruri_id), len = 12}, r2_val = {s = 0x2 Address 0x2 out of
bounds,
len = 137380436}}
next_hop = {user = {s = 0x9f1a0c0 \001, len = -1077719656},
passwd = {s = 0x65eba1 \203\304\004[]É\366\215\274',
len = 166830272}, host = {s = 0x765c00 DXv, len =
-1077719624}, port = {
s = 0x683588
\213]\364\213u\370\213}\374\211\354]Ã\276|\003, len = 166830272},
params = {s = 0x0, len = -1077719560},
headers = {s = 0xc8817c
[OpenSIPS-Users] SIP-WEB browser Telephony
Maybe this can be useful for OpenSIPs users and their applications: We can build click2talk / webphone application empowering webpages with SIP/Web Telephony using online SIP webphone and opensips For Instance, a web phone link to call: a href= http://widget.doddlephone.com/embed/web3phone.jsp?sipserver=proxy.ideasip.comusername=deglk1password=palindrucallto=1234567890auto=yesstun=stun.ideasip.com Tel: +1 234 567 890 /a It can be also used as a regular webphone (http://widget.doddlephone.com - web driven softphone) Regards Sergio ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
