Re: [OpenSIPS-Users] Critical:core:anchor_lump: offset exceeds message size (1033 1000), abort
Hi Bobby, What version of opensips are you running (also, what the svnrevision - get it with opensips -V). Regards, Bogdan bobby.sm...@gmail.com wrote: And lastly, doing some research on the forums, it seems similar to this issue that's happened lately: http://sourceforge.net/tracker/?func=detailatid=1086410aid=2649267group_id=232389 Again, thanks everyone for their pointers/advice. ___ 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] subscriber
Hi Chamo,
chamo wrote:
Hi,
i have few questions about opensips configuration
1) i use opensips as sip proxy , for decision of route i use
if(uri=~^sip:0XXX@) {
rewritehostport(XX.XXX.162.7:5060);
append_hf(P-hint:
outbound-voice4\r\n);
route(1);
return;
};
it works great but it would be much better to do it from database, i was
looking in modules, but didn't find any usable for this
i think there could be a solution trough perl script, but i don't know
if it will be fast enought
You can use avpops module (avp_db_query) or alias_db or even more
complex drouting or dialplan modules.
2) i need to do LNP (local number portability) where i need to decide to
which trunk will be connection made
i can also do it this way
if(uri=~^sip:0) {
rewritehostport(XX.XXX.XXX.XX:5060);
append_hf(P-hint:
outbound-to-sip-provider\r\n);
route(1);
return;
};
but it can't be prefix driven, because of LNP and are plenty of records
more or less the mechanisms as above.
3) i found in some article (here) problem with Restriction Caller ID,
only i know is that i should use rpid field from subscriber table
,but i didn't find how
in my case is rpid usualy group of numbers
but what you try to achieve ??
4) last problem i have is need to lock subscriber to IP address (or range)
To allow registration from only one IP? nothing simple - add a new
column in subscriber table and provision there the IP address. During
registration (either use av_db_load() or use load_credential if you do
auth) load the IP and check with the src_ip.
Regards,
Bogdan
any help will be great,
thanks, chamo
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Re: [OpenSIPS-Users] Critical:core:anchor_lump: offset exceeds message size (1033 1000), abort
A little more information into the problem. We enabled some additional logging around the point where the anchor_lump function was being called to print out the contents of the SIP message. You can see that after processing this message several times, it craps out and kills all the children processes. If someone could once over the routing script that I included to see, if for example, that I'm trying to correct a nat'ed contact where I shouldn't be, that would be much appreciated. Further, if that's not the case, is there a way where we can gracefully kill one child without killing the whole application? Last -- this is with an emergency update to production to Opensips 1.4.5 last night. This happened this morning around 9. More importantly, Opensips consumed temporarily almost 2 GB of memory before dying, and then cached it. Huge memory spike. Again, tunables: 32 MB private per process, 16 processes total (at this point, we figured less private memory, more processes to work the CPU), 512 Shared memory, on a 4 GB, 2 x Dual Core box. Mar 24 09:09:41 serbox2 /sbin/opensips[2460]: ERROR:registrar:update_contacts: invalid cseq for aor vh13669 Mar 24 09:09:43 serbox2 /sbin/opensips[2468]: CRITICAL:core:anchor_lump: offset exceeds message size (1207 1204 message type 0) aborting... Mar 24 09:09:43 serbox2 /sbin/opensips[2468]: CRITICAL:core:anchor_lump: sip message: INVITE sip:4079641...@my-domain.domain.com:5060 SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK9b74ccab7^M Max-Forwards: 69^M Content-Length: 566^M To: 4079641506 sip:4079641...@my-domain.domain.com:5060^M From: sip:vh11...@my-domain.domain.com:5060;tag=09a4248cffc7eac^M Call-ID: 832e7673f6690afbbda3f438d...@192.168.1.101^m CSeq: 1845720778 INVITE^M Supported: timer^M Allow-Events: talk,hold,conference^M Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO^M Content-Type: application/sdp^M Contact: sip:vh11...@172.16.0.2^m Supported: replaces^M User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26^M ^M v=0^M o=MxSIP 0 2095421055 IN IP4 192.168.1.101^M s=SIP Call^M c=IN IP4 192.168.1.101^M t=0 0^M m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 97 98 2 99 8 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:18 G729/8000^M a=rtpmap:96 BV16/8000^M a=rtpmap:102 BV32/16000^M a=rtpmap:107 L16/16000^M Mar 24 09:09:44 serbox2 /sbin/opensips[2463]: CRITICAL:core:anchor_lump: offset exceeds message size (1207 1204 message type 0) aborting... Mar 24 09:09:48 serbox2 /sbin/opensips[2463]: CRITICAL:core:anchor_lump: sip message: INVITE sip:4079641...@my-domain.domain.com:5060 SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK9b74ccab7^M Max-Forwards: 69^M Content-Length: 566^M To: 4079641506 sip:4079641...@my-domain.domain.com:5060^M From: sip:vh11...@my-domain.domain.com:5060;tag=09a4248cffc7eac^M Call-ID: 832e7673f6690afbbda3f438d...@192.168.1.101^m CSeq: 1845720778 INVITE^M Supported: timer^M Allow-Events: talk,hold,conference^M Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO^M Content-Type: application/sdp^M Contact: sip:vh11...@172.16.0.2^m Supported: replaces^M User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26^M ^M v=0^M o=MxSIP 0 2095421055 IN IP4 192.168.1.101^M s=SIP Call^M c=IN IP4 192.168.1.101^M t=0 0^M m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 97 98 2 99 8 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:18 G729/8000^M a=rtpmap:96 BV16/8000^M a=rtpmap:102 BV32/16000^M a=rtpmap:107 L16/16000^M Mar 24 09:10:12 serbox2 /sbin/opensips[2466]: ERROR:registrar:update_contacts: invalid cseq for aor vh12317 Mar 24 09:10:31 serbox2 /sbin/opensips[2465]: CRITICAL:core:anchor_lump: offset exceeds message size (1205 1204 message type 0) aborting... Mar 24 09:10:31 serbox2 /sbin/opensips[2465]: CRITICAL:core:anchor_lump: sip message: INVITE sip:4079641...@my-domain.domain.com:5060 SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK9b74ccab7^M Max-Forwards: 69^M Content-Length: 566^M To: 4079641506 sip:4079641...@my-domain.domain.com:5060^M From: sip:vh11...@my-domain.domain.com:5060;tag=09a4248cffc7eac^M Call-ID: 832e7673f6690afbbda3f438d...@192.168.1.101^m CSeq: 1845720778 INVITE^M Supported: timer^M Allow-Events: talk,hold,conference^M Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO^M Content-Type: application/sdp^M Contact: sip:vh11...@172.16.0.2^m Supported: replaces^M User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26^M ^M v=0^M o=MxSIP 0 2095421055 IN IP4 192.168.1.101^M s=SIP Call^M c=IN IP4 192.168.1.101^M t=0 0^M m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 97 98 2 99 8 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:18 G729/8000^M a=rtpmap:96 BV16/8000^M a=rtpmap:102 BV32/16000^M a=rtpmap:107 L16/16000^M Mar 24 09:10:31 serbox2 /sbin/opensips[2470]: ERROR:registrar:update_contacts: invalid cseq for aor vh13669 Mar 24
[OpenSIPS-Users] R: DB version error in upgrading to 1.5
Hi Carlo, you need to modify also the table_version value in the version table. Regards, MD Ciao Carlo, c'è una tabella in opensips chiamata version nel tuo caso devi impostare il valore del campo table_version a 5 in corrispondenza della riga table_name=trusted. Buon Lavoro MD -Messaggio originale- Da: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] Per conto di Carlo Dimaggio Inviato: martedì 24 marzo 2009 15:08 A: users@lists.opensips.org Oggetto: [OpenSIPS-Users] DB version error in upgrading to 1.5 Hi all, I have upgraded my server to opensips 1.5 (with the migration of db) but when I restart I have this problem: Mar 24 15:01:29 sip /sbin/opensips[4122]: ERROR:core:db_check_table_version: invalid version 4 for table trusted found, expected 5 Mar 24 15:01:29 sip /sbin/opensips[4122]: ERROR:permissions:init_trusted: error during table version check. Mar 24 15:01:29 sip /sbin/opensips[4122]: ERROR:permissions:mod_init: failed to initialize the allow_trusted function Mar 24 15:01:29 sip /sbin/opensips[4122]: ERROR:core:init_mod: failed to initialize module permissions Mar 24 15:01:29 sip /sbin/opensips[4122]: ERROR:core:main: error while initializing modules I have tried to delete the old db (migrated from 1.4) and to create a new db (with opensipsdbctl create) but I have still the same problem. What is the error? Thank you, Carlo Dimaggio ___ 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] DB version error in upgrading to 1.5
Il giorno 24/mar/09, alle ore 15:22, Bogdan-Andrei Iancu ha scritto: Hi Carlo, to update your database to 1.5 (from 1.4 version) you need to do opensipdbctl migrate - this should solve your problem. Regards, Bogdan Hi Bogdan, I've used the command 'opensipsdbctl migrate' but I have the same problem (the version of trusted table is 4 instead of 5). Do I change manually the number to 5 as Mauro suggested? Thanks, Carlo The output of opensipsdbctl migrate is: # opensipsdbctl migrate opensips opensips_15 MySQL password for root: INFO: Creating new Database opensips_new INFO: test server charset INFO: creating database opensips_new ... INFO: Core OpenSIPS tables succesfully created. Install presence related tables? (y/n): y INFO: creating presence tables into opensips_new ... INFO: Presence tables succesfully created. Install tables for imc cpl siptrace domainpolicy carrierroute userblacklist? (y/n): y INFO: creating extra tables into opensips_new ... INFO: Extra tables succesfully created. INFO: Migrating data from opensips to opensips_new INFO: -- Migrating opensips.acc to opensips_new.acc.OK INFO: -- Migrating opensips.missed_calls to opensips_new.missed_calls.OK INFO: -- Migrating opensips.aliases to opensips_new.aliases.OK INFO: -- Migrating opensips.dbaliases to opensips_new.dbaliases.OK INFO: -- Migrating opensips.grp to opensips_new.grp.OK INFO: -- Migrating opensips.re_grp to opensips_new.re_grp.OK INFO: -- Migrating opensips.silo to opensips_new.silo.OK INFO: -- Migrating opensips.domain to opensips_new.domain.OK INFO: -- Migrating opensips.uri to opensips_new.uri.OK INFO: -- Migrating opensips.usr_preferences to opensips_new.usr_preferences.OK INFO: -- Migrating opensips.trusted to opensips_new.trusted.OK INFO: -- Migrating opensips.address to opensips_new.address.OK INFO: -- Migrating opensips.speed_dial to opensips_new.speed_dial.OK ERROR: failed to migrate opensips.gw to opensips_new.gw!!! Skip it and continue (y/n)? y -- Migrating opensips.gw_grp to opensips_new.gw_grp.SKIPPED (no source) INFO: -- Migrating opensips.lcr to opensips_new.lcr.OK INFO: -- Migrating opensips.pdt to opensips_new.pdt.OK INFO: -- Migrating opensips.subscriber to opensips_new.subscriber.OK INFO: -- Migrating opensips.cpl to opensips_new.cpl.OK -- Migrating opensips.siptrace to opensips_new.siptrace.SKIPPED (no source) INFO: -- Migrating opensips.imc_rooms to opensips_new.imc_rooms.OK -- Migrating opensips.im_members to opensips_new.imc_members.SKIPPED (no source) INFO: Migration successfully completed. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Critical:core:anchor_lump: offset exceeds message size (1033 1000), aborting -- exited by signal 6
And a core dump, looks like the message header is getting an out of bounds
exception or something.
#6 0x0805f9f5 in anchor_lump (msg=0x81a65f8, offset=909, len=0,
type=HDR_OTHER_T) at data_lump.c:346
tmp = value optimized out
prev = value optimized out
t = value optimized out
list = value optimized out
__FUNCTION__ = anchor_lump
#7 0xb7b38f44 in alter_mediaip (msg=0x81a65f8, body=0xbfd11a3c,
oldip=0xbfd11a44, oldpf=2, newip=0xbfd11a4c, newpf=2, preserve=1) at
nathelper.c:1848
anchor = value optimized out
__FUNCTION__ = alter_mediaip
#8 0xb7b44356 in fix_nated_sdp_f (msg=0x81a65f8, str1=value optimized
out, str2=0x0) at nathelper.c:1621
body = {
s = 0x817b332 v=0\r\no=- 1237869463 1237869463 IN IP4
192.168.1.97\r\ns=Polycom IP Phone\r\nc=IN IP4 192.168.1.97\r\nt=0
0\r\na=sendrecv\r\nm=audio 2230 RTP/AVP 0 8 18 101\r\na=rtpmap:0
PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:..., len = 251}
ip = {s = 0x0, len = 0}
anchor = (struct lump *) 0x4
__FUNCTION__ = fix_nated_sdp_f
#9 0x0805333e in do_action (a=0x81a5d58, msg=0x81a65f8) at action.c:846
ret = value optimized out
v = value optimized out
to = value optimized out
p = value optimized out
tmp = value optimized out
end = value optimized out
crt = value optimized out
len = value optimized out
user = value optimized out
uri = {user = {s = 0x81a5a68 \002, len = 135947768}, passwd = {s =
0x0, len = 1}, host = {s = 0x301ab12c Address 0x301ab12c out of bounds,
len = 2}, port = {s = 0x1 Address 0x1 out of bounds,
len = -1212909168}, params = {s = 0x81a6b68 \003, len = 135947992},
headers = {s = 0x1 Address 0x1 out of bounds, len = 0}, port_no = 7312,
proto = 49105, type = 70, transport = {
s = 0xbfd11ca8 \030\037Ñ¿3\005\bøe\032\bØ\016\032\b, len =
-1209763042}, ttl = {s = 0x81a65f8 o®, len = 70}, user_param = {s =
0xbfd11c90 ø²\027\b\002, len = -1212724828}, maddr = {
s = 0x81a4678 \017, len = 135947768}, method = {s = 0xbfd11c90
ø²\027\b\002, len = 135947768}, lr = {s = 0xbfd11d10 \210\032\b\b, len
= 0}, r2 = {s = 0xb7b4ec20 74.167.19.120,
len = -1750208088}, transport_val = {
s = 0x817b2f8 69\r\nContent-Type: application/sdp\r\nContent-Length:
251\r\n\r\nv=0\r\no=- 1237869463 1237869463 IN IP4 192.168.1.97\r\ns=Polycom
IP Phone\r\nc=IN IP4 192.168.1.97\r\nt=0 0\r\na=sendrecv\r\nm=audio 2230
RTP/AVP 0 8 18 ..., len = 2}, ttl_val = {s = 0xbfd11ce8
(\035Ñ¿\237Z\005\bX]\032\bøe\032\b\020\035Ñ¿ è´·\026, len = 135936264},
user_param_val = {s = 0x81a65f8 o®, len = 0},
maddr_val = {s = 0xbfd11f18
X\037Ñ¿\237Z\005\b\b9\032\bøe\032\bË9ìe\200!Ñ¿Ñ°\027\b\005, len =
134558526}, method_val = {s = 0x81a65f8 o®, len = 135925464}, lr_val = {s
= 0x0, len = 0}, r2_val = {
s = 0x0, len = 0}}
next_hop = {user = {s = 0xbfd11bc8 è\034Ñ¿pɺ·, len = 0}, passwd =
{s = 0x819bc60 , len = 135925576}, host = {s = 0x81a1078 \r\n=on=5068,
len = 135925816}, port = {
s = 0xe Address 0xe out of bounds, len = 19}, params = {s = 0x816acc1
_out, len = 0}, headers = {s = 0x7 Address 0x7 out of bounds, len = 1},
port_no = 29768, proto = 2074, type = 134961223,
transport = {s = 0xbfd11bd0 , len = 135952121}, ttl = {
s = 0x817b2fc Content-Type: application/sdp\r\nContent-Length:
251\r\n\r\nv=0\r\no=- 1237869463 1237869463 IN IP4 192.168.1.97\r\ns=Polycom
IP Phone\r\nc=IN IP4 192.168.1.97\r\nt=0 0\r\na=sendrecv\r\nm=audio 2230
RTP/AVP 0 8 18 101\r..., len = -1212476996}, user_param = {s = 0x0, len =
135950312}, maddr = {s = 0xbfd11ce8
(\035Ñ¿\237Z\005\bX]\032\bøe\032\b\020\035Ñ¿ è´·\026, len = -1212495504},
method = {s = 0x0, len = -1076814576}, lr = {s = 0x1 Address 0x1 out of
bounds, len = 0}, r2 = {s = 0xbfd11c90 ø²\027\b\002, len = 0},
transport_val = {s = 0xbfd11c58 ¨\034Ñ¿\036{ä·øe\032\bF,
len = -1209757902}, ttl_val = {s = 0x81a65f8 o®, len = 256},
user_param_val = {s = 0x0, len = 0}, maddr_val = {s = 0xbfd11d14 \b, len =
-1076814736}, method_val = {s = 0x0, len = 135944856},
lr_val = {s = 0x81a65f8 o®, len = 4}, r2_val = {s = 0xbfd11d28
\230\037Ñ¿ûF\005\bÈZ\032\bøe\032\b, len = 134891743}}
u = value optimized out
port = value optimized out
cmatch = value optimized out
aitem = value optimized out
adefault = value optimized out
spec = value optimized out
---Type return to continue, or q return to quit---
model = value optimized out
val = {rs = {s = 0x0, len = -1212460488}, ri = 135947768, flags = 0}
__FUNCTION__ = do_action
#10 0x08055a9f in run_action_list (a=0x81a5ac8, msg=0x81a65f8) at
action.c:138
ret = 1
t = (struct action *) 0x81a5d58
__FUNCTION__ = run_action_list
#11 0x080546fb in do_action (a=0x81a5ea8, msg=0x81a65f8) at action.c:718
ret = value optimized out
v = 1
to = value optimized out
p = value optimized out
tmp = value
Re: [OpenSIPS-Users] serialize_branches/next_branches problem
Hi Bogdan, Debug level was 6 for get_redirects(*), serialize_branches(1) and next_branches(). The contact header from the 302 was as follows: Contact:sip:+1303...@ww.xx.116.46:5060;user=phone;q=0.5,sip:+1303 0...@ww.xx.119.46:5060;user=phone;q=0.25 Debug output: DBG:uac_redirect:get_redirect: resume branch=0 DBG:uac_redirect:get_redirect: checking branch=0 (added=0) DBG:uac_redirect:get_redirect: branch=0 is a redirect (added=0) DBG:core:parse_headers: flags=7 DBG:core:get_hdr_field: content_length=0 DBG:core:get_hdr_field: found end of header DBG:uac_redirect:sort_contacts: sort_contacts: sip:+1303...@ww.xx.119.46:5060;user=phone q=250 DBG:uac_redirect:sort_contacts: sort_contacts: sip:+1303...@ww.xx.116.46:5060;user=phone q=500 DBG:uac_redirect:shmcontact2dset: adding contact sip:+1303...@ww.xx.119.46:5060;user=phone DBG:uac_redirect:shmcontact2dset: adding contact sip:+1303...@ww.xx.116.46:5060;user=phone DBG:core:serialize_branches: loaded sip:+1303...@ww.xx.119.46:5060;user=phone, q=-1 q_flag 0 DBG:core:serialize_branches: loaded sip:+1303...@ww.xx.116.46:5060;user=phone, q=500 q_flag 16 DBG:core:next_branches: branch is sip:+1303...@ww.xx.116.46:5060;user=phone The Opensips build is from an SVN checkout of branches/1.5 about 15:00 GMT today. - Jeff On 3/23/09 10:38 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Jeff, please post the debug=6 logs - also be sure you are using the latest version as a similar bug was fixed one or two weeks ago. Regards, Bogdan Jeff Pyle wrote: Hello, I catch a 302 in a failure_route that runs: get_redirects(³*²), serialize_branches and next_branches. The subsequent t_relay() causes a parallel fork to both contacts in the 302¹s Contact header. The 302¹s Contact header looks like this: Contact:sip:+1303...@qq.rr.ss.tt:5060;user=phone;q=0.5,sip:+130300 0...@qq.rr.ww.tt:5060;user=phone;q=0.25 I would expect it to load only the q=0.5 route at first, no? - Jeff ___ 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] [OpenSIPS-Devel] mi_xmlrpc module with opensips 1.4.2-notls (x86_64/linux) undefined symbol: xmlrpc_array_size
Hi Ollie, be sure you do proper clean (do make proper) before recompiling - all version 10310 - 10630 should work. maybe you just have mixed some objects from the prev compiling. Regards, Bogdan ollie eillo wrote: Hi, /Centos 5.2 x86_64/ /opensips 1.4.2-notls/ I have opensips compiled and running successfully on Centos 5.2 without the mi_xmplrpc module. I followed this method: http://www.smartvox.co.uk/serfaq_install_opensips142.htm I am trying to enable mi_xmlrpc so that openxcap can work with opensips. The key problem arises with the dependency: xmlrpc-c. The documentation says that version 0.9.10 should be used but this is a very old version and I cannot find a compiled 64bit version. I have tried compiling the source version of 0.9.10 but .\configure fails because it has not been tested with 64bit and I don't know how to patch it! In the end I tried installing newer tarball versions: *xmlrpc-c-1.06.18-1.el5.kb.x86_64.rpm xmlrpc-c-devel-1.06.18-1.el5.kb.x86_64.rpm* Success, Opensips then copiled with no errors. *However, it no longer starts.* _opensips.log_ Mar 24 10:09:43 opensips: ERROR:core:sr_load_module: could not open module /usr/local/lib64/opensips/modules/mi_xmlrpc.so: */usr/local/lib64/opensips/modules/mi_xmlrpc.so: undefined symbol: xmlrpc_array_size *Mar 24 10:09:43 opensips: CRITICAL:core:yyerror: parse error in config file, line 95, column 13-14: failed to load module Mar 24 10:09:43 opensips: ERROR:core:set_mod_param_regex: no module matching mi_xmlrpc found | Mar 24 10:09:43 opensips: CRITICAL:core:yyerror: parse error in config file, line 153, column 21-22: Can't set module parameter Mar 24 10:09:43 opensips: ERROR:core:main: bad config file (2 errors) Mar 24 10:09:43 opensips: NOTICE:presence:destroy: destroy module ... _opensips.cfg_ loadmodule xcap_client.so loadmodule presence.so loadmodule presence_xml.so loadmodule mi_xmlrpc.so modparam(presence, server_address, sip:s...@private_nic_ip:5060 ) modparam(presence, clean_period, 100) modparam(presence, max_expires, 3600) modparam(presence_xml, integrated_xcap_server, 1) modparam(presence_xml, pidf_manipulation, 1) modparam(presence_xml, force_active, 0) modparam(presence_xml, xcap_table, xcap) modparam(mi_xmlrpc,port, 8080) _find /usr/include/ | grep -i xmlrpc_ /usr/include/xmlrpc_server_w32httpsys.h /usr/include/xmlrpc_client.h /usr/include/xmlrpc_server_abyss.h /usr/include/xmlrpc_cgi.h /usr/include/xmlrpc.h /usr/include/XmlRpcCpp.h /usr/include/xmlrpc_server.h /usr/include/xmlrpc-c /usr/include/xmlrpc-c/girerr.hpp /usr/include/xmlrpc-c/server_abyss.h /usr/include/xmlrpc-c/client.hpp /usr/include/xmlrpc-c/server_w32httpsys.h /usr/include/xmlrpc-c/server_abyss.hpp /usr/include/xmlrpc-c/server_cgi.h /usr/include/xmlrpc-c/girmem.hpp /usr/include/xmlrpc-c/xml.hpp /usr/include/xmlrpc-c/client.h /usr/include/xmlrpc-c/registry.hpp /usr/include/xmlrpc-c/transport.h /usr/include/xmlrpc-c/oldxmlrpc.h /usr/include/xmlrpc-c/base.hpp /usr/include/xmlrpc-c/timeout.hpp /usr/include/xmlrpc-c/client_transport.hpp /usr/include/xmlrpc-c/util.h /usr/include/xmlrpc-c/client_global.h /usr/include/xmlrpc-c/client_simple.hpp /usr/includ e/xmlrpc-c/base.h /usr/include/xmlrpc-c/oldcppwrapper.hpp /usr/include/xmlrpc-c/config.h /usr/include/xmlrpc-c/server.h /usr/include/xmlrpc-c/abyss.h _find /usr/local | grep -i mi_xmlrpc_ /usr/local/lib64/opensips/modules/mi_xmlrpc.so Please let me know how to work around this. Perhaps a way of compiling 0.9.10 with 64bit. Thank you very much in advance, Ollie ___ Devel mailing list de...@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/devel ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] DB version error in upgrading to 1.5
I just checked and creating a new DB will set a 5 version for the trusted table. Are you sure you are not mixing somehow the old and new DB ? Regards, Bogdan Carlo Dimaggio wrote: Il giorno 24/mar/09, alle ore 15:22, Bogdan-Andrei Iancu ha scritto: Hi Carlo, to update your database to 1.5 (from 1.4 version) you need to do opensipdbctl migrate - this should solve your problem. Regards, Bogdan Hi Bogdan, I've used the command 'opensipsdbctl migrate' but I have the same problem (the version of trusted table is 4 instead of 5). Do I change manually the number to 5 as Mauro suggested? Thanks, Carlo The output of opensipsdbctl migrate is: # opensipsdbctl migrate opensips opensips_15 MySQL password for root: INFO: Creating new Database opensips_new INFO: test server charset INFO: creating database opensips_new ... INFO: Core OpenSIPS tables succesfully created. Install presence related tables? (y/n): y INFO: creating presence tables into opensips_new ... INFO: Presence tables succesfully created. Install tables for imc cpl siptrace domainpolicy carrierroute userblacklist? (y/n): y INFO: creating extra tables into opensips_new ... INFO: Extra tables succesfully created. INFO: Migrating data from opensips to opensips_new INFO: -- Migrating opensips.acc to opensips_new.acc.OK INFO: -- Migrating opensips.missed_calls to opensips_new.missed_calls.OK INFO: -- Migrating opensips.aliases to opensips_new.aliases.OK INFO: -- Migrating opensips.dbaliases to opensips_new.dbaliases.OK INFO: -- Migrating opensips.grp to opensips_new.grp.OK INFO: -- Migrating opensips.re_grp to opensips_new.re_grp.OK INFO: -- Migrating opensips.silo to opensips_new.silo.OK INFO: -- Migrating opensips.domain to opensips_new.domain.OK INFO: -- Migrating opensips.uri to opensips_new.uri.OK INFO: -- Migrating opensips.usr_preferences to opensips_new.usr_preferences.OK INFO: -- Migrating opensips.trusted to opensips_new.trusted.OK INFO: -- Migrating opensips.address to opensips_new.address.OK INFO: -- Migrating opensips.speed_dial to opensips_new.speed_dial.OK ERROR: failed to migrate opensips.gw to opensips_new.gw!!! Skip it and continue (y/n)? y -- Migrating opensips.gw_grp to opensips_new.gw_grp.SKIPPED (no source) INFO: -- Migrating opensips.lcr to opensips_new.lcr.OK INFO: -- Migrating opensips.pdt to opensips_new.pdt.OK INFO: -- Migrating opensips.subscriber to opensips_new.subscriber.OK INFO: -- Migrating opensips.cpl to opensips_new.cpl.OK -- Migrating opensips.siptrace to opensips_new.siptrace.SKIPPED (no source) INFO: -- Migrating opensips.imc_rooms to opensips_new.imc_rooms.OK -- Migrating opensips.im_members to opensips_new.imc_members.SKIPPED (no source) INFO: Migration successfully completed. ___ 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] Critical:core:anchor_lump: offset exceeds message size (1033 1000), abort
Hi Bobby, can you get the logs with debug=6 and post (or send priv to me) ? Also, what is the rev number your are using? Regards, Bogdan bobby.sm...@gmail.com wrote: A little more information into the problem. We enabled some additional logging around the point where the anchor_lump function was being called to print out the contents of the SIP message. You can see that after processing this message several times, it craps out and kills all the children processes. If someone could once over the routing script that I included to see, if for example, that I'm trying to correct a nat'ed contact where I shouldn't be, that would be much appreciated. Further, if that's not the case, is there a way where we can gracefully kill one child without killing the whole application? Last -- this is with an emergency update to production to Opensips 1.4.5 last night. This happened this morning around 9. More importantly, Opensips consumed temporarily almost 2 GB of memory before dying, and then cached it. Huge memory spike. Again, tunables: 32 MB private per process, 16 processes total (at this point, we figured less private memory, more processes to work the CPU), 512 Shared memory, on a 4 GB, 2 x Dual Core box. Mar 24 09:09:41 serbox2 /sbin/opensips[2460]: ERROR:registrar:update_contacts: invalid cseq for aor vh13669 Mar 24 09:09:43 serbox2 /sbin/opensips[2468]: CRITICAL:core:anchor_lump: offset exceeds message size (1207 1204 message type 0) aborting... Mar 24 09:09:43 serbox2 /sbin/opensips[2468]: CRITICAL:core:anchor_lump: sip message: INVITE sip:4079641...@my-domain.domain.com:5060 SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK9b74ccab7^M Max-Forwards: 69^M Content-Length: 566^M To: 4079641506 sip:4079641...@my-domain.domain.com:5060^M From: sip:vh11...@my-domain.domain.com:5060;tag=09a4248cffc7eac^M Call-ID: 832e7673f6690afbbda3f438d...@192.168.1.101^m CSeq: 1845720778 INVITE^M Supported: timer^M Allow-Events: talk,hold,conference^M Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO^M Content-Type: application/sdp^M Contact: sip:vh11...@172.16.0.2^m Supported: replaces^M User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26^M ^M v=0^M o=MxSIP 0 2095421055 IN IP4 192.168.1.101^M s=SIP Call^M c=IN IP4 192.168.1.101^M t=0 0^M m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 97 98 2 99 8 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:18 G729/8000^M a=rtpmap:96 BV16/8000^M a=rtpmap:102 BV32/16000^M a=rtpmap:107 L16/16000^M Mar 24 09:09:44 serbox2 /sbin/opensips[2463]: CRITICAL:core:anchor_lump: offset exceeds message size (1207 1204 message type 0) aborting... Mar 24 09:09:48 serbox2 /sbin/opensips[2463]: CRITICAL:core:anchor_lump: sip message: INVITE sip:4079641...@my-domain.domain.com:5060 SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK9b74ccab7^M Max-Forwards: 69^M Content-Length: 566^M To: 4079641506 sip:4079641...@my-domain.domain.com:5060^M From: sip:vh11...@my-domain.domain.com:5060;tag=09a4248cffc7eac^M Call-ID: 832e7673f6690afbbda3f438d...@192.168.1.101^m CSeq: 1845720778 INVITE^M Supported: timer^M Allow-Events: talk,hold,conference^M Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO^M Content-Type: application/sdp^M Contact: sip:vh11...@172.16.0.2^m Supported: replaces^M User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26^M ^M v=0^M o=MxSIP 0 2095421055 IN IP4 192.168.1.101^M s=SIP Call^M c=IN IP4 192.168.1.101^M t=0 0^M m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 97 98 2 99 8 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:18 G729/8000^M a=rtpmap:96 BV16/8000^M a=rtpmap:102 BV32/16000^M a=rtpmap:107 L16/16000^M Mar 24 09:10:12 serbox2 /sbin/opensips[2466]: ERROR:registrar:update_contacts: invalid cseq for aor vh12317 Mar 24 09:10:31 serbox2 /sbin/opensips[2465]: CRITICAL:core:anchor_lump: offset exceeds message size (1205 1204 message type 0) aborting... Mar 24 09:10:31 serbox2 /sbin/opensips[2465]: CRITICAL:core:anchor_lump: sip message: INVITE sip:4079641...@my-domain.domain.com:5060 SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK9b74ccab7^M Max-Forwards: 69^M Content-Length: 566^M To: 4079641506 sip:4079641...@my-domain.domain.com:5060^M From: sip:vh11...@my-domain.domain.com:5060;tag=09a4248cffc7eac^M Call-ID: 832e7673f6690afbbda3f438d...@192.168.1.101^m CSeq: 1845720778 INVITE^M Supported: timer^M Allow-Events: talk,hold,conference^M Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO^M Content-Type: application/sdp^M Contact: sip:vh11...@172.16.0.2^m Supported: replaces^M User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26^M ^M v=0^M o=MxSIP 0 2095421055 IN IP4 192.168.1.101^M s=SIP Call^M c=IN IP4 192.168.1.101^M t=0 0^M m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 97 98 2 99 8 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:18 G729/8000^M
Re: [OpenSIPS-Users] Critical:core:anchor_lump: offset exceeds message size (1033 1000), abort
Opensips 1.4.5 -- the latest release as of yesterday (non-1.5). Working on getting you your trace now, will take a few minutes. Much thanks. 2009/3/24 Bogdan-Andrei Iancu bog...@voice-system.ro Hi Bobby, can you get the logs with debug=6 and post (or send priv to me) ? Also, what is the rev number your are using? Regards, Bogdan bobby.sm...@gmail.com wrote: A little more information into the problem. We enabled some additional logging around the point where the anchor_lump function was being called to print out the contents of the SIP message. You can see that after processing this message several times, it craps out and kills all the children processes. If someone could once over the routing script that I included to see, if for example, that I'm trying to correct a nat'ed contact where I shouldn't be, that would be much appreciated. Further, if that's not the case, is there a way where we can gracefully kill one child without killing the whole application? Last -- this is with an emergency update to production to Opensips 1.4.5 last night. This happened this morning around 9. More importantly, Opensips consumed temporarily almost 2 GB of memory before dying, and then cached it. Huge memory spike. Again, tunables: 32 MB private per process, 16 processes total (at this point, we figured less private memory, more processes to work the CPU), 512 Shared memory, on a 4 GB, 2 x Dual Core box. Mar 24 09:09:41 serbox2 /sbin/opensips[2460]: ERROR:registrar:update_contacts: invalid cseq for aor vh13669 Mar 24 09:09:43 serbox2 /sbin/opensips[2468]: CRITICAL:core:anchor_lump: offset exceeds message size (1207 1204 message type 0) aborting... Mar 24 09:09:43 serbox2 /sbin/opensips[2468]: CRITICAL:core:anchor_lump: sip message: INVITE sip:4079641...@my-domain.domain.com:5060 SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK9b74ccab7^M Max-Forwards: 69^M Content-Length: 566^M To: 4079641506 sip:4079641...@my-domain.domain.com:5060^M From: sip:vh11...@my-domain.domain.com:5060;tag=09a4248cffc7eac^M Call-ID: 832e7673f6690afbbda3f438d...@192.168.1.101^m CSeq: 1845720778 INVITE^M Supported: timer^M Allow-Events: talk,hold,conference^M Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO^M Content-Type: application/sdp^M Contact: sip:vh11...@172.16.0.2sip%3avh11...@172.16.0.2^M Supported: replaces^M User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/ 1.5.1.0 MxSF/v3.2.6.26^M ^M v=0^M o=MxSIP 0 2095421055 IN IP4 192.168.1.101^M s=SIP Call^M c=IN IP4 192.168.1.101^M t=0 0^M m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 97 98 2 99 8 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:18 G729/8000^M a=rtpmap:96 BV16/8000^M a=rtpmap:102 BV32/16000^M a=rtpmap:107 L16/16000^M Mar 24 09:09:44 serbox2 /sbin/opensips[2463]: CRITICAL:core:anchor_lump: offset exceeds message size (1207 1204 message type 0) aborting... Mar 24 09:09:48 serbox2 /sbin/opensips[2463]: CRITICAL:core:anchor_lump: sip message: INVITE sip:4079641...@my-domain.domain.com:5060 SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK9b74ccab7^M Max-Forwards: 69^M Content-Length: 566^M To: 4079641506 sip:4079641...@my-domain.domain.com:5060^M From: sip:vh11...@my-domain.domain.com:5060;tag=09a4248cffc7eac^M Call-ID: 832e7673f6690afbbda3f438d...@192.168.1.101^m CSeq: 1845720778 INVITE^M Supported: timer^M Allow-Events: talk,hold,conference^M Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO^M Content-Type: application/sdp^M Contact: sip:vh11...@172.16.0.2sip%3avh11...@172.16.0.2^M Supported: replaces^M User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/ 1.5.1.0 MxSF/v3.2.6.26^M ^M v=0^M o=MxSIP 0 2095421055 IN IP4 192.168.1.101^M s=SIP Call^M c=IN IP4 192.168.1.101^M t=0 0^M m=audio 3000 RTP/AVP 0 18 96 102 107 104 105 106 97 98 2 99 8 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:18 G729/8000^M a=rtpmap:96 BV16/8000^M a=rtpmap:102 BV32/16000^M a=rtpmap:107 L16/16000^M Mar 24 09:10:12 serbox2 /sbin/opensips[2466]: ERROR:registrar:update_contacts: invalid cseq for aor vh12317 Mar 24 09:10:31 serbox2 /sbin/opensips[2465]: CRITICAL:core:anchor_lump: offset exceeds message size (1205 1204 message type 0) aborting... Mar 24 09:10:31 serbox2 /sbin/opensips[2465]: CRITICAL:core:anchor_lump: sip message: INVITE sip:4079641...@my-domain.domain.com:5060 SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK9b74ccab7^M Max-Forwards: 69^M Content-Length: 566^M To: 4079641506 sip:4079641...@my-domain.domain.com:5060^M From: sip:vh11...@my-domain.domain.com:5060;tag=09a4248cffc7eac^M Call-ID: 832e7673f6690afbbda3f438d...@192.168.1.101^m CSeq: 1845720778 INVITE^M Supported: timer^M Allow-Events: talk,hold,conference^M Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO^M Content-Type: application/sdp^M Contact: sip:vh11...@172.16.0.2sip%3avh11...@172.16.0.2^M Supported: replaces^M User-Agent: Aastra 480i/1.4.1.2000 Brcm Callctrl/ 1.5.1.0 MxSF/v3.2.6.26^M ^M v=0^M
Re: [OpenSIPS-Users] DB version error in upgrading to 1.5
Il giorno 24/mar/09, alle ore 16:15, Bogdan-Andrei Iancu ha scritto: I just checked and creating a new DB will set a 5 version for the trusted table. Are you sure you are not mixing somehow the old and new DB ? I don't think, but I'll investigate better... For now, I have set manually the version table. Thank you, Carlo ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] DB version error in upgrading to 1.5
And only the trusted table was affected ? the rest of the tables are ok? Thanks and regards, Bogdan Carlo Dimaggio wrote: Il giorno 24/mar/09, alle ore 16:15, Bogdan-Andrei Iancu ha scritto: I just checked and creating a new DB will set a 5 version for the trusted table. Are you sure you are not mixing somehow the old and new DB ? I don't think, but I'll investigate better... For now, I have set manually the version table. Thank you, Carlo ___ 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] DB version error in upgrading to 1.5
Il giorno 24/mar/09, alle ore 17:05, Bogdan-Andrei Iancu ha scritto: And only the trusted table was affected ? the rest of the tables are ok? Hi Bogdan, At the end, I have modified trusted, gw, address, lcr, subscriber, dialog, grp, location, userblacklist tables. It seems a little bit strange, but for now all is fine. Best regards, Carlo ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] loose_route: loop on ACK requests
Hi Bogdan,
You are right. I had the same issue as Jeff. The IP was in the domain table :(.
Thanks again,
Noel
On Mon, Mar 23, 2009 at 12:03 PM, Bogdan-Andrei Iancu
bog...@voice-system.ro wrote:
Hi Noel,
maybe you have the same issue as Jeff - maybe you have the 92.168.193.20 IP
in domain table, as local domain.
Regards,
Bogdan
Noel R. Morais wrote:
Hi Bogdan,
I'm not using alias.
I've realized that if I request authorization for the initial INVITE,
everything works like a charm.
If I just by pass (using allow_trusted) it doesn't work and opensips
route the ACK to itself.
If you don't mind follow attached my cfg. It's simple.
Thanks,
Noel
On Fri, Mar 20, 2009 at 2:54 PM, Bogdan-Andrei Iancu
bog...@voice-system.ro wrote:
Hi Noel,
it looks like OpenSIPS is doing strict routing on the received ACK.
This happens if it finds out in RURI an IP/address which is considered
local
- in the case the RURI will be consumed and use the Route as new RURI...
So are you sure there is no misconfiguration in the alias params ?
Regards,
Bogdan
Noel R. Morais wrote:
Hi guys,
I'm having problems about loose_route(). Opensips is routing ACK
requests to itself.
I know that posting code and traces are ugly, but I think I do not
have choices. Sorry.
Follow bellow the code regarding loose_route:
if (has_totag()) {
if (loose_route()) {
if(method==INVITE) {
route(5); #Check authentication of
re-invites
}
route(1);
} else {
if ( is_method(ACK) ) {
if ( t_check_trans() ) {
route(1);
}
}
}
}
Follow bellow the trace, 192.168.191.188 is the opensips ip address:
U 2009/03/11 14:46:53.950565 192.168.191.188:5060 -
192.168.192.233:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.192.233;branch=z9hG4bKac74079177.
Contact: sip:xx...@192.168.193.20:5060.
Record-Route:
sip:192.168.191.188;lr=on;ftag=1c74077990;did=f9e.d32b00d2.
Call-ID: 74077637112200051...@192.168.192.233.
From: Jeff002 sip:yyy...@192.168.193.20;tag=1c74077990.
To:
sip:xx...@192.168.191.188;user=phone;tag=a94c095b773be1dd6e8d668a785a9c8469ec.
CSeq: 1 INVITE.
Server: Cantata-SIP/10.3.2.51932 IMG 0.
Allow: INVITE, BYE, REGISTER, ACK, OPTIONS, CANCEL, INFO.
Supported: path.
Accept: application/sdp.
Content-Type: application/sdp.
Content-Length: 236.
.
v=0.
o=Cantata_SDP 0 1 IN IP4 192.168.193.20.
s=Cantata-SIP.
c=IN IP4 192.168.193.21.
t=0 0.
m=audio 8944 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=silenceSupp:off - - - -.
a=ptime:20.
U 2009/03/11 14:46:53.997019 192.168.192.233:5060 -
192.168.191.188:5060
ACK sip:xx...@192.168.193.20:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.192.233;branch=z9hG4bKac82192814.
Max-Forwards: 70.
From: Jeff002 sip:yyy...@192.168.193.20;tag=1c74077990.
To:
sip:xx...@192.168.191.188;user=phone;tag=a94c095b773be1dd6e8d668a785a9c8469ec.
Call-ID: 74077637112200051...@192.168.192.233.
CSeq: 1 ACK.
Contact: sip:yyy...@192.168.192.233.
Route: sip:192.168.191.188;lr=on;ftag=1c74077990;did=f9e.d32b00d2.
Supported: em,timer,replaces,path.
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE.
User-Agent: Audiocodes-Sip-Gateway-MP-102 FXS/v.4.60A.035.001.
Content-Length: 0.
.
U 2009/03/11 14:46:53.998178 192.168.191.188:5060 -
192.168.191.188:5060
ACK sip:192.168.191.188;lr=on;ftag=1c74077990;did=f9e.d32b00d2 SIP/2.0.
Record-Route: sip:192.168.191.188;lr=on;ftag=1c74077990.
Via: SIP/2.0/UDP 192.168.191.188;branch=z9hG4bKef17.5d8b81f4.2.
Via: SIP/2.0/UDP 192.168.192.233;branch=z9hG4bKac82192814.
Max-Forwards: 69.
From: Jeff002 sip:yyy...@192.168.193.20;tag=1c74077990.
To:
sip:xx...@192.168.191.188;user=phone;tag=a94c095b773be1dd6e8d668a785a9c8469ec.
Call-ID: 74077637112200051...@192.168.192.233.
CSeq: 1 ACK.
Supported: em,timer,replaces,path.
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE.
User-Agent: Audiocodes-Sip-Gateway-MP-102 FXS/v.4.60A.035.001.
Content-Length: 0.
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[OpenSIPS-Users] prefix to account
Hey All,I was just wanting to get some feedback from the community on how you may handle this. I have a number of clients who like to use prefixes in the dialed number coupled with IP address authentication to link calls to a specific account.. It starts out simple.. Customer A sends me calls from 1.2.3.4.. Great. I have a table that links IP to account.. So now I can account those calls.. But now customer A, has subcustomer A.1, or A.2 They still send calls from 1.2.3.4, but they'll send prefix 001234 before the dialed number (like 00123415125551212). In this case, I want to identify the 1.2.3.4 + prefix of 001234 as being customer A.1, then strip off 001234. So in general, I do an avp_db_query (to be replaced by a cache_fetch) for $si + substr($rU)... Which works fine.. BUT if the prefix is not of a fixed length.. I'm not even really sure hwo to go about it.. (pardon the messy sql, it's really just to prove a point) with the avp_db_query, I can simply do a like select ala: select account from customertrunks where ip=$si and to_did like concat($rU,'%') But if I do a cache_fetch, I can't do the pattern match.. So how do you guys do this? or do you do it at all. :) I see a lot of clients asking for some sort of call prefixes.. usually a fixed length will make them happy, but I've got some now that don't have a fixed length. Thanks, Brett ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] acc log 100 Trying
Hey All,I'm trying to log the 100 Trying in the acc module. I tried the
acc_db_request, but it can't be called from onreply.
So I tried editing acc_logic.c, but that isn't working either?! :/ Any
ideas? Maybe I should edit acc_db_request so it can be called from ONREPLY?
static inline int should_acc_reply(struct sip_msg *req,struct sip_msg *rpl,
int code)
{
/* negative transactions reported otherwise only if explicitly
* demanded */
if (code == 100)
return 1;
if ( !is_failed_acc_on(req) code =300 )
return 0;
if ( !is_acc_on(req) )
return 0;
if ( code200 !(early_media
parse_headers(rpl,HDR_CONTENTLENGTH_F, 0)==0 rpl-content_length
get_content_length(rpl)0 ) )
return 0;
return 1; /* seed is through, we will account this reply */
}
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[OpenSIPS-Users] CDRTool 6.7.4, lost Rating Info
Hello, I upgraded from CDRTool 6.7.2 to 6.7.4 and I lost the Rating Information on the CDR display page. I use CDRTool with rating disabled. I manually populate the Rate field of the radius records from Opensips from an AVP that contains the rate (our cost) of the call depending on a bunch of different factors. The value of the Rate field used to show up under the Rating Information under 6.7.2. Under 6.7.4, nothing. Did something change that might affect this? If so, is there a way I can get it to display once again? Thanks, Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] upgrade to 1.5 - module versions are mixed
after upgrading to 1.5, i now get this error when running opensips start r...@sfsip:/etc/init.d# opensips start Mar 24 18:24:53 [4389] DBG:core:yyparse: loading module //lib/opensips/modules/signaling.so Mar 24 18:24:53 [4389] ERROR:core:version_control: module version mismatch for //lib/opensips/modules/signaling.so; core: opensips 1.4.4-notls (i386/linux); module: opensips 1.5.0-notls (i386/linux) r...@sfsip:/etc/init.d# I can tell that it is still starting the 1.4 opensips, but I cannot figure out why/where it is and how to force 1.5 to start. To install I ran make prefix=/ all make prefix=/ install Or as an alternative to this question, can anyone tell me how to uninstall 1.5 so that i can go back? I may have jumped the gun. -- View this message in context: http://n2.nabble.com/upgrade-to-1.5---module-versions-are-mixed-tp2530005p2530005.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
