[OpenSIPS-Users] permissions module: subnet_table_insert: subnet table is full
Hello:
I'm Using Opensips-1.6.4 with permissions module to classify clients in
different groups depending of the source IP. Everything works fine but
today, after inserting a new subnet in table address and doing opensipsctl
address reload i saw in the log the message:
CRITICAL:permissions:subnet_table_insert: subnet table is full
In address table I have 178 hosts (mask 32) and 129 networks (mask != 32)
Doing opensipsctl fifo address_dump y saw 178 host entries (all OK), but
doing the command opensipsctl fifo subnet_dump shows only 128 entries.
Checking the sources I saw the filter in hash.c
count = table[PERM_MAX_SUBNETS].grp;
if (count == PERM_MAX_SUBNETS) {
LM_CRIT(subnet table is full\n);
return 0;
}
and the value of PERM_MAX_SUBNETS in hash.h is 128
I didn't found anything about this limit in module Readme.
Looking at sources of 1.8.3 seems to have the same limit (#define
PERM_MAX_SUBNETS 128 in hash.h)
My questions are:
¿Is this networks limit real? ¿Can I increase it? ¿how?
Thanks
Carlos Oliva
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Re: [OpenSIPS-Users] memory consumed by t_relay
Well, this indicate that you have a bug in your test call flows - the BYE ending the calls should also end the dialogs - there is no need for them to expire. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/23/2013 06:46 AM, microx wrote: Hi Bogdan-Andrei, Thanks so much for your help. With modparam(dialog, default_timeout, n), the dialogs are removed and the memory consumption issue is resolved. Best wishes, Chen-Che -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/memory-consumed-by-t-relay-tp7586016p7586470.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] permissions module: subnet_table_insert: subnet table is full
On May 23, 2013, at 10:24 AM, Carlos Oliva wrote:
Hello:
I'm Using Opensips-1.6.4 with permissions module to classify clients in
different groups depending of the source IP. Everything works fine but today,
after inserting a new subnet in table address and doing opensipsctl address
reload i saw in the log the message:
CRITICAL:permissions:subnet_table_insert: subnet table is full
In address table I have 178 hosts (mask 32) and 129 networks (mask != 32)
Doing opensipsctl fifo address_dump y saw 178 host entries (all OK), but
doing the command opensipsctl fifo subnet_dump shows only 128 entries.
Checking the sources I saw the filter in hash.c
count = table[PERM_MAX_SUBNETS].grp;
if (count == PERM_MAX_SUBNETS) {
LM_CRIT(subnet table is full\n);
return 0;
}
and the value of PERM_MAX_SUBNETS in hash.h is 128
I didn't found anything about this limit in module Readme.
Looking at sources of 1.8.3 seems to have the same limit (#define
PERM_MAX_SUBNETS 128 in hash.h)
My questions are:
¿Is this networks limit real? ¿Can I increase it? ¿how?
I feel your pain, I hate compile time limitations with a passion. Right now the
only way around is to modify that define to a higher value and recompile. FWIW
I increased it to an absurdly high value (1 IIRC) and that did the job.
--
Saúl Ibarra Corretgé
AG Projects
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Re: [OpenSIPS-Users] Use_next-$avp(gw_id) parameter
Hi Khaled, Check the return code of the use_next_gw() - if there is no other GW to be used (all used), the function will return false and, of course, no gw_id or rule_attrs will be available. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/22/2013 09:41 PM, M.Khaled W Chehab wrote: Dears I am using opensips 1.8.3 with do_routing module From documentation: The name of the avp for storing the id of the current selected gateway/destination - once a new destination is selected (via the use_next_gw() function), the AVP will be updated with the ID of the new selected gateway/destination. xlog(after issue 2 gw: $avp(gw_id) ... and attrs : $avp(rule_attrs) --); use_next_gw(); xlog(before issue 2 gw: $avp(gw_id) .. and attrs : $avp(rule_attrs) --); debug after issue 2 gw: Snell_93 ... and attrs : 1 -- before issue 2 gw: null .. and attrs : null -- How to force do_routing function to store the : $avp(gw_id)after use_next_gw(); Regards ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Registrar not saving received from Path header
Hello Nathaniel,
My test was just to see if your opensips hits the fixup function for the
flags - and it seems it does (the value I gave you is bogus, so it ended
up with error).
Could you try this new patch (remove old one) ? (of course, put back the
p0 param ).
Thanks and regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05/23/2013 01:43 AM, Nathaniel L Keeling III wrote:
Hello Bogdan,
I am using opensips v1.8.3. I was using v1.8.2 earlier but I upgraded
thinking it might fix my issue. When I changed the script to the
save(location, $((ff))) I get this config error when starting
opensips:
May 22 17:39:11 [14757] ERROR:core:pv_parse_spec: pvar (inner_name)
not found
May 22 17:39:11 [14757] ERROR:core:pv_parse_spec: wrong char [f/102]
in [$((ff))] at [3 (2)]
May 22 17:39:11 [14757] ERROR:core:fixup_spve: wrong format[$((ff))]
May 22 17:39:11 [14757] ERROR:core:fix_actions: fixing failed
(code=-1) at cfg line 767
May 22 17:39:11 [14757] CRITICAL:core:fix_expr: fix_actions error
May 22 17:39:11 [14757] ERROR:core:main: failed to fix configuration
with err code -1
Thanks
Nathaniel L Keeling
On 5/22/13 11:46 AM, Bogdan-Andrei Iancu wrote:
Could you try : save(location,$((ff))) ?
Do you get any error ?
What is your opensips version ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05/22/2013 05:49 PM, Nathaniel L Keeling III wrote:
Hello Bogdan,
I have validated the script and that i am passing a parameter. I
also changed the debug log statement that I displayed right before
the save() and I still get the same output. Here is the code that I
use in the script:
xlog(SAVING THE SUBSCRIBER INTO THE LOCATION TABLE, test with
extra debug ...);
if (!save(location, p1))
{
xlog(L_ERR, ERR:callerid:$ci|end|System error trying to
save Register's request location);
sl_reply_error();
}
xlog(L_INFO, INFO:callerid:$ci|end|The subscriber has
successfully registered with Akan Voice);
exit;
Is there a better way to validate? I am not sure of what else to check.
Thanks
Nathaniel L Keeling
On 5/22/13 6:02 AM, Bogdan-Andrei Iancu wrote:
Hi Nathaniel,
Well, the logs shows that save() does not receive any flags as
params...everything indicates that you do not have the params or
you are using the wrong config file.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05/22/2013 08:22 AM, Nathaniel L Keeling III wrote:
Hello Bogdan,
Here is the output from the opensips log. I have also attached a
snippet from the log file.
May 21 23:39:15 OpenSips[14397]: [ID 257313 local1.debug]
DBG:registrar:save_aux: xXx - flags param is
May 21 23:39:15 OpenSips[14397]: [ID 154992 local1.debug]
DBG:registrar:save_aux: xXx - flags bitmask is 0
May 21 23:39:15 OpenSips[14397]: [ID 269964 local1.debug]
DBG:registrar:pack_ci: xXx - flags are 0
Thanks
Nathaniel L Keeling
On 5/20/13 11:56 AM, Bogdan-Andrei Iancu wrote:
Hello Nathaniel,
See the attached patch - it logs more from the part where the
params are handled .
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05/18/2013 09:33 AM, Nathaniel L Keeling III wrote:
Hello Bogdan,
Here are snippets from my script. I only have one place where I
execute the save function. Just wondering, could it be
truncating a byte?
modparam(usrloc, nat_bflag, 10)
modparam(usrloc, use_domain, 1)
modparam(usrloc, db_mode, 3)
modparam(usrloc, db_url,
postgres://opensips:opensip...@ama.akan.net/opensips181t)
modparam(registrar, tcp_persistent_flag, 7)
modparam(registrar, received_avp, $avp(received_nh))
xlog(SAVING THE SUBSCRIBER INTO THE LOCATION TABLE
);
if (!save(location,p1))
{
xlog(L_ERR, ERR:callerid:$ci|end|System error trying
to save Register's request location);
sl_reply_error();
}
xlog(L_NOTICE, NOTICE:callerid:$ci|end|The subscriber has
successfully registered with Akan Voice);
exit;
Thanks
Nathaniel
On 5/17/13 6:07 AM, Bogdan-Andrei Iancu wrote:
Hello Nathaniel,
That is odd.it's like you do not set the p1 flag
I tested and I with p1 flag I get:
May 17 14:05:03 [7944] DBG:registrar:pack_ci: xXx - flags
are 10
Are you sure your script gets to the right save() ??
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05/17/2013 09:37 AM, Nathaniel L Keeling III wrote:
Hello Bogdan,
I added the patch and here is what I found: OpenSips[4378]:
[ID 269964 local1.debug] DBG:registrar:pack_ci: xXx - flags
are 0. I have also included the log file.
Thanks
Nathaniel Keeling
On 5/16/13 3:47 AM, Bogdan-Andrei Iancu wrote:
Hello Nathaniel,
Re: [OpenSIPS-Users] Sip server dipping/advice
And starting with trunk version, OpenSIPS implements a REST client - see
http://www.opensips.org/html/docs/modules/devel/rest_client.html
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05/23/2013 08:14 AM, Daniel Yu wrote:
M.Khaled,
It may be easier to perform HTTP Queries to perform the LRN lookup.
Have a look at www.bulkvs.com http://www.bulkvs.com where we offer
HTTP based LRN DIPs.
Cheers -
Daniel
-- Forwarded message --
From: *M.Khaled W Chehab* kche...@icucall.com
mailto:kche...@icucall.com
Date: Tue, May 14, 2013 at 6:17 AM
Subject: [OpenSIPS-Users] Sip server dipping/advice
To: users@lists.opensips.org mailto:users@lists.opensips.org,
users-boun...@lists.opensips.org
mailto:users-boun...@lists.opensips.org
Hi,
I am using opensips 1.8.x with do_routing module ,and trying to
implement the US dipping service in my script
I am sending the call to the dipping sip server gateway and
receive the 302 message containing the LRN number ,
As as result I can have the rn number but I am sending the invite
to the dipping provider twice ,please can you advice how to setup
it in the right way
if ($rU=~^.) {
route(7);
route(1);
exit;
route[1] {
if (subst_uri('/(sip:.*);nat=yes/\1/')) {
setbflag(6);
}
if (isflagset(5)) {
search_append('Contact:.*sip:[^[:cntrl:]]*',
';nat=yes');
}
# for INVITEs enable some additional helper routes
if (is_method(INVITE)) {
t_on_branch(2);
t_on_reply(2);
t_on_failure(1);
avp_db_query(UPDATE `opensips`.`invites` set
`trunkip` ='$rd' where `CALLID` = '$ci' );
} else if (is_method(BYE)) {
setflag(1); # do accounting ...
setflag(3); #transaction falis
setflag(4); #CDR Table
xlog(Route 1
Bye---);
} else if (is_method(ACK)) {
# call answered an ACKed, start billing here
} else if (is_method(CANCEL)) {
# call cancelled by caller, do clean up here' );
}
if (!t_relay()) {
xlog(L_INFO, --Debug Customer
ID:$avp(Cusid)/IP:$si--#11###Reply: $T_reply_code\n);
send_reply(500,Internal Error);
};
exit;
}
route[7]{
if (!do_routing($avp(Cusid),FW)) {
drop();
exit;
}
}
route[6] {
if ( use_next_gw() ) {
$var(prefix) =
$(avp(gw_attrs){csv.value,1});
$rU = $var(prefix) + $avp(dst);
xlog(L_INFO, --Debug Customer
ID:$avp(Cusid)/IP:$si-Calling number to Next Provier $rU\n);
setflag(26); #Missed calls
t_on_failure(1);
t_relay();
exit;
}
}
branch_route[2] {
xlog(L_INFO, --Debug Customer
ID:$avp(Cusid)/IP:$si-new branch at $ru\n);
route(7);
}
failure_route[1] {
if (!t_check_status(302)) {
if (!next_routing()){
xlog(L_INFO, LRN - Unable to DIP);
t_reply(500,Unable to DIP);
exit;
}
xlog(L_INFO, LRN - Unable to DIP - Trying Next);
t_on_failure(1);
t_relay();
exit;
}
if (!$(replyct.fields(uri){param.value,rn})){
xlog(L_INFO, LRN - No
redirect information found);
route(1);
}else if
($(replyct.fields(uri){param.value,rn}) == $tU){
xlog(L_INFO, LRN - Returned same number, no
need to redirect);
route(1);
}else{
xlog(LRN-$rU---Else lRN
$avp(lrnct)-);
$rU=$avp(lrnct);
Re: [OpenSIPS-Users] Slight problem routing 100s and 183s
Hi Nick, if INVITE goes .11 - .5 - .10 - .20 - .122, why the 100 reply from .122 goes to .5 ??? replies have to be relaid back exactly on the same path as the request. So, the 100 reply from .122 must go to .20, not to .5 !!! Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/22/2013 08:57 PM, Nick Khamis wrote: Hello Bogdan, Thank you so much for your response, and your time! The log is for the same call, only, the callid is getting changed by asterisk. What is happening is: 192.168.2.11 (UAC) - 192.168.2.5 (OpenSIPSIn) INVITE Call-ID: 4737d441-5fb15ea7-7142c0d8@192.168.2.11 mailto:4737d441-5fb15ea7-7142c0d8@192.168.2.11. 192.168.2.5 (OpenSIPSIn) - 192.168.2.10 (Asterisk) INVITE Call-ID: 4737d441-5fb15ea7-7142c0d8@192.168.2.11 mailto:4737d441-5fb15ea7-7142c0d8@192.168.2.11. 192.168.2.10 (Asterisk) - 192.168.2.20 (OpenSIPSOut) INVITE Call-ID: 1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060 http://1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060. 192.168.2.20 (OpenSIPSOut) - 94.101.2.122 (Service Provider) INVITE Call-ID: 1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060 http://1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060. 94.101.2.122:5060 http://94.101.2.122:5060 (ServicProvider) - 192.168.2.5:5060 http://192.168.2.5:5060 (OpenSIPSIn) Giving a Try Call-ID: 1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060 http://1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060. I am assuming because the callid coming into OpenSIPSIn from the service provider has been changed by asterisk, and OpensipsIn is not aware traffic with that callid, the 183 and 200s are being ignored? I experienced something similar with BYEs and 404, due to changed callid where Vlad solved the problem by explicitly forcing dialog matching using match_dialog. I am not sure if that is possible here too? http://lists.opensips.org/pipermail/users/2013-April/025322.html I also thought about trying to relay the 183 and 200s coming in from the service provider to asterisk. The reason for this is because asterisk has the two callid mapped, and can relay the traffic with the original callid back to the proxy. However, to limit the traffic going back and forth, if I can use the match_dialog approach again it would be perfect!! This is the last piece of the elephant!!! I hope I can put it together :) Kind Regards, Nick. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence BLF TLS
Sorry send wrong trace T 2013/05/23 13:24:13.215773 10.222.1.253:37827 - 10.222.0.17:5060 [AP] SUBSCRIBE sip:2...@g-voip.stb.ua SIP/2.0. Via: SIP/2.0/TCP 10.222.1.253:5060;branch=z9hG4bK1334330933;rport;alias. From: sip:9...@g-voip.stb.ua;tag=1005928247. To: sip:2...@g-voip.stb.ua. Call-ID: 310582788-5060...@ba.ccc.b.cfd. CSeq: 20070 SUBSCRIBE. Contact: sip:@10.222.1.253:5060;transport=tcp. X-Grandstream-PBX: true. Max-Forwards: 70. User-Agent: Grandstream GXP2120 1.0.5.24. Expires: 300. Supported: replaces, path, timer, eventlist. Event: dialog. Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Length: 0. . T 2013/05/23 13:24:13.216339 10.222.0.17:5060 - 10.222.1.253:37827 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 10.222.1.253:5060;received=10.222.1.253;branch=z9hG4bK1334330933;rport=37827;alias. From: sip:9...@g-voip.stb.ua;tag=1005928247. To: sip:2...@g-voip.stb.ua;tag=6b081e8cb2be0cc593bdd08e52ae2811-dae0. Call-ID: 310582788-5060...@ba.ccc.b.cfd. CSeq: 20070 SUBSCRIBE. Expires: 180. Contact: sip:sa@10.222.0.17:5060;transport=tcp. Server: OpenSIPS (1.8.3-tls (x86_64/linux)). Content-Length: 0. May 23 13:24:13 stb-bsip-TEST opensips[14672]: INFO:presence:update_subscription: notify May 23 13:24:13 stb-bsip-TEST opensips[14672]: ERROR:tm:msg_send: tcp_send failed May 23 13:24:13 stb-bsip-TEST opensips[14672]: ERROR:tm:t_uac: attempt to send to 'sip:@10.222.1.253:5060;transport=tcp' failed May 23 13:24:13 stb-bsip-TEST opensips[14672]: INFO:presence:send_notify_request: NOTIFY sip:9...@g-voip.stb.ua via sip:@10.222.1.253:5060;transport=tcp on behalf of sip:2...@g-voip.stb.ua for event dialog, to_tag=6b081e8cb2be0cc593bdd08e52ae2811-dae0, cseq=1 2013/5/23 Chusov Alexsandr chusov.alexsa...@gmail.com: Hello, Log Opensips: May 23 12:29:37 stb-bsip-TEST opensips[13572]: INFO:presence:update_subscription: notify May 23 12:29:37 stb-bsip-TEST opensips[13572]: ERROR:tm:msg_send: tcp_send failed May 23 12:29:37 stb-bsip-TEST opensips[13572]: ERROR:tm:t_uac: attempt to send to 'sip:@10.222.1.253:5060;transport=tcp' failed May 23 12:29:37 stb-bsip-TEST opensips[13572]: INFO:presence:send_notify_request: NOTIFY sip:9...@g-voip.stb.ua via sip:@10.222.1.253:5060;transport=tcp on behalf of sip:2...@g-voip.stb.ua for event dialog, to_tag=6b081e8cb2be0cc593bdd08e52ae2811-fcef, cseq=1 SIP trace tcp: T 2013/05/23 12:29:37.439107 10.222.1.253:45393 - 10.222.0.17:5060 [AP] SUBSCRIBE sip:2...@g-voip.stb.ua SIP/2.0. Via: SIP/2.0/TCP 10.222.1.253:5060;branch=z9hG4bK1476817771;rport;alias. From: sip:9...@g-voip.stb.ua;tag=971365295. To: sip:2...@g-voip.stb.ua. Call-ID: 1552429319-5060...@ba.ccc.b.cfd. CSeq: 20070 SUBSCRIBE. Contact: sip:@10.222.1.253:5060;transport=tcp. X-Grandstream-PBX: true. Max-Forwards: 70. User-Agent: Grandstream GXP2120 1.0.5.24. Expires: 300. Supported: replaces, path, timer, eventlist. Event: dialog. Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Length: 0. . T 2013/05/23 12:29:37.439608 10.222.0.17:5060 - 10.222.1.253:45393 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP 10.222.1.253:5060;received=10.222.1.253;branch=z9hG4bK1476817771;rport=45393;alias. From: sip:9...@g-voip.stb.ua;tag=971365295. To: sip:2...@g-voip.stb.ua;tag=6b081e8cb2be0cc593bdd08e52ae2811-fcef. Call-ID: 1552429319-5060...@ba.ccc.b.cfd. CSeq: 20070 SUBSCRIBE. Expires: 180. Contact: sip:sa@10.222.0.17:5061;transport=tcp. Server: OpenSIPS (1.8.3-tls (x86_64/linux)). Content-Length: 0. 2013/5/22 Bogdan-Andrei Iancu bog...@opensips.org: The registration info is used only for routing the SUBSCRIBE (which is ok). For the NOTIFY routing, the info from SUBSCRIBE + its 200 OK is used - this is why I need the pcap. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/22/2013 08:24 PM, Chusov Alexsandr wrote: I do not have pcap because use TLS. But I can tomorrow create pcap TCP. Two phone one enable Use Actual Ephemeral Port in Contact with TCP/TLS ( 2...@g-voip.stb.ua ) AOR:: 9...@g-voip.stb.ua Contact:: sip:@10.222.1.253:5060;transport=tls Q= Expires:: 247 Callid:: 311684349-506...@ba.ccc.b.cfd Cseq:: 2053 User-agent:: Grandstream GXP2120 1.0.5.24 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tls:10.222.0.17:5061 Methods:: 6015 AOR:: 2...@g-voip.stb.ua Contact:: sip:2083@10.222.1.221:45266;transport=tls Q= Expires:: 88 Callid::
Re: [OpenSIPS-Users] Slight problem routing 100s and 183s
Hello Bogdan, thank you so much. The reason for that is the port forwarding that is from NAT to .5 for SIP and RTP traffic. Do we have any options to relay the .100 to .5? I tried to set some flags on .5 (i.e., if(status==183 || status=100)) in the reply_route, but could not catch the replies coming from .122, even though the sip trace shows the traffic coming in. Would opensips just ignore sip traffic with callids it is not aware of? Finally, could we not force the dialog matching for traffic coming in from the outside with new callids? Kind Regards, Nick. On 5/23/13, Bogdan-Andrei Iancu bog...@opensips.org wrote: Hi Nick, if INVITE goes .11 - .5 - .10 - .20 - .122, why the 100 reply from .122 goes to .5 ??? replies have to be relaid back exactly on the same path as the request. So, the 100 reply from .122 must go to .20, not to .5 !!! Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/22/2013 08:57 PM, Nick Khamis wrote: Hello Bogdan, Thank you so much for your response, and your time! The log is for the same call, only, the callid is getting changed by asterisk. What is happening is: 192.168.2.11 (UAC) - 192.168.2.5 (OpenSIPSIn) INVITE Call-ID: 4737d441-5fb15ea7-7142c0d8@192.168.2.11 mailto:4737d441-5fb15ea7-7142c0d8@192.168.2.11. 192.168.2.5 (OpenSIPSIn) - 192.168.2.10 (Asterisk) INVITE Call-ID: 4737d441-5fb15ea7-7142c0d8@192.168.2.11 mailto:4737d441-5fb15ea7-7142c0d8@192.168.2.11. 192.168.2.10 (Asterisk) - 192.168.2.20 (OpenSIPSOut) INVITE Call-ID: 1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060 http://1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060. 192.168.2.20 (OpenSIPSOut) - 94.101.2.122 (Service Provider) INVITE Call-ID: 1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060 http://1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060. 94.101.2.122:5060 http://94.101.2.122:5060 (ServicProvider) - 192.168.2.5:5060 http://192.168.2.5:5060 (OpenSIPSIn) Giving a Try Call-ID: 1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060 http://1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060. I am assuming because the callid coming into OpenSIPSIn from the service provider has been changed by asterisk, and OpensipsIn is not aware traffic with that callid, the 183 and 200s are being ignored? I experienced something similar with BYEs and 404, due to changed callid where Vlad solved the problem by explicitly forcing dialog matching using match_dialog. I am not sure if that is possible here too? http://lists.opensips.org/pipermail/users/2013-April/025322.html I also thought about trying to relay the 183 and 200s coming in from the service provider to asterisk. The reason for this is because asterisk has the two callid mapped, and can relay the traffic with the original callid back to the proxy. However, to limit the traffic going back and forth, if I can use the match_dialog approach again it would be perfect!! This is the last piece of the elephant!!! I hope I can put it together :) Kind Regards, Nick. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Carrier route Failover
Hi,
I am using opensips 1.8.3,
In the do_routing($cusid,FW) I allow fail over and weight,
Moreover I create a two dial rules with same priority and prefix
1578,157 and assign to 1578 gwlist #USA that includes gw1 and gw2 to
it
for dial rule 157 I assign gwlist gw3
The dialed number is: 15782626262
As I allow fail over so, if the call didn't succeed in connecting through
#USA (dial rule prefix 1578) it will jump to the shortest match rule
that has a dial rule prefix 157 and go to gw3
Since I am in need to have both scenarios (failover and denting failover) ,
My question is :how can I set a flag or anything can do after rule
prefix 1578 to allow or disallow the failing over
I try to add an attribute flag (1 is blocked ,0 allow fail over) on
dr_rules and get it before use_next_gw() ,but in this case only the call
will go to gw1
$dlg_val(routeblock)=$avp(rule_attrs);
if($dlg_val(routeblock)==1) {
xlog(L_INFO, --Route 6---StickCheck--avp :
$avp(rule_attrs)---$dlg_val(\routeblock\)\n);
exit;
}
Please advice
Regards
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[OpenSIPS-Users] Opensips topology_hiding() , uac_auth() , B2B
Hello! Could you please just direct me little bit... centos 6.3 opensips-1.9.1 rtpproxy-1.2.1 I am trying to build a node that will play a b2b role. What I am looking for: 1. use several send_socket IP (depends on route) 2. use rtpproxy with different IP (depends on route) 3. use INVITE auth if required 4. Hide topology only from the called (leg B) party 5. I don't need NAT 6. I don't need to handle REGISTER What I got: 1. listen on several interfaces 2. force_send_socket for setting IP 3. use different rtpproxy sets for selecting proxy IP 4. use dialog topology_hiding() for hiding 5. use uac_auth() for adding auth info + fix of Cseq header 6. use UAC functions for changing from and to I don't ask you to guide me how to write a config, just let me know if I've selected correct way for the task. Or it is much simpler to solve with b2b module? Cause I was not able to implement all I need with b2b, probably I did it wrong... P.S. It is my first email here Best Regards, Roman Davydov, site: www.openvoip.co email: openvoip...@gmail.com cell: +1 (347) 637-8703 skype: openvoip.co ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Registrar not saving received from Path header
Hello Bogdan,
The change that you did is now causing things to work. I have attached a
snippet of the log file and I have listed the output from the extra
debug lines from your patch within the email:
May 23 09:31:28 OpenSips[23458]: [ID 197553 local1.error] SAVING THE
SUBSCRIBER INTO THE LOCATION TABLE, test with extra debug
...
May 23 09:31:28 OpenSips[23458]: [ID 795617 local1.debug]
DBG:registrar:save_aux: xXx - flags param is 10038f8a0
May 23 09:31:28 OpenSips[23458]: [ID 956053 local1.debug]
DBG:registrar:save_aux: xXx - flags string is p0
May 23 09:31:28 OpenSips[23458]: [ID 154992 local1.debug]
DBG:registrar:save_aux: xXx - flags bitmask is 20
May 23 09:31:28 OpenSips[23458]: [ID 497291 local1.debug]
DBG:core:parse_headers: flags=
.
.
May 23 09:31:28 OpenSips[23458]: [ID 497291 local1.debug]
DBG:core:parse_headers: flags=800
May 23 09:31:28 OpenSips[23458]: [ID 269964 local1.debug]
DBG:registrar:pack_ci: xXx - flags are 20
May 23 09:31:28 OpenSips[23458]: [ID 692724 local1.debug]
DBG:registrar:pack_ci: xXx - saving path into usrloc
May 23 09:31:28 OpenSips[23458]: [ID 497291 local1.debug]
DBG:core:parse_headers: flags=
Thanks
Nathaniel L Keeling
On 5/23/13 4:44 AM, Bogdan-Andrei Iancu wrote:
Hello Nathaniel,
My test was just to see if your opensips hits the fixup function for
the flags - and it seems it does (the value I gave you is bogus, so it
ended up with error).
Could you try this new patch (remove old one) ? (of course, put back
the p0 param ).
Thanks and regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
{\rtf1\ansi\ansicpg1252\cocoartf1138\cocoasubrtf510
{\fonttbl\f0\fswiss\fcharset0 Helvetica;}
{\colortbl;\red255\green255\blue255;}
\margl1440\margr1440\vieww21340\viewh14540\viewkind0
\pard\tx720\tx1440\tx2160\tx2880\tx3600\tx4320\tx5040\tx5760\tx6480\tx7200\tx7920\tx8640\pardirnatural
\f0\fs24 \cf0 May 23 09:31:28 OpenSips[23458]: [ID 197553 local1.error] SAVING THE SUBSCRIBER INTO THE LOCATION TABLE, test with extra debug ...\
May 23 09:31:28 OpenSips[23458]: [ID 795617 local1.debug]
\b DBG:registrar:save_aux: xXx - flags param is 10038f8a0
\b0 \
May 23 09:31:28 OpenSips[23458]: [ID 956053 local1.debug]
\b DBG:registrar:save_aux: xXx - flags string is p0
\b0 \
May 23 09:31:28 OpenSips[23458]: [ID 154992 local1.debug]
\b DBG:registrar:save_aux: xXx - flags bitmask is 20
\b0 \
May 23 09:31:28 OpenSips[23458]: [ID 497291 local1.debug] DBG:core:parse_headers: flags=\
May 23 09:31:28 OpenSips[23458]: [ID 421386 local1.debug] DBG:core:parse_uri: parsed uri:\
May 23 09:31:28type=1 user=nkeeling(8)\
May 23 09:31:28passwd=(0)\
May 23 09:31:28host=akanvoice.com(13)\
May 23 09:31:28port=(0): 0\
May 23 09:31:28params=(0)\
May 23 09:31:28headers=(0)\
May 23 09:31:28 OpenSips[23458]: [ID 883594 local1.debug] DBG:core:parse_uri: uri params:\
May 23 09:31:28 transport=, val=, proto=0\
May 23 09:31:28 OpenSips[23458]: [ID 996803 local1.debug] DBG:core:parse_uri:user-param=, val=\
May 23 09:31:28 OpenSips[23458]: [ID 784403 local1.debug] DBG:core:parse_uri:method=, val=\
May 23 09:31:28 OpenSips[23458]: [ID 185236 local1.debug] DBG:core:parse_uri:ttl=, val=\
May 23 09:31:28 OpenSips[23458]: [ID 628781 local1.debug] DBG:core:parse_uri:maddr=, val=\
May 23 09:31:28 OpenSips[23458]: [ID 949013 local1.debug] DBG:core:parse_uri:lr=, val=\
May 23 09:31:28 OpenSips[23458]: [ID 595631 local1.debug] DBG:core:parse_uri:r2=, val=\
May 23 09:31:28 OpenSips[23458]: [ID 323951 local1.debug] DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 8 chars, out: 8 chars\
May 23 09:31:28 OpenSips[23458]: [ID 323951 local1.debug] DBG:db_postgres:db_postgres_val2str: PQescapeStringConn: in: 13 chars, out: 13 chars\
May 23 09:31:28 OpenSips[23458]: [ID 181770 local1.debug] DBG:db_postgres:db_postgres_submit_query: 100391220 PQsendQuery(select contact,expires,q,callid,cseq,flags,cflags,user_agent,received,path,socket,methods,last_modified,sip_instance from location where username='nkeeling' AND domain='akanvoice.com' order by q)\
May 23 09:31:28 OpenSips[23458]: [ID 529718 local1.debug] DBG:core:db_new_result: allocate 48 bytes for result set at 1003960d0\
May 23 09:31:28 OpenSips[23458]: [ID 911073 local1.debug] DBG:db_postgres:db_postgres_store_result: 100391220 PQresultStatus(PGRES_TUPLES_OK) PQgetResult(100782900)\
May 23 09:31:28 OpenSips[23458]: [ID 460951 local1.debug] DBG:db_postgres:db_postgres_get_columns: 14 columns returned from the query\
May 23 09:31:28 OpenSips[23458]: [ID 586195 local1.debug] DBG:core:db_allocate_columns: allocate 392 bytes for result columns at 100396358\
May 23 09:31:28 OpenSips[23458]: [ID 412218 local1.debug] DBG:db_postgres:db_postgres_get_columns:
[OpenSIPS-Users] Event-RabbitMQ (and Event in general): pvar for event names
Greetings list, I'd like to be able to use a pvar in the subscribe function for the event interface like so: subscribe($avp(event_name), $avp(rabbit_url)) Where event_name would be like E_CUSTOM_SUBSCRIBE and url would be rabbitmq:localhost/$fU I know this is not supported yet and I'd be interested in implementing it, but understanding how pvars work is kinda tough without documentation. Also, it seems like you have to have a sip_msg in context for the function to translate a pvar into it's data type, but in the event interface it's not passed in. Any thoughts? I'd prefer not to use an infinite case statement: my use case is having a 1.9.x route block with param() calls that pass in the event name and the pvar for the queue name to subscribe to. Thanks list! ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
