Re: [OpenSIPS-Users] ACK and NAT problem

2010-11-26 Thread Nawfel Oujdi
Hello Bogdan,

 Thanks for reply my email and just say i fixed the problem.It was exactly
like you explain in your email so i did 2 changes in my opensips.cfg: first
i preset the record route with my public ip  ""record_route_preset("Public
IP")"  and second  i put  aliases= "Public IP" and then everything seems
work well.

Best Regards and great job!!
Nawfel



2010/11/25 Bogdan-Andrei Iancu 

> Hi Nawfel,
>
> The problem is that opensips, as running on a private address, will
> advertise in Route hdr that private IP (that will be useless when GW tries
> to route the ACK).
>
> So, currently, what is the way your opensips is reachable from outside ?
>
> Best regards,
> Bogdan
>
> Nawfel Oujdi wrote:
>
>> Hi!!
>>
>>  I load balancing  between 2 asterisk,the load_balancer it s done
>> correctly  but the calls hangs up after 10 second approximately. My setup is
>> (just with 1 asterisk):
>>   Cisco GW (NAT)> Opensips > Asterisk
>>
>> My problem is that opensips never receive the ACK from Cisco GW even if
>> this last one sent it.
>>
>> CISCO logs:
>> ACK sip:911126...@*public ip of asterisk* :5060 SIP/2.0 Route:
>> 
>>  I think cisco reply ACK to Asterisk through the internal ip of opensips
>> cause opensips did not provide Cisco with his public IP (NAT).
>>
>> Can somebody help me with this??
>>
>> I heard aboutfix_nated_contact() but i don't  know how and where in my
>> script¿Somebody send me examples ?
>>
>>
>> Thank you very much!!
>>
>> my opensips.cfg:
>> route{
>>
>>if (!mf_process_maxfwd_header("10")) {
>>sl_send_reply("483","looping");
>>exit;
>>}
>>if ($rU==NULL) {
>># request with no Username in RURI
>>   sl_send_reply("484","Address Incomplete");
>>   exit;
>>}
>>if (!has_totag()) {
>># initial request
>>record_route();
>>} else {
>># sequential request -> obey Route indication
>>loose_route();
>>t_relay();
>>exit;
>>}
>>   if ( is_method("CANCEL") ) {
>>if ( t_check_trans() )
>>t_relay();
>>exit;
>>}
>>
>>
>># from now on we have only the initial requests
>>if (!is_method("INVITE")) {
>>send_reply("405","Method Not Allowed");
>>exit;
>>}
>>
>># detect resources and do balancing
>>if (method=="INVITE") {
>>  load_balance("1","calls"); }
>>
>># LB function returns negative if no suitable destination (for
>> requested resources) is found,
>># or if all destinations are full
>>if ($retcode<0) {
>> sl_send_reply("500","Service full");
>> exit;
>>}
>>
>>xlog("Selected destination is: $du\n");
>>
>># send it out
>>if (!t_relay()) {
>>sl_reply_error();
>>}
>> }
>>
>>
>>
>>
>>
>>
>>
>>
>> 
>>
>> Aviso legal: Este mensaje electrónico está dirigido únicamente a la(s)
>> dirección(es) indicadas anteriormente; el carácter confidencial, personal e
>> intransferible del mismo está protegido legalmente. Cualquier revelación,
>> uso o reenvío no autorizado, completo o en parte, está prohibido. Si ha
>> recibido este mensaje por equivocación, notifíquelo inmediatamente a la
>> persona que lo ha enviado y borre el mensaje original junto con sus ficheros
>> anexos sin leerlo ni grabarlo, total o parcialmente.
>>
>>
>> 
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Bootcamp
> 15 - 19 November 2010, Edison, New Jersey, USA
> www.voice-system.ro
>
>
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>



-- 
Nawfel Oujdi
*Ingeniero VoIP*
nou...@egtelecom.es
 EG telecom S.A | www.egtelecom.es
Oficina: *902 050 080*
Agustín de Foxá, 25 - 9B | 28036 Madrid
--

Aviso legal: Este mensaje electrónico está dirigido únicamente a la(s)
dirección(es) indicadas anteriormente; el carácter confidencial, personal e
intransferible del mismo está protegido legalmente. Cualquier revelación,
uso o reenvío no autorizado, completo o en parte, está prohibido. Si ha
recibido este mensaje por equivocación, notifíquelo inmediatamente a la
persona que lo ha enviado y borre el mensaje original junto con sus ficheros
anexos sin leerlo ni grabarlo, total o parcialmente.
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[OpenSIPS-Users] opensips 100% cpu

2010-11-26 Thread Ангелов К.Н.

Hello,

my opensips process goes to 100% cpu after first call with default 
script and auth_aaa module enabled.

No such problem at all when auth_aaa disabled. OS FreeBSD 8.1/amd64

#loadmodule "auth_aaa.so"
#modparam("auth_aaa", "aaa_url", 
"radius:/usr/local/etc/radiusclient-ng/radiusclient.conf")

#modparam("auth_aaa", "service_type", 15)

It hangs at modules/tm/t_reply.c on LOCK_REPLIES( t );

*
+if (!onreply_avp_mode)   LM_DBG("!onreply_avp_mode!\n");
+if (!has_reply_route)LM_DBG("!has_reply_route!\n");

if (!onreply_avp_mode || !has_reply_route)   {
+   LM_DBG("Before lock\n");
/* lock the reply*/
LOCK_REPLIES( t );
+LM_DBG("After lock\n");
}


Here logs, without loadmodule "auth_aaa.so" and modparam auth_aaa

Nov 26 15:33:14 noc-pbx /usr/local/sbin/opensips[70850]: 
DBG:tm:reply_received: org. status uas=0, uac[0]=0 local=0 is_invite=0)
Nov 26 15:33:14 noc-pbx /usr/local/sbin/opensips[70850]: 
DBG:tm:reply_received: !onreply_avp_mode!
Nov 26 15:33:14 noc-pbx /usr/local/sbin/opensips[70850]: 
DBG:tm:reply_received: !has_reply_route!
Nov 26 15:33:14 noc-pbx /usr/local/sbin/opensips[70850]: 
DBG:tm:reply_received: Before lock
Nov 26 15:33:14 noc-pbx /usr/local/sbin/opensips[70850]: 
DBG:tm:reply_received: After lock
Nov 26 15:33:14 noc-pbx /usr/local/sbin/opensips[70850]: 
DBG:tm:t_should_relay_response: T_code=0, new_code=200
Nov 26 15:33:14 noc-pbx /usr/local/sbin/opensips[70850]: 
DBG:tm:relay_reply: branch=0, save=0, relay=0



with them

Nov 26 15:35:33 noc-pbx /usr/local/sbin/opensips[70900]: 
DBG:tm:reply_received: org. status uas=0, uac[0]=0 local=0 is_invite=0)
Nov 26 15:35:33 noc-pbx /usr/local/sbin/opensips[70900]: 
DBG:tm:reply_received: !onreply_avp_mode!
Nov 26 15:35:33 noc-pbx /usr/local/sbin/opensips[70900]: 
DBG:tm:reply_received: !has_reply_route!
Nov 26 15:35:33 noc-pbx /usr/local/sbin/opensips[70900]: 
DBG:tm:reply_received: Before lock


..100%CPU.


gdb

Loaded symbols for /usr/local/lib/libradiusclient-ng.so.2
Reading symbols from /usr/local/lib/opensips/modules/auth_aaa.so...(no 
debugging symbols found)...done.

Loaded symbols for /usr/local/lib/opensips/modules/auth_aaa.so
Reading symbols from /libexec/ld-elf.so.1...(no debugging symbols 
found)...done.

Loaded symbols for /libexec/ld-elf.so.1
[Switching to Thread 800c041c0 (LWP 100216)]
0x00080089e74c in sched_yield () from /lib/libc.so.7
(gdb) bt
#0  0x00080089e74c in sched_yield () from /lib/libc.so.7
#1  0x00080073e10f in pthread_mutex_getyieldloops_np ()
   from /lib/libthr.so.3
#2  0x0008014f98a2 in reply_received ()
   from /usr/local/lib/opensips/modules/tm.so
#3  0x0041c5fe in forward_reply ()
#4  0x00444078 in receive_msg ()
#5  0x004746d4 in udp_rcv_loop ()
#6  0x00422b5a in main ()
*


Any tips ?

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Re: [OpenSIPS-Users] CANCELing the connection - no totag in ACK

2010-11-26 Thread Maciej Bylica
Thank You Bogdan.

This is the way i am going to follow.

Maciej.

> Maciej,
>
> skip auth challenge for the ACK requests as you cannot send replies for an
> ACK
>
> So you have 2 options:
>   1) if you do not want to auth ACK at all, simple skip them from auth
>   2) if you want the auth ACK, if the ACK does not have an Authorize hdr
> from beginning (as RFC sais) you cannot do much about it.
>
> Regards,
> Bogdan
>
> Maciej Bylica wrote:
>>
>> Iñaki
>>
>>
>>
>>>
>>> It's well explained in RFC 3261.
>>> An ACK for a [3456]XX response must have same branch and same CSeq
>>> number (but "ACK" method) as the INVITE of the transaction.
>>>
>>
>> I meant some hints regarding script configuration, because as far as i
>> understand i should double check my .cfg
>> Okay i may proxy auth only INVITE methods - at this moment i do have
>>        if (!(method=="REGISTER") && from_uri==myself)  /*no multidomain
>> version*/ {
>>                if (!proxy_authorize("", "subscriber")) {
>>                        xlog("L_INFO","proxy auth");
>>                        proxy_challenge("", "0");
>>                        exit;
>>                }
>> so there wont be any problem to filter that out, but how to inspect
>> branch, CSeq - isn't that functionality hardcoded?
>>
>> Thx,
>> Maciej.
>>
>> ___
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>>
>>
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Bootcamp
> 15 - 19 November 2010, Edison, New Jersey, USA
> www.voice-system.ro
>
>
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>

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Re: [OpenSIPS-Users] nat transversal

2010-11-26 Thread Adrian Georgescu
The port is on the m line

On Nov 23, 2010, at 9:04 PM, abdelghafour harraz wrote:

> hey, 
> 
> actually i wanted to know how the nat helper module manage to inform the 
> rtpproxy of the port number used by a symmetric nat, cause in the sdp packet, 
> the only thing that is mentionned is the public address of the nat, and not 
> the port number used by the nat.
> 
> thanks 
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[OpenSIPS-Users] MySQL LAST_INSERT_ID() safe?

2010-11-26 Thread Chris Maciejewski
Hi,

I am trying to figure out if is it safe to use in OpenSIPs config file:

avp_db_query("INSERT INTO some_table VALUES ('data1')");
avp_db_query("SELECT LAST_INSERT_ID()","$avp(last_id);");

// do stuff with $avp(last_id);

How does OpenSIPs connects to MySQL sever? Does it use some smart
connection pooling maybe, which could result in unexpected behaviour?

Regards,
Chris

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[OpenSIPS-Users] Opensisp behind a Firewall and NAT

2010-11-26 Thread Stefano Pisani

Hallo,
my architecture needs the Opensips behind a firewall than does static 
NAT between the private IP of the opensips and the public ip.
I would use advertise_host to give correct public IP only for the 
messages directed outside and not for those directed inside; in 
different words, if an internal client calls another internal client I 
do not want to use advertise_host but If an internal client calls an 
external client I need to use it.
I can't see how to manage different behavior of advertise_host. The is a 
way to specify which of the networks are local?

Thanks.

s

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