Re: [OpenSIPS-Users] dns srv question

2014-02-11 Thread Miha 

tnx for you answer.

br
miha

Dne 2/11/2014 4:38 PM, piše Bogdan-Andrei Iancu:

Hello,

OpenSIPS will automatically try NAPTR/SRV (of course, depending on the 
URI form - like port must be unknown in order to go for SRV lookup).


If I got it right, your hackish DNS configuration may work - using the 
same name for the SRV and for the A record - it should not be any 
conflict at all - phones with SRV capabilities will go via SRV, the 
ones without will do A lookup (in both cases on the same domain name).


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 11.02.2014 16:26, Miha wrote:

Hi,

opensips in my case does not use dns srv domin. I am using dns srv 
domain just for UAC's, for geo redundancy. As some phones does not 
support dns srv (small amount) it is easier (for me) to use the same 
domain for registration's. I have tftp server for all configration 
and it is easy and nicers that registrar in all config is the same. 
That is why I would like to define same A record as is dns srv domain.


So, I will have two A record and if I understand you right, Opensips 
will sometimes bind to one and sometimes to another record? Could be 
this causing any problems?


tnx
miha

Dne 2/11/2014 2:53 PM, piše Olle E. Johansson:

On 11 Feb 2014, at 14:47, Miha  wrote:


Hi

Now I have one DNS SRV domain (sip.domain.com) which points to two 
A record inputs with different weight (sip1.sss.domain.com and 
sip2.sss.domain.com).


If I add for sip.domain.com  A record which  will pont to the same 
IP as sip1.sss.domain.com, could this be causing any problems for 
opensips (there will be two A record two the same IP, one with the 
same name as dns srv record)?
If you have configured your proxy to look up SRV records, those A 
records for "sip.domain.com" will not be asked for when doing SRV 
lookups.


If your proxy is not configured for doing SRV lookups, then it will 
find the A records and use them. Debugging will be hard.


I would start with adding SRV records in DNS, test everything. When 
I have confirmed that it works with software that implements SRV 
properly, I would add the A records for the same domain.


/O
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Re: [OpenSIPS-Users] Memory leakage in Opensips 1.9.1

2014-02-11 Thread Ahsan Hasan 
The Opensips version I am using is 1.9.1 compiled from src tar. The output
of opensips -V is below:

===
version: opensips 1.9.1-tls (x86_64/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST,
SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
svnrevision: unknown
@(#) $Id: main.c 9790 2013-02-15 10:14:34Z bogdan_iancu $
main.c compiled on 03:11:32 Feb  6 2014 with gcc 4.4.5
===


The backtrace from last 3 crashes are:

===
#0  0x7fc3d12fe1b5 in raise () from /lib/libc.so.6
(gdb) bt
#0  0x7fc3d12fe1b5 in raise () from /lib/libc.so.6
#1  0x7fc3d1300fc0 in abort () from /lib/libc.so.6
#2  0x004e8b19 in qm_free (qm=0x7fc3b8e59000, p=0x7fc3b92d90c0,
file=0x596ed9 "parser/sdp/sdp.c", func=0x596da0 "free_cloned_sdp_stream",
line=874) at mem/q_malloc.c:450
#3  0x0051ab2d in free_cloned_sdp_stream (_stream=) at parser/sdp/sdp.c:874
#4  0x0051ada3 in free_cloned_sdp_session (_session=) at parser/sdp/sdp.c:894
#5  0x7fc3cc380012 in destroy_qos (qos_sdp=0x7fc3b92aaba0) at
qos_ctx_helpers.c:73
#6  0x7fc3cc383aeb in remove_sdp (qos_ctx=, dir=1,
_m=0x7fc3d01ba410, role=, other_role=1) at
qos_ctx_helpers.c:539
#7  0x7fc3cc384220 in qos_dialog_response_CB (did=, type=, params=) at
qos_handlers.c:271
#8  0x7fc3cd01a82e in run_dlg_callbacks (type=512, dlg=0x7fc3b9333ec8,
msg=, dir=4294967295, dlg_data=0x0) at dlg_cb.c:253
#9  0x7fc3cdedc9a5 in run_trans_callbacks (type=8,
trans=0x7fc3b9f94c18, req=0x7fc3ba5fa7c0, rpl=,
code=) at t_hooks.c:212
#10 0x7fc3cdefa966 in relay_reply (t=0x7fc3b9f94c18,
p_msg=0x7fc3d01ba410, branch=0, msg_status=491, cancel_bitmap=) at t_reply.c:1218
#11 0x7fc3cdefb579 in reply_received (p_msg=0x7fc3d01ba410) at
t_reply.c:1549
#12 0x0042fa45 in forward_reply (msg=0x7fc3d01ba410) at
forward.c:575
#13 0x0047c80d in receive_msg (
buf=0x7e3620 "SIP/2.0 491 Another INVITE transaction in
progress\r\nVia: SIP/2.0/UDP
XXX.XXX.XXX.XXX:6000;received=77.66.2.136;branch=z9hG4bKc83b.cbf7d513.0\r\nVia:
SIP/2.0/UDP XXX.XXX.XXX.XXX:5090;rport=5090;received=77.66.2."..., len=517,
rcv_info=0x7fff1ecac640) at receive.c:207
#14 0x004e03d1 in udp_rcv_loop () at udp_server.c:424
#15 0x00438296 in main_loop (argc=,
argv=) at main.c:884
#16 main (argc=, argv=) at
main.c:1557
===

===
#0  0x7f32f24fd1b5 in raise () from /lib/libc.so.6
(gdb) bt
#0  0x7f32f24fd1b5 in raise () from /lib/libc.so.6
#1  0x7f32f24fffc0 in abort () from /lib/libc.so.6
#2  0x004e8b19 in qm_free (qm=0x7f32da058000, p=0x7f32da46f370,
file=0x596ed9 "parser/sdp/sdp.c", func=0x596da0 "free_cloned_sdp_stream",
line=874) at mem/q_malloc.c:450
#3  0x0051ab2d in free_cloned_sdp_stream (_stream=) at parser/sdp/sdp.c:874
#4  0x0051ada3 in free_cloned_sdp_session (_session=) at parser/sdp/sdp.c:894
#5  0x7f32ed57f012 in destroy_qos (qos_sdp=0x7f32da48efe0) at
qos_ctx_helpers.c:73
#6  0x7f32ed582aeb in remove_sdp (qos_ctx=, dir=1,
_m=0x7f32f13b8de0, role=, other_role=1) at
qos_ctx_helpers.c:539
#7  0x7f32ed583220 in qos_dialog_response_CB (did=, type=, params=) at
qos_handlers.c:271
#8  0x7f32ee21982e in run_dlg_callbacks (type=512, dlg=0x7f32da472ef0,
msg=, dir=4294967295, dlg_data=0x0) at dlg_cb.c:253
#9  0x7f32ef0db9a5 in run_trans_callbacks (type=8,
trans=0x7f32da4ba1f0, req=0x7f32da4d99a8, rpl=,
code=) at t_hooks.c:212
#10 0x7f32ef0f9966 in relay_reply (t=0x7f32da4ba1f0,
p_msg=0x7f32f13b8de0, branch=0, msg_status=491, cancel_bitmap=) at t_reply.c:1218
#11 0x7f32ef0fa579 in reply_received (p_msg=0x7f32f13b8de0) at
t_reply.c:1549
#12 0x0042fa45 in forward_reply (msg=0x7f32f13b8de0) at
forward.c:575
#13 0x0047c80d in receive_msg (
buf=0x7e3620 "SIP/2.0 491 Another INVITE transaction in
progress\r\nVia: SIP/2.0/UDP
XXX.XXX.XXX.XXX:6000;received=77.66.2.136;branch=z9hG4bK4d4d.6b9c34d6.0\r\nVia:
SIP/2.0/UDP XXX.XXX.XXX.XXX:5090;rport=5090;received=77.66.2."..., len=517,
rcv_info=0x7fffd3e54750) at receive.c:207
#14 0x004e03d1 in udp_rcv_loop () at udp_server.c:424
#15 0x00438296 in main_loop (argc=,
argv=) at main.c:884
#16 main (argc=, argv=) at
main.c:1557
===

===
#0  0x7fe13b0e31b5 in raise () from /lib/libc.so.6
(gdb) bt
#0  0x7fe13b0e31b5 in raise () from /lib/libc.so.6
#1  0x7fe13b0e5fc0 in abort () from /lib/libc.so.6
#2  0x004e8b19 in qm_free (qm=0x7fe122c3e000, p=0x7fe1230a02a0,
file=0x596ed9 "parser/sdp/sdp.c", func=0x596da0 "free_cloned_sdp_stream",
line=874) at mem/q_malloc.c:450
#3  0x0051ab2d in free_cloned_sdp_stream (_stream=) at parser/sdp/sdp.c:874
#4  0x0051ada3 in free_cloned_sdp_s

Re: [OpenSIPS-Users] Setting do_routing

2014-02-11 Thread Nick Altmann 
As I write before... Look into these avps after do_routing:

modparam("drouting", "ruri_avp", '$avp(dr_ruri)')
modparam("drouting", "gw_attrs_avp", '$avp(dr_gw_attrs)')
modparam("drouting", "gw_prefixes_avp", '$avp(dr_gw_pref)')

Try to change order there (in all avps the same order).
use_next_gw() just extract next gw from these avps.
If you would change order in avps, use_next_gw() will use YOUR order.

I use method like this to change order to very-very custom in dispatcher
module.
Don't see problems to do this with drouting.

P.S. You may write me personal email if you don't understand how to do it.
I'd can try to test this solution when will have free time.

--
Nick


2014-02-11 23:37 GMT+04:00 Nick Cameo :

>
> On Thu, Feb 6, 2014 at 1:54 PM, Nick Altmann wrote:
>
>> Carrier is just a gateways list. You may also set weight for each gw
>> and drouting will choose gw's due its weight.
>> Route_to_carrier() gets carrier parameters (gateways list) into avps
>> and set $ru to first gw address.
>> Use_next_gw() extract next gw parameters from avps and set $ru to the
>> new gateway.
>> So, you may try to work with these avps to change routing behaviour.
>>
>>
> Hello Nick,
>
> Thank you for your response. After looking into it further I realized that
> this approach is not very scalable. The reason why
> is because we have over 30 interconnects (30 gateways), and would have to
> maintain all of them to the carrier table.
>
> Is there not way to pass a gwlist list to a function dynamically (at run
> time) that will route to that call with failover? do_routing
> would be perfect! If it adheres to the order of the gateway list.
>
> PS I really don't want to change the ruri in perl either if this
> functionality can be made available in the script.
>
> Kind Regards,
>
> Nick.
>
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Re: [OpenSIPS-Users] Adjusting Headers

2014-02-11 Thread Alec Doran-Twyford 
Hi,
I have attached a copy from the pbx and the opensips/ I been told by one of
my colleagues that the contact header maybe the one which is effecting the
call working. I can't seem to find away to modify the contact header.
However I found this am I on the right track if I need to modify the
contact header?


Alec Doran-Twyford

| Junior Support Enginner for IVSTel
| E-mail: a.dorantwyf...@ivstel.com | Phone: +61 2 9288 8890 |


OPENSIPS.pcap
Description: Binary data


FREEPBXa.pcap
Description: Binary data
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Re: [OpenSIPS-Users] Setting do_routing

2014-02-11 Thread Nick Cameo 
On Thu, Feb 6, 2014 at 1:54 PM, Nick Altmann  wrote:

> Carrier is just a gateways list. You may also set weight for each gw
> and drouting will choose gw's due its weight.
> Route_to_carrier() gets carrier parameters (gateways list) into avps
> and set $ru to first gw address.
> Use_next_gw() extract next gw parameters from avps and set $ru to the
> new gateway.
> So, you may try to work with these avps to change routing behaviour.
>
>
Hello Nick,

Thank you for your response. After looking into it further I realized that
this approach is not very scalable. The reason why
is because we have over 30 interconnects (30 gateways), and would have to
maintain all of them to the carrier table.

Is there not way to pass a gwlist list to a function dynamically (at run
time) that will route to that call with failover? do_routing
would be perfect! If it adheres to the order of the gateway list.

PS I really don't want to change the ruri in perl either if this
functionality can be made available in the script.

Kind Regards,

Nick.
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Re: [OpenSIPS-Users] Interaction between OpenSIPS as UAC and real UAC

2014-02-11 Thread Bogdan-Andrei Iancu 

Hello,

Please keep the list CC'ed all the time !

You did no inserted the topo hiding triggering in the write place - you 
need to do it only for initial INVITEs ; for sequential requests you 
need the be sure to invoke the match_dialog() function. See:

http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id295144

It will be helpful to provide more info on what "it is not working" - 
like do you see any changes on the INVITE sent out by OpenSIPS, any 
script errors, etc



Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 11.02.2014 16:21, Александр Пучков wrote:

11.02.2014 16:37, Bogdan-Andrei Iancu пишет:

Hello,

Use the dialog based topology hiding
http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id296001

when forwarding the INVITE to Asterisk (in OpenSIPS).

Thank you.

I tried it like this code fragment:

#--8<--

route {


if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}

#force_rport();

if(avp_db_load("$fu","$avp(trace)")) {
$avp(traceuser)=$fu;
setflag(22);
sip_trace();
xlog("L_INFO","User $fu being traced");
}
#...
if (db_is_user_in("$fu", "asterisk"))
{
if(!has_totag() && is_method("INVITE")) {
topology_hiding();
}
}

if (has_totag()) {
...
}

...
#--8<--

But it not worked :( Perhaps, this code is wrong.

Could you indicate where the error? Could you indicate where the 
error? Or do I need to follow the documentation on the function 
topology_hiding()?


Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 11.02.2014 10:12, Александр Пучков wrote:


10.02.2014 13:24, Bogdan-Andrei Iancu пишет:

Hello,

In this scenario:

Astеrisk <-- 3 -- OpenSIPS <-- 4 -- UAC

What SIP requests the UAC is sending ? REGISTER ? INVITES ?

Hello!

UAC registered to Opensips, Opensips registered as UAC on Asterisk. 
When the INVITE request comes from the UAC, Opensips need to sent 
INVITE to Asterisk as UAC.


Note that the UAC does not know about Asterisk and Asterisk does not 
know about UAC. Asterisk know about only Opensips.


It is necessary to any UAC registered on Opensips could imagine how 
extension on Asterisk, without changing the configuration of UAC.


Thank you.




Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 20.01.2014 13:05, Александр Пучков wrote:

Hi!

OpenSIPS version 1.8.

We have the following diagram:

PSTN<-- 1 --> OpenSIPS<-- 2 --> Astеrisk<-- 3 --> UAC
192.168.1.1 <-- 1 --> 192.168.1.2 <-- 2 --> 192.168.1.3 <-- 3 --> 
192.168.1.*


I would like to make the following scheme:

PSTN<-- 1 --> OpenSIPS<-- 2 --> Astеrisk<-- 3 --> 
OpenSIPS<-- 4 --> UAC
192.168.1.1 <-- 1 --> 192.168.1.2 <-- 2 --> 192.168.1.3 <-- 3 --> 
192.168.1.2 <-- 4 --> 192.168.1.*


Interestingly the following schema fragment:
Astеrisk <-- 3 --> OpenSIPS<-- 4 --> UAC

Here OpenSIPS need to increase control over the services for UAC. 
UAC should not know about the existence of Asterisk and UAC must 
be registered on the server OpenSIPS, and any SIP request OpenSIPS 
should redirect to Asterisk.


I tried to use the module in UAC_REGISTRANT OpenSIPS, it works 
fine in the direction:


Astеrisk -- 3 --> OpenSIPS -- 4 --> UAC

But how to implement the scheme in the direction:

Astеrisk <-- 3 -- OpenSIPS <-- 4 -- UAC

I do not really imagine. Please tell me how it can be implemented.

Thank!


*Александр Пучков,
Системный администратор,
Тел.: +7(496) 569-24-24 доб.тел. 255;
e-mail: mai...@poig.ru;
*ООО "ПОИГ" (Интернет-провайдер г. Щелково)
Факс: +7(496) 569-24-24 доб. 103;
Адрес: 141108, М.О., г.Щелково, Пролетарский пр., д.11;
WEB: http://www.schelkovo-net.ru



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*Александр Пучков,
Системный администратор,
Тел.: +7(496) 569-24-24 доб.тел. 255;
*ООО "ПОИГ" (Интернет-провайдер г. Щелково)
Факс: +7(496) 569-24-24 доб. 103;
Адрес: 141108, М.О., г.Щелково, Пролетарский пр., д.11;
WEB: http://www.schelkovo-net.ru







*Александр Пучков,
Системный администратор,
Тел.: +7(496) 569-24-24 доб.тел. 255;
*ООО "ПОИГ" (Интернет-провайдер г. Щелково)
Факс: +7(496) 569-24-24 доб. 103;
Адрес: 141108, М.О., г.Щелково, Пролетарский пр., д.11;
WEB: http://www.schelkovo-net.ru



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Re: [OpenSIPS-Users] dns srv question

2014-02-11 Thread Bogdan-Andrei Iancu 

Hello,

OpenSIPS will automatically try NAPTR/SRV (of course, depending on the 
URI form - like port must be unknown in order to go for SRV lookup).


If I got it right, your hackish DNS configuration may work - using the 
same name for the SRV and for the A record - it should not be any 
conflict at all - phones with SRV capabilities will go via SRV, the ones 
without will do A lookup (in both cases on the same domain name).


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 11.02.2014 16:26, Miha wrote:

Hi,

opensips in my case does not use dns srv domin. I am using dns srv 
domain just for UAC's, for geo redundancy. As some phones does not 
support dns srv (small amount) it is easier (for me) to use the same 
domain for registration's. I have tftp server for all configration and 
it is easy and nicers that registrar in all config is the same. That 
is why I would like to define same A record as is dns srv domain.


So, I will have two A record and if I understand you right, Opensips 
will sometimes bind to one and sometimes to another record? Could be 
this causing any problems?


tnx
miha

Dne 2/11/2014 2:53 PM, piše Olle E. Johansson:

On 11 Feb 2014, at 14:47, Miha  wrote:


Hi

Now I have one DNS SRV domain (sip.domain.com) which points to two A 
record inputs with different weight (sip1.sss.domain.com and 
sip2.sss.domain.com).


If I add for sip.domain.com  A record which  will pont to the same 
IP as sip1.sss.domain.com, could this be causing any problems for 
opensips (there will be two A record two the same IP, one with the 
same name as dns srv record)?
If you have configured your proxy to look up SRV records, those A 
records for "sip.domain.com" will not be asked for when doing SRV 
lookups.


If your proxy is not configured for doing SRV lookups, then it will 
find the A records and use them. Debugging will be hard.


I would start with adding SRV records in DNS, test everything. When I 
have confirmed that it works with software that implements SRV 
properly, I would add the A records for the same domain.


/O
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Re: [OpenSIPS-Users] Memory leakage in Opensips 1.9.1

2014-02-11 Thread Bogdan-Andrei Iancu 

Hello,

Two things:

1) be sure you have the latest 1.9.1 code (GIT or tar balls)

2) post the actual backtrace  (from the core file), please

Thanks and regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 11.02.2014 06:25, Ahsan Hasan wrote:

So it crashed again today. Here is the log

Feb 10 18:26:24 RTSIP rtsip-service[10443]: CRITICAL:core:qm_free: 
freeing already freed pointer, first free: parser/sdp/sdp.c: 
free_cloned_sdp_stream(874) - aborting
Feb 10 18:26:25 RTSIP rtsip-service[10464]: CRITICAL:core:receive_fd: 
EOF on 25





On Mon, Feb 10, 2014 at 2:45 PM, Ahsan Hasan 
mailto:ahsanhasanja...@gmail.com>> wrote:


My opensips has crashed again. Here is the core dump.

= core dump =
Reading symbols from /lib/libnss_compat.so.2...(no debugging
symbols found)...done.
Loaded symbols for /lib/libnss_compat.so.2
Reading symbols from /lib/libnss_nis.so.2...(no debugging symbols
found)...done.
Loaded symbols for /lib/libnss_nis.so.2
Reading symbols from /lib/libnss_files.so.2...(no debugging
symbols found)...done.
Loaded symbols for /lib/libnss_files.so.2
Core was generated by `/usr/local/sbin/opensips -P
/var/run/opensips/opensips.pid -m 256 -M 16 -u root'.
Program terminated with signal 6, Aborted.
#0  0x7fe13b0e31b5 in raise () from /lib/libc.so.6

===

The memlogs are also attached.

-- 
Ahsan Hasan




On Thu, Jan 9, 2014 at 3:14 PM, Ahsan Hasan
mailto:ahsanhasanja...@gmail.com>> wrote:

Hi Razvan,

Let me enable it and wait for the next crash.


Regards,
-- 
Ahsan Hasan




On Thu, Jan 9, 2014 at 12:44 PM, Ra(zvan Crainea
mailto:raz...@opensips.org>> wrote:

Hi, Ahsan!

Can you please enable the the memory debugging
(DBG_QM_MALLOC flag in menuconfig)? You can follow this
tutorial[1].

[1]
http://www.opensips.org/Documentation/TroubleShooting-OutOfMem

Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com


On 01/09/2014 07:54 AM, Ahsan Hasan wrote:


We are using opensips-1.9.1 and it is crashing
randomly every other
week, the core dumps generated are always related to
memory leakage. The
last four core dumps are

Program terminated with signal 11, Segmentation fault.
#0  0x7f4c12e20c77 in write_dialog_profiles
(links=0x7f4bff03d7f0)
at dlg_db_handler.c:1085
1085l += link->profile->name.len + 1 +
link->value.len + 1;

Program terminated with signal 11, Segmentation fault.
#0  0x004e5da2 in fm_remove_free
(qm=0x7f4bfec5a000, size=112)
at mem/f_malloc.c:172
172  *pf=n->u.nxt_free;

Program terminated with signal 11, Segmentation fault.
#0  free_cloned_sdp_session
(_session=0x362e373740333532) at
parser/sdp/sdp.c:893
893session = l_session->next;

Program terminated with signal 11, Segmentation fault.
#0  0x004e5da2 in fm_remove_free
(qm=0x7f71a9524000, size=96) at
mem/f_malloc.c:172
172  *pf=n->u.nxt_free;

What can be the cause?

--
Ahsan Hasan



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+92.333.438.2041


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Re: [OpenSIPS-Users] Registrar - max_contacts and 503 logging

2014-02-11 Thread Bogdan-Andrei Iancu 

Hello,

OK, thanks for testing - I will push the fix I did in the public code. I 
will also give a bit more thinking to see what should be the best 
solution to expose the internal error.


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 10.02.2014 18:29, Adrien Martin wrote:

Hello,

I tested it, and it works.

About the save() return codes, there are a lot of cases, so I would have
sorted it this way:
- no error,
- error codes in parsing SIP,
- server side errors (like manipulating usrloc),
- and service errors (like too many registers).

Here is a patch by way of example, but I can't say if it's the right way
to do.

Regards,



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Re: [OpenSIPS-Users] dns srv question

2014-02-11 Thread Miha 

Hi,

opensips in my case does not use dns srv domin. I am using dns srv 
domain just for UAC's, for geo redundancy. As some phones does not 
support dns srv (small amount) it is easier (for me) to use the same 
domain for registration's. I have tftp server for all configration and 
it is easy and nicers that registrar in all config is the same. That is 
why I would like to define same A record as is dns srv domain.


So, I will have two A record and if I understand you right, Opensips 
will sometimes bind to one and sometimes to another record? Could be 
this causing any problems?


tnx
miha

Dne 2/11/2014 2:53 PM, piše Olle E. Johansson:

On 11 Feb 2014, at 14:47, Miha  wrote:


Hi

Now I have one DNS SRV domain (sip.domain.com) which points to two A record 
inputs with different weight (sip1.sss.domain.com and sip2.sss.domain.com).

If I add for sip.domain.com  A record which  will pont to the same IP as 
sip1.sss.domain.com, could this be causing any problems for opensips (there 
will be two A record two the same IP, one with the same name as dns srv record)?

If you have configured your proxy to look up SRV records, those A records for 
"sip.domain.com" will not be asked for when doing SRV lookups.

If your proxy is not configured for doing SRV lookups, then it will find the A 
records and use them. Debugging will be hard.

I would start with adding SRV records in DNS, test everything. When I have 
confirmed that it works with software that implements SRV properly, I would add 
the A records for the same domain.

/O
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Re: [OpenSIPS-Users] dns srv question

2014-02-11 Thread Olle E. Johansson 

On 11 Feb 2014, at 14:47, Miha  wrote:

> Hi
> 
> Now I have one DNS SRV domain (sip.domain.com) which points to two A record 
> inputs with different weight (sip1.sss.domain.com and sip2.sss.domain.com).
> 
> If I add for sip.domain.com  A record which  will pont to the same IP as 
> sip1.sss.domain.com, could this be causing any problems for opensips (there 
> will be two A record two the same IP, one with the same name as dns srv 
> record)?

If you have configured your proxy to look up SRV records, those A records for 
"sip.domain.com" will not be asked for when doing SRV lookups.

If your proxy is not configured for doing SRV lookups, then it will find the A 
records and use them. Debugging will be hard.

I would start with adding SRV records in DNS, test everything. When I have 
confirmed that it works with software that implements SRV properly, I would add 
the A records for the same domain.

/O
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[OpenSIPS-Users] dns srv question

2014-02-11 Thread Miha 

Hi

Now I have one DNS SRV domain (sip.domain.com) which points to two A 
record inputs with different weight (sip1.sss.domain.com and 
sip2.sss.domain.com).


If I add for sip.domain.com  A record which  will pont to the same IP as 
sip1.sss.domain.com, could this be causing any problems for opensips 
(there will be two A record two the same IP, one with the same name as 
dns srv record)?


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Re: [OpenSIPS-Users] Interaction between OpenSIPS as UAC and real UAC

2014-02-11 Thread Bogdan-Andrei Iancu 

Hello,

Use the dialog based topology hiding
http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id296001

when forwarding the INVITE to Asterisk (in OpenSIPS).

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 11.02.2014 10:12, Александр Пучков wrote:


10.02.2014 13:24, Bogdan-Andrei Iancu пишет:

Hello,

In this scenario:

Astеrisk <-- 3 -- OpenSIPS <-- 4 -- UAC

What SIP requests the UAC is sending ? REGISTER ? INVITES ?

Hello!

UAC registered to Opensips, Opensips registered as UAC on Asterisk. 
When the INVITE request comes from the UAC, Opensips need to sent 
INVITE to Asterisk as UAC.


Note that the UAC does not know about Asterisk and Asterisk does not 
know about UAC. Asterisk know about only Opensips.


It is necessary to any UAC registered on Opensips could imagine how 
extension on Asterisk, without changing the configuration of UAC.


Thank you.




Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 20.01.2014 13:05, Александр Пучков wrote:

Hi!

OpenSIPS version 1.8.

We have the following diagram:

PSTN<-- 1 --> OpenSIPS<-- 2 --> Astеrisk<-- 3 --> UAC
192.168.1.1 <-- 1 --> 192.168.1.2 <-- 2 --> 192.168.1.3 <-- 3 --> 
192.168.1.*


I would like to make the following scheme:

PSTN<-- 1 --> OpenSIPS<-- 2 --> Astеrisk<-- 3 --> 
OpenSIPS<-- 4 --> UAC
192.168.1.1 <-- 1 --> 192.168.1.2 <-- 2 --> 192.168.1.3 <-- 3 --> 
192.168.1.2 <-- 4 --> 192.168.1.*


Interestingly the following schema fragment:
Astеrisk <-- 3 --> OpenSIPS<-- 4 --> UAC

Here OpenSIPS need to increase control over the services for UAC. 
UAC should not know about the existence of Asterisk and UAC must be 
registered on the server OpenSIPS, and any SIP request OpenSIPS 
should redirect to Asterisk.


I tried to use the module in UAC_REGISTRANT OpenSIPS, it works fine 
in the direction:


Astеrisk -- 3 --> OpenSIPS -- 4 --> UAC

But how to implement the scheme in the direction:

Astеrisk <-- 3 -- OpenSIPS <-- 4 -- UAC

I do not really imagine. Please tell me how it can be implemented.

Thank!


*Александр Пучков,
Системный администратор,
Тел.: +7(496) 569-24-24 доб.тел. 255;
e-mail: mai...@poig.ru;
*ООО "ПОИГ" (Интернет-провайдер г. Щелково)
Факс: +7(496) 569-24-24 доб. 103;
Адрес: 141108, М.О., г.Щелково, Пролетарский пр., д.11;
WEB: http://www.schelkovo-net.ru



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*Александр Пучков,
Системный администратор,
Тел.: +7(496) 569-24-24 доб.тел. 255;
*ООО "ПОИГ" (Интернет-провайдер г. Щелково)
Факс: +7(496) 569-24-24 доб. 103;
Адрес: 141108, М.О., г.Щелково, Пролетарский пр., д.11;
WEB: http://www.schelkovo-net.ru



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Re: [OpenSIPS-Users] Authentication using Username, Password, IP address or Just IP address

2014-02-11 Thread Bogdan-Andrei Iancu 

Hello Alec,

I'm happy you managed to make it work.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 11.02.2014 06:18, Alec Doran-Twyford wrote:

Hi Bogdan,

Thanks for the help it working a charm now the code look like this for 
anyone else who is interested:


if(check_source_address("0")) {
xlog("IP Allow Routing");
} else {
xlog("authentication required for call from $si");
if (!is_method("REGISTER")) {
# EC - auth
if (!proxy_authorize("", "subscriber")) {
xlog("proxy challenge!");
proxy_challenge("", "0");  # Realm 
will be autogenerated

exit;
}
if (!db_check_from()) {
 xlog("forbidden!");
 sl_send_reply("403","Forbidden auth ID");
 exit;
}
consume_credentials();
}
 else {
if (!www_authorize("", "subscriber")) {
xlog("www challenge!");
www_challenge("", "0");  # Realm will 
be autogenerated

exit;
}

if (!db_check_from()) {
 xlog("forbidden!");
 sl_send_reply("403","Forbidden auth ID");
 exit;
}

if (!save("location")) {
xlog("failed to save location!");
sl_reply_error();
}


xlog("registered - $from");
exit;
}
}


and currently add IP like this in to the database
insert into opensips.address (ip, port) VALUES ("192.168.168.55", 5060);

and then use the command to refresh the current cache of entries
opensipsctl fifo address_reload

Alec Doran-Twyford

| Junior Support Enginner for IVSTel
| E-mail: a.dorantwyf...@ivstel.com  
| Phone: +61 2 9288 8890 |




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Re: [OpenSIPS-Users] Interaction between OpenSIPS as UAC and real UAC

2014-02-11 Thread Александр Пучков 


10.02.2014 13:24, Bogdan-Andrei Iancu пишет:

Hello,

In this scenario:

Astеrisk <-- 3 -- OpenSIPS <-- 4 -- UAC

What SIP requests the UAC is sending ? REGISTER ? INVITES ?

Hello!

UAC registered to Opensips, Opensips registered as UAC on Asterisk. When 
the INVITE request comes from the UAC, Opensips need to sent INVITE to 
Asterisk as UAC.


Note that the UAC does not know about Asterisk and Asterisk does not 
know about UAC. Asterisk know about only Opensips.


It is necessary to any UAC registered on Opensips could imagine how 
extension on Asterisk, without changing the configuration of UAC.


Thank you.




Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 20.01.2014 13:05, Александр Пучков wrote:

Hi!

OpenSIPS version 1.8.

We have the following diagram:

PSTN<-- 1 --> OpenSIPS<-- 2 --> Astеrisk<-- 3 --> UAC
192.168.1.1 <-- 1 --> 192.168.1.2 <-- 2 --> 192.168.1.3 <-- 3 --> 
192.168.1.*


I would like to make the following scheme:

PSTN<-- 1 --> OpenSIPS<-- 2 --> Astеrisk<-- 3 --> 
OpenSIPS<-- 4 --> UAC
192.168.1.1 <-- 1 --> 192.168.1.2 <-- 2 --> 192.168.1.3 <-- 3 --> 
192.168.1.2 <-- 4 --> 192.168.1.*


Interestingly the following schema fragment:
Astеrisk <-- 3 --> OpenSIPS<-- 4 --> UAC

Here OpenSIPS need to increase control over the services for UAC. UAC 
should not know about the existence of Asterisk and UAC must be 
registered on the server OpenSIPS, and any SIP request OpenSIPS 
should redirect to Asterisk.


I tried to use the module in UAC_REGISTRANT OpenSIPS, it works fine 
in the direction:


Astеrisk -- 3 --> OpenSIPS -- 4 --> UAC

But how to implement the scheme in the direction:

Astеrisk <-- 3 -- OpenSIPS <-- 4 -- UAC

I do not really imagine. Please tell me how it can be implemented.

Thank!


*Александр Пучков,
Системный администратор,
Тел.: +7(496) 569-24-24 доб.тел. 255;
e-mail: mai...@poig.ru;
*ООО "ПОИГ" (Интернет-провайдер г. Щелково)
Факс: +7(496) 569-24-24 доб. 103;
Адрес: 141108, М.О., г.Щелково, Пролетарский пр., д.11;
WEB: http://www.schelkovo-net.ru



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--
С уважением,
Александр Пучков,
Системный администратор,
Тел.:  +7(496) 569-24-24 доб.тел. 255;
ООО "ПОИГ" (Интернет-провайдер г. Щелково)
Факс:  +7(496) 569-24-24 доб. 103;
Адрес: 141108, М.О., г.Щелково, Пролетарский пр., д.11;
WEB:   http://www.schelkovo-net.ru
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