Re: [OpenSIPS-Users] forwarding custom variables

[Stefano Pisani]

Yes. It's possible.
The "variable" must be in a custom header.

Ciao
s


Il 15/03/2017 19:41, Roberto Cantalapiedra ha scritto:

Hi,

I would like to know if it is possible this scenario:


- Caller initialize the call and sending specific variable to opensips.
- opensips match one of those variable and route the call to a provider.
- if the call is answered the call is forwarded to an asterisk server 
with all the variables that the caller sent in the initialization 
request to the proxy.


Thanks in advance,






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Re: [OpenSIPS-Users] how to catch hangup event inside opensips

[Stefano Pisani]

please explain better

Il 25 Gennaio 2017 13:24:57 CET, Khalil Khamlichi  
ha scritto:
>there is one more question we still struggling with, how to run a route
>at
>each hangup, we never found any way to do it inside opensips. right now
>we
>catching the event upon cdr insert in db.

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Re: [OpenSIPS-Users] Registrar with IP authentication - selecting variables from DB

[Stefano Pisani]

Use 0.0.0.0/0 for those without IP filter.

s

Il 13/01/2017 12:09, maatohewetbi ha scritto:

I think You don't understand. My Opensips should work in this scenario:

1. When user wants to register, I have to check whether his sip login is in
address table (which can be stored in context_info for example). If it is
there then check IP, which is in this record, for this sip login. If this IP
is the same as real IP of this user (from $si value), the user can be
registered. If this IP is not the same - user can't be registered.
2. When user wants to register, but his sip login doesn't exist in address
table - this user can be registered without IP checking.

So I have to check IP, only for users, whom login is in address table.
Others, can registered without checking. If I use one of function in
permission module I always have to check IP. I have to check login first,
then IP. I hope that now everything is clear.



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Re: [OpenSIPS-Users] tls_mgm

[Stefano Pisani]

Hi Răzvan,
there is not openssl 1.0.2h available for ubuntu 16.04 so I installed it 
from the tarball and fixed the symlink to the correct libraries libssl 
and libcrypto. That's all.


Thanks.
Stefano

Il 20/08/2016 22:28, Răzvan Crainea ha scritto:

Hi, Stefano!

Sorry, I missed that email. So what was your solution to solve the 
conflict? Deployed a custom deb?


Best regards,

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

On 08/20/2016 11:14 PM, Stefano Pisani wrote:

Hello Răzvan,
in my previous email I told you that I have upgraded openssl to 1.0.2h
then the error was different.
After that I realized that there was a conflict between ubuntu openssl
package and new openssl.
Finally, after fixed that, tls_mgm module is working properly.

I'm using ubuntu 16.04 LTS

Thanks
Stefano

Il 20/08/2016 22:09, Răzvan Crainea ha scritto:

Hello!

Is there any chance you could upgrade your openssl library? This
version has a known bug.
Also, could you tell us what OS you're running, perhaps we can manage
to replicate this and track it down.

Best regards.

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

On 08/20/2016 10:56 PM, Venkatesh Macha wrote:


Hi all,

 I am trying to install OpenSIPS with WSS support. But i am getting
following error on OpenSIPS Startup.

ERROR:tls_mgm:mod_init: unable to set the memory allocation functions
Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 1.0.1e-fips,
as this
version is known to be broken; if so, you need to upgrade or
downgrade to a
different openssl version !!
Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
ERROR:tls_mgm:mod_init: extra: malloc=0x7f7490dd25f8/0x7f749d13c550
realloc=0x7f7490dd2624/0x7f749d13cc40 
free=0x7f7490dd265c/0x7f749d13ca70

version=OpenSSL 1.0.2g-fips  1 Mar 2016
Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
ERROR:core:init_mod:
failed to initialize module tls_mgm
Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]: 
ERROR:core:main:

error while initializing modules

My OpenSIPS version:
 opensips -V
version: opensips 2.3.0-dev (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
QM_MALLOC,
DBG_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.
git revision: 326a1c6

My OpenSSL Version:
openssl version
OpenSSL 1.0.2g-fips  1 Mar 2016

I am using Ubuntu 16.04 LTS.

Thank you in advance.



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Re: [OpenSIPS-Users] tls_mgm

[Stefano Pisani]

Hello Răzvan,
in my previous email I told you that I have upgraded openssl to 1.0.2h 
then the error was different.
After that I realized that there was a conflict between ubuntu openssl 
package and new openssl.

Finally, after fixed that, tls_mgm module is working properly.

I'm using ubuntu 16.04 LTS

Thanks
Stefano

Il 20/08/2016 22:09, Răzvan Crainea ha scritto:

Hello!

Is there any chance you could upgrade your openssl library? This 
version has a known bug.
Also, could you tell us what OS you're running, perhaps we can manage 
to replicate this and track it down.


Best regards.

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

On 08/20/2016 10:56 PM, Venkatesh Macha wrote:


Hi all,

 I am trying to install OpenSIPS with WSS support. But i am getting
following error on OpenSIPS Startup.

ERROR:tls_mgm:mod_init: unable to set the memory allocation functions
Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 1.0.1e-fips, 
as this
version is known to be broken; if so, you need to upgrade or 
downgrade to a

different openssl version !!
Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]:
ERROR:tls_mgm:mod_init: extra: malloc=0x7f7490dd25f8/0x7f749d13c550
realloc=0x7f7490dd2624/0x7f749d13cc40 free=0x7f7490dd265c/0x7f749d13ca70
version=OpenSSL 1.0.2g-fips  1 Mar 2016
Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]: 
ERROR:core:init_mod:

failed to initialize module tls_mgm
Aug 20 19:46:12 webrtc /usr/local/sbin/opensips[20545]: ERROR:core:main:
error while initializing modules

My OpenSIPS version:
 opensips -V
version: opensips 2.3.0-dev (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
QM_MALLOC,

DBG_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.
git revision: 326a1c6

My OpenSSL Version:
openssl version
OpenSSL 1.0.2g-fips  1 Mar 2016

I am using Ubuntu 16.04 LTS.

Thank you in advance.



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Re: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory allocation functions

[Stefano Pisani]

I upgraded openssl version to 1.0.2h.

The error now is different.

Aug 17 12:46:03 [5229] ERROR:core:sr_load_module: could not open module 
: /lib/x86_64-l 
inux-gnu/libssl.so.1.0.0: undefined symbol: EVP_idea_cbc


It looks like it miss a cipher.

Stefano


Il 17/08/2016 16:09, Răzvan Crainea ha scritto:

Can you try to downgrade it[1]? At least just for testing purposes.
BTW, I am using 1.0.2h and it works fine.

[1] 
http://askubuntu.com/questions/292314/how-to-downgrade-packages-on-ubuntu


Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/17/2016 04:07 PM, Stefano Pisani wrote:
It seems to be the last available version for ubuntu 16.04 LTS. What 
can I do?


#apt-get install openssl
Reading package lists... Done
Building dependency tree
Reading state information... Done
openssl is already the newest version (1.0.2g-1ubuntu4.1)

Thanks
Stefano




Il 17/08/2016 14:52, Răzvan Crainea ha scritto:
Seems to be a similar problem to the 1.0.1e-fips library. Could you 
try to upgrade the openssl package?


Thanks,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/17/2016 12:48 PM, Stefano Pisani wrote:


I miss the last two lines

(gdb) info line *0x77672a70
Line 2935 of "malloc.c" starts at address 0x77672a70 
<__GI___libc_free> and ends at 0x77672a7a <__GI___libc_free+10>.

(gdb) info line *0x7161b60e
Line 101 of "tls.h" starts at address 0x7161b60e  and 
ends at 0x7161b61a .


I hope this helps.

Aug 17 05:41:30 [10867] INFO:tls_mgm:mod_init: initializing TLS 
protocol
Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: unable to set the 
memory allocation functions
Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: NOTE: check if you 
have openssl 1.0.1e-fips, as this version is known to be broken; if 
so, you need to upgrade or downgrade to a different openssl version !!
Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: extra: 
malloc=0x7161b5aa/0x77672550 
realloc=0x7161b5d6/0x77672c40 
free=0x7161b60e/0x77672a70 version=OpenSSL 1.0.2g-fips  1 
Mar 2016
Aug 17 05:41:30 [10867] ERROR:core:init_mod: failed to initialize 
module tls_mgm
Aug 17 05:41:30 [10867] ERROR:core:main: error while initializing 
modules

(gdb) info line *0x77672550
Line 2903 of "malloc.c" starts at address 0x77672550 
<__GI___libc_malloc> and ends at 0x77672556 <__GI___libc_malloc+6>.

(gdb) info line *0x77672c40
Line 2975 of "malloc.c" starts at address 0x77672c40 
<__GI___libc_realloc> and ends at 0x77672c54 
<__GI___libc_realloc+20>.

(gdb) info line *0x7161b5aa
Line 89 of "tls.h" starts at address 0x7161b5aa  and 
ends at 0x7161b5b6 .

(gdb) info line *0x7161b5d6
Line 95 of "tls.h" starts at address 0x7161b5d6  
and ends at 0x7161b5e6 .



Il 17/08/2016 11:16, Răzvan Crainea ha scritto:

Can you run 'gdb ./opensips' and run the following commands:
info line *0x7f0ce10d8a70
info line *0x7f0ce10d8550

Thanks,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/17/2016 12:06 PM, Stefano Pisani wrote:


Hi Răzvan.

These are the new logs. I hope it helps to fix this.

Aug 17 05:04:40 [3997] INFO:tls_mgm:mod_init: initializing TLS 
protocol
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: unable to set the 
memory allocation functions
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: NOTE: check if you 
have openssl 1.0.1e-fips, as this version is known to be broken; 
if so, you need to upgrade or downgrade to a different openssl 
version !!
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: extra: 
malloc=0x7f0cdb0845c8/0x7f0ce10d8550 
realloc=0x7f0cdb0845f4/0x7f0ce10d8c40 
free=0x7f0cdb08462c/0x7f0ce10d8a70 version=OpenSSL 1.0.2g-fips  1 
Mar 2016
Aug 17 05:04:40 [3997] ERROR:core:init_mod: failed to initialize 
module tls_mgm
Aug 17 05:04:40 [3997] ERROR:core:main: error while initializing 
modules


Thanks.
Stefano

Il 17/08/2016 09:53, Răzvan Crainea ha scritto:


Hi, Stefano!

Apologies, I did not push the changes. Could you please re-clone 
and try again?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/16/2016 07:26 PM, Stefano Pisani wrote:

I have this result.

# ./opensips -V
version: opensips 2.3.0-dev (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, 
PKG_MALLOC, QM_MALLOC, DBG_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.
git revision: 058991d
main.c compiled on 11:50:11 Aug 16 2016 with gcc 5.4.0



Il 16/08/2016 18:23, Răzvan Crainea ha scritto:


Hi, Stefano!

I don't think you are using the latest sources. Either you 
have not installed them, or you are using a wrong path. Could 
you please double-check?


Best regards,

Răzvan Crainea
OpenSIPS Solutions

Re: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory allocation functions

[Stefano Pisani]

It seems to be the last available version for ubuntu 16.04 LTS. What can 
I do?


#apt-get install openssl
Reading package lists... Done
Building dependency tree
Reading state information... Done
openssl is already the newest version (1.0.2g-1ubuntu4.1)

Thanks
Stefano




Il 17/08/2016 14:52, Răzvan Crainea ha scritto:
Seems to be a similar problem to the 1.0.1e-fips library. Could you 
try to upgrade the openssl package?


Thanks,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/17/2016 12:48 PM, Stefano Pisani wrote:


I miss the last two lines

(gdb) info line *0x77672a70
Line 2935 of "malloc.c" starts at address 0x77672a70 
<__GI___libc_free> and ends at 0x77672a7a <__GI___libc_free+10>.

(gdb) info line *0x7161b60e
Line 101 of "tls.h" starts at address 0x7161b60e  and 
ends at 0x7161b61a .


I hope this helps.

Aug 17 05:41:30 [10867] INFO:tls_mgm:mod_init: initializing TLS protocol
Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: unable to set the 
memory allocation functions
Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: NOTE: check if you 
have openssl 1.0.1e-fips, as this version is known to be broken; if 
so, you need to upgrade or downgrade to a different openssl version !!
Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: extra: 
malloc=0x7161b5aa/0x77672550 
realloc=0x7161b5d6/0x77672c40 
free=0x7161b60e/0x77672a70 version=OpenSSL 1.0.2g-fips  1 Mar 
2016
Aug 17 05:41:30 [10867] ERROR:core:init_mod: failed to initialize 
module tls_mgm

Aug 17 05:41:30 [10867] ERROR:core:main: error while initializing modules
(gdb) info line *0x77672550
Line 2903 of "malloc.c" starts at address 0x77672550 
<__GI___libc_malloc> and ends at 0x77672556 <__GI___libc_malloc+6>.

(gdb) info line *0x77672c40
Line 2975 of "malloc.c" starts at address 0x77672c40 
<__GI___libc_realloc> and ends at 0x77672c54 
<__GI___libc_realloc+20>.

(gdb) info line *0x7161b5aa
Line 89 of "tls.h" starts at address 0x7161b5aa  and 
ends at 0x7161b5b6 .

(gdb) info line *0x7161b5d6
Line 95 of "tls.h" starts at address 0x7161b5d6  and 
ends at 0x7161b5e6 .



Il 17/08/2016 11:16, Răzvan Crainea ha scritto:

Can you run 'gdb ./opensips' and run the following commands:
info line *0x7f0ce10d8a70
info line *0x7f0ce10d8550

Thanks,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/17/2016 12:06 PM, Stefano Pisani wrote:


Hi Răzvan.

These are the new logs. I hope it helps to fix this.

Aug 17 05:04:40 [3997] INFO:tls_mgm:mod_init: initializing TLS protocol
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: unable to set the 
memory allocation functions
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: NOTE: check if you 
have openssl 1.0.1e-fips, as this version is known to be broken; if 
so, you need to upgrade or downgrade to a different openssl version !!
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: extra: 
malloc=0x7f0cdb0845c8/0x7f0ce10d8550 
realloc=0x7f0cdb0845f4/0x7f0ce10d8c40 
free=0x7f0cdb08462c/0x7f0ce10d8a70 version=OpenSSL 1.0.2g-fips  1 
Mar 2016
Aug 17 05:04:40 [3997] ERROR:core:init_mod: failed to initialize 
module tls_mgm
Aug 17 05:04:40 [3997] ERROR:core:main: error while initializing 
modules


Thanks.
Stefano

Il 17/08/2016 09:53, Răzvan Crainea ha scritto:


Hi, Stefano!

Apologies, I did not push the changes. Could you please re-clone 
and try again?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/16/2016 07:26 PM, Stefano Pisani wrote:

I have this result.

# ./opensips -V
version: opensips 2.3.0-dev (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
QM_MALLOC, DBG_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.
git revision: 058991d
main.c compiled on 11:50:11 Aug 16 2016 with gcc 5.4.0



Il 16/08/2016 18:23, Răzvan Crainea ha scritto:


Hi, Stefano!

I don't think you are using the latest sources. Either you have 
not installed them, or you are using a wrong path. Could you 
please double-check?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/16/2016 06:56 PM, Stefano Pisani wrote:


I cloned the source from

|git clone https://github.com/OpenSIPS/opensips.git 
opensips_head the error appears the same Aug 16 11:55:14 
[25630] INFO:tls_mgm:mod_init: initializing TLS protocol Aug 16 
11:55:14 [25630] ERROR:tls_mgm:mod_init: unable to set the 
memory allocation functions Aug 16 11:55:14 [25630] 
ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 
1.0.1e-fips, as this version is know to be broken; if so, you 
need to upgrade or downgrade to a differen openssl version !! 
Aug 16 11:55:14 [25630] ERROR:core:init_mod: failed to 

Re: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory allocation functions

[Stefano Pisani]

I miss the last two lines

(gdb) info line *0x77672a70
Line 2935 of "malloc.c" starts at address 0x77672a70 
<__GI___libc_free> and ends at 0x77672a7a <__GI___libc_free+10>.

(gdb) info line *0x7161b60e
Line 101 of "tls.h" starts at address 0x7161b60e  and ends 
at 0x7161b61a .


I hope this helps.

Aug 17 05:41:30 [10867] INFO:tls_mgm:mod_init: initializing TLS protocol
Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: unable to set the memory 
allocation functions
Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: NOTE: check if you have 
openssl 1.0.1e-fips, as this version is known to be broken; if so, you 
need to upgrade or downgrade to a different openssl version !!
Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: extra: 
malloc=0x7161b5aa/0x77672550 
realloc=0x7161b5d6/0x77672c40 free=0x7161b60e/0x77672a70 
version=OpenSSL 1.0.2g-fips  1 Mar 2016
Aug 17 05:41:30 [10867] ERROR:core:init_mod: failed to initialize module 
tls_mgm

Aug 17 05:41:30 [10867] ERROR:core:main: error while initializing modules
(gdb) info line *0x77672550
Line 2903 of "malloc.c" starts at address 0x77672550 
<__GI___libc_malloc> and ends at 0x77672556 <__GI___libc_malloc+6>.

(gdb) info line *0x77672c40
Line 2975 of "malloc.c" starts at address 0x77672c40 
<__GI___libc_realloc> and ends at 0x77672c54 <__GI___libc_realloc+20>.

(gdb) info line *0x7161b5aa
Line 89 of "tls.h" starts at address 0x7161b5aa  and ends 
at 0x7161b5b6 .

(gdb) info line *0x7161b5d6
Line 95 of "tls.h" starts at address 0x7161b5d6  and 
ends at 0x7161b5e6 .



Il 17/08/2016 11:16, Răzvan Crainea ha scritto:

Can you run 'gdb ./opensips' and run the following commands:
info line *0x7f0ce10d8a70
info line *0x7f0ce10d8550

Thanks,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/17/2016 12:06 PM, Stefano Pisani wrote:


Hi Răzvan.

These are the new logs. I hope it helps to fix this.

Aug 17 05:04:40 [3997] INFO:tls_mgm:mod_init: initializing TLS protocol
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: unable to set the 
memory allocation functions
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: NOTE: check if you 
have openssl 1.0.1e-fips, as this version is known to be broken; if 
so, you need to upgrade or downgrade to a different openssl version !!
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: extra: 
malloc=0x7f0cdb0845c8/0x7f0ce10d8550 
realloc=0x7f0cdb0845f4/0x7f0ce10d8c40 
free=0x7f0cdb08462c/0x7f0ce10d8a70 version=OpenSSL 1.0.2g-fips  1 Mar 
2016
Aug 17 05:04:40 [3997] ERROR:core:init_mod: failed to initialize 
module tls_mgm

Aug 17 05:04:40 [3997] ERROR:core:main: error while initializing modules

Thanks.
Stefano

Il 17/08/2016 09:53, Răzvan Crainea ha scritto:


Hi, Stefano!

Apologies, I did not push the changes. Could you please re-clone and 
try again?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/16/2016 07:26 PM, Stefano Pisani wrote:

I have this result.

# ./opensips -V
version: opensips 2.3.0-dev (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
QM_MALLOC, DBG_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.
git revision: 058991d
main.c compiled on 11:50:11 Aug 16 2016 with gcc 5.4.0



Il 16/08/2016 18:23, Răzvan Crainea ha scritto:


Hi, Stefano!

I don't think you are using the latest sources. Either you have 
not installed them, or you are using a wrong path. Could you 
please double-check?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/16/2016 06:56 PM, Stefano Pisani wrote:


I cloned the source from

|git clone https://github.com/OpenSIPS/opensips.git opensips_head 
the error appears the same Aug 16 11:55:14 [25630] 
INFO:tls_mgm:mod_init: initializing TLS protocol Aug 16 11:55:14 
[25630] ERROR:tls_mgm:mod_init: unable to set the memory 
allocation functions Aug 16 11:55:14 [25630] 
ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 
1.0.1e-fips, as this version is know to be broken; if so, you 
need to upgrade or downgrade to a differen openssl version !! Aug 
16 11:55:14 [25630] ERROR:core:init_mod: failed to initialize 
module tls_mgm Aug 16 11:55:14 [25630] ERROR:core:main: error 
while initializing modules Aug 16 11:55:14 [25630] 
INFO:core:cleanup: cleanup Aug 16 11:55:14 [25630] 
DBG:dispatcher:destroy: destroying module ... Aug 16 11:55:14 
[25630] DBG:tm:tm_shutdown: tm_shutdown : start Aug 16 11:55:14 
[25630] DBG:tm:unlink_timer_lists: emptying DELETE list for set 0 
Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: emptying hash table 
Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: releasing timers Aug 
16 11:55:14 [25630] DBG:tm:tm_shutdown: rem

Re: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory allocation functions

[Stefano Pisani]

I hope this helps.

Aug 17 05:41:30 [10867] INFO:tls_mgm:mod_init: initializing TLS protocol
Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: unable to set the memory 
allocation functions
Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: NOTE: check if you have 
openssl 1.0.1e-fips, as this version is known to be broken; if so, you 
need to upgrade or downgrade to a different openssl version !!
Aug 17 05:41:30 [10867] ERROR:tls_mgm:mod_init: extra: 
malloc=0x7161b5aa/0x77672550 
realloc=0x7161b5d6/0x77672c40 free=0x7161b60e/0x77672a70 
version=OpenSSL 1.0.2g-fips  1 Mar 2016
Aug 17 05:41:30 [10867] ERROR:core:init_mod: failed to initialize module 
tls_mgm

Aug 17 05:41:30 [10867] ERROR:core:main: error while initializing modules
(gdb) info line *0x77672550
Line 2903 of "malloc.c" starts at address 0x77672550 
<__GI___libc_malloc> and ends at 0x77672556 <__GI___libc_malloc+6>.

(gdb) info line *0x77672c40
Line 2975 of "malloc.c" starts at address 0x77672c40 
<__GI___libc_realloc> and ends at 0x77672c54 <__GI___libc_realloc+20>.

(gdb) info line *0x7161b5aa
Line 89 of "tls.h" starts at address 0x7161b5aa  and ends 
at 0x7161b5b6 .

(gdb) info line *0x7161b5d6
Line 95 of "tls.h" starts at address 0x7161b5d6  and 
ends at 0x7161b5e6 .



Il 17/08/2016 11:16, Răzvan Crainea ha scritto:

Can you run 'gdb ./opensips' and run the following commands:
info line *0x7f0ce10d8a70
info line *0x7f0ce10d8550

Thanks,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/17/2016 12:06 PM, Stefano Pisani wrote:


Hi Răzvan.

These are the new logs. I hope it helps to fix this.

Aug 17 05:04:40 [3997] INFO:tls_mgm:mod_init: initializing TLS protocol
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: unable to set the 
memory allocation functions
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: NOTE: check if you 
have openssl 1.0.1e-fips, as this version is known to be broken; if 
so, you need to upgrade or downgrade to a different openssl version !!
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: extra: 
malloc=0x7f0cdb0845c8/0x7f0ce10d8550 
realloc=0x7f0cdb0845f4/0x7f0ce10d8c40 
free=0x7f0cdb08462c/0x7f0ce10d8a70 version=OpenSSL 1.0.2g-fips  1 Mar 
2016
Aug 17 05:04:40 [3997] ERROR:core:init_mod: failed to initialize 
module tls_mgm

Aug 17 05:04:40 [3997] ERROR:core:main: error while initializing modules

Thanks.
Stefano

Il 17/08/2016 09:53, Răzvan Crainea ha scritto:


Hi, Stefano!

Apologies, I did not push the changes. Could you please re-clone and 
try again?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/16/2016 07:26 PM, Stefano Pisani wrote:

I have this result.

# ./opensips -V
version: opensips 2.3.0-dev (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
QM_MALLOC, DBG_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.
git revision: 058991d
main.c compiled on 11:50:11 Aug 16 2016 with gcc 5.4.0



Il 16/08/2016 18:23, Răzvan Crainea ha scritto:


Hi, Stefano!

I don't think you are using the latest sources. Either you have 
not installed them, or you are using a wrong path. Could you 
please double-check?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/16/2016 06:56 PM, Stefano Pisani wrote:


I cloned the source from

|git clone https://github.com/OpenSIPS/opensips.git opensips_head 
the error appears the same Aug 16 11:55:14 [25630] 
INFO:tls_mgm:mod_init: initializing TLS protocol Aug 16 11:55:14 
[25630] ERROR:tls_mgm:mod_init: unable to set the memory 
allocation functions Aug 16 11:55:14 [25630] 
ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 
1.0.1e-fips, as this version is know to be broken; if so, you 
need to upgrade or downgrade to a differen openssl version !! Aug 
16 11:55:14 [25630] ERROR:core:init_mod: failed to initialize 
module tls_mgm Aug 16 11:55:14 [25630] ERROR:core:main: error 
while initializing modules Aug 16 11:55:14 [25630] 
INFO:core:cleanup: cleanup Aug 16 11:55:14 [25630] 
DBG:dispatcher:destroy: destroying module ... Aug 16 11:55:14 
[25630] DBG:tm:tm_shutdown: tm_shutdown : start Aug 16 11:55:14 
[25630] DBG:tm:unlink_timer_lists: emptying DELETE list for set 0 
Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: emptying hash table 
Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: releasing timers Aug 
16 11:55:14 [25630] DBG:tm:tm_shutdown: removing semaphores Aug 
16 11:55:14 [25630] DBG:tm:tm_shutdown: destroying callback lists 
Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown : done 
Aug 16 11:55:14 [25630] DBG:core:shm_mem_destroy: destroying the 
shared memory lock Aug 16 11:55:14 [25630] NOTICE:core:main: 
Exiting Thanks Stefano |


Il 16/08/2016 17:18, Răzv

Re: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory allocation functions

[Stefano Pisani]

# gdb ./opensips
GNU gdb (Ubuntu 7.11.1-0ubuntu1~16.04) 7.11.1
Copyright (C) 2016 Free Software Foundation, Inc.
License GPLv3+: GNU GPL version 3 or later 
<http://gnu.org/licenses/gpl.html>

This is free software: you are free to change and redistribute it.
There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
and "show warranty" for details.
This GDB was configured as "x86_64-linux-gnu".
Type "show configuration" for configuration details.
For bug reporting instructions, please see:
<http://www.gnu.org/software/gdb/bugs/>.
Find the GDB manual and other documentation resources online at:
<http://www.gnu.org/software/gdb/documentation/>.
For help, type "help".
Type "apropos word" to search for commands related to "word"...
Reading symbols from ./opensips...done.
(gdb) info line *0x7f0ce10d8a70
No line number information available for address 0x7f0ce10d8a70
(gdb) info line *0x7f0ce10d8550
No line number information available for address 0x7f0ce10d8550
(gdb)

Thanks
Stefano


Il 17/08/2016 11:16, Răzvan Crainea ha scritto:

Can you run 'gdb ./opensips' and run the following commands:
info line *0x7f0ce10d8a70
info line *0x7f0ce10d8550

Thanks,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/17/2016 12:06 PM, Stefano Pisani wrote:


Hi Răzvan.

These are the new logs. I hope it helps to fix this.

Aug 17 05:04:40 [3997] INFO:tls_mgm:mod_init: initializing TLS protocol
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: unable to set the 
memory allocation functions
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: NOTE: check if you 
have openssl 1.0.1e-fips, as this version is known to be broken; if 
so, you need to upgrade or downgrade to a different openssl version !!
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: extra: 
malloc=0x7f0cdb0845c8/0x7f0ce10d8550 
realloc=0x7f0cdb0845f4/0x7f0ce10d8c40 
free=0x7f0cdb08462c/0x7f0ce10d8a70 version=OpenSSL 1.0.2g-fips  1 Mar 
2016
Aug 17 05:04:40 [3997] ERROR:core:init_mod: failed to initialize 
module tls_mgm

Aug 17 05:04:40 [3997] ERROR:core:main: error while initializing modules

Thanks.
Stefano

Il 17/08/2016 09:53, Răzvan Crainea ha scritto:


Hi, Stefano!

Apologies, I did not push the changes. Could you please re-clone and 
try again?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/16/2016 07:26 PM, Stefano Pisani wrote:

I have this result.

# ./opensips -V
version: opensips 2.3.0-dev (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
QM_MALLOC, DBG_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.
git revision: 058991d
main.c compiled on 11:50:11 Aug 16 2016 with gcc 5.4.0



Il 16/08/2016 18:23, Răzvan Crainea ha scritto:


Hi, Stefano!

I don't think you are using the latest sources. Either you have 
not installed them, or you are using a wrong path. Could you 
please double-check?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/16/2016 06:56 PM, Stefano Pisani wrote:


I cloned the source from

|git clone https://github.com/OpenSIPS/opensips.git opensips_head 
the error appears the same Aug 16 11:55:14 [25630] 
INFO:tls_mgm:mod_init: initializing TLS protocol Aug 16 11:55:14 
[25630] ERROR:tls_mgm:mod_init: unable to set the memory 
allocation functions Aug 16 11:55:14 [25630] 
ERROR:tls_mgm:mod_init: NOTE: check if you have openssl 
1.0.1e-fips, as this version is know to be broken; if so, you 
need to upgrade or downgrade to a differen openssl version !! Aug 
16 11:55:14 [25630] ERROR:core:init_mod: failed to initialize 
module tls_mgm Aug 16 11:55:14 [25630] ERROR:core:main: error 
while initializing modules Aug 16 11:55:14 [25630] 
INFO:core:cleanup: cleanup Aug 16 11:55:14 [25630] 
DBG:dispatcher:destroy: destroying module ... Aug 16 11:55:14 
[25630] DBG:tm:tm_shutdown: tm_shutdown : start Aug 16 11:55:14 
[25630] DBG:tm:unlink_timer_lists: emptying DELETE list for set 0 
Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: emptying hash table 
Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: releasing timers Aug 
16 11:55:14 [25630] DBG:tm:tm_shutdown: removing semaphores Aug 
16 11:55:14 [25630] DBG:tm:tm_shutdown: destroying callback lists 
Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown : done 
Aug 16 11:55:14 [25630] DBG:core:shm_mem_destroy: destroying the 
shared memory lock Aug 16 11:55:14 [25630] NOTICE:core:main: 
Exiting Thanks Stefano |


Il 16/08/2016 17:18, Răzvan Crainea ha scritto:

Stefano!

I've added some extra debugging in the master branch that might 
help you debug this. Could you please take the latest git 
version/deb and try to run again?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 08/16/2016

Re: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory allocation functions

[Stefano Pisani]

Hi Răzvan.

These are the new logs. I hope it helps to fix this.

Aug 17 05:04:40 [3997] INFO:tls_mgm:mod_init: initializing TLS protocol
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: unable to set the memory 
allocation functions
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: NOTE: check if you have 
openssl 1.0.1e-fips, as this version is known to be broken; if so, you 
need to upgrade or downgrade to a different openssl version !!
Aug 17 05:04:40 [3997] ERROR:tls_mgm:mod_init: extra: 
malloc=0x7f0cdb0845c8/0x7f0ce10d8550 
realloc=0x7f0cdb0845f4/0x7f0ce10d8c40 free=0x7f0cdb08462c/0x7f0ce10d8a70 
version=OpenSSL 1.0.2g-fips  1 Mar 2016
Aug 17 05:04:40 [3997] ERROR:core:init_mod: failed to initialize module 
tls_mgm

Aug 17 05:04:40 [3997] ERROR:core:main: error while initializing modules

Thanks.
Stefano

Il 17/08/2016 09:53, Răzvan Crainea ha scritto:


Hi, Stefano!

Apologies, I did not push the changes. Could you please re-clone and 
try again?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/16/2016 07:26 PM, Stefano Pisani wrote:

I have this result.

# ./opensips -V
version: opensips 2.3.0-dev (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
QM_MALLOC, DBG_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.
git revision: 058991d
main.c compiled on 11:50:11 Aug 16 2016 with gcc 5.4.0



Il 16/08/2016 18:23, Răzvan Crainea ha scritto:


Hi, Stefano!

I don't think you are using the latest sources. Either you have not 
installed them, or you are using a wrong path. Could you please 
double-check?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/16/2016 06:56 PM, Stefano Pisani wrote:


I cloned the source from

|git clone https://github.com/OpenSIPS/opensips.git opensips_head 
the error appears the same Aug 16 11:55:14 [25630] 
INFO:tls_mgm:mod_init: initializing TLS protocol Aug 16 11:55:14 
[25630] ERROR:tls_mgm:mod_init: unable to set the memory allocation 
functions Aug 16 11:55:14 [25630] ERROR:tls_mgm:mod_init: NOTE: 
check if you have openssl 1.0.1e-fips, as this version is know to 
be broken; if so, you need to upgrade or downgrade to a differen 
openssl version !! Aug 16 11:55:14 [25630] ERROR:core:init_mod: 
failed to initialize module tls_mgm Aug 16 11:55:14 [25630] 
ERROR:core:main: error while initializing modules Aug 16 11:55:14 
[25630] INFO:core:cleanup: cleanup Aug 16 11:55:14 [25630] 
DBG:dispatcher:destroy: destroying module ... Aug 16 11:55:14 
[25630] DBG:tm:tm_shutdown: tm_shutdown : start Aug 16 11:55:14 
[25630] DBG:tm:unlink_timer_lists: emptying DELETE list for set 0 
Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: emptying hash table Aug 
16 11:55:14 [25630] DBG:tm:tm_shutdown: releasing timers Aug 16 
11:55:14 [25630] DBG:tm:tm_shutdown: removing semaphores Aug 16 
11:55:14 [25630] DBG:tm:tm_shutdown: destroying callback lists Aug 
16 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown : done Aug 16 
11:55:14 [25630] DBG:core:shm_mem_destroy: destroying the shared 
memory lock Aug 16 11:55:14 [25630] NOTICE:core:main: Exiting 
Thanks Stefano |


Il 16/08/2016 17:18, Răzvan Crainea ha scritto:

Stefano!

I've added some extra debugging in the master branch that might 
help you debug this. Could you please take the latest git 
version/deb and try to run again?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 08/16/2016 10:52 AM, Stefano Pisani wrote:

I'm trying to enable wss on opensips 2.2.1 and Ubuntu 16.04.1 LTS

version: opensips 2.2.1 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
F_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.
main.c compiled on  with gcc 5.3.1

It's not working:
Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing TLS 
protocol
Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to set the 
memory allocation functions
Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check if you 
have openssl 1.0.1e-fips, as this version is know to be broken; 
if so, you need to upgrade or downgrade to a differen openssl 
version !!
Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to initialize 
module tls_mgm


OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016

The opensips installed packages are:
opensipsinstall
opensips-tls-module:amd64   install
opensips-tlsmgm-module:amd64install
opensips-wss-module:amd64   install


There is something I can do?
Thanks






___
Users mailing list
Users@lists.opensips.org
http://lists.

Re: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory allocation functions

[Stefano Pisani]

I have this result.

# ./opensips -V
version: opensips 2.3.0-dev (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
QM_MALLOC, DBG_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.
git revision: 058991d
main.c compiled on 11:50:11 Aug 16 2016 with gcc 5.4.0



Il 16/08/2016 18:23, Răzvan Crainea ha scritto:


Hi, Stefano!

I don't think you are using the latest sources. Either you have not 
installed them, or you are using a wrong path. Could you please 
double-check?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 08/16/2016 06:56 PM, Stefano Pisani wrote:


I cloned the source from

|git clone https://github.com/OpenSIPS/opensips.git opensips_head the 
error appears the same Aug 16 11:55:14 [25630] INFO:tls_mgm:mod_init: 
initializing TLS protocol Aug 16 11:55:14 [25630] 
ERROR:tls_mgm:mod_init: unable to set the memory allocation functions 
Aug 16 11:55:14 [25630] ERROR:tls_mgm:mod_init: NOTE: check if you 
have openssl 1.0.1e-fips, as this version is know to be broken; if 
so, you need to upgrade or downgrade to a differen openssl version !! 
Aug 16 11:55:14 [25630] ERROR:core:init_mod: failed to initialize 
module tls_mgm Aug 16 11:55:14 [25630] ERROR:core:main: error while 
initializing modules Aug 16 11:55:14 [25630] INFO:core:cleanup: 
cleanup Aug 16 11:55:14 [25630] DBG:dispatcher:destroy: destroying 
module ... Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown : 
start Aug 16 11:55:14 [25630] DBG:tm:unlink_timer_lists: emptying 
DELETE list for set 0 Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: 
emptying hash table Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: 
releasing timers Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: removing 
semaphores Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: destroying 
callback lists Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: 
tm_shutdown : done Aug 16 11:55:14 [25630] DBG:core:shm_mem_destroy: 
destroying the shared memory lock Aug 16 11:55:14 [25630] 
NOTICE:core:main: Exiting Thanks Stefano |


Il 16/08/2016 17:18, Răzvan Crainea ha scritto:

Stefano!

I've added some extra debugging in the master branch that might help 
you debug this. Could you please take the latest git version/deb and 
try to run again?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 08/16/2016 10:52 AM, Stefano Pisani wrote:

I'm trying to enable wss on opensips 2.2.1 and Ubuntu 16.04.1 LTS

version: opensips 2.2.1 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
F_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.
main.c compiled on  with gcc 5.3.1

It's not working:
Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing TLS 
protocol
Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to set the 
memory allocation functions
Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check if you 
have openssl 1.0.1e-fips, as this version is know to be broken; if 
so, you need to upgrade or downgrade to a differen openssl version !!
Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to initialize 
module tls_mgm


OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016

The opensips installed packages are:
opensipsinstall
opensips-tls-module:amd64   install
opensips-tlsmgm-module:amd64install
opensips-wss-module:amd64   install


There is something I can do?
Thanks






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Re: [OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory allocation functions

[Stefano Pisani]

I cloned the source from

|git clone https://github.com/OpenSIPS/opensips.git opensips_head the 
error appears the same Aug 16 11:55:14 [25630] INFO:tls_mgm:mod_init: 
initializing TLS protocol Aug 16 11:55:14 [25630] 
ERROR:tls_mgm:mod_init: unable to set the memory allocation functions 
Aug 16 11:55:14 [25630] ERROR:tls_mgm:mod_init: NOTE: check if you have 
openssl 1.0.1e-fips, as this version is know to be broken; if so, you 
need to upgrade or downgrade to a differen openssl version !! Aug 16 
11:55:14 [25630] ERROR:core:init_mod: failed to initialize module 
tls_mgm Aug 16 11:55:14 [25630] ERROR:core:main: error while 
initializing modules Aug 16 11:55:14 [25630] INFO:core:cleanup: cleanup 
Aug 16 11:55:14 [25630] DBG:dispatcher:destroy: destroying module ... 
Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: tm_shutdown : start Aug 16 
11:55:14 [25630] DBG:tm:unlink_timer_lists: emptying DELETE list for set 
0 Aug 16 11:55:14 [25630] DBG:tm:tm_shutdown: emptying hash table Aug 16 
11:55:14 [25630] DBG:tm:tm_shutdown: releasing timers Aug 16 11:55:14 
[25630] DBG:tm:tm_shutdown: removing semaphores Aug 16 11:55:14 [25630] 
DBG:tm:tm_shutdown: destroying callback lists Aug 16 11:55:14 [25630] 
DBG:tm:tm_shutdown: tm_shutdown : done Aug 16 11:55:14 [25630] 
DBG:core:shm_mem_destroy: destroying the shared memory lock Aug 16 
11:55:14 [25630] NOTICE:core:main: Exiting Thanks Stefano |



Il 16/08/2016 17:18, Răzvan Crainea ha scritto:

Stefano!

I've added some extra debugging in the master branch that might help 
you debug this. Could you please take the latest git version/deb and 
try to run again?


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 08/16/2016 10:52 AM, Stefano Pisani wrote:

I'm trying to enable wss on opensips 2.2.1 and Ubuntu 16.04.1 LTS

version: opensips 2.2.1 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
F_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.
main.c compiled on  with gcc 5.3.1

It's not working:
Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing TLS protocol
Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to set the 
memory allocation functions
Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check if you 
have openssl 1.0.1e-fips, as this version is know to be broken; if 
so, you need to upgrade or downgrade to a differen openssl version !!
Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to initialize 
module tls_mgm


OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016

The opensips installed packages are:
opensipsinstall
opensips-tls-module:amd64   install
opensips-tlsmgm-module:amd64install
opensips-wss-module:amd64   install


There is something I can do?
Thanks






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[OpenSIPS-Users] ERROR:tls_mgm:mod_init: unable to set the memory allocation functions

[Stefano Pisani]

I'm trying to enable wss on opensips 2.2.1 and Ubuntu 16.04.1 LTS

version: opensips 2.2.1 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
F_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.
main.c compiled on  with gcc 5.3.1

It's not working:
Aug 16 03:43:02 [3915] INFO:tls_mgm:mod_init: initializing TLS protocol
Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: unable to set the memory 
allocation functions
Aug 16 03:43:02 [3915] ERROR:tls_mgm:mod_init: NOTE: check if you have 
openssl 1.0.1e-fips, as this version is know to be broken; if so, you 
need to upgrade or downgrade to a differen openssl version !!
Aug 16 03:43:02 [3915] ERROR:core:init_mod: failed to initialize module 
tls_mgm


OpenSSL version is OpenSSL 1.0.2g-fips  1 Mar 2016

The opensips installed packages are:
opensipsinstall
opensips-tls-module:amd64   install
opensips-tlsmgm-module:amd64install
opensips-wss-module:amd64   install


There is something I can do?
Thanks






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Re: [OpenSIPS-Users] opensips transparent technology

[Stefano Pisani]

Where is their real phone number?
Do you have it in a database?
You can change the From header to show the real phone number.



Il 24/01/2016 12.22, MichaelLeung ha scritto:

Hi all

i was trying to make my opensips users to sent their real phone number 
when they call .


what is the name of this technology ? transmit transparently ?

i search google find nothing, and where can i read document of this 
technology ?


thanks.


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Re: [OpenSIPS-Users] X-Auth-IP How?

[Stefano Pisani]

to be precise:

append_hf("X-Auth-IP: $i\r\n");

according to documentation

Il 13/01/2016 18.43, Tim King ha scritto:
I have read countless articles now talking about using x-auth-ip as a 
method for using OpenSIPs as a load balancer serving to a cluster of 
Freeswtich servers and having a method to maintain the original IP 
address.


Direct from the Freeswitch wiki it states:

apply-proxy-acl

Use the IP specified in X-AUTH-IP header sent from proxy for 
apply-inbound-acl Note: You'll need to configure your proxy to add 
this header



However I am not able to find any example of how to make OpenSIPs add 
this header. Can anyone provide me some guidance as to how to 
accomplish this?



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Re: [OpenSIPS-Users] X-Auth-IP How?

[Stefano Pisani]

what about

appendHf("X-Auth-IP: $si");

in your script?

however there are the "via" headers already to do this job

Il 13/01/2016 18.43, Tim King ha scritto:
I have read countless articles now talking about using x-auth-ip as a 
method for using OpenSIPs as a load balancer serving to a cluster of 
Freeswtich servers and having a method to maintain the original IP 
address.


Direct from the Freeswitch wiki it states:

apply-proxy-acl

Use the IP specified in X-AUTH-IP header sent from proxy for 
apply-inbound-acl Note: You'll need to configure your proxy to add 
this header



However I am not able to find any example of how to make OpenSIPs add 
this header. Can anyone provide me some guidance as to how to 
accomplish this?



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Re: [OpenSIPS-Users] uac_replace_from multiple times?

[Stefano Pisani]

No you can't.
Use a variable to store the from and replace it once, just before to 
send out the message.




Il 13/01/2016 14.32, Søren Andersen ha scritto:


Hello,

I’m wondering if it’s possible to use uac_replace_from multiple times? 
– fx. Inbound call gets changed by uac_replace_from and removed the 
+45 prefix. – But sometimes I need to forward the call back to my ISP, 
and they need to have +45 in the from header. But if I try to use the 
function two times the sip headers gets invalid  like the below header:


From: 
;tag=c675aa4668ba7a0c3150f682eea7a54b-7701


Perhaps there are a better option to do this?

/Søren



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Re: [OpenSIPS-Users] Can OpenSIPS can be used as a WebRTC gateway for JsSIP client and WebRTC client?

[Stefano Pisani]

Take a look to OverSIP.

Il 05/01/2016 07.42, suganthi karthick ha scritto:

Hi all,

I need to implement a WebRTC gateway for an existing conference 
bridge. The WebRTC gateway has to support Signaling, ICE, DTLS-SRTP. 
The webrtc clients can be JsSIP or any JSON based webrtc client.


The conference bridge is an existing working one for SIP clients, and 
I am trying to add webrtc support for that.


The webrtc gateway needs to be implemented in a way like a library 
because it needs to be integrated into the existing platform.


There are some init functions and config function from the existing 
conference platform, based on which the webrtc gateway has to  be 
configured.


Also, when a webrtc call come from a webrtc client, it needs to handle 
the signaling and the media(RTP) has to go to the conference bridge 
platform.


Do you have some suggestion on whether openSIPS can be used for this 
purpose?


Your suggestions will be helpful.

Thanks.




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Re: [OpenSIPS-Users] Try to setup ENUM NRENUM support...

[Stefano Pisani]

Ciao Michele,
per uscire con il contact corretto devi usare l'advertised_address / 
advertised_port se il serves ha solo un IP interno.
Questo fa nascere vari problemi perché la correzione viene fatta anche 
per verso i client interni.

Non c'è il concetto di Lan come per asterisk, almeno finora.
Nelle ultime versioni ho visto dei parametri in più quindi le cose 
potrebbero essere migliorate.


La miglior configurazione nel tuo caso sarebbe un server dual homed con 
IP interno ed esterno.


Ciao
s





Il 03/11/2015 10:43, Michele Pinassi ha scritto:

Hi all,

i'm trying to setup enum (NRENUM) infrastructure for our university. 
We have a public IP voip server, VOIP01, with a private network for 
voip phones (172.20.x.x). To let RTP flow through the private and 
public network, i set up a RTP Proxy:


/onreply_route[enum_answer] {//
//if(has_body("application/sdp")) {//
//rtpproxy_answer();//
//}//
//}//
//
//onreply_route[enum_offer] {//
//if(has_body("application/sdp")) {//
//rtpproxy_offer();//
//}//
//}//
//
//route[enum] {//
//xlog("L_INFO","Route to ENUM [$fd/$fu/$rd/$ru/$si/]\n");//

//if (is_method("INVITE")) {//
//if(has_body("application/sdp")) {//
//if (rtpproxy_offer()) {//
//t_on_reply("enum_answer");//
//}//
//} else {//
//t_on_reply("enum_offer");//
//}//
//}//
//if (is_method("ACK") && has_body("application/sdp")) {//
//rtpproxy_answer();//
//}//

//t_on_failure("pstn");//
//
//if(!t_relay()) {//
//sl_reply_error();//
//}//
//exit;//
//}/


but on establishing call, this is the tcpdump trace between VOIP01 and 
VOIP02 i get this:


/IP VOIP01.5060 > VOIP02.5060: UDP, length 1170//
//
//INVITE sip:86472@VOIP02 SIP/2.0//
//Record-Route: //
//Via: SIP/2.0/UDP VOIP01:5060;branch=z9hG4bKe567.edebdee3.0//
//Via: SIP/2.0/UDP 
172.20.1.47:57907;received=172.20.1.47;branch=z9hG4bK-7l3vckv00z2m;rport=57907//

//From: "Michele Pinassi" ;tag=gujliebxxu//
//To: //
//Call-ID: 313434363534333236353330393830-j09a5i32z5kd//
//CSeq: 2 INVITE//
//Max-Forwards: 69//
//User-Agent: snom760/8.7.5.17//
//Contact: ;reg-id=1//
//X-Serialnumber: 00041371928A//
//P-Key-Flags: resolution="31x13", keys="4"//
//Accept: application/sdp//
//Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, 
PRACK, MESSAGE, INFO, UPDATE//

//Allow-Events: talk, hold, refer, call-info//
//Supported: timer, 100rel, replaces, from-change//
//Session-Expires: 3600//
//Min-SE: 90//
//Content-Type: application/sdp//
//Content-Length: 228//
//
//v=0//
//o=root 846474428 846474428 IN IP4 172.20.1.47//
//s=call//
//c=IN IP4 VOIP01//
//t=0 0//
//m=audio 63194 RTP/AVP 9 0 8//
//a=rtpmap:9 G722/8000//
//a=rtpmap:0 PCMU/8000//
//a=rtpmap:8 PCMA/8000//
//a=ptime:20//
//a=sendrecv//
//a=nortpproxy:yes//
//
//IP VOIP02.5060 > VOIP01.5060: UDP, length 722//
//
//SIP/2.0 100 Trying//
//Via: SIP/2.0/UDP 
VOIP01:5060;branch=z9hG4bKe567.edebdee3.0;received=VOIP01;rport=5060//
//Via: SIP/2.0/UDP 
172.20.1.47:57907;received=172.20.1.47;branch=z9hG4bK-7l3vckv00z2m;rport=57907//

//Record-Route: //
//From: "Michele Pinassi" ;tag=gujliebxxu//
//To: //
//Call-ID: 313434363534333236353330393830-j09a5i32z5kd//
//CSeq: 2 INVITE//
//Server: FPBX-2.11.0(11.7.0)//
//Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH//

//Supported: replaces, timer//
//Session-Expires: 1800;refresher=uas//
//Contact: //
//Content-Length: 0//
//
//
//IP VOIP02.5060 > VOIP01.5060: UDP, length 590//
//
//SIP/2.0 603 Declined//
//Via: SIP/2.0/UDP 
VOIP01:5060;branch=z9hG4bKe567.edebdee3.0;received=VOIP01;rport=5060//
//Via: SIP/2.0/UDP 
172.20.1.47:57907;received=172.20.1.47;branch=z9hG4bK-7l3vckv00z2m;rport=57907//

//From: "Michele Pinassi" ;tag=gujliebxxu//
//To: ;tag=as15312d47//
//Call-ID: 313434363534333236353330393830-j09a5i32z5kd//
//CSeq: 2 INVITE//
//Server: FPBX-2.11.0(11.7.0)//
//Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH//

//Supported: replaces, timer//
//Content-Length: 0//
//
//
//IP VOIP01.5060 > VOIP02.5060: UDP, length 375//
//
//ACK sip:86472@VOIP02 SIP/2.0//
//Via: SIP/2.0/UDP VOIP01:5060;branch=z9hG4bKe567.edebdee3.0//
//From: "Michele Pinassi" ;tag=gujliebxxu//
//Call-ID: 313434363534333236353330393830-j09a5i32z5kd//
//To: ;tag=as15312d47//
//CSeq: 2 ACK//
//Max-Forwards: 70//
//User-Agent: VoIP Unisi.it//
//Content-Length: 0/

The main doubt is: /a=nortpproxy:yes/ ...why ?

Thanks, Michele



--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 -central...@unisi.it

Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo,http://www.faq.unisi.it  



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Re: [OpenSIPS-Users] rtpproxy vs. STUN/TURN/ICE

[Stefano Pisani]

Stun/Turn/Ice are usefull where Client is behind a NAT and OpenSIPS has 
public IP.
You could use nathelper modules instead of Stun, to set the right IPs in 
the messages from client.
If OpenSIPS is behind a NAT too (it has private IP) you must use 
RTPProxy too with a proper configuration.



Il 29/08/2015 17:47, Nabeel ha scritto:


Hi,

I would like to know which is more effective for NAT traversal, 
rtpproxy or STUN/TURN/ICE implementation.


I heard that TURN server with one public IP can function equivalent to 
rtpproxy, and TURN server with two public IPs is more effective than 
rtpproxy.


Is that true?



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[OpenSIPS-Users] Save in on_reply route

[Stefano Pisani]

Hi Guys,
I'm facing an issue. I'm using opensips as proxy for REGISTER too.
So I need to save the registration only when 200 OK returns from media 
server.
I do save("location","r") in onreply_route and it works but there is a 
problem.
The original contact was fixed using fix_nated_contact() but the 
original one in saved in location, so I have an internal IP in contact 
field and the sipping (the reason for doing that) is not working.


What I'm doing wrong?
Thanks

s

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[OpenSIPS-Users] Save with AOR

[Stefano Pisani]

Hi Guys,
I'm using OpenSIPS 1.11.

save("location","r","sip:user@1.1.1.1:3")  in REPLY ROUTE is not working

in the location table I find always a different value from the AOR I set.

Is it a bug?

Thanks
s

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[OpenSIPS-Users] transformation

[Stefano Pisani]

Hello guys,
I have problems with these to simple transformations.

$var(uri) = "$ct.fields(uri)";
$var(ct) = "$(var(uri){uri.user})" + "@" + "$(var(uri){uri.host})";

what's wrong with them?

s

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Re: [OpenSIPS-Users] forwarding calls from Asterisk to OpenSIPs

[Stefano Pisani]

Just define a peer in sip.conf

then Dial(SIP/peer_name/OpensipsAccount) on extensions.conf. Very simple.

regards,
s

Il 07/05/2015 08:32, Julian Kay ha scritto:


Hello Everyone;

I want to be able to forward calls from Asterisk to a phone registered 
with OpenSIPs, can anyone give me some information on how this can be 
done? Or a link to the info I would need to do this.


Thanks for any help

Juls



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Re: [OpenSIPS-Users] SIP and RTP Proxy without local user base

[Stefano Pisani]

Remember that if the audio path is p2p the DMTF tones do not work 
anymore, so you can't transfer a call, etc. You should use SIP INFO if 
supported.
For the audio path give a look to the asterisk options /directmedia/=yes 
and directrtpsetup=yes. You could not need to use openSIPS and rtpproxy.




Il 24/04/2015 17:36, Russell Treleaven ha scritto:

As an alternative you could try bypass media.
https://freeswitch.org/confluence/display/FREESWITCH/Bypass+Media+Overview


On Fri, Apr 24, 2015 at 11:17 AM, Roman Dissauer > wrote:


Dear all,

I’m running a centralized Freeswitch based PBX for use on several
sites. All phones register against this Freeswitch instance.

Now I want to install SIP and RTP Proxies on every site to keep
RTP traffic locally on site. Registration should still be done by
freeswitch. Can anybody give me a hint if this is possible with
opensips and rtpproxy?

Maybe I can clarify it a bit more:

|  Freeswitch  |

   |
   |Public Internet
   |
   |- Phone 3 external
   |
   |
--
| Firewall / NAT |
--
   |

|Proxy |   Site 1

   |
   |
   |- Phone 1 internal
   |- Phone 2 internal

Phone 1 - 3 are all registered at Freeswitch
Phone 1 calls Phone 3: SIP and RTP over Freeswitch
Phone 1 calls Phone 2: SIP over Freeswitch but RTP over Proxy

Does this make sense?

Thanks,
Roman
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Re: [OpenSIPS-Users] OPENSIPS + IVR CALL CONTROL

[Stefano Pisani]

You should say something more about your issue.



Il 14/03/2015 17:22, mahan77 ha scritto:
Hi Danilo, I’m having problem with OpenSips => Asterisk connection. 
Can you able to mail me your working OpenSips scripts. mail at 
Sathees.co.uk appreciate sathees


View this message in context: Re: OPENSIPS + IVR CALL CONTROL 

Sent from the OpenSIPS - Users mailing list archive 
 
at Nabble.com.



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Re: [OpenSIPS-Users] OpenSipS as a simple proxy

[Stefano Pisani]

The purpose is to create a privileged network path beetween your server 
and the local clients.

Is it?

Il 04/02/2015 20.58, Dovid Bender ha scritto:


Hi,

We have a cluster in the US running custom software. We have clients 
in countries where their ISP’s traffic out of the country is not the 
best. We were thinking of getting a local server and then simply 
having all the traffic go through an OpenSipS box. We would like for 
two things to happen.


1)Any request that comes to the OpenSipS server simply gets passed to 
our OpenSipS servers in the US.


2)That the server proxy all RTP media.

Is there any “simple” configuration out there for such a set up?

Thanks in advance.

Regards,

Dovid



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[OpenSIPS-Users] ACK never leaves opensips

[Stefano Pisani]

I have a strange issue with an ACK that never leaves Opensips. It 
disappears.

This is the ACK message incoming dumped with ngrep

U publicIP1:32769 -> publicIPOpenSIPS:5172
ACK sip:s@publicIP2:6050 SIP/2.0.
Via: SIP/2.0/UDP 192.168.4.53:32769;branch=z9hG4bK-nt6kbhw2yq7b;rport.
Route: 
.

From: "103" ;tag=dwlhrdursy.
To: ;tag=as24f5fc71.
Call-ID: 313432313936303531313339303433-apx44rudybcq.
CSeq: 1 ACK.
Max-Forwards: 70.
User-Agent: snom710/8.7.5.13.
Contact: ;reg-id=1.
Content-Length: 0.

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Re: [OpenSIPS-Users] Some phones can call, others not...

[Stefano Pisani]

In the first case the last UDP packet (INVITE with authentication 
header) is fragmented.

It's very common to lost fragmented UDP packet.

s

Il 28/11/2014 09.42, Michele Pinassi ha scritto:

Hi all,

i'm experiencing a strange issue. Some VoIP phones, like mine (5002),
cannot call other phones (like 5023) but i receive calls from 5023. Same
config, same context.

I did some sipgrep:

Call from 5002 to 5023 (FAILED): http://pastebin.com/BF6YyWHr
Call from 5023 to 5002 (SUCCESS): http://pastebin.com/rW3AKr22

My config: http://pastebin.com/9gP9xncd

Thanks for all your help.

Michele



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Re: [OpenSIPS-Users] URGENT! uac_auth PSTN gateway authentication issue

[Stefano Pisani]

Check if the cseq was incremented by one in the second try.
Use ngrep.



Il 24/08/2014 22.24, Satish Patel ha scritto:


Hi,

my Opensips (UAC) registered to PSTN gateway and now i am trying to 
call using my SIPphone which is register to opensip but no success. I 
am getting 407 Proxy authentication issue..  I am using following 
method but it didn't work. I need solution badly..


PSTN gateway sending 407 Proxy auth and then my Opensip sending 407 
proxy auth to SIP phone.


Does anyone has any working example or some kind of document? I 
haven;t see any single doc anywhere in Internet about uac_auth issue




modparam("uac","credential","username:domain:password")

route {

t_on_failure("2");
t_relay( "udp:ip_addr:5060" );
...
}

failure_route[2] {
  uac_auth();
  t_relay("udp:ip_addr:5060");
}


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Re: [OpenSIPS-Users] 403 Rely forbidden

[Stefano Pisani]

Is your installation working?
Did you try to add a user? It worked?



Il 28/04/2014 11.47, Schneur Rosenberg ha scritto:


You missed the fifo

On Apr 27, 2014 3:11 AM, <mailto:i...@vintageelectronics.ca>> wrote:


Help says it's opensips domain add.
This is what I get:

# opensipsctl domain add 10.10.10.3
INFO: execute '/sbin/opensipsctl domain reload' to synchronize
cache and database
[root@lion opensips]# opensipsctl domain reload
500 command 'domain_reload' not available

What's wrong again?

Thank you!

    On 04/26/2014 01:17 PM, Stefano Pisani wrote:

opensipsctl add domain ... etc.

"opensipsctl --help" can help



Il 26/04/2014 16.13, i...@vintageelectronics.ca
<mailto:i...@vintageelectronics.ca> ha scritto:

Stephano,

Just realized that I only replied directly to you so this
probably never hit the maillist.
Ok, now I understand that a local domain has to be defined, but
don't know how to achieve that.
Any tips?

Thank you!

On 04/25/2014 09:14 AM, Stefano Pisani wrote:

If you do not define a local domain OpenSIPS can't understand
that SIP messase is for him.

Il 25/04/2014 13.36, i...@vintageelectronics.ca
<mailto:i...@vintageelectronics.ca> ha scritto:

Most likely not, as I did not make any configuration changes
besides changing listen IP to 10.10.10.3 and port to 5066.
My install is the same as came from the rpm.
What do I need to do to define the local domains?

Thank you!

On 04/25/2014 03:18 AM, Stefano Pisani wrote:

Did you defined local domains in OpenSIPS? 10.10.10.3 should
be one of them.

Il 25/04/2014 00.40, i...@vintageelectronics.ca
<mailto:i...@vintageelectronics.ca> ha scritto:

Would this be sufficient? I just ran `ngrep -d lo` - let me
know if I should use different options:

# ngrep -d lo
interface: lo (127.0.0.0/255.0.0.0 <http://127.0.0.0/255.0.0.0>)
#
T 10.10.10.3:43236 <http://10.10.10.3:43236> ->
10.10.10.3:5066 <http://10.10.10.3:5066> [AP]
  OPTIONS sip:10.10.10.3 SIP/2.0..Call-ID:
d1c9a44f67f3c5dcfd0d917fad134e47@0:0:0:0:0:0:0:0..CSeq
<mailto:d1c9a44f67f3c5dcfd0d917fad134e47@0:0:0:0:0:0:0:0..CSeq>:
111 OPTIONS..From: "user0@SIP" 
<mailto:sip:user0@10.10.10.3>;tag=7b34c5d
  6..To: "user0@SIP" 
<mailto:sip:user0@10.10.10.3>..Via: SIP/2.0/TCP

10.10.10.3:43236;branch=z9hG4bK-313736-2cb10f1cb92f117c858e374b091830eb..Max-Forwards:
70..Contact: "a
  lexim@SIP"


<mailto:sip:user0@10.10.10.3:43236;transport=tcp;registering_acc=10_10_10_3>..User-Agent:
Jitsi2.5.5193Linux..Allow:
INFO,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,A
CK,CANCEL,PUBLISH,NOTIFY,INVITE..Allow-Events:
refer..Content-Length: 0
#
T 10.10.10.3:5066 <http://10.10.10.3:5066> ->
10.10.10.3:43236 <http://10.10.10.3:43236> [AP]
  SIP/2.0 403 Rely forbidden..Call-ID:
d1c9a44f67f3c5dcfd0d917fad134e47@0:0:0:0:0:0:0:0..CSeq
<mailto:d1c9a44f67f3c5dcfd0d917fad134e47@0:0:0:0:0:0:0:0..CSeq>:
111 OPTIONS..From: "user0@SIP" 
<mailto:sip:user0@10.10.10.3>;tag=7b34c5d6..T
  o: "user0@SIP" 

<mailto:sip:user0@10.10.10.3>;tag=ed31db4a4567c982708589607dc4dd7d.6e52..Via:
SIP/2.0/TCP
10.10.10.3:43236;branch=z9hG4bK-313736-2cb10f1cb92f117c858e374b0
  91830eb..Server: OpenSIPS (1.10.1-tls
(x86_64/linux))..Content-Length: 0
##
T 10.10.10.3:43236 <http://10.10.10.3:43236> ->
10.10.10.3:5066 <http://10.10.10.3:5066> [AP]
  MESSAGE sip:user1@10.10.10.3 <mailto:sip:user1@10.10.10.3>
SIP/2.0..Call-ID:
cf8432a003f376bea826a03b1d6c2791@0:0:0:0:0:0:0:0..CSeq
<mailto:cf8432a003f376bea826a03b1d6c2791@0:0:0:0:0:0:0:0..CSeq>:
694827676 MESSAGE..From: "user0@SIP" 
<mailto:sip:user0@10.10.10.3>;
  tag=29fbaeac..To: 
<mailto:sip:user1@10.10.10.3>..Via: SIP/2.0/TCP

10.10.10.3:43236;branch=z9hG4bK-313736-c420c7a03fba6256d5863512fbdd6683..Max-Forwards:
70..Content-Type: t
  ext/plain..Contact: "user0@SIP"


<mailto:sip:user0@10.10.10.3:43236;transport=tcp;registering_acc=10_10_10_3>..User-Agent:
Jitsi2.5.5193Linux..Content-Length: 44test
  .   . . .   . .  .   ..  .   .. ..
#
T 10.10.10.3:5066 <http://10.10.10.3:5066> ->
10.10.10.3:43236 <http://10.10.10.3:43236> [AP]
  SIP/2.0 403 Rely forbidden..Call-ID:
cf8432a003f376bea826a03b1d6c2791@0:0:0:0:0:0:0:0..CSeq
<mailto:cf8432a003f376bea826a03b1d6c2791@0:0:0:0:0:0:0:0..CSeq>:
694827676 MESSAGE..From: "user0@SIP" 
<mailto:sip:user0@10.10.10.3>;tag=29fba
  eac..To: 

<mailto:sip:

Re: [OpenSIPS-Users] 403 Rely forbidden

[Stefano Pisani]

opensipsctl add domain ... etc.

"opensipsctl --help" can help



Il 26/04/2014 16.13, i...@vintageelectronics.ca ha scritto:

Stephano,

Just realized that I only replied directly to you so this probably 
never hit the maillist.
Ok, now I understand that a local domain has to be defined, but don't 
know how to achieve that.

Any tips?

Thank you!

On 04/25/2014 09:14 AM, Stefano Pisani wrote:
If you do not define a local domain OpenSIPS can't understand that 
SIP messase is for him.


Il 25/04/2014 13.36, i...@vintageelectronics.ca ha scritto:
Most likely not, as I did not make any configuration changes besides 
changing listen IP to 10.10.10.3 and port to 5066.

My install is the same as came from the rpm.
What do I need to do to define the local domains?

Thank you!

On 04/25/2014 03:18 AM, Stefano Pisani wrote:
Did you defined local domains in OpenSIPS? 10.10.10.3 should be one 
of them.


Il 25/04/2014 00.40, i...@vintageelectronics.ca ha scritto:
Would this be sufficient? I just ran `ngrep -d lo` - let me know 
if I should use different options:


# ngrep -d lo
interface: lo (127.0.0.0/255.0.0.0)
#
T 10.10.10.3:43236 -> 10.10.10.3:5066 [AP]
  OPTIONS sip:10.10.10.3 SIP/2.0..Call-ID: 
d1c9a44f67f3c5dcfd0d917fad134e47@0:0:0:0:0:0:0:0..CSeq: 111 
OPTIONS..From: "user0@SIP" ;tag=7b34c5d
  6..To: "user0@SIP" ..Via: SIP/2.0/TCP 
10.10.10.3:43236;branch=z9hG4bK-313736-2cb10f1cb92f117c858e374b091830eb..Max-Forwards: 
70..Contact: "a
  lexim@SIP" 
..User-Agent: 
Jitsi2.5.5193Linux..Allow: INFO,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,A
  CK,CANCEL,PUBLISH,NOTIFY,INVITE..Allow-Events: 
refer..Content-Length: 0

#
T 10.10.10.3:5066 -> 10.10.10.3:43236 [AP]
  SIP/2.0 403 Rely forbidden..Call-ID: 
d1c9a44f67f3c5dcfd0d917fad134e47@0:0:0:0:0:0:0:0..CSeq: 111 
OPTIONS..From: "user0@SIP" ;tag=7b34c5d6..T
  o: "user0@SIP" 
;tag=ed31db4a4567c982708589607dc4dd7d.6e52..Via: 
SIP/2.0/TCP 
10.10.10.3:43236;branch=z9hG4bK-313736-2cb10f1cb92f117c858e374b0
  91830eb..Server: OpenSIPS (1.10.1-tls 
(x86_64/linux))..Content-Length: 0

##
T 10.10.10.3:43236 -> 10.10.10.3:5066 [AP]
  MESSAGE sip:user1@10.10.10.3 SIP/2.0..Call-ID: 
cf8432a003f376bea826a03b1d6c2791@0:0:0:0:0:0:0:0..CSeq: 694827676 
MESSAGE..From: "user0@SIP" ;
  tag=29fbaeac..To: ..Via: SIP/2.0/TCP 
10.10.10.3:43236;branch=z9hG4bK-313736-c420c7a03fba6256d5863512fbdd6683..Max-Forwards: 
70..Content-Type: t
  ext/plain..Contact: "user0@SIP" 
..User-Agent: 
Jitsi2.5.5193Linux..Content-Length: 44test

  .   . . .   . .  .   ..  .   .. ..
#
T 10.10.10.3:5066 -> 10.10.10.3:43236 [AP]
  SIP/2.0 403 Rely forbidden..Call-ID: 
cf8432a003f376bea826a03b1d6c2791@0:0:0:0:0:0:0:0..CSeq: 694827676 
MESSAGE..From: "user0@SIP" ;tag=29fba
  eac..To: 
;tag=ed31db4a4567c982708589607dc4dd7d.3249..Via: 
SIP/2.0/TCP 
10.10.10.3:43236;branch=z9hG4bK-313736-c420c7a03fba6256d5863512fbdd6683..

  Server: OpenSIPS (1.10.1-tls (x86_64/linux))..Content-Length: 0
#^Cexit
52 received, 0 dropped


On 04/24/2014 06:26 PM, Stefano Pisani wrote:
Could you dump the SIP dialog using ngrep on opensips server? and 
post it?


Il 25/04/2014 00.17, i...@vintageelectronics.ca ha scritto:
My installation is essentially vanilla, whatever came with the 
rpms except for listen port/IP.
Where in which file should I seek the ruri header and what 
should it be set to?
I did a global search in /etc/opensips and found its only 
mention at line 229 of opensips.cfg (it's a comment):


if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}

Thank you!

On 04/23/2014 08:12 PM, Tito Cumpen wrote:
it means that the server isnt identifying itself as the host. 
Please take a look at the ruri header host portion and verify 
that your server is aware of all alias hostnames.






On Wed, Apr 23, 2014 at 7:10 PM, <mailto:i...@vintageelectronics.ca>> wrote:


Getting this error from Jitsi when sending text messages or
trying to place a voice call between the registered contacts:


 The above message could not be delivered
A network problem occurred. Please check your network
configuration and try again. Error was: 403 Rely forbidden



What is Rely and what does that error mean?

Thank you!













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Re: [OpenSIPS-Users] 403 Rely forbidden

[Stefano Pisani]

Did you defined local domains in OpenSIPS? 10.10.10.3 should be one of them.

Il 25/04/2014 00.40, i...@vintageelectronics.ca ha scritto:
Would this be sufficient? I just ran `ngrep -d lo` - let me know if I 
should use different options:


# ngrep -d lo
interface: lo (127.0.0.0/255.0.0.0)
#
T 10.10.10.3:43236 -> 10.10.10.3:5066 [AP]
  OPTIONS sip:10.10.10.3 SIP/2.0..Call-ID: 
d1c9a44f67f3c5dcfd0d917fad134e47@0:0:0:0:0:0:0:0..CSeq: 111 
OPTIONS..From: "user0@SIP" ;tag=7b34c5d
  6..To: "user0@SIP" ..Via: SIP/2.0/TCP 
10.10.10.3:43236;branch=z9hG4bK-313736-2cb10f1cb92f117c858e374b091830eb..Max-Forwards: 
70..Contact: "a
  lexim@SIP" 
..User-Agent: 
Jitsi2.5.5193Linux..Allow: INFO,OPTIONS,MESSAGE,BYE,REFER,SUBSCRIBE,A
  CK,CANCEL,PUBLISH,NOTIFY,INVITE..Allow-Events: 
refer..Content-Length: 0

#
T 10.10.10.3:5066 -> 10.10.10.3:43236 [AP]
  SIP/2.0 403 Rely forbidden..Call-ID: 
d1c9a44f67f3c5dcfd0d917fad134e47@0:0:0:0:0:0:0:0..CSeq: 111 
OPTIONS..From: "user0@SIP" ;tag=7b34c5d6..T
  o: "user0@SIP" 
;tag=ed31db4a4567c982708589607dc4dd7d.6e52..Via: 
SIP/2.0/TCP 
10.10.10.3:43236;branch=z9hG4bK-313736-2cb10f1cb92f117c858e374b0
  91830eb..Server: OpenSIPS (1.10.1-tls 
(x86_64/linux))..Content-Length: 0

##
T 10.10.10.3:43236 -> 10.10.10.3:5066 [AP]
  MESSAGE sip:user1@10.10.10.3 SIP/2.0..Call-ID: 
cf8432a003f376bea826a03b1d6c2791@0:0:0:0:0:0:0:0..CSeq: 694827676 
MESSAGE..From: "user0@SIP" ;
  tag=29fbaeac..To: ..Via: SIP/2.0/TCP 
10.10.10.3:43236;branch=z9hG4bK-313736-c420c7a03fba6256d5863512fbdd6683..Max-Forwards: 
70..Content-Type: t
  ext/plain..Contact: "user0@SIP" 
..User-Agent: 
Jitsi2.5.5193Linux..Content-Length: 44test

  .   . . .   . .  .   ..  .   .. ..
#
T 10.10.10.3:5066 -> 10.10.10.3:43236 [AP]
  SIP/2.0 403 Rely forbidden..Call-ID: 
cf8432a003f376bea826a03b1d6c2791@0:0:0:0:0:0:0:0..CSeq: 694827676 
MESSAGE..From: "user0@SIP" ;tag=29fba
  eac..To: 
;tag=ed31db4a4567c982708589607dc4dd7d.3249..Via: 
SIP/2.0/TCP 
10.10.10.3:43236;branch=z9hG4bK-313736-c420c7a03fba6256d5863512fbdd6683..

  Server: OpenSIPS (1.10.1-tls (x86_64/linux))..Content-Length: 0
#^Cexit
52 received, 0 dropped


On 04/24/2014 06:26 PM, Stefano Pisani wrote:
Could you dump the SIP dialog using ngrep on opensips server? and 
post it?


Il 25/04/2014 00.17, i...@vintageelectronics.ca ha scritto:
My installation is essentially vanilla, whatever came with the rpms 
except for listen port/IP.
Where in which file should I seek the ruri header and what should it 
be set to?
I did a global search in /etc/opensips and found its only mention at 
line 229 of opensips.cfg (it's a comment):


if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}

Thank you!

On 04/23/2014 08:12 PM, Tito Cumpen wrote:
it means that the server isnt identifying itself as the host. 
Please take a look at the ruri header host portion and verify that 
your server is aware of all alias hostnames.






On Wed, Apr 23, 2014 at 7:10 PM, <mailto:i...@vintageelectronics.ca>> wrote:


Getting this error from Jitsi when sending text messages or
trying to place a voice call between the registered contacts:


 The above message could not be delivered
A network problem occurred. Please check your network
configuration and try again. Error was: 403 Rely forbidden



What is Rely and what does that error mean?

Thank you!





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Re: [OpenSIPS-Users] 403 Rely forbidden

[Stefano Pisani]

Could you dump the SIP dialog using ngrep on opensips server? and post it?

Il 25/04/2014 00.17, i...@vintageelectronics.ca ha scritto:
My installation is essentially vanilla, whatever came with the rpms 
except for listen port/IP.
Where in which file should I seek the ruri header and what should it 
be set to?
I did a global search in /etc/opensips and found its only mention at 
line 229 of opensips.cfg (it's a comment):


if ($rU==NULL) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}

Thank you!

On 04/23/2014 08:12 PM, Tito Cumpen wrote:
it means that the server isnt identifying itself as the host. Please 
take a look at the ruri header host portion and verify that your 
server is aware of all alias hostnames.






On Wed, Apr 23, 2014 at 7:10 PM, > wrote:


Getting this error from Jitsi when sending text messages or
trying to place a voice call between the registered contacts:


 The above message could not be delivered
A network problem occurred. Please check your network
configuration and try again. Error was: 403 Rely forbidden



What is Rely and what does that error mean?

Thank you!


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Re: [OpenSIPS-Users] Rewriting Contact Header -- Should I or Shouldn't I?

[Stefano Pisani]

There are more than one Record-Route.
There is one record with proxy ip? Where you read the messages? on the 
uac interface?
The message changes according to the point where it is. Proxy shoud add 
its own record route  header.




Il 29/03/2014 18.46, Peter Kust ha scritto:

Record-Route: 

I believe the answer is yes.  The above header field is in the OK message.

Cordially,
  
Peter Nayland Kust

Director of Technologies
BusinesSuites
24624 Interstate 45 North, Suite 200
Houston, TX 77070
peter.k...@businessuites.com

-Original Message-
From: Stefano Pisani [mailto:stefano.pis...@omnianet.it]
Sent: Saturday, March 29, 2014 12:23 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Rewriting Contact Header -- Should I or Shouldn't 
I?

Is the request route header present in OK message?

Il 29/03/2014 18.13, Peter Kust ha scritto:

Is the Contact header in the OK message correct?  That  is the question of the 
moment.

When the call gets answered, (after the 180 "RINGING" messages), the Asterisk 
server sends an OK message to the proxy with a contact header containing the IP address 
of the Asterisk server in the Contact URI.  The Proxy then sends that OK message onto the 
UAC with the same contact header (i.e., with the IP Address of the Asterisk server in the 
Contact URI).  The UAC then sends the ACK directly to the Asterisk server and bypasses 
the proxy.  As a result, the Asterisk server sends the BYE message directly to the UAC 
and not the proxy.

This is the Contact header in the OK message the proxy sends to the UAC:

Contact: 

*.*.*.102 is the IP address of my Asterisk server.  My Proxy server is at 
*.*.*.200.

S..should the Contact header in the OK message the proxy sends
to the UAC have the IP address of the proxy or the original IP address
of the Asterisk server?  Is the contact header correct as is, or
should it read

Contact: 

That is where I am getting stumped.  That, and what the best/safest and most 
stable method is for altering that header, if necessary.

Cordially,
   
Peter Nayland Kust

Director of Technologies
BusinesSuites
24624 Interstate 45 North, Suite 200
Houston, TX 77070
peter.k...@businessuites.com
-Original Message-
From: Stefano Pisani [mailto:stefano.pis...@omnianet.it]
Sent: Saturday, March 29, 2014 11:23 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Rewriting Contact Header -- Should I or Shouldn't 
I?

BYE was never received.
Check the Contact header in OK message. Is it right?
Check also the request route. Are they present? Probably NOT because BYE go to 
The UAC and not to the PROXY.

Cheers,
s

Il 29/03/2014 17.17, Peter Kust ha scritto:

Also, this is how the SIP messaging is proceeding, starting with the
INVITE from the GenBand eSBC

**.***.***.110  INVITE SDP (g711U telephone-event)
   (5060)   -->  (5060)   **.***.***.200
**.***.***.110  100 Giving a try
   (5060)   <--  (5060)   **.***.***.200
  **.***.***.200 INVITE 
SDP (g711U telephone-event)
 (5060)   
-->  (5060)   **.***.***.102
  **.***.***.200 100 
Trying
 (5060)   
<--  (5060)   **.***.***.102
  **.***.***.200 180 
Ringing
 (5060)   
<--  (5060)   **.***.***.102
**.***.***.110  180 Ringing
   (5060)   <--  (5060)   **.***.***.200
  **.***.***.200 180 
Ringing
 (5060)   
<--  (5060)   **.***.***.102
**.***.***.110  180 Ringing
   (5060)   <--  (5060)   **.***.***.200
  **.***.***.200 200 OK 
SDP (g711U telephone-event)
 (5060)   
<--  (5060)   **.***.***.102
**.***.***.110  200 OK SDP (g711U telephone-event)
   (5060)   <--  (5060)   **.***.***.200
**.***.***.110  ACK
   (5060)   
>  
(5060)   **.***.***.102
**.***.***.110  BYE
   (5060)   
-  
(5060)   **.***.***.102
**.***.***.110  BYE
   (5060)   
-  
(5060)   **.***

Re: [OpenSIPS-Users] Rewriting Contact Header -- Should I or Shouldn't I?

[Stefano Pisani]

Is the request route header present in OK message?

Il 29/03/2014 18.13, Peter Kust ha scritto:

Is the Contact header in the OK message correct?  That  is the question of the 
moment.

When the call gets answered, (after the 180 "RINGING" messages), the Asterisk 
server sends an OK message to the proxy with a contact header containing the IP address 
of the Asterisk server in the Contact URI.  The Proxy then sends that OK message onto the 
UAC with the same contact header (i.e., with the IP Address of the Asterisk server in the 
Contact URI).  The UAC then sends the ACK directly to the Asterisk server and bypasses 
the proxy.  As a result, the Asterisk server sends the BYE message directly to the UAC 
and not the proxy.

This is the Contact header in the OK message the proxy sends to the UAC:

Contact: 

*.*.*.102 is the IP address of my Asterisk server.  My Proxy server is at 
*.*.*.200.

S..should the Contact header in the OK message the proxy sends to the 
UAC have the IP address of the proxy or the original IP address of the Asterisk 
server?  Is the contact header correct as is, or should it read

Contact: 

That is where I am getting stumped.  That, and what the best/safest and most 
stable method is for altering that header, if necessary.

Cordially,
  
Peter Nayland Kust

Director of Technologies
BusinesSuites
24624 Interstate 45 North, Suite 200
Houston, TX 77070
peter.k...@businessuites.com
-Original Message-----
From: Stefano Pisani [mailto:stefano.pis...@omnianet.it]
Sent: Saturday, March 29, 2014 11:23 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Rewriting Contact Header -- Should I or Shouldn't 
I?

BYE was never received.
Check the Contact header in OK message. Is it right?
Check also the request route. Are they present? Probably NOT because BYE go to 
The UAC and not to the PROXY.

Cheers,
s

Il 29/03/2014 17.17, Peter Kust ha scritto:

Also, this is how the SIP messaging is proceeding, starting with the
INVITE from the GenBand eSBC

**.***.***.110  INVITE SDP (g711U telephone-event)
  (5060)   -->  (5060)   **.***.***.200
**.***.***.110  100 Giving a try
  (5060)   <--  (5060)   **.***.***.200
 **.***.***.200 INVITE 
SDP (g711U telephone-event)
(5060)   
-->  (5060)   **.***.***.102
 **.***.***.200 100 
Trying
(5060)   
<--  (5060)   **.***.***.102
 **.***.***.200 180 
Ringing
(5060)   
<--  (5060)   **.***.***.102
**.***.***.110  180 Ringing
  (5060)   <--  (5060)   **.***.***.200
 **.***.***.200 180 
Ringing
(5060)   
<--  (5060)   **.***.***.102
**.***.***.110  180 Ringing
  (5060)   <--  (5060)   **.***.***.200
 **.***.***.200 200 OK 
SDP (g711U telephone-event)
(5060)   
<--  (5060)   **.***.***.102
**.***.***.110  200 OK SDP (g711U telephone-event)
  (5060)   <--  (5060)   **.***.***.200
**.***.***.110  ACK
  (5060)   
>  
(5060)   **.***.***.102
**.***.***.110  BYE
  (5060)   
-  
(5060)   **.***.***.102
**.***.***.110  BYE
  (5060)   
-  
(5060)   **.***.***.102
**.***.***.110  BYE
  (5060)   
-  
(5060)   **.***.***.102
**.***.***.110  BYE
  (5060)   
-  
(5060)   **.***.***.102
**.***.***.110  BYE
  (5060)   
-  
(5060)   **.***.***.102
**.***.***.110  BYE
  (5060)   
-  
(5060)   **.***.***.102
**.***.***.110  BYE
  (5060)   
-  
(5060)   **.***.***.102
**.***.***.110  BYE
  (5060)   
-

Re: [OpenSIPS-Users] Rewriting Contact Header -- Should I or Shouldn't I?

[Stefano Pisani]

BYE was never received.
Check the Contact header in OK message. Is it right?
Check also the request route. Are they present? Probably NOT because BYE 
go to The UAC and not to the PROXY.


Cheers,
s

Il 29/03/2014 17.17, Peter Kust ha scritto:

Also, this is how the SIP messaging is proceeding, starting with the INVITE 
from the GenBand eSBC

**.***.***.110  INVITE SDP (g711U telephone-event)
 (5060)   -->  (5060)   **.***.***.200
**.***.***.110  100 Giving a try
 (5060)   <--  (5060)   **.***.***.200
**.***.***.200 INVITE 
SDP (g711U telephone-event)
   (5060)   
-->  (5060)   **.***.***.102
**.***.***.200 100 
Trying
   (5060)   
<--  (5060)   **.***.***.102
**.***.***.200 180 
Ringing
   (5060)   
<--  (5060)   **.***.***.102
**.***.***.110  180 Ringing
 (5060)   <--  (5060)   **.***.***.200
**.***.***.200 180 
Ringing
   (5060)   
<--  (5060)   **.***.***.102
**.***.***.110  180 Ringing
 (5060)   <--  (5060)   **.***.***.200
**.***.***.200 200 OK 
SDP (g711U telephone-event)
   (5060)   
<--  (5060)   **.***.***.102
**.***.***.110  200 OK SDP (g711U telephone-event)
 (5060)   <--  (5060)   **.***.***.200
**.***.***.110  ACK
 (5060)   
>  
(5060)   **.***.***.102
**.***.***.110  BYE
 (5060)   
-  
(5060)   **.***.***.102
**.***.***.110  BYE
 (5060)   
-  
(5060)   **.***.***.102
**.***.***.110  BYE
 (5060)   
-  
(5060)   **.***.***.102
**.***.***.110  BYE
 (5060)   
-  
(5060)   **.***.***.102
**.***.***.110  BYE
 (5060)   
-  
(5060)   **.***.***.102
**.***.***.110  BYE
 (5060)   
-  
(5060)   **.***.***.102
**.***.***.110  BYE
 (5060)   
-  
(5060)   **.***.***.102
**.***.***.110  BYE
 (5060)   
-  
(5060)   **.***.***.102
**.***.***.110  481 Call leg/transaction does not exist
 (5060)   
-  
(5060)   **.***.***.102

Cordially,

Peter Nayland Kust
Director of Technologies
BusinesSuites
24624 Interstate 45 North, Suite 200
Houston, TX 77386
peter.k...@businessuites.com

From: Peter Kust
Sent: Saturday, March 29, 2014 10:38 AM
To: 'users@lists.opensips.org'
Subject: Rewriting Contact Header -- Should I or Shouldn't I?

I am currently testing an OpenSIPS/Asterisk combination with a GenBand eSBC 
(Quantix QFlex).

My basic architecture looks like this

Phone (Cisco SPA525G2) → OpenSIPS proxy → Asterisk Media Server
Asterisk Media Server → OpenSIPS proxy → GenBand QFlex eSBC (→PSTN)

The GenBand is handling both the SIP and RTP protocols, which means the 
Asterisk Media Server is sending the RTP stream direct to the GenBand.

A problem arises on inbound calls (from PSTN through GenBand to 
OpenSIPS/Asterisk).  During the call setup the GenBand sends a SIP ACK message 
directly to my Asterisk server, which seems to be causing the Asterisk server 
to send the BYE message at the end of the call directly to the GenBand instead 
of via the OpenSIPS proxy.  The result is that the external call end point 
(i.e., my cell phone), never gets a BYE message and that call leg stays open.

In the OK message from the proxy to the GenBand, the Contact header contains 
the IP address of my Asterisk server, and not the proxy.  I am being told this 
is what prompts the GenBand to send to the Asterisk server and not the proxy.

>From a packet capture I have run on the offending call scenario, the OK 
message in questi

Re: [OpenSIPS-Users] RES: RES: Error in Module Permissions

[Stefano Pisani]

So I was right :-)

Il 18/03/2014 20.11, Alcindo Schleder ha scritto:


Hi all..

I found the error. The version of the module was incorrect. The 
opensips was loaded through the rpm package version 1.9.2. I 
downloaded the source and recompiled ... voilà it worked.


[]s

*De:*users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] *Em nome de *Laszlo

*Enviada em:* terça-feira, 18 de março de 2014 15:41
*Para:* OpenSIPS users mailling list
*Assunto:* Re: [OpenSIPS-Users] RES: Error in Module Permissions

that sounds interesting.

if you replace your check_address row with this, what do you get back?

if (check_address("1","$si","$sp","$proto","")) {

2014-03-18 18:56 GMT+01:00 Alcindo Schleder >:


Hi Laszli.

With 4 parameters returns the same error.

*De:*users-boun...@lists.opensips.org 
 
[mailto:users-boun...@lists.opensips.org 
] *Em nome de *Laszlo

*Enviada em:* terça-feira, 18 de março de 2014 14:24


*Para:* OpenSIPS users mailling list
*Assunto:* Re: [OpenSIPS-Users] Error in Module Permissions

the function expects 4 mandatory parameters, not 3

- Laszli

2014.03.18. 18:04, "Alcindo Schleder" > ezt írta:


I'm trying to use the function and check_address get an error 
compiling the script.


Script line 390.

if (check_address("1", "$si", "$sp")) {

xlog("IP Allow Routing to $si");

}

Error:

CRITICAL:core:yyerror: parse error in config file 
/etc/opensips/opensips.cfg, line 396, column 38-39: unknown command 
, missing loadmodule?


Alcindo Schleder

Gerente de Negócios -- comerc...@renovaretelecom.com.br 



(51) 3564-4156

(51) 9790-9437

Renovare Telecom 


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--

Kind regards,

Laszlo Bekesi

http://voipfreak.net



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Re: [OpenSIPS-Users] Error in Module Permissions

[Stefano Pisani]

missing loadmodule?

Il 18/03/2014 18.03, Alcindo Schleder ha scritto:


I'm trying to use the function and check_address get an error 
compiling the script.


Script line 390.

if (check_address("1", "$si", "$sp")) {

xlog("IP Allow Routing to $si");

}

Error:

CRITICAL:core:yyerror: parse error in config file 
/etc/opensips/opensips.cfg, line 396, column 38-39: unknown command 
, missing loadmodule?


Alcindo Schleder

Gerente de Negócios -- comerc...@renovaretelecom.com.br 



(51) 3564-4156

(51) 9790-9437

Renovare Telecom 



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Re: [OpenSIPS-Users] too many hops

[Stefano Pisani]

It seems an addressing issue.
Could you post your opensips.cfg?
Are you able to log the sip session using ngrep? Post it.

Il 10/03/2014 11.03, Mike Claudi Pedersen ha scritto:

im trying to establish connection between 2 phones
user1: 43384001
user2: 43384002

i have user 2 added to usrloc with the cmd:

opensipsctl ul add 43384003 sip:+1043384003*2...@voip.local.com 


which is an asterisk server that handles calls in a specific way.

but when i try to call i get "too many hops"

this is my invite handling

if (method=="INVITE")
{
t_relay();
exit;
};

do i need some sort of line added to include using usrloc
--
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ipnordic A/S

Mike Claudi Pedersen
Tekniker

Telefon: 79301033
www.ipnordic.dk 


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Re: [OpenSIPS-Users] Redirect

[Stefano Pisani]

Simply replace the $ru ($ru = NEWURI) and the call goes to the right 
destination


If the remapping is fixed you and use ALIAS. for example.

Il 10/03/2014 08.11, Mike Claudi Pedersen ha scritto:
Can someone please help me ind the right direction, i need to 
implement a system to rewrite the destination URI from a list of 
possible destinations.

eg
call from 20304050
if(method == "INVITE") {
 "check where 20304050 is"

then return and replace 20304050 with something like 
+4520304050@voip123.local and redirect


--
Med venlig hilsen
ipnordic A/S

Mike Claudi Pedersen
Tekniker

Telefon: 79301033
www.ipnordic.dk 


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Re: [OpenSIPS-Users] Renegotiation

[Stefano Pisani]

You can trap the 415 response from the called peer and send the call 
through asterisk to get transcoding.


Il 27/02/2014 16.51, Jorge Ortea ha scritto:

Hi all,

I have a scenario with OpenSIPS 1.8 and Asterisks 1.4.   Proxy SIP has 
two ways to manage a call, the first is B2BUA and second is be relay 
between UAC and Asterisk.


I have a problem, when OpenSIPS works as B2BUA and both UAC can't 
negotiate codec then this call failed. I would like restart this same 
call at second way (with Asterisk). Is that possible?


Thanks.
Regards.


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Re: [OpenSIPS-Users] Adding Proxy-Authorization header

[Stefano Pisani]

You can use module UAC_AUTH

Il 24/02/2014 16.18, Diego Barberio ha scritto:

Hi all,

I have opensips registered to an IP-PBX using registrant module and I 
want to make an outbound call to that PBX through the proxy.


I'm sending and INVITE from my application to the proxy with a From 
that is actually registered by the proxy, however OpenSIPs is not 
adding the Proxy-Authorization header so the INVITE is rejected with a 
401 Unauthorized and that response is forwarded to my application.


I just want opensips to add the Proxy-Authorization header so the call 
is not rejected by the IP-PBX. Is it possible to achieve this?


Thanks
Diego


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Re: [OpenSIPS-Users] Initializing SIP messages from routing

[Stefano Pisani]

You can develop something using perl and Net::SIP module.

Il 13/02/2014 19.45, Jayesh Nambiar ha scritto:

Hi,
CRBT is caller ring back tone. What you are primarily looking at is 
sending the INVITE to some b2bua like FreeSWITCH or Asterisk where you 
control both legs of the call. So when you get a ringing signal from 
the B-leg, you play some media file on the A-leg.


--- Jayesh

On Friday, February 7, 2014, Bogdan-Andrei Iancu > wrote:


Hello,

No sure what CRBT stands for, but it looks to me that you need to
use B2B module - what you are trying to do is something more than
simply proxying a call.

Regards,

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

On 07.02.2014 19:00, H Yavari wrote:

Hi Bogdan,
thanks for your answer. I want to implement CRBT. For this I want
when the invite received,I send an invite to media server and
play something.
I can do this with B2BUA? I write a module for this or do with
script? script running has side effect on performance when load
is high?

Regards,
H.Yavari



Hello,

Typically you process in OpenSIPS script an incoming request (and
you fwd or reply it). It is unusual to generate a new request
while processing another one.

May I ask about your scenario ?

Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com  
On 28.01.2014 10:02, H Yavari wrote:

Hi to all openSIPSer,
I want to initialize a sip message from routing, is it possible?
If only way that do this is writing script, is this efficient
when load on openSIPS is high?
If answer is yes, can you give me some example to how do this?

Regards,
H.Yavari


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Re: [OpenSIPS-Users] media server behind nat

[Stefano Pisani]

There are a problem in the network configuration.
Opensips has only one network card, in this way it cound only use one IP 
but it actually has two IPs (one external and one interna).
You need to enable a second network card with a public IP (in the DMZ)  
and use mhomed=1 in configuration to change IP in SIP header according 
to interface used.


Or you can use (if available and working properly) the SIP ALG in you 
firewall instead of advertising_address.


The third option is to create DNAT (with different port) also for media 
server so all server are on public IP.


You can't mix public and private IP as you done using a single network card.

s

Il 06/02/2014 03.47, Tony Ward ha scritto:


*From:*Tony Ward
*Sent:* Wednesday, February 05, 2014 3:43 PM
*To:* 'stefano.pis...@omnianet.it'; 'OpenSIPS users mailling list'
*Subject:* RE: [OpenSIPS-Users] media server behind nat

Sorry for the lack of clarity.  Network drawing is at 
http://www.ais-rx.com/pub/network.jpg


Thanks,

Tony



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Re: [OpenSIPS-Users] media server behind nat

[Stefano Pisani]

Please post a link with your network drawing. Your description is unclear.

Il 05/02/2014 20.19, Tony Ward ha scritto:


Hello,

I currently have a media server behind a nat firewall with calls 
delivered via a PSTN Trunk.  I want to add a 2nd media server and 
route calls to either depending upon the dialed number.  I've been 
 trying to do this using drouting in opensips 1.10.0, but cannot get a 
configuration that works.


I started by generating the 'trunking script' using make menuconfig, 
and populated mysql to accept my PSTN trunk and route to my media 
server.   When an incoming call arrives, it is directed to opensips, 
and forwarded to media server with a record-route header containing my 
private ip.  This confuses my PSTN partner and we are unable to 
establish the rtp stream.


After reviewing the mailing lists I tried setting alias and 
advertised_address=my public ip.  Now when an incoming call arrives it 
is directed to opensips and forwarded to the media server with a 
record-route header containing my public ip.  Call setup completes 
successfully. Call teardown initiated from PSTN trunk completes 
successfully.  Call teardown initiated from media server fails because 
the media servers sends BYE  to the public IP, and the NAT router does 
not know what to do with it (destination unreachable).


It seems as though the invite to my media server needs to have a 
record-route header with my private ip, while the ok response back to 
my PSTN provider needs to have a record-route header with my public 
ip.  Is this the right approach?  I've briefly toyed with rtpproxy and 
also b2bua without much luck, and was hoping this simpler solution 
could be made to work.


Thanks,

Tony



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Re: [OpenSIPS-Users] check if ip address belongs to ip and subnet subscriber

[Stefano Pisani]

if( ( $(avp(sourceip_mask){ip.pton}) & $(avp(sourceip){ip.pton}) ) == (
$(avp(sourceip_mask){ip.pton}) & $(si{ip.pton}) ) )
{
 xlog("L_INFO", " ip $si belongs to $au\n");
}
else
{
 xlog("L_INFO", " ip $si does not belong to $au\n");
 sl_send_reply("403", "Forbidden");
 exit;
}

Why you do not use directly the net address? Why do you need to do 
($(avp(sourceip_mask){ip.pton}) & $(avp(sourceip){ip.pton})) each time 
if the result is always the same? Put the result in your db instead of 
sourceip.


if( ($(avp(sourceip_mask){ip.pton}) & $(si{ip.pton}) ) == 
$avp(sourceip_net) ) ...




Il 01/02/2014 18.15, Edwin ha scritto:

I tried () also, but this resulted in an error too ):

For the [] part i followed the docs:
http://www.opensips.org/Documentation/Script-Operators
Arithmetic expressions can be used in condition expressions via test
operator ' [ ... ] '.

But I'm not the expert here, any help is appreciated!



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Re: [OpenSIPS-Users] check if ip address belongs to ip and subnet subscriber

[Stefano Pisani]

why you are using "[]"? use "()" instead.

Il 01/02/2014 17.44, Edwin ha scritto:

This helped a bit, so I came up with:

$var(sourceip_net) = $(avp(sourceip_mask){ip.pton}) &
$(avp(sourceip){ip.pton});
$var(si_net) = $(avp(sourceip_mask){ip.pton}) & $(si{ip.pton});

if($var(sourceip_net) == $var(si_net))
{
 xlog("L_INFO", " ip $si belongs to $au\n");
}
else
{
 xlog("L_INFO", " ip $si does not belong to $au\n");
 sl_send_reply("403", "Forbidden");
 exit;
}

But I like to write i like this:

if( [ $(avp(sourceip_mask){ip.pton}) & $(avp(sourceip){ip.pton}) ] == [
$(avp(sourceip_mask){ip.pton}) & $(si{ip.pton}) ] )
{
 xlog("L_INFO", " ip $si belongs to $au\n");
}
else
{
 xlog("L_INFO", " ip $si does not belong to $au\n");
 sl_send_reply("403", "Forbidden");
 exit;
}

But this gives an error (column 121-123: syntax error, column 121-123: bad
command!)



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Re: [OpenSIPS-Users] check if ip address belongs to ip and subnet subscriber

[Stefano Pisani]

$var(mask) = "255.255.0.0";
$var(ip) = "192.168.2.134";
$var(net) = $(var(mask){ip.pton}) & $(var(ip){ip.pton});

if ($(var(net){ip.ntop}) == "192.168.0.0")
 xlog("IP is in 192.168.0.0/16 network\n");

Il 30/01/2014 20.47, Edwin ha scritto:

Stefano,

I know,  and maybe I will use the permission module if needed, but the
question / topic is about using the ip.pton function.

I would appreciate if you (or anyone else) could help me with this question.
Use and output of ip.pton is still opaque to me...

Thanks,

Edwin



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Re: [OpenSIPS-Users] check if ip address belongs to ip and subnet subscriber

[Stefano Pisani]

You can create an external perl script that to the job.

Il 30/01/2014 12.38, Edwin Haselhoff ha scritto:

Stefano,

I tested the permission module but changes to the table are not 'real 
time', I have to reload the table every time (or did I miss something?).




Stefano Pisani schreef op 30-1-2014 12:32:

Hi,
use module permission.

s

Il 30/01/2014 12.21, Edwin Haselhoff ha scritto:

Hi all,

For security reasons I want to check if the $si ip is part of ip and 
subnet of a subscriber so added '$(avp(sourceip)' and 
'$(avp(sourceip_mask)' to the subscriber table.
(I know I can use permissions module, but this is in cache and we 
like to make changes real time without haveing to reload the table 
in cache)


I tried something like this where sourceip_net and sourceip_mask is 
the ip subnet belonging to the subscriber:


$si = 10.100.5.42 (1011010000101010)
$avp(sourceip_net) = 10.20.30.40; (1011010000101000)
$avp(sourceip_mask) = 29;

if($si{ip.isip} && $(si{ip.pton}{s.substr,0,$avp(sourceip_mask)}) == 
$(avp(sourceip_net){ip.pton}{s.substr,0,$avp(sourceip_mask)}))

{
xlog("L_INFO", " ip $si belongs to $au\n");
}
else
{
xlog("L_INFO", " ip $si does not belong to $au\n");
sl_send_reply("403", "Forbidden");
exit;
}


So I expect the ip is valid and the comparison is true 
(1011010000101 = 1011010000101) but 
it doesn't seem to work like I expect.


It's difficult to output ip.pton to xlog (unreadable). Does it 
output a binary format like I expect?


Any ideas how to accomplish this?

Thanks,

Edwin


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Re: [OpenSIPS-Users] How to connect opensips to esternal VOIP server?

[Stefano Pisani]

UAC_REGISTRAR is not enought becouse you need something else to make 
call with proxy autentication like UAC_AUTH and failure_route


Il 30/01/2014 17.40, Nikita Tarasov ha scritto:
Where are any ability to make calls from opensips with registration? 
For example registered with uac_registrant?


*From: *Tito Cumpen
*Sent: *Thursday, 30 January 2014 20:36
*To: *OpenSIPS users mailling list
*Reply To: *OpenSIPS users mailling list
*Subject: *Re: [OpenSIPS-Users] How to connect opensips to esternal VOIP
server?


Fabio,

It depends on what your carrier requires you to do. Are you dependent 
on registration ? if this is simply fowarding an invite you just need 
to follow the tutorial below but take in mind the opensips does not 
proxy media unless you apply rtp proxy or media proxy. Simply replace 
the ip in rewritehostport with the carriers ip.

http://vidodz.wordpress.com/2009/07/28/route-calls-from-openseropensips-to-asterisk/


On Thu, Jan 30, 2014 at 10:36 AM, > wrote:


Hi to all!
I joined to this mailing list to do a simple (maybe) question.

I've setup an opensips server, i can register voip phone, but i
don't understand how to connect opensips to external (internet)
trunks, how i was doing it with asterisk (is just an example to
better explain ). Is this possible? Or i haven't understand for
what opensips is made to? If is it possible how i could do it?
Thanks in advice for your reply.

Regards, Fabrizio Pappolla.

PS I sent this message in reply to  mailing list subscription, i
hope this isn't a duplicate.

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Re: [OpenSIPS-Users] check if ip address belongs to ip and subnet subscriber

[Stefano Pisani]

Hi,
use module permission.

s

Il 30/01/2014 12.21, Edwin Haselhoff ha scritto:

Hi all,

For security reasons I want to check if the $si ip is part of ip and 
subnet of a subscriber so added '$(avp(sourceip)' and 
'$(avp(sourceip_mask)' to the subscriber table.
(I know I can use permissions module, but this is in cache and we like 
to make changes real time without haveing to reload the table in cache)


I tried something like this where sourceip_net and sourceip_mask is 
the ip subnet belonging to the subscriber:


$si = 10.100.5.42 (1011010000101010)
$avp(sourceip_net) = 10.20.30.40; (1011010000101000)
$avp(sourceip_mask) = 29;

if($si{ip.isip} && $(si{ip.pton}{s.substr,0,$avp(sourceip_mask)}) == 
$(avp(sourceip_net){ip.pton}{s.substr,0,$avp(sourceip_mask)}))

{
xlog("L_INFO", " ip $si belongs to $au\n");
}
else
{
xlog("L_INFO", " ip $si does not belong to $au\n");
sl_send_reply("403", "Forbidden");
exit;
}


So I expect the ip is valid and the comparison is true 
(1011010000101 = 1011010000101) but it 
doesn't seem to work like I expect.


It's difficult to output ip.pton to xlog (unreadable). Does it output 
a binary format like I expect?


Any ideas how to accomplish this?

Thanks,

Edwin


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Re: [OpenSIPS-Users] changing header contact

[Stefano Pisani]

If you have public and private IP on different eth ports you can use 
mhost = 1


Il 27/01/2014 20.58, discodo...@aol.com ha scritto:

Hello all,
I am trying to setup a opensips server that has 2 ip's  one internal 
and one external IP.  I have set the rewritehostport to an external 
IP.  When I send the call to the internal NICs IP the invite makes it 
to the external server but my to: from: and header contact all show 
the original IPs and the header contact is the internal NICs IP.


Should I be able to configure the open sips server to branch a new 
call that shows the call originating from the external NIC IP of the 
opensips server?  I have setup rtpproxy that is working fine changing 
the SDP contact and media ports to the external NIC's IP.  Can anyone 
point me in the correct direction?


Thanks,

James


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Re: [OpenSIPS-Users] Ref: how can we replace URI Header more than once per call?

[Stefano Pisani]

The uac_replace_to can be used once.
Are you sure you need to replace to header? Could be enough set a new 
RURI instead.

What do you want do exactly?

regards,

Il 09/12/2013 12.57, AMPTEL PTY LTD | RuvixTel ha scritto:


Hi all

Just wondering, if anyone able to assist us with below:

We are using Opensips 1.6 and we like to know: how can we replace URI 
Header more than once per call?


uac_replace_to function is there but when we try this second time: 
instead of replacing it. It just appended two headers.


What we are trying to achieve here is call Fail Over function? When 
one IP is unable to terminate a call it will try next IP from the 
priority list.


Your assistance will be highly appreciated.

If this issue has been discussed earlier please direct me to the 
thread please.



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Anup




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Re: [OpenSIPS-Users] Can I use alias_db to change credentials?

[Stefano Pisani]

uac_auth() is buggy. It does not increment the CSeq and that is a big 
problem.




Il 25/11/2013 09.33, Walter Klomp ha scritto:
Actually, I found this before, but this only works for 1 specific 
general username and password for the PSTN gateway. I need this "per 
user", how do I configure that ?


I.e. call A with userA,passwordA and call B with userB, passwordB ...

Nobody can help me ?

Walter.


On 25/11/13 3:52 pm, Rik Broers wrote:
I think this article contains everything you need exactly :) (at 
least for the authentication part)


http://docs.huihoo.com/opensips/tutorials/uac/ar01s06.html

You need the UAC modules for this.

Regards,
Met vriendelijke groet,

Rik Broers
Voice Engineer








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-Original Message-
From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Walter Klomp

Sent: zondag 24 november 2013 16:43
To: stefano.pis...@omnianet.it; OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Can I use alias_db to change credentials?

Hi Stefano,

Thanks for the confidence it can be done easily, but I have not seen 
any example of how to do it... I do not want to have to resort to 
sending the call to asterisk where I then call the pstn softswitch 
with the proper credentials, I'd like to do that directly in opensips...


i.e. PSTN -> opensips (replacing username/password & new destination
number) -> Softswitch -> PSTN

is there a module I can use, (using opensips 1.8.x), and how exactly 
do I use it?  (I know how to pull the username and password and 
destination number from the database)


an example would greatly help me.

Thanks
Walter.

On 24/11/13 4:46 pm, Stefano Pisani wrote:

You can query a ad using avp_query and get new credentials, then use
them in the routing script.
This is not complicated.



Il 24/11/2013 01.38, Walter Klomp ha scritto:

Hi,

I want to enable call forwarding feature to pstn, but the outbound
calls need to be authenticated to the outbound proxy server with the
credentials of the user (callee) to be billed. How do I go around
doing this? Callerid should remain intact, would optionally want to
add the countrycode in front.

Anybody has done this before and can give me some pointers?

Thanks in advance
Walter
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Re: [OpenSIPS-Users] Can I use alias_db to change credentials?

[Stefano Pisani]

You can query a ad using avp_query and get new credentials, then use 
them in the routing script.

This is not complicated.



Il 24/11/2013 01.38, Walter Klomp ha scritto:

Hi,

I want to enable call forwarding feature to pstn, but the outbound calls need 
to be authenticated to the outbound proxy server with the credentials of the 
user (callee) to be billed. How do I go around doing this? Callerid should 
remain intact, would optionally want to add the countrycode in front.

Anybody has done this before and can give me some pointers?

Thanks in advance
Walter
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Re: [OpenSIPS-Users] Re A bridge Call via Opensips

[Stefano Pisani]

That is the parallel forking.
You should see documentation about it.

The simplest solution is to register the 3 SIP phones using the same SIP 
account (ID:200)


Il 11/11/2013 07.01, steven chew ha scritto:

Hi All,

How are you?

I would like to know how to configure a bridge call in opensips.cfg, 
the bridge call's situation would be like:


- One SIP Phone (ID:100) can dial to a ID:200 which will make 3 SIP 
Phones ringing.
- One of three SIP Ringing Phones answers the call from 200, the other 
two sip phones of three will stop ringing.


Is it possible to achieve this kind of the bridge call?

I hope to hear from your kindly reply a.s.a.p..

Thanks
Kind Regards,
Steven


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Re: [OpenSIPS-Users] Modify/Edit INVITE

[Stefano Pisani]

take a look to textops module

Il 09/11/2013 16.06, Aziz Bayd ha scritto:

Hello.

  I have a question about OpenSIPS configuration.
Can I edit or modify the INVITE message content? For example, remove
  "To" or "Via" fields in the INVITE content.
  If yes, how can I do?
  Thanks.

2013/11/9, Aziz Bayd :

Hello.

I have a question about OpenSIPS configuration.
Can I edit or modify the INVITE message content? For example, remove
"To" or "Via" fields in the INVITE content.
If yes, how can I do?
Thanks.


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Re: [OpenSIPS-Users] Multiple aliases for one sip account

[Stefano Pisani]

I agree.

Il 30/10/2013 01.38, Dario Busso ha scritto:


La tua richiesta è veramente poco chiara.
Your question is really confused. May you try to describe it better?

-ddB

Il 28/ott/2013 18:00 "Manuela Pigini" > ha scritto:


Hello,
we are developing a voip card for a traditional pbx. The voip card
contains 4
ATA for the link with the PBX. We installed Opensips on the board
and it's
all ok, but now we would like to support calls from remote
offices, where we
would like to dial as a local phone of our traditional pbx, for
example: we
have 418 as local number of the traditional pbx, and from the
remote office
we would like to dial 418 and speak with the 418 local end. What
is the best
solution? We thought to use "aliases" for the 4 ATAs on the Voip
card, but we
would need several different aliases for the ATAs, as many as the
local
numbers of the traditional PBX (is this possible?)...otherwise?

Thank you all for the support.
Manuela


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Re: [OpenSIPS-Users] Opensip as transaprent inbound proxy for asterisk

[Stefano Pisani]

Just FYI
you need to write down some code in asterisk to manage the new header
obviusly



Il 11/10/2013 02.30, bluerain ha scritto:

Just FYI, I tried, I insert your line in the method invite and right before
the routing, Asterisk didn't seem to care.  It still care about the prior
Hop IP.

So what I mean is that

from 199.33.33.33 --> opensip 22.55.33.33 (and then I put your line) -->
Asterisk server.

Asterisk server identified the call came from 22.55.33.33 and not
199.33.33.33

Frank



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Re: [OpenSIPS-Users] Opensip as transaprent inbound proxy for asterisk

[Stefano Pisani]

You do not need to manipulate core variables. You have to add a header 
to pass the source ip to asterisk.


esample append_hf("X-src-ip: $si\r\n")


Il 10/10/2013 02.05, bluerain ha scritto:

Are you sure?  Can you tell my which function call in opensips?  I know how
to manipulate the core variable, but $si is read only.  And I think if you
define a "peering" resource in asterisk, it will try to match it by the
source IP at the network layer and not within the INVITE.  Please tell me
which function to manipulate in opensips and I can try.

Thx



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Re: [OpenSIPS-Users] Opensip as transaprent inbound proxy for asterisk

[Stefano Pisani]

opensips can add an header with the real IP
and asterisk can use that header to know the real IP

Il 09/10/2013 17.02, bluerain ha scritto:

I've try to search on internet but not much info.  I currently have Asterisk
server setup to have sip trunk with customers on a "peer" type.  This way,
no registration need and that asterisk server will identify the inbound call
base on "IP address" matching.  But now I would like to put OPENSIPS in
front of asterisk server (so I can load balance).  But once I do that,
asterisk loses the capability of getting the "original" source IP.  All the
call from different customer now only have OPENSIPS server IP address.  $si
is not ediable.  It seems asterisk server read source IP at a higher layer (
I am no network guru so pardon my stupid language).

Is this something I have to tackle at the OS level and not at OpenSIP leve?
Opensips only manipulate sip message where as asterisk is reading the source
IP at the network layer?

Or is there something Opensips can do?  Or can someone point me to a
software (prefer open source) that would make Debian a "transparent proxy"?
I've search for that but mostly come up with 'SQUID' which is a transparent
HTTP server, I don't think that is for sip protocol.

Thank you!



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Re: [OpenSIPS-Users] calling external command with sudo

[Stefano Pisani]

are you sure to know how to configure sudo?
please post the user used by openSIPS and sudo configuration (sudoers)



Il 20/09/2013 19.11, Dragomir Haralambiev ha scritto:

Hello everyone,
I am trying to execute 'iptables' from opensips in the script, which 
works if opensips runs as root. However if opensips is configured to 
run as non-privileged user, in order to control iptables, I have to 
call iptables via sudo. Command works on the command prompt when 
executed manually under the user opensips run as, but does not seem to 
execute if run by opensips itself.

This is how I try to call it:
exec_msg("/usr/bin/sudo /sbin/iptables -A INPUT -s $si -p udp -j DROP");
log file says command is executed, but the firewall rules do not get 
updated.

Any idea why?


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Re: [OpenSIPS-Users] WebRTC : Integration with opensips.org free VoIP service & Tutorial

[Stefano Pisani]

Hi Vlad,
yes, it show connected.
I'm trying to call an external SIP URI.

It works only inside @opensips.org?

Thanks
Stefano

Il 05/07/2013 14.14, Vlad Paiu ha scritto:

Hello,

So when you go to 'web calls' and hit the 'Login' button, the app 
successfully registers your SIP account against opensips.org and shows 
'Connected' ?
If yes, the username that you are trying to call, is it also logged in 
on the website, in the 'web calls' section ? If it's using a regular 
soft/hard phone, does the phone have webRTC capabilities ?


Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 07/04/2013 07:35 PM, Stefano Pisani wrote:

I tried to call a SIP URI but it do not seems to be working.
I used crome. The connection works but it cannot place the call.

s

Il 04/07/2013 15.55, Vlad Paiu ha scritto:

Hello,

The free VoIP service offered by opensips.org has now been enhanced 
in order to support WebRTC calls.
In order to test it, you can login to your account at [1] and go to 
'web calls' in the left menu. The integrated client supports both 
audio and video calls between two parties.


Also, we have added a new tutorial, available at [2], which shows 
how to add WebRTC capabilities to any existing OpenSIPS-based 
deployment.
The tutorial makes use of an OpenSIPS deployment with NAT support, 
and adds WebRTC capabilities on top of that by using OverSIP as a WS 
to SIP gateway and sipML5 as the web client.


[1] https://www.opensips.org/account/
[2] http://www.opensips.org/Documentation/Tutorials-WebSocket

Best Regards,
--
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com


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Re: [OpenSIPS-Users] WebRTC : Integration with opensips.org free VoIP service & Tutorial

[Stefano Pisani]

I tried to call a SIP URI but it do not seems to be working.
I used crome. The connection works but it cannot place the call.

s

Il 04/07/2013 15.55, Vlad Paiu ha scritto:

Hello,

The free VoIP service offered by opensips.org has now been enhanced in 
order to support WebRTC calls.
In order to test it, you can login to your account at [1] and go to 
'web calls' in the left menu. The integrated client supports both 
audio and video calls between two parties.


Also, we have added a new tutorial, available at [2], which shows how 
to add WebRTC capabilities to any existing OpenSIPS-based deployment.
The tutorial makes use of an OpenSIPS deployment with NAT support, and 
adds WebRTC capabilities on top of that by using OverSIP as a WS to 
SIP gateway and sipML5 as the web client.


[1] https://www.opensips.org/account/
[2] http://www.opensips.org/Documentation/Tutorials-WebSocket

Best Regards,
--
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com


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Re: [OpenSIPS-Users] uac_replace_to problem

[Stefano Pisani]

Hi ,
have you changed to twice?
This happens if you try to use uac_replace_to (or uac_replace_from) twice.

s

Il 19/06/2013 15.18, M.Khaled W Chehab ha scritto:


while i am using uac_replace_to in failover route branch i can find 
that TO header is not changed(sip user part ) but appended an new raw 
 as I want it to be To: "971552448304" 


SIP to address: 
sip:835822971552448...@xx.xx.xx.55sip:808971552448...@xx.xx.xx.55


SIP to address User Part: 835822971552448304

SIP to address Host Part: xx.xx.xx.55sip

SIP to address Host Port: 808971552448...@xx.xx.xx.55 



please advice

regards

*From:*M.Khaled W Chehab [mailto:kche...@icucall.com]
*Sent:* Wednesday, June 19, 2013 2:41 PM
*To:* users@lists.opensips.org
*Cc:* users-boun...@lists.opensips.org
*Subject:* uac_replace_to problem

Hi,

I am running opensips 1.8.3 with  do_routing module

A dial_rule prefix has 3 trunk gateways ( gw1,gw2,gw3)

After do_routing(,)

I am setting the $rU and fixing  the To Tag header using 
uac_replace_to( and the call go to gw1 with correct TO tag as I set it ,


but when calls fails on gw1 ,then  I set the $rU in route[6] to go to 
second in route and it goes with bad TO header, since it goes with the 
same To header in the 1^st invite


That target gw1

1-how to fix the To header in the second invite to gw2

loadmodule "uac.so"

modparam("uac","restore_mode","auto")

uac_replace_to("$avp(dst)","sip:$var(prefix)$avp(dst)@$td 
");


t_on_failure("1");

on failure_route[1] {

.

if (!t_check_status("487")) {

#xlog("route6---\n");

$avp(failure_count) = $avp(failure_count) + 1; #480|486|603

route(6);

}

Route[6]{

.

.

.

$rU = $var(prefix) + $avp(dst);

uac_replace_to("sip:$var(prefix)$avp(dst)@$td 
");


}

Regards



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Re: [OpenSIPS-Users] OpenSips and Ekiga

[Stefano Pisani]

Could you get some log using ngrep?

regards
s

Il 04/06/2013 00.50, Alberto Ayala ha scritto:

Hello subscribers

I'm trying to integrate OpenSips and Ekiga

I was able to install using EPEL packages in a CentOS 5.x machine

opensips mysql is working. (I create two users for testing purpouse)

when I login to the server using ekiga (sip server I see my status as 
register)


and when I try to call I get in the log resource busy:

May 30 10:37:30 localhost /usr/sbin/opensips[2797]: ACC: call missed: 
timestamp=1369935450;method=INVITE;from_tag=981a3e3c-080a-1910-909d-08002700c8d5;to_tag=d20e3a3c-080a-1910-8dd7-005056c1;call_id=981a3e3c-080a-1910-909e-08002700c8d5@bluebox;code=487;reason=Request 
Terminated
May 30 10:42:20 localhost /usr/sbin/opensips[5148]: ACC: call missed: 
timestamp=1369935740;method=INVITE;from_tag=4a45ef3d-080a-1910-8dda-005056c1;to_tag=e173f33d-080a-1910-90af-08002700c8d5;call_id=ae45ef3d-080a-1910-8dda-005056c1@fei-pc;code=487;reason=Request 
Terminated
May 30 10:58:35 localhost /usr/sbin/opensips[5147]: ACC: call missed: 
timestamp=1369936715;method=INVITE;from_tag=f697b243-080a-1910-9351-08002700c8d5;to_tag=;call_id=f697b243-080a-1910-9352-08002700c8d5@bluebox;code=487;reason=Request 
Terminated
May 30 10:59:24 localhost /usr/sbin/opensips[5150]: ACC: call missed: 
timestamp=1369936764;method=INVITE;from_tag=647d1944-080a-1910-9355-08002700c8d5;to_tag=;call_id=967d1944-080a-1910-9355-08002700c8d5@bluebox;code=487;reason=Request 
Terminated
May 30 11:03:36 localhost /usr/sbin/opensips[5150]: ACC: call missed: 
timestamp=1369937016;method=INVITE;from_tag=18a19c45-080a-1910-999a-08002700c8d5;to_tag=09ac9c45-080a-1910-999c-08002700c8d5;call_id=18a19c45-080a-1910-999b-08002700c8d5@bluebox;code=486;reason=Busy 
Here
May 30 11:05:17 localhost /usr/sbin/opensips[5149]: ACC: call missed: 
timestamp=1369937117;method=INVITE;from_tag=16da3646-080a-1910-99b7-08002700c8d5;to_tag=a8e33646-080a-1910-99b9-08002700c8d5;call_id=16da3646-080a-1910-99b8-08002700c8d5@bluebox;code=486;reason=Busy 
Here
May 30 13:38:31 localhost /usr/sbin/opensips[29295]: ACC: call missed: 
timestamp=1369946311;method=INVITE;from_tag=d535d47c-080a-1910-9377-08002700c8d5;to_tag=a53dd47c-080a-1910-9379-08002700c8d5;call_id=d535d47c-080a-1910-9378-08002700c8d5@bluebox;code=486;reason=Busy 
Here
May 30 13:39:25 localhost /usr/sbin/opensips[29295]: ACC: call missed: 
timestamp=1369946365;method=INVITE;from_tag=ecc0277d-080a-1910-9384-08002700c8d5;to_tag=e2ca277d-080a-1910-9385-08002700c8d5;call_id=ecc0277d-080a-1910-9385-08002700c8d5@bluebox;code=486;reason=Busy 
Here


any help is going to be appreciated.

cheers.




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Re: [OpenSIPS-Users] How to get only the ip from contact?

[Stefano Pisani]

Use the core variables $(ct.fields(uri){uri.host}) or something like that

best regards
s

Il 29/05/2013 15.43, microx ha scritto:

Hi all,

An example contact uri of an INVIT looks like "sip:111@61.60.228.221:5060".
OpenSIPS seems not to have a
specific variable to store the IP part. Is there an efficient way to get
only the IP part from contact?
Any idea is very welcome.

Best regards,
Chen-Che



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Re: [OpenSIPS-Users] How do I remove an unwanted header w OpenSIPS?

[Stefano Pisani]

  
  
Use 

remove_hf("X-BroadWorks-DNC:network-address");

to remove the unwanted header

s

Il 26/04/2013 19.30, Stacy Trippe ha
  scritto:


  
  
  
  
  
We are running into a problem with the VPN
  overhead, our packets getting too big.  Because of this, we
  are told to reduce the packet size to 1400 bytes.  I’m trying
  to remove all non-necessary headers from the SIP invite,
  generated by our Broadsoft platform.
 
OpenSIPS 1.5.3-notls
 
I need to remove the following header from
  session:
 
X-BroadWorks-DNC:network-address="sip:+15015525956@10.135.224.130;user=phone"
 
In addition, I saw where OpenSER used to
  allow the following core commands:
pmtu_discovery
= yes
udp_mtu = 1400
udp_mtu_try_proto
= TCP
 
I
don’t see any documentation where this is allowed in
OpenSIPS 1.5.x. 
 
How best to handle this?
 

  

  
 
Stacy
Trippe
Network
Operations
TecInfo,
Inc
662.686.9009
662.686.5656
Fax
o...@corp.tecinfo.net
  
  
 

  
http://www.tecinfo.net/
  

  

 
The
information transmitted is intended only for the person or
entity to which it is addressed and may contain
confidential, proprietary, and/or privileged material. Any
review, retransmission, dissemination or other use of, or
taking of any action in reliance upon this information by
persons or entities without the written permission of
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contact the sender and delete the material from all
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Re: [OpenSIPS-Users] How to protect OpenSIPS from undesidered requests (DoS attack?)

[Stefano Pisani]

Il 06/03/2013 19:58, leo ha scritto:

I've also added Nick's suggestion:
if ($ua =~ "friendly-scanner") {
xlog("L_ERR", "Attack attempt - Request dropped");
drop();
}

But i don't have neither those events in the opensips.log file.


it depends where in the script you added these lines, have you use the 
right place?


s

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Re: [OpenSIPS-Users] mediaproxy behavior

[Stefano Pisani]

Use rtpproxy :-)

s

Il 16/01/2013 09:53, Jorge Ortea ha scritto:

Hi all,

I use OpenSIPS + Mediaproxy and several asterisk behind. I have the 
next problem: I would like SIP Proxy with a public IP and the 
asterisks in private network, but it isn't possible because mediaproxy 
only can forward RTP on the same network interface, this forces that 
each asterisk must have public IP.


How can I resolve it?

Someone know if rtp proxy behaves same or it be able forward rtp from 
public interface to private interface ?


Thanks.
Regards.


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Re: [OpenSIPS-Users] NAT issues on client and server

[Stefano Pisani]

Could you explaint your scenario better?
The server is in a private network?
The cliets are in different private networks?

s

Il 04/01/2013 05:23, Mark Currie ha scritto:


Thanks for the advice Flavio.

Currently I am actually pretty close with my NAT'ed OpenSIPS and 
NAT'ed clients. I am assuming this is because I have a simple NAT 
scenario with full-cone NATs.


I get as far as making a call, but when I pick-up the call only one 
side gets RTP through. I traced the problem to the fact that a local 
address is being put in the "Record Route" section of an ACK response.


I would like to try to get this more controllable scenario working 
first before I launch into new complexity with RTP proxies etc. I 
think that I just need to know how to get the fix_nat_sip() and 
fix_nat_contact() to work. Any pointers with that?


Regards

Mark

*From:*users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] *On Behalf Of *Flavio Goncalves

*Sent:* 03 January 2013 12:27
*To:* OpenSIPS users mailling list
*Subject:* Re: [OpenSIPS-Users] NAT issues on client and server

Hi Mark,

There is no simple way to traverse NAT. Unless all your routers use a 
non symmetric NAT, you will have to use rtpproxy or mediaproxy (you 
can check this with a stun client). OpenSIPS behind NAT make things 
even more complicated. So I suggest that you follow an example with 
rtpproxy or media proxy and also use the OpenSIPS in a valid IP 
address. The setup you are trying to do with OpenSIPS behind NAT is 
possible, but it is even more complex.



Flavio E. Goncalves

2013/1/3 Mark Currie mailto:m...@ziliant.com>>

Hi,

I have a very simple setup for a closed network of users (all NAT'ed) with
one OpenSIPS server (also NAT'ed).

I have managed to solve my first problem with registration by following
previous posts and using fix_nat_register(), but I am still having 
problems

with NAT issues during a call. I know that I probably need to use
fix_nat_contact() and fix_nat_sip() but I can't figure out how to use 
these

properly through the documentation.

I have tried to search for examples of opensips.cfg that suit my scenario
but all the ones I found are complicated with proxies etc. Can someone 
point

me to simple example of opensips.cfg that takes care of NAT?

Regards
Mark Currie


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Re: [OpenSIPS-Users] android native sip client

[Stefano Pisani]

Hi Nick, I'm using Sipdroid without any issue.

s

Il 24/12/2012 06:58, Nick Chang ha scritto:


Hello

Do everyone use android native sip client with opensips??

I try it. I can dial to B phone. B can answer. But it is interrupt 
immediately.


If I change to linphone. It's OK.

Thanks

Nick



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Re: [OpenSIPS-Users] call duration problem

[Stefano Pisani]

That seems to be fine.
Are you sure that the call duration are more than OK-BYE time?

s

Il 25/07/2012 14:02, Francisco Franco ha scritto:

I am using this options

/# - acc params -

modparam("acc", "early_media", 0)
modparam("acc", "report_ack", 0)
modparam("acc", "report_cancels", 0)

modparam("acc", "detect_direction", 0)

modparam("acc", "failed_transaction_flag", 3)
modparam("acc", "log_level", 1)
modparam("acc", "log_flag", 1)
modparam("acc", "log_missed_flag", 2)

modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 2)
modparam("acc", "db_url", "mysql://opensips:***@localhost/opensips")
modparam("acc", "aaa_url", "radius:/etc/opensips/radius/client.conf")
modparam("acc", "aaa_flag", 1)
modparam("acc", "aaa_missed_flag", 2)
modparam("acc", "aaa_extra", "User-Name=$Au; \
 Calling-Station-Id=$from; \
 Called-Station-Id=$to; \
Sip-Translated-Request-URI=$ru; \
 Sip-RPid=$avp(s:rpid); \
 Source-IP=$avp(s:source_ip); \
Source-Port=$avp(s:source_port); \
SIP-Proxy-IP=$avp(s:sip_proxy_ip); \
Canonical-URI=$avp(s:can_uri); \
Billing-Party=$avp(s:billing_party); \
Divert-Reason=$avp(s:divert_reason); \
 User-Agent=$hdr(user-agent); \
 Contact=$hdr(contact); \
 Event=$hdr(event); \
 ENUM-TLD=$avp(s:enum_tld)")/
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Tlf: +34 968 970 037
El 24/07/12 20:53, Stefano Pisani escribió:

Try using this option

modparam("acc", "early_media", 0)

regards,
s

Il 24/07/2012 19:55, Francisco Franco ha scritto:

Hi,

I have opensips 1.6 runing with mediaproxy and have a  problem with 
call duration accounting.


The session duration that is stored in database is total time from 
call start ringing, but for billing correctly, time should be start 
when callee pick up.


How can i solve it?

regards.


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Re: [OpenSIPS-Users] call duration problem

[Stefano Pisani]

Try using this option

modparam("acc", "early_media", 0)

regards,
s

Il 24/07/2012 19:55, Francisco Franco ha scritto:

Hi,

I have opensips 1.6 runing with mediaproxy and have a  problem with 
call duration accounting.


The session duration that is stored in database is total time from 
call start ringing, but for billing correctly, time should be start 
when callee pick up.


How can i solve it?

regards.


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Re: [OpenSIPS-Users] private domain to IP resolution

[Stefano Pisani]

Take a look to drouting module

Il 18/07/2012 13:59, asd asd ha scritto:

Hi Rebecca,

thank you for the proposal.

The requirement is to keep all the configuration inside opensips so 
there would be no need to correlate with any other external system. 
All the proposed solution require an external system to take care of 
private domain to ip address resolution.


In my scenario private domains are not used anywhere outside of sip 
signalling, so there is no need to set up DNS, hosts, ENUM, whatever 
infrastructure for it.


I simply want the opensips to have a table in a database to look for a 
mapping between a private domain received in RURI and the IP addresses 
that serve this private domain. In order to spare the effort I wanted 
to reuse any of available modules for this.


It looks like there is no standard solution for it so I'll have to 
define my own table, but then I'll also have to define all the 
distribution and failover mechanisms as well. This is something I want 
to avoid.


Greetings,
Asd

We have Puppet Enterprise and in your situation, I would use Puppet to 
maintain the state of the host file and to be able to push all the 
needed changes to all of your nodes.


I know it's not a module that you are looking for, but it should 
accomplish your task.


Sent from




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Re: [OpenSIPS-Users] private domain to IP resolution

[Stefano Pisani]

Just create your own DNS. Simply.

s

Il 17/07/2012 21:18, asd asd ha scritto:

Hi,

thanks for an idea, but this would not be manageble for a farm of 
opensips nodes.


I'm looking for a a solution using standard opensips modules or at 
least a db solution using standard opensips db access functions.


Greetings,
Asd

Hi,

Put your domains and their ip addresses in /etc/hosts file (for 
Centos). This would be a linux feature not an OpenSIPS solution.


Regards,
Ali Pey


On Tue, Jul 17, 2012 at 2:27 PM, asd asd  wrote:

Hi,

Given a number of SIP systems connected to opensips that use
private domain names not available in the DNS.

What is the best way to provide interconnection between them
without using DNS or ENUM?

Is there a standard opensips module with a database that can be
used to provide local domain to ip resolution, or do I have to
create a custom table for it?

Thanks!


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[OpenSIPS-Users] ERROR:core:pv_printf: buffer overflow

[Stefano Pisani]

Hello folks,
I get this error using avp_query. How I can increase the buffer size?

Jul 12 17:00:28 ks363596 opensips[24980]: ERROR:core:pv_printf: no more 
space for spec value  [75][1192]
Jul 12 17:00:28 ks363596 opensips[24980]: ERROR:core:pv_printf: buffer 
overflow -- increase the buffer size from [1023]...
Jul 12 17:00:28 ks363596 opensips[24980]: ERROR:avpops:ops_dbquery_avps: 
cannot print the query


Thanks
s

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Re: [OpenSIPS-Users] PROXY behind a NAT

[Stefano Pisani]

Yes, you need advertised_address. Remember, you cannot have internal and 
external client. All client MUST be external


Il 10/07/2012 21:59, Ignacio Gonzalez ha scritto:

So adding this to my configuration file, do I fix the problem?

Do I need to set advertised_address or not?



2012/7/10 Ali Pey mailto:ali...@gmail.com>>

On the reply route try something like this:

onreply_route[1] {
xlog("reply route 1: sequential requests ($rs $rr)\n");
if ( src_ip != onebox_asterisk1_ip && src_ip != onebox_asterisk2_ip ){
if ( nat_uac_test( "31") ) {
fix_nated_contact();
force_rport();
if( has_body( "application/sdp" ) ) {
fix_nated_sdp( "3" );
}
}
}
}


Regards,
Ali Pey


On Tue, Jul 10, 2012 at 2:00 PM, Ignacio Gonzalez
mailto:mylan...@gmail.com>> wrote:

Hello everybody, I installed and configured opensips with a
residential configuration setting NAT Traversal and
Multidomain options with the opensips configuration tool. When
I try to make a call with two users, the ACK method of the
caller party never reaches the called party.

This are the messages received in the called party:


INVITE sip:ralvarez@domain;transport=udp SIP/2.0
Via: SIP/2.0/UDP

10.151.199.52:45084;branch=z9hG4bK-d8754z-06f2fafe119979f6-1---d8754z-;rport
Max-Forwards: 70
Contact: 
To: 
From: ;tag=2627e587
Call-ID: ZTg3YjI0ZGNiYjVjZjZlMDRlNjU1Mzg1ZGJkMWQ0NmQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY,
MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest

username="2427001",realm=:"domain",nonce="4ffbbd4200478de4c68512245fee27f7dadd4549c048",uri="sip:ralvarez@domain;transport=udp",response="8fe20af4d2a381cc6a4ebd3ac1162f69",algorithm=MD5
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 410

v=0
o=- 12986371568156250 1 IN IP4 10.151.199.52
s=CounterPath X-Lite 4.1
c=IN IP4 10.151.166.52
t=0 0
a=ice-ufrag:6079ef
a=ice-pwd:8febe129544f5ec86474270bef1a9214
m=audio 53146 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 10.151.199.52 53146 typ host
a=candidate:1 2 UDP 659134 10.151.199.52 53147 typ host

SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP

10.151.199.52:45084;received=201.255.1.30;branch=z9hG4bK-d8754z-06f2fafe119979f6-1---d8754z-;rport=45084
To: 
From: ;tag=2627e587
Call-ID: ZTg3YjI0ZGNiYjVjZjZlMDRlNjU1Mzg1ZGJkMWQ0NmQ.
CSeq: 2 INVITE
Server: OpenSIPS (1.8.0-notls (i386/linux))
Content-Length: 0

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP

10.151.199.52:45084;received=201.255.1.30;branch=z9hG4bK-d8754z-06f2fafe119979f6-1---d8754z-;rport=45084
Record-Route: 
*Contact:
*
To: ;tag=e637fa0f
From: ;tag=2627e587
Call-ID: ZTg3YjI0ZGNiYjVjZjZlMDRlNjU1Mzg1ZGJkMWQ0NmQ.
CSeq: 2 INVITE
User-Agent: X-Lite release 1002tx stamp 29712
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP

10.151.199.52:45084;received=201.255.1.30;branch=z9hG4bK-d8754z-06f2fafe119979f6-1---d8754z-;rport=45084
Record-Route: 
*Contact:
*
To: ;tag=e637fa0f
From: ;tag=2627e587
Call-ID: ZTg3YjI0ZGNiYjVjZjZlMDRlNjU1Mzg1ZGJkMWQ0NmQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY,
MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp

 More OK responses ( retries )

BYE sip:2427001@201.255.1.30:45084;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.100.1.220;branch=z9hG4bKe265.bd09554.0
Via: SIP/2.0/UDP

192.100.1.200:30612;received=192.168.1.200;branch=z9hG4bK-d87543-4677b2285a79346c-1--d87543-;rport=30612
Max-Forwards: 69
Contact:

To: ;tag=2627e587
From: ;tag=e637fa0f
Call-ID: ZTg3YjI0ZGNiYjVjZjZlMDRlNjU1Mzg1ZGJkMWQ0NmQ.
CSeq: 2 BYE
User-Agent: X-Lite release 1002tx stamp 29712
Reason: SIP;description="ACK not received"
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.100.1.220;branch=z9hG4bKe265.bd09554.0;received=24.94.9.135
Via: SIP/2.0/UDP

192.100.1.200:30612;received=192.100.1.200;branch=z9hG4bK-d87543-4677b2285a79346c-1--d87543-;rport=30612
Contact: 
To: ;tag=2627e587
From: ;tag=e637fa0f
Call-ID: ZTg3YjI0ZGNiYjVjZjZlMDRlNjU1Mzg1ZGJkMWQ0NmQ.
CSeq: 2 BYE
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0

---

Re: [OpenSIPS-Users] Detecting retransmissions

[Stefano Pisani]

You have to check for same callid, same from/to tag, and same cseq.
If you have the same values of a previus record, discard it.

s

Il 08/04/2011 10:21, Pete Kelly ha scritto:

Hi

I am performing some database logging actions on responses in a reply 
route, is it possible to detect if a response is a retransmission, so 
I don't perform the same logging action on it twice?


Pete


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Re: [OpenSIPS-Users] Need ideas to tamper with CSeq

[Stefano Pisani]

What are you trying to do exactly?

s

Il 31/03/2011 16:37, Cindy Leung ha scritto:

I guess I wasn't being clear enough in the call flow.  I assume the CSeq in the 
CANCEL has to be the same as the second INVITE.

1. Phone sends out INVITE #1, OpenSIPS responds with 401, Phone ACK'd.  I 
believe the transaction is over at this point.
2. Phone sends out INVITE #2 with auth, OpenSIPS accepts the INVITE and send 
back 180.  Phone now sends out a CANCEL, but the CSeq is not the same as INVITE 
#2.

As far as I can tell, everything else (ruri, call-id...) is the same except for 
CSeq.

Thanks.




On Mar 31, 2011, at 5:03 AM, Anca Vamanu wrote:


On 03/31/2011 03:21 AM, Cindy Leung wrote:

I know I'm doing something bad here.  However, we are having a problem with one 
of the SIP phones that we support.  When it sends out an INVITE and then 
CANCEL, the CANCEL is not being forwarded.  We are suspecting that it is caused 
by a wrong CSeq value.

INVITE #1 gets challenged.
INVITE #2 accepted.
CANCEL is sent, but CSeq is the same as the one in INVITE #1.


It is ok (RFC compliant) for the Cseq in CANCEL to be the same as the Cseq in 
INVITE:
RFC 3261 - section 9.1:

"The Request-URI, Call-ID, To, the numeric part of CSeq, and From header
   fields in the CANCEL request MUST be identical to those in the
   request being cancelled, including tags. "

Regards,

--
Anca Vamanu
OpenSIPS Developer




It is less than ideal for us to contact their support and we'd like to get it 
fixed asap.  I've tried subst(), remove_hf and append_hf to play with CSeq with 
no luck.

Any suggestions?  Thanks!


Cinthi

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Re: [OpenSIPS-Users] Dedicated Presence Service

[Stefano Pisani]

to do what?

s

Il 31/03/2011 11:07, Paris Stamatopoulos ha scritto:

Anyone who could give a helping hand here?

Regards,
Paris


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Re: [OpenSIPS-Users] Modify a URI

[Stefano Pisani]

You can do that in several ways.
A simple way could be:

$ru = "sip:12345#" + $rU + "@" + $rd

other ways are possible

Il 23/02/2011 19:31, Brian Artigas ha scritto:


Can anyone give me a code example how I would take a URI and inject a 
string between the sip: and the actual number. I found the 
rewriteuri() function but cannot find any code examples on how to use it.


For instance take the request URI:   
sip:5615551212@111.222.333.444


And change it to:  
sip:12345#5615551212@111.222.333.444


Thanks in advance,

Brian


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Re: [OpenSIPS-Users] OpenSIPS handling B2B features

[Stefano Pisani]

It's very simple setup a Conference server using OpenSIPS and Asterisk. 
So use asterisk.


Regards,
s

Il 27/01/2011 17:39, Anca Vamanu ha scritto:

Toyima,

I am sorry, I don't have experience in setting up conference systems, 
so I can not make a recommendation.


Regards,



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Re: [OpenSIPS-Users] force_rtp_proxy no longer supported

[Stefano Pisani]

My experience using the opensips 1.6.3 is that engage_rtp_proxy doesn't 
work, and force_rtp_proxy works fine, despite the documentation.


Il 10/02/2011 14:57, chris ha scritto:

Hi,

Thanks for that. The new docs mention engage_rtp_proxy but this does not
seem to be invoked or described in the module anywhere? Should this be
present and how would it then interact with offer and answer?

Thanks

Chris

-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Ovidiu Sas
Sent: 10 February 2011 13:11
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] force_rtp_proxy no longer supported

The docs for 1.6.4 are the actual trunk/devel and there the reference
to force_rtp_proxy was removed:
http://www.opensips.org/html/docs/modules/devel/nathelper.html

A new link needs to be created for 1.6.4 docs.


Regards,
Ovidiu Sas

On Thu, Feb 10, 2011 at 4:20 AM, chris  wrote:


Hi,



Can someone update the various docs to indicate that this function is

no

longer supported and to revert to the rtpproxy_offer/answer methods.

It is still refered to in the module source and examples with a small
reference to it being removed in nathelper.c spelt wrong so impossible

to

find!



Regards



Chris

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Re: [OpenSIPS-Users] How to test if a message is from myself

[Stefano Pisani]

Hi Dave
you could try

if ($si == $hdr(X-src-ip)){...}


Il 03/02/2011 12:59, Bogdan-Andrei Iancu ha scritto:

Hi Dave,

Unfortunately does not work with variables.

Regards,
Bogdan

Dave Singer wrote:

Wow I missed that one. Thanks.
Does that work for PVs so I can test other IPs like one from another
header, say "X-src-ip:192.168.0.5". Last I tried I couldn't get it to
work. Not sure if that was 1.6.2 or 1.6.3. I'm using 1.6.4 now. :)

Thanks Again
Dave

On Wed, Feb 2, 2011 at 4:37 AM, Bogdan-Andrei Iancu 
 wrote:

Hi Dave,

do :  if (src_ip==myself) {}

Regards,
Bogdan

Dave Singer wrote:

Is there any way to check if the source IP/port is one that opensips
is listening on or one ? something like if ("sip:$si:$sp" == myself) {
...bla; bla;}

Thanks
Dave

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2 - 4 February 2011, ITExpo, Miami,  USA
OpenSIPS solutions and "know-how"


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Re: [OpenSIPS-Users] call forwarding with replace from uri

[Stefano Pisani]

Hi Jesse,
in your script you are replacing from header twice.
Double check to your script and delete the second uac_replace_from.
This function can be used just once a phone call.

ciao
s

Il 02/02/2011 16:41, Jesse Cloutier ha scritto:
Thanks for the answer, I was not very clear in my first email though, 
sorry for that. Basically when caller A initiates a call, his $fu may 
be pulled from the db and replaced with a uac_replace_from. Then if 
the call progresses and is redirected to a new destination the call 
fails to authenticate because the new FROM URI is not in the DB. I 
have tried to restore his original FROM URI using uac_restore_from, 
but this doesnt seem to do it and if I store the original uri and call 
uac_replace_from a second time I get something like 
"sip:111@1.1.1.1sip:222@1.1.1.1"


Multumesc!

On Wed 02 Feb 2011 07:28:28 AM EST, Bogdan-Andrei Iancu wrote:

Hi Jesse,

Lost me a bit between those pieces of script.

Anyhow, as far I understood from your problem, you want to change the 
FROM URI in order to use the new value later in the script (in the 
auth part). Well, this does not work - in opensips, the changes you 
do over the message are not visible until the message is sent out. So 
you cannot use your own changes later in script.


To avoid this issue, you can use a variable to store the "correct" 
value of the FROM URI ($fu directly or the value from DB) - later in 
script, use for auth (or other purposes) this variable in order to 
deal with the FROM URI value (the right one).


Regards,
Bogdan

Jesse Cloutier wrote:

Hi list,

I having trouble with my script when trying to call forward by 
reseting the $ru and doing a route(1)


My problem seems to be coming from the fact that I am changing my 
$fu with uac_replace_from. When I xlog the $fu right before the 
route() It shows the correct value (the original $fu before it was 
changed by uac_replace_from). But on the request to the forwarded 
number it tries to authenticate the user using the new value (the 
value that uac_replace_from put in)


If I don't replace the $fu everything works fine.

Thanks A lot for any help!!

here is the relavant parts of my script:

Replacing the uri in the original request:

if (is_avp_set("$avp(s:uri)")) {
if (is_avp_set("$avp(s:fromname)")) {
xlog("L_INFO","Fromname set to $avp(s:fromname) and URI set to 
$avp(s:uri)");

uac_replace_from("$avp(s:fromname)","$avp(s:uri)");
} else {
uac_replace_from("","$avp(s:uri)");
xlog("L_INFO","Only Fromname Set");
}
}


The fowrwarding:

if(avp_db_load("$ru","$avp(s:unavailcallfwd)")) {
#xlog("call forward is set to: $avp(s:unavailcallfwd)");
avp_pushto("$ru","$avp(s:unavailcallfwd)");
xlog("call forward is set to: $ru from $fu");

route(1);

exit;
}


And the proxy authorize


xlog("Checking if we should attempt authentication on $fu");
if (!(method=="REGISTER"))
{
#Do not authenticate calls from the gateways
xlog("Checking if its from a gateway");
if(!is_from_gw()) # This check is from the drouting module
{
xlog("Checking if it is an IP Authed IP");
if(!check_source_address("0", "$avp(i:9)")) #This check looks in the 
address table

{
xlog("Checking if it is a subscriber");

xlog("from is $fu");

if (!proxy_authorize("", "subscriber")) {
proxy_challenge("", "0");
xlog("Sent proxy challange to $fu");
exit;
}
if (!db_check_from()) {
sl_send_reply("403","Forbidden auth ID");
exit;
}

}
}
}


--
Jesse Cloutier
Network Administrator
Cronomagic Canada
5143411579 x210
je...@cronomagic.com
 



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Re: [OpenSIPS-Users] Lookup contact from user part of RURI

[Stefano Pisani]

Hi,
you could set OpenSIPS to not use domain part of uri, so your issue is 
solved.


stefano

Il 02/02/2011 15:30, Nauman Sulaiman ha scritto:

Hi, using opensips 1.6.2. I am trying to use the user part of incoming RURI
to look up a contact, reason being is full RURI is incorrect, this is due to 
bogus proxy upstream so need a workaround.

lookup("location") seems to be only if you use AOR.

For exmaple i need to reroute incoming ACK to real address of UA
So i would like to lookup 1234 user part of RURI below and rewrite the
RURI with the correct ip. 12.34.56.78 is Opensips IP rather than end UA ip.

1...@domain.com is stored in db. How do i lookup contact just with user part 
and rewrite the RURI.

ie ACK sip:1234@12.34.56.78;rinstance=A89B5393

Need something for below
  if(method=="ACK")
  {
   xlog("ACK received  \n");
   if( $rd == "12.34.56.78")  // check if opensips ip
   {
lookup(user);  // ???   // need to lookup with user or rinstance
// rewrite RURI with correct address
  }
  }




Hope its clear, thanks




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[OpenSIPS-Users] RTP Proxy with 2 opensips

[Stefano Pisani]

Hello,
I have this scenario:

UAC --> Opensips1 --> Opensips2 + RTPProxy --> Internet

Opensips1 is on LAN, Opensips2 is dual homed (LAN + Public IP) and on 
the same machine there is a RTP Proxy in Bridge Mode.


RTP Proxy doesn't work. It uses, as callee IP, the opensips1 IP instead 
of UAC IP.

What's wrong in this architecture?

regards,
s

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Re: [OpenSIPS-Users] Opensips B2B + RTP proxy in bridged mode

[Stefano Pisani]

Could you post your cfg?
regards,
s

Il 13/12/2010 11:48, beci345 ha scritto:

Hello to all,
i'm using the Opensips as proxy in multihomed mode (one public IPaddr and
one internal), with relaying RTP traffic through RTP proxy in bridged mode:

UA --->OpenSipsIP1-OpenSipsIP2 >UA (signalling)
UA>RTPproxyIP1-RTPproxyIP2--->UA (RTP), RTP proxy in bridged mode.

Everything works OK, SDP information is modified correctly in subsequent
INVITE messages (using the force_rtp_proxy("ei")):

UA (SIP INVITE, SDP Connection information IPofUA) --->  Opensips (SIP
INVITE, SDP Connection information IP2)>  UA

However, i have problem if i would like to extend the configuration with
topology hiding functionality - byusing the B2B modules.
By calling the scenario with b2b_init_request("top hiding"), Opensips fires
out the INVITE with wrong SDP Connection information (IP of calling UA)
towards the called party.
I was not able to set the SDP in correct way...
I've tried tricks from textops module, not successful. Is there anybody who
could advice how to fix it, or is this configuration possible? Im' using
OpenSIPS 1.6.2.

Thanks and regards,
Bela



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[OpenSIPS-Users] opensips cross border configuration

[Stefano Pisani]

I guys,
I'm try to configure opensips 1.6.3 + rtpproxy on a dual homed server 
with one interface on private network and one on public network.

I want to make possibile the calls from inside to outside and viceversa.
There is a smart way to do that?
I'm going to write a cfg to manually change the IPs in the outgoing 
messages.


Thanks
s

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Re: [OpenSIPS-Users] registration to other SIP proxies?

[Stefano Pisani]

I developed my perl script using Net::SIP to do that (and also to play 
audio message without asterisk).


s


Il 02/12/2010 15:24, Erik Dekkers ha scritto:

Bogdan,

Would it be possible to run sipak from the opensips.cfg script? I'm also 
looking for something like this.

Kind regards,

Erik

-Oorspronkelijk bericht-
Van: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org] 
Namens Bogdan-Andrei Iancu
Verzonden: donderdag 2 december 2010 13:12
Aan: OpenSIPS users mailling list
Onderwerp: Re: [OpenSIPS-Users] registration to other SIP proxies?

Hi Paul,

OpenSIPS it self cannot register with other servers (the UAC capabilities are 
limited).

But you can easily do that with sipsak - you can use this utility to register 
opensips's contact to another registrar server.

Regards,
Bogdan

Paul Wise wrote:

Hi all,

Is it possible for OpenSIPS to register to other SIP proxies? Or if
not has anyone found a way to do this indirectly?


--
--

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15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro


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[OpenSIPS-Users] Opensisp behind a Firewall and NAT

[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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Re: [OpenSIPS-Users] rewritehost() and AVP

[Stefano Pisani]

use $du = $avp(...)

regards,
s

Il 18/11/2010 10:38, Anton Zagorskiy ha scritto:

Thanks, this is work.

Can you explain why rewritehost() doesn't accept AVP? Where AVP doesn't work
too?






WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru




-Original Message-
From: users-boun...@lists.opensips.org [mailto:users-
boun...@lists.opensips.org] On Behalf Of Ovidiu Sas
Sent: Wednesday, November 17, 2010 7:35 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] rewritehost() and AVP

Deal with the host and port via PVs:
http://www.opensips.org/Resources/DocsCoreVar16#toc59
http://www.opensips.org/Resources/DocsCoreVar16#toc64


Regards,
Ovidiu Sas

On Wed, Nov 17, 2010 at 10:58 AM, Anton Zagorskiy
  wrote:

Hello.
How to pass a value to the rewritehost() function using AVP?

1. Code
$avp(i:50) = "192.168.0.01";
rewritehostport("$avp(i:50)");
route(1)

raises errors:
ERROR:core:parse_uri: bad port in uri (error at char ) in state 8)

parsed:

(21) /  (22)
ERROR:core:parse_sip_msg_uri: bad uri

2. Code
$avp(i:50) = "192.168.0.01";
rewritehostport($avp(i:50)); # Without qoutes
route(1)

raises errors during stratup:
CRITICAL:core:yyerror: parse error in config file, line 747, column

28-38:

syntax error
CRITICAL:core:yyerror: parse error in config file, line 747, column

38-39:

bad argument, string expected



WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru





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[OpenSIPS-Users] Two OpenSIPS one database

[Stefano Pisani]

Hello,
I need to divide the load on two opensips (using DNS round robin)
I would use only one database (mysql).
my worries are about the writing on database.
Both opensips can use the db to record the user locations without problems?

thanks
stefano

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Re: [OpenSIPS-Users] Two OpenSIPS one database

[Stefano Pisani]

Hello,
my worries are about writing on database.
Both opensips can use db to record locations?

s

Il 26/10/2010 22:45, thrillerbee ha scritto:

Stefano,

Yes, when you specify the db_url, it doesn't have to be local.  So 
multiple OpenSIPS instances can all pull from the same db if you want.


That being said, how are you wanting to round robin?


On Tue, Oct 26, 2010 at 3:42 PM, Stefano Pisani 
mailto:stefano.pis...@omnianet.it>> wrote:


Hello,
is it possible to share the database beteween two opensips?
I would to use round robin to divide the load on more servers.

Thanks

s

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[OpenSIPS-Users] Two OpenSIPS one database

[Stefano Pisani]

Hello,
is it possible to share the database beteween two opensips?
I would to use round robin to divide the load on more servers.

Thanks

s

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Re: [OpenSIPS-Users] How to change Contact header

[Stefano Pisani]

  Why don't you explain the problem you want to solve?
Maybe that to modify contact in this way is not the right solution.

s


Il 05/10/2010 18:34, David Santiago ha scritto:
> Hi all,
>
> I need to modify the host part of a contact header. I'm trying something like:
>
> if ( subst('/^Contact: /ig') ) {
>  xlog("contact modified!");
>  };
>
> but the resulting Contact header is wrong and cannot be processed.
>
> Having a look at the header with wireshark shows that the "Contact
> Binding" entry is missing the ending ">", but the "Contact", "URI" or
> "SIP contact address" have the">" at the end  :L
>
> May be this is not the right way to modify a Contact header...
>
>
> Thanks in advance,
> David
>
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Re: [OpenSIPS-Users] $fU read-only, calling number modification problem

[Stefano Pisani]

  Use replace_from :-)

ciao
s

Il 10/10/2010 19:19, Maciej Bylica ha scritto:
> Hello
>
> I have a question regarding $fU pseudo variable.
> As mentioned at http://www.opensips.org/Resources/DocsCoreVar16 and on
> the basis of opensips outputs:
> ERROR:dialplan:dp_trans_fixup: the output PV is read-only!!
> it clearly means that $fU is read-only.
>
> Unfortunately it is quite big problem for me, because what im
> struggling with is to achieve proper calling number presentation.
> In my scenario all endpoints located in subscriber table do have full
> username with country code, so there are for instance:
> - 48111223344 (48 country code)
> - 49222334455 (49 country code)
> - 44333445566 (44 country code)
> ...
>
> If there is a national call inside the 48 country code the calling
> number should be changed by striping first two digits (48) -
> 48999887766--->999887766
> In case of international call, i should add two digits (00) -
> 49222334455--->0049222334455.
>
> I am using diaplan module in this case and following entry gives me
> the error I mentioned.
> dp_translate("2", "$fU/$fU");
>
> If there are any workaround.
> Any help would be highly appreaciated.
>
> Thanks,
> Maciej
>
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