Re: [OpenSIPS-Users] Opensips 2.4.x and CP 8

[qasimak...@gmail.com]

Hi,

Here is your error:
ERROR:mi_json:mi_json_answer_to_connection: unexpected method [POST]

If you look up in documentation it says on first line:
JSON support via HTTP GET for Management Interface

Regards
Qasim

On Wed, 6 Nov 2019 at 6:58 AM, Jeff Wilkie  wrote:

> Attempting to get CP8 and 2.4.x talking to each other.  I have modules in
> the config enabled
>
> loadmodule "httpd.so"
> modparam("httpd", "ip", "127.0.0.1")
> modparam("httpd", "port", )
> loadmodule "mi_json.so"
> modparam("mi_json", "mi_json_root", "json")
>
> When attempting to execute any MI_json command from CP we're seeing this
> from the opensips logs.
>
> ERROR:mi_json:mi_json_answer_to_connection: unexpected method [POST]
>
> Where is this error being generated?  How do we correct this?
>
> Thanks
> Jeff
>
>
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Re: [OpenSIPS-Users] Recommended Radius client/server for AAA on 2.4.x under Debian 9

[qasimak...@gmail.com]

You can use latest version of freeradius it has both client and server.

Regards,
Qasim

On Wed, 6 Nov 2019 at 8:27 AM, Jeff Wilkie  wrote:

> Attempting to find current docs since radiusclient-ng is referenced in
> several old docs but is no longer available.  Currently, what is the
> recommending radius packages to use for accounting purposes on opensips for
> CDRs and ACC?
>
> Thanks
> Jeff
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Re: [OpenSIPS-Users] Website Down.

[qasimak...@gmail.com]

I think there was a short flux for around 10-15 minutes.

Regards,
Qasim

On Tue, Dec 26, 2017 at 8:21 PM, Impala Tux  wrote:

> Hi
>
> In Brazil it's right, no problems
>
> Em 26 de dez de 2017 08:05, "qasimak...@gmail.com" 
> escreveu:
>
>> Hi,
>>
>> I am getting 504 Gateway Timeout on opensips.org. Is anyone else facing
>> the same issue?
>>
>> Regards,
>> Qasim
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
> ___
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>
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[OpenSIPS-Users] Website Down.

[qasimak...@gmail.com]

Hi,

I am getting 504 Gateway Timeout on opensips.org. Is anyone else facing the
same issue?

Regards,
Qasim
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Re: [OpenSIPS-Users] Opensips 2.2.3 crash on async.

[qasimak...@gmail.com]

Thanks bogdan, ill see how I can work around it keeping launch() in mind.

-Qasim

On Apr 13, 2017 4:28 PM, "Bogdan-Andrei Iancu"  wrote:

> The async(), as based on transactions, works only for requests, not for
> replies. This is limitation that will be changed in the next releases (but
> not in 2.3).
>
> What you could do in 2.3 is to use launch() statement to run your async
> stuff.
> launch( blocking_function(...) );
> The launch will start your function in background and continue your script.
>
> Bogdan-Andrei Iancu
>   OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>
> On 04/13/2017 02:00 PM, qasimak...@gmail.com wrote:
>
> Hi Bogdan,
>
> Yes i have to do some accounting calls based on reply recieved. So i guess
> that means that async is not available in reply route? Is it still planned
> for future release or maybe in 2.3?
>
> Regards,
> Qasim
>
> On Thu, Apr 13, 2017 at 3:15 PM, Bogdan-Andrei Iancu 
> wrote:
>
>> Hi Qasim,
>>
>> This looks very very similar to an old report:
>> http://lists.opensips.org/pipermail/users/2017-January/036231.html
>>
>> And this was fixed in 2.2:
>> https://github.com/OpenSIPS/opensips/commit/4023a5d797960d24
>> 5b52eb73dc9fc26b8cdf2914
>>
>> I guess you try to do some async stuff in the reply route, right ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>   OpenSIPS Founder and Developer
>>   http://www.opensips-solutions.com
>>
>> OpenSIPS Summit May 2017 Amsterdam
>>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>>
>> On 04/13/2017 12:56 PM, qasimak...@gmail.com wrote:
>>
>> Hi Bogdan,
>> PFA required logs. Also fund below some more info on the setup i am
>> using. *OS:* Distributor ID: SUSE LINUX Description:openSUSE Leap
>> 42.1 (x86_64) Release:42.1
>> *Opensips:*
>> version: opensips 2.2.3 (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. git revision: 815a81abb main.c compiled on
>> 13:56:49 Apr 12 2017 with gcc 4.8
>> Please also note that i have tried compiling it from git but the problem
>> persists.
>> Regards,
>> Qasim
>> On Thu, Apr 13, 2017 at 2:26 PM, Bogdan-Andrei Iancu > > wrote:
>>>
>>> Hi Qasim, Thank you for your report. Could you please run a "bt full" in
>>> gdb a post the output ? Best regards,
>>>
>>> Bogdan-Andrei Iancu
>>>   OpenSIPS Founder and Developer
>>>   http://www.opensips-solutions.com
>>>
>>> OpenSIPS Summit May 2017 Amsterdam
>>>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>>>
>>> On 04/13/2017 12:09 PM, qasimak...@gmail.com wrote:
>>>
>>> Hi,
>>> I have upgraded my script from 1.11 to 2.2.3 which works fine until i
>>> put async function on a single rest_get query. When the async line is
>>> executed i get following errors:
>>>>
>>>> 2017-04-13T14:03:45.320300+05:00 sip01 kernel: [31710929.448312]
>>>> opensips[11475]: segfault at 10 ip 00427260 sp 7ffe8c29f6e8
>>>> error 6 in opensips[40+228000] 2017-04-13T14:03:45.908172+05:00
>>>> sip01 ./sbin/opensips[10993]: WARNING:core:utimer_ticker: utimer task
>>>>  already scheduled for 19470 ms (now 19570 ms), it may overlap..
>>>> 2017-04-13T14:03:46.008338+05:00 sip01 ./sbin/opensips[10993]:
>>>> WARNING:core:utimer_ticker: utimer task  already scheduled for
>>>> 19470 ms (now 19670 ms), it may overlap..
>>>
>>> I also have a core dump available, after opening in gdb i get the
>>> following info:
>>>
>>>> [Thread debugging using libthread_db enabled] Using host libthread_db
>>>> library "/lib64/libthread_db.so.1". Core was generated by `./sbin/opensips
>>>> -P /var/run/opensips.pid'. Program terminated with signal SIGSEGV,
>>>> Segmentation fault. #0  0x00427260 in context_put_int
>>>> (type=CONTEXT_GLOBAL, ctx=0x0, pos=4, data=1) at context.c:173
>>>> 173 ((int *)ctx)[pos] = data;
>>>>
>>> Please note i will keep core dumps if you guys need some more info on
>>> it.
>>> Regards,
>>> Qasim
>>>
>>> ___
>>> Users mailing 
>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
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Re: [OpenSIPS-Users] Opensips 2.2.3 crash on async.

[qasimak...@gmail.com]

Hi Bogdan,

Yes i have to do some accounting calls based on reply recieved. So i guess
that means that async is not available in reply route? Is it still planned
for future release or maybe in 2.3?

Regards,
Qasim

On Thu, Apr 13, 2017 at 3:15 PM, Bogdan-Andrei Iancu 
wrote:

> Hi Qasim,
>
> This looks very very similar to an old report:
> http://lists.opensips.org/pipermail/users/2017-January/036231.html
>
> And this was fixed in 2.2:
> https://github.com/OpenSIPS/opensips/commit/
> 4023a5d797960d245b52eb73dc9fc26b8cdf2914
>
> I guess you try to do some async stuff in the reply route, right ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>   OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>
> On 04/13/2017 12:56 PM, qasimak...@gmail.com wrote:
>
> Hi Bogdan,
>
> PFA required logs. Also fund below some more info on the setup i am using.
>
> *OS:*
> Distributor ID: SUSE LINUX
> Description:openSUSE Leap 42.1 (x86_64)
> Release:42.1
>
> *Opensips:*
> version: opensips 2.2.3 (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.
> git revision: 815a81abb
> main.c compiled on 13:56:49 Apr 12 2017 with gcc 4.8
>
> Please also note that i have tried compiling it from git but the problem
> persists.
>
> Regards,
> Qasim
>
>
>
> On Thu, Apr 13, 2017 at 2:26 PM, Bogdan-Andrei Iancu <
> bog...@opensips.org> wrote:
>
>> Hi Qasim,
>>
>> Thank you for your report. Could you please run a "bt full" in gdb a post
>> the output ?
>>
>> Best regards,
>>
>> Bogdan-Andrei Iancu
>>   OpenSIPS Founder and Developer
>>   http://www.opensips-solutions.com
>>
>> OpenSIPS Summit May 2017 Amsterdam
>>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>>
>> On 04/13/2017 12:09 PM, qasimak...@gmail.com wrote:
>>
>> Hi,
>> I have upgraded my script from 1.11 to 2.2.3 which works fine until i put
>> async function on a single rest_get query. When the async line is executed
>> i get following errors:
>>>
>>> 2017-04-13T14:03:45.320300+05:00 sip01 kernel: [31710929.448312]
>>> opensips[11475]: segfault at 10 ip 00427260 sp 7ffe8c29f6e8
>>> error 6 in opensips[40+228000] 2017-04-13T14:03:45.908172+05:00
>>> sip01 ./sbin/opensips[10993]: WARNING:core:utimer_ticker: utimer task
>>>  already scheduled for 19470 ms (now 19570 ms), it may overlap..
>>> 2017-04-13T14:03:46.008338+05:00 sip01 ./sbin/opensips[10993]:
>>> WARNING:core:utimer_ticker: utimer task  already scheduled for
>>> 19470 ms (now 19670 ms), it may overlap..
>>
>> I also have a core dump available, after opening in gdb i get the
>> following info:
>>
>>> [Thread debugging using libthread_db enabled] Using host libthread_db
>>> library "/lib64/libthread_db.so.1". Core was generated by `./sbin/opensips
>>> -P /var/run/opensips.pid'. Program terminated with signal SIGSEGV,
>>> Segmentation fault. #0  0x00427260 in context_put_int
>>> (type=CONTEXT_GLOBAL, ctx=0x0, pos=4, data=1) at context.c:173
>>> 173 ((int *)ctx)[pos] = data;
>>>
>> Please note i will keep core dumps if you guys need some more info on it.
>> Regards,
>> Qasim
>>
>> ___
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
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Re: [OpenSIPS-Users] Opensips 2.2.3 crash on async.

[qasimak...@gmail.com]

Hi Bogdan,

PFA required logs. Also fund below some more info on the setup i am using.

*OS:*
Distributor ID: SUSE LINUX
Description:openSUSE Leap 42.1 (x86_64)
Release:42.1

*Opensips:*
version: opensips 2.2.3 (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.
git revision: 815a81abb
main.c compiled on 13:56:49 Apr 12 2017 with gcc 4.8

Please also note that i have tried compiling it from git but the problem
persists.

Regards,
Qasim



On Thu, Apr 13, 2017 at 2:26 PM, Bogdan-Andrei Iancu 
wrote:

> Hi Qasim,
>
> Thank you for your report. Could you please run a "bt full" in gdb a post
> the output ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
>   OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
>
> OpenSIPS Summit May 2017 Amsterdam
>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>
> On 04/13/2017 12:09 PM, qasimak...@gmail.com wrote:
>
> Hi,
>
> I have upgraded my script from 1.11 to 2.2.3 which works fine until i put
> async function on a single rest_get query. When the async line is executed
> i get following errors:
>
> 2017-04-13T14:03:45.320300+05:00 sip01 kernel: [31710929.448312]
>> opensips[11475]: segfault at 10 ip 00427260 sp 7ffe8c29f6e8
>> error 6 in opensips[40+228000]
>> 2017-04-13T14:03:45.908172+05:00 sip01 ./sbin/opensips[10993]:
>> WARNING:core:utimer_ticker: utimer task  already scheduled for
>> 19470 ms (now 19570 ms), it may overlap..
>> 2017-04-13T14:03:46.008338+05:00 sip01 ./sbin/opensips[10993]:
>> WARNING:core:utimer_ticker: utimer task  already scheduled for
>> 19470 ms (now 19670 ms), it may overlap..
>>
>
> I also have a core dump available, after opening in gdb i get the
> following info:
>
> [Thread debugging using libthread_db enabled]
>> Using host libthread_db library "/lib64/libthread_db.so.1".
>> Core was generated by `./sbin/opensips -P /var/run/opensips.pid'.
>> Program terminated with signal SIGSEGV, Segmentation fault.
>> #0  0x00427260 in context_put_int (type=CONTEXT_GLOBAL, ctx=0x0,
>> pos=4, data=1) at context.c:173
>> 173 ((int *)ctx)[pos] = data;
>>
>
> Please note i will keep core dumps if you guys need some more info on it.
>
> Regards,
> Qasim
>
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
#0  0x00427260 in context_put_int (type=CONTEXT_GLOBAL, ctx=0x0, pos=4, 
data=1) at context.c:173
No locals.
#1  0x7f808e4d27ec in dlg_onreply (t=0x7f80916b9498, type=, 
param=) at dlg_handlers.c:490
rpl = 0x7f80d13dbb08
req = 0x7f80916baef0
dlg = 0x7f80916b8598
new_state = 1
old_state = -1855224504
unref = 487
event = 4
mangled_from = {s = 0x0, len = 0}
mangled_to = {s = 0x0, len = 0}
req_out_buff = 
#2  0x7f8090d5d4c9 in run_trans_callbacks (type=type@entry=64, 
trans=trans@entry=0x7f80916b9498, req=req@entry=0x7f80916baef0, rpl=, code=)
at t_hooks.c:209
params = {req = 0x7f80916baef0, rpl = 0x7f80d13dbb08, code = 487, param 
= 0x7f80916b9120, extra1 = 0x7ffc399fe790, extra2 = 0x7f80916b95b0}
cbp = 0x7f80916b9110
backup = 0x847678 
trans_backup = 0x
__FUNCTION__ = "run_trans_callbacks"
#3  0x7f8090d5d822 in run_trans_callbacks_locked (type=type@entry=64, 
trans=trans@entry=0x7f80916b9498, req=0x7f80916baef0, 
rpl=rpl@entry=0x7f80d13dbb08, code=code@entry=487)
at t_hooks.c:262
No locals.
#4  0x7f8090d1523c in relay_reply (t=, p_msg=, branch=, msg_status=, 
cancel_bitmap=) at t_reply.c:1244
relay = 
save_clone = 
buf = 0x7f80d14014e0 "SIP/2.0 487 Request Terminated\r\nVia: 
SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK029e.5080a3a5.0\r\nRecord-Route: 
,,
bm = {to_tag_val = {s = 0x7f80d13af6c8 "\b", len = 112}}
totag_retr = 0
uas_rb = 
cb_s = {
  s = 0x7f80d14014e0 "SIP/2.0 487 Request Terminated\r\nVia: 
SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK029e.5080a3a5.0\r\nRecord-Route: 
,, "", len = -784575864}
__FUNCTION__ = "relay_reply"
#5  0x7f8090d1b6a6 in reply_received (p_msg=0x7f80d13dbb08) at 
t_reply.c:1505
msg_status = 487
last_uac_status = 180
branch = 0
reply_status = 
timer = 140190162580653
cancel_bitmap = 0
uac = 0x7f80916b9670
t = 0x7f80916b9498

[OpenSIPS-Users] Opensips 2.2.3 crash on async.

[qasimak...@gmail.com]

Hi,

I have upgraded my script from 1.11 to 2.2.3 which works fine until i put
async function on a single rest_get query. When the async line is executed
i get following errors:

2017-04-13T14:03:45.320300+05:00 sip01 kernel: [31710929.448312]
> opensips[11475]: segfault at 10 ip 00427260 sp 7ffe8c29f6e8
> error 6 in opensips[40+228000]
> 2017-04-13T14:03:45.908172+05:00 sip01 ./sbin/opensips[10993]:
> WARNING:core:utimer_ticker: utimer task  already scheduled for
> 19470 ms (now 19570 ms), it may overlap..
> 2017-04-13T14:03:46.008338+05:00 sip01 ./sbin/opensips[10993]:
> WARNING:core:utimer_ticker: utimer task  already scheduled for
> 19470 ms (now 19670 ms), it may overlap..
>

I also have a core dump available, after opening in gdb i get the following
info:

[Thread debugging using libthread_db enabled]
> Using host libthread_db library "/lib64/libthread_db.so.1".
> Core was generated by `./sbin/opensips -P /var/run/opensips.pid'.
> Program terminated with signal SIGSEGV, Segmentation fault.
> #0  0x00427260 in context_put_int (type=CONTEXT_GLOBAL, ctx=0x0,
> pos=4, data=1) at context.c:173
> 173 ((int *)ctx)[pos] = data;
>

Please note i will keep core dumps if you guys need some more info on it.

Regards,
Qasim
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Re: [OpenSIPS-Users] opensips performance

[qasimak...@gmail.com]

I think there is no such direct command that will calculate CPS for you,
However there are certainly ways you can calculate this CPS (One example
given by Aqs) but since you are crunching maximum CPS from opensips i would
recommend that you dont use opensips for its calculation, reason being that
you will have to invoke extra modules and functions to achieve the task
hence increasing complexity to your configuration.

Having said that i would recommend that you simply put a log for every
incoming invite and offload the TPS calculation to some external script
e.g. *#grep -c "Apr  7 14:" *this will give you total logs in one hour and
you divide it by 3600 gives you average CPS per hour. You can set the
resolution to your requirement.

I know this seems a bit dumb solution but then again i guess 'it aint
stupid if it works' :), and you can always come up with some fancy way of
doing the same thing.

Regards,
Qasim

On Thu, Apr 6, 2017 at 2:06 AM, deizeppe  wrote:

> Hi, is there any command that shows the value of CPS?
>
> Thanks in advance
>
>
>
> --
> View this message in context: http://opensips-open-sip-
> server.1449251.n2.nabble.com/opensips-performance-tp7605389p7606871.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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>
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Re: [OpenSIPS-Users] Opensips Late reply.

[qasimak...@gmail.com]

@Daniel, I know i dont need 1 child per request but i did it because it
might keep delays short as it has more threads to work with. I have limited
input call origination's from mi_datagram's UDP interface. Also maximum
number of calls are capped using dialog profiling.


Also i am using opensips 1.11 and i have got a few more logs to share. This
time i deliberately put maximum load on opensips and let it crash, Here are
is some info collected from core dumps:

*Log File:*
2017-04-05T17:45:04.986687+05:00 sip01 /usr/local/sbin/opensips[29092]:
Total number of dropped replies = 0
2017-04-05T17:45:04.986764+05:00 sip01 /usr/local/sbin/opensips[29344]:
Total number of dropped replies = 0
2017-04-05T17:45:04.987717+05:00 sip01 /usr/local/sbin/opensips[29344]:
Real SHMEM used size is 126444552
2017-04-05T17:45:04.988274+05:00 sip01 /usr/local/sbin/opensips[29344]:
Free SHMEM available is 947297272
2017-04-05T17:45:04.988425+05:00 sip01 /usr/local/sbin/opensips[29344]: The
total number of SHMEM fragments is 99819

2017-04-05T17:45:35.519928+05:00 sip01 /usr/local/sbin/opensips[28681]:
INFO:core:handle_sigs: child process 28887 exited by a signal 6
2017-04-05T17:45:35.730406+05:00 sip01 /usr/local/sbin/opensips[28681]:
INFO:core:handle_sigs: core was generated
2017-04-05T17:45:35.730879+05:00 sip01 /usr/local/sbin/opensips[28681]:
INFO:core:handle_sigs: terminating due to SIGCHLD

*gdb bt full:*
#5  0x7f3a81d17f7e in mi_uac_dlg_hdl (t=,
type=, ps=0x7ffe99c7eeb0) at mi.c:378
mi_hdl = 0x7f3a44b04b18
rpl_tree = 0x7f3a825f9d60
text = {s = 0x3 , len =
0}
__FUNCTION__ = "mi_uac_dlg_hdl"


-Qasim


On Wed, Apr 5, 2017 at 5:53 PM, Daniel Zanutti 
wrote:

> Hi Qasim
>
> How did you limit CPS? Because i have a similar scenario (150cps) but i
> set children to 20 or 24, never 200. You don't need 1 children per request.
>
> On Wed, Apr 5, 2017 at 9:44 AM, qasimak...@gmail.com  > wrote:
>
>> Hi,
>>
>> I have this scenario where i originate calls from mi_datagram and the
>> calls are cancelled as soon as it starts ringing. The problem i am facing
>> is that are running for a few minutes the response from opensips becomes
>> slow i.e. it send packets back to far end after a few seconds. Keeping it
>> running for a few hours and it crashes. I have currently limited the calls
>> per second to 200 and max call session to 600, but still there are calls
>> where opensips responds slow to SIP packets.
>>
>> Here are a few configurations that i am using:
>>
>> fork=yes
>> children=200
>>
>> loadmodule "mi_datagram.so"
>> modparam("mi_datagram", "socket_name", "/tmp/opensips.sock")
>> modparam("mi_datagram", "socket_name", "udp:localhost:2000")
>> modparam("mi_datagram", "children_count", 200)
>>
>>
>> Do you have any pointers where i should start looking at? I have also
>> generated core dump files let me know if you need some more info on this.
>>
>> Regards,
>> Qasim
>>
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[OpenSIPS-Users] Opensips Late reply.

[qasimak...@gmail.com]

Hi,

I have this scenario where i originate calls from mi_datagram and the calls
are cancelled as soon as it starts ringing. The problem i am facing is that
are running for a few minutes the response from opensips becomes slow i.e.
it send packets back to far end after a few seconds. Keeping it running for
a few hours and it crashes. I have currently limited the calls per second
to 200 and max call session to 600, but still there are calls where
opensips responds slow to SIP packets.

Here are a few configurations that i am using:

fork=yes
children=200

loadmodule "mi_datagram.so"
modparam("mi_datagram", "socket_name", "/tmp/opensips.sock")
modparam("mi_datagram", "socket_name", "udp:localhost:2000")
modparam("mi_datagram", "children_count", 200)


Do you have any pointers where i should start looking at? I have also
generated core dump files let me know if you need some more info on this.

Regards,
Qasim
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Re: [OpenSIPS-Users] ACC Db Duration

[qasimak...@gmail.com]

Thanks Daniel :)... You saved the day!

Regards,
Qasim

On Tue, Mar 28, 2017 at 6:36 PM, Daniel Zanutti 
wrote:

> Hi
>
> Did you check "cdr_flag" on Acc module?
>
> Otherwise it will generate 2 registers, 1 for INVITE and 1 for BYE.
>
> Regards
>
> On Tue, Mar 28, 2017 at 10:29 AM, qasimak...@gmail.com <
> qasimak...@gmail.com> wrote:
>
>> Hi,
>>
>> Sorry for the spam last email i miss-clicked on send amidst writing the
>> email.
>>
>> Anyways the problem i am facing is that my ACC module is configured with
>> MySQL DB backend and the CDR's are being written. However the problem i am
>> facing is that it is not logging duration into DB or syslog. Here are debug
>> logs where query is being prepared and inserted
>>
>>
>>- db_mysql_do_prepared_query: new query=|insert into acc(
>>
>> *method,from_tag,to_tag,callid,sip_code,sip_reason,time,caller_id,callee_id,serverid,info,billResponse,balance*
>>) values (?,?,?,?,?,?,?,?,?,?,?,?,?)|
>>- DBG:db_mysql:re_init_statement:  query  is >
>> *method,from_tag,to_tag,callid,sip_code,sip_reason,time,caller_id,callee_id,serverid,info,billResponse,balance*
>>) values(?,?,?,?,?,?,?,?,?,?,?,?,?)>, ptr=(nil)
>>
>> The problem is that i dont see duration in this query. Am i missing some
>> flag or something that needs to be set for duration to be logged in DB?
>>
>> *P.S. I am using opensips 1.11*
>>
>> Regards,
>>
>> Qasim
>>
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>>
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[OpenSIPS-Users] ACC Db Duration

[qasimak...@gmail.com]

Hi,

Sorry for the spam last email i miss-clicked on send amidst writing the
email.

Anyways the problem i am facing is that my ACC module is configured with
MySQL DB backend and the CDR's are being written. However the problem i am
facing is that it is not logging duration into DB or syslog. Here are debug
logs where query is being prepared and inserted


   - db_mysql_do_prepared_query: new query=|insert into acc(
   
*method,from_tag,to_tag,callid,sip_code,sip_reason,time,caller_id,callee_id,serverid,info,billResponse,balance*
   ) values (?,?,?,?,?,?,?,?,?,?,?,?,?)|
   - DBG:db_mysql:re_init_statement:  query  is , ptr=(nil)

The problem is that i dont see duration in this query. Am i missing some
flag or something that needs to be set for duration to be logged in DB?

*P.S. I am using opensips 1.11*

Regards,

Qasim
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[OpenSIPS-Users] ACC db duration

[qasimak...@gmail.com]

Hi,

I have enabled acc module in my opensips installation with db, My CDR's are
being written in MySQL backend but for every call the duration remains 0, I
have checked but according to documentation duration is automatically
logged in ACC module. PLease note following debug filtered where query is
made and executed
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Re: [OpenSIPS-Users] Opensips switch from SQL to NoSQL

[qasimak...@gmail.com]

Dear Feroz,

>From past experience DO NOT ATTEMPT this :). You will surely run into many
bottle necks from switching to NoSQL. NoSQL is good for caching during call
processing not for storing persistent data.

Regards,
Qasim

On Thu, Jul 14, 2016 at 11:38 PM, Jim DeVito  wrote:

> I addition to what Eric said I suspect you are failing to load
> cachedb_mongodb as stated "The following modules must be loaded before this
> module: At least one NoSQL cachedb_* module."
>
> Thanks!!
>
> ---
> Jim DeVito
> Mobile 440.941.3860
>
> On 2016-07-14 10:31, feroze waris wrote:
>
>> Hi Daniel and Benajmin,
>>
>> Thank you for reply and sharing link of mongodb basics. Yes there is a
>> performance issue. I want that opensips writes and get information of
>> dialog, location, subscriber etc from cachedb(mongodb). For this task
>> i came across with the module named DB_CACHEDB (
>> http://www.opensips.org/html/docs/modules/2.1.x/db_cachedb.html [3] ).
>>
>>
>>  According to documentation what this module do is " THE DB_CACHEDB
>> MODULE WILL EXPOSE THE SAME FRONT DB API, HOWEVER IT WILL RUN ON TOP
>> OF A NOSQL BACK-END, EMULATING THE SQL CALLS TO THE BACK-END SPECIFIC
>> QUERIES."
>>
>> I implemented this module and also created collections in mongodb
>> according to mysql tables list. Now when i start opensips it gives me
>> following error
>>
>> Jul 14 17:27:45 localhost [23610]: ERROR:core:db_check_api: module
>> db_mongodb does not export db_use_table function
>> Jul 14 17:27:45 localhost [23610]: ERROR:uri:mod_init: No database
>> module found
>> Jul 14 17:27:45 localhost [23610]: ERROR:core:init_mod: failed to
>> initialize module uri
>> Jul 14 17:27:45 localhost [23610]: ERROR:core:main: error while
>> initializing modules
>>
>> my opensips version is 2.1.3
>>
>> On Wed, Jul 13, 2016 at 6:52 PM, Benjamin Cropley
>>  wrote:
>>
>> You don't have to explicitly define a schema when you insert a
>>> document into MongoDB.
>>>
>>> For example, you could create a MongoDB database, and then
>>> immediately do an insert into an insert into a collection without
>>> even creating it or a 'schema'.
>>>
>>> Maybe you should do some reading about MongoDB and NoSQL in general,
>>> as a basic understanding should have answered that question :)
>>>
>>> https://docs.mongodb.com/manual/faq/fundamentals/ [2]
>>>
>>> On Wed, Jul 13, 2016 at 2:44 PM, feroze waris
>>>  wrote:
>>>
>>> Hi,

 Can anyone tell me how to move opensips DB from SQL to NoSQL. I
 have seen a module name db_cachedb which has a support for mongodb
 but how to move mysql schema to mongodb schema.

 Regards
 Feroze

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>>>
>>> --
>>>
>>> All the best,
>>> Ben Cropley
>>> 07539 366 905
>>> ___
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>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users [1]
>>>
>>
>>
>>
>> Links:
>> --
>> [1] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> [2] https://docs.mongodb.com/manual/faq/fundamentals/
>> [3] http://www.opensips.org/html/docs/modules/2.1.x/db_cachedb.html
>>
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Re: [OpenSIPS-Users] horizontal scaling / dimensioning

[qasimak...@gmail.com]

I guess you can reffer to this -> (
http://www.opensips.org/About/PerformanceTests-StressTests)  document for
some estimates and +1 Eric.

Regards,
Qasim

On Wed, Jul 13, 2016 at 11:09 PM, Eric Tamme  wrote:

> Short answer,  no.  There are many variables that will change what a
> system is capable of and without understanding all the pieces involved, you
> can not make any estimate at sizing.
>
>
> On 07/13/2016 11:51 AM, Owais Ahmad wrote:
>
> Hi all,
>
> Is there a way I can theoretically determine the max number of dialogs and
> cps a system of known specifications can handle?
>
> Also share useful tools for dimensioning, and your findings of max
> concurrent calls and cps on single opensips instances with specs.
>
> Thanks,
> Owais
>
>
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>
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>
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Re: [OpenSIPS-Users] Segfault in opensips 2.2

[qasimak...@gmail.com]

Hi,

No we are not using any save() or lookup() in our code.

Regards,
Qasim

On Thu, Jun 2, 2016 at 12:25 PM, Bogdan-Andrei Iancu 
wrote:

> Hi Qasim,
>
> Thank you for your report.
>
> In your opensips cfg, do you have any save() or lookup() ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 01.06.2016 22:21, qasimak...@gmail.com wrote:
>
> Dear Team,
>
> There is another segfault when i try to run with my old configuration from
> opensips 1.11. as far as i can understand it dosent go beyond loading the
> modules. Please find below logs and backtrace.
>
> syslog: http://pastebin.com/EAqTKu1n
> backtrace: http://pastebin.com/rP9JDeDW
>
> Regards,
> Qasim
>
>
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>
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[OpenSIPS-Users] Segfault in opensips 2.2

[qasimak...@gmail.com]

Dear Team,

There is another segfault when i try to run with my old configuration from
opensips 1.11. as far as i can understand it dosent go beyond loading the
modules. Please find below logs and backtrace.

syslog: http://pastebin.com/EAqTKu1n
backtrace: http://pastebin.com/rP9JDeDW

Regards,
Qasim
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Re: [OpenSIPS-Users] Segfault using Loadbalancer Module.

[qasimak...@gmail.com]

Thank Liviu... Its fixed now.

Regards,
Qasim

On Wed, Jun 1, 2016 at 4:48 PM, Liviu Chircu  wrote:

> Thank you for the backtrace, Qasim! It was very helpful.
>
> I managed to fix the issue on the "master" and "2.2" git branches [1].
> Please pull the latest sources and redo your tests!
>
> [1]: https://github.com/OpenSIPS/opensips.git
>
> Liviu Chircu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 01.06.2016 13:17, qasimak...@gmail.com wrote:
>
> Dear Razvan,
>
> Please find below backtrace of the core file:
>
> http://pastebin.com/WpLtezAS
>
> Regards,
> Qasim Ayyaz Khan
>
> On Wed, Jun 1, 2016 at 11:46 AM, Răzvan Crainea < 
> raz...@opensips.org> wrote:
>
>> Hi, Qasim!
>>
>>
>> Yes, please post a backtrace of the core dump on pastebin for further
>> investigation.
>>
>> Best regards,
>>
>> Răzvan Crainea
>> OpenSIPS Solutionswww.opensips-solutions.com
>>
>> On 05/31/2016 10:12 PM, qasimak...@gmail.com wrote:
>>
>> Hi,
>>
>> I was using default script generated by opensips menuconfig and it gives
>> the following segfault
>>
>> http://pastebin.com/6zuimn5N
>>
>> I was evaluating opensips 2.2 latest release. Please let me know if core
>> dump is required
>>
>> Regards,
>> Qasim Ayyaz Khan
>>
>>
>> ___
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>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
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>
>
>
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Re: [OpenSIPS-Users] Segfault using Loadbalancer Module.

[qasimak...@gmail.com]

Dear Razvan,

Please find below backtrace of the core file:

http://pastebin.com/WpLtezAS

Regards,
Qasim Ayyaz Khan

On Wed, Jun 1, 2016 at 11:46 AM, Răzvan Crainea  wrote:

> Hi, Qasim!
>
>
> Yes, please post a backtrace of the core dump on pastebin for further
> investigation.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 05/31/2016 10:12 PM, qasimak...@gmail.com wrote:
>
> Hi,
>
> I was using default script generated by opensips menuconfig and it gives
> the following segfault
>
> http://pastebin.com/6zuimn5N
>
> I was evaluating opensips 2.2 latest release. Please let me know if core
> dump is required
>
> Regards,
> Qasim Ayyaz Khan
>
>
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>
>
>
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[OpenSIPS-Users] Segfault using Loadbalancer Module.

[qasimak...@gmail.com]

Hi,

I was using default script generated by opensips menuconfig and it gives
the following segfault

http://pastebin.com/6zuimn5N

I was evaluating opensips 2.2 latest release. Please let me know if core
dump is required

Regards,
Qasim Ayyaz Khan
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Re: [OpenSIPS-Users] Problem using radius_send_auth

[qasimak...@gmail.com]

Hi John,

I am successfully using these functions (radius_send_auth)/dictionaries
with FreeRadius as radius server, I dont know about your particular radius
setup. As far as i know you dont need to load any other dictionary for your
radius related modules/functions to work. If your Opensip's dictionary is
properly loaded that should cater all your radius related functionality in
opensips.

Having said that you should make sure that the dictionary is loaded on both
Radius Server as well as on Client end that is the only requirement as far
as dictionaries are concerned.

Regards,
Qasim


On Mon, Mar 24, 2014 at 10:23 PM, John Quick wrote:

> Hi Quasim,
>
> I appreciate your help. However, I am not using FreeRadius as the Radius
> server and have already got all the basic dictionaries loaded (like
> dictionary.opensips, dictionary.sip)
> aaa_www_authorize and writing of Radius CDR's is working ok. That is not
> the
> problem.
>
> It is only when I try to use the radius_send_auth(set1, set2) function that
> I had problems.
> Please can you confirm if you have used this function?
>
> I just tried a change to the dictionaries I use. No longer using
> dictionary.rfc2869. Instead using dictionary.rfc2865.
> In set1, the attribs that are sent to the server, I now specify 'User-Name'
> and 'User-Password'. This seems to have fixed the problem whereby OpenSIPS
> required the Message-Authenticator attribute. However, to get
> dictionary.rfc2865 to work, I had to comment out all the attributes of type
> "octets".
>
> I still have the second problem: OpenSIPS and radiusclient-ng does not
> recognise the attribute type "octets".
>
> John Quick
> Smartvox Limited
>
>
> From: qasimak...@gmail.com [mailto:qasimak...@gmail.com]
> Sent: 24 March 2014 16:27
> To: john.qu...@smartvox.co.uk
> Cc: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Problem using radius_send_auth
>
> hmm... here are the settings that i am using that works perfectly for me:
> These files are required on opensips radiusclient-ng side:
> /etc/radiusclient-ng/dictionary
> ...
> $INCLUDE/etc/radiusclient-ng/dictionary.sip
> ...
> ATTRIBUTE   User-Name   1   string
> ATTRIBUTE   Password2   string
> ATTRIBUTE   CHAP-Password   3   string
> ...
> /etc/radiusclient-ng/dictionary.sip (This is the opensips dictionary)
> ## $Id: dictionary.opensips 7139 2010-08-17 14:06:00Z razvancrainea $
> ...
> ATTRIBUTE Sip-Uri-User 208  string # Proprietary, auth_radius
> ATTRIBUTE Sip-Group211  string # Proprietary, group_radius
> ATTRIBUTE Sip-Rpid 213  string # Proprietary, auth_radius
> ATTRIBUTE SIP-AVP  225  string # Proprietary, avp_radius
> ATTRIBUTE Sip-Call-Duration227  integer
> ATTRIBUTE Sip-Call-Setuptime   228  integer
> ...
>
> On freeradius end:
> /usr/local/etc/raddb
>
> $INCLUDE/usr/local/share/freeradius/dictionary
>
>
> /usr/local/share/freeradius/dictionary
> ...
> $INCLUDE dictionary.sip
> ...
>
> /usr/local/share/freeradius/dictionary.sip (This is the opensips
> dictionary)
>
> ## $Id: dictionary.opensips 7139 2010-08-17 14:06:00Z razvancrainea $
> ...
> P.S. If you need these dictionary files just PM me and i will send them to
> you i think these are not required on the forum it will just clutter things
> if anything.
>
> Regards,
> Qasim
>
> On Mon, Mar 24, 2014 at 6:05 PM, John Quick 
> wrote:
> I am already using the opensips dictionary.
> It does not contain the Message-Authenticator attribute.
>
> When I do not use dictionary.rfc2869, I get this error every time the
> radius_send_auth function is called:
> rc_avpair_gen: received unknown attribute 80 of length 18: 0x..
>
> When I include dictionary.rfc2869, I get this error on startup:
> rc_read_dictionary: invalid type on line 13 of dictionary
> /usr/local/etc/radiusclient-ng/dictionary.rfc2869
>
> John
>
>
> From: qasimak...@gmail.com [mailto:qasimak...@gmail.com]
> Sent: 24 March 2014 11:57
> To: john.qu...@smartvox.co.uk; OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Problem using radius_send_auth
>
> Try using opensips dictionary.
> Regards,
> Qasim
>
>
> On Mon, Mar 24, 2014 at 4:37 PM, John Quick 
> wrote:
> I'm using OpenSIPS version 1.8.2 with radiusclient-ng.
> I need to be able to make custom radius authentication requests using
> radius_send_auth (a function in the aaa_radius module).
>
> The first time I tried, it failed and reported an error that
> Message-Authenticator was an unknown attribute.
> I found the missin

Re: [OpenSIPS-Users] An Error in Opensips 1.9.1

[qasimak...@gmail.com]

I think a little more information than this would be required if you need
help :).

Regards,
Qasim


On Mon, Mar 24, 2014 at 4:36 PM, dpa  wrote:

> Hello!
>
>
>
> 1.   In log file I see many errors
>
> "CRITICAL:core:comp_scriptvar: invalid operation 20/3/4!!
>
> WARNING:core:do_action: error in expression (l=662)"
>
>
>
> As I understand there is a problem somewhere in opensips.cfg
>
> How can I understand where problem is?
>
>
>
> 2.   I have "modparam("dialog", "db_mode", 1)" in opensips.cfg.
>
> In normal operation of opensips I can see in statistics, for example, 800
> active dialogs, but after restart I see much more active dialogs. Why can
> it be?
>
>
>
> Thank you for any help.
>
>
>
>
>
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Re: [OpenSIPS-Users] Problem using radius_send_auth

[qasimak...@gmail.com]

hmm... here are the settings that i am using that works perfectly for me:

These files are required on opensips radiusclient-ng side:


*/etc/radiusclient-ng/dictionary*...
$INCLUDE/etc/radiusclient-ng/dictionary.sip
...
ATTRIBUTE   User-Name   1   string
ATTRIBUTE   Password2   string
ATTRIBUTE   CHAP-Password   3   string
...


*/etc/radiusclient-ng/dictionary.sip (This is the opensips dictionary)*
## $Id: dictionary.opensips 7139 2010-08-17 14:06:00Z razvancrainea $
...
ATTRIBUTE Sip-Uri-User 208  string # Proprietary, auth_radius
ATTRIBUTE Sip-Group211  string # Proprietary, group_radius
ATTRIBUTE Sip-Rpid 213  string # Proprietary, auth_radius
ATTRIBUTE SIP-AVP  225  string # Proprietary, avp_radius
ATTRIBUTE Sip-Call-Duration227  integer
ATTRIBUTE Sip-Call-Setuptime   228  integer
...


On freeradius end:

*/usr/local/etc/raddb*
$INCLUDE/usr/local/share/freeradius/dictionary



*/usr/local/share/freeradius/dictionary*...
$INCLUDE dictionary.sip
...

*/usr/local/share/freeradius/dictionary.sip* *(This is the opensips
dictionary)*

## $Id: dictionary.opensips 7139 2010-08-17 14:06:00Z razvancrainea $
...

P.S. If you need these dictionary files just PM me and i will send them to
you i think these are not required on the forum it will just clutter things
if anything.

Regards,
Qasim

On Mon, Mar 24, 2014 at 6:05 PM, John Quick wrote:

> I am already using the opensips dictionary.
> It does not contain the Message-Authenticator attribute.
>
> When I do not use dictionary.rfc2869, I get this error every time the
> radius_send_auth function is called:
> rc_avpair_gen: received unknown attribute 80 of length 18: 0x..
>
> When I include dictionary.rfc2869, I get this error on startup:
> rc_read_dictionary: invalid type on line 13 of dictionary
> /usr/local/etc/radiusclient-ng/dictionary.rfc2869
>
> John
>
>
> From: qasimak...@gmail.com [mailto:qasimak...@gmail.com]
> Sent: 24 March 2014 11:57
> To: john.qu...@smartvox.co.uk; OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Problem using radius_send_auth
>
> Try using opensips dictionary.
> Regards,
> Qasim
>
>
> On Mon, Mar 24, 2014 at 4:37 PM, John Quick 
> wrote:
> I'm using OpenSIPS version 1.8.2 with radiusclient-ng.
> I need to be able to make custom radius authentication requests using
> radius_send_auth (a function in the aaa_radius module).
>
> The first time I tried, it failed and reported an error that
> Message-Authenticator was an unknown attribute.
> I found the missing attribute in dictionary.rfc2869, but when I include
> this
> dictionary, OpenSIPS fails to start and reports an error that seems to
> point
> to the "octets" attribute type being unrecognised.
>
> Any help with this would be greatly appreciated.
>
> John Quick
> Smartvox Limited
> Web: www.smartvox.co.uk
>
>
>
>
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>
>
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Re: [OpenSIPS-Users] Problem using radius_send_auth

[qasimak...@gmail.com]

Try using opensips dictionary.

Regards,
Qasim



On Mon, Mar 24, 2014 at 4:37 PM, John Quick wrote:

> I'm using OpenSIPS version 1.8.2 with radiusclient-ng.
> I need to be able to make custom radius authentication requests using
> radius_send_auth (a function in the aaa_radius module).
>
> The first time I tried, it failed and reported an error that
> Message-Authenticator was an unknown attribute.
> I found the missing attribute in dictionary.rfc2869, but when I include
> this
> dictionary, OpenSIPS fails to start and reports an error that seems to
> point
> to the "octets" attribute type being unrecognised.
>
> Any help with this would be greatly appreciated.
>
> John Quick
> Smartvox Limited
> Web: www.smartvox.co.uk
>
>
>
>
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Re: [OpenSIPS-Users] installing opensips on Debian / Ubuntu

[qasimak...@gmail.com]

Hi,

Please follow this tutorial line by line.

http://saevolgo.blogspot.com/2012/05/installing-opensips-on-ubuntu-server.html

Regards,
Qasim


On Tue, Feb 25, 2014 at 6:07 PM, Tomasz Chmielewski  wrote:

> I'm trying to install opensips on Debian or Ubuntu.
>
> However, the provided deb packages seem to have wrong dependencies which
> prevent the installation:
>
> # apt-get install opensips opensips-console opensips-http-modules
> opensips-mysql-module opensips-identity-module
> Reading package lists... Done
> Building dependency tree
> Reading state information... Done
> Some packages could not be installed. This may mean that you have
> requested an impossible situation or if you are using the unstable
> distribution that some required packages have not yet been created
> or been moved out of Incoming.
> The following information may help to resolve the situation:
>
> The following packages have unmet dependencies:
>  opensips-mysql-module : Depends: libmysqlclient16 (>= 5.1.21-1) but it is
> not installable
> E: Unable to correct problems, you have held broken packages.
>
>
>
> I've tried using from "deb http://apt.opensips.org/ stable110 main"
> http://apt.opensips.org/ on Debian 7 and Ubuntu 12.04.4 LTS.
> Both have similarly broken dependencies.
>
>
> How to best install on Ubuntu or Debian?
>
> --
> Tomasz Chmielewski
> http://wpkg.org
>
>
>
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Re: [OpenSIPS-Users] Dialog timeout_avp.

[qasimak...@gmail.com]

yes.

Regards,
Qasim


On Mon, Jun 24, 2013 at 6:51 PM, Laszlo  wrote:

> Do you use create_dialog("B"); ?
>
> -Laszlo
>
>
> 2013/6/24 qasimak...@gmail.com 
>
>> Hi,
>>
>> I am using radius accounting and during that accounting i calculate the
>> maximum amount to call duration. I am setting dialog timout_avp in my route
>> but the call doesn't hangup. From documentation i see that i should use
>> timeout_avp before loose_route().
>>
>> My question is that how i can use timeout_avp when i authenticate the
>> call after loose_route()? I don't want to use call control module as i
>> don't like external dependency.
>>
>> Thanks & Regards,
>> Qasim
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
>
> --
> Kind regards,
> Laszlo Bekesi
> http://voipfreak.net
>
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[OpenSIPS-Users] Dialog timeout_avp.

[qasimak...@gmail.com]

Hi,

I am using radius accounting and during that accounting i calculate the
maximum amount to call duration. I am setting dialog timout_avp in my route
but the call doesn't hangup. From documentation i see that i should use
timeout_avp before loose_route().

My question is that how i can use timeout_avp when i authenticate the call
after loose_route()? I don't want to use call control module as i don't
like external dependency.

Thanks & Regards,
Qasim
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Re: [OpenSIPS-Users] radius acc in local_route on dialog timeout

[qasimak...@gmail.com]

Hi Dani,

You most probably don't have correct dictionary files placed. You can turn
debug=6 and it then see if you have any dictionary items missing. Every
time i install a new opensips with radius accounting i end up missing
dictionary file in one or more places and opensips does not show it to you
unless you have debug on.

Regards,
Qasim


On Thu, Jun 20, 2013 at 11:25 PM, Dani Popa  wrote:

> any ideea ?
>
>
> On Tue, Jun 18, 2013 at 7:10 PM, Dani Popa  wrote:
>
>> Hi all,
>> I use acc with radius and  when i set accountig flag in local_route  i
>> dont receive any accountig request on radius server.  As I see local_route
>> was hit twice on "dialog timeout" and i dont understand when and how many
>> request should i receive on accounting if should i receive accounting
>> request. Or should i user radius_send_acc in this case.
>>
>>
>>
>> this is my local_route
>>
>> local_route {
>>xlog"local route");
>> if (is_method("BYE")) {
>> xlog("acc 1");
>> setflag(ACCOUNTING_FLAG);
>> #acc_db_request("200 Dialog Timeout", "acc");
>>
>> }
>> }
>>
>>
>> Thanks,
>> --
>> Dani Popa
>>
>
>
>
> --
> Dani Popa
>
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Re: [OpenSIPS-Users] VIA relay error using mhomed=1

[qasimak...@gmail.com]

Hi Bodgan,

Thanks for all your help. It turns out that i had configured multiple IP's
on virtual ports and needed to set "listen=" param according to my
priority. That fixed the problem for me.

Regards,
Qasim


On Fri, May 24, 2013 at 8:30 PM, Bogdan-Andrei Iancu wrote:

> **
> You actually have a loop - not actually a loop, but double processing or
> so. On loose_route() true branch, the route(2) has not "exit" neither at
> the end (of route 2 block), nor after the route(2) invocation -> your
> script will continue and probably does moe stuff which was not intended.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 05/24/2013 04:58 PM, qasimak...@gmail.com wrote:
>
>  Dear Bodgan,
> This is what i am doing for ACK path (This is minus all the other crap
> like auth, redis stuff). I dont think that there would be any loop here.
>
>
> if (nat_uac_test("19")) {
> if (is_method("REGISTER")) {
>   fix_nated_register();
>} else {
>   fix_nated_contact();
>};
>  }
>
>  force_rport();
>
>  if (loose_route()) {
> if (loose_route()) {
># route it out to whatever destination was set by loose_route()
>   # in $du (destination URI).
>   route(2);
> }
>  } else {
> if ( is_method("ACK") ) {
>if (t_check_trans()) {
>   t_relay();
>   exit;
>} else {
>   exit;
>}
> }
>  }
>
>  route[2] {
> if (is_direction("downstream")) {
>   xlog("L_NOTICE", "[$pr:$fU@$si:$sp]: Sequencial '$rm' request from
> caller '$fU' for call from '$fu' to '$ru' \n");
>} else {
>   xlog("L_NOTICE", "[$pr:$fU@$si:$sp]: Sequencial '$rm' request from
> callee '$fU' for call from '$ru' to '$fu' \n");
> };
>if(is_method("ACK")) {
>$avp(pdd) = 0;
>$avp(pdd) = $Ts - $(avp(pdd){s.int});
>xlog("L_NOTICE", "[$pr:$fU@$si:$sp]: Post Dial Delay of Call-ID
> '$ci' from '$fu' to '$ru' is '$avp(pdd)' at '$time(%F %T %Z)' \n");
> }
> if (!t_relay()) {
>sl_reply_error();
> }
>  }
>
>
>
>
>
>  On Wed, May 22, 2013 at 10:01 PM, Bogdan-Andrei Iancu <
> bog...@opensips.org> wrote:
>
>>  No, it is not a retransmission as it is the same process and there is
>> no second set of logs for receiving a message from network:
>>
>> May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: DBG:core:parse_msg:
>> SIP Request:
>> May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: DBG:core:parse_msg:
>> method:  
>> May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: DBG:core:parse_msg:
>> uri:
>> 
>> May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: DBG:core:parse_msg:
>> version: 
>> 
>> May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]:
>> DBG:core:receive_msg: preparing to run routing scripts...
>> ...
>> May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: DBG:rr:after_loose:
>> Topmost route URI: '
>> sip:622190004...@xx.xx.xx.xx:6000;lr;ftag=2e76e266;did=c7c.34372c92' is
>> me
>> ...
>> May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: [
>> udp:622190004001@39.42.183.233:7085]: Sequencial 'ACK' request from
>> caller '622..' for call from ...
>> 
>> May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: DBG:rr:after_loose:
>> Topmost route URI: '
>> sip:622190004...@xx.xx.xx.xx:6000;lr;ftag=2e76e266;did=c7c.34372c92' is
>> me
>> ...
>> May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: [
>> udp:622190004001@39.42.183.233:7085]: Sequencial 'ACK' request from
>> caller '622.' for call from 
>>
>>
>> It is clearly a loop.
>>
>> Regards,
>>
>>  Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>>
>>   On 05/22/2013 03:21 PM, qasimak...@gmail.com wrote:
>>
>> I think that is retransmission of ACK packet because it didn't get its
>> 200 ok back.
>>
>> Regards,
>> Qasim
>>
>>
>> On Tue, May 21, 2013 at 10:08 PM, Bogdan-Andrei Iancu <
>> bog...@opensips.org> wrote:
>>
>>>  Hi Qasim,
>>>
>>> Looking at the ACK related logs, I see you get the script log
>>>  Sequencial 

Re: [OpenSIPS-Users] RTPProxy Forced Symmetric RTP.

[qasimak...@gmail.com]

Thanks Razvan :).

Regards,
Qasim


On Mon, May 27, 2013 at 3:37 PM, Răzvan Crainea  wrote:

> Hi, Qasim!
>
> The s and w flag do the exact same thing - instruct RTPProxy that
> symmetric NAT should be used. You can use either of them and you will get
> the same behavior. It is not similar to the i/e flags.
>
> Best regards,
>
> Razvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.**com <http://www.opensips-solutions.com>
>
>
> On 05/27/2013 09:53 AM, qasimak...@gmail.com wrote:
>
>> Hi,
>>
>> I wanted to know that there are two flags to force Symmetric RTP in
>> rtpproxy module i.e. s/w. Does these two flags have different working
>> principles or does it work exactly like the i/e flags which work in
>> bridging mode?
>>
>> Regards,
>> Qasim
>>
>>
>> __**_
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>> http://lists.opensips.org/cgi-**bin/mailman/listinfo/users<http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>>
>>
> __**_
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[OpenSIPS-Users] RTPProxy Forced Symmetric RTP.

[qasimak...@gmail.com]

Hi,

I wanted to know that there are two flags to force Symmetric RTP in
rtpproxy module i.e. s/w. Does these two flags have different working
principles or does it work exactly like the i/e flags which work in
bridging mode?

Regards,
Qasim
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[OpenSIPS-Users] RTPProxy Forced Symmetric RTP.

[qasimak...@gmail.com]

Hi,

I wanted to know that there are two flags to force Symmetric RTP in
rtpproxy module i.e. s/w. Does these two flags have different working
principles or does it work exactly like the i/e flags which work in
bridging mode?

Regards,
Qasim
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Re: [OpenSIPS-Users] VIA relay error using mhomed=1

[qasimak...@gmail.com]

Dear Bodgan,
This is what i am doing for ACK path (This is minus all the other crap like
auth, redis stuff). I dont think that there would be any loop here.


if (nat_uac_test("19")) {
   if (is_method("REGISTER")) {
  fix_nated_register();
   } else {
  fix_nated_contact();
   };
}

force_rport();

if (loose_route()) {
   if (loose_route()) {
  # route it out to whatever destination was set by loose_route()
  # in $du (destination URI).
  route(2);
   }
} else {
   if ( is_method("ACK") ) {
  if (t_check_trans()) {
 t_relay();
 exit;
  } else {
 exit;
  }
   }
}

route[2] {
   if (is_direction("downstream")) {
  xlog("L_NOTICE", "[$pr:$fU@$si:$sp]: Sequencial '$rm' request from
caller '$fU' for call from '$fu' to '$ru' \n");
   } else {
  xlog("L_NOTICE", "[$pr:$fU@$si:$sp]: Sequencial '$rm' request from
callee '$fU' for call from '$ru' to '$fu' \n");
   };
   if(is_method("ACK")) {
  $avp(pdd) = 0;
  $avp(pdd) = $Ts - $(avp(pdd){s.int});
  xlog("L_NOTICE", "[$pr:$fU@$si:$sp]: Post Dial Delay of Call-ID '$ci'
from '$fu' to '$ru' is '$avp(pdd)' at '$time(%F %T %Z)' \n");
   }
   if (!t_relay()) {
  sl_reply_error();
   }
}





On Wed, May 22, 2013 at 10:01 PM, Bogdan-Andrei Iancu
wrote:

> **
> No, it is not a retransmission as it is the same process and there is no
> second set of logs for receiving a message from network:
>
> May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: DBG:core:parse_msg:
> SIP Request:
> May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: DBG:core:parse_msg:
> method:  
> May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: DBG:core:parse_msg:
> uri:
> 
> May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: DBG:core:parse_msg:
> version: 
> 
> May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: DBG:core:receive_msg:
> preparing to run routing scripts...
> ...
> May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: DBG:rr:after_loose:
> Topmost route URI: '
> sip:622190004...@xx.xx.xx.xx:6000;lr;ftag=2e76e266;did=c7c.34372c92' is me
> ...
> May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: [
> udp:622190004001@39.42.183.233:7085]: Sequencial 'ACK' request from
> caller '622..' for call from ...
> 
> May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: DBG:rr:after_loose:
> Topmost route URI: '
> sip:622190004...@xx.xx.xx.xx:6000;lr;ftag=2e76e266;did=c7c.34372c92' is me
> ...
> May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: [
> udp:622190004001@39.42.183.233:7085]: Sequencial 'ACK' request from
> caller '622.' for call from 
>
>
> It is clearly a loop.
>
> Regards,
>
>  Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 05/22/2013 03:21 PM, qasimak...@gmail.com wrote:
>
> I think that is retransmission of ACK packet because it didn't get its 200
> ok back.
>
> Regards,
> Qasim
>
>
> On Tue, May 21, 2013 at 10:08 PM, Bogdan-Andrei Iancu  > wrote:
>
>>  Hi Qasim,
>>
>> Looking at the ACK related logs, I see you get the script log
>>  Sequencial 'ACK' request from caller '622190004001' for call from
>> .
>>
>> twice - also the logs from the loose_route() function - I suspect you
>> loop somehow in your script and a route is triggered twice (the route doing
>> loose_route)
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>>
>>   On 05/20/2013 02:46 PM, qasimak...@gmail.com wrote:
>>
>>  Hi Bodgan,
>>
>>  Sorry for the late reply as i was traveling this weekend. Please find
>> attached call logs with debug mode 4.
>>
>> Regards,
>> Qasim
>>
>>
>>  On Fri, May 17, 2013 at 8:50 PM, Bogdan-Andrei Iancu <
>> bog...@opensips.org> wrote:
>>
>>>  Funny, as I do not see anything wrong on a first look - while running
>>> in debug mode (4), please send me the logs corresponding to the ACK
>>> processing.
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>>
>>>   On 05/17/2013 02:34 PM, qasimak...@gmail.com wrote:
>>>
>>>  Hi,
>>>
>>>  Please find attached trace. This is server on Public IP that is why i
&

Re: [OpenSIPS-Users] VIA relay error using mhomed=1

[qasimak...@gmail.com]

I think that is retransmission of ACK packet because it didn't get its 200
ok back.

Regards,
Qasim


On Tue, May 21, 2013 at 10:08 PM, Bogdan-Andrei Iancu
wrote:

> **
> Hi Qasim,
>
> Looking at the ACK related logs, I see you get the script log
>  Sequencial 'ACK' request from caller '622190004001' for call from
> .
>
> twice - also the logs from the loose_route() function - I suspect you loop
> somehow in your script and a route is triggered twice (the route doing
> loose_route)
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 05/20/2013 02:46 PM, qasimak...@gmail.com wrote:
>
>  Hi Bodgan,
>
>  Sorry for the late reply as i was traveling this weekend. Please find
> attached call logs with debug mode 4.
>
> Regards,
> Qasim
>
>
>  On Fri, May 17, 2013 at 8:50 PM, Bogdan-Andrei Iancu  > wrote:
>
>>  Funny, as I do not see anything wrong on a first look - while running
>> in debug mode (4), please send me the logs corresponding to the ACK
>> processing.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>>
>>   On 05/17/2013 02:34 PM, qasimak...@gmail.com wrote:
>>
>>  Hi,
>>
>>  Please find attached trace. This is server on Public IP that is why i
>> cannot send the trace on the list. I am listening to IP's as follows
>>
>> listen=udp:202.152.203.195:5060
>> listen=udp:202.152.203.195:6000
>> listen=udp:192.168.226.142:5060
>> listen=udp:192.168.226.142:6000
>>
>> disable_tcp=no
>> listen=tcp:202.152.203.195:5060
>> listen=tcp:202.152.203.195:6000
>> listen=tcp:192.168.226.142:5060
>> listen=tcp:192.168.226.142:6000
>>
>>  If you need anything else i would be happy to provide it to you.
>>
>>  Regards,
>> Qasim
>>
>>
>>
>> On Fri, May 17, 2013 at 3:50 PM, Bogdan-Andrei Iancu > > wrote:
>>
>>>  Hello Qasim,
>>>
>>> So you have multiple interfaces in OpenSIPS - are all of them the same
>>> protocol ?
>>>
>>> Please try to post a SIP capture of the full call, to see how the RR
>>> part is done.
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>>
>>>   On 05/16/2013 01:07 PM, qasimak...@gmail.com wrote:
>>>
>>> On further investigation i see that i only face this issue when both
>>> caller and callee are on the same network. If both are on separate network
>>> it works fine.
>>>
>>> Regards,
>>> Qasim
>>>
>>>
>>> On Thu, May 16, 2013 at 3:05 PM, qasimak...@gmail.com <
>>> qasimak...@gmail.com> wrote:
>>>
>>>>  yes.
>>>>
>>>>  Regards,
>>>> Qasim
>>>>
>>>>
>>>> On Thu, May 16, 2013 at 2:50 PM, Bogdan-Andrei Iancu <
>>>> bog...@opensips.org> wrote:
>>>>
>>>>>  And do you have UDP 202.152.203.195 port 6000 as listener defined in
>>>>> OpenSIPS ??
>>>>>
>>>>> Regards,
>>>>>
>>>>> Bogdan-Andrei Iancu
>>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>>
>>>>>
>>>>>   On 05/16/2013 12:32 PM, qasimak...@gmail.com wrote:
>>>>>
>>>>>  Hi Bodgan,
>>>>>
>>>>>  Yes i see the following route header in my packet.
>>>>>
>>>>> Route:
>>>>> 
>>>>>
>>>>>
>>>>>  And yes i am routing it through loose_route.
>>>>>
>>>>>  Regards,
>>>>> Qasim
>>>>>
>>>>>
>>>>> On Wed, May 15, 2013 at 10:40 PM, Bogdan-Andrei Iancu <
>>>>> bog...@opensips.org> wrote:
>>>>>
>>>>>>  Hello Qasim,
>>>>>>
>>>>>> The ACK should be routed via loose_route() based on the "Route"
>>>>>> headers from it. Could you check if the Route hdrs (from the ACK) are
>>>>>> correctly reflecting your opensips interfaces ?
>>>>>>
>>>>>> Best regards,
>>>>>>
>>>>>> Bogdan-Andrei Iancu
>>>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>>>
>>>>>>
>>>>>> On 05/14/2013 07:55 AM, qasimak...@gmail.com wrote:
>>>>>>
>>>>>>   Hi,
>>>>>>
>>>>>>  I am using OpenSIPs in Public<->Private bridging mode and have
>>>>>> enabled mhomed=1. But the problem is that when we have a call in which 
>>>>>> both
>>>>>> parties are on Public interface the INVITE gets relayed properly but and
>>>>>> ACK of that invite gives the following error.
>>>>>>
>>>>>> ERROR:core:get_out_socket: no socket found
>>>>>> ERROR:core:forward_request: cannot forward to af 2, proto 1 no
>>>>>> correspondinglistening socket
>>>>>>
>>>>>>  Regards,
>>>>>> Qasim
>>>>>>
>>>>>>
>>>>>> ___
>>>>>> Users mailing 
>>>>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>
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Re: [OpenSIPS-Users] VIA relay error using mhomed=1

[qasimak...@gmail.com]

Hi Bodgan,

I have sent you SIP capture in private as the server was on public IP.

Regards,
Qasim


On Fri, May 17, 2013 at 3:50 PM, Bogdan-Andrei Iancu wrote:

> **
> Hello Qasim,
>
> So you have multiple interfaces in OpenSIPS - are all of them the same
> protocol ?
>
> Please try to post a SIP capture of the full call, to see how the RR part
> is done.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 05/16/2013 01:07 PM, qasimak...@gmail.com wrote:
>
> On further investigation i see that i only face this issue when both
> caller and callee are on the same network. If both are on separate network
> it works fine.
>
> Regards,
> Qasim
>
>
> On Thu, May 16, 2013 at 3:05 PM, qasimak...@gmail.com <
> qasimak...@gmail.com> wrote:
>
>>  yes.
>>
>>  Regards,
>> Qasim
>>
>>
>> On Thu, May 16, 2013 at 2:50 PM, Bogdan-Andrei Iancu > > wrote:
>>
>>>  And do you have UDP 202.152.203.195 port 6000 as listener defined in
>>> OpenSIPS ??
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>>
>>>   On 05/16/2013 12:32 PM, qasimak...@gmail.com wrote:
>>>
>>>  Hi Bodgan,
>>>
>>>  Yes i see the following route header in my packet.
>>>
>>> Route:
>>> 
>>>
>>>
>>>  And yes i am routing it through loose_route.
>>>
>>>  Regards,
>>> Qasim
>>>
>>>
>>> On Wed, May 15, 2013 at 10:40 PM, Bogdan-Andrei Iancu <
>>> bog...@opensips.org> wrote:
>>>
>>>>  Hello Qasim,
>>>>
>>>> The ACK should be routed via loose_route() based on the "Route" headers
>>>> from it. Could you check if the Route hdrs (from the ACK) are correctly
>>>> reflecting your opensips interfaces ?
>>>>
>>>> Best regards,
>>>>
>>>> Bogdan-Andrei Iancu
>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>
>>>>
>>>> On 05/14/2013 07:55 AM, qasimak...@gmail.com wrote:
>>>>
>>>>   Hi,
>>>>
>>>>  I am using OpenSIPs in Public<->Private bridging mode and have enabled
>>>> mhomed=1. But the problem is that when we have a call in which both parties
>>>> are on Public interface the INVITE gets relayed properly but and ACK of
>>>> that invite gives the following error.
>>>>
>>>> ERROR:core:get_out_socket: no socket found
>>>> ERROR:core:forward_request: cannot forward to af 2, proto 1 no
>>>> correspondinglistening socket
>>>>
>>>>  Regards,
>>>> Qasim
>>>>
>>>>
>>>> ___
>>>> Users mailing 
>>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>
>>
>
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Re: [OpenSIPS-Users] VIA relay error using mhomed=1

[qasimak...@gmail.com]

On further investigation i see that i only face this issue when both caller
and callee are on the same network. If both are on separate network it
works fine.

Regards,
Qasim


On Thu, May 16, 2013 at 3:05 PM, qasimak...@gmail.com
wrote:

> yes.
>
> Regards,
> Qasim
>
>
> On Thu, May 16, 2013 at 2:50 PM, Bogdan-Andrei Iancu 
> wrote:
>
>> **
>> And do you have UDP 202.152.203.195 port 6000 as listener defined in
>> OpenSIPS ??
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>>
>> On 05/16/2013 12:32 PM, qasimak...@gmail.com wrote:
>>
>>  Hi Bodgan,
>>
>>  Yes i see the following route header in my packet.
>>
>> Route:
>> 
>>
>>
>>  And yes i am routing it through loose_route.
>>
>>  Regards,
>> Qasim
>>
>>
>> On Wed, May 15, 2013 at 10:40 PM, Bogdan-Andrei Iancu <
>> bog...@opensips.org> wrote:
>>
>>>  Hello Qasim,
>>>
>>> The ACK should be routed via loose_route() based on the "Route" headers
>>> from it. Could you check if the Route hdrs (from the ACK) are correctly
>>> reflecting your opensips interfaces ?
>>>
>>> Best regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>>
>>> On 05/14/2013 07:55 AM, qasimak...@gmail.com wrote:
>>>
>>>   Hi,
>>>
>>>  I am using OpenSIPs in Public<->Private bridging mode and have enabled
>>> mhomed=1. But the problem is that when we have a call in which both parties
>>> are on Public interface the INVITE gets relayed properly but and ACK of
>>> that invite gives the following error.
>>>
>>> ERROR:core:get_out_socket: no socket found
>>> ERROR:core:forward_request: cannot forward to af 2, proto 1 no
>>> correspondinglistening socket
>>>
>>>  Regards,
>>> Qasim
>>>
>>>
>>> ___
>>> Users mailing 
>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>
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Re: [OpenSIPS-Users] VIA relay error using mhomed=1

[qasimak...@gmail.com]

yes.

Regards,
Qasim


On Thu, May 16, 2013 at 2:50 PM, Bogdan-Andrei Iancu wrote:

> **
> And do you have UDP 202.152.203.195 port 6000 as listener defined in
> OpenSIPS ??
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 05/16/2013 12:32 PM, qasimak...@gmail.com wrote:
>
>  Hi Bodgan,
>
>  Yes i see the following route header in my packet.
>
> Route:
> 
>
>
>  And yes i am routing it through loose_route.
>
>  Regards,
> Qasim
>
>
> On Wed, May 15, 2013 at 10:40 PM, Bogdan-Andrei Iancu  > wrote:
>
>>  Hello Qasim,
>>
>> The ACK should be routed via loose_route() based on the "Route" headers
>> from it. Could you check if the Route hdrs (from the ACK) are correctly
>> reflecting your opensips interfaces ?
>>
>> Best regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>>
>> On 05/14/2013 07:55 AM, qasimak...@gmail.com wrote:
>>
>>   Hi,
>>
>>  I am using OpenSIPs in Public<->Private bridging mode and have enabled
>> mhomed=1. But the problem is that when we have a call in which both parties
>> are on Public interface the INVITE gets relayed properly but and ACK of
>> that invite gives the following error.
>>
>> ERROR:core:get_out_socket: no socket found
>> ERROR:core:forward_request: cannot forward to af 2, proto 1 no
>> correspondinglistening socket
>>
>>  Regards,
>> Qasim
>>
>>
>> ___
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
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Re: [OpenSIPS-Users] VIA relay error using mhomed=1

[qasimak...@gmail.com]

Hi Bodgan,

Yes i see the following route header in my packet.

Route: 


And yes i am routing it through loose_route.

Regards,
Qasim


On Wed, May 15, 2013 at 10:40 PM, Bogdan-Andrei Iancu
wrote:

> **
> Hello Qasim,
>
> The ACK should be routed via loose_route() based on the "Route" headers
> from it. Could you check if the Route hdrs (from the ACK) are correctly
> reflecting your opensips interfaces ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 05/14/2013 07:55 AM, qasimak...@gmail.com wrote:
>
>  Hi,
>
>  I am using OpenSIPs in Public<->Private bridging mode and have enabled
> mhomed=1. But the problem is that when we have a call in which both parties
> are on Public interface the INVITE gets relayed properly but and ACK of
> that invite gives the following error.
>
> ERROR:core:get_out_socket: no socket found
> ERROR:core:forward_request: cannot forward to af 2, proto 1 no
> correspondinglistening socket
>
>  Regards,
> Qasim
>
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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[OpenSIPS-Users] VIA relay error using mhomed=1

[qasimak...@gmail.com]

Hi,

I am using OpenSIPs in Public<->Private bridging mode and have enabled
mhomed=1. But the problem is that when we have a call in which both parties
are on Public interface the INVITE gets relayed properly but and ACK of
that invite gives the following error.

ERROR:core:get_out_socket: no socket found
ERROR:core:forward_request: cannot forward to af 2, proto 1 no
correspondinglistening socket

Regards,
Qasim
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Re: [OpenSIPS-Users] RTPProxy for users behind NAT.

[qasimak...@gmail.com]

Hi Razvan,

Thanks for your replies but i figured that i was using wrong flags along
with ie. Its working fine after fixing the flags.

Regards,
Qasim


On Fri, May 10, 2013 at 2:13 PM, Răzvan Crainea  wrote:

> Hi, Qasim!
>
> I am not sure what's your problem then. Are you saying that the SDP is
> properly changed for both Invite and 200 OK? Can you send a trace? Or at
> least explain exactly how each message (INVITE and 200 OK) is sent out by
> OpenSIPS.
>
>
> Best regards,
>
> Razvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.**com <http://www.opensips-solutions.com>
>
> On 05/10/2013 06:41 AM, qasimak...@gmail.com wrote:
>
>> Yes exactly that is being done perfectly but what i want to do is to
>> handle NAT on client's end. The IP of client that comes in the SDP's c=
>> param is his local IP address and rtpproxy swaps that IP with server's
>> local IP but on the other way arround it tries to send the IP back to
>> client's local IP address which is not visible to server.
>>
>> Actually we have two nated acenerios. One on the server end and the
>> other on the client's end.
>>
>> Regards,
>> Qasim
>>
>>
>> On Thu, May 9, 2013 at 5:59 PM, Răzvan Crainea > <mailto:raz...@opensips.org>> wrote:
>>
>> Hi, Qasim!
>>
>> Basically this is what the rtpproxy module does: when you call
>> rtpproxy_offer("ei") function, opensips tells the rtpproxy server
>> that a new session has to be created and the media flow will be from
>> external to internal. Rtpproxy assigns the proper interface(IP) and
>> port and returns them to OpenSIPS, which advertises in the ongoing
>> INVITE. So, considering the rtpproxy server has been configured
>> correctly, all you have to do is call rtpproxy_offer() with the
>> proper direction.
>>
>>
>> Best regards,
>>
>> Razvan Crainea
>> OpenSIPS Core Developer
>> http://www.opensips-solutions.**__com <http://www.opensips-**
>> solutions.com <http://www.opensips-solutions.com>>
>>
>> On 05/09/2013 02:54 PM, qasimak...@gmail.com
>>
>> <mailto:qasimak...@gmail.com> wrote:
>>
>> Hi Razvan,
>>
>> My scenerio is like this
>>
>> Client <---> NAT <---> OpenSIPs/RTPProxy <---> Client
>>
>>
>> in this scenerio left side of OpenSIPs is public side and the
>> right side
>> is on private network. Secondly i have tried using
>> rtpproxy_offer/answer() but the same problem. I will try using
>> rtpproxy_offer/answer() again in a bit more detail now specially
>> after
>> hearing about problems in engage_rtpproxy in brigding mode. Now
>> can you
>> point me how i can achieve nat handling in rtpproxy module?
>>
>> Regards,
>> Qasim
>>
>>
>> On Thu, May 9, 2013 at 5:39 PM, Răzvan Crainea
>> mailto:raz...@opensips.org>
>> <mailto:raz...@opensips.org <mailto:raz...@opensips.org>>> wrote:
>>
>>  Hi, Qasim!
>>
>>  There are two problems with your approach: the first one is
>> that you
>>  are using the engage_rtp_proxy() function in a bridging mode
>>  scenario. The behavior of this is undefined, because the
>> rtpproxy
>>  module cannot fully determine your scenario (for example
>> what's the
>>  direction of the media flow in the reply). That's why you
>> should use
>>  the rtpproxy_offer() and rtpproxy_answer() functions to
>> explicitly
>>  indicate the direction in INVITE and replies.
>>  The second problem is that you try to change the SDP twice:
>> first by
>>  the fix_nated_sdp() and then by engage_rtp_proxy(). These
>> changes
>>  confuse OpenSIPS, who tries to apply both of them. Try to
>> use only
>>  one. My suggestion is to rtpproxy_offer/answer() to fix the
>> SDP,
>>  without calling fix_nated_sdp().
>>
>>  Best regards,
>>
>>  Razvan Crainea
>>  OpenSIPS Core Developer
>>
>
> __**_
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Re: [OpenSIPS-Users] RTPProxy for users behind NAT.

[qasimak...@gmail.com]

Yes exactly that is being done perfectly but what i want to do is to handle
NAT on client's end. The IP of client that comes in the SDP's c= param is
his local IP address and rtpproxy swaps that IP with server's local IP but
on the other way arround it tries to send the IP back to client's local IP
address which is not visible to server.

Actually we have two nated acenerios. One on the server end and the other
on the client's end.

Regards,
Qasim


On Thu, May 9, 2013 at 5:59 PM, Răzvan Crainea  wrote:

> Hi, Qasim!
>
> Basically this is what the rtpproxy module does: when you call
> rtpproxy_offer("ei") function, opensips tells the rtpproxy server that a
> new session has to be created and the media flow will be from external to
> internal. Rtpproxy assigns the proper interface(IP) and port and returns
> them to OpenSIPS, which advertises in the ongoing INVITE. So, considering
> the rtpproxy server has been configured correctly, all you have to do is
> call rtpproxy_offer() with the proper direction.
>
>
> Best regards,
>
> Razvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.**com <http://www.opensips-solutions.com>
>
> On 05/09/2013 02:54 PM, qasimak...@gmail.com wrote:
>
>> Hi Razvan,
>>
>> My scenerio is like this
>>
>> Client <---> NAT <---> OpenSIPs/RTPProxy <---> Client
>>
>>
>> in this scenerio left side of OpenSIPs is public side and the right side
>> is on private network. Secondly i have tried using
>> rtpproxy_offer/answer() but the same problem. I will try using
>> rtpproxy_offer/answer() again in a bit more detail now specially after
>> hearing about problems in engage_rtpproxy in brigding mode. Now can you
>> point me how i can achieve nat handling in rtpproxy module?
>>
>> Regards,
>> Qasim
>>
>>
>> On Thu, May 9, 2013 at 5:39 PM, Răzvan Crainea > <mailto:raz...@opensips.org>> wrote:
>>
>> Hi, Qasim!
>>
>> There are two problems with your approach: the first one is that you
>> are using the engage_rtp_proxy() function in a bridging mode
>> scenario. The behavior of this is undefined, because the rtpproxy
>> module cannot fully determine your scenario (for example what's the
>> direction of the media flow in the reply). That's why you should use
>> the rtpproxy_offer() and rtpproxy_answer() functions to explicitly
>> indicate the direction in INVITE and replies.
>> The second problem is that you try to change the SDP twice: first by
>> the fix_nated_sdp() and then by engage_rtp_proxy(). These changes
>> confuse OpenSIPS, who tries to apply both of them. Try to use only
>> one. My suggestion is to rtpproxy_offer/answer() to fix the SDP,
>> without calling fix_nated_sdp().
>>
>> Best regards,
>>
>> Razvan Crainea
>> OpenSIPS Core Developer
>> http://www.opensips-solutions.**__com <http://www.opensips-**
>> solutions.com <http://www.opensips-solutions.com>>
>>
>>
>>
>> On 05/09/2013 02:33 PM, Nick Khamis wrote:
>>
>> It's not a bug, many of us here use RTP proxy in the same
>> scenario.
>>     Can you please provide a sip trace using ngrep
>> 
>> (http://wiki.freeswitch.org/__**wiki/Packet_Capture<http://wiki.freeswitch.org/__wiki/Packet_Capture>
>> 
>> <http://wiki.freeswitch.org/**wiki/Packet_Capture<http://wiki.freeswitch.org/wiki/Packet_Capture>>).
>> Secondly,
>>
>> post the
>> relevant far end nat related scripting please.
>>
>>
>> Nick.
>>
>> On 5/9/13, qasimak...@gmail.com <mailto:qasimak...@gmail.com>
>>
>> mailto:qasimak...@gmail.com>> wrote:
>>
>> Hi,
>>
>> I am facing a problem when a client connects to opensips
>> from NATed
>> network. I am using rtpproxy in bridging mode i.e. from
>> publicnetwork to
>> private network. When i use fix_nated_sdp function from
>> nathelper the local
>> IP address of the caller is replaced by its public IP but
>> the problem
>> starts when i use engage_rtp_proxy instead of replacing
>> server's public ip
>> to private it embeds private ip after the caller's publicIP
>> like
>> X.X.X.XY.Y.Y.Y. I have tried fix_nated_ip with flag 3 and
>> engag

Re: [OpenSIPS-Users] RTPProxy for users behind NAT.

[qasimak...@gmail.com]

Hi Razvan,

My scenerio is like this

Client <---> NAT <---> OpenSIPs/RTPProxy <---> Client


in this scenerio left side of OpenSIPs is public side and the right side is
on private network. Secondly i have tried using rtpproxy_offer/answer() but
the same problem. I will try using rtpproxy_offer/answer() again in a bit
more detail now specially after hearing about problems in engage_rtpproxy
in brigding mode. Now can you point me how i can achieve nat handling in
rtpproxy module?

Regards,
Qasim


On Thu, May 9, 2013 at 5:39 PM, Răzvan Crainea  wrote:

> Hi, Qasim!
>
> There are two problems with your approach: the first one is that you are
> using the engage_rtp_proxy() function in a bridging mode scenario. The
> behavior of this is undefined, because the rtpproxy module cannot fully
> determine your scenario (for example what's the direction of the media flow
> in the reply). That's why you should use the rtpproxy_offer() and
> rtpproxy_answer() functions to explicitly indicate the direction in INVITE
> and replies.
> The second problem is that you try to change the SDP twice: first by the
> fix_nated_sdp() and then by engage_rtp_proxy(). These changes confuse
> OpenSIPS, who tries to apply both of them. Try to use only one. My
> suggestion is to rtpproxy_offer/answer() to fix the SDP, without calling
> fix_nated_sdp().
>
> Best regards,
>
> Razvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.**com <http://www.opensips-solutions.com>
>
>
> On 05/09/2013 02:33 PM, Nick Khamis wrote:
>
>> It's not a bug, many of us here use RTP proxy in the same scenario.
>> Can you please provide a sip trace using ngrep
>> (http://wiki.freeswitch.org/**wiki/Packet_Capture<http://wiki.freeswitch.org/wiki/Packet_Capture>).
>> Secondly, post the
>> relevant far end nat related scripting please.
>>
>>
>> Nick.
>>
>> On 5/9/13, qasimak...@gmail.com  wrote:
>>
>>> Hi,
>>>
>>> I am facing a problem when a client connects to opensips from NATed
>>> network. I am using rtpproxy in bridging mode i.e. from publicnetwork to
>>> private network. When i use fix_nated_sdp function from nathelper the
>>> local
>>> IP address of the caller is replaced by its public IP but the problem
>>> starts when i use engage_rtp_proxy instead of replacing server's public
>>> ip
>>> to private it embeds private ip after the caller's publicIP like
>>> X.X.X.XY.Y.Y.Y. I have tried fix_nated_ip with flag 3 and
>>> engage_rtp_proxy
>>> with flag rie.
>>>
>>> The question is am i using something wrong here or should it be counted
>>> as
>>> a bug.
>>>
>>> If you want some more clarification can draw a flow diagram also.
>>>
>>> Regards,
>>> Qasim
>>>
>>>
>> __**_
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>>
>>
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Re: [OpenSIPS-Users] OpenSIPS with public/private interface and RTPProxy

[qasimak...@gmail.com]

For engage_rtpproxy there are two flags that are used i.e. i for LAN
interface and E for WAN interface. you can use these two flags to specify
your direction of bridging. e.g. ie for LAN to WAN bridging and ei for WAN
to LAN bridging. Meanwhile look at this documentation for detailed flag
usage.

http://www.opensips.org/html/docs/modules/1.8.x/rtpproxy.html#id292744


Regards,
Qasim


On Thu, May 9, 2013 at 4:09 PM, Michele Pinassi wrote:

> Hi all,
>
> i have an OpenSIPS server with two interface, PUBLIC (xxx) and PRIVATE
> (172.20.1.2). The PRIVATE interface works inside a LAN dedicated to
> VoIP, with a MediaServer (172.20.1.5) and a Patton Gateway for PSTN
> (172.20.1.4).
>
> Users phone's can register on both interface and i use RTPProxy (in
> bridging mode) to ensure that both side can talk togheter.
>
> But something don't work as expected
>
> Here's my OpenSIPS routing logic:
>
> ===
> route{
> if (!mf_process_maxfwd_header("10")) {
> sl_send_reply("483","Too Many Hops");
> exit;
> }
>
> if (msg:len >=  2048 ) {
> sl_send_reply("513", "Message too big");
> exit;
> };
>
> if(is_method("INVITE") && has_totag()) {
> engage_rtp_proxy();
> }
>
> if (has_totag()) {
> # sequential request withing a dialog should
> # take the path determined by record-routing
> if (loose_route()) {
> if (is_method("BYE")) {
> setflag(1); # do accounting ...
> setflag(3); # ... even if the transaction
> fails
> } else if (is_method("INVITE")) {
> # even if in most of the cases is useless,
> do RR for
> # re-INVITEs alos, as some buggy clients
> do change route set
> # during the dialog.
> record_route();
> }
> # route it out to whatever destination was set by
> loose_route()
> # in $du (destination URI).
> route(1);
> } else {
> /* uncomment the following lines if you want to
> enable presence */
> if (is_method("SUBSCRIBE") && $rd == "
> voip.unisi.it") {
> # in-dialog subscribe requests
> route(2);
> exit;
> }
> if ( is_method("ACK") ) {
> if ( t_check_trans() ) {
> # non loose-route, but stateful
> ACK; must be an ACK after
> # a 487 or e.g. 404 from upstream
> server
> t_relay();
> exit;
> } else {
> # ACK without matching transaction
> ->
> # ignore and discard
> exit;
> }
> }
> sl_send_reply("404","Not here");
> }
> exit;
> }
>
> #initial requests
>
> # CANCEL processing
> if (is_method("CANCEL"))
> {
> if (t_check_trans())
> t_relay();
> exit;
> }
>
> t_check_trans();
>
> # authenticate if from local subscriber (uncomment to enable auth)
> # authenticate all initial non-REGISTER request that pretend to be
> # generated by local subscriber (domain from FROM URI is local)
> # if (!(method=="REGISTER") && from_uri==myself) /*no multidomain
> version*/
> if (!(method=="REGISTER") && is_from_local())  /*multidomain
> version*/
> {
> if(!check_source_address("0")){
> if (!proxy_authorize("", "subscriber")) {
> proxy_challenge("", "0");
> exit;
> }
> if (!db_check_from()) {
> sl_send_reply("403","Forbidden auth ID");
> exit;
> }
>
> consume_credentials();
> # caller authenticated
> }
> }
>
> # preloaded route checking
> if (loose_route()) {
> xlog("L_ERR", "Attempt to route with preloaded Route's
> [$fu/$tu/$ru/$ci]");
> if (!is_method("ACK"))
> sl_send_reply("403","Preload Route denied");
> exit;
> }
>
> # record routing
>  

[OpenSIPS-Users] RTPProxy for users behind NAT.

[qasimak...@gmail.com]

Hi,

I am facing a problem when a client connects to opensips from NATed
network. I am using rtpproxy in bridging mode i.e. from publicnetwork to
private network. When i use fix_nated_sdp function from nathelper the local
IP address of the caller is replaced by its public IP but the problem
starts when i use engage_rtp_proxy instead of replacing server's public ip
to private it embeds private ip after the caller's publicIP like
X.X.X.XY.Y.Y.Y. I have tried fix_nated_ip with flag 3 and engage_rtp_proxy
with flag rie.

The question is am i using something wrong here or should it be counted as
a bug.

If you want some more clarification can draw a flow diagram also.

Regards,
Qasim
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Re: [OpenSIPS-Users] OpenSIPs Radius Accounting.

[qasimak...@gmail.com]

Hi,

Thanks Bogdan for your reply.

Now for my question, I want to keep my STOP event on reply as i have faced
issues when generating event on request time. The thing is how should i
cater the fact that the device has gone offline and there is no response
generated and hence no accounting STOP event.

Regards,
Qasim


On Tue, Apr 30, 2013 at 2:26 PM, Bogdan-Andrei Iancu wrote:

> **
> Hello,
>
> All accounting triggers (START/STOP or CDR based) are on replies, so when
> the transaction is completed. Of course, all transactions are terminated in
> OpenSIPS  - either by received replies, either by a timeout (if no reply
> received).
>
> If you want to generate the STOP event at BYE request time (versus BYE
> reply time), you can manually do it from script via the acc function
> acc_db_request() (instead of setting the acc flag and letting the acc
> module to generate automatically the event) - the generate event is the
> same. See:
> http://www.opensips.org/html/docs/modules/1.9.x/acc.html#id294346
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 04/30/2013 08:00 AM, qasimak...@gmail.com wrote:
>
>  I have tried this scenario. Still if the User B is behind a NAT or is
> unreachable the opensips generates the BYE with retransmitted BYE's and the
> dialog is closed but there is no response to BYE received from that user
> hence no radius acct request.
>
>  Regards,
> Qasim
>
>
> On Mon, Apr 29, 2013 at 8:36 PM, Muhammad Shahzad 
> wrote:
>
>> Per my understanding, accounting event is sent when BYE completes,
>> whether if destination replies with 200 OK or BYE re-transmission times out
>> and opensips responds with 408 Request timeout. In each case SIP response
>> code is set appropriately and you should use stop time as accounting end
>> time rather then the time your receive account stop request on radius (they
>> both may differ, e.g. under high load scenarios).
>>
>>  Thank you.
>>
>>
>>
>>  On Mon, Apr 29, 2013 at 3:27 PM, qasimak...@gmail.com <
>> qasimak...@gmail.com> wrote:
>>
>>>Hi,
>>>
>>>  I wanted to confirm if radius accounting requests are generated on a
>>> successful transaction or it can be generated on a received BYE only. To
>>> elaborate my question you can look at 2 diagrams below. Is first scenario
>>> correct based on RFC's? My next question is that if scenario A is correct
>>> then how can we account the call if say user B has gone offline and we do
>>> not receive 200 OK of the BYE sent?
>>>
>>> Can we send a manual accounting request to Radius with acc_aaa_request
>>> in accounting module?
>>>
>>>  *Scenario A:*
>>>  User AOpenSIPsRadius   User B
>>>
>>> |---BYE--->|  |
>>> |
>>> |-BYE>|
>>>  |   |---acct-BYE--->|
>>>
>>> *Scenario B:*
>>> User AOpenSIPsRadius   User B
>>> |---BYE--->|
>>> |   |
>>> |
>>> |-BYE>|
>>>  |   |<---200 OK
>>> -|
>>>  |<200 OK -|
>>> |   |---acct-BYE--->|
>>>
>>>
>>>  Regards,
>>> Qasim Ayyaz Khan
>>>
>>>  ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
>>  --
>> Mit freundlichen Grüßen
>> Muhammad Shahzad
>> ---
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +49 176 99 83 10 85 <%2B49%20176%2099%2083%2010%2085>
>> MSN: shari_78...@hotmail.com
>> Email: shaherya...@googlemail.com
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
> ___
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> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] OpenSIPs Radius Accounting.

[qasimak...@gmail.com]

I have tried this scenario. Still if the User B is behind a NAT or is
unreachable the opensips generates the BYE with retransmitted BYE's and the
dialog is closed but there is no response to BYE received from that user
hence no radius acct request.

Regards,
Qasim


On Mon, Apr 29, 2013 at 8:36 PM, Muhammad Shahzad wrote:

> Per my understanding, accounting event is sent when BYE completes, whether
> if destination replies with 200 OK or BYE re-transmission times out and
> opensips responds with 408 Request timeout. In each case SIP response code
> is set appropriately and you should use stop time as accounting end time
> rather then the time your receive account stop request on radius (they both
> may differ, e.g. under high load scenarios).
>
> Thank you.
>
>
>
> On Mon, Apr 29, 2013 at 3:27 PM, qasimak...@gmail.com <
> qasimak...@gmail.com> wrote:
>
>> Hi,
>>
>> I wanted to confirm if radius accounting requests are generated on a
>> successful transaction or it can be generated on a received BYE only. To
>> elaborate my question you can look at 2 diagrams below. Is first scenario
>> correct based on RFC's? My next question is that if scenario A is correct
>> then how can we account the call if say user B has gone offline and we do
>> not receive 200 OK of the BYE sent?
>>
>> Can we send a manual accounting request to Radius with acc_aaa_request in
>> accounting module?
>>
>> *Scenario A:*
>> User AOpenSIPsRadius   User B
>> |---BYE--->|
>> |
>> |
>> |-BYE>|
>> |   |---acct-BYE--->|
>>
>> *Scenario B:*
>> User AOpenSIPsRadius   User B
>> |---BYE--->|  |
>> |
>> |
>> |-BYE>|
>> |   |<---200 OK -|
>> |<200 OK -|
>> |   |---acct-BYE--->|
>>
>>
>> Regards,
>> Qasim Ayyaz Khan
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
> Mit freundlichen Grüßen
> Muhammad Shahzad
> ---
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_78...@hotmail.com
> Email: shaherya...@googlemail.com
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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[OpenSIPS-Users] OpenSIPs Radius Accounting.

[qasimak...@gmail.com]

Hi,

I wanted to confirm if radius accounting requests are generated on a
successful transaction or it can be generated on a received BYE only. To
elaborate my question you can look at 2 diagrams below. Is first scenario
correct based on RFC's? My next question is that if scenario A is correct
then how can we account the call if say user B has gone offline and we do
not receive 200 OK of the BYE sent?

Can we send a manual accounting request to Radius with acc_aaa_request in
accounting module?

*Scenario A:*
User AOpenSIPsRadius   User B
|---BYE--->|  |
|   |-BYE>|
|   |---acct-BYE--->|

*Scenario B:*
User AOpenSIPsRadius   User B
|---BYE--->|  |   |
|   |-BYE>|
|   |<---200 OK -|
|<200 OK -|
|   |---acct-BYE--->|


Regards,
Qasim Ayyaz Khan
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Re: [OpenSIPS-Users] Need to transcoding

[qasimak...@gmail.com]

Just forward your call to any Media Server capable of transcoding and let
it forward the call to destination. You can use Asterisk or Freeswitch.
This should be a simple scenerio.

Regards,
Qasim


On Thu, Apr 25, 2013 at 1:49 AM, Dragomir Haralambiev wrote:

> I use follow scheme:
> ClientA >> Opensips  Asterisk ( for trancoding)
> ClientB << Opensisp 
>
>
>
>
> 2013/4/24 pa...@eremina.net 
>
>> I need to transcoding inbound traffic from one GW. I know that i can use
>> that scheme: ClientA >> Operator_729 >> sip_opensips_sip >>
>> 729_Asterisk_711 >> 711_ClientB.
>>
>> But in perfect i want to use something like rtpproxy with opensips and
>> scheme will be that:
>>
>> ClientA >> Operator_729 >> 729_opensips+rtpproxy_711 >> ClientB.
>>
>> Can you help me? How can i do it? What software i can use?
>>
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
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Re: [OpenSIPS-Users] acc module flags.

[qasimak...@gmail.com]

Yes i replaced Flag with Named Flag and its working fine now.

Thanks for the Help :)

-Qasim


On Sun, Apr 21, 2013 at 1:23 AM, Liviu Chircu  wrote:

>  Hello Qasim,
>
> The named flags are 100% compatible with the integer flags in pre 1.9
> OpenSIPS. You should only change your script to use named flags if you want
> to get rid of the acc module deprecation warnings in the startup phase.
>
> Regarding your issue, I have two quick requests:
> - please confirm you receive WARNING deprecation messages from the acc
> module (at least) when restarting OpenSIPS (to confirm the 1.9 modules were
> built and replaced in your system)
> - if possible, are you able to confirm that the aaa flags are not properly
> set? (with "isflagset")
>
> Best regards,
>
> Liviu Chircu
> OpenSIPS devhttp://www.opensips-solutions.com
>
> On 04/20/2013 12:37 PM, qasimak...@gmail.com wrote:
>
>  Hi,
>
>  Just wondering how to use new string flags in acc module in opensips
> version 1.9. My script works fine on opensips 1.8 but in 1.9 i dont get
> accounting packet in radius.
>
>  Regards,
> Qasim
>
>
> ___
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> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
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>
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Re: [OpenSIPS-Users] radius keep alive (Accounting Interim Update)

[qasimak...@gmail.com]

You can turn on in dialog ping using pP flag i.e. "create_dialog("Pp");"
and send acct packet to radius on its reply.

-Qasim


On Sun, Apr 21, 2013 at 9:18 AM, Ewgeny  wrote:

> Hi!
>
> I use Opensips 1.9 with RADIUS accounting functions - "radius_send_auth"
> and  "radius_send_acct" (AAA RADIUS MODULE).
> Now I want to make a more reliable RADIUS accounting by sending
> periodically RADIUS packets - Interim Update (or RADIUS keep-alive).
> Ie Keep-alive packet to be sent every 10 seconds, provided that the active
> SIP dialog.
> Tell me how to do it?
> In which route or what function it is possible to carry out periodical
> actions?
>
> __**_
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-**bin/mailman/listinfo/users
>
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[OpenSIPS-Users] acc module flags.

[qasimak...@gmail.com]

Hi,

Just wondering how to use new string flags in acc module in opensips
version 1.9. My script works fine on opensips 1.8 but in 1.9 i dont get
accounting packet in radius.

Regards,
Qasim
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Re: [OpenSIPS-Users] Opensips and Asterisk

[qasimak...@gmail.com]

If you are using Asterisk then you dont need media proxy as asterisk can
handle NAT and Media issues. You just need to forward SIP messages to
asterisk. Use OpenSIPs in LoadBalancer/Dispatcher scenerio.

Regards,
Qasim

On Thu, Mar 28, 2013 at 12:07 AM, Jagadish Thoutam
wrote:

> HI All,
>
>   I am New Here, i am getting Confusion while i am useing Openisps with My
> asterisk Cluster My Implimentation Plan is Like this
>
>
> (NAT)Opensips
> 1---  |
> Asterisk1
>  Inbound & Outbound
> |  |   Asterisk2
>
>  |--|   Asterisk3
> Povider --->
>   |  |   Asterisk4
> (NAT)Opensips 2
> -  |
> Asterisk5
>  Inbound & Outbound(Fail Over)
>
>
>
> Can anyone suggest how can i go with will i need the rtpproxy for Media or
> i can just bypass the sip Messages
>
> Thanks in Advance
>
>
> Jagadish
>
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Re: [OpenSIPS-Users] open sips 1.9 issue

[qasimak...@gmail.com]

Also since you copied script from 1.8 you should consider going through
this document.

http://www.opensips.org/Resources/DocsMigration180to190

-Qasim

On Mon, Mar 18, 2013 at 2:34 PM, qasimak...@gmail.com
wrote:

> Set debug level to 6 and then send the logs again.
>
> -Qasim
>
>
> On Mon, Mar 18, 2013 at 2:14 PM, Peter Zoltan Keresztes <
> zozo6...@gmail.com> wrote:
>
>> Yes, the table/DB name is correct. the same as in config.h in the sources.
>> #define VERSION_TABLE "version" /*!< Table holding
>> versions of other opensips tables */
>> #define VERSION_COLUMN"table_version"   /*!< Column name for the
>> version value in version table */
>> #define TABLENAME_COLUMN  "table_name"  /*!< Column name of the
>> table name column in the version table */
>>
>> mysql> describe version;
>> +---+--+--+-+-+---+
>> | Field | Type | Null | Key | Default | Extra |
>> +---+--+--+-+-+---+
>> | table_name| char(32) | NO   | PRI | NULL|   |
>> | table_version | int(10) unsigned | NO   | | 0   |   |
>> +---+--+--+-+-+---+
>> 2 rows in set (0.00 sec)
>>
>>
>> On 18 Mar 2013, at 08:08, "qasimak...@gmail.com" 
>> wrote:
>>
>> Have you confirmed that table/DB name is correct? You can verify your
>> version table from config.h in your sources.
>>
>> -Qasim
>>
>> On Sun, Mar 17, 2013 at 10:32 PM, Peter Zoltan Keresztes <
>> zozo6...@gmail.com> wrote:
>>
>>> Hello,
>>>
>>> I have installed an opensips 1.9
>>> I have the configuration copied from an 1.8 setup created a brand new
>>> database  when I start it I am getting the following errors:
>>> Mar 17 09:56:35 freya /usr/local/sbin/opensips[434]:
>>> ERROR:core:db_check_table_version: invalid version 0 for table dialog
>>> found, expected 8
>>> Mar 17 09:56:35 freya /usr/local/sbin/opensips[434]:
>>> ERROR:dialog:init_dlg_db: error during table version check.
>>> Mar 17 09:56:35 freya /usr/local/sbin/opensips[434]:
>>> ERROR:dialog:mod_init: failed to initialize the DB support
>>> Mar 17 09:56:35 freya /usr/local/sbin/opensips[434]:
>>> ERROR:core:init_mod: failed to initialize module dialog
>>> Mar 17 09:56:35 freya /usr/local/sbin/opensips[434]: ERROR:core:main:
>>> error while initializing modules
>>>
>>> The error is telling about invalid version of the dialog table however
>>> if I check the version table the dialog has the value 8 as it supposed to
>>> have it.
>>>
>>> Thanks
>>> Peter
>>> ___
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>>>
>>
>> ___
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>>
>>
>>
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Re: [OpenSIPS-Users] open sips 1.9 issue

[qasimak...@gmail.com]

Have you confirmed that table/DB name is correct? You can verify your
version table from config.h in your sources.

-Qasim

On Sun, Mar 17, 2013 at 10:32 PM, Peter Zoltan Keresztes  wrote:

> Hello,
>
> I have installed an opensips 1.9
> I have the configuration copied from an 1.8 setup created a brand new
> database  when I start it I am getting the following errors:
> Mar 17 09:56:35 freya /usr/local/sbin/opensips[434]:
> ERROR:core:db_check_table_version: invalid version 0 for table dialog
> found, expected 8
> Mar 17 09:56:35 freya /usr/local/sbin/opensips[434]:
> ERROR:dialog:init_dlg_db: error during table version check.
> Mar 17 09:56:35 freya /usr/local/sbin/opensips[434]:
> ERROR:dialog:mod_init: failed to initialize the DB support
> Mar 17 09:56:35 freya /usr/local/sbin/opensips[434]: ERROR:core:init_mod:
> failed to initialize module dialog
> Mar 17 09:56:35 freya /usr/local/sbin/opensips[434]: ERROR:core:main:
> error while initializing modules
>
> The error is telling about invalid version of the dialog table however if
> I check the version table the dialog has the value 8 as it supposed to have
> it.
>
> Thanks
> Peter
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Re: [OpenSIPS-Users] Opensips Crash - Dialog Ping dlg_ping_routine

[qasimak...@gmail.com]

Thanks :). I will update my opensips and update you.

Regards,
Qasim

On Wed, Feb 13, 2013 at 2:18 PM, Bogdan-Andrei Iancu wrote:

> **
> Hi Qasim,
>
> Current revision on 1.8 branch is 9787, so you have an older version. I
> remember some recent fixes in the dlg pinging, so I would advice you to
> update from SVN (1.8 branch) and give it another try.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 02/13/2013 11:04 AM, qasimak...@gmail.com wrote:
>
> Hi,
>
> Sorry for the spamming but my opensips version is 1.8 SVN Rev 9447.
>
> Regards,
> Qasim
>
>  On Wed, Feb 13, 2013 at 2:00 PM, qasimak...@gmail.com <
> qasimak...@gmail.com> wrote:
>
>> Hi,
>>
>> My opensips configuration was running fine until I enabled dialog ping
>> flag in create_dialog. Now after enabling ping my opensips crashes randomly
>> 3-4 times daily. I have collected opensips logs which are as follows:
>>
>> P.S: Please let me know if anything else is needed. I am preserving my
>> core dumps and my opensips logs.
>>
>> *Syslog:*
>> **
>> opensips[3251]: segfault at 19d7c ip 7f6f2ddd2df6 sp
>> 7fffacd946b0 error 4 in dialog.so[7f6f2dd9b000+4b000]
>>
>> *CoreDump:*
>> **
>> Core was generated by `/usr/local/sbin/opensips -P
>> /var/run/opensips/opensips.pid -m 512 -M 4 -u root'.
>> Program terminated with signal 11, Segmentation fault.
>> #0  dlg_ping_routine (ticks=, attr=> out>) at dlg_timer.c:525
>> 525 dlg->pl = 0;
>> (gdb) bt
>> #0  dlg_ping_routine (ticks=, attr=> out>) at dlg_timer.c:525
>> #1  0x004b5956 in timer_ticker () at timer.c:360
>> #2  run_timer_process () at timer.c:404
>> #3  start_timer_processes () at timer.c:527
>> #4  0x00432c60 in main_loop (argc=,
>> argv=) at main.c:945
>> #5  main (argc=, argv=) at
>> main.c:1541
>> (gdb) bt full
>> #0  dlg_ping_routine (ticks=, attr=> out>) at dlg_timer.c:525
>> expired = 
>> it = 
>> curr = 0x7f0f15926a20
>> dlg = 0x10
>> __FUNCTION__ = "dlg_ping_routine"
>> #1  0x004b5956 in timer_ticker () at timer.c:360
>> t = 0x7f0f3b586fc8
>> #2  run_timer_process () at timer.c:404
>> multiple = 1
>> cnt = 
>> tv = {tv_sec = 0, tv_usec = 0}
>> #3  start_timer_processes () at timer.c:527
>> tpl = 0x7f0f3b586568
>> pid = 
>> __FUNCTION__ = "start_timer_processes"
>> #4  0x00432c60 in main_loop (argc=,
>> argv=) at main.c:945
>> i = 4
>> pid = 24
>> si = 0x0
>> startup_done = 0x0
>> chd_rank = 8
>> load_p = 0x7f0f15840628
>> #5  main (argc=, argv=) at
>> main.c:1541
>> cfg_log_stderr = 0
>> cfg_stream = 
>> c = 
>> r = 
>> tmp = 0x7fff330839d5 ""
>> tmp_len = 
>> port = 
>> proto = 
>> ret = 
>> seed = 2204336599
>> rfd = 
>> __FUNCTION__ = "main"
>>
>> Regards,
>> Qasim Ayyaz
>>
>
>
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>
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Re: [OpenSIPS-Users] Opensips Crash - Dialog Ping dlg_ping_routine

[qasimak...@gmail.com]

Hi,

Sorry for the spamming but my opensips version is 1.8 SVN Rev 9447.

Regards,
Qasim

On Wed, Feb 13, 2013 at 2:00 PM, qasimak...@gmail.com
wrote:

> Hi,
>
> My opensips configuration was running fine until I enabled dialog ping
> flag in create_dialog. Now after enabling ping my opensips crashes randomly
> 3-4 times daily. I have collected opensips logs which are as follows:
>
> P.S: Please let me know if anything else is needed. I am preserving my
> core dumps and my opensips logs.
>
> *Syslog:*
> **
> opensips[3251]: segfault at 19d7c ip 7f6f2ddd2df6 sp
> 7fffacd946b0 error 4 in dialog.so[7f6f2dd9b000+4b000]
>
> *CoreDump:*
> **
> Core was generated by `/usr/local/sbin/opensips -P
> /var/run/opensips/opensips.pid -m 512 -M 4 -u root'.
> Program terminated with signal 11, Segmentation fault.
> #0  dlg_ping_routine (ticks=, attr= out>) at dlg_timer.c:525
> 525 dlg->pl = 0;
> (gdb) bt
> #0  dlg_ping_routine (ticks=, attr= out>) at dlg_timer.c:525
> #1  0x004b5956 in timer_ticker () at timer.c:360
> #2  run_timer_process () at timer.c:404
> #3  start_timer_processes () at timer.c:527
> #4  0x00432c60 in main_loop (argc=,
> argv=) at main.c:945
> #5  main (argc=, argv=) at
> main.c:1541
> (gdb) bt full
> #0  dlg_ping_routine (ticks=, attr= out>) at dlg_timer.c:525
> expired = 
> it = 
> curr = 0x7f0f15926a20
> dlg = 0x10
> __FUNCTION__ = "dlg_ping_routine"
> #1  0x004b5956 in timer_ticker () at timer.c:360
> t = 0x7f0f3b586fc8
> #2  run_timer_process () at timer.c:404
> multiple = 1
> cnt = 
> tv = {tv_sec = 0, tv_usec = 0}
> #3  start_timer_processes () at timer.c:527
> tpl = 0x7f0f3b586568
> pid = 
> __FUNCTION__ = "start_timer_processes"
> #4  0x00432c60 in main_loop (argc=,
> argv=) at main.c:945
> i = 4
> pid = 24
> si = 0x0
> startup_done = 0x0
> chd_rank = 8
> load_p = 0x7f0f15840628
> #5  main (argc=, argv=) at
> main.c:1541
> cfg_log_stderr = 0
> cfg_stream = 
> c = 
> r = 
> tmp = 0x7fff330839d5 ""
> tmp_len = 
> port = 
> proto = 
> ret = 
> seed = 2204336599
> rfd = 
> __FUNCTION__ = "main"
>
> Regards,
> Qasim Ayyaz
>
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[OpenSIPS-Users] Opensips Crash - Dialog Ping dlg_ping_routine

[qasimak...@gmail.com]

Hi,

My opensips configuration was running fine until I enabled dialog ping flag
in create_dialog. Now after enabling ping my opensips crashes randomly 3-4
times daily. I have collected opensips logs which are as follows:

P.S: Please let me know if anything else is needed. I am preserving my core
dumps and my opensips logs.

*Syslog:*
**
opensips[3251]: segfault at 19d7c ip 7f6f2ddd2df6 sp
7fffacd946b0 error 4 in dialog.so[7f6f2dd9b000+4b000]

*CoreDump:*
**
Core was generated by `/usr/local/sbin/opensips -P
/var/run/opensips/opensips.pid -m 512 -M 4 -u root'.
Program terminated with signal 11, Segmentation fault.
#0  dlg_ping_routine (ticks=, attr=) at dlg_timer.c:525
525 dlg->pl = 0;
(gdb) bt
#0  dlg_ping_routine (ticks=, attr=) at dlg_timer.c:525
#1  0x004b5956 in timer_ticker () at timer.c:360
#2  run_timer_process () at timer.c:404
#3  start_timer_processes () at timer.c:527
#4  0x00432c60 in main_loop (argc=,
argv=) at main.c:945
#5  main (argc=, argv=) at
main.c:1541
(gdb) bt full
#0  dlg_ping_routine (ticks=, attr=) at dlg_timer.c:525
expired = 
it = 
curr = 0x7f0f15926a20
dlg = 0x10
__FUNCTION__ = "dlg_ping_routine"
#1  0x004b5956 in timer_ticker () at timer.c:360
t = 0x7f0f3b586fc8
#2  run_timer_process () at timer.c:404
multiple = 1
cnt = 
tv = {tv_sec = 0, tv_usec = 0}
#3  start_timer_processes () at timer.c:527
tpl = 0x7f0f3b586568
pid = 
__FUNCTION__ = "start_timer_processes"
#4  0x00432c60 in main_loop (argc=,
argv=) at main.c:945
i = 4
pid = 24
si = 0x0
startup_done = 0x0
chd_rank = 8
load_p = 0x7f0f15840628
#5  main (argc=, argv=) at
main.c:1541
cfg_log_stderr = 0
cfg_stream = 
c = 
r = 
tmp = 0x7fff330839d5 ""
tmp_len = 
port = 
proto = 
ret = 
seed = 2204336599
rfd = 
__FUNCTION__ = "main"

Regards,
Qasim Ayyaz
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Re: [OpenSIPS-Users] Bulding Own Distro

[qasimak...@gmail.com]

I would start from here...

https://help.ubuntu.com/community/LiveCDCustomization

Regards,
Qasim

On Thu, Dec 27, 2012 at 9:13 PM, M.Khaled W Chehab wrote:

> -How can I make an iso image in order to install this distro in other
> servers to avoid working from scratch ?
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Re: [OpenSIPS-Users] Recover running configuration of OpenSIPs

[qasimak...@gmail.com]

I think a better way than adding copy command to init script would be to
use incron utility.
This utility triggers cron jobs based on file system triggers so
you can backup a file when it is changed. You can do SVN/Git commit or even
do an rsync so whenever you change the file it is automatically backed up.

Regards,
Qasim


On Tue, Nov 6, 2012 at 10:12 PM, Ovidiu Sas  wrote:

> Just use git to track down your changes.
>
> On Tue, Nov 6, 2012 at 11:25 AM, Muhammad Shahzad
>  wrote:
> > No, file was not removed, but only changed with default opensips
> > configuration file that comes with its source code, so i guess any file
> > system utility won't help, specially when using EXT3 FS.
> >
> > To avoid such situation in future i suggest to create backup
> configuration
> > file from start up script i.e. /etc/init.d/opensips, e.g. whenever we run
> > "/etc/init.d/opensips start", it should make a copy of opensips.cfg
> > somewhere safe like,
> >
> > cp -arvf /usr/local/opensips/opensips.cfg
> > /var/run/opensips/opensips.cfg-
> >
> > This backup is good not only for recovery but also to see what went to
> > production when..
> >
> > But for current situation, it seems i am helpless.
> >
> > Thank you.
> >
> >
> > On Tue, Nov 6, 2012 at 4:22 PM, alexandre Moutot 
> > wrote:
> >>
> >> Hi,
> >>
> >> Have you try to recover your file from your FS ?
> >>
> >> Regards,
> >>
> >> M. A.
> >>
> >> - Original Message -
> >> > From: "Muhammad Shahzad" 
> >> > To: "OpenSIPS users mailling list" 
> >> > Sent: Tuesday, November 6, 2012 3:57:33 PM
> >> > Subject: [OpenSIPS-Users] Recover running configuration of OpenSIPs
> >> > Hi,
> >> >
> >> >
> >> > Is there anyway to get currently loaded opensips.cfg in a running
> >> > instance of opensips? i have accidently lost the opensips
> >> > configuration of a running production server? The backup of
> >> > configuration file is bit old and does not contains most recent
> >> > changes so i am worried next time we restart opensips we may loose
> >> > some important changes in configuration file.
> >> >
> >> >
> >> >
> >> > I know its a bit stupid / awkward situation but any help will be
> >> > highly appreciated.
> >> >
> >> >
> >> > Thank you.
> >> >
> >> >
> >> >
>
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Re: [OpenSIPS-Users] Help: Understanding ACK loop

[qasimak...@gmail.com]

Try replacing your ACK block:

t_relay();
exit;

with:

if (src_ip == 10.9.6.40) {
route(1)
}

if (src_ip == 10.9.6.3) {
route(2)
}

Regards,
Qasim
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Re: [OpenSIPS-Users] Help to Understand Loop

[qasimak...@gmail.com]

Can you share your Config and SIP Dump? I don't seems to have it in this
email chain.


On Tue, Nov 6, 2012 at 3:05 PM, spady  wrote:

> Hi, can someone help me understand this issue?
>
> Thanks
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/Help-to-Understand-Loop-tp7582655p7582785.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS 1.9.0 major release

[qasimak...@gmail.com]

VIA Parser Patch for WS & WSS:

http://sourceforge.net/tracker/?func=detail&aid=3545859&group_id=232389&atid=1086412

Regards,
Qasim


On Fri, Nov 2, 2012 at 6:43 AM, Binan AL Halabi wrote:

>
> Hi All,
>
> If oversip uses Path extension OpenSIPS must support it.
>
> 1- Sending Path header values in 200 ok REGISTER response
> 2- Path header files syntax must confom to Route syntax
> 3- When look up the opensips must copies the stored path header fileds
> into Route header fileds - preloaded route.
> Reference: RFC 3327
>
>
> Here i think adding one Route header pointing to OverSIP (second Path URI)
> is enough in simple case (UA  OverSIP--- OpenSIPS).
> OverSIP removes the Route header and route the request based on RURI
> (first Path URI).
>
>
>
> OpenSIPs support this
> http://www.opensips.org/html/docs/modules/devel/registrar.html#id248705
> flag "px" (path support) of save function in Registrar module.
>
> So still parsing VIA header is required to add WS and WSS. I think this
> patch is already posted.
>
>
> Binan.
>
>   --
> *Från:* Iñaki Baz Castillo 
> *Till:* Bogdan-Andrei Iancu 
> *Kopia:* OpenSIPS users mailling list 
> *Skickat:* torsdag, 1 november 2012 20:36
> *Ämne:* Re: [OpenSIPS-Users] [OpenSIPS-Devel] [RELEASES] Planing OpenSIPS
> 1.9.0 major release
>
> 2012/11/1 Bogdan-Andrei Iancu 
> >
> > Hi Inaki,
> >
> > Please correct me if I'm wrong, but reading the draft and listing you
> guys all, I would say the right approach is to : (1) use OverSIP as gw (to
> extract SIP traffic from WebSocket) and (2) make OpenSIPS to support SIP
> traffic resulted from websocket extraction.
> >
> > If so, OpenSIPS has nothing to do with the WebSocket protocol itself,
> but only to support the extensions from the draft (like new protocols and
> eventually the SIP server location).
>
> Right. As Saul pointed out, this scenario (which is a pure RFC 5626
> "Outbound" scenario with a Edge Proxy in front of the
> registrar/authentication-proxy) requires:
>
> - Path support in OpenSIPS for storing the Path URI(s). Note: It's
> important to increase the "path" column size in the location table.
> The current value is to small and cannot store two URI's (OverSIP adds
> double Path headers).
>
> - OpenSIPS should improve the parser of the Via transport field since
> currently it only accepts UDP, TCP, TLS and SCTP. It should also
> accept WS and WSS, but better if it accepts any token (as the RFC 3261
> BNF grammar states). Otherwise OpenSIPS will discard SIP requests
> coming from OverSIP (since the non top Via header, that created by the
> SIP WebSocket client, has "WS" as transport protocol).
>
> And nothing else at all, but the above two points are important.
>
> Regards.
>
>
>
>
>
> --
> Iñaki Baz Castillo
> 
>
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>
>
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Re: [OpenSIPS-Users] Transfering a call by opensips

[qasimak...@gmail.com]

I haven’t tried doing this before but if i am not wrong you can write a
script in perl using opensips perl module.

Regards,
Qasim

On Thu, Oct 18, 2012 at 11:53 AM, Engineer Voip  wrote:

> Hello all,
> I want to transfert the call to user C when user A calls user B in
> interval of time for example: 11h-14h
> I can do that by asterisk but i prefer to do it by opensips
> It's possible to do that by opensips?
>
> Cordialement.
> Envoyé de mon iPhone
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Re: [OpenSIPS-Users] attack from friendly-scanner

[qasimak...@gmail.com]

Advise: Read threads initiated by you thoroughly.

Read: http://blog.sipvicious.org/ to know more about the tool we all face
> every once a while.
>

Regards,
Qasim

On Tue, Oct 9, 2012 at 2:27 PM, Engineer voip  wrote:

> Hi All,
> thank you for your reply, Know i want to simulate an attacker to test if
> my fail2ban and pike module  works good.
> someone has an idea to do that?
>
> 2012/10/9 SamyGo 
>
>> Hi,
>>
>> Very nice suggestions by Brett. I remember there are regular thread like
>> these on the mailing lists and people share a lot of experiences. AFAIR
>> there was some service which contains the IP addresses of known attackers
>> available for users. OP needs to do some searching in this regard to
>> collect more ideas.
>>
>> * Nothing is _NOT_ CPU cycles free *
>>
>> I'm not sure about sip vicious but if I were to detect and hack a SIP
>> server I'd first start by sending OPTIONS on its ports. Mostly that's where
>> things kick off. Changing the user-agent field is nothing big, so question
>> is how do you know a hacker is about to get angry !!
>>
>> I'd say it needs a time populated repository and a well crafted shell
>> script to maintain the list of Hacker IPs captured in the past and use it
>> across all the servers or devices. Let me explain the idea.
>>
>> * ii)* - For any incoming packets one needs to look-up the hacker's
>> listing and detect if a known hacker or not.
>>  *i)* - Take fail2ban for example, or pike module , or iptables rate
>> limit mechanism to initially detect a new born hacker trying to access your
>> sip server (yes will take few minutes to finally conclude that a particular
>> source IP is hacker) - Store that IP in your hacker's listing.
>> *iii)* - Use an intelligent script to share the detected hacker's IP
>> across all the other SIP servers and router devices/firewall to block the
>> traffic at network layer.
>>
>> *Critical Exceptions:*
>> Always ensure that the IP which is going to get blocked across the whole
>> network perimeter is not your own server or within the same subnet as
>> your's. It shouldn't be localhost as well.(Hint: IP spoofing)
>>
>> *Focus on Security rather Friendly-scanner:*
>> *
>> *
>> One need to secure each and everything when it comes to security, just
>> one layer security  i.e fail2ban or iptables or pike module is never
>> enough. Like Brett said you can drop packets once detected a "very friendly
>> scanner", how about a customer who wants to toy with your service ! how
>> about a massive DoS attack !! drop() won't help alone. iptables needs to be
>> there to stop the packets from even reaching the SIP server app, then again
>> why should the server's NIC be chocked up by that massive DoS ! your
>> firewall or networking device should stop the packets from entering the
>> network !
>>
>> This is just not enough: How about a different unique new tool which
>> sends malicious or malformed SIP packets to crash the server !! its just
>> one packet but malformed -- all the above measures WILL fail !! Obviously
>> needs to go one step ahead and use SNORT or anything like IDS+IPS to verify
>> that the packet going through the network is not malformed.
>>
>> Thats pretty much it for now. There are things which I've forgotten to
>> write at the moment OR might not even know which I expect some one else may
>> like to add.
>>
>> Networks and Data Security is a huge field, and VoIP security alone has
>> hundreds of book on the topic.
>>
>> *Interesting threads to read: *
>> *
>> *
>> http://lists.opensips.org/pipermail/users/2010-November/015243.html
>> http://lists.opensips.org/pipermail/users/2011-June/018271.html
>> Read: http://blog.sipvicious.org/ to know more about the tool we all
>> face every once a while.
>> Fail2ban for openSIPS ::
>> http://www.opensips.org/Resources/DocsTutFail2ban
>>
>>
>> --
>> Best Regards
>> Sammy
>>
>>
>>
>>
>>
>>
>>
>>
>> On Mon, Oct 8, 2012 at 6:31 PM, Brett Nemeroff wrote:
>>
>>> First of all,
>>> This is an attack from sipvicious. It is an *attack*. It will be very
>>> high rate (cps) and you do *not* want to use anything that consumes
>>> resources to attempt to block it.
>>>
>>> First recommendation is to use iptables. In addition, you *should* put a
>>> check in your config for friendly-scanner and drop() the packet. Do not
>>> reply with a sip code. You want to be invisible to the attacker. If you
>>> reply with a sip code, they'll just scan you attempting to find a request
>>> combination that will return a usable result.
>>>
>>> 1. Do whatever you can to not use CPU resources to block this
>>> 2. Don't look like a SIP server to source IPs you do not recognize
>>>
>>> I guarantee, if you look like a SIP server, you will get brutally
>>> attacked from unsolicited sources.
>>>
>>> Read up on the fail2ban docs for asterisk. They have some good ideas in
>>> there on how to perform intrusion detection and how to automatically add
>>> offending traffic to fail2ban. You can do something similar in OpenSIPs.

Re: [OpenSIPS-Users] Functioning 1.7 + RTP Proxy Configuration

[qasimak...@gmail.com]

I dunno what you are trying to explain but a simple scenario for topology
hiding would look something liek this:

UA   OSIPs/RTPP *  CARRIER
<--<>--<>-->
<<><>##>

In the above scenario your OpenSIPs and RTPProxy are both running on the
same server so both your SIP signaling and RTP stream will be relayed to
OpenSIPs/RTP box. Now your OSIPs will relay these requests to your * server
and similarly RTPProxy will relay your RTP stream to the * box. Now if you
look on UA(User Agent's) side only one node will be visible to it. To
furthur secure your * box you can remove it from public domain and assign
it a local IP. Possibilities are endless :).

Lastly i am sure i explained every thing right. Please correct me if i am
wrong anywhere.

Regards,
Qasim


On Fri, Aug 31, 2012 at 6:20 PM, Nick Khamis  wrote:

> >> RTP is used for many purposes most commonly to hide your internal
> network topology i.e. in your case your * server also it helps in NAT
> traversal.
>
> That is exactly what we are trying to accomplish. Would like all SIP
> traffic to go through the OpenSIPS server,
>
>
> Could someone kindly illustrate graphically how the flow of traffic is
> suppose to look in a topology hidden network using NAT traversal
> please? Should there be port forwarding
> of RTP traffic to the Proxy? Meaning, should the incoming ITSP traffic
> be directed at OpenSIPS:
>
>
> ->: Initial Stream
> *> Asterisk Stream
>
> What I think we have now, and why we don't need RTP NAT traversal yet:
>
> Scenario 1
>
> UC > Proxy > Asterisk > ITSP > * Router 192.168.2.1  * > Back to
> Asterisk > Back to Proxy > Back to UC (Final)
>
>   Proxy--->Asterisk>  ITSP
> UC  ---> *   *  *   *
>   <** *  ** Router***
>
>
> What we should have and need eventually (Topology Hiding using NAT
> traversal):
>
> Scenario 2
>
> UC > Proxy > Asterisk > ITSP >  *Router 192.168.2.1 * > Back to
> OpenSIPS > Back to UC (Final)
>
>
>   Proxy--->Asterisk>ITSP
> UC  ---> * *
>   <** * *
>  *Router 192.168.2.1
>
>
>
> i.e., there has to be no direct connection between the UCs or ITSP and
> asterisk. I think we have accomplished this without using RTP proxy
> since everything is on the same subnet,
> or am I incorrect? And as you mentioned, the RTP stream is now being
> directed at the Asterisk server? Will get a NGREP trace posted when I
> get to work. Please remember
> that this is done using virtual machines as a test bed for now...
>
>
>
> Thank you Kindly for your Help,
>
> Nick.
>
>
> On Fri, Aug 31, 2012 at 12:11 AM, qasimak...@gmail.com
>  wrote:
> > Most probably your RTP stream is directly bein connected to * server.
> RTP is
> > used for many purposes most commonly to hide your internal network
> topology
> > i.e. in your case your * server also it helps in NAT traversal.
> >
> > Regards,
> > Qasim
> >
> > On Fri, Aug 31, 2012 at 8:05 AM, Nick Khamis  wrote:
> >>
> >> Thank you guys for your response. It's the strangest thing. I removed
> >> all the rtpproxy and nathelper stuff, and it works perfectly?
> >> Everything is on the same subnet, i.e.:
> >>
> >> Router
> >> |
> >> - MySQL 192.168.2.105 (Virtual Box)
> >> |
> >> - OpenSIPS 192.168.2.102 (Virtual Box)
> >> |
> >> -Asterisk 192.168.2.110 (Virtual Box)
> >> |
> >> - Polycom 192.168.2.102 (Virtual Box)
> >>
> >>
> >> The flow of transmition is:
> >>
> >> UC--->Proxy--->Asterisk
> >>
> >>
> >> And I have two way audio without rtp proxy or port forwarding. I'm now
> >> scared and confused. Can someone please
> >> explain to me how this is?
> >>
> >> Thanks in Advnace,
> >>
> >> Nick.
> >>
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> >
> >
> >
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Re: [OpenSIPS-Users] Functioning 1.7 + RTP Proxy Configuration

[qasimak...@gmail.com]

Most probably your RTP stream is directly bein connected to * server. RTP
is used for many purposes most commonly to hide your internal network
topology i.e. in your case your * server also it helps in NAT traversal.

Regards,
Qasim

On Fri, Aug 31, 2012 at 8:05 AM, Nick Khamis  wrote:

> Thank you guys for your response. It's the strangest thing. I removed
> all the rtpproxy and nathelper stuff, and it works perfectly?
> Everything is on the same subnet, i.e.:
>
> Router
> |
> - MySQL 192.168.2.105 (Virtual Box)
> |
> - OpenSIPS 192.168.2.102 (Virtual Box)
> |
> -Asterisk 192.168.2.110 (Virtual Box)
> |
> - Polycom 192.168.2.102 (Virtual Box)
>
>
> The flow of transmition is:
>
> UC--->Proxy--->Asterisk
>
>
> And I have two way audio without rtp proxy or port forwarding. I'm now
> scared and confused. Can someone please
> explain to me how this is?
>
> Thanks in Advnace,
>
> Nick.
>
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Re: [OpenSIPS-Users] Functioning 1.7 + RTP Proxy Configuration

[qasimak...@gmail.com]

My friend has this good walkthrough's for Opensips configuration and RTP.
and example is

http://saevolgo.blogspot.com/2012/03/making-rtpproxy-work.html

You can also find other posts there. Just go through them and you will be
good to go.

PS: I also learned using rtpproxy using above mentioned page.

Regards,
Qasim

On Wed, Aug 29, 2012 at 7:58 AM, Nick Khamis  wrote:

> Hello Everyone,
>
> After a week of tinkering with opensips rtpproxy functions, I have a
> quite messy config file. Was wondering if anyone would
> be kind enough to share or walkthrough a configuration that will get
> two way audio working. Presently I have single
> outgoing audio. Seems like I am not able to pick up the callee's RTP.
>
> INFO:remove_session: RTP stats: 0 in from callee, 872 in from caller,
> 872 relayed, 0 dropped
> INFO:remove_session: RTCP stats: 8 in from callee, 2 in from caller,
> 10 relayed, 0 dropped
>
> Basic layout of the network
>
> router 192.168.2.1
> opensips 192.168.2.102 (bridged virutal box, ports forwarded)
> asterisk 192.168.2.110 (bridged virutal box)
> Polycom 192.168.2.11
>
> [router]-[opensips]---[asterisk][SIP Trunk]
>
> Please bare with the virtual box setup, I am just trying to get all
> the configs together before deploying onto the servers. I know i'm
> really
> close, and would love to be able to move on to the other parts (i.e.,
> dialplan, routing etc...)
>
> I pasted an ngrep trace at http://pastebin.com/A39vBG3t.
>
> Thank you Kindly,
>
> Nick.
>
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Re: [OpenSIPS-Users] OpenSIPS 1.7 + NAT + rtpproxy

[qasimak...@gmail.com]

Have you installed and started rtpproxy? if not just scroll through this
website .

Regards,
Qasim

On Fri, Aug 3, 2012 at 2:27 AM, Ashish Kundu  wrote:

> Opensips is a great product, but I have been having problem in configuring
> the nat traversal + rtpproxy with opensips and have spent about a week on
> this.  I am a novice in this... when opensips runs with the following
> opensips.cfg relevant portions -- it raises the following rtpproxy problem:
>
> "ERROR:rtpproxy:select_rtpp_node: script error -no valid set selected"
> "ERROR:rtpproxy:force_rtp_proxy: no available proxies"
>
> # nat_traversal params -
> modparam("nat_traversal", "keepalive_interval", 30)
> modparam("nat_traversal", "keepalive_method", "OPTIONS")
> modparam("nat_traversal", "keepalive_from", "sip:keepalive@a.b.c.d")
> modparam("nat_traversal", "keepalive_state_file",
> "/var/run/opensips/keepalive_state")
>
> #ak# --- rtpproxy -
> # single rtproxy with specific weight
> modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:")
> modparam("rtpproxy", "nortpproxy_str", "a=sdpmangled:yes\r\n")
> modparam("rtpproxy", "db_url", "mysql://opensips:opensipsrw@localhost
> /opensips")
> #modparam("rtpproxy", "db_table", "nh_rtpp")
> modparam("rtpproxy", "rtpp_socket_col", "rtpproxy_sock")
>
>
> ### Routing Logic 
>
>
> # main request routing logic
>
> route{
> nat_traversal info
> force_rport();
> if (client_nat_test("7")) {
> fix_contact();
> setflag(5);
> }
>
> if ((method=="REGISTER" || method=="SUBSCRIBE" ||
> (method=="INVITE" && !has_totag())) &&
> client_nat_test("7"))
> {
> nat_keepalive();
> }
> nat_traversal info ends
>
> if (!mf_process_maxfwd_header("10")) {
> sl_send_reply("483","Too Many Hops");
> exit;
> }
>
> ##ak#
> if ((is_method("INVITE")) && has_totag()) {
> #(has_body("application/sdp"))) {
> engage_rtp_proxy();
> }
>
> if (has_totag()) {
> # sequential request withing a dialog should
> # take the path determined by record-routing
> if (loose_route()) {
> if (is_method("BYE")) {
> setflag(1); # do accounting ...
> setflag(3); # ... even if the transaction
> fails
> } else if (is_method("INVITE")) {
> # even if in most of the cases is useless,
> do RR for
> # re-INVITEs alos, as some buggy clients
> do change route set
> # during the dialog.
> record_route();
> }
> # route it out to whatever destination was set by
> loose_route()
> # in $du (destination URI).
> route(1);
> } else {
> /* uncomment the following lines if you want to
> enable presence */
> ##if (is_method("SUBSCRIBE") && $rd == "your.server.ip.address") {
> ##  # in-dialog subscribe requests
> ##  route(2);
> ##  exit;
> ##}
> if ( is_method("ACK") ) {
> if ( t_check_trans() ) {
> # non loose-route, but stateful
> ACK; must be an ACK after
> # a 487 or e.g. 404 from upstream
> server
> t_relay();
> exit;
> } else {
> # ACK without matching transaction
> ->
> # ignore and discard
> exit;
> }
> }
> sl_send_reply("404","Not here");
> }
> exit;
> }
>
> #initial requests
>
> # CANCEL processing
> if (is_method("CANCEL"))
> {
> if (t_check_trans())
> t_relay();
> #unforce_rtpproxy();
> exit;
> }
>
> t_check_trans();
>
> # preloaded route checking
> if (loose_route()) {
> xlog("L_ERR",
> "Attempt to route with preloaded Route's
> [$fu/$tu/$ru/$ci]");
> if (!is_method("ACK"))
> sl_send_reply("403","Preload Route denied");
> exit;
> }
>
> # record routing
> if (!is_method("REGISTER|MESSAGE"))
> record_route();
>
> # account o

Re: [OpenSIPS-Users] ACC module with AAA dont send failed transaction

[qasimak...@gmail.com]

Maybe this would help

"For failed SIP sessions a radius packet type FAILED is generated. A failed
SIP session is a session that has been rejected by the Proxy server or by
the destination. Unfortunately Freeradius server is not able to cope with
FAILED packets. The server must be patched and recompiled if you wish to
support accounting for failed SIP sessions."

Regards,
Qasim

On Wed, Aug 1, 2012 at 3:29 PM, SamyGo  wrote:

> Hello,
>
> I've setup OpenSIPS with RADIUS and I've to send ACC events for failed
> transactions. I'm already receiving the missed calls, early media, call
> start & stop events on my radius server.
> Now what I'm trying to accomplish is that if a transaction returns with a
> failure i.e 486 Busy etc, these  should be communictaed to radius-server
> too.
>
> For this I'm trying to use the 
> *failed_transaction_flag
> *  but I don't get any failure codes on my radius-server.
>
> I've this in my modparam:
>
> modparam("acc", "failed_transaction_flag", 8)
>
> and I'm also setting the flag setflag(8) in my code.
>
> Please help me on this one.
>
> Regards,
> Sammy
>
>
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