Re: [OpenSIPS-Users] Question regarding b2b_bridge function

[Ping Han]

Hi Bogdan,

Thank you very much for your reply.

I have tried to use the module parameter "b2bl_key_avp" as described in the
document as below.


modparam("b2b_logic", "b2bl_key_avp", "$avp(99)")


However, I got the following errors when the Opensips is restarted.

ERROR:core:set_mod_param_regex: parameter  not found in
module 


I am using the Opensips version opensips-2.1.2-1.el6.x86_64 (rpm).

Thanks,
Chris

On Tue, May 31, 2016 at 6:01 PM, Bogdan-Andrei Iancu 
wrote:

> Hi Chris,
>
> The "dialog_id" is actually the b2b key, that is expose by the b2b_logic
> via the module parameter b2bl_key_avp. See:
>
> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294094
>
> That key can also be found in the b2b_logic table in DB.
>
> At signaling level, the key is the Call-ID of the outbound calls from b2b.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 23.05.2016 07:32, Ping Han wrote:
>
> Hi Bogdan,
>
> I asked the question a few days ago but have not got a response.
>
> I am just wondering if I could get some advice from you.
>
> Any advice will be appreciated.
>
> Thanks,
> Chris
>
> On Wed, May 18, 2016 at 4:39 PM, Ping Han  wrote:
>
>> Hi,
>>
>> I would like to use the b2b_bridge fifo function as specified at
>> http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id248916.
>>
>> The function will be triggered by a third party. I will need to pass the
>> parameters to the third party for it to trigger the function. One of the
>> parameters is the "dialog-id".
>>
>> The problem is that I am not sure how the value of the dialog-id can be
>> available in the Opensips config. Is there any Opensips modules/function
>> that can retrieve the value of the dialog-id?
>>
>> I tried to get the value from the "b2b_entities" and "b2b_logic" table.
>> However, it seems that it does not work this way because the two tables do
>> not pop the data in real time. Sometimes I can see the data but sometimes I
>> am not able to see it.
>>
>> It is appreciated that you can give me some idea.
>>
>> Thanks,
>>
>> Ping
>>
>
>
>
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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] Out of memory problem

[Denis]

Hello!

I had increased PKG memory twice (to 1000 M) but the problem is still exists.

May 31 19:09:29 opensips-main /usr/local/opensips2.1/sbin/opensips[20681]: 
ERROR:core:db_allocate_rows: no memory left
May 31 19:09:29 opensips-main /usr/local/opensips2.1/sbin/opensips[20681]: 
ERROR:db_mysql:db_mysql_fetch_result: no memory left
May 31 19:09:29 opensips-main /usr/local/opensips2.1/sbin/opensips[20681]: 
ERROR:dialplan:dp_load_db: failed to fetch
May 31 19:09:29 opensips-main /usr/local/opensips2.1/sbin/opensips[20681]: 
ERROR:dialplan:dp_load_all_db: unable to load ast_dialplan table
May 31 19:09:29 opensips-main /usr/local/opensips2.1/sbin/opensips[20681]: 
ERROR:dialplan:mi_reload_rules: failed to reload database
May 31 19:09:29 opensips-main /usr/local/opensips2.1/sbin/opensips[20681]: 
ERROR:mi_fifo:mi_fifo_server: command (dp_reload) processing failed

Haw else can i make memory debug without Opensips stopped?

Thank you.

 mailto:denis7...@mail.ru


No, more like the 2.2+ LTS, due to be released towards the end of March, a fork 
of the current "master" branch on git.
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 29.02.2016 12:58, Денис Путято wrote:
Re: [OpenSIPS-Users] Out of memory problem Newer version = 2.1?

mailto:denis7...@mail.ru


This is PKG memory (i.e. packaged / private / per-process), so you should 
actually increase "-M" CLI switch!

Note: Newer versions of OpenSIPS will have improved error reporting for easier 
troubleshooting of oom (out-of-memory) errors.
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 29.02.2016 10:20, Денис Путято wrote:
Re: [OpenSIPS-Users] Out of memory problem ERROR:core:db_allocate_rows: no 
memory left
ERROR:db_mysql:db_mysql_fetch_result: no memory left
ERROR:dialplan:dp_load_db: failed to fetch
ERROR:dialplan:dp_load_all_db: unable to load ast_dialplan table
ERROR:dialplan:mi_reload_rules: failed to reload database
ERROR:mi_fifo:mi_fifo_server: command () processing failed


mailto:denis7...@mail.ru

> What is the exact text of the error?
> ‎
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 303 Perimeter Center North, Suite 300
> Atlanta, GA 30346
> United States

> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
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> Sent from my BlackBerry.
>   Original Message  
> From: Денис Путято
> Sent: Monday, February 29, 2016 00:58
> To: Alex Balashov; OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Out of memory problem


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Re: [OpenSIPS-Users] Fraud module

[Liviu Chircu]

Hi Denis,

Currently this statistic is not reset between intervals, but only when a 
user dials a different number than the one(s) before.
This makes it so that the module never loses track of the sequence of 
calls a particular user has made.


In your particular case, does it make more sense for the module to do a 
reset?


Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 31.05.2016 15:50, Денис Путято wrote:

Fraud module Hello!

I am using Opensips 2.1 with fraud module.
Should the module resets current statistics when new time stamp begin?
I use 00:00 - 23:59 time interval for fraud detection and see that, 
for example, "sequential calls" increases for every call every day.


Thank you

mailto:denis7...@mail.ru


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[OpenSIPS-Users] Fraud_detection module

[Denis]

Hello!

I am using Opensips 2.1 with fraud module.
Should the module resets current statistics when new time stamp begin?
I use 00:00 - 23:59 time interval for fraud detection and see that, for 
example, "sequential calls" increases for every call every day.

Thank you

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Re: [OpenSIPS-Users] Opensips CP DROUTING GATEWAY enable/disable issue

[Jeff Wilkie]

OpenSIPS Control Panel version 5.0

Jeff Wilkie
Chief Technology Officer
US IP Communications
919.297.1057


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On Tue, May 31, 2016 at 3:28 AM, Bogdan-Andrei Iancu 
wrote:

> Hi Jeff,
>
> What OpenSIPS Control Panel version are you using ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 21.05.2016 00:15, Jeff Wilkie wrote:
>
> OPENSIPS 1.10.x
> I have the following set
>
> opensips.cfg
>
>  FIFO Management Interface
>
> loadmodule "mi_fifo.so"
>
> modparam("mi_fifo", "fifo_name", "/tmp/opensips_proxy_fifo")
>
> modparam("mi_fifo", "fifo_mode", 0666)
>
> opensipsctlrc
>
> ## path to FIFO file
>
> OSIPS_FIFO="/tmp/opensips_proxy_fifo"
>
>
> Attempting to disable gateways via the CP gives the following errors:
>
>
> From DROUTING-Gateway interface:
>
>
> Error while disabling gateway 2
> (the GWID is 2 for the gateway I'm attempting to disable)
>
> From the MI Commands:
> Initiating the following command: *dr_gw_status 2 0*
> 404 GW ID not found
>
> From the DROUTING-Gateway interface I am able to enable the interface if
> it is disabled
> I'm able to also enable the Gateway from the MI Commands section.
>
> I'm also able to enable and disable the Gateway using opensipsctl fifo
> commands
>
> opensipsctl fifo dr_gw_status 2
>
> Enabled:: yes
>
> opensipsctl fifo dr_gw_status 2 0
>
> opensipsctl fifo dr_gw_status 2
>
> Enabled:: no
>
> opensipsctl fifo dr_gw_status 2 1
>
> opensipsctl fifo dr_gw_status 2
>
> Enabled:: yes
>
> Not sure where the problem is but I feel its somewhere in the syntax of
> how its delivered.  I'm sure it's something easy I've overlooked.  Any help
> on this?
>
> Thanks
> Jeff
>
>
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>
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Re: [OpenSIPS-Users] engage_rtp_proxy()

[Sasmita Panda]

In my case its working great . So I haven't done such experiments to know
what is happening with dialog module  . We are using this form years .

 If you got to know then let me know also . That may help me in future .

*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*

On Tue, May 31, 2016 at 6:56 PM, Miha  wrote:

> @Sasmita I had writen cfg script like this and it works. I tried than with
> engage_rtp_proxy() but did not work automatically, that is why i asked :)
>
> So, can be some issue with dialog module? Not configured properly?
>
>
> tnx
>
> miha
>
>
> On 31/05/2016 15:21, Sasmita Panda wrote:
>
> Yes . This should happen . But I don't know the exact problem . What I
> explain is the way we are using rtpproxy .
> This is clearly mention in the document also .. You can go through
> opensips.org
>
> This is what we are doing .  Rest I am not an expertise in opensips .
>
> route {
> ...
> if (is_method("INVITE")) {
> if (has_body("application/sdp")) {
> if (rtpproxy_offer())
> t_on_reply("1");
> } else {
> t_on_reply("2");
> }
> }
> if (is_method("ACK") && has_body("application/sdp"))
> rtpproxy_answer();
> ...
> }
>
> onreply_route[1]
> {
> ...
> if (has_body("application/sdp"))
> rtpproxy_answer();
> ...
> }
>
> onreply_route[2]
> {
> ...
> if (has_body("application/sdp"))
> rtpproxy_offer();
> ...
> }
>
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
> On Tue, May 31, 2016 at 6:32 PM, Max Mühlbronner  wrote:
>
>> Hi,
>>
>>
>> @Miha: Are you sure that it does not automatically set the rtpproxies for
>> 200OK & ACK?
>>
>> @Sasmita: According to the documentation it is not necessary to invoke
>> engage_rtp_proxy() for replies as this is handled by the dialog module.
>>
>>
>> "Function must only be called for the initial INVITE and internally takes
>> care of rewriting the body of 200 OKs and ACKs. "
>>
>>
>>
>> Best Regards
>>
>> Max M.
>>
>>
>> On 31.05.2016 14:42, Miha wrote:
>>
>> @Sasmita, totally clear :)
>>
>> I asked wrong question :)
>>
>>
>> What is the difference between using engage_rtp_proxy() or using
>> rtpproxy_offer(), rtpproxy_answer()?
>>
>>
>> tnx
>>
>> miha
>>
>>
>> On 31/05/2016 14:39, Sasmita Panda wrote:
>>
>> If you are using in INVITE , then it should be offer . Because
>> firstly we are offering media to someone . If its 200 Ok then it will be
>> answer because the 2nd party is answering the call .
>>
>>   If there is no sdp in INVITE but in ACK , then it will get reversed
>> . In 200 OK you should offer and in ACK you have to answer .
>> This can be done in loop .
>>
>>  I hope I make you understand .
>>
>> */Thanks & Regards/*
>> /Sasmita Panda/
>> /Network Testing and Software Engineer/
>> /3CLogic , ph:07827611765/
>>
>> On Tue, May 31, 2016 at 6:02 PM, Miha < m...@softnet.si
>>  > wrote:
>>
>> ok tnx. I understand documentation on wrong way.
>>
>> But then, what is the difference with  using rtpproxy offer, answer ?
>>
>>
>> br
>>
>> mia
>>
>>
>> On 31/05/2016 14:17, Sasmita Panda wrote:
>>
>> If there is sdp in ACK and u wanted to engage rtp proxy , the
>>  you have to write it inside ACK also ... By writing for INVITE
>> cant help you to update ACK also . For 200 OK , you must write it
>> in reply route .
>>
>> */Thanks & Regards/*
>> /Sasmita Panda/
>> /Network Testing and Software Engineer/
>> /3CLogic , ph:07827611765/
>>
>> On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq
>> < jo...@democon.be 
>> > wrote:
>>
>> put it also in reply route.
>>
>> 2016-05-31 13:42 GMT+02:00 Miha < 
>> m...@softnet.si
>>  >:
>>
>> HI
>>
>> if I use engage_rtp_proxy(), I can use it only on initial
>> INVITE and opensips should automatically rewritten also
>> 200 OK and ACK with SDP, right?
>> But when I am using this function, I can see from trace
>> that only SDP for initial invite is rewritten, 200 ok
>> with sdp is not changed. Must I do something else?
>>
>> Rtpproxy is not running in bridge mode.
>>
>>
>> tnx
>> miha
>>
>>
>>
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>>
>>
>>
>> 

Re: [OpenSIPS-Users] OpenSips as Load Balancer - Handling Failure Scenarios

[Bogdan-Andrei Iancu]

Hi Chandan,

If the call is rejected by callee, your opensips should receive a 
negative reply and trigger the failure route - what is the reply code 
you get back from callee ? Is your failure route triggered ?


Regards,

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

On 31.05.2016 16:31, Chandan PR wrote:

Hi Guys,

I am trying to configure OpenSips as Load Balancer for our outbound 
dialling.


I am following the example from:
http://www.opensips.org/Documentation/Tutorials-LoadBalancing-1-9.

Right now when the user rejects the call, instead of ending up in 486 
we are ending up in 408.


This is due to the call being ended up in the failure_route after the 
timeout (fr_inv_timer).


Is there a way or configuration to send the response codes from the 
reply back to client, instead of waiting for timeout?



Regards,
Chandan


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Re: [OpenSIPS-Users] how to make resume invite on opensips

[Bogdan-Andrei Iancu]

Hello Tuan,

Not sure what you mean by "caching the INVITE package" but how usually 
PN in implemented is to delay (at server level) the relay of the INVITE 
(to the callee) while you run an extern script (triggered when the 
INVITE is received on the proxy) to do the PN.


Regards,

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

On 31.05.2016 12:56, Tuan Phan Ngoc wrote:


Dear,

I cached invite packages from caller when callee offline on 
redis-cache, push notification to wake up callee and resend to callee. 
I don't know how to opensips know package (Ringing,... ) from callee 
and forward to calller.


I can't find any documents about follow route call for opensips.

Thanks for your help.


--
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Phone 0985-251-257.
U29B-U31B-U33B Road Tan Thuan, Tan Thuan EPZ, District 7, HCM City, 
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[OpenSIPS-Users] OpenSips as Load Balancer - Handling Failure Scenarios

[Chandan PR]

Hi Guys,

I am trying to configure OpenSips as Load Balancer for our outbound
dialling.

I am following the example from:
http://www.opensips.org/Documentation/Tutorials-LoadBalancing-1-9.

Right now when the user rejects the call, instead of ending up in 486 we
are ending up in 408.

This is due to the call being ended up in the failure_route after the
timeout (fr_inv_timer).

Is there a way or configuration to send the response codes from the reply
back to client, instead of waiting for timeout?


Regards,
Chandan
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Re: [OpenSIPS-Users] engage_rtp_proxy()

[Miha]

@Sasmita I had writen cfg script like this and it works. I tried than 
with engage_rtp_proxy() but did not work automatically, that is why i 
asked :)


So, can be some issue with dialog module? Not configured properly?


tnx

miha


On 31/05/2016 15:21, Sasmita Panda wrote:
Yes . This should happen . But I don't know the exact problem . What I 
explain is the way we are using rtpproxy .
This is clearly mention in the document also .. You can go through 
opensips.org 


This is what we are doing .  Rest I am not an expertise in opensips .
route {
...
 if (is_method("INVITE")) {
 if (has_body("application/sdp")) {
 if (rtpproxy_offer())
 t_on_reply("1");
 } else {
 t_on_reply("2");
 }
 }
 if (is_method("ACK") && has_body("application/sdp"))
 rtpproxy_answer();
...
}

onreply_route[1]
{
...
 if (has_body("application/sdp"))
 rtpproxy_answer();
...
}

onreply_route[2]
{
...
 if (has_body("application/sdp"))
 rtpproxy_offer();
...
}

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 6:32 PM, Max Mühlbronner > wrote:


Hi,


@Miha: Are you sure that it does not automatically set the
rtpproxies for 200OK & ACK?

@Sasmita: According to the documentation it is not necessary to
invoke engage_rtp_proxy() for replies as this is handled by the
dialog module.


"Function must only be called for the initial INVITE and
internally takes care of rewriting the body of 200 OKs and ACKs. "



Best Regards

Max M.


On 31.05.2016 14:42, Miha wrote:

@Sasmita, totally clear :)

I asked wrong question :)


What is the difference between using engage_rtp_proxy() or using
rtpproxy_offer(), rtpproxy_answer()?


tnx

miha


On 31/05/2016 14:39, Sasmita Panda wrote:

If you are using in INVITE , then it should be offer .
Because firstly we are offering media to someone . If its 200 Ok
then it will be answer because the 2nd party is answering the
call .

  If there is no sdp in INVITE but in ACK , then it will get
reversed . In 200 OK you should offer and in ACK you have to
answer .
This can be done in loop .

 I hope I make you understand .

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 6:02 PM, Miha  
> wrote:

ok tnx. I understand documentation on wrong way.

But then, what is the difference with  using rtpproxy offer,
answer ?


br

mia


On 31/05/2016 14:17, Sasmita Panda wrote:

If there is sdp in ACK and u wanted to engage rtp proxy , the
 you have to write it inside ACK also ... By writing for
INVITE
cant help you to update ACK also . For 200 OK , you must
write it
in reply route .

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq

 > wrote:

put it also in reply route.

2016-05-31 13:42 GMT+02:00 Miha 
 >:

HI

if I use engage_rtp_proxy(), I can use it only on
initial
INVITE and opensips should automatically rewritten
also
200 OK and ACK with SDP, right?
But when I am using this function, I can see from
trace
that only SDP for initial invite is rewritten, 200 ok
with sdp is not changed. Must I do something else?

Rtpproxy is not running in bridge mode.


tnx
miha



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Re: [OpenSIPS-Users] engage_rtp_proxy()

[Miha]

@Max, yes I am sure. Did some traces...

That is way I asked what I could be doing wrong. I read that there is an 
issue doing it with rtpproxy in bridge, but in my case it is not in bridge.



tnx

miha


On 31/05/2016 15:02, Max Mühlbronner wrote:

Hi,


@Miha: Are you sure that it does not automatically set the rtpproxies 
for 200OK & ACK?


@Sasmita: According to the documentation it is not necessary to invoke 
engage_rtp_proxy() for replies as this is handled by the dialog module.



"Function must only be called for the initial INVITE and internally 
takes care of rewriting the body of 200 OKs and ACKs. "




Best Regards

Max M.


On 31.05.2016 14:42, Miha wrote:

@Sasmita, totally clear :)

I asked wrong question :)


What is the difference between using engage_rtp_proxy() or using 
rtpproxy_offer(), rtpproxy_answer()?



tnx

miha


On 31/05/2016 14:39, Sasmita Panda wrote:
If you are using in INVITE , then it should be offer . 
Because firstly we are offering media to someone . If its 200 Ok 
then it will be answer because the 2nd party is answering the call .


  If there is no sdp in INVITE but in ACK , then it will get 
reversed . In 200 OK you should offer and in ACK you have to answer .

This can be done in loop .

 I hope I make you understand .

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 6:02 PM, Miha > wrote:


ok tnx. I understand documentation on wrong way.

But then, what is the difference with  using rtpproxy offer, 
answer ?



br

mia


On 31/05/2016 14:17, Sasmita Panda wrote:

If there is sdp in ACK and u wanted to engage rtp proxy , the
 you have to write it inside ACK also ... By writing for INVITE
cant help you to update ACK also . For 200 OK , you must write it
in reply route .

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq
> wrote:

put it also in reply route.

2016-05-31 13:42 GMT+02:00 Miha >:

HI

if I use engage_rtp_proxy(), I can use it only on initial
INVITE and opensips should automatically rewritten also
200 OK and ACK with SDP, right?
But when I am using this function, I can see from trace
that only SDP for initial invite is rewritten, 200 ok
with sdp is not changed. Must I do something else?

Rtpproxy is not running in bridge mode.


tnx
miha



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Re: [OpenSIPS-Users] engage_rtp_proxy()

[Sasmita Panda]

Yes . This should happen . But I don't know the exact problem . What I
explain is the way we are using rtpproxy .
This is clearly mention in the document also .. You can go through
opensips.org

This is what we are doing .  Rest I am not an expertise in opensips .

route {
...
if (is_method("INVITE")) {
if (has_body("application/sdp")) {
if (rtpproxy_offer())
t_on_reply("1");
} else {
t_on_reply("2");
}
}
if (is_method("ACK") && has_body("application/sdp"))
rtpproxy_answer();
...
}

onreply_route[1]
{
...
if (has_body("application/sdp"))
rtpproxy_answer();
...
}

onreply_route[2]
{
...
if (has_body("application/sdp"))
rtpproxy_offer();
...
}


*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*

On Tue, May 31, 2016 at 6:32 PM, Max Mühlbronner  wrote:

> Hi,
>
>
> @Miha: Are you sure that it does not automatically set the rtpproxies for
> 200OK & ACK?
>
> @Sasmita: According to the documentation it is not necessary to invoke
> engage_rtp_proxy() for replies as this is handled by the dialog module.
>
>
> "Function must only be called for the initial INVITE and internally takes
> care of rewriting the body of 200 OKs and ACKs. "
>
>
>
> Best Regards
>
> Max M.
>
>
> On 31.05.2016 14:42, Miha wrote:
>
> @Sasmita, totally clear :)
>
> I asked wrong question :)
>
>
> What is the difference between using engage_rtp_proxy() or using
> rtpproxy_offer(), rtpproxy_answer()?
>
>
> tnx
>
> miha
>
>
> On 31/05/2016 14:39, Sasmita Panda wrote:
>
> If you are using in INVITE , then it should be offer . Because
> firstly we are offering media to someone . If its 200 Ok then it will be
> answer because the 2nd party is answering the call .
>
>   If there is no sdp in INVITE but in ACK , then it will get reversed
> . In 200 OK you should offer and in ACK you have to answer .
> This can be done in loop .
>
>  I hope I make you understand .
>
> */Thanks & Regards/*
> /Sasmita Panda/
> /Network Testing and Software Engineer/
> /3CLogic , ph:07827611765/
>
> On Tue, May 31, 2016 at 6:02 PM, Miha   > wrote:
>
> ok tnx. I understand documentation on wrong way.
>
> But then, what is the difference with  using rtpproxy offer, answer ?
>
>
> br
>
> mia
>
>
> On 31/05/2016 14:17, Sasmita Panda wrote:
>
> If there is sdp in ACK and u wanted to engage rtp proxy , the
>  you have to write it inside ACK also ... By writing for INVITE
> cant help you to update ACK also . For 200 OK , you must write it
> in reply route .
>
> */Thanks & Regards/*
> /Sasmita Panda/
> /Network Testing and Software Engineer/
> /3CLogic , ph:07827611765/
>
> On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq
>  >
> wrote:
>
> put it also in reply route.
>
> 2016-05-31 13:42 GMT+02:00 Miha   >:
>
> HI
>
> if I use engage_rtp_proxy(), I can use it only on initial
> INVITE and opensips should automatically rewritten also
> 200 OK and ACK with SDP, right?
> But when I am using this function, I can see from trace
> that only SDP for initial invite is rewritten, 200 ok
> with sdp is not changed. Must I do something else?
>
> Rtpproxy is not running in bridge mode.
>
>
> tnx
> miha
>
>
>
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>
>
>
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>
>
>
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>
>
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>
>
>
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>
>
>
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>
>

Re: [OpenSIPS-Users] engage_rtp_proxy()

[Max Mühlbronner]

Hi,


@Miha: Are you sure that it does not automatically set the rtpproxies 
for 200OK & ACK?


@Sasmita: According to the documentation it is not necessary to invoke 
engage_rtp_proxy() for replies as this is handled by the dialog module.



"Function must only be called for the initial INVITE and internally 
takes care of rewriting the body of 200 OKs and ACKs. "




Best Regards

Max M.


On 31.05.2016 14:42, Miha wrote:

@Sasmita, totally clear :)

I asked wrong question :)


What is the difference between using engage_rtp_proxy() or using 
rtpproxy_offer(), rtpproxy_answer()?



tnx

miha


On 31/05/2016 14:39, Sasmita Panda wrote:
If you are using in INVITE , then it should be offer . 
Because firstly we are offering media to someone . If its 200 Ok then 
it will be answer because the 2nd party is answering the call .


  If there is no sdp in INVITE but in ACK , then it will get 
reversed . In 200 OK you should offer and in ACK you have to answer .

This can be done in loop .

 I hope I make you understand .

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 6:02 PM, Miha > wrote:


ok tnx. I understand documentation on wrong way.

But then, what is the difference with  using rtpproxy offer, 
answer ?



br

mia


On 31/05/2016 14:17, Sasmita Panda wrote:

If there is sdp in ACK and u wanted to engage rtp proxy , the
 you have to write it inside ACK also ... By writing for INVITE
cant help you to update ACK also . For 200 OK , you must write it
in reply route .

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq
> wrote:

put it also in reply route.

2016-05-31 13:42 GMT+02:00 Miha >:

HI

if I use engage_rtp_proxy(), I can use it only on initial
INVITE and opensips should automatically rewritten also
200 OK and ACK with SDP, right?
But when I am using this function, I can see from trace
that only SDP for initial invite is rewritten, 200 ok
with sdp is not changed. Must I do something else?

Rtpproxy is not running in bridge mode.


tnx
miha



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[OpenSIPS-Users] Fraud module

[Денис Путято]

Hello!

I am using Opensips 2.1 with fraud module.
Should the module resets current statistics when new time stamp begin?
I use 00:00 - 23:59 time interval for fraud detection and see that, for 
example, "sequential calls" increases for every call every day.

Thank you

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Re: [OpenSIPS-Users] engage_rtp_proxy()

[Miha]

@Sasmita, totally clear :)

I asked wrong question :)


What is the difference between using engage_rtp_proxy() or using 
rtpproxy_offer(), rtpproxy_answer()?



tnx

miha


On 31/05/2016 14:39, Sasmita Panda wrote:
If you are using in INVITE , then it should be offer . Because 
firstly we are offering media to someone . If its 200 Ok then it will 
be answer because the 2nd party is answering the call .


  If there is no sdp in INVITE but in ACK , then it will get 
reversed . In 200 OK you should offer and in ACK you have to answer .

This can be done in loop .

 I hope I make you understand .

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 6:02 PM, Miha > wrote:


ok tnx. I understand documentation on wrong way.

But then, what is the difference with  using rtpproxy offer, answer ?


br

mia


On 31/05/2016 14:17, Sasmita Panda wrote:

If there is sdp in ACK and u wanted to engage rtp proxy , the
 you have to write it inside ACK also ... By writing for INVITE
cant help you to update ACK also . For 200 OK , you must write it
in reply route .

*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq
> wrote:

put it also in reply route.

2016-05-31 13:42 GMT+02:00 Miha >:

HI

if I use engage_rtp_proxy(), I can use it only on initial
INVITE and opensips should automatically rewritten also
200 OK and ACK with SDP, right?
But when I am using this function, I can see from trace
that only SDP for initial invite is rewritten, 200 ok
with sdp is not changed. Must I do something else?

Rtpproxy is not running in bridge mode.


tnx
miha



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Re: [OpenSIPS-Users] engage_rtp_proxy()

[Sasmita Panda]

If you are using in INVITE , then it should be offer . Because
firstly we are offering media to someone . If its 200 Ok then it will be
answer because the 2nd party is answering the call .

  If there is no sdp in INVITE but in ACK , then it will get reversed .
In 200 OK you should offer and in ACK you have to answer .
This can be done in loop .

 I hope I make you understand .

*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*

On Tue, May 31, 2016 at 6:02 PM, Miha  wrote:

> ok tnx. I understand documentation on wrong way.
>
> But then, what is the difference with  using rtpproxy offer, answer ?
>
>
> br
>
> mia
>
> On 31/05/2016 14:17, Sasmita Panda wrote:
>
> If there is sdp in ACK and u wanted to engage rtp proxy , the  you have to
> write it inside ACK also ... By writing for INVITE cant help you to update
> ACK also . For 200 OK , you must write it in reply route .
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
> On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq  wrote:
>
>> put it also in reply route.
>>
>> 2016-05-31 13:42 GMT+02:00 Miha :
>>
>>> HI
>>>
>>> if I use engage_rtp_proxy(), I can use it only on initial INVITE and
>>> opensips should automatically rewritten also 200 OK and ACK with SDP, right?
>>> But when I am using this function, I can see from trace that only SDP
>>> for initial invite is rewritten, 200 ok with sdp is not changed. Must I do
>>> something else?
>>>
>>> Rtpproxy is not running in bridge mode.
>>>
>>>
>>> tnx
>>> miha
>>>
>>>
>>>
>>> ___
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>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
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>>
>
>
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Re: [OpenSIPS-Users] engage_rtp_proxy()

[Miha]

ok tnx. I understand documentation on wrong way.

But then, what is the difference with  using rtpproxy offer, answer ?


br

mia


On 31/05/2016 14:17, Sasmita Panda wrote:
If there is sdp in ACK and u wanted to engage rtp proxy , the  you 
have to write it inside ACK also ... By writing for INVITE cant help 
you to update ACK also . For 200 OK , you must write it in reply route .


*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq > wrote:


put it also in reply route.

2016-05-31 13:42 GMT+02:00 Miha >:

HI

if I use engage_rtp_proxy(), I can use it only on initial
INVITE and opensips should automatically rewritten also 200 OK
and ACK with SDP, right?
But when I am using this function, I can see from trace that
only SDP for initial invite is rewritten, 200 ok with sdp is
not changed. Must I do something else?

Rtpproxy is not running in bridge mode.


tnx
miha



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Re: [OpenSIPS-Users] engage_rtp_proxy()

[Sasmita Panda]

If there is sdp in ACK and u wanted to engage rtp proxy , the  you have to
write it inside ACK also ... By writing for INVITE cant help you to update
ACK also . For 200 OK , you must write it in reply route .

*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*

On Tue, May 31, 2016 at 5:35 PM, Johan De Clercq  wrote:

> put it also in reply route.
>
> 2016-05-31 13:42 GMT+02:00 Miha :
>
>> HI
>>
>> if I use engage_rtp_proxy(), I can use it only on initial INVITE and
>> opensips should automatically rewritten also 200 OK and ACK with SDP, right?
>> But when I am using this function, I can see from trace that only SDP for
>> initial invite is rewritten, 200 ok with sdp is not changed. Must I do
>> something else?
>>
>> Rtpproxy is not running in bridge mode.
>>
>>
>> tnx
>> miha
>>
>>
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
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Re: [OpenSIPS-Users] compiling opensips on a raspberry pi

[Steve Woolley]

Thanks. Recompiled with 2.2 and everything’s quick and peppy. Thanks for the 
help Bogdan.

--
Steve Woolley
steve.wool...@me.com



> On May 30, 2016, at 5:25 AM, Bogdan-Andrei Iancu  wrote:
> 
> Hi Steve,
> 
> Please update from git and give it a try again (there were couple of small 
> fixes). Still, I recommend to go for 2.2 .
> 
> Regards,
>  Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com 
> On 27.05.2016 22:21, Steve Woolley wrote:
>> Am trying to run opensips on a raspberry pi OS:Raspbian (based on Debian 
>> Jessie).
>> 
>> Compiled successfully but take a long time to start up opensips. 
>> (35-45 minutes).
>> Shows the following in the log (during the long pause):
>>  DBG:tm:lock_initialize: lock initialization started
>> 
>> 
>> Bogan-Andrei Iancu has suggested this is probably due to "System V locking 
>> support”.
>> He suggested trying to set USE_PTHREAD_MUTEX or USE_POSIX_SEMS compile 
>> options.
>> However, both of these options fail on:
>> 
>> Compiling net/net_tcp.c
>> net/net_tcp.c: In function ‘tcpconn_new’:
>> net/net_tcp.c:844:7: error: used union type value where scalar is required
>>if (c->write_lock) lock_destroy(>write_lock);
>>^
>> Makefile.rules:25: recipe for target 'net/net_tcp.o' failed
>> make: *** [net/net_tcp.o] Error 1
>> 
>> Anyone had any luck on setting these compile options and/or getting opensips 
>> (efficiently) running on a Raspberry Pi?
>> 
>> --
>> Steve Woolley
>> steve.wool...@me.com 
>> 
>> 
>> 
>> 
>> 
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>> 
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Re: [OpenSIPS-Users] engage_rtp_proxy()

[Johan De Clercq]

put it also in reply route.

2016-05-31 13:42 GMT+02:00 Miha :

> HI
>
> if I use engage_rtp_proxy(), I can use it only on initial INVITE and
> opensips should automatically rewritten also 200 OK and ACK with SDP, right?
> But when I am using this function, I can see from trace that only SDP for
> initial invite is rewritten, 200 ok with sdp is not changed. Must I do
> something else?
>
> Rtpproxy is not running in bridge mode.
>
>
> tnx
> miha
>
>
>
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[OpenSIPS-Users] engage_rtp_proxy()

[Miha]

HI

if I use engage_rtp_proxy(), I can use it only on initial INVITE and 
opensips should automatically rewritten also 200 OK and ACK with SDP, right?
But when I am using this function, I can see from trace that only SDP 
for initial invite is rewritten, 200 ok with sdp is not changed. Must I 
do something else?


Rtpproxy is not running in bridge mode.


tnx
miha



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[OpenSIPS-Users] how to make resume invite on opensips

[Tuan Phan Ngoc]

Dear,

I cached invite packages from caller when callee offline on redis-cache,
push notification to wake up callee and resend to callee. I don't know how
to opensips know package (Ringing,... ) from callee and forward to calller.

I can't find any documents about follow route call for opensips.

Thanks for your help.


-- 

Thanks and Best Regards,

 * PHAN NGOC TUAN*


  Phone 0985-251-257.

  U29B-U31B-U33B Road Tan Thuan, Tan Thuan EPZ,
District 7, HCM City, Vietnam
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Re: [OpenSIPS-Users] Question regarding b2b_bridge function

[Bogdan-Andrei Iancu]

Hi Chris,

The "dialog_id" is actually the b2b key, that is expose by the b2b_logic 
via the module parameter b2bl_key_avp. See:

http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id294094

That key can also be found in the b2b_logic table in DB.

At signaling level, the key is the Call-ID of the outbound calls from b2b.

Regards,

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

On 23.05.2016 07:32, Ping Han wrote:

Hi Bogdan,

I asked the question a few days ago but have not got a response.

I am just wondering if I could get some advice from you.

Any advice will be appreciated.

Thanks,
Chris

On Wed, May 18, 2016 at 4:39 PM, Ping Han > wrote:


Hi,

I would like to use the b2b_bridge fifo function as specified at
http://www.opensips.org/html/docs/modules/2.1.x/b2b_logic.html#id248916.

The function will be triggered by a third party. I will need to
pass the parameters to the third party for it to trigger the
function. One of the parameters is the "dialog-id".

The problem is that I am not sure how the value of the dialog-id
can be available in the Opensips config. Is there any Opensips
modules/function that can retrieve the value of the dialog-id?

I tried to get the value from the "b2b_entities" and "b2b_logic"
table. However, it seems that it does not work this way because
the two tables do not pop the data in real time. Sometimes I can
see the data but sometimes I am not able to see it.

It is appreciated that you can give me some idea.

Thanks,

Ping




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Re: [OpenSIPS-Users] Opensips starting error?

[Bogdan-Andrei Iancu]

Hi,

OpenSIPS 1.6 is really old and deprecated - I strongly suggest you to 
use one of the maintained versions like 1.11 or 2.2


Regards,

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

On 24.05.2016 10:09, mir_as82 wrote:

now i am watching log file of opensips. There are some errors.
May 24 09:51:42 as-VirtualBox /usr/local/sbin/opensips[7322]:
NOTICE:core:main: version: opensips 1.6.0-notls (i386/linux)
May 24 09:51:42 as-VirtualBox /usr/local/sbin/opensips[7322]:
NOTICE:signaling:mod_init: initializing module ...
May 24 09:51:42 as-VirtualBox /usr/local/sbin/opensips[7322]:
ERROR:db_text:dbt_query: table does not exist!
May 24 09:51:42 as-VirtualBox /usr/local/sbin/opensips[7322]:
ERROR:core:db_table_version: error in db_query
May 24 09:51:42 as-VirtualBox /usr/local/sbin/opensips[7322]:
ERROR:core:db_check_table_version: querying version for table subscriber
May 24 09:51:42 as-VirtualBox /usr/local/sbin/opensips[7322]:
ERROR:auth_db:auth_fixup: error during table version check.
May 24 09:51:42 as-VirtualBox /usr/local/sbin/opensips[7322]:
ERROR:core:fix_actions: fixing failed (code=-1) at cfg line 404
May 24 09:51:42 as-VirtualBox /usr/local/sbin/opensips[7322]:
CRITICAL:core:fix_expr: fix_actions error
May 24 09:51:42 as-VirtualBox /usr/local/sbin/opensips[7322]:
ERROR:core:main: failed to fix configuration with err code -1




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Re: [OpenSIPS-Users] Opensips CP DROUTING GATEWAY enable/disable issue

[Bogdan-Andrei Iancu]

Hi Jeff,

What OpenSIPS Control Panel version are you using ?

Regards,

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

On 21.05.2016 00:15, Jeff Wilkie wrote:

OPENSIPS 1.10.x
I have the following set

opensips.cfg

 FIFO Management Interface

loadmodule "mi_fifo.so"

modparam("mi_fifo", "fifo_name", "/tmp/opensips_proxy_fifo")

modparam("mi_fifo", "fifo_mode", 0666)


opensipsctlrc

## path to FIFO file

OSIPS_FIFO="/tmp/opensips_proxy_fifo"


Attempting to disable gateways via the CP gives the following errors:


From DROUTING-Gateway interface:


Error while disabling gateway 2
(the GWID is 2 for the gateway I'm attempting to disable)

From the MI Commands:
Initiating the following command: *dr_gw_status 2 0*
404 GW ID not found

From the DROUTING-Gateway interface I am able to enable the interface 
if it is disabled

I'm able to also enable the Gateway from the MI Commands section.

I'm also able to enable and disable the Gateway using opensipsctl fifo 
commands


opensipsctl fifo dr_gw_status 2

Enabled:: yes

opensipsctl fifo dr_gw_status 2 0

opensipsctl fifo dr_gw_status 2

Enabled:: no

opensipsctl fifo dr_gw_status 2 1

opensipsctl fifo dr_gw_status 2

Enabled:: yes


Not sure where the problem is but I feel its somewhere in the syntax 
of how its delivered.  I'm sure it's something easy I've overlooked.  
Any help on this?


Thanks
Jeff


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Re: [OpenSIPS-Users] Opensips starting error?

[mir_as82]

now i am watching log file of opensips. There are some errors.
May 24 09:51:42 as-VirtualBox /usr/local/sbin/opensips[7322]:
NOTICE:core:main: version: opensips 1.6.0-notls (i386/linux)
May 24 09:51:42 as-VirtualBox /usr/local/sbin/opensips[7322]:
NOTICE:signaling:mod_init: initializing module ...
May 24 09:51:42 as-VirtualBox /usr/local/sbin/opensips[7322]:
ERROR:db_text:dbt_query: table does not exist!
May 24 09:51:42 as-VirtualBox /usr/local/sbin/opensips[7322]:
ERROR:core:db_table_version: error in db_query
May 24 09:51:42 as-VirtualBox /usr/local/sbin/opensips[7322]:
ERROR:core:db_check_table_version: querying version for table subscriber
May 24 09:51:42 as-VirtualBox /usr/local/sbin/opensips[7322]:
ERROR:auth_db:auth_fixup: error during table version check.
May 24 09:51:42 as-VirtualBox /usr/local/sbin/opensips[7322]:
ERROR:core:fix_actions: fixing failed (code=-1) at cfg line 404
May 24 09:51:42 as-VirtualBox /usr/local/sbin/opensips[7322]:
CRITICAL:core:fix_expr: fix_actions error
May 24 09:51:42 as-VirtualBox /usr/local/sbin/opensips[7322]:
ERROR:core:main: failed to fix configuration with err code -1




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[OpenSIPS-Users] Opensips radius prepaid

[sunter]

How can I implement a mechanism when the first message to the Radius Auth we
get a quota from BE , for example, 10 seconds ,  after 10 seconds, do re-
auth request to obtain the radius and again 10 seconds or hang up , so there
is no money. Now look toward the dialog module, write function to make
re-auth.



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[OpenSIPS-Users] Opensips starting error?

[mir_as82]

I am new in Opensips. I have installed opensips than i tried to run opensips.
but when when type "opensipsctl start" so i get following error message. How
can i know the reason of this problem? 
ERROR: PID file /var/run/opensips.pid does not exist -- OpenSIPS start
failed




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Re: [OpenSIPS-Users] sip.instance

[Bogdan-Andrei Iancu]

Hi,

You NEED to authenticate the calls / INVITEs too, but you do in wrong 
way - you need to use the proxy_challenge() for INVITEs and not 
www_challenge() (which is to be used only for REGISTER requests).


Regards,

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

On 24.05.2016 17:17, Johan De Clercq wrote:

NO, because you still authenticate on registratin.

2016-05-24 15:20 GMT+02:00 Frank Paaske >:


Thanks, but won’t this allow anyone to use this OpenSIPS instance
to make SIP calls?

Best regards,

Frank

*From: *> on behalf of Johan De
Clercq >
*Reply-To: *OpenSIPS users mailling list >
*Date: *tirsdag 24. mai 2016 08.16
*To: *OpenSIPS users mailling list >
*Subject: *Re: [OpenSIPS-Users] sip.instance

you need to remove authentication on invite.

Check in opensips.cfg: if(method=="INVITE")
{


www_challenge()

}

comment that part out and restart opensips.

2016-05-24 0:12 GMT+02:00 Frank Paaske >:

Hi,

I’m trying to get a liblinphone based client connected to
OpenSIPS.

The REGISTER is challenged with a 401, which is automatically
replied to with a proper Digest by liblinphone, which in turn
gives 200 OK back from OpenSIPS.

However, when trying to make a call, the INVITE is also
challenged with a 401. In this case the liblinphone is not
responding automatically.

According to the developers of the app, this happens because
OpenSIPS is not configured to support sip.instance.

The sequence is roughly like this:

Client Server

|-REGISTER-->|

|<---401-|

|-REGISTER-with-auth>|

|<---200-|

|-INVITE>|

|<---401-|

Then nothing

So my question is how can I enable support for sip.instance in
OpenSIPS?

Best regards,

Frank


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