date:20221020

Re: [OpenSIPS-Users] early dialog termination

2022-10-20 Thread Bogdan-Andrei Iancu


Ivan,

Actually a simpler approach will be to use t_wait_for_new_branches() 
instead of that t_write function, it should do the same trick 
(postponing the deletion of the transaction), but without any side effects.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Bootcamp 5-16 Dec 2022, online
  https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/

On 10/19/22 10:21 AM, Ryzhik Ivan wrote:
Sorry, I mean no sleep, i mean async( sleep($var(wait_time)), 
after_sleep );

Regards, Ivan.

вт, 18 окт. 2022 г. в 14:42, Bogdan-Andrei Iancu >:


Hi,

yes, call it before ending the REQUEST route. I'm 100% the
transaction is not deleted before the end of the route. And try to
use the unix sock flavor for the function, not the fifo one.

DO NOT use the sleep, you will block your whole opensips.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com  
OpenSIPS Bootcamp 5-16 Dec 2022, online
   https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/  


On 10/17/22 11:56 AM, Ryzhik Ivan wrote:

Hi, did you mean that i must call t_write_req once before
REQUEST_ROUTE is finished? In this case the transaction was removed.
"even if you do not have to actually write anything to outer
world - just fake it." - i must use fifo and i must read data
from it, in else we got:
ERROR:tm:write_to_fifo: nobody listening on [/tmp/moh.fifo] fifo
for reading!
ERROR:tm:t_write_req: write_to_fifo failed
And last question is may I use sleep(20) at the end of route to
keep transaction? or can i use modparam("tm", "wt_timer", 20)?
Regards, Ivan

пн, 17 окт. 2022 г. в 09:38, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>>:

Hi Ryzhik,

Right, the transaction must be forced to stay until you are
done with a final reply. Unfortunately there is no clear way
to do this from script (this may be subject of further small
improvements), but you can try taking advantage of the
`t_write_req` [1] for this purpose, even if you do not have
to actually write anything to outer world - just fake it.


[1]
https://opensips.org/html/docs/modules/3.2.x/tm.html#func_t_write_req


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com  

OpenSIPS Summit 27-30 Sept 2022, Athens
   https://www.opensips.org/events/Summit-2022Athens/  


On 10/13/22 2:45 PM, Ryzhik Ivan wrote:

Hi.
One more question.
Everything works fine except the transaction was deleted
after 15 sec after the initial route was finished.
on INVITE i last do t_reply_with_body(183, "Session
progress", ...) and than exit;

on end route log :

2022-10-13T10:58:01.994598+00:00  DBG:tm:_reply_light: reply
sent out. buf=0x7f558a087d98: SIP/2.0 1...,
shmem=0x7f5549797470: SIP/2.0 1
2022-10-13T10:58:01.994676+00:00  DBG:tm:_reply_light: finished

2022-10-13T10:58:01.995835+00:00  DBG:tm:do_t_cleanup:
transaction 0x7f5549793b18 already updated! Skipping update!
2022-10-13T10:58:01.996020+00:00  DBG:tm:cleanup_uac_timers:
RETR/FR timers reset
2022-10-13T10:58:01.996202+00:00
*DBG:tm:insert_timer_unsafe: [2]: 0x7f5549793b98 (12)*
2022-10-13T10:58:01.996317+00:00 * DBG:tm:t_unref:
UNREF_UNSAFE: [0x7f5549793b18] after is 0*
2022-10-13T10:58:01.996488+00:00  DBG:core:destroy_avp_list:
destroying list (nil)
2022-10-13T10:58:01.996673+00:00  DBG:core:receive_msg:
cleaning up

2022-10-13T10:58:07.651091+00:00* DBG:tm:timer_routine:
timer routine:2,tl=0x7f5549793b98 next=(nil), timeout=12*
2022-10-13T10:58:07.651332+00:00  DBG:tm:wait_handler:
removing 0x7f5549793b18 from table
2022-10-13T10:58:07.651425+00:00  DBG:tm:delete_ce*ll:
delete transaction 0x7f5549793b18*
2022-10-13T10:58:07.651513+00:00  DBG:tm:wait_handler: done

Can you tell me how I can i fix this? Transaction marked
safe for deletion...
Regards, Ivan

ср, 12 окт. 2022 г. в 13:11, Bogdan-Andrei Iancu
mailto:bog...@opensips.org>>:

Perfect !!!

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com  

OpenSIPS Summit 27-30 Sept 2022, Athens
   https://www.opensips.org/even

[OpenSIPS-Users] append body to reply?

2022-10-20 Thread Kingsley Tart

Hi,

Is it possible with OpenSIPS to append a body to a reply? If so, is
there also an equivalent of $ml for how long the outbound reply would
be? (I appreciate that until send_reply() is called, the exact size
would be impossible to calculate).

I'm finding it useful to use custom INFO calls to share metadata with
some of our other hosts. I could add the info in headers, but the
message body in some cases seems more appropriate.

Cheers,
Kingsley.


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Re: [OpenSIPS-Users] append body to reply?

2022-10-20 Thread Bogdan-Andrei Iancu


Hi Kingsley,

Docs are your best friend:
https://opensips.org/html/docs/modules/3.2.x/tm.html#func_t_reply_with_body

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Bootcamp 5-16 Dec 2022, online
  https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/

On 10/20/22 2:08 PM, Kingsley Tart wrote:

Hi,

Is it possible with OpenSIPS to append a body to a reply? If so, is
there also an equivalent of $ml for how long the outbound reply would
be? (I appreciate that until send_reply() is called, the exact size
would be impossible to calculate).

I'm finding it useful to use custom INFO calls to share metadata with
some of our other hosts. I could add the info in headers, but the
message body in some cases seems more appropriate.

Cheers,
Kingsley.


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Re: [OpenSIPS-Users] Dispatcher module and partition

2022-10-20 Thread Bogdan-Andrei Iancu


Hi Alberto,

Unfortunately all the mentioned settings are global, equally impacting 
all the partitions.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Bootcamp 5-16 Dec 2022, online
  https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/

On 10/19/22 6:20 PM, Alberto wrote:

Hi,

I'm using opensips 3.3.1 and I have a question about the dispatcher 
module.


Can I set different ds_ping_from, ds_ping_method, ds_probing_mode, and 
ds_ping_interval per partition?


Thanks

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Re: [OpenSIPS-Users] SIP user statuses to websocket

2022-10-20 Thread Bogdan-Andrei Iancu


Hi Anonymous,

Check the events interface in OpenSIPS [1]. Different modules (see their 
documentation) may raise different events, like when a user is 
registered, when a call is placed, etc.


[1] https://www.opensips.org/Documentation/Interface-Events-3-2

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Bootcamp 5-16 Dec 2022, online
  https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/

On 10/19/22 9:32 AM, ideanet help wrote:

Hi all,

I am looking for a way to use WebSocket with opensips. Basically, I 
want to show my all sip users statuses/activity on a web page. I am 
currently using an API, which is called after a few seconds so this is 
a bit slow. I am looking for a way that opensips sends event to the 
socket and on the other hand, my webpage will be connected to a 
WebSocket server. This way I can show all the events instantly.
For example, If a user receives a call I will have it on the web page 
straight away. if a user gets unregistered i will see on my page that 
users sent Expiry 0 something like that.


Will it be possible with the proto_wss module. any pointers will help. 
I saw what proto_wss can do but I did not understand it properly. it 
will send traces to a server i don't need. I just need it if a user is 
busy or idle. or he gets unregistered that's all.


Thanks

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Re: [OpenSIPS-Users] append body to reply?

2022-10-20 Thread Kingsley Tart

Hi,

thanks - I hunted through docs and googled for several hours but didn't
find that, though I did figure out that sipmsgops wasn't the answer.

Cheers,
Kingsley.

On Thu, 2022-10-20 at 15:41 +0300, Bogdan-Andrei Iancu wrote:
> Hi Kingsley,
> 
> Docs are your best friend:
> 
https://opensips.org/html/docs/modules/3.2.x/tm.html#func_t_reply_with_body
> 
> Regards,
> 
> Bogdan-Andrei Iancu
> 
> OpenSIPS Founder and Developer
>https://www.opensips-solutions.com
> OpenSIPS Bootcamp 5-16 Dec 2022, online
>https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
> 
> On 10/20/22 2:08 PM, Kingsley Tart wrote:
> > Hi,
> > 
> > Is it possible with OpenSIPS to append a body to a reply? If so, is
> > there also an equivalent of $ml for how long the outbound reply
> > would
> > be? (I appreciate that until send_reply() is called, the exact size
> > would be impossible to calculate).
> > 
> > I'm finding it useful to use custom INFO calls to share metadata
> > with
> > some of our other hosts. I could add the info in headers, but the
> > message body in some cases seems more appropriate.
> > 
> > Cheers,
> > Kingsley.
> > 
> > 
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 


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Re: [OpenSIPS-Users] - INVITE (SDP) includes Originators IP info

2022-10-20 Thread Nitesh Divecha

Bela

Much appreciated!

That fixed the "User-Agent" problem.

Cheers,
Nitesh


On Wed, Oct 19, 2022 at 8:03 PM Bela H  wrote:

> I hope this helps:
>
> https://opensips.org/docs/modules/3.2.x/sipmsgops.html#func_remove_hf
>
>
>
>
>
> *From: *Nitesh Divecha 
> *Sent: *Thursday, 20 October 2022 12:29
> *To: *OpenSIPS users mailling list 
> *Subject: *Re: [OpenSIPS-Users] - INVITE (SDP) includes Originators IP
> info
>
>
>
> Bela,
>
>
>
> Much appreciated!
>
>
>
> Changing topolgy_hiding("C"); fixed the Call-ID issue.
>
>
>
> Call-ID shows clean Call-ID:
> *DLGCH_W0xtTFgVXWleUV1fVgFvEiVSRVdabgccAltXbUFf*
>
>
>
> Now gotta figure out how to stop sending Originator User-Agent to outbound
> provider and how to configure rtpproxy behind NAT.
>
>
>
> Cheers,
>
> Nitesh
>
>
>
>
>
>
>
> On Wed, Oct 19, 2022 at 5:35 PM Bela H  wrote:
>
>
>
> For the first problem check this:
>
> *C* - Encode the callid header
>
> *Note:* Changing the callid of the call using the "C" flag is only
> available when doing topology_hiding with *dialog support*. Using this
> flag without dialog support will not change the callid at all!.
>
>
>
> *From: *Nitesh Divecha 
> *Sent: *Thursday, 20 October 2022 10:09
> *To: *Bogdan-Andrei Iancu 
> *Cc: *OpenSIPS users mailling list 
> *Subject: *Re: [OpenSIPS-Users] - INVITE (SDP) includes Originators IP
> info
>
>
>
> Hello All,
>
>
>
> So I had some success using topology_hiding and rtpproxy but found few
> problems.
>
>
>
> After implementing topology_hiding(), SIP INVITE was much better but still
> showing following:
>
>
>
> INVITE sip:aaabbbc...@outboundprovider.com:5060 SIP/2.0
>
> Call-ID: 4ed41738da10faa5@172.16.16.250 *<<<-- showing originators Device
> LAN IP —>>>*
>
> Content-Length: 329
> CSeq: 8002 INVITE
> From: ;tag=SP39b79130abfb7487f
> Max-Forwards: 69
> To: 
> Via: SIP/2.0/UDP 3.xxx.xxx.49:5060;branch=z9hG4bK1dcb.5bb78035.0
> User-Agent: OBIHAI/OBi302-3.2.2.6259 *<<<-- showing originators
> User-Agent —>>>*
> Contact: 
> Expires: 60
> Supported: replaces
> Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE
> Content-Type: application/sdp
>
> ===
>
> 1) How can I remove IP from Call-ID and rewrite Originators User-Agent to
> local OpenSIPS User-Agent?
>
> ===
>
>
>
>
>
> Now issue with rtpproxy - I'm running this OpenSIPS on AWS cloud... AWS
> cloud does natting by default, so my Public IP is 3.xxx.xxx.49 and actual
> VM IP is *172.31.29.47. *
>
>
>
> After implement rtpproxy (https://www.rtpproxy.org/), it is running on
> local IP:
>
> └─183589 /usr/local/bin/rtpproxy -s udp:172.31.29.47 2 -u rtpproxy
> rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 172.31.29.47 -m 1000 -M 2000
> -d INFO LOG_LOCAL5
>
> As it shows from SIP INVITE and due to that no audio or RTP because IP is
> not reachable...
>
>
>
> v=0
> o=- 16210664 1 IN IP4 *172.31.29.47 <<<-- OpenSIPS NAT IP —>>>*
> s=-
> c=IN IP4 *172.31.29.47 <<<-- OpenSIPS NAT IP —>>>*
> t=0 0
> m=audio 1958 RTP/AVP 0 8 18 104 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:104 G726-32/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
> a=ptime:20
> a=xg726bitorder:big-endian
> a=nortpproxy:yes
>
>
>
> ===
>
> 2. How can I configure rtpproxy with Public IP? Or do I start rtpproxy
> with Public IP 3.xxx.xxx.49 and reconfigure OpenSIPS with Public IP?
>
> modparam("rtpproxy", "rtpproxy_sock", "udp:172.31.29.47:2")
>
>
>
> Thanking in advance...
>
>
>
> Cheers,
>
> Nitesh
>
>
>
>
>
>
>
>
>
>
>
> On Wed, Oct 19, 2022 at 10:17 AM Nitesh Divecha <
> aviator.nites...@gmail.com> wrote:
>
> Hello,
>
>
>
> Thank y'all for the input... I will try to read the documentation and work
> on implementing these modules.
>
>
>
> By any chance do either of you have any working examples which I can refer
> to? I'm a work in progress and every time I change something I break
> OpenSIPS and it takes me hours to troubleshoot! :-)
>
>
>
> Thanking in advance...
>
>
>
> Cheers,
>
> Nitesh
>
>
>
>
>
>
>
> On Wed, Oct 19, 2022 at 2:20 AM Bogdan-Andrei Iancu 
> wrote:
>
> Hi there,
>
> Actually you do not need the B2B, you can achieve the same kind of privacy
> (at SIP level) with dialog module and topology_hiding module together.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
>
>
> OpenSIPS Founder and Developer
>
>   https://www.opensips-solutions.com
>
> OpenSIPS Bootcamp 5-16 Dec 2022, online
>
>   https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
>
> On 10/19/22 1:23 AM, Abdul Basit wrote:
>
> Nitesh,
>
>
>
> You need a B2BUA function
> 
>  with
> the help of a topo-hiding module with opensips as Bela shared in his email.
>
> Also, install the RTP proxy on the same opensips box (not necessary if you
> need separate signaling and media boxes).
>
>
>
> Far end party will not be able to see the A-party information.
>
>
>
> https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2
>

Re: [OpenSIPS-Users] - INVITE (SDP) includes Originators IP info

2022-10-20 Thread Nitesh Divecha

Hello,

After reading the rtpproxy documentation again, I was able to resolve the
rtpproxy NAT issue.

-A *advaddr1[/advaddr2]*

Set advertised address of rtpproxy. Useful if the rtpproxy is behind a NAT
firewall. (Amazon EC2) When the rtpproxy receives a session request from a
SIP controller it will return the IP address(es) specified by the -A option.

CGroup: /system.slice/rtpproxy.service

 └─247521 /usr/local/bin/rtpproxy -s udp:172.31.29.47 2 -u
rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -A 3.xxx.xxx.49 -l
172.31.29.47 -m 1000 -M 2000 -d INFO LOG_LOCAL5



Just for my understanding... What is the difference between rtpproxy and
Mediaproxy Module in OpenSIPS? Do I need both or can I achieve the same
with Mediaproxy? I have to monitor two services now (rtpproxy and OpenSIPS).

Cheers,
Nitesh



On Wed, Oct 19, 2022 at 5:06 PM Nitesh Divecha 
wrote:

> Hello All,
>
> So I had some success using topology_hiding and rtpproxy but found few
> problems.
>
> After implementing topology_hiding(), SIP INVITE was much better but still
> showing following:
>
> INVITE sip:aaabbbc...@outboundprovider.com:5060 SIP/2.0
> Call-ID: 4ed41738da10faa5@172.16.16.250 *<<<-- showing originators Device
> LAN IP —>>>*
> Content-Length: 329
> CSeq: 8002 INVITE
> From: ;tag=SP39b79130abfb7487f
> Max-Forwards: 69
> To: 
> Via: SIP/2.0/UDP 3.xxx.xxx.49:5060;branch=z9hG4bK1dcb.5bb78035.0
> User-Agent: OBIHAI/OBi302-3.2.2.6259 *<<<-- showing originators
> User-Agent —>>>*
> Contact: 
> Expires: 60
> Supported: replaces
> Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE
> Content-Type: application/sdp
>
> ===
> 1) How can I remove IP from Call-ID and rewrite Originators User-Agent to
> local OpenSIPS User-Agent?
> ===
>
>
> Now issue with rtpproxy - I'm running this OpenSIPS on AWS cloud... AWS
> cloud does natting by default, so my Public IP is 3.xxx.xxx.49 and actual
> VM IP is *172.31.29.47. *
>
> After implement rtpproxy (https://www.rtpproxy.org/), it is running on
> local IP:
> └─183589 /usr/local/bin/rtpproxy -s udp:172.31.29.47 2 -u rtpproxy
> rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 172.31.29.47 -m 1000 -M 2000
> -d INFO LOG_LOCAL5
>
> As it shows from SIP INVITE and due to that no audio or RTP because IP is
> not reachable...
>
> v=0
> o=- 16210664 1 IN IP4 *172.31.29.47 <<<-- OpenSIPS NAT IP —>>>*
> s=-
> c=IN IP4 *172.31.29.47 <<<-- OpenSIPS NAT IP —>>>*
> t=0 0
> m=audio 1958 RTP/AVP 0 8 18 104 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:104 G726-32/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
> a=ptime:20
> a=xg726bitorder:big-endian
> a=nortpproxy:yes
>
> ===
> 2. How can I configure rtpproxy with Public IP? Or do I start rtpproxy
> with Public IP 3.xxx.xxx.49 and reconfigure OpenSIPS with Public IP?
> modparam("rtpproxy", "rtpproxy_sock", "udp:172.31.29.47:2")
>
> Thanking in advance...
>
> Cheers,
> Nitesh
>
>
>
>
>
> On Wed, Oct 19, 2022 at 10:17 AM Nitesh Divecha <
> aviator.nites...@gmail.com> wrote:
>
>> Hello,
>>
>> Thank y'all for the input... I will try to read the documentation and
>> work on implementing these modules.
>>
>> By any chance do either of you have any working examples which I can
>> refer to? I'm a work in progress and every time I change something I break
>> OpenSIPS and it takes me hours to troubleshoot! :-)
>>
>> Thanking in advance...
>>
>> Cheers,
>> Nitesh
>>
>>
>>
>> On Wed, Oct 19, 2022 at 2:20 AM Bogdan-Andrei Iancu 
>> wrote:
>>
>>> Hi there,
>>>
>>> Actually you do not need the B2B, you can achieve the same kind of
>>> privacy (at SIP level) with dialog module and topology_hiding module
>>> together.
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>>
>>> OpenSIPS Founder and Developer
>>>   https://www.opensips-solutions.com
>>> OpenSIPS Bootcamp 5-16 Dec 2022, online
>>>   https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
>>>
>>> On 10/19/22 1:23 AM, Abdul Basit wrote:
>>>
>>> Nitesh,
>>>
>>> You need a B2BUA function
>>> 
>>>  with
>>> the help of a topo-hiding module with opensips as Bela shared in his email.
>>> Also, install the RTP proxy on the same opensips box (not necessary if
>>> you need separate signaling and media boxes).
>>>
>>> Far end party will not be able to see the A-party information.
>>>
>>> https://www.opensips.org/Documentation/Tutorials-B2BUA-3-2
>>>
>>> I hope this will help.
>>>
>>> --
>>> regards,
>>>
>>> abdul basit
>>>
>>> On Wed, 19 Oct 2022 at 03:14, Bela H  wrote:
>>>
 Hi Nitesh,



1. Check the topology hiding function:
https://opensips.org/docs/modules/3.2.x/topology_hiding.html
2. Use e.g. rtpproxy:


 https://opensips.org/docs/modules/3.2.x/rtpproxy.html#func_rtpproxy_offer


 http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_

Re: [OpenSIPS-Users] SIP user statuses to websocket

2022-10-20 Thread ideanet help

Hi Bogdan,

Thank you so much for the guidance. I have gone through the link and added
some custom events. I am able to subscribe and send them to an interface.
The document for jsonrpc event is broken. If you go to Transport Protocols
and click on event_jsonrpc
 it says Not
Found. can someone fix this link, please? I just want to look at the json
rpc events. Thank you.

Regards,
Jazzi




On Fri, Oct 21, 2022 at 1:47 AM Bogdan-Andrei Iancu 
wrote:

> Hi Anonymous,
>
> Check the events interface in OpenSIPS [1]. Different modules (see their
> documentation) may raise different events, like when a user is registered,
> when a call is placed, etc.
>
> [1] https://www.opensips.org/Documentation/Interface-Events-3-2
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
> OpenSIPS Bootcamp 5-16 Dec 2022, online
>   https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/
>
> On 10/19/22 9:32 AM, ideanet help wrote:
>
> Hi all,
>
> I am looking for a way to use WebSocket with opensips. Basically, I want
> to show my all sip users statuses/activity on a web page. I am currently
> using an API, which is called after a few seconds so this is a bit slow. I
> am looking for a way that opensips sends event to the socket and on the
> other hand, my webpage will be connected to a WebSocket server. This way I
> can show all the events instantly.
> For example, If a user receives a call I will have it on the web page
> straight away. if a user gets unregistered i will see on my page that users
> sent Expiry 0 something like that.
>
> Will it be possible with the proto_wss module. any pointers will help. I
> saw what proto_wss can do but I did not understand it properly. it will
> send traces to a server i don't need. I just need it if a user is busy or
> idle. or he gets unregistered that's all.
>
> Thanks
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
___
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[OpenSIPS-Users] can several SIP phones with same SIP account ring together and hang up asynchronous

2022-10-20 Thread ???? via Users

for example, phone A/B configured with the same SIP account.  when phone C 
call the sip account , phone A/B will ring at the same time. At the ringing 
state, when A hang up the call, C will be terminated by opensips. is there a 
way to let B still calling instead of terminated?___
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[OpenSIPS-Users] can several SIP phones with same SIP account ring together and hang up asynchronous

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