date:20221011

Re: [OpenSIPS-Users] Failed to trigger pkg stats in logs

2022-10-11 Thread Yury Kirsanov

Hi Bogdan,
No, after I removed all the autoscaling I'm not getting any
'core:signal_pkg_status' errors.

Regards,
Yury.

On Tue, Oct 11, 2022 at 10:56 PM Bogdan-Andrei Iancu 
wrote:

> Hi Yury,
>
> trying to recap here - after completely removing the auto-scaling, do you
> still have issues with fetching the pkg stats?
>
> 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 9/22/22 5:39 PM, Yury Kirsanov wrote:
>
> Hi Bogdan,
> Sorry, I just realized that I haven't turned off the auto-scaler for UDP
> processes so you're right - that error is for them. I've switched TCP
> auto-scaling off due to all my previous issues that I wasn't able to
> overcome.
>
> auto_scaling_profile = PROFILE_UDP
>  scale up to 32 on 80% for 4 cycles within 5
>  scale down to 2 on 10% for 5 cycles
>
> udp_workers=1 use_auto_scaling_profile PROFILE_UDP
>
> socket=udp:10.x.x.x:5060
> socket=udp:103.x.x.x:7060
>
> In regards to pkg memory - I only have 32MB as -M parameter, maybe that's
> not enough? I'm not sure how to do 'pkg dump' - never tried to do that
> before, is this what you're asking about?
>
> Sep 21 18:52:09 ERROR:core:signal_pkg_status: failed to trigger pkg stats
> for process 45
>  opensips-cli -x mi ps
> {
> "Processes": [
> ...
> {
> "ID": 45,
> "PID": 310115,
> "Type": "SIP receiver udp:103.x.x.x:7060"
> }
> ]
> }
> root@osgw1:~# opensips-cli -x mi mem_pkg_dump 310115
> ERROR: command 'mem_pkg_dump' returned: 500: Internal error
>
> Best regards,
> Yury.
>
>
>
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Re: [OpenSIPS-Users] dr_partitions reload

2022-10-11 Thread Marcin Groszek

Well, after some more testing I noticed that dr_partitions does not get 
reloaded with "opensips-cli -x mi dr_reload"


How can one reload the content of dr_partitions without restarting 
opensips process?



On 10/11/2022 10:19 AM, Marcin Groszek wrote:

opensips 3.1.5

opensips-cli -x mi dr_reload part_name  reloads the partition part_name

When new entry is added or removed from  dr_partitions table 
opensips-cli -x mi dr_reload is needed to reload the content of 
dr_partitions, but this also reloads all partitions.


Is there a way to do a dr_reload without all partitions, or perhaps 
reload only dr_partitions table?



--
Best Regards:
Marcin Groszek
Business Phone Service
https://www.voipplus.net


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[OpenSIPS-Users] dr_partitions reload

2022-10-11 Thread Marcin Groszek


opensips 3.1.5

opensips-cli -x mi dr_reload part_name  reloads the partition part_name

When new entry is added or removed from  dr_partitions table 
opensips-cli -x mi dr_reload is needed to reload the content of 
dr_partitions, but this also reloads all partitions.


Is there a way to do a dr_reload without all partitions, or perhaps 
reload only dr_partitions table?


--
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Marcin Groszek
Business Phone Service
https://www.voipplus.net


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Re: [OpenSIPS-Users] Problem proxying a SIP connection with t_relay

2022-10-11 Thread Thomas Pircher via Users


Thomas Pircher via Users wrote:

sure, please find the trace attached. This was captured using
sudo tcpdump -lnn -i any udp port 5060 -w opensips-any.pcap
i.e. it contains both OpenSIPS interfaces in one file:
- ens4, 10.30.8.201, external
- ens5, 10.30.9.10, internal

The other IPs are:
- 10.30.8.204: sipp uac
- 10.30.9.11: sipp uas


This time with actual attachment; sorry...

Thomas


opensips-any.pcap
Description: application/vnd.tcpdump.pcap
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Re: [OpenSIPS-Users] Problem proxying a SIP connection with t_relay

2022-10-11 Thread Thomas Pircher via Users


Bogdan-Andrei Iancu wrote:
Could you post a pcap or ngrep capture of the failing call (with the 
broken ACK) - PM me if info is too sensitive.


Hi Bogdan-Andrei,

sure, please find the trace attached. This was captured using
sudo tcpdump -lnn -i any udp port 5060 -w opensips-any.pcap
i.e. it contains both OpenSIPS interfaces in one file:
- ens4, 10.30.8.201, external
- ens5, 10.30.9.10, internal

The other IPs are:
- 10.30.8.204: sipp uac
- 10.30.9.11: sipp uas

Thanks,
Thomas

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Re: [OpenSIPS-Users] Creating branches inside a while loop

2022-10-11 Thread mayamatakeshi

Hi Bogdan,
yes. It works.
I am using it like this:





















*$var(idx) = 0;$var(aor) =
$(var(aors){s.select,$var(idx),,});while($var(aor)
!= null) {if($var(idx) > 0) {
append_branch();}
seturi("sip:$var(aor)@$fd");$var(idx) = $var(idx) + 1;
  $var(aor) = $(var(aors){s.select,$var(idx),!});}
  if (lookup("location", "r")) {
t_on_branch("USERAGENT_BRANCH");route(RELAY_REQUEST);
  } else {sl_send_reply(480, "subscriber not online
retcode=$retcode");}*

Thanks and regards,
Takeshi

On Tue, Oct 11, 2022 at 6:32 PM Bogdan-Andrei Iancu 
wrote:

> Hi, and it works now, right?
>
> 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 10/11/22 10:29 AM, mayamatakeshi wrote:
>
> Hi Bogdan,
> indeed, I think I removed the append_branch() line by mistake when
> removing some debug lines I added to check function return code etc.
> Thanks,
> Takeshi
>
> On Mon, Oct 10, 2022 at 9:39 PM Bogdan-Andrei Iancu 
> wrote:
>
>> Hi Takeshi,
>>
>> Aren't you missing the "seturi()" + "append_branch()" in the loop ??
>>
>> 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 9/28/22 4:26 PM, mayamatakeshi wrote:
>>
>>
>> On Wed, Sep 28, 2022 at 2:21 PM mayamatakeshi 
>> wrote:
>>
>>> Hi,
>>> I'm testing latest commit b243666098be44226ade6a7df2b62851efcb5de8 of
>>> opensips-3.2.
>>>
>>> I tested adding branches to an INVITE for a fixed size list of AORs this
>>> way:
>>>
>>> $var(aors) = "sip:us...@test1.com,sip:us...@test1.com,
>>> sip:us...@test1.com";
>>>
>>> seturi($(var(aors){s.select,0,,}));
>>>
>>> append_branch();
>>> seturi($(var(aors){s.select,1,,}));
>>>
>>> append_branch();
>>> seturi($(var(aors){s.select,2,,}));
>>>
>>> lookup("location", "r")
>>>
>>> The above works fine and all 3 destinations resolved by AOR lookup are
>>> called (max of contact per AOR).
>>>
>>> However, in case of a a list of unknown size, I tried to use a while
>>> loop like this:
>>> $var(aors) = "sip:us...@test1.com,sip:us...@test1.com,
>>> sip:us...@test1.com";
>>>
>>> $var(idx) = 0;
>>> $var(aor) = $(var(aors){s.select,$var(idx),,});
>>>
>>> while($var(aor) != null) {
>>> seturi($var(aor));
>>>
>>> $var(idx) = $var(idx) + 1;
>>> $var(aor) = $(var(aors){s.select,$var(idx),,});
>>> }
>>>
>>> lookup("location", "r")
>>>
>>> But with the above, only the last destination (lookup of us...@test1.com)
>>> is called.
>>> I confirmed this is not related to the lookup function because I tried
>>> with fixed destinations like this:
>>>
>>> $var(aors) = "sip:user1@10.0.0.1:5072,
>>> sip:user2@10.0.0.1:5074,sip:user3@10.0.0.1:5076";
>>>
>>> $var(idx) = 0;
>>> $var(aor) = $(var(aors){s.select,$var(idx),,});
>>>
>>> while($var(aor) != null) {
>>> seturi($var(aor));
>>>
>>> $var(idx) = $var(idx) + 1;
>>> $var(aor) = $(var(aors){s.select,$var(idx),,});
>>> }
>>>
>>> and the same problem happens: only the last destination
>>> sip:user3@10.0.0.1:5076 is called.
>>>
>>> So, is there a way to append a non-fixed number of branches to an INVITE?
>>>
>>> Regards,
>>> Takeshi
>>>
>>
>> Sorry, I think I did something wrong.
>> I was able to make append_branch to work inside a while loop.
>> So there is no problem.
>> Regards,
>> Takeshi
>>
>>
>> ___
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>>
>>
>>
>
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Re: [OpenSIPS-Users] Expires value - [ZVP-JQSVP-142]

2022-10-11 Thread Artem Fomenko via Users

Yes, it is right.

Regards,

![LiveAgent](https://www.qualityunit.com/mail/mail-logo-la.png) **Artem 
Fomenko**

**Development Team**

+421 2 33 456 826 (EU  Worldwide)

+1-888-257-8754 (USA  Canada)

[www.liveagent.com](https://www.liveagent.com/)

  

**How 
nice was my reply?**

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-Original message-
From: Bogdan-Andrei Iancu bog...@opensips.org
Sent: 2022-10-11 12:37:40



> Hi Artem,
>
>  I see - so at each re-registration cycle you loose 1 secs, dropping from the 
> 120 provisioned all the way the minimum of 29, right ?
>
>  Regards,
>  ```
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com; 
> rel="noreferrer" target="_blank">https://www.opensips-solutions.com
> OpenSIPS Summit 27-30 Sept 2022, Athens
>href="https://www.opensips.org/events/Summit-2022Athens/; rel="noreferrer" 
> target="_blank">https://www.opensips.org/events/Summit-2022Athens/
> ```
>
> On 10/11/22 11:23 AM, Artem Fomenko wrote:
> 

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Re: [OpenSIPS-Users] b2bua top hiding behind NAT

2022-10-11 Thread Bogdan-Andrei Iancu


Hi Alex,

Could you detail the prev UPDATE message (prev frame) that loops on 
udp:10.130.23:5070 - that is the UPDATE as generated by OpenSIPS, right ?


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 9/21/22 5:07 PM, Alex wrote:

Hello,

Not only do I need force messages from internet to lan through 
udp:10.130.23:5070 but also I need send messages from lan 
(udp:10.130.23:5070) to internet (udp:10.130.23:5060). So I make rule 
with if and $socket_in .
But Messages (Updates, reInvites) and replies go directly between LAN 
and 10.130.23:5060 thought.
All messages (input, output, replies) must go between two sockets 
10.130.23:5070 <-> 10.130.23:5060 .
    b2b_server_new("server1",$avp(b2b_hdrs), 
$avp(b2b_hdr_bodies));
                if ($socket_in == "udp:10.130.0.23:5070 
") {

force_send_socket("udp:10.130.23:5060");
                } else {
force_send_socket("udp:10.130.23:5070");
                }
                b2b_client_new("Unistar","sip:09876543...@provider.com 
","sip:provider","test","sip:1234567@1.1.1.1 
");

                #force_send_socket("udp:10.130.23:5070");
                b2b_init_request("top hiding");
                exit;

But Updates go from 5060 directly to lan. I made a screenshot you can 
see by the url 
https://drive.google.com/file/d/1bpwmXCB6qRxbDk8KZ6GuCg3-hwfABvwk/view?usp=sharing 







вт, 23 авг. 2022 г. в 12:44, Bogdan-Andrei Iancu >:


Hi Alex,

Have you tried something like this (for calls from Internet) :

b2b_server_new("server1",$avp(b2b_hdrs), $avp(b2b_hdr_bodies));
force_send_socket("udp:10.130.23:5070");
b2b_client_new("Unistar","sip:09876543...@provider.com

","sip:provider","test","sip:1234567@1.1.1.1
");
b2b_init_request("top hiding");

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 8/19/22 6:12 PM, Alex wrote:

Hello,

My opensips is behind NAT in cloud
internet - (1.1.1.1)Cone_NAT -
port_5060__10.130.0.23(opensips)__port_5070 -
main_registrar_sip_server

So I made 2 sockets
socket=udp:10.130.0.23:5070    # for LAN
socket=udp:10.130.0.23:5060  as
1.1.1.1:5060  #for Internet

I try to make call from lan to Tel provider with using b2bua.

Is any way to setup b2bua with more than one socket?

loadmodule "b2b_entities.so"
loadmodule "b2b_logic.so"
modparam("b2b_logic", "custom_headers", "P-Asserted-Identity")
#"User-Agent;Date")
#modparam("b2b_logic", "contact_user", 1)
modparam("b2b_logic", "server_address", "sip:1234567@1.1.1.1
")


#force_send_socket("udp:10.130.23:5060");
b2b_server_new("server1",$avp(b2b_hdrs), $avp(b2b_hdr_bodies));
force_send_socket("udp:10.130.23:5060");
b2b_client_new("Unistar","sip:09876543...@provider.com

","sip:provider","test","sip:1234567@1.1.1.1
");
#force_send_socket("udp:10.130.23:5070");
b2b_init_request("top hiding");
exit;

I use this construction in route[relay]. And I`ve tried to insert
in before b2b_server_new. But  it didn`t help
if ($socket_in == "udp:10.130.0.23:5070 ") {
                $socket_out = "udp:10.130.0.23:5060
";
        } else {
                $socket_out = "udp:10.130.0.23:5070
";
        }

1) with force_send_socket I made to send requests from :5070 to
:5060 . But my provider`s Udpates and re-Invites go to my LAN
directly from socket :5060 . They don`t go  through :5070
2) I can`t change contact in request to provider. Default contact
looks like this Contact: . I whant to change to
"sip:1234567@1.1.1.1 "

-- 
С уважением,

Якимкин Алексей

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--
С уважением,
Якимкин Алексей


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Re: [OpenSIPS-Users] OpenSIPS 3.2.7 tracer module for sip dialogs leads to an endless loop

2022-10-11 Thread Bogdan-Andrei Iancu


Hi PAvel,

How do you define the tid you use for tracing ?

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 9/21/22 7:45 PM, Pavel Ekshin wrote:

Hi Bogdan,

Thanks for the answer.
I tried this module, but nothing weird was found. On reply ACK 
messages in sngrep capture I see the correct "To/Contact'' header, but 
in log the "To" header looks different, but it's not lead to any loop.


Sep 20 16:51:55 openSIPS /usr/sbin/opensips[15619]: [Script 
Trace][/etc/opensips/opensips_residential.cfg:430][script_trace][core 
if] -> (INVITE from 172.18.53.131, ruri=sip:4002@172.16.34.173:5060 
, 
contact=>)
Sep 20 16:51:55 openSIPS /usr/sbin/opensips[15619]: [Script 
Trace][/etc/opensips/opensips_residential.cfg:427][script_trace][module 
t_relay] -> (INVITE from 172.18.53.131, 
ruri=sip:4002@172.16.34.173:5060 , 
contact=>)
Sep 20 16:51:55 openSIPS /usr/sbin/opensips[15619]: [Script 
Trace][/etc/opensips/opensips_residential.cfg:430][script_trace][core 
exit] -> (INVITE from 172.18.53.131, ruri=sip:4002@172.16.34.173:5060 
, 
contact=>)
Sep 20 16:51:55 openSIPS /usr/sbin/opensips[15619]: [Script 
Trace][/etc/opensips/opensips_residential.cfg:221][script_trace][core 
if] -> (ACK from 172.18.53.131, 
ruri=sip:172.16.34.173:5060;transport=udp, 
contact=>)
Sep 20 16:51:55 openSIPS /usr/sbin/opensips[15619]: [Script 
Trace][/etc/opensips/opensips_residential.cfg:214][script_trace][module 
mf_process_maxfwd_header] -> (ACK from 172.18.53.131, 
ruri=sip:172.16.34.173:5060;transport=udp, 
contact=>)
Sep 20 16:51:55 openSIPS /usr/sbin/opensips[15619]: [Script 
Trace][/etc/opensips/opensips_residential.cfg:258][script_trace][core 
if] -> (ACK from 172.18.53.131, 
ruri=sip:172.16.34.173:5060;transport=udp, 
contact=>)
Sep 20 16:51:55 openSIPS /usr/sbin/opensips[15619]: [Script 
Trace][/etc/opensips/opensips_residential.cfg:221][script_trace][module 
has_totag] -> (ACK from 172.18.53.131, 
ruri=sip:172.16.34.173:5060;transport=udp, 
contact=>)



172.18.53.131:5060___172.16.34.91:5060__172.16.34.173:5060___172.16.34.173:64087 


__qqwq__qqwq__qqwq__qqwqx
__16:06:48.772459___xINVITE_(SDP)_x_x_x_
+0.001900___x_qq>_x_x_x_
__16:06:48.774359___x__407_Proxy_Authentication_R_x_x_x_
+0.014197___x__x_x_x_
__16:06:48.788659___xINVITE_(SDP)_x_x_x_
+0.005221___x_qq>_x_x_x_
__16:06:48.793880___x_100_Giving_it_a_try_x_x_x_
+0.000664___x__x_x_
__16:06:48.799099___x_x100_Trying_x_
+0.039449___x_x__x_x_x_
__16:06:48.851924___x_x_ACK_x_x_
+8.768408___x_x_qq>_x_x_
__16:06:57.620332___x_BYE_x_x_x_
+0.000966___x_qq>_x_x_x_
__16:06:57.621298___x_x_BYE_x_x_
+0.003895___x_x_qq>_x_x
__16:06:57.625193___x_x__200_Ok___x_
+0.000271___x_x_If I disable trace("tid", "d/t/m", "sip") module at route level, I 
never faced

Re: [OpenSIPS-Users] Failed to trigger pkg stats in logs

2022-10-11 Thread Bogdan-Andrei Iancu


Hi Yury,

trying to recap here - after completely removing the auto-scaling, do 
you still have issues with fetching the pkg stats?


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 9/22/22 5:39 PM, Yury Kirsanov wrote:

Hi Bogdan,
Sorry, I just realized that I haven't turned off the auto-scaler for 
UDP processes so you're right - that error is for them. I've switched 
TCP auto-scaling off due to all my previous issues that I wasn't able 
to overcome.


auto_scaling_profile = PROFILE_UDP
     scale up to 32 on 80% for 4 cycles within 5
     scale down to 2 on 10% for 5 cycles

udp_workers=1 use_auto_scaling_profile PROFILE_UDP

socket=udp:10.x.x.x:5060
socket=udp:103.x.x.x:7060

In regards to pkg memory - I only have 32MB as -M parameter, maybe 
that's not enough? I'm not sure how to do 'pkg dump' - never tried to 
do that before, is this what you're asking about?


Sep 21 18:52:09 ERROR:core:signal_pkg_status: failed to trigger pkg 
stats for process 45

 opensips-cli -x mi ps
{
    "Processes": [
...
        {
  "ID": 45,
            "PID": 310115,
            "Type": "SIP receiver udp:103.x.x.x:7060"
        }
    ]
}
root@osgw1:~# opensips-cli -x mi mem_pkg_dump 310115
ERROR: command 'mem_pkg_dump' returned: 500: Internal error

Best regards,
Yury.


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Re: [OpenSIPS-Users] Problem proxying a SIP connection with t_relay

2022-10-11 Thread Bogdan-Andrei Iancu


Hi Thomas,

Could you post a pcap or ngrep capture of the failing call (with the 
broken ACK) - PM me if info is too sensitive.


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 10/11/22 12:06 PM, Thomas Pircher via Users wrote:

Bogdan-Andrei Iancu wrote:

Hi Thomas,

Your handling of sequential requests is broken, see here for a 
correct sample:


https://github.com/OpenSIPS/opensips/blob/master/etc/opensips.cfg#L109


Hi Bogdan-Andrei,

thanks for your reply. I had a look at my config (both the config
attached to the first mail, and the fragment in my second mail) and it
does match -- modulo whitespace changes and one additional
route[byNumber] -- the config in git.

The bracket placement in the config fragment in my second mail was
misleading, I made a copy error, that might have created a
confusion? Or is there something I continue missing when I read and
compare the config files?

Thanks,
Thomas

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Re: [OpenSIPS-Users] Expires value - [ZVP-JQSVP-142]

2022-10-11 Thread Bogdan-Andrei Iancu


Hi Artem,

I see - so at each re-registration cycle you loose 1 secs, dropping from 
the 120 provisioned all the way the minimum of 29, right ?


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 10/11/22 11:23 AM, Artem Fomenko wrote:


Hello Bogdan-Andrei,

I have a conversation with VoIP provider, he decrease "expires" time 
by 1 sec on each registration from 120 sec (our expiry value) to 29 
sec (according 30 sec - minimal provider's value). So registration 
become to process too often. It is not make any problems for now. The 
provider motivates its behavior by the following:


"

In fact, this is the time left to the end of the registration,
which is indicated in the "expires" parameter sent in our 200 OK.
"The time left to the end of the registration" was my
interpretation, but I think it's a reasonable one.  When your side
requests "expires=120", our side honors this request, but due to
the actual record in the location registrar occurring milliseconds
later, the required "expires" time becomes 119.

Imagine that a bigger delay happens between receiving INVITE and
the actual recording to the location registrar, let's say about 2
seconds. In this case, if 200 OK was generated with "expires=120"
and sent 2 seconds later after receiving the first INVITE, your
side would think that the registration ends 2 seconds later than
actually it was requested.

"

this is not a violation of RFC3261 Section 10.3 step 8:

"
The registrar returns a 200 (OK) response.  The response MUST contain Contact header 
field values enumerating all current bindings.  Each Contact value MUST feature an 
"expires" parameter indicating its expiration interval chosen by the registrar.

"

But in my opinion, the interpretation of the provider is not entirely 
correct. Despite this, I would like to know your opinion on this issue.


Regards,

LiveAgent   

*Artem Fomenko*

*Development Team*

+421 2 33 456 826 (EU & Worldwide)

+1-888-257-8754 (USA & Canada)

www.liveagent.com 



*How nice was my reply?*

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Re: [OpenSIPS-Users] early dialog termination

2022-10-11 Thread Bogdan-Andrei Iancu


Hi Ivan,

you can use timer_route, but as there is no way to send a reply for a 
particular transaction from script level (only to the currently 
processed request), you will have to trigger the MI cmds from the timer 
route, which is a bit hackish 


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 10/11/22 11:47 AM, Ryzhik Ivan wrote:

Hi, Bogdan!
What d' you think, can we use timer_route instead of an external script?
Regards, Ivan.

пн, 10 окт. 2022 г. в 17:04, Bogdan-Andrei Iancu >:


Hi Ryzhik,

Without a t_relay() it makes not much sense to have an dialog
structure at all - the dialog module in opensips is actually
design for proxied calls, not for UAC calls.

IMO, you should keep it a transaction level, by sending replies
back only. When getting the INVITE, put its call-id into a DB
table (to keep only the "active" session) together with a lifetime
/ expiration time. When getting a CANCEL, update the table (set
lifetime to 0), to know it is terminated. And use an simple
external script that keeps scanning the DB for (1) sending 487
Terminated via MI if the record has 0 lifetime or (2) send a 408
Timeout via MI if the lifetime exceeded.
In a similar way you can handle the BYE - send back 200OK for the
BYE and set 0 in lifetime, to send a 487 canceled back

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 10/10/22 4:33 PM, Ryzhik Ivan wrote:

Hello!
My opensips version is 3.1 with tm,dialog and rtpengine modules.
On incoming INVITE i'm creating an early dialog with 183 replies
and i'm playing audio to caller with rtpengine, no t_relay() on
this step, OS is acting as UAS endpoint.
If the caller cancels the invite with a CANCEL message - all
works great.
But some users terminate dialog with BYE message.
1) on BYE with to-tag OS can't find dialog with match_dialog(),
because to-tag presents.
2) if i use load_dialog_ctx($ci) -  it is possible to handle BYE.
3) in early dialog termination with BYE we also need to send
final response to the INVITE transaction.

Maybe I did something wrong, but I can't handle the final
response to INVITE in this case.

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Re: [OpenSIPS-Users] Formating issue for string that contains pv

2022-10-11 Thread Serdar GUCLUER


Hi Liviu,

I tried /s.eval/ transformation for opensips 3.3.1 and it worked, thanks 
for your helping.


Serdar



Hi Serdar,

I remember that at some point, there was some support for string 
evaluation added as part of the {s.eval} transformation[1].  
Unfortunately, I don't recall an equivalent function, so you may have 
to play around with the transformation only.


[1]: https://www.opensips.org/Documentation/Script-Tran-3-4#s.eval

Best regards,



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Re: [OpenSIPS-Users] Expires value - [ZVP-JQSVP-142]

2022-10-11 Thread Artem Fomenko via Users

Hello Bogdan-Andrei,

I have a conversation with VoIP provider, he decrease "expires" time by 1 sec 
on each registration from 120 sec (our expiry value) to 29 sec (according 30 
sec - minimal provider's value). So registration become to process too often. 
It is not make any problems for now. The provider motivates its behavior by the 
following:

> "
>
> In fact, this is the time left to the end of the registration, which is 
> indicated in the "expires" parameter sent in our 200 OK. "The time left to 
> the end of the registration" was my interpretation, but I think it's a 
> reasonable one. When your side requests "expires=120", our side honors this 
> request, but due to the actual record in the location registrar occurring 
> milliseconds later, the required "expires" time becomes 119.
>
> Imagine that a bigger delay happens between receiving INVITE and the actual 
> recording to the location registrar, let's say about 2 seconds. In this case, 
> if 200 OK was generated with "expires=120" and sent 2 seconds later after 
> receiving the first INVITE, your side would think that the registration ends 
> 2 seconds later than actually it was requested.
>
> "

this is not a violation of RFC3261 Section 10.3 step 8:

> ```
>
> "
> The registrar returns a 200 (OK) response.  The response MUST contain Contact 
> header field values enumerating all current bindings.  Each Contact value 
> MUST feature an "expires" parameter indicating its expiration interval chosen 
> by the registrar.
> ```
>
> "

But in my opinion, the interpretation of the provider is not entirely correct. 
Despite this, I would like to know your opinion on this issue.

Regards,

![LiveAgent](https://www.qualityunit.com/mail/mail-logo-la.png) **Artem 
Fomenko**

**Development Team**

+421 2 33 456 826 (EU  Worldwide)

+1-888-257-8754 (USA  Canada)

[www.liveagent.com](https://www.liveagent.com/)

  

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-Original message-
From: Bogdan-Andrei Iancu bog...@opensips.org
Sent: 2022-10-10 16:50:42



> Hi Artem,
>
>  This is the intended behavior of the module, re-register with the last 
> "expires" provided by the server - if the server had a good reason to lower 
> the registration time on first register, it will most probably do it for the 
> re-register's also. So does not make too much of a sense to keep using the 
> initial DB value - or do you have some good reason not to update ?
>
>  Regards,
>  ```
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com; 
> rel="noreferrer" target="_blank">https://www.opensips-solutions.com
> OpenSIPS Summit 27-30 Sept 2022, Athens
>href="https://www.opensips.org/events/Summit-2022Athens/; rel="noreferrer" 
> target="_blank">https://www.opensips.org/events/Summit-2022Athens/
> ```
>
> On 9/23/22 7:15 PM, Artem Fomenko via Users 
> wrote:
> 

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Re: [OpenSIPS-Users] Creating branches inside a while loop

2022-10-11 Thread Bogdan-Andrei Iancu


Hi, and it works now, right?

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 10/11/22 10:29 AM, mayamatakeshi wrote:

Hi Bogdan,
indeed, I think I removed the append_branch() line by mistake when 
removing some debug lines I added to check function return code etc.

Thanks,
Takeshi

On Mon, Oct 10, 2022 at 9:39 PM Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


Hi Takeshi,

Aren't you missing the "seturi()" + "append_branch()" in the loop ??

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 9/28/22 4:26 PM, mayamatakeshi wrote:


On Wed, Sep 28, 2022 at 2:21 PM mayamatakeshi
mailto:mayamatake...@gmail.com>> wrote:

Hi,
I'm testing latest
commit b243666098be44226ade6a7df2b62851efcb5de8 of opensips-3.2.

I tested adding branches to an INVITE for a fixed size list
of AORs this way:

            $var(aors) = "sip:us...@test1.com
,sip:us...@test1.com
,sip:us...@test1.com
";

            seturi($(var(aors){s.select,0,,}));

            append_branch();
            seturi($(var(aors){s.select,1,,}));

            append_branch();
            seturi($(var(aors){s.select,2,,}));

            lookup("location", "r")

The above works fine and all 3 destinations resolved by AOR
lookup are called (max of contact per AOR).

However, in case of a a list of unknown size, I tried to use
a while loop like this:
            $var(aors) = "sip:us...@test1.com
,sip:us...@test1.com
,sip:us...@test1.com
";

            $var(idx) = 0;
            $var(aor) = $(var(aors){s.select,$var(idx),,});

            while($var(aor) != null) {
                seturi($var(aor));

                $var(idx) = $var(idx) + 1;
                $var(aor) = $(var(aors){s.select,$var(idx),,});
            }

            lookup("location", "r")

But with the above, only the last destination (lookup of
us...@test1.com ) is called.
I confirmed this is not related to the lookup function
because I tried with fixed destinations like this:

            $var(aors) = "sip:user1@10.0.0.1:5072
,sip:user2@10.0.0.1:5074
,sip:user3@10.0.0.1:5076
";

            $var(idx) = 0;
            $var(aor) = $(var(aors){s.select,$var(idx),,});

            while($var(aor) != null) {
                seturi($var(aor));

                $var(idx) = $var(idx) + 1;
                $var(aor) = $(var(aors){s.select,$var(idx),,});
            }

and the same problem happens: only the last destination
sip:user3@10.0.0.1:5076  is
called.

So, is there a way to append a non-fixed number of branches
to an INVITE?

Regards,
Takeshi


Sorry, I think I did something wrong.
I was able to make append_branch to work inside a while loop.
So there is no problem.
Regards,
Takeshi


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Re: [OpenSIPS-Users] Problem proxying a SIP connection with t_relay

2022-10-11 Thread Thomas Pircher via Users


Bogdan-Andrei Iancu wrote:

Hi Thomas,

Your handling of sequential requests is broken, see here for a correct 
sample:


https://github.com/OpenSIPS/opensips/blob/master/etc/opensips.cfg#L109


Hi Bogdan-Andrei,

thanks for your reply. I had a look at my config (both the config
attached to the first mail, and the fragment in my second mail) and it
does match -- modulo whitespace changes and one additional
route[byNumber] -- the config in git.

The bracket placement in the config fragment in my second mail was
misleading, I made a copy error, that might have created a
confusion? Or is there something I continue missing when I read and
compare the config files?

Thanks,
Thomas

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Re: [OpenSIPS-Users] Problem proxying a SIP connection with t_relay and rtpproxy

2022-10-11 Thread Thomas Pircher via Users


John Quick wrote:

ACK messages are normally loose routed. Perhaps you need to call
loose_route() before t_relay().
You could try reading my article here which may help explain things:
https://kb.smartvox.co.uk/opensips/contact-and-record-route-headers-explained/


Hi John,

thanks for the reply and sorry for the late response. Your articles are
quite informative, they made a few bits clearer.
I tried tracing the sequence diagram from your page on the opensips.cfg
file attached in my first post, and I believe all the necessary bits are
there -- I have taken the residential configuration from the config
generator and added an additiona "route[byNumber]" in the flow.

Unless I'm missing something when looking at the config file, I do
believe my config does match the flow you described. However, I have not
looked at the headers in the trace in detail, perhaps that's where my
problem lies?

Thanks,
Thomas

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Re: [OpenSIPS-Users] Request-Disposition: no-fork

2022-10-11 Thread Bogdan-Andrei Iancu


That's only for BLF (dialog/info presence), nothing to do with the calling.

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 10/10/22 10:13 PM, Bela H wrote:


Or is the dialoginfo_set_branch_callee(callee) function the key here?

*From: *Bela H 
*Sent: *Tuesday, 11 October 2022 08:09
*To: *Bogdan-Andrei Iancu ; OpenSIPS users 
mailling list 

*Subject: *Re: [OpenSIPS-Users] Request-Disposition: no-fork

Thanks Bogdan!

However, I am talking about serial forking, call forwarding busy/no 
answer scenario.


Is there a way to avoid that in the cfg without messing up with the to 
tags?


How do I achieve “proxy to only a single address ("no-fork")”?

According to fork-directive in 
https://www.rfc-editor.org/rfc/rfc3841#section-9.1.


Cheers,

Bela

*From: *Bogdan-Andrei Iancu 
*Sent: *Tuesday, 11 October 2022 01:49
*To: *OpenSIPS users mailling list ; 
Bela H 

*Subject: *Re: [OpenSIPS-Users] Request-Disposition: no-fork

Hi Bela,

What you are trying to do (messing with the TO-tags) is a bad idea, as 
you will be breaking the upstream parallel forking.


If the GW does not support forking, what you can do is to avoid doing 
parallel forking in your cfg (like when routing to users via lookup). 
You do not need any special support.


Best 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 9/29/22 7:10 AM, Bela H wrote:

Hello,

I have call forwarding busy/no answer scenario: A number is from a
gateway, B and C numbers are our own subs.

The gateway is sending us the INVITE message with
“Request-Disposition: no-fork” header field.

That means we must use one dialog for the mentioned scenario.

Currently the To tag we are sending to the GW in the first 180
ringing/181 Call is being forwarded messages are different to the
To tag in the second 180 ringing and 200 OK (SDP).

Gateway      OpenSips

      INVITE

-->

100 GIVING IT A TRY

<-- -

  180 RINGING

<- ---

181 CALL IS BEING FORWARDED

<- ---

  180 RINGING

<- ---

  200 OK (SDP)

<- ---

What would be the easiest way from OpenSIPS to send the same To
tag (it should be the same from the first 180 ringing through to
the 200 OK) and using one dialog for this scenario?

Cheers,

Bela

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Re: [OpenSIPS-Users] Request-Disposition: no-fork

2022-10-11 Thread Bogdan-Andrei Iancu


Hi Bela,

If the directive is indicated in the INVITE, simply avoid doing any 
forking in your cfg, nothing more.


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 10/10/22 10:06 PM, Bela H wrote:


Thanks Bogdan!

However, I am talking about serial forking, call forwarding busy/no 
answer scenario.


Is there a way to avoid that in the cfg without messing up with the to 
tags?


How do I achieve “proxy to only a single address ("no-fork")”?

According to fork-directive in 
https://www.rfc-editor.org/rfc/rfc3841#section-9.1.


Cheers,

Bela

*From: *Bogdan-Andrei Iancu 
*Sent: *Tuesday, 11 October 2022 01:49
*To: *OpenSIPS users mailling list ; 
Bela H 

*Subject: *Re: [OpenSIPS-Users] Request-Disposition: no-fork

Hi Bela,

What you are trying to do (messing with the TO-tags) is a bad idea, as 
you will be breaking the upstream parallel forking.


If the GW does not support forking, what you can do is to avoid doing 
parallel forking in your cfg (like when routing to users via lookup). 
You do not need any special support.


Best 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 9/29/22 7:10 AM, Bela H wrote:

Hello,

I have call forwarding busy/no answer scenario: A number is from a
gateway, B and C numbers are our own subs.

The gateway is sending us the INVITE message with
“Request-Disposition: no-fork” header field.

That means we must use one dialog for the mentioned scenario.

Currently the To tag we are sending to the GW in the first 180
ringing/181 Call is being forwarded messages are different to the
To tag in the second 180 ringing and 200 OK (SDP).

Gateway      OpenSips

      INVITE

-->

100 GIVING IT A TRY

<-- -

  180 RINGING

<- ---

181 CALL IS BEING FORWARDED

<- ---

  180 RINGING

<- ---

  200 OK (SDP)

<- ---

What would be the easiest way from OpenSIPS to send the same To
tag (it should be the same from the first 180 ringing through to
the 200 OK) and using one dialog for this scenario?

Cheers,

Bela



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Re: [OpenSIPS-Users] Creating branches inside a while loop

2022-10-11 Thread mayamatakeshi

Hi Bogdan,
indeed, I think I removed the append_branch() line by mistake when removing
some debug lines I added to check function return code etc.
Thanks,
Takeshi

On Mon, Oct 10, 2022 at 9:39 PM Bogdan-Andrei Iancu 
wrote:

> Hi Takeshi,
>
> Aren't you missing the "seturi()" + "append_branch()" in the loop ??
>
> 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 9/28/22 4:26 PM, mayamatakeshi wrote:
>
>
> On Wed, Sep 28, 2022 at 2:21 PM mayamatakeshi 
> wrote:
>
>> Hi,
>> I'm testing latest commit b243666098be44226ade6a7df2b62851efcb5de8 of
>> opensips-3.2.
>>
>> I tested adding branches to an INVITE for a fixed size list of AORs this
>> way:
>>
>> $var(aors) = "sip:us...@test1.com,sip:us...@test1.com,
>> sip:us...@test1.com";
>>
>> seturi($(var(aors){s.select,0,,}));
>>
>> append_branch();
>> seturi($(var(aors){s.select,1,,}));
>>
>> append_branch();
>> seturi($(var(aors){s.select,2,,}));
>>
>> lookup("location", "r")
>>
>> The above works fine and all 3 destinations resolved by AOR lookup are
>> called (max of contact per AOR).
>>
>> However, in case of a a list of unknown size, I tried to use a while loop
>> like this:
>> $var(aors) = "sip:us...@test1.com,sip:us...@test1.com,
>> sip:us...@test1.com";
>>
>> $var(idx) = 0;
>> $var(aor) = $(var(aors){s.select,$var(idx),,});
>>
>> while($var(aor) != null) {
>> seturi($var(aor));
>>
>> $var(idx) = $var(idx) + 1;
>> $var(aor) = $(var(aors){s.select,$var(idx),,});
>> }
>>
>> lookup("location", "r")
>>
>> But with the above, only the last destination (lookup of us...@test1.com)
>> is called.
>> I confirmed this is not related to the lookup function because I tried
>> with fixed destinations like this:
>>
>> $var(aors) = "sip:user1@10.0.0.1:5072,sip:user2@10.0.0.1:5074
>> ,sip:user3@10.0.0.1:5076";
>>
>> $var(idx) = 0;
>> $var(aor) = $(var(aors){s.select,$var(idx),,});
>>
>> while($var(aor) != null) {
>> seturi($var(aor));
>>
>> $var(idx) = $var(idx) + 1;
>> $var(aor) = $(var(aors){s.select,$var(idx),,});
>> }
>>
>> and the same problem happens: only the last destination
>> sip:user3@10.0.0.1:5076 is called.
>>
>> So, is there a way to append a non-fixed number of branches to an INVITE?
>>
>> Regards,
>> Takeshi
>>
>
> Sorry, I think I did something wrong.
> I was able to make append_branch to work inside a while loop.
> So there is no problem.
> Regards,
> Takeshi
>
>
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