[OpenSIPS-Users] Are alias_db_.. blocking functions?

[John Quick]

Please can someone confirm if alias_db_lookup() and alias_db_find() are
blocking functions.
If they are, can they be used in the async function?
I'm using v2.4.6 and 2.4.7 of OpenSIPS.
Thanks.

John Quick
Smartvox Limited
Web: www.smartvox.co.uk



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Re: [OpenSIPS-Users] rtpengine documentation

[Callum Guy]

Hi Rob,

I'm interested to follow your thread to hear more about this, I have
found that some flags are valid yet undocumented during initial setup
of some RTC compatable proxies.

Two in particular: DTLS-passive and SDES-disable both of which appear
to influence behaviour of RTPEngine in accordance with the RTPEngine
documentation.

YMMV however it might help you to try this format as you proceed with testing.

Good luck!

Callum


On Sat, 16 May 2020 at 20:38, Robert Dyck  wrote:
>
> I am wanting to convert my config/script to use rtpengine instead of rtpproxy.
> I think it would better deal with webrtc. After looking at some examples I
> found, I see a couple of parameters that are not mentioned in the opensips
> documentation. First there is the offer/answer option ice=force-relay and
> secondly DTLS=passive.
>
> Are these options obsolete/deprecated/intentionally omitted?
>
> On the subject of DTLS I noticed that when I use ice=force in offer and answer
> rtpengine sends new DTLS fingerprints to the parties. I appears to operate as
> back-to-back DTLS agent. I know this because both UAs sent SHA-256
> fingerprints but they received SHA-1 fingerprints. This may have worked but
> one UA will only accept SHA-256 and it drops the call.
>
> The documentation does not mention that the ice= option can influence DTLS.
>
> Regards, Rob
>
>
>
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[OpenSIPS-Users] OpenSips Configuration with RTP Proxy

[Burhan Khan]

Hi

I have complied RTP Proxy and its working also I have added rtpproxy module
in opensips.cfg file but it is not working.
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Re: [OpenSIPS-Users] UDP fragmentation in reply routes

[Maxim Sobolev]

Smells like a OS/kernel bug to me. There is little application can do in
that regard, UDP fragmentation/reassembly happens at much lower layers of
the OSI stack.

However, as a workaround as long as SIP goes you can try to reduce your SIP
signalling packet size by using compact version of SIP headers, as well as
dropping headers that are not used. That would save you 100-150 bytes per
SIP message perhaps. I don't know if OpenSIP can do that in the proxy mode
out of the box though, so you might want to add b2b into the flow.

-Max

On Mon., May 18, 2020, 12:34 p.m. Olle Frimanson,  wrote:

> Hi,
>
>
>
> We have an issue on our home proxy (opensips 2.4.6), when it receives  200
> OK (over UDP)  from our Freeswitch and the package size is higher than the
> MTU size , we sometimes get fragmentation  of the UDP packets, but only the
> first part of the fragmented package is sent to our edge proxy. Is this a
> known issue or is it a OS bug?
>
>
>
> We have not yet spotted any pattern on this and in most cases bigger
> packets with MTU around 1600 bytes gets through without an issue.
>
>
>
> I can add that in the other direction in the normal request routes we
> don’t have any issue at all can have packets > 2000 bytes without any
> issues.
>
> Does Opensips use IP_MTU_DISCOVER or how is fragmentation controlled and is 
> it expected to have different behavior in reply routes vs other routes?
>
> We use Centos 7 in a virtual server environment.
>
>
>
> I was hoping someone can share some light on this strange issue.
>
>
>
> BR/Olle
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Re: [OpenSIPS-Users] Question about regex

[Bogdan-Andrei Iancu]

Hey Michael,

Maybe this {ip.isprivate} is a better options:
    https://www.opensips.org/Documentation/Script-Tran-3-1#toc80

Regards,

Bogdan-Andrei Iancu

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

On 5/18/20 8:29 PM, Saint Michael wrote:

I need to identify all private IPs vs public IPs
Right now I am doing
$rd =~ "192.168" || $rd =~ "10." || $rd =~ "172.16."
but in regex there is a faster way, chaining several ORs
like
$rd =~ "192.168|10.|172.16."
what is the correct way to do this in opensips?


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[OpenSIPS-Users] UDP fragmentation in reply routes

[Olle Frimanson]

Hi,

 

We have an issue on our home proxy (opensips 2.4.6), when it receives  200
OK (over UDP)  from our Freeswitch and the package size is higher than the
MTU size , we sometimes get fragmentation  of the UDP packets, but only the
first part of the fragmented package is sent to our edge proxy. Is this a
known issue or is it a OS bug?

 

We have not yet spotted any pattern on this and in most cases bigger packets
with MTU around 1600 bytes gets through without an issue.

 

I can add that in the other direction in the normal request routes we don't
have any issue at all can have packets > 2000 bytes without any issues.

Does Opensips use IP_MTU_DISCOVER or how is fragmentation controlled and is
it expected to have different behavior in reply routes vs other routes?

We use Centos 7 in a virtual server environment.

 

I was hoping someone can share some light on this strange issue.

 

BR/Olle

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[OpenSIPS-Users] add_rr_param(";r2=on"); not adding

[James Hogbin]

My script has

if (is_method("INVITE") && !has_totag() && check_source_address(0)) {
xlog("[ROUTE]Incoming call to MS: RURI=$ruri, SI=$si, M=$rm\n");
trace("tid");
create_dialog();
do_routing(1);  
strip(1);
prefix("+44");
record_route_preset("sbc.ip-sentinel.com:5091;transport=tls", 
"10.0.0.5:5060");
add_rr_param(";r2=on");
route(relay);

Which is turning this:

James Hogbin
Director 
 
IP Sentinel 
t. +44 (0)20 3011 4150
m. +44 7786910895
w. https://www.ip-sentinel.com

INVITE sip:08435577...@sbc.ip-sentinel.com SIP/2.0
Via: SIP/2.0/UDP 13.80.245.144:5080;rport;branch=z9hG4bK1tDv8cQgjXXaj
Max-Forwards: 68
From: "James Hogbin" ;tag=8tKmNZc4FpF3D
To: 

Into:

INVITE sip:+448435577...@sip.pstnhub.microsoft.com SIP/2.0
Record-Route: 

Record-Route: 
Via: SIP/2.0/TLS 137.117.136.143:5091;branch=z9hG4bKb195.04f15415.0
Via: SIP/2.0/UDP 
13.80.245.144:5080;received=13.80.245.144;rport=5080;branch=z9hG4bK1tDv8cQgjXXaj
Max-Forwards: 68
From: "James Hogbin" ;tag=8tKmNZc4FpF3D
To: 

BUT the Record routes do not have ;r2=on in them

I think this is causing an issue with ACK & BYE from the PBX hanging off the 
5050 port

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delivered and neither it nor any of its contents should be passed on to or used 
by any other person. IP Sentinel Ltd is a limited company registered in England 
and Wales under Registered Number 08648097. Registered Office: Newnhams Wood, 
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[OpenSIPS-Users] Running OpenSIPS on same box as Asterisk/FreeSwitch

[John Quick]

Whenever I have tried to run OpenSIPS on the same host server as either
Asterisk or FreeSwitch (using different ports), I have always hit a snag
with routing of sequential loose-routed requests. Possibly this only happens
when using double Record-Route headers which I need for protocol conversion
(TLS to UDP or TLS to TCP).

Where the address in the topmost Route header is the same as it's local
address, OpenSIPS copies the URI (and parameters) from a Route header to the
R-URI which, in this use-case, causes the request to loop back to itself. It
should instead delete the two Route headers and copy the URI value from the
Contact header, picking up the correct port for the Asterisk/FreeSwitch app.

I've tried Googling to see if others have experienced this, but didn't find
any nice solution other than using two IP's on the same box.
A post from Flavio mentions this problem in June 2013. Is there a simple
solution?

John Quick
Smartvox Limited
Web: www.smartvox.co.uk



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[OpenSIPS-Users] Question about regex

[Saint Michael]

I need to identify all private IPs vs public IPs
Right now I am doing
$rd =~ "192.168" || $rd =~ "10." || $rd =~ "172.16."
but in regex there is a faster way, chaining several ORs
like
$rd =~ "192.168|10.|172.16."
what is the correct way to do this in opensips?
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Re: [OpenSIPS-Users] Running OpenSIPS on same box as Asterisk/FreeSwitch

[James Hogbin]

I thought I had just done some bad config.  I have almost exactly the same 
thing.

I have Freeswitch And Opensips on Azure VM’s

Freeswitch = 10.0.0.4 + External IP
Opensips = 10.0.0.5 + External IP

Both are configured to use the external addresses

Freeswitch will start the conversation on its external Gateway IP address with 
the Opensips external IP address

However the moment I put the Record_Route in the message to Teams Freeswitch 
starts using the local 10.x.x.x. address on opensips rather than the external 
address for ACK and BYE which rogers the whole thing up.

I’m sure there is probably some way to fix it in the opensips plan as the 
dialog/transaction ID’s are still there but it’s beyond me.

James

James Hogbin
Director 
 
IP Sentinel 
t. +44 (0)20 3011 4150
m. +44 7786910895
w. https://www.ip-sentinel.com

> On 18 May 2020, at 16:28, John Quick  wrote:
> 
> Whenever I have tried to run OpenSIPS on the same host server as either
> Asterisk or FreeSwitch (using different ports), I have always hit a snag
> with routing of sequential loose-routed requests. Possibly this only happens
> when using double Record-Route headers which I need for protocol conversion
> (TLS to UDP or TLS to TCP).
> 
> Where the address in the topmost Route header is the same as it's local
> address, OpenSIPS copies the URI (and parameters) from a Route header to the
> R-URI which, in this use-case, causes the request to loop back to itself. It
> should instead delete the two Route headers and copy the URI value from the
> Contact header, picking up the correct port for the Asterisk/FreeSwitch app.
> 
> I've tried Googling to see if others have experienced this, but didn't find
> any nice solution other than using two IP's on the same box.
> A post from Flavio mentions this problem in June 2013. Is there a simple
> solution?
> 
> John Quick
> Smartvox Limited
> Web: www.smartvox.co.uk
> 
> 
> 
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[OpenSIPS-Users] [Blog] Cross dialog data accessing

[Bogdan-Andrei Iancu]

There are several calling scenarios – typical Class V – where multiple 
SIP dialogs may be involved. And to make it work, you need, /from one 
dialog, to access the data that belongs to another dialog/.


https://blog.opensips.org/2020/05/18/cross-dialog-data-accessing/

Enjoy,

--
Bogdan-Andrei Iancu

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

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Re: [OpenSIPS-Users] Upgrade 2.4 to 3.1

[Bogdan-Andrei Iancu]

Hi Mark,

First of all, all the upgrades are incremental, from one version to next 
one, so you should do 2.4 - > 3.0  and 3.0 -> 3.1


Secondly, we haven;t yet prepared the migration docs/scripts for 3.0 -> 
3.1, they will be available upon beta release.


Regards,

Bogdan-Andrei Iancu

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

On 5/18/20 5:16 PM, Mark Farmer wrote:

Hi everyone

Just wondering if there's a 'supported' upgrade process to jump from 
2.4 to 3.1?

Or will I need to do it in 2 steps (2.4 to 3.0 & 3.0 to 3.1)?

Many thanks
Mark.


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Re: [OpenSIPS-Users] Upgrade 2.4 to 3.1

[Mark Farmer]

Yes, I realise I might be getting a little ahead of myself :)
I think my question really should be 'can I upgrade the DB direct from 2.4
to 3.1 format?'

Mark.


On Mon, 18 May 2020 at 15:48, Giovanni Maruzzelli  wrote:

> On Mon, May 18, 2020 at 4:19 PM Mark Farmer  wrote:
>
>>
>> Just wondering if there's a 'supported' upgrade process to jump from 2.4
>> to 3.1?
>> Or will I need to do it in 2 steps (2.4 to 3.0 & 3.0 to 3.1)?
>>
>>
> Ahem...
> 3.1 is not even in beta, I believe...
> -giovanni
>
>
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-- 
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Re: [OpenSIPS-Users] OpenSIPS cdr variables

[Bogdan-Andrei Iancu]

Hi Asteriskman,

As per doc [1], the module will provide the timestamp of the 200 OK 
(call answering) . If doing "cdrs", you will also get the call duration, 
already computed.


For anything extra, you should use extra accounting data [2].

The accounting engine does not natively support noSQL backends, but you 
can use the accounting CDR event [3] in script [4] in order to perform a 
raw mongo query [5].



[1] https://opensips.org/html/docs/modules/3.0.x/acc.html#overview

[2] https://opensips.org/html/docs/modules/3.0.x/acc.html#ACC-extra-id

[3] https://opensips.org/html/docs/modules/3.0.x/acc.html#event_E_ACC_EVENT

[4] https://opensips.org/html/docs/modules/3.0.x/event_route.html#overview

[5] https://www.opensips.org/Documentation/Script-CoreFunctions-3-0#toc10

Regards,

Bogdan-Andrei Iancu

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

On 5/16/20 11:20 PM, Asteriskman wrote:

Hello everyone,

I basically know Asterisk and SIP but new to OpenSIPS. I am working on 
a call flow where I need to receive a call from the provider and send 
it to an endpoint or a trunk and store Call start timestamp(INVITE), 
connect timestamp(200 OK) and end timestamp(BYE) in MongoDB.

Now my questions are
(1) What is the best way to get those values in opensips script? Use 
$Ts at different stages and store the values in dialogue?
(2)What is the best way to perform insert into MongoDB? I see there is 
a cachedb_mongodb module and you can use cache_raw_query.

(3)Will my above approaches work when I do parallel forking too?

BR,


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Re: [OpenSIPS-Users] Are alias_db_.. blocking functions?

[Bogdan-Andrei Iancu]

Hi John,

Yes, these functions are performing DB queries in runtime and these 
queries are potentially blocking.


The db_aliases module does not offer async support, but the queries it 
is doing are trivial and you can do them via avp_db_query() + async() 
support.


Regards,

Bogdan-Andrei Iancu

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

On 5/18/20 11:56 AM, John Quick wrote:

Please can someone confirm if alias_db_lookup() and alias_db_find() are
blocking functions.
If they are, can they be used in the async function?
I'm using v2.4.6 and 2.4.7 of OpenSIPS.
Thanks.

John Quick
Smartvox Limited
Web: www.smartvox.co.uk



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Re: [OpenSIPS-Users] Adjust T1 and T2 timers per transaction

[Grant Bagdasarian]

Hi Bogdan-Andrei,

Yes, I know, but I meant similar to $T_fr_timeout and
$T_fr_inv_timeout, which can be set in the script during message
processing.
Sorry if my questions wasn't clear enough.

Regards,

Grant

On Mon, May 18, 2020 at 1:49 PM Bogdan-Andrei Iancu  wrote:
>
> Hi Grant,
>
> There are the 'T1_timer' and 'T2_timer' module parameters for that.
>
> https://opensips.org/html/docs/modules/3.0.x/tm.html#param_T1_timer
>
> Best regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
>
> On 5/16/20 7:53 PM, Grant Bagdasarian wrote:
>
> Hello,
>
> I couldn’t find anything related to adjusting the T1 and T2 timers on a per 
> transaction basis, similar to the fr and fr_inv timeouts.
>
> Is there a way in OpenSips to adjust these values at runtime?
>
> Regards,
>
> Grant
>
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[OpenSIPS-Users] Upgrade 2.4 to 3.1

[Mark Farmer]

Hi everyone

Just wondering if there's a 'supported' upgrade process to jump from 2.4 to
3.1?
Or will I need to do it in 2 steps (2.4 to 3.0 & 3.0 to 3.1)?

Many thanks
Mark.
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Re: [OpenSIPS-Users] REFER from MS teams to Kamailio/Opensips

[Bogdan-Andrei Iancu]

Hi Yuriy,

As you are using Kamailio and FreeSWITCH, it is more appropriate to post 
the question to their mailing lists.


Regards,

Bogdan-Andrei Iancu

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

On 5/14/20 6:09 PM, Yuriy Nasida wrote:

I use kamilio but I think it works in the same way for Opensips.

Well, I saw similar questions in the list already but looks like 
nobody has answer.


Please look at REFER below.

Kamilio gets REFER from MS and sends it to FS node. Next, FS node try 
to make 3th call for some reason.I expect that FS will not do 3th call 
and just will connect Alice and Bob itself.



2020/05/14 12:32:00.637027 KAM_IP:5060 -> FS_IP:5060
REFER sip:Alice_number@FS_IP:5060;transport=udp SIP/2.0
FROM: 
Customer1;tag=a860c50a3fb54d08b4e5740fa2dfb3d6

TO: ;user=phone;tag=e8ct9S6ty13va
CSEQ: 4 REFER
CALL-ID: 2c71b2a6669b5343a231e1244b19c945
MAX-FORWARDS: 50
Via: SIP/2.0/UDP 
FQDN_OF_TRUNK:5060;branch=z9hG4bK10ae.2c42897feca117121a23bf0c8d54cd19.0;i=c

VIA: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bK7e3e8998
CONTACT: 

CONTENT-LENGTH: 0
REFER-TO: 

REFERRED-BY: 

acbabfce07;x-tt=aHR0cHM6Ly9hcGktZHUtYS1ldXdlLnBzdG5odWIubWljcm9zb2Z0LmNvbS92MS9uZ2MvY2FsbG5vdGlmaWNhdGlvbj9kY2k9YzIxMjE3MzEyNTQ2NDk1ZjlhYTcwODliYTkwNGIxZGQ%3D>
USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.5.6.2 i.EUWE.4
ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
P-ASSERTED-IDENTITY: 
,>

PRIVACY: id
X-AUTH-IP: 52.114.75.24
X-AUTH-PORT: 3136


Any advice?

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Re: [OpenSIPS-Users] Adjust T1 and T2 timers per transaction

[Bogdan-Andrei Iancu]

Ah, gotcha no, there is nothing per transaction, these are global 
options.


Regards,

Bogdan-Andrei Iancu

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

On 5/18/20 2:51 PM, Grant Bagdasarian wrote:

Hi Bogdan-Andrei,

Yes, I know, but I meant similar to $T_fr_timeout and
$T_fr_inv_timeout, which can be set in the script during message
processing.
Sorry if my questions wasn't clear enough.

Regards,

Grant

On Mon, May 18, 2020 at 1:49 PM Bogdan-Andrei Iancu  wrote:

Hi Grant,

There are the 'T1_timer' and 'T2_timer' module parameters for that.

https://opensips.org/html/docs/modules/3.0.x/tm.html#param_T1_timer

Best regards,

Bogdan-Andrei Iancu

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

On 5/16/20 7:53 PM, Grant Bagdasarian wrote:

Hello,

I couldn’t find anything related to adjusting the T1 and T2 timers on a per 
transaction basis, similar to the fr and fr_inv timeouts.

Is there a way in OpenSips to adjust these values at runtime?

Regards,

Grant

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Re: [OpenSIPS-Users] OpenSips Configuration with RTP Proxy

[SamyGo]

Hi,

Could you share further details as whats not working?
check the following:
1 - OpenSIPS is able to connect tot he rtpproxy socket
2 - OpenSIPS is calling the right rtpproxy function from the script
3 - the RTPproxy function have correct parameters
4 - Check the SDP for incoming leg and the outgoing leg and check if
rtpproxy is modifying it or not at all.

Regards,
Sammy

On Mon, May 18, 2020 at 7:37 AM Burhan Khan  wrote:

> Hi
>
> I have complied RTP Proxy and its working also I have added rtpproxy
> module in opensips.cfg file but it is not working.
>
>
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Re: [OpenSIPS-Users] Upgrade 2.4 to 3.1

[Giovanni Maruzzelli]

On Mon, May 18, 2020 at 4:19 PM Mark Farmer  wrote:

>
> Just wondering if there's a 'supported' upgrade process to jump from 2.4
> to 3.1?
> Or will I need to do it in 2 steps (2.4 to 3.0 & 3.0 to 3.1)?
>
>
Ahem...
3.1 is not even in beta, I believe...
-giovanni
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