date:20210202

Re: [OpenSIPS-Users] Global variable $rm gives number when using $json

2021-02-02 Thread Liviu Chircu


On 03.02.2021 00:14, Alex Kinch wrote:
but there now appears to be another issue whereby $json is outputting 
$socket_in(port) as a string instead of numeric? It's numeric as 
expected if I go back to 3.1.1


Well, you have just uncovered another slight inaccuracy in the $socket() 
variable code, which is now fixed [1]!  Thank you!


[1]: https://github.com/OpenSIPS/opensips/commit/de72c4f829ae0

Cheers,

--
Liviu Chircu
www.twitter.com/liviuchircu | www.opensips-solutions.com


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Re: [OpenSIPS-Users] Global variable $rm gives number when using $json

2021-02-02 Thread Alex Kinch

On Tue, 2 Feb 2021 at 20:11, Liviu Chircu  wrote:

> This problem is identical to the previous one and the patch for the
> $json variable should cover both of them.  Make sure you're using the
> nightly 3.1 packages or wait for the 3.1.2 release, after which you can
> test out the fix.

Sorry, me again -  $socket_in(af) is working as expected in nightly 3.1
(thanks again!) but there now appears to be another issue whereby $json is
outputting $socket_in(port) as a string instead of numeric? It's numeric as
expected if I go back to 3.1.1

Thanks,
Alex
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Re: [OpenSIPS-Users] Global variable $rm gives number when using $json

2021-02-02 Thread Alex Kinch

On Tue, 2 Feb 2021 at 20:11, Liviu Chircu  wrote:

This problem is identical to the previous one and the patch for the
> $json variable should cover both of them.  Make sure you're using the
> nightly 3.1 packages or wait for the 3.1.2 release, after which you can
> test out the fix.
>

Thanks Liviu :)

Alex
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Re: [OpenSIPS-Users] OPenSIPs 3.1 + CDRTool 9.9, Nodes IP Address and Call Direction for SIP Trace

2021-02-02 Thread Ahmed Chohan

Hi,

Trying to follow up on this as seeking advice.


>
> Hi,
>
> I've recently installed and configured OpenSIPs 3.1 + CDRTool 9.9 on CentOS
> 7 platform. As per functionality based not facing any issues as able to get
> SIP traces and accounting.
>
> As per CDRTool 9.9 release, they've added feature nodes IP addresses along
> with directions for  SIP trace. On OPenSIPs configuration, I've already
> declared and configured tracer modules however, the issue I'm experiencing
> is not showing call direction (INVITE, Trying, OK, etc) and nodes IP
> address but only showing BYE header direction.
>
> Please advise additional configuration I may be missing out as tried as
>  followed tracer module document i.e. loaded module, modparameter and
> configured in routing script along with CDRTool installation document.
>
> --
> Regards,
>
> Ahmed Munir Chohan
> -- next part --
> An HTML attachment was scrubbed...
> URL: <
> http://lists.opensips.org/pipermail/users/attachments/20210126/e4c007aa/attachment-0001.html
> >
>
>
-- 
Regards,

Ahmed Munir Chohan
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Re: [OpenSIPS-Users] Global variable $rm gives number when using $json

2021-02-02 Thread Liviu Chircu


On 02.02.2021 18:32, Alex Kinch wrote:
Just found another similar related issue so thought I'd pop it on the 
same thread.


$socket_in(af) returns INET or INET6 as expected but after passing 
through $json it's numeric and 0 in both cases.


Redacted examples below. I'm on 3.1.1 from Debian/Ubuntu packages.


Hi Alex,

This problem is identical to the previous one and the patch for the 
$json variable should cover both of them.  Make sure you're using the 
nightly 3.1 packages or wait for the 3.1.2 release, after which you can 
test out the fix.


Kind regards,

--
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www.twitter.com/liviuchircu | www.opensips-solutions.com


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Re: [OpenSIPS-Users] opensips dialplan regex

2021-02-02 Thread volga629 via Users


  
  
Hello Johan,
Huge thank you that resolve the issue.

volga629
  
On 2/2/21 12:58 PM, Johan De Clercq
  wrote:


  
  
if your _expression_ means : any number of digits, then try
  something like below: 



try ([0-9]*)([A-Z]*)([0-9]*)


replace with \1\3. 

  
  
  
Op di 2 feb. 2021 om 17:45
  schreef volga629 via Users :


   Hello Everyone,
Here are some test result 

Feb  3 00:38:46 voice-proxy /usr/sbin/opensips[1337808]:
[REQ_ROUTE] [INVITE] got incorrect 1506855JMGTJ4566
adjusting...
Feb  3 00:38:46 voice-proxy /usr/sbin/opensips[1337808]:
[REQ_ROUTE] [INVITE] new username -> [1506855JMGTJ4566]
~> 1506855

As you see digits on the end got lost.

volga629.

On 2/2/21 11:47 AM, volga629 via Users wrote:

 Hello Everyone,
  I am trying clean up with dialplan  any characters
  from $rU except digits.
  I tried regex rule
  
  Matching Regular _expression_ 
   ([0-9]*)

  Substitution
  Regular _expression_
  ([0-9]*)
  
  Replacement _expression_
  \1
  
  
  Example: 
  fhgg592199477719hh#
  
I
  see that rule match regex, but rewrite is not
  happening.
  
  Any help thank you.
  
  volga629.
 
  
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Re: [OpenSIPS-Users] Media Exchange API media_fork_to_uri

2021-02-02 Thread Eugene Christensen

I believe I have made some headway on this today.  It appears that in my 
opensips.cfg, the line towards the top that is 

socket=tcp:0.0.0.0:5060 AS 52.24.166.239:5060 

is creating a listener for me (from your earlier question).

In the script where I call media_fork_to_uri, I've added the transport=tcp to 
the uri.  That seems to get me passed the issue and a call is now being sent to 
my other endpoint.

I'm not out of the woods but I'm a step closer.

Is there any documentation on the type of thing I'm trying to do?  I see the 
media_exchange module documentation but perhaps for a beginner to OpenSIPS and 
media_exchange, something more would be very useful.

Thank you.

Eugene

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-Original Message-
From: Users  On Behalf Of Eugene Christensen
Sent: Tuesday, February 2, 2021 8:36 AM
To: OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] Media Exchange API media_fork_to_uri

[EXTERNAL] 

Hi Răzvan. 

Thank you for looking at this with me.

Attached is my opensips.cfg file.

In addition to starting OpenSIPS, the only other thing I am doing is starting 
RTPProxy using the following command.  If there is more that I should be doing 
to configure listeners, I am unaware.

rtpproxy -u admin -s udp:127.0.0.1 12345 -l 172.31.6.99 -A 52.24.166.239

I don't expect that rtpproxy and the configured addresses for it are having any 
bearing on this but I could be wrong.  I'll explain the addresses here and in 
the config file.
172.31.6.99 and 52.24.166.239 are the private and public addresses on the 
server running OpenSIPS and the rtpproxy (both on the same machine).  

The 123.123.123.123 address in the config file in the media_fork_to_uri command 
has been altered for this thread.  The unaltered address is a public reachable 
address where I have a SIP device listening for an incoming SIP call, 
presumably from the OpenSIPS server when I run the media_fork_to_uri command.

Does this answer the question about the listeners or am I missing something 
that I need to do for listeners (or other)?

Please advise.

Thank you.

Eugene Christensen.

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previous e-mail messages attached to it, may contain confidential and 
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responsible for delivering it to the intended recipient, you are hereby 
notified that any disclosure, copying, distribution or use of any of the 
information contained in or attached to this message is STRICTLY PROHIBITED. If 
you have received this transmission in error, please immediately notify me by 
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and destroy the original transmission and its attachments without reading them 
or saving them to disk.

-Original Message-
From: Users  On Behalf Of Razvan Crainea
Sent: Tuesday, February 2, 2021 7:22 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Media Exchange API media_fork_to_uri

[EXTERNAL] 

Hi, Eugene!

Could you give us a bit more information about the listeners (sockets) you are 
using?

Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 2/1/21 11:22 PM, Eugene Christensen wrote:
> Hello,
> 
> I’m trying to use the Media Exchange module and can’t seem to get it 
> to work as I wish.
> 
> I’ve enabled RTPProxy and have that working to anchor media for a SIP 
> call with the OpenSIPS server.  In the script I have the following 
> code in the route section of the code:
> 
> …
> 
>      # account only INVITEs
> 
>      if (is_method("INVITE")) {
> 
>      xlog("Received INVITE ... start dialog, rtpproxy and 
> then fork media\n");
> 
>      create_dialog();
> 
>      rtpproxy_engage();
> 
>  
> media_fork_to_uri("sip:1222333@123.123.123.123:5060", "caller");
> 
>      do_accounting("log");
> 
>      }
> 
> …
> 
> I see the following errors:
> 
> Jan 29 22:31:51 ip-172-31-6-99 /usr/local/sbin/opensips[17084]: 
> ERROR:media_exchange:uri2sock: no corresponding socket for af 2
> 
> Jan 29 22:31:51 ip-172-31-6-99 /usr/local/sbin/opensips[17084]: 
> ERROR:media_exchange:media_fork_to_uri: could not

Re: [OpenSIPS-Users] opensips dialplan regex

2021-02-02 Thread Johan De Clercq

if your expression means : any number of digits, then try something like
below:

try ([0-9]*)([A-Z]*)([0-9]*)

replace with \1\3.

Op di 2 feb. 2021 om 17:45 schreef volga629 via Users <
users@lists.opensips.org>:

> Hello Everyone,
> Here are some test result
>
> Feb  3 00:38:46 voice-proxy /usr/sbin/opensips[1337808]: [REQ_ROUTE]
> [INVITE] got incorrect 1506855JMGTJ4566 adjusting...
> Feb  3 00:38:46 voice-proxy /usr/sbin/opensips[1337808]: [REQ_ROUTE]
> [INVITE] new username -> [1506855JMGTJ4566] ~> 1506855
>
> As you see digits on the end got lost.
>
> volga629.
>
> On 2/2/21 11:47 AM, volga629 via Users wrote:
>
> Hello Everyone,
> I am trying clean up with dialplan  any characters from $rU except digits.
> I tried regex rule
>
> Matching Regular Expression
>  ([0-9]*)
>
> Substitution Regular Expression
> ([0-9]*)
>
> Replacement Expression
> \1
>
>
> Example:
> fhgg592199477719hh#
>
> I see that rule match regex, but rewrite is not happening.
>
> Any help thank you.
>
> volga629.
>
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> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] To-tag value in ACK

2021-02-02 Thread John Quick

Johan,
Thanks for your comment, but in this instance the problem is something very 
subtle.
OpenSIPS is acting as a Proxy, not as an endpoint. So the Contact header in the 
ACK contains the address of the UAC.
OpenSIPS identified itself earlier in the dialog using the correct FQDN in the 
topmost Record-Route header of the INVITE request and using TLS with a 
certificate whose subject name matches the FQDN.

Like I said, I have been able to put this ACK side-by-side with the ACK in a 
similar case where the call works correctly.
Doing an A-B comparison, the only obvious difference I could identify was in 
the order of the headers.
For example, the Route header is before the Via headers in one case and after 
in the other. I don't believe this is important.

So then I looked at values in the 200 OK to see if they were the same in the 
ACK:
The R-URI in the ACK is the same as the Contact in the 200 OK, including 
parameters.
The From tag is identical in both
The To tag in the ACK is not the same as the To tag in the 200 OK, but in my 
sip trace for a call that worked okay, the To tag did not get changed.

My question is about the To tag. Should it be the same in the ACK as it was in 
the 200 OK?

John Quick
Smartvox Limited
Tel:  +44-1727-221221


> From: Johan De Clercq  
> Sent: 02 February 2021 14:42
> To: John Q ; OpenSIPS users mailling list 
> 
> Subject: Re: [OpenSIPS-Users] To-tag value in ACK
>
> is contact an fqdn ? 
> If not, look no further. 


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Re: [OpenSIPS-Users] opensips dialplan regex

2021-02-02 Thread volga629 via Users


  
  
Hello Everyone,
Here are some test result 

Feb  3 00:38:46 voice-proxy /usr/sbin/opensips[1337808]: [REQ_ROUTE]
[INVITE] got incorrect 1506855JMGTJ4566 adjusting...
Feb  3 00:38:46 voice-proxy /usr/sbin/opensips[1337808]: [REQ_ROUTE]
[INVITE] new username -> [1506855JMGTJ4566] ~> 1506855

As you see digits on the end got lost.

volga629.

On 2/2/21 11:47 AM, volga629 via Users
  wrote:


  
  Hello Everyone,
  I am trying clean up with dialplan  any characters from $rU
  except digits.
  I tried regex rule
  
  Matching Regular _expression_ 
   ([0-9]*)

  Substitution Regular
  _expression_
  ([0-9]*)
  
  Replacement _expression_
  \1
  
  
  Example: 
  fhgg592199477719hh#
  
I see that
  rule match regex, but rewrite is not happening.
  
  Any help thank you.
  
  volga629.
 
  
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Re: [OpenSIPS-Users] Global variable $rm gives number when using $json

2021-02-02 Thread Alex Kinch

Hi Liviu,

Just found another similar related issue so thought I'd pop it on the same
thread.

$socket_in(af) returns INET or INET6 as expected but after passing through
$json it's numeric and 0 in both cases.

Redacted examples below. I'm on 3.1.1 from Debian/Ubuntu packages.

Thanks,
Alex

Feb  2 16:24:11 [1095] INVITE request from XX to XX received
Feb  2 16:24:11 [1095] AF is INET proto is udp
Feb  2 16:24:11 [1095] Making request to invite service...
Feb  2 16:24:11 [1095] Request is { "call_id": "XX", "ts": "2021-02-02
16:24:11Z", "src_ip": "XX", "src_port": 63412, "dst_ip": "XX", "dst_port":
5060, "dst_af": 0, "proto": "udp", "method": "INVITE", "sip_from":
"sip:XX", "sip_to": "sip:XX", "dialled": "XX" }

Feb  2 16:24:13 [1119] INVITE request from XX to XX received
Feb  2 16:24:13 [1119] AF is INET6 proto is udp
Feb  2 16:24:13 [1119] Making request to invite service...
Feb  2 16:24:13 [1119] Request is { "call_id": "XX", "ts": "2021-02-02
16:24:13Z", "src_ip": "XX", "src_port": 50505, "dst_ip": "XX", "dst_port":
5060, "dst_af": 0, "proto": "udp", "method": "INVITE", "sip_from":
"sip:XX", "sip_to": "sip:XX", "dialled": "XX" }


On Tue, 19 Jan 2021 at 17:22, Alex Kinch  wrote:

> On Tue, 19 Jan 2021 at 17:17, Liviu Chircu  wrote:
>
>
>> For 3.1, I think we're still in the right time window to backport this
>> fix without breaking any opensips.cfg files.
>
>
> I'm using 3.1 so quite happy with that proposal :) Thanks for the fast
> response and fix, much appreciated.
>
> Alex
>
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[OpenSIPS-Users] opensips dialplan regex

2021-02-02 Thread volga629 via Users


  
  
Hello Everyone,
I am trying clean up with dialplan  any characters from $rU
except digits.
I tried regex rule

Matching Regular _expression_ 
 ([0-9]*)
  
Substitution Regular
_expression_
([0-9]*)

Replacement _expression_
\1


Example: 
fhgg592199477719hh#

  I see that
rule match regex, but rewrite is not happening.

Any help thank you.

volga629.
  
  


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Re: [OpenSIPS-Users] Media Exchange API media_fork_to_uri

2021-02-02 Thread Eugene Christensen

Hi Răzvan. 

Thank you for looking at this with me.

Attached is my opensips.cfg file.

In addition to starting OpenSIPS, the only other thing I am doing is starting 
RTPProxy using the following command.  If there is more that I should be doing 
to configure listeners, I am unaware.

rtpproxy -u admin -s udp:127.0.0.1 12345 -l 172.31.6.99 -A 52.24.166.239

I don't expect that rtpproxy and the configured addresses for it are having any 
bearing on this but I could be wrong.  I'll explain the addresses here and in 
the config file.
172.31.6.99 and 52.24.166.239 are the private and public addresses on the 
server running OpenSIPS and the rtpproxy (both on the same machine).  

The 123.123.123.123 address in the config file in the media_fork_to_uri command 
has been altered for this thread.  The unaltered address is a public reachable 
address where I have a SIP device listening for an incoming SIP call, 
presumably from the OpenSIPS server when I run the media_fork_to_uri command.

Does this answer the question about the listeners or am I missing something 
that I need to do for listeners (or other)?

Please advise.

Thank you.

Eugene Christensen.

CONFIDENTIALITY NOTICE. This e-mail transmission, and any documents, files or 
previous e-mail messages attached to it, may contain confidential and 
proprietary information. If you are not the intended recipient, or a person 
responsible for delivering it to the intended recipient, you are hereby 
notified that any disclosure, copying, distribution or use of any of the 
information contained in or attached to this message is STRICTLY PROHIBITED. If 
you have received this transmission in error, please immediately notify me by 
reply e-mail at echristen...@sorenson.com or by telephone at +1 (801) 287-9419, 
and destroy the original transmission and its attachments without reading them 
or saving them to disk.

-Original Message-
From: Users  On Behalf Of Razvan Crainea
Sent: Tuesday, February 2, 2021 7:22 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Media Exchange API media_fork_to_uri

[EXTERNAL] 

Hi, Eugene!

Could you give us a bit more information about the listeners (sockets) you are 
using?

Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 2/1/21 11:22 PM, Eugene Christensen wrote:
> Hello,
> 
> I’m trying to use the Media Exchange module and can’t seem to get it 
> to work as I wish.
> 
> I’ve enabled RTPProxy and have that working to anchor media for a SIP 
> call with the OpenSIPS server.  In the script I have the following 
> code in the route section of the code:
> 
> …
> 
>      # account only INVITEs
> 
>      if (is_method("INVITE")) {
> 
>      xlog("Received INVITE ... start dialog, rtpproxy and 
> then fork media\n");
> 
>      create_dialog();
> 
>      rtpproxy_engage();
> 
>  
> media_fork_to_uri("sip:1222333@123.123.123.123:5060", "caller");
> 
>      do_accounting("log");
> 
>      }
> 
> …
> 
> I see the following errors:
> 
> Jan 29 22:31:51 ip-172-31-6-99 /usr/local/sbin/opensips[17084]: 
> ERROR:media_exchange:uri2sock: no corresponding socket for af 2
> 
> Jan 29 22:31:51 ip-172-31-6-99 /usr/local/sbin/opensips[17084]: 
> ERROR:media_exchange:media_fork_to_uri: could not find suitable socket 
> for originating traffic to sip:1222333444@123.123.123.123:5060
> 
> 
> Any ideas what I am missing in my attempt?
> 
> Is there a primer on the use of the media exchange module?  
> Specifically with this particular API?
> 
> Thank you.
> 
> 
> Eugene Christensen.
> 
> *CONFIDENTIALITY NOTICE.*This e-mail transmission, and any documents, 
> files or previous e-mail messages attached to it, may contain 
> confidential and proprietary information. If you are not the intended 
> recipient, or a person responsible for delivering it to the intended 
> recipient, you are hereby notified that any disclosure, copying, 
> distribution or use of any of the information contained in or attached 
> to this message is STRICTLY PROHIBITED. If you have received this 
> transmission in error, please immediately notify me by reply e-mail at 
> echristen...@sorenson.com or by 
> telephone at +1 (801) 287-9419, and destroy the original transmission 
> and its attachments without reading them or saving them to disk.
> 
> 
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opensips.cfg
Description: opensips.cfg
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[OpenSIPS-Users] dispatcher cluster

2021-02-02 Thread volga629 via Users


  
  
Hello Everyone,
Dispatcher in cluster causing destinations goes down
unexpectedly.  
Scenario: 
Two nodes with last ip octet 61 and 41.
Freeswitch stats to calculate  weight.
Issue:
61 is  set to ping all destinations and report to the 41 the
status via cluster, the issue that 41 bring down all
destinations into Inactive state after cluster update message
until ds_reload issued again.
Attempt to fix: 
I tried set limit on which groups which node can ping, but seems
like in cluster it should be relevant, because 61 should  get
status of all groups and send to  41.
Relevant Config:
Node 61:
  
 Dispatcher
loadmodule "dispatcher.so"
modparam("dispatcher", "db_url", "postgres://")
modparam("dispatcher", "table_name", "dispatcher")
modparam("dispatcher", "setid_col", "setid")
modparam("dispatcher", "priority_col", "priority")
modparam("dispatcher", "destination_col", "destination")
modparam("dispatcher", "cnt_avp", "$avp(274)")
modparam("dispatcher", "grp_avp", "$avp(275)")
modparam("dispatcher", "hash_pvar", "$avp(273)")
modparam("dispatcher", "dst_avp", "$avp(271)")
modparam("dispatcher", "sock_avp", "$avp(276)")
modparam("dispatcher", "ds_ping_from", "sip:proxy@10.30.100.61")
modparam("dispatcher", "ds_ping_method", "OPTIONS")
modparam("dispatcher", "ds_ping_interval", 45)
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "ds_probing_threshold", 5)
modparam("dispatcher", "ds_probing_list", "2,3,4")
modparam("dispatcher", "fetch_freeswitch_stats", 1)
modparam("dispatcher", "options_reply_codes", "501,403,404,400,200")
modparam("dispatcher", "cluster_id", 1)

Node 41:

 Dispatcher
loadmodule "dispatcher.so"
modparam("dispatcher", "")
modparam("dispatcher", "table_name", "dispatcher")
modparam("dispatcher", "setid_col", "setid")
modparam("dispatcher", "priority_col", "priority")
modparam("dispatcher", "destination_col", "destination")
modparam("dispatcher", "cnt_avp", "$avp(274)")
modparam("dispatcher", "grp_avp", "$avp(275)")
modparam("dispatcher", "hash_pvar", "$avp(273)")
modparam("dispatcher", "dst_avp", "$avp(271)")
modparam("dispatcher", "sock_avp", "$avp(276)")
modparam("dispatcher", "ds_ping_from", "sip:proxy@10.30.100.41")
modparam("dispatcher", "ds_ping_method", "OPTIONS")
modparam("dispatcher", "ds_ping_interval", 45)
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "ds_probing_threshold", 5)
modparam("dispatcher", "fetch_freeswitch_stats", 1)
modparam("dispatcher", "options_reply_codes", "501,403,404,400,200")
modparam("dispatcher", "ds_probing_list", "1")
modparam("dispatcher", "cluster_id", 1)
modparam("dispatcher", "cluster_sharing_tag", "vip")



Comments:
I think in cluster 41 should not do any operation or decisions regard node states and it should rely on 61 only 

Log: 

 41 node
[root@vprx00 ~]# grep EVENT /var/log/opensips/opensips.log
Feb  1 12:44:57 vprx00 /usr/sbin/opensips[250547]: [EVENT_ROUTE] [DISPATCHER] received group=0 ~> address=sip:10.30.100.57:5160 ~> status=inactive
Feb  1 12:50:54 vprx00 /usr/sbin/opensips[250545]: [EVENT_ROUTE] [DISPATCHER] received group=0 ~> address=sip:10.30.100.48:5160 ~> status=inactive
Feb  1 12:52:15 vprx00 /usr/sbin/opensips[250538]: [EVENT_ROUTE] [DISPATCHER] received group=0 ~> address=sip:10.30.100.49:5160 ~> status=inactive

   41 node
   Feb  1 14:24:49 vprx00 /usr/sbin/opensips[250540]:
DBG:dispatcher:w_ds_select: ds_select: 1 1 1000 1
   Feb  1 14:24:49 vprx00 /usr/sbin/opensips[250540]:
DBG:dispatcher:ds_select_dst: no active destinations in set [1] !
 
volga629.


 
  
  

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Re: [OpenSIPS-Users] To-tag value in ACK

2021-02-02 Thread Johan De Clercq

is contact an fqdn ?
If not, look no further.

Op di 2 feb. 2021 om 15:06 schreef John Quick :

> I am seeing a problem in calls made to MS Teams via OpenSIPS configured as
> an SBC.
> The usual INVITE, 180, 183, 200 OK sequence looks okay, but the ACK request
> is not accepted by MS Teams.
> When I say "not accepted", I mean an ACK is sent to the Teams Proxy, but
> Teams responds after 30 seconds with BYE, reason is:
> "Call Controller timed out while waiting for acknowledgement"
>
> I am able to compare the sip trace for the failing call with a very similar
> call scenario on another Teams system that works.
> The only explanation I can suggest is that the UAC did not use the same
> value for the To-tag as it received in the 200 OK.
> Could this explain why Teams is failing to match it against the previous
> dialogue?
>
> John Quick
> Smartvox Limited
> Web: www.smartvox.co.uk
>
>
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Re: [OpenSIPS-Users] Media Exchange API media_fork_to_uri

2021-02-02 Thread Răzvan Crainea


Hi, Eugene!

Could you give us a bit more information about the listeners (sockets) 
you are using?


Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 2/1/21 11:22 PM, Eugene Christensen wrote:

Hello,

I’m trying to use the Media Exchange module and can’t seem to get it to 
work as I wish.


I’ve enabled RTPProxy and have that working to anchor media for a SIP 
call with the OpenSIPS server.  In the script I have the following code 
in the route section of the code:


…

     # account only INVITEs

     if (is_method("INVITE")) {

     xlog("Received INVITE ... start dialog, rtpproxy and 
then fork media\n");


     create_dialog();

     rtpproxy_engage();

 
media_fork_to_uri("sip:1222333@123.123.123.123:5060", "caller");


     do_accounting("log");

     }

…

I see the following errors:

Jan 29 22:31:51 ip-172-31-6-99 /usr/local/sbin/opensips[17084]: 
ERROR:media_exchange:uri2sock: no corresponding socket for af 2


Jan 29 22:31:51 ip-172-31-6-99 /usr/local/sbin/opensips[17084]: 
ERROR:media_exchange:media_fork_to_uri: could not find suitable socket 
for originating traffic to sip:1222333444@123.123.123.123:5060 



Any ideas what I am missing in my attempt?

Is there a primer on the use of the media exchange module?  Specifically 
with this particular API?


Thank you.


Eugene Christensen.

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[OpenSIPS-Users] To-tag value in ACK

2021-02-02 Thread John Quick

I am seeing a problem in calls made to MS Teams via OpenSIPS configured as
an SBC.
The usual INVITE, 180, 183, 200 OK sequence looks okay, but the ACK request
is not accepted by MS Teams.
When I say "not accepted", I mean an ACK is sent to the Teams Proxy, but
Teams responds after 30 seconds with BYE, reason is:
"Call Controller timed out while waiting for acknowledgement"

I am able to compare the sip trace for the failing call with a very similar
call scenario on another Teams system that works.
The only explanation I can suggest is that the UAC did not use the same
value for the To-tag as it received in the 200 OK.
Could this explain why Teams is failing to match it against the previous
dialogue?

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


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Re: [OpenSIPS-Users] Global variable $rm gives number when using $json

Re: [OpenSIPS-Users] Global variable $rm gives number when using $json

Re: [OpenSIPS-Users] Global variable $rm gives number when using $json

Re: [OpenSIPS-Users] OPenSIPs 3.1 + CDRTool 9.9, Nodes IP Address and Call Direction for SIP Trace

Re: [OpenSIPS-Users] Global variable $rm gives number when using $json

Re: [OpenSIPS-Users] opensips dialplan regex

Re: [OpenSIPS-Users] Media Exchange API media_fork_to_uri

Re: [OpenSIPS-Users] opensips dialplan regex

Re: [OpenSIPS-Users] To-tag value in ACK

Re: [OpenSIPS-Users] opensips dialplan regex

Re: [OpenSIPS-Users] Global variable $rm gives number when using $json

[OpenSIPS-Users] opensips dialplan regex

Re: [OpenSIPS-Users] Media Exchange API media_fork_to_uri

[OpenSIPS-Users] dispatcher cluster

Re: [OpenSIPS-Users] To-tag value in ACK

Re: [OpenSIPS-Users] Media Exchange API media_fork_to_uri

[OpenSIPS-Users] To-tag value in ACK

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