[OpenSIPS-Users] Media Exchange API media_fork_to_uri

2021-02-01 Thread Eugene Christensen
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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responsible for delivering it to the intended recipient, you are hereby 
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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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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.

*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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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 <mailto: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.
> 
> 
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 

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opensips.cfg
Description: opensips.cfg
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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

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 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.

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]: 
> 

Re: [OpenSIPS-Users] Media Exchange API media_fork_to_uri

2021-02-04 Thread Răzvan Crainea

Hi, Eugene!

The idea is simple: a socket/listener represents an 
interface+protocol+port OpenSIPS listens on and is able to use.
In your initial tests, you were trying to send the call to a UDP 
endpoint, but you were only defining a TCP listener - hence the failure.


Please let us know *what* you are trying to achieve so we can guide you 
to the right documentation.


Best regards,

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

On 2/2/21 9:08 PM, Eugene Christensen wrote:

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

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 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.

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 for

Re: [OpenSIPS-Users] Media Exchange API media_fork_to_uri

2021-02-04 Thread Eugene Christensen
Thanks Răzvan.

What I'm exploring is the ability to fork all the media of a call to another 
SIP entity.  When I do this, it could be at any time during the call.

I did look into SIPREC and played with that a bit too but since I wish to be 
able to do this at any time during a call, SIPREC doesn't seem like the best 
solution since it appears to want to get negotiated at the beginning of a call.

Ideally, I would like to do the following:

Identify a call that I would like to be able to receive both the caller's and 
calle's media.  Once identified, send the appropriate commands to fork the 
media (A/V/T) from both endpoints in the point to point call to another SIP 
entity.  I do not wish to have the receiving device send any media back in 
return.

If I am understanding things correctly, I should be able to create a dialog and 
engage RTPProxy (or RTPEngine) in the original call to anchor the media so that 
if/when I wish to fork the media, the media and dialog are already established 
where needed.

I am able to get the receiving device called now but I do not see media being 
sent to it.

Does this help explain my intentions and needs?

Thanks,
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: Thursday, February 4, 2021 7:48 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Media Exchange API media_fork_to_uri

[EXTERNAL] 

Hi, Eugene!

The idea is simple: a socket/listener represents an 
interface+protocol+port OpenSIPS listens on and is able to use.
In your initial tests, you were trying to send the call to a UDP endpoint, but 
you were only defining a TCP listener - hence the failure.

Please let us know *what* you are trying to achieve so we can guide you to the 
right documentation.

Best regards,

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

On 2/2/21 9:08 PM, Eugene Christensen wrote:
> 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
> 
> 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 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 rtppro

Re: [OpenSIPS-Users] Media Exchange API media_fork_to_uri

2021-02-09 Thread Eugene Christensen
Hi Răzvan,

Any further directions on this please?

I did get it to the point that it will fork the media for one or two calls but 
thereafter it requires the opensips to be restarted before it works again.  Is 
this a bug in the module or is it because of the way I'm using it?

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: Eugene Christensen 
Sent: Thursday, February 4, 2021 10:54 AM
To: OpenSIPS users mailling list 
Subject: RE: [OpenSIPS-Users] Media Exchange API media_fork_to_uri

Thanks Răzvan.

What I'm exploring is the ability to fork all the media of a call to another 
SIP entity.  When I do this, it could be at any time during the call.

I did look into SIPREC and played with that a bit too but since I wish to be 
able to do this at any time during a call, SIPREC doesn't seem like the best 
solution since it appears to want to get negotiated at the beginning of a call.

Ideally, I would like to do the following:

Identify a call that I would like to be able to receive both the caller's and 
calle's media.  Once identified, send the appropriate commands to fork the 
media (A/V/T) from both endpoints in the point to point call to another SIP 
entity.  I do not wish to have the receiving device send any media back in 
return.

If I am understanding things correctly, I should be able to create a dialog and 
engage RTPProxy (or RTPEngine) in the original call to anchor the media so that 
if/when I wish to fork the media, the media and dialog are already established 
where needed.

I am able to get the receiving device called now but I do not see media being 
sent to it.

Does this help explain my intentions and needs?

Thanks,
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: Thursday, February 4, 2021 7:48 AM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Media Exchange API media_fork_to_uri

[EXTERNAL] 

Hi, Eugene!

The idea is simple: a socket/listener represents an 
interface+protocol+port OpenSIPS listens on and is able to use.
In your initial tests, you were trying to send the call to a UDP endpoint, but 
you were only defining a TCP listener - hence the failure.

Please let us know *what* you are trying to achieve so we can guide you to the 
right documentation.

Best regards,

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

On 2/2/21 9:08 PM, Eugene Christensen wrote:
> 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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