From s.kainz at wnt.at  Thu Oct  1 05:30:45 2020
From: s.kainz at wnt.at (Stefan Kainz)
Date: Thu, 1 Oct 2020 05:30:45 +0000
Subject: [Freeswitch-users] Microsoft Teams integration
In-Reply-To: <A94AB550-C012-416A-BACB-0009121B0761@drachtio.org>
References: <VI1P190MB04944EF9D1570CDE35F3E78C863D0@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <D89498C5-DB0D-4C06-ADAC-66E09FBCCBEE@drachtio.org>
 <VI1P190MB04949434E780D5D125633E4386380@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>,
 <A94AB550-C012-416A-BACB-0009121B0761@drachtio.org>
Message-ID: <VI1P190MB0494E51F6504798137148DEB86300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>

Im following your advice now, and dont add record-route headers.
Im trying an outbound call ( teams -> pstn ) and it disconnects soon after freeswitch sends the OK.

Here is the OK:


send 1695 bytes to tls/[52.114.75.24]:10176 at 07:12:26.210854:

------------------------------------------------------------------------

SIP/2.0 200 OK

Via: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bKbc16542e;rport=10176

Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>

FROM: Stefan Kainz<sip:+FROMNUMBER at sip.pstnhub.microsoft.com:5061;user=phone>;tag=b88e3a220f48472cb452fec5fe46579e

To: <sip:+TONUMBER at xxx.example.at:5061;user=phone>;tag=9FFZm2y4cBmrS

CALL-ID: 6a933a0ac454552081ca931b3daf42de

CSEQ: 1 INVITE

Contact: <sip:+TONUMBER at xxx.xxx.xxx.xxx:5061;transport=tls>

User-Agent: TTG01 1.0

Accept: application/sdp

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY

Require: timer

Supported: timer, path, replaces

Allow-Events: talk, hold, conference, refer

Session-Expires: 3600;refresher=uac

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 801

P-Asserted-Identity: "+TONUMBER" <sip:+TONUMBER at xxx.example.at>


v=0

o=FreeSWITCH 1601512094 1601512095 IN IP4 xxx.xxx.xxx.xxx

s=FreeSWITCH

c=IN IP4 xxx.xxx.xxx.xxx

t=0 0

a=msid-semantic: WMS pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF

m=audio 17052 RTP/SAVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-mux

a=rtcp:17052 IN IP4 xxx.xxx.xxx.xxx

a=ice-ufrag:czL8C9GwWgQdfSIK

a=ice-pwd:gICzT6c1eiFHGhUJGvI8vevi

a=candidate:2023834489 1 udp 659136 xxx.xxx.xxx.xxx 17052 typ host generation 0

a=end-of-candidates

a=ssrc:1804998514 cname:Fy91tlEMiigNDyLm

a=ssrc:1804998514 msid:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF a0

a=ssrc:1804998514 mslabel:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF

a=ssrc:1804998514 label:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cFa0

a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:aoXrZjxmNQmdkxShUkmMO526J+5laS1UL8ZWiKyS

I compared it with your examples, and noticed that i have an ip-address in contact-header, you on the other hand have a domain.
Do you think that could be the problem?

regards,


________________________________
Von: David Horton <daveh at drachtio.org>
Gesendet: Mittwoch, 23. September 2020 19:24
An: Stefan Kainz <s.kainz at wnt.at>
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Betreff: Re: [Freeswitch-users] Microsoft Teams integration

In my implementation I am using a drachtio server (https://drachtio.org), more specifically configured as part of the jambonz CPaaS platform I build (https://docs.jambonz.org/teams/)

A Record-Route header only makes sense if you are creating a sip proxy, AFAIC.  In my implementation I built a B2BUA so on requests I send to Teams there is no RR.  Of course, on responses I preserve their RR headers.

For your scenario, I think I passed on the trace but just in case here is a link to it again: https://gist.github.com/davehorton/9ea3d79e8eeda3cbb5e2e87209dcfe63

One question: for the domain you are putting in the Contact header of your 200 OK response, does that resolve to a single server or multiple?  I have found in the latter case Teams will gladly send a mid-call request to a different server that heretofore was not involved in the call at all.  On the server that initially handled the call, it will simply look like Teams is not being responsive..

Dave

On Sep 23, 2020, at 3:56 AM, Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>> wrote:

Thank you for your Answer!

May I ask what kind of SBC you are using?
We also use Audiocodes SBC’s, which are working great, but even those send Record-Route Headers.

Maybe I need to explain the szenario a little better…
In the teams client I dial an external number ( my mobile phone for example )
I pick up, and then push hold in the teams client. ( So far so good )
When I try to unhold, of course also in teams, it becomes silent on both ends. ( even the moh stops )

And then as I said I see a few OK Messages being retransmitted to the Microsoft servers.

Its funny that you say you don’t need Record-Route Headers at all, because without setting those, I cant even make incoming calls work.
With those headers, everything works perfectly except for the thing explained above.
Even when trying to get incoming calls back from hold on teams. ( PSTN – freeswitch – teams, hold / unhold on teams )

I also found a tutorial from kamailio on how to connect it to teams.
There they also state that the record-route headers a needed.
https://skalatan.de/en/blog/kamailio-sbc-teams

Thank you again David!

Regards,
Stefan


From: David Horton <daveh at drachtio.org<mailto:daveh at drachtio.org>>
Sent: Montag, 21. September 2020 22:36
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Cc: Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>>
Subject: Re: [Freeswitch-users] Microsoft Teams integration

It has not been my experience that Microsoft Teams direct routing requires a Record-Route header on requests / responses you send to it (though their requests will have an RR and you must Route accordingly)

Here is a sip trace showing a successful on-hold / off-hold scenario from one of my servers (not a freeswitch server).  This is an on-hold triggered from the MS teams client, not sure if you are trying to do the reverse.

One thing: make sure the Contact header in your INVITEs resolve to a single server, otherwise (if you have mutiple A records for the DNS name) you will find that MS Teams will gladly respond to a reINVITE to a completely different server than you sent from — this will manifest as not seeing the 200 OK to a re-INVITE you send, or not seeing the ACK when you respond 200 OK to a re-INVITE.

daveh

One thing to make sure of is that

On Sep 20, 2020, at 12:05 PM, Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>> wrote:

Hello,

we're trying to use Freeswitch as a Microsoft Teams direct routing SBC.

So far everything works perfectly, except getting the call back from hold on an outbound call.
The Problem is that Microsoft expects a Record-Route Header in every Request and every Response.
I have been able to set/rewrite this Record-Route header with sip_h_ and sip_rh_ in most szenarios, but like i said - unhold on an outgoing call does not work.

The hold is initiated with a reinvite with a=inactive in sdp, which does work.
However, the unhold - also a reinvite with a=sendrecv in sdp does not work correctly.
In the sip-trace i see that the OK ( in response to the INVITE for unhold ) gets retransmitted a couple of times, because the ACK from microsoft isn't received.

This is due to the missing Record-Route Header in the OK Message.
I have seen this behavior on inbound calls too, when i didnt set the Record-Route Header.

Now my question:
Is there any way to add the Record-Route Header to every Message on a sip-profile or gateway?
Or if not, is it possible to handle a reinvite in dialplan so i could set the header manually?
Maybe there is somebody who already has this working that could point me in the right direction.

I know freeswitch isn't a SBC, but it would be nice to be able to use freeswitch as a Teams-SBC without Kamailio or something similar.

Any help is very much appreciated 🙂

regards,
Stefan







_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com<https://signalwire.com/>
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community<https://signalwire.community/>

Professional FreeSWITCH Services
sales at freeswitch.com<mailto:sales at freeswitch.com>
https://freeswitch.com<https://freeswitch.com/>

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com<https://cluecon.com/>

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From daveh at drachtio.org  Thu Oct  1 12:19:48 2020
From: daveh at drachtio.org (David Horton)
Date: Thu, 1 Oct 2020 08:19:48 -0400
Subject: [Freeswitch-users] Microsoft Teams integration
In-Reply-To: <VI1P190MB0494E51F6504798137148DEB86300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
References: <VI1P190MB04944EF9D1570CDE35F3E78C863D0@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <D89498C5-DB0D-4C06-ADAC-66E09FBCCBEE@drachtio.org>
 <VI1P190MB04949434E780D5D125633E4386380@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <A94AB550-C012-416A-BACB-0009121B0761@drachtio.org>
 <VI1P190MB0494E51F6504798137148DEB86300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
Message-ID: <A384C846-2B00-456F-9ACC-73ED703FBD39@drachtio.org>

Yes, you must have a domain in the Contact header.  That’s another ridiculous requirement Microsoft came up with for this integration...

On Oct 1, 2020, at 1:30 AM, Stefan Kainz <s.kainz at wnt.at> wrote:

Im following your advice now, and dont add record-route headers. 
Im trying an outbound call ( teams -> pstn ) and it disconnects soon after freeswitch sends the OK.

Here is the OK: 

send 1695 bytes to tls/[52.114.75.24]:10176 at 07:12:26.210854:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bKbc16542e;rport=10176
Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>>
FROM: Stefan Kainz<sip:+FROMNUMBER at sip.pstnhub.microsoft.com:5061;user=phone <sip:+FROMNUMBER at sip.pstnhub.microsoft.com:5061;user=phone>>;tag=b88e3a220f48472cb452fec5fe46579e
To: <sip:+TONUMBER at xxx.example.at:5061;user=phone <sip:+TONUMBER at xxx.example.at:5061;user=phone>>;tag=9FFZm2y4cBmrS
CALL-ID: 6a933a0ac454552081ca931b3daf42de
CSEQ: 1 INVITE
Contact: <sip:+TONUMBER at xxx.xxx.xxx.xxx:5061;transport=tls <sip:+TONUMBER at xxx.xxx.xxx.xxx:5061;transport=tls>>
User-Agent: TTG01 1.0
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY
Require: timer
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Session-Expires: 3600;refresher=uac
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 801
P-Asserted-Identity: "+TONUMBER" <sip:+TONUMBER at xxx.example.at <sip:+TONUMBER at xxx.example.at>>

v=0
o=FreeSWITCH 1601512094 1601512095 IN IP4 xxx.xxx.xxx.xxx
s=FreeSWITCH
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
a=msid-semantic: WMS pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF
m=audio 17052 RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-mux
a=rtcp:17052 IN IP4 xxx.xxx.xxx.xxx
a=ice-ufrag:czL8C9GwWgQdfSIK
a=ice-pwd:gICzT6c1eiFHGhUJGvI8vevi
a=candidate:2023834489 1 udp 659136 xxx.xxx.xxx.xxx 17052 typ host generation 0
a=end-of-candidates
a=ssrc:1804998514 cname:Fy91tlEMiigNDyLm
a=ssrc:1804998514 msid:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF a0
a=ssrc:1804998514 mslabel:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF
a=ssrc:1804998514 label:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cFa0
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:aoXrZjxmNQmdkxShUkmMO526J+5laS1UL8ZWiKyS

I compared it with your examples, and noticed that i have an ip-address in contact-header, you on the other hand have a domain. 
Do you think that could be the problem?

regards, 


Von: David Horton <daveh at drachtio.org>
Gesendet: Mittwoch, 23. September 2020 19:24
An: Stefan Kainz <s.kainz at wnt.at>
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Betreff: Re: [Freeswitch-users] Microsoft Teams integration
 
In my implementation I am using a drachtio server (https://drachtio <https://drachtio/>.org), more specifically configured as part of the jambonz CPaaS platform I build (https://docs.jambonz.org/teams/ <https://docs.jambonz.org/teams/>)

A Record-Route header only makes sense if you are creating a sip proxy, AFAIC.  In my implementation I built a B2BUA so on requests I send to Teams there is no RR.  Of course, on responses I preserve their RR headers.

For your scenario, I think I passed on the trace but just in case here is a link to it again: https://gist.github.com/davehorton/9ea3d79e8eeda3cbb5e2e87209dcfe63 <https://gist.github.com/davehorton/9ea3d79e8eeda3cbb5e2e87209dcfe63>

One question: for the domain you are putting in the Contact header of your 200 OK response, does that resolve to a single server or multiple?  I have found in the latter case Teams will gladly send a mid-call request to a different server that heretofore was not involved in the call at all.  On the server that initially handled the call, it will simply look like Teams is not being responsive..

Dave

On Sep 23, 2020, at 3:56 AM, Stefan Kainz <s.kainz at wnt.at <mailto:s.kainz at wnt.at>> wrote:

Thank you for your Answer!
 
May I ask what kind of SBC you are using?
We also use Audiocodes SBC’s, which are working great, but even those send Record-Route Headers.
 
Maybe I need to explain the szenario a little better…
In the teams client I dial an external number ( my mobile phone for example )
I pick up, and then push hold in the teams client. ( So far so good )
When I try to unhold, of course also in teams, it becomes silent on both ends. ( even the moh stops )
 
And then as I said I see a few OK Messages being retransmitted to the Microsoft servers.
 
Its funny that you say you don’t need Record-Route Headers at all, because without setting those, I cant even make incoming calls work.
With those headers, everything works perfectly except for the thing explained above.
Even when trying to get incoming calls back from hold on teams. ( PSTN – freeswitch – teams, hold / unhold on teams )
 
I also found a tutorial from kamailio on how to connect it to teams.
There they also state that the record-route headers a needed.
https://skalatan.de/en/blog/kamailio-sbc-teams <https://skalatan.de/en/blog/kamailio-sbc-teams>
 
Thank you again David!
 
Regards,
Stefan
 
 
From: David Horton <daveh at drachtio.org <mailto:daveh at drachtio.org>> 
Sent: Montag, 21. September 2020 22:36
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
Cc: Stefan Kainz <s.kainz at wnt.at <mailto:s.kainz at wnt.at>>
Subject: Re: [Freeswitch-users] Microsoft Teams integration
 
It has not been my experience that Microsoft Teams direct routing requires a Record-Route header on requests / responses you send to it (though their requests will have an RR and you must Route accordingly)
 
Here is a sip trace showing a successful on-hold / off-hold scenario from one of my servers (not a freeswitch server).  This is an on-hold triggered from the MS teams client, not sure if you are trying to do the reverse.
 
One thing: make sure the Contact header in your INVITEs resolve to a single server, otherwise (if you have mutiple A records for the DNS name) you will find that MS Teams will gladly respond to a reINVITE to a completely different server than you sent from — this will manifest as not seeing the 200 OK to a re-INVITE you send, or not seeing the ACK when you respond 200 OK to a re-INVITE.
 
daveh
 
One thing to make sure of is that 
 
On Sep 20, 2020, at 12:05 PM, Stefan Kainz <s.kainz at wnt.at <mailto:s.kainz at wnt.at>> wrote:
 
Hello, 
 
we're trying to use Freeswitch as a Microsoft Teams direct routing SBC.
 
So far everything works perfectly, except getting the call back from hold on an outbound call. 
The Problem is that Microsoft expects a Record-Route Header in every Request and every Response.
I have been able to set/rewrite this Record-Route header with sip_h_ and sip_rh_ in most szenarios, but like i said - unhold on an outgoing call does not work.
 
The hold is initiated with a reinvite with a=inactive in sdp, which does work. 
However, the unhold - also a reinvite with a=sendrecv in sdp does not work correctly.
In the sip-trace i see that the OK ( in response to the INVITE for unhold ) gets retransmitted a couple of times, because the ACK from microsoft isn't received. 
 
This is due to the missing Record-Route Header in the OK Message.
I have seen this behavior on inbound calls too, when i didnt set the Record-Route Header.
 
Now my question:
Is there any way to add the Record-Route Header to every Message on a sip-profile or gateway? 
Or if not, is it possible to handle a reinvite in dialplan so i could set the header manually?
Maybe there is somebody who already has this working that could point me in the right direction.
 
I know freeswitch isn't a SBC, but it would be nice to be able to use freeswitch as a Teams-SBC without Kamailio or something similar.
 
Any help is very much appreciated 🙂
 
regards, 
Stefan
 
 
 
 
 
 
 
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>

Professional FreeSWITCH Services
sales at freeswitch.com <mailto:sales at freeswitch.com>
https://freeswitch.com <https://freeswitch.com/>

Official FreeSWITCH Sites
https://freeswitch.com/oss <https://freeswitch.com/oss>
https://freeswitch.org/confluence <https://freeswitch.org/confluence>
https://cluecon.com <https://cluecon.com/>

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
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From s.kainz at wnt.at  Thu Oct  1 14:20:52 2020
From: s.kainz at wnt.at (Stefan Kainz)
Date: Thu, 1 Oct 2020 14:20:52 +0000
Subject: [Freeswitch-users] Microsoft Teams integration
In-Reply-To: <A384C846-2B00-456F-9ACC-73ED703FBD39@drachtio.org>
References: <VI1P190MB04944EF9D1570CDE35F3E78C863D0@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <D89498C5-DB0D-4C06-ADAC-66E09FBCCBEE@drachtio.org>
 <VI1P190MB04949434E780D5D125633E4386380@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <A94AB550-C012-416A-BACB-0009121B0761@drachtio.org>
 <VI1P190MB0494E51F6504798137148DEB86300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <A384C846-2B00-456F-9ACC-73ED703FBD39@drachtio.org>
Message-ID: <VI1P190MB049482B1DCA37658CB9A87C586300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>

Ok, then i might have found whats causing all my problems.
Would you happen to know how I achive this in freeswitch?

Ive tried a couple of things already, but none seem to work.
The gateway has “contact-host” set correctly, I also tried sip_contact_host …

Thank you very much!

Regards,

From: David Horton <daveh at drachtio.org>
Sent: Donnerstag, 1. Oktober 2020 14:20
To: Stefan Kainz <s.kainz at wnt.at>
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Microsoft Teams integration

Yes, you must have a domain in the Contact header.  That’s another ridiculous requirement Microsoft came up with for this integration...

On Oct 1, 2020, at 1:30 AM, Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>> wrote:

Im following your advice now, and dont add record-route headers.
Im trying an outbound call ( teams -> pstn ) and it disconnects soon after freeswitch sends the OK.

Here is the OK:

send 1695 bytes to tls/[52.114.75.24]:10176 at 07:12:26.210854:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bKbc16542e;rport=10176
Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>
FROM: Stefan Kainz<sip:+FROMNUMBER at sip.pstnhub.microsoft.com:5061;user=phone>;tag=b88e3a220f48472cb452fec5fe46579e
To: <sip:+TONUMBER at xxx.example.at:5061;user=phone>;tag=9FFZm2y4cBmrS
CALL-ID: 6a933a0ac454552081ca931b3daf42de
CSEQ: 1 INVITE
Contact: <sip:+TONUMBER at xxx.xxx.xxx.xxx:5061;transport=tls>
User-Agent: TTG01 1.0
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY
Require: timer
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Session-Expires: 3600;refresher=uac
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 801
P-Asserted-Identity: "+TONUMBER" <sip:+TONUMBER at xxx.example.at>

v=0
o=FreeSWITCH 1601512094 1601512095 IN IP4 xxx.xxx.xxx.xxx
s=FreeSWITCH
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
a=msid-semantic: WMS pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF
m=audio 17052 RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-mux
a=rtcp:17052 IN IP4 xxx.xxx.xxx.xxx
a=ice-ufrag:czL8C9GwWgQdfSIK
a=ice-pwd:gICzT6c1eiFHGhUJGvI8vevi
a=candidate:2023834489 1 udp 659136 xxx.xxx.xxx.xxx 17052 typ host generation 0
a=end-of-candidates
a=ssrc:1804998514 cname:Fy91tlEMiigNDyLm
a=ssrc:1804998514 msid:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF a0
a=ssrc:1804998514 mslabel:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF
a=ssrc:1804998514 label:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cFa0
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:aoXrZjxmNQmdkxShUkmMO526J+5laS1UL8ZWiKyS

I compared it with your examples, and noticed that i have an ip-address in contact-header, you on the other hand have a domain.
Do you think that could be the problem?

regards,


________________________________
Von: David Horton <daveh at drachtio.org<mailto:daveh at drachtio.org>>
Gesendet: Mittwoch, 23. September 2020 19:24
An: Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>>
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Betreff: Re: [Freeswitch-users] Microsoft Teams integration

In my implementation I am using a drachtio server (https://drachtio<https://drachtio/>.org), more specifically configured as part of the jambonz CPaaS platform I build (https://docs.jambonz.org/teams/)

A Record-Route header only makes sense if you are creating a sip proxy, AFAIC.  In my implementation I built a B2BUA so on requests I send to Teams there is no RR.  Of course, on responses I preserve their RR headers.

For your scenario, I think I passed on the trace but just in case here is a link to it again: https://gist.github.com/davehorton/9ea3d79e8eeda3cbb5e2e87209dcfe63

One question: for the domain you are putting in the Contact header of your 200 OK response, does that resolve to a single server or multiple?  I have found in the latter case Teams will gladly send a mid-call request to a different server that heretofore was not involved in the call at all.  On the server that initially handled the call, it will simply look like Teams is not being responsive..

Dave

On Sep 23, 2020, at 3:56 AM, Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>> wrote:

Thank you for your Answer!

May I ask what kind of SBC you are using?
We also use Audiocodes SBC’s, which are working great, but even those send Record-Route Headers.

Maybe I need to explain the szenario a little better…
In the teams client I dial an external number ( my mobile phone for example )
I pick up, and then push hold in the teams client. ( So far so good )
When I try to unhold, of course also in teams, it becomes silent on both ends. ( even the moh stops )

And then as I said I see a few OK Messages being retransmitted to the Microsoft servers.

Its funny that you say you don’t need Record-Route Headers at all, because without setting those, I cant even make incoming calls work.
With those headers, everything works perfectly except for the thing explained above.
Even when trying to get incoming calls back from hold on teams. ( PSTN – freeswitch – teams, hold / unhold on teams )

I also found a tutorial from kamailio on how to connect it to teams.
There they also state that the record-route headers a needed.
https://skalatan.de/en/blog/kamailio-sbc-teams

Thank you again David!

Regards,
Stefan


From: David Horton <daveh at drachtio.org<mailto:daveh at drachtio.org>>
Sent: Montag, 21. September 2020 22:36
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Cc: Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>>
Subject: Re: [Freeswitch-users] Microsoft Teams integration

It has not been my experience that Microsoft Teams direct routing requires a Record-Route header on requests / responses you send to it (though their requests will have an RR and you must Route accordingly)

Here is a sip trace showing a successful on-hold / off-hold scenario from one of my servers (not a freeswitch server).  This is an on-hold triggered from the MS teams client, not sure if you are trying to do the reverse.

One thing: make sure the Contact header in your INVITEs resolve to a single server, otherwise (if you have mutiple A records for the DNS name) you will find that MS Teams will gladly respond to a reINVITE to a completely different server than you sent from — this will manifest as not seeing the 200 OK to a re-INVITE you send, or not seeing the ACK when you respond 200 OK to a re-INVITE.

daveh

One thing to make sure of is that

On Sep 20, 2020, at 12:05 PM, Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>> wrote:

Hello,

we're trying to use Freeswitch as a Microsoft Teams direct routing SBC.

So far everything works perfectly, except getting the call back from hold on an outbound call.
The Problem is that Microsoft expects a Record-Route Header in every Request and every Response.
I have been able to set/rewrite this Record-Route header with sip_h_ and sip_rh_ in most szenarios, but like i said - unhold on an outgoing call does not work.

The hold is initiated with a reinvite with a=inactive in sdp, which does work.
However, the unhold - also a reinvite with a=sendrecv in sdp does not work correctly.
In the sip-trace i see that the OK ( in response to the INVITE for unhold ) gets retransmitted a couple of times, because the ACK from microsoft isn't received.

This is due to the missing Record-Route Header in the OK Message.
I have seen this behavior on inbound calls too, when i didnt set the Record-Route Header.

Now my question:
Is there any way to add the Record-Route Header to every Message on a sip-profile or gateway?
Or if not, is it possible to handle a reinvite in dialplan so i could set the header manually?
Maybe there is somebody who already has this working that could point me in the right direction.

I know freeswitch isn't a SBC, but it would be nice to be able to use freeswitch as a Teams-SBC without Kamailio or something similar.

Any help is very much appreciated 🙂

regards,
Stefan







_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com<https://signalwire.com/>
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community<https://signalwire.community/>

Professional FreeSWITCH Services
sales at freeswitch.com<mailto:sales at freeswitch.com>
https://freeswitch.com<https://freeswitch.com/>

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
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From daveh at drachtio.org  Thu Oct  1 15:02:41 2020
From: daveh at drachtio.org (David Horton)
Date: Thu, 1 Oct 2020 11:02:41 -0400
Subject: [Freeswitch-users] Microsoft Teams integration
In-Reply-To: <VI1P190MB049482B1DCA37658CB9A87C586300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
References: <VI1P190MB04944EF9D1570CDE35F3E78C863D0@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <D89498C5-DB0D-4C06-ADAC-66E09FBCCBEE@drachtio.org>
 <VI1P190MB04949434E780D5D125633E4386380@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <A94AB550-C012-416A-BACB-0009121B0761@drachtio.org>
 <VI1P190MB0494E51F6504798137148DEB86300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <A384C846-2B00-456F-9ACC-73ED703FBD39@drachtio.org>
 <VI1P190MB049482B1DCA37658CB9A87C586300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
Message-ID: <CD20FD48-EB37-4E4A-866C-F5555ECA742D@drachtio.org>

I’m sure there is a way to do it in Freeswitch, but I myself don’t know how as I’ve never needed to do that.  Maybe others can chime in..

It may be obvious, but I should have mentioned that the DNS name in the Contact header must resolve to your server either via SRV or A records.  Secondly, you do not want to have multiple A records for the DNS name (as for instance if you had a bank of freeswitch servers) because if you do Microsoft will happily send a mid-call reINVITE (or BYE) to a different server than sent it the INVITE in the first place.

On Oct 1, 2020, at 10:20 AM, Stefan Kainz <s.kainz at wnt.at> wrote:

Ok, then i might have found whats causing all my problems.
Would you happen to know how I achive this in freeswitch?
 
Ive tried a couple of things already, but none seem to work.
The gateway has “contact-host” set correctly, I also tried sip_contact_host …
 
Thank you very much!
 
Regards,
 
From: David Horton <daveh at drachtio.org> 
Sent: Donnerstag, 1. Oktober 2020 14:20
To: Stefan Kainz <s.kainz at wnt.at>
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Microsoft Teams integration
 
Yes, you must have a domain in the Contact header.  That’s another ridiculous requirement Microsoft came up with for this integration...
 
On Oct 1, 2020, at 1:30 AM, Stefan Kainz <s.kainz at wnt.at <mailto:s.kainz at wnt.at>> wrote:
 
Im following your advice now, and dont add record-route headers. 
Im trying an outbound call ( teams -> pstn ) and it disconnects soon after freeswitch sends the OK.
 
Here is the OK: 
 
send 1695 bytes to tls/[52.114.75.24]:10176 at 07:12:26.210854:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bKbc16542e;rport=10176
Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>>
FROM: Stefan Kainz<sip:+FROMNUMBER at sip.pstnhub.microsoft.com:5061;user=phone <sip:+FROMNUMBER at sip.pstnhub.microsoft.com:5061;user=phone>>;tag=b88e3a220f48472cb452fec5fe46579e
To: <sip:+TONUMBER at xxx.example.at:5061;user=phone <sip:+TONUMBER at xxx.example.at:5061;user=phone>>;tag=9FFZm2y4cBmrS
CALL-ID: 6a933a0ac454552081ca931b3daf42de
CSEQ: 1 INVITE
Contact: <sip:+TONUMBER at xxx.xxx.xxx.xxx:5061;transport=tls <sip:+TONUMBER at xxx.xxx.xxx.xxx:5061;transport=tls>>
User-Agent: TTG01 1.0
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY
Require: timer
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Session-Expires: 3600;refresher=uac
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 801
P-Asserted-Identity: "+TONUMBER" <sip:+TONUMBER at xxx.example.at <sip:+TONUMBER at xxx.example.at>>
 
v=0
o=FreeSWITCH 1601512094 1601512095 IN IP4 xxx.xxx.xxx.xxx
s=FreeSWITCH
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
a=msid-semantic: WMS pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF
m=audio 17052 RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-mux
a=rtcp:17052 IN IP4 xxx.xxx.xxx.xxx
a=ice-ufrag:czL8C9GwWgQdfSIK
a=ice-pwd:gICzT6c1eiFHGhUJGvI8vevi
a=candidate:2023834489 1 udp 659136 xxx.xxx.xxx.xxx 17052 typ host generation 0
a=end-of-candidates
a=ssrc:1804998514 cname:Fy91tlEMiigNDyLm
a=ssrc:1804998514 msid:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF a0
a=ssrc:1804998514 mslabel:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF
a=ssrc:1804998514 label:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cFa0
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:aoXrZjxmNQmdkxShUkmMO526J+5laS1UL8ZWiKyS
 
I compared it with your examples, and noticed that i have an ip-address in contact-header, you on the other hand have a domain. 
Do you think that could be the problem?
 
regards, 
 
 
Von: David Horton <daveh at drachtio.org <mailto:daveh at drachtio.org>>
Gesendet: Mittwoch, 23. September 2020 19:24
An: Stefan Kainz <s.kainz at wnt.at <mailto:s.kainz at wnt.at>>
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
Betreff: Re: [Freeswitch-users] Microsoft Teams integration
 
In my implementation I am using a drachtio server (https://drachtio <https://drachtio/>.org), more specifically configured as part of the jambonz CPaaS platform I build (https://docs.jambonz.org/teams/ <https://docs.jambonz.org/teams/>)
 
A Record-Route header only makes sense if you are creating a sip proxy, AFAIC.  In my implementation I built a B2BUA so on requests I send to Teams there is no RR.  Of course, on responses I preserve their RR headers.
 
For your scenario, I think I passed on the trace but just in case here is a link to it again: https://gist.github.com/davehorton/9ea3d79e8eeda3cbb5e2e87209dcfe63 <https://gist.github.com/davehorton/9ea3d79e8eeda3cbb5e2e87209dcfe63>
 
One question: for the domain you are putting in the Contact header of your 200 OK response, does that resolve to a single server or multiple?  I have found in the latter case Teams will gladly send a mid-call request to a different server that heretofore was not involved in the call at all.  On the server that initially handled the call, it will simply look like Teams is not being responsive..
 
Dave
 
On Sep 23, 2020, at 3:56 AM, Stefan Kainz <s.kainz at wnt.at <mailto:s.kainz at wnt.at>> wrote:
 
Thank you for your Answer!
 
May I ask what kind of SBC you are using?
We also use Audiocodes SBC’s, which are working great, but even those send Record-Route Headers.
 
Maybe I need to explain the szenario a little better…
In the teams client I dial an external number ( my mobile phone for example )
I pick up, and then push hold in the teams client. ( So far so good )
When I try to unhold, of course also in teams, it becomes silent on both ends. ( even the moh stops )
 
And then as I said I see a few OK Messages being retransmitted to the Microsoft servers.
 
Its funny that you say you don’t need Record-Route Headers at all, because without setting those, I cant even make incoming calls work.
With those headers, everything works perfectly except for the thing explained above.
Even when trying to get incoming calls back from hold on teams. ( PSTN – freeswitch – teams, hold / unhold on teams )
 
I also found a tutorial from kamailio on how to connect it to teams.
There they also state that the record-route headers a needed.
https://skalatan.de/en/blog/kamailio-sbc-teams <https://skalatan.de/en/blog/kamailio-sbc-teams>
 
Thank you again David!
 
Regards,
Stefan
 
 
From: David Horton <daveh at drachtio.org <mailto:daveh at drachtio.org>> 
Sent: Montag, 21. September 2020 22:36
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
Cc: Stefan Kainz <s.kainz at wnt.at <mailto:s.kainz at wnt.at>>
Subject: Re: [Freeswitch-users] Microsoft Teams integration
 
It has not been my experience that Microsoft Teams direct routing requires a Record-Route header on requests / responses you send to it (though their requests will have an RR and you must Route accordingly)
 
Here is a sip trace showing a successful on-hold / off-hold scenario from one of my servers (not a freeswitch server).  This is an on-hold triggered from the MS teams client, not sure if you are trying to do the reverse.
 
One thing: make sure the Contact header in your INVITEs resolve to a single server, otherwise (if you have mutiple A records for the DNS name) you will find that MS Teams will gladly respond to a reINVITE to a completely different server than you sent from — this will manifest as not seeing the 200 OK to a re-INVITE you send, or not seeing the ACK when you respond 200 OK to a re-INVITE.
 
daveh
 
One thing to make sure of is that 
 
On Sep 20, 2020, at 12:05 PM, Stefan Kainz <s.kainz at wnt.at <mailto:s.kainz at wnt.at>> wrote:
 
Hello, 
 
we're trying to use Freeswitch as a Microsoft Teams direct routing SBC.
 
So far everything works perfectly, except getting the call back from hold on an outbound call. 
The Problem is that Microsoft expects a Record-Route Header in every Request and every Response.
I have been able to set/rewrite this Record-Route header with sip_h_ and sip_rh_ in most szenarios, but like i said - unhold on an outgoing call does not work.
 
The hold is initiated with a reinvite with a=inactive in sdp, which does work. 
However, the unhold - also a reinvite with a=sendrecv in sdp does not work correctly.
In the sip-trace i see that the OK ( in response to the INVITE for unhold ) gets retransmitted a couple of times, because the ACK from microsoft isn't received. 
 
This is due to the missing Record-Route Header in the OK Message.
I have seen this behavior on inbound calls too, when i didnt set the Record-Route Header.
 
Now my question:
Is there any way to add the Record-Route Header to every Message on a sip-profile or gateway? 
Or if not, is it possible to handle a reinvite in dialplan so i could set the header manually?
Maybe there is somebody who already has this working that could point me in the right direction.
 
I know freeswitch isn't a SBC, but it would be nice to be able to use freeswitch as a Teams-SBC without Kamailio or something similar.
 
Any help is very much appreciated 🙂
 
regards, 
Stefan
 
 
 
 
 
 
 
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>

Professional FreeSWITCH Services
sales at freeswitch.com <mailto:sales at freeswitch.com>
https://freeswitch.com <https://freeswitch.com/>

Official FreeSWITCH Sites
https://freeswitch.com/oss <https://freeswitch.com/oss>
https://freeswitch.org/confluence <https://freeswitch.org/confluence>
https://cluecon.com <https://cluecon.com/>

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
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From mbrancaleoni at voismart.it  Thu Oct  1 15:04:42 2020
From: mbrancaleoni at voismart.it (Matteo)
Date: Thu, 1 Oct 2020 17:04:42 +0200 (CEST)
Subject: [Freeswitch-users] Microsoft Teams integration
In-Reply-To: <VI1P190MB049482B1DCA37658CB9A87C586300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
References: <VI1P190MB04944EF9D1570CDE35F3E78C863D0@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <D89498C5-DB0D-4C06-ADAC-66E09FBCCBEE@drachtio.org>
 <VI1P190MB04949434E780D5D125633E4386380@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <A94AB550-C012-416A-BACB-0009121B0761@drachtio.org>
 <VI1P190MB0494E51F6504798137148DEB86300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <A384C846-2B00-456F-9ACC-73ED703FBD39@drachtio.org>
 <VI1P190MB049482B1DCA37658CB9A87C586300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
Message-ID: <934828332.44218.1601564682210.JavaMail.zimbra@voismart.it>

Put your hostname in the sip-ip sofia profile config.

Be sure that the hostname is resolvable by FS and point to the ip address you're going to use (put in etc hosts for example) and should do the trick. You can check if is set by using "sofia status profile foo" and look into the profile details (url, bind-url and so on), it should do the trick.

For example:

freeswitch@**redacted**> sofia status profile external
=================================================================================================
Name             	external
Domain Name      	N/A
Auto-NAT         	false
DBName           	sofia_reg_external
Pres Hosts       	
Dialplan         	XML
Context          	public
Challenge Realm  	auto_to
RTP-IP           	1.2.3.4
SIP-IP           	sbc.**redacted**.com
URL              	sip:mod_sofia at sbc.**redacted**.com:5060
BIND-URL         	sip:mod_sofia at sbc.**redacted**.com:5060;transport=udp,tcp
TLS-URL          	sip:mod_sofia at sbc.**redacted**.com:5061
TLS-BIND-URL     	sips:mod_sofia at sbc.**redacted**.com:5061;transport=tls
HOLD-MUSIC       	local_stream://moh
OUTBOUND-PROXY   	N/A
....

Relevant part of the sofia profile config:

[root@**redacted** ~]# grep sip-ip /etc/freeswitch/sip_profiles/external.xml 
    <param name="sip-ip" value="sbc.**redacted**.com"/>
    <param name="ext-sip-ip" value="$${external_rtp_ip}"/>


I've tried to dig into the sources, and seems that the contact is built directly from that sip-ip parameter and currently don't know if is possible to override it easily.

Is super important that the hostname is resolvable correctly with the FS box.

Also, the TLS certi must be valid (we use LE) and have the contact hostname among the validaded entries.

jm2c,
Mat

----- Il 1-ott-20, alle 16:20, Stefan Kainz s.kainz at wnt.at ha scritto:

> Ok, then i might have found whats causing all my problems.
> 
> Would you happen to know how I achive this in freeswitch?
> 
> 
> 
> Ive tried a couple of things already, but none seem to work.
> 
> The gateway has “contact-host” set correctly, I also tried sip_contact_host …
> 
> 
> 
> Thank you very much!
> 
> 
> 
> Regards,
> 
> 
> 
> 
> From: David Horton <daveh at drachtio.org>
> Sent: Donnerstag, 1. Oktober 2020 14:20
> To: Stefan Kainz <s.kainz at wnt.at>
> Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Microsoft Teams integration
> 
> 
> 
> 
> Yes, you must have a domain in the Contact header. That’s another ridiculous
> requirement Microsoft came up with for this integration...
> 
> 
> 
> 
> 
> On Oct 1, 2020, at 1:30 AM, Stefan Kainz < [ mailto:s.kainz at wnt.at |
> s.kainz at wnt.at ] > wrote:
> 
> 
> 
> 
> 
> Im following your advice now, and dont add record-route headers.
> 
> 
> Im trying an outbound call ( teams -> pstn ) and it disconnects soon after
> freeswitch sends the OK.
> 
> 
> 
> 
> 
> Here is the OK:
> 
> 
> 
> 
> 
> send 1695 bytes to tls/[52.114.75.24]:10176 at 07:12:26.210854:
> 
> 
> ------------------------------------------------------------------------
> 
> 
> SIP/2.0 200 OK
> 
> 
> Via: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bKbc16542e;rport=10176
> 
> 
> Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>
> 
> 
> FROM: Stefan
> Kainz<sip:+FROMNUMBER at sip.pstnhub.microsoft.com:5061;user=phone>;tag=b88e3a220f48472cb452fec5fe46579e
> 
> 
> To: <sip:+TONUMBER at xxx.example.at:5061;user=phone>;tag=9FFZm2y4cBmrS
> 
> 
> CALL-ID: 6a933a0ac454552081ca931b3daf42de
> 
> 
> CSEQ: 1 INVITE
> 
> 
> Contact: <sip:+TONUMBER at xxx.xxx.xxx.xxx:5061;transport=tls>
> 
> 
> User-Agent: TTG01 1.0
> 
> 
> Accept: application/sdp
> 
> 
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> NOTIFY
> 
> 
> Require: timer
> 
> 
> Supported: timer, path, replaces
> 
> 
> Allow-Events: talk, hold, conference, refer
> 
> 
> Session-Expires: 3600;refresher=uac
> 
> 
> Content-Type: application/sdp
> 
> 
> Content-Disposition: session
> 
> 
> Content-Length: 801
> 
> 
> P-Asserted-Identity: "+TONUMBER" <sip:+TONUMBER at xxx.example.at>
> 
> 
> 
> 
> 
> v=0
> 
> 
> o=FreeSWITCH 1601512094 1601512095 IN IP4 xxx.xxx.xxx.xxx
> 
> 
> s=FreeSWITCH
> 
> 
> c=IN IP4 xxx.xxx.xxx.xxx
> 
> 
> t=0 0
> 
> 
> a=msid-semantic: WMS pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF
> 
> 
> m=audio 17052 RTP/SAVP 0 101
> 
> 
> a=rtpmap:0 PCMU/8000
> 
> 
> a=rtpmap:101 telephone-event/8000
> 
> 
> a=fmtp:101 0-16
> 
> 
> a=ptime:20
> 
> 
> a=rtcp-mux
> 
> 
> a=rtcp:17052 IN IP4 xxx.xxx.xxx.xxx
> 
> 
> a=ice-ufrag:czL8C9GwWgQdfSIK
> 
> 
> a=ice-pwd:gICzT6c1eiFHGhUJGvI8vevi
> 
> 
> a=candidate:2023834489 1 udp 659136 xxx.xxx.xxx.xxx 17052 typ host generation 0
> 
> 
> a=end-of-candidates
> 
> 
> a=ssrc:1804998514 cname:Fy91tlEMiigNDyLm
> 
> 
> a=ssrc:1804998514 msid:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF a0
> 
> 
> a=ssrc:1804998514 mslabel:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF
> 
> 
> a=ssrc:1804998514 label:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cFa0
> 
> 
> a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:aoXrZjxmNQmdkxShUkmMO526J+5laS1UL8ZWiKyS
> 
> 
> 
> 
> 
> I compared it with your examples, and noticed that i have an ip-address in
> contact-header, you on the other hand have a domain.
> 
> 
> Do you think that could be the problem?
> 
> 
> 
> 
> 
> regards,
> 
> 
> 
> 
> 
> 
> 
> 
> 
> Von: David Horton < [ mailto:daveh at drachtio.org | daveh at drachtio.org ] >
> Gesendet: Mittwoch, 23. September 2020 19:24
> An: Stefan Kainz < [ mailto:s.kainz at wnt.at | s.kainz at wnt.at ] >
> Cc: FreeSWITCH Users Help < [ mailto:freeswitch-users at lists.freeswitch.org |
> freeswitch-users at lists.freeswitch.org ] >
> Betreff: Re: [Freeswitch-users] Microsoft Teams integration
> 
> 
> 
> 
> 
> In my implementation I am using a drachtio server ( [ https://drachtio/ |
> https://drachtio ] .org), more specifically configured as part of the jambonz
> CPaaS platform I build ( [ https://docs.jambonz.org/teams/ |
> https://docs.jambonz.org/teams/ ] )
> 
> 
> 
> 
> 
> A Record-Route header only makes sense if you are creating a sip proxy, AFAIC.
> In my implementation I built a B2BUA so on requests I send to Teams there is no
> RR. Of course, on responses I preserve their RR headers.
> 
> 
> 
> 
> 
> For your scenario, I think I passed on the trace but just in case here is a link
> to it again: [
> https://gist.github.com/davehorton/9ea3d79e8eeda3cbb5e2e87209dcfe63 |
> https://gist.github.com/davehorton/9ea3d79e8eeda3cbb5e2e87209dcfe63 ]
> 
> 
> 
> 
> 
> One question: for the domain you are putting in the Contact header of your 200
> OK response, does that resolve to a single server or multiple? I have found in
> the latter case Teams will gladly send a mid-call request to a different server
> that heretofore was not involved in the call at all. On the server that
> initially handled the call, it will simply look like Teams is not being
> responsive..
> 
> 
> 
> 
> 
> Dave
> 
> 
> 
> 
> 
> On Sep 23, 2020, at 3:56 AM, Stefan Kainz < [ mailto:s.kainz at wnt.at |
> s.kainz at wnt.at ] > wrote:
> 
> 
> 
> 
> 
> Thank you for your Answer!
> 
> 
> 
> 
> 
> May I ask what kind of SBC you are using?
> 
> 
> We also use Audiocodes SBC’s, which are working great, but even those send
> Record-Route Headers.
> 
> 
> 
> 
> 
> Maybe I need to explain the szenario a little better…
> 
> 
> In the teams client I dial an external number ( my mobile phone for example )
> 
> 
> I pick up, and then push hold in the teams client. ( So far so good )
> 
> 
> When I try to unhold, of course also in teams, it becomes silent on both ends. (
> even the moh stops )
> 
> 
> 
> 
> 
> And then as I said I see a few OK Messages being retransmitted to the Microsoft
> servers.
> 
> 
> 
> 
> 
> Its funny that you say you don’t need Record-Route Headers at all, because
> without setting those, I cant even make incoming calls work.
> 
> 
> With those headers, everything works perfectly except for the thing explained
> above.
> 
> 
> Even when trying to get incoming calls back from hold on teams. ( PSTN –
> freeswitch – teams, hold / unhold on teams )
> 
> 
> 
> 
> 
> I also found a tutorial from kamailio on how to connect it to teams.
> 
> 
> There they also state that the record-route headers a needed.
> 
> 
> [ https://skalatan.de/en/blog/kamailio-sbc-teams |
> https://skalatan.de/en/blog/kamailio-sbc-teams ]
> 
> 
> 
> 
> 
> Thank you again David!
> 
> 
> 
> 
> 
> Regards,
> 
> 
> Stefan
> 
> 
> 
> 
> 
> 
> 
> 
> From: David Horton < [ mailto:daveh at drachtio.org | daveh at drachtio.org ] >
> Sent: Montag, 21. September 2020 22:36
> To: FreeSWITCH Users Help < [ mailto:freeswitch-users at lists.freeswitch.org |
> freeswitch-users at lists.freeswitch.org ] >
> Cc: Stefan Kainz < [ mailto:s.kainz at wnt.at | s.kainz at wnt.at ] >
> Subject: Re: [Freeswitch-users] Microsoft Teams integration
> 
> 
> 
> 
> 
> It has not been my experience that Microsoft Teams direct routing requires a
> Record-Route header on requests / responses you send to it (though their
> requests will have an RR and you must Route accordingly)
> 
> 
> 
> 
> 
> Here is a sip trace showing a successful on-hold / off-hold scenario from one of
> my servers (not a freeswitch server). This is an on-hold triggered from the MS
> teams client, not sure if you are trying to do the reverse.
> 
> 
> 
> 
> 
> One thing: make sure the Contact header in your INVITEs resolve to a single
> server, otherwise (if you have mutiple A records for the DNS name) you will
> find that MS Teams will gladly respond to a reINVITE to a completely different
> server than you sent from — this will manifest as not seeing the 200 OK to a
> re-INVITE you send, or not seeing the ACK when you respond 200 OK to a
> re-INVITE.
> 
> 
> 
> 
> 
> daveh
> 
> 
> 
> 
> 
> One thing to make sure of is that
> 
> 
> 
> 
> 
> On Sep 20, 2020, at 12:05 PM, Stefan Kainz < [ mailto:s.kainz at wnt.at |
> s.kainz at wnt.at ] > wrote:
> 
> 
> 
> 
> 
> Hello,
> 
> 
> 
> 
> 
> we're trying to use Freeswitch as a Microsoft Teams direct routing SBC.
> 
> 
> 
> 
> 
> So far everything works perfectly, except getting the call back from hold on an
> outbound call.
> 
> 
> The Problem is that Microsoft expects a Record-Route Header in every Request and
> every Response.
> 
> 
> I have been able to set/rewrite this Record-Route header with sip_h_ and sip_rh_
> in most szenarios, but like i said - unhold on an outgoing call does not work.
> 
> 
> 
> 
> 
> The hold is initiated with a reinvite with a=inactive in sdp, which does work.
> 
> 
> However, the unhold - also a reinvite with a=sendrecv in sdp does not work
> correctly.
> 
> 
> In the sip-trace i see that the OK ( in response to the INVITE for unhold ) gets
> retransmitted a couple of times, because the ACK from microsoft isn't received.
> 
> 
> 
> 
> 
> This is due to the missing Record-Route Header in the OK Message.
> 
> 
> I have seen this behavior on inbound calls too, when i didnt set the
> Record-Route Header.
> 
> 
> 
> 
> 
> Now my question:
> 
> 
> Is there any way to add the Record-Route Header to every Message on a
> sip-profile or gateway?
> 
> 
> Or if not, is it possible to handle a reinvite in dialplan so i could set the
> header manually?
> 
> 
> Maybe there is somebody who already has this working that could point me in the
> right direction.
> 
> 
> 
> 
> 
> I know freeswitch isn't a SBC, but it would be nice to be able to use freeswitch
> as a Teams-SBC without Kamailio or something similar.
> 
> 
> 
> 
> 
> Any help is very much appreciated 🙂
> 
> 
> 
> 
> 
> regards,
> 
> 
> Stefan
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire [ https://signalwire.com/ |
> https://signalwire.com ]
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time [ https://signalwire.community/ |
> https://signalwire.community ]
> 
> Professional FreeSWITCH Services
> [ mailto:sales at freeswitch.com | sales at freeswitch.com ]
> [ https://freeswitch.com/ | https://freeswitch.com ]
> 
> Official FreeSWITCH Sites
> [ https://freeswitch.com/oss | https://freeswitch.com/oss ]
> [ https://freeswitch.org/confluence | https://freeswitch.org/confluence ]
> [ https://cluecon.com/ | https://cluecon.com ]
> 
> FreeSWITCH-users mailing list
> [ mailto:FreeSWITCH-users at lists.freeswitch.org |
> FreeSWITCH-users at lists.freeswitch.org ]
> [ http://lists.freeswitch.org/mailman/listinfo/freeswitch-users |
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users ]
> UNSUBSCRIBE: [ http://lists.freeswitch.org/mailman/options/freeswitch-users |
> http://lists.freeswitch.org/mailman/options/freeswitch-users ]
> [ https://freeswitch.com/ | https://freeswitch.com ]
> 
> 
> 
> 
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com


From s.kainz at wnt.at  Thu Oct  1 15:21:09 2020
From: s.kainz at wnt.at (Stefan Kainz)
Date: Thu, 1 Oct 2020 15:21:09 +0000
Subject: [Freeswitch-users] Microsoft Teams integration
In-Reply-To: <CD20FD48-EB37-4E4A-866C-F5555ECA742D@drachtio.org>
References: <VI1P190MB04944EF9D1570CDE35F3E78C863D0@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <D89498C5-DB0D-4C06-ADAC-66E09FBCCBEE@drachtio.org>
 <VI1P190MB04949434E780D5D125633E4386380@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <A94AB550-C012-416A-BACB-0009121B0761@drachtio.org>
 <VI1P190MB0494E51F6504798137148DEB86300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <A384C846-2B00-456F-9ACC-73ED703FBD39@drachtio.org>
 <VI1P190MB049482B1DCA37658CB9A87C586300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <CD20FD48-EB37-4E4A-866C-F5555ECA742D@drachtio.org>
Message-ID: <VI1P190MB049472C0AE2452F2F7D3E69886300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>

Ok, our fqdn does resolve to just one ip. No multiple entries.
Thank you so far David, I’ll try to get the fqdn in contact header and if I cant do it, maybe send another mail to the list.

Thank you very much!

Regards,

From: David Horton <daveh at drachtio.org>
Sent: Donnerstag, 1. Oktober 2020 17:03
To: Stefan Kainz <s.kainz at wnt.at>
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Microsoft Teams integration

I’m sure there is a way to do it in Freeswitch, but I myself don’t know how as I’ve never needed to do that.  Maybe others can chime in..

It may be obvious, but I should have mentioned that the DNS name in the Contact header must resolve to your server either via SRV or A records.  Secondly, you do not want to have multiple A records for the DNS name (as for instance if you had a bank of freeswitch servers) because if you do Microsoft will happily send a mid-call reINVITE (or BYE) to a different server than sent it the INVITE in the first place.

On Oct 1, 2020, at 10:20 AM, Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>> wrote:

Ok, then i might have found whats causing all my problems.
Would you happen to know how I achive this in freeswitch?

Ive tried a couple of things already, but none seem to work.
The gateway has “contact-host” set correctly, I also tried sip_contact_host …

Thank you very much!

Regards,

From: David Horton <daveh at drachtio.org<mailto:daveh at drachtio.org>>
Sent: Donnerstag, 1. Oktober 2020 14:20
To: Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>>
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Microsoft Teams integration

Yes, you must have a domain in the Contact header.  That’s another ridiculous requirement Microsoft came up with for this integration...

On Oct 1, 2020, at 1:30 AM, Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>> wrote:

Im following your advice now, and dont add record-route headers.
Im trying an outbound call ( teams -> pstn ) and it disconnects soon after freeswitch sends the OK.

Here is the OK:

send 1695 bytes to tls/[52.114.75.24]:10176 at 07:12:26.210854:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bKbc16542e;rport=10176
Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>
FROM: Stefan Kainz<sip:+FROMNUMBER at sip.pstnhub.microsoft.com:5061;user=phone>;tag=b88e3a220f48472cb452fec5fe46579e
To: <sip:+TONUMBER at xxx.example.at:5061;user=phone>;tag=9FFZm2y4cBmrS
CALL-ID: 6a933a0ac454552081ca931b3daf42de
CSEQ: 1 INVITE
Contact: <sip:+TONUMBER at xxx.xxx.xxx.xxx:5061;transport=tls>
User-Agent: TTG01 1.0
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY
Require: timer
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Session-Expires: 3600;refresher=uac
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 801
P-Asserted-Identity: "+TONUMBER" <sip:+TONUMBER at xxx.example.at>

v=0
o=FreeSWITCH 1601512094 1601512095 IN IP4 xxx.xxx.xxx.xxx
s=FreeSWITCH
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
a=msid-semantic: WMS pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF
m=audio 17052 RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-mux
a=rtcp:17052 IN IP4 xxx.xxx.xxx.xxx
a=ice-ufrag:czL8C9GwWgQdfSIK
a=ice-pwd:gICzT6c1eiFHGhUJGvI8vevi
a=candidate:2023834489 1 udp 659136 xxx.xxx.xxx.xxx 17052 typ host generation 0
a=end-of-candidates
a=ssrc:1804998514 cname:Fy91tlEMiigNDyLm
a=ssrc:1804998514 msid:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF a0
a=ssrc:1804998514 mslabel:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF
a=ssrc:1804998514 label:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cFa0
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:aoXrZjxmNQmdkxShUkmMO526J+5laS1UL8ZWiKyS

I compared it with your examples, and noticed that i have an ip-address in contact-header, you on the other hand have a domain.
Do you think that could be the problem?

regards,


________________________________
Von: David Horton <daveh at drachtio.org<mailto:daveh at drachtio.org>>
Gesendet: Mittwoch, 23. September 2020 19:24
An: Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>>
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Betreff: Re: [Freeswitch-users] Microsoft Teams integration

In my implementation I am using a drachtio server (https://drachtio<https://drachtio/>.org), more specifically configured as part of the jambonz CPaaS platform I build (https://docs.jambonz.org/teams/)

A Record-Route header only makes sense if you are creating a sip proxy, AFAIC.  In my implementation I built a B2BUA so on requests I send to Teams there is no RR.  Of course, on responses I preserve their RR headers.

For your scenario, I think I passed on the trace but just in case here is a link to it again: https://gist.github.com/davehorton/9ea3d79e8eeda3cbb5e2e87209dcfe63

One question: for the domain you are putting in the Contact header of your 200 OK response, does that resolve to a single server or multiple?  I have found in the latter case Teams will gladly send a mid-call request to a different server that heretofore was not involved in the call at all.  On the server that initially handled the call, it will simply look like Teams is not being responsive..

Dave

On Sep 23, 2020, at 3:56 AM, Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>> wrote:

Thank you for your Answer!

May I ask what kind of SBC you are using?
We also use Audiocodes SBC’s, which are working great, but even those send Record-Route Headers.

Maybe I need to explain the szenario a little better…
In the teams client I dial an external number ( my mobile phone for example )
I pick up, and then push hold in the teams client. ( So far so good )
When I try to unhold, of course also in teams, it becomes silent on both ends. ( even the moh stops )

And then as I said I see a few OK Messages being retransmitted to the Microsoft servers.

Its funny that you say you don’t need Record-Route Headers at all, because without setting those, I cant even make incoming calls work.
With those headers, everything works perfectly except for the thing explained above.
Even when trying to get incoming calls back from hold on teams. ( PSTN – freeswitch – teams, hold / unhold on teams )

I also found a tutorial from kamailio on how to connect it to teams.
There they also state that the record-route headers a needed.
https://skalatan.de/en/blog/kamailio-sbc-teams

Thank you again David!

Regards,
Stefan


From: David Horton <daveh at drachtio.org<mailto:daveh at drachtio.org>>
Sent: Montag, 21. September 2020 22:36
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Cc: Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>>
Subject: Re: [Freeswitch-users] Microsoft Teams integration

It has not been my experience that Microsoft Teams direct routing requires a Record-Route header on requests / responses you send to it (though their requests will have an RR and you must Route accordingly)

Here is a sip trace showing a successful on-hold / off-hold scenario from one of my servers (not a freeswitch server).  This is an on-hold triggered from the MS teams client, not sure if you are trying to do the reverse.

One thing: make sure the Contact header in your INVITEs resolve to a single server, otherwise (if you have mutiple A records for the DNS name) you will find that MS Teams will gladly respond to a reINVITE to a completely different server than you sent from — this will manifest as not seeing the 200 OK to a re-INVITE you send, or not seeing the ACK when you respond 200 OK to a re-INVITE.

daveh

One thing to make sure of is that

On Sep 20, 2020, at 12:05 PM, Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>> wrote:

Hello,

we're trying to use Freeswitch as a Microsoft Teams direct routing SBC.

So far everything works perfectly, except getting the call back from hold on an outbound call.
The Problem is that Microsoft expects a Record-Route Header in every Request and every Response.
I have been able to set/rewrite this Record-Route header with sip_h_ and sip_rh_ in most szenarios, but like i said - unhold on an outgoing call does not work.

The hold is initiated with a reinvite with a=inactive in sdp, which does work.
However, the unhold - also a reinvite with a=sendrecv in sdp does not work correctly.
In the sip-trace i see that the OK ( in response to the INVITE for unhold ) gets retransmitted a couple of times, because the ACK from microsoft isn't received.

This is due to the missing Record-Route Header in the OK Message.
I have seen this behavior on inbound calls too, when i didnt set the Record-Route Header.

Now my question:
Is there any way to add the Record-Route Header to every Message on a sip-profile or gateway?
Or if not, is it possible to handle a reinvite in dialplan so i could set the header manually?
Maybe there is somebody who already has this working that could point me in the right direction.

I know freeswitch isn't a SBC, but it would be nice to be able to use freeswitch as a Teams-SBC without Kamailio or something similar.

Any help is very much appreciated 🙂

regards,
Stefan







_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com<https://signalwire.com/>
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community<https://signalwire.community/>

Professional FreeSWITCH Services
sales at freeswitch.com<mailto:sales at freeswitch.com>
https://freeswitch.com<https://freeswitch.com/>

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com<https://cluecon.com/>

FreeSWITCH-users mailing list
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From daveh at drachtio.org  Thu Oct  1 15:43:21 2020
From: daveh at drachtio.org (David Horton)
Date: Thu, 1 Oct 2020 11:43:21 -0400
Subject: [Freeswitch-users] Microsoft Teams integration
In-Reply-To: <VI1P190MB049472C0AE2452F2F7D3E69886300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
References: <VI1P190MB04944EF9D1570CDE35F3E78C863D0@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <D89498C5-DB0D-4C06-ADAC-66E09FBCCBEE@drachtio.org>
 <VI1P190MB04949434E780D5D125633E4386380@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <A94AB550-C012-416A-BACB-0009121B0761@drachtio.org>
 <VI1P190MB0494E51F6504798137148DEB86300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <A384C846-2B00-456F-9ACC-73ED703FBD39@drachtio.org>
 <VI1P190MB049482B1DCA37658CB9A87C586300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <CD20FD48-EB37-4E4A-866C-F5555ECA742D@drachtio.org>
 <VI1P190MB049472C0AE2452F2F7D3E69886300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
Message-ID: <137C0D75-CF83-4A16-ABE4-0AC1F11D5D0D@drachtio.org>

good…just a point: putting it in /etc/hosts won’t be enough — it needs to be resolvable by Microsoft, so there need to be DNS records propagated

On Oct 1, 2020, at 11:21 AM, Stefan Kainz <s.kainz at wnt.at> wrote:

Ok, our fqdn does resolve to just one ip. No multiple entries.
Thank you so far David, I’ll try to get the fqdn in contact header and if I cant do it, maybe send another mail to the list.
 
Thank you very much!
 
Regards,
 
From: David Horton <daveh at drachtio.org> 
Sent: Donnerstag, 1. Oktober 2020 17:03
To: Stefan Kainz <s.kainz at wnt.at>
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Microsoft Teams integration
 
I’m sure there is a way to do it in Freeswitch, but I myself don’t know how as I’ve never needed to do that.  Maybe others can chime in..
 
It may be obvious, but I should have mentioned that the DNS name in the Contact header must resolve to your server either via SRV or A records.  Secondly, you do not want to have multiple A records for the DNS name (as for instance if you had a bank of freeswitch servers) because if you do Microsoft will happily send a mid-call reINVITE (or BYE) to a different server than sent it the INVITE in the first place.
 
On Oct 1, 2020, at 10:20 AM, Stefan Kainz <s.kainz at wnt.at <mailto:s.kainz at wnt.at>> wrote:
 
Ok, then i might have found whats causing all my problems.
Would you happen to know how I achive this in freeswitch?
 
Ive tried a couple of things already, but none seem to work.
The gateway has “contact-host” set correctly, I also tried sip_contact_host …
 
Thank you very much!
 
Regards,
 
From: David Horton <daveh at drachtio.org <mailto:daveh at drachtio.org>> 
Sent: Donnerstag, 1. Oktober 2020 14:20
To: Stefan Kainz <s.kainz at wnt.at <mailto:s.kainz at wnt.at>>
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Microsoft Teams integration
 
Yes, you must have a domain in the Contact header.  That’s another ridiculous requirement Microsoft came up with for this integration...
 
On Oct 1, 2020, at 1:30 AM, Stefan Kainz <s.kainz at wnt.at <mailto:s.kainz at wnt.at>> wrote:
 
Im following your advice now, and dont add record-route headers. 
Im trying an outbound call ( teams -> pstn ) and it disconnects soon after freeswitch sends the OK.
 
Here is the OK: 
 
send 1695 bytes to tls/[52.114.75.24]:10176 at 07:12:26.210854:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bKbc16542e;rport=10176
Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>>
FROM: Stefan Kainz<sip:+FROMNUMBER at sip.pstnhub.microsoft.com:5061;user=phone <sip:+FROMNUMBER at sip.pstnhub.microsoft.com:5061;user=phone>>;tag=b88e3a220f48472cb452fec5fe46579e
To: <sip:+TONUMBER at xxx.example.at:5061;user=phone <sip:+TONUMBER at xxx.example.at:5061;user=phone>>;tag=9FFZm2y4cBmrS
CALL-ID: 6a933a0ac454552081ca931b3daf42de
CSEQ: 1 INVITE
Contact: <sip:+TONUMBER at xxx.xxx.xxx.xxx:5061;transport=tls <sip:+TONUMBER at xxx.xxx.xxx.xxx:5061;transport=tls>>
User-Agent: TTG01 1.0
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY
Require: timer
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Session-Expires: 3600;refresher=uac
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 801
P-Asserted-Identity: "+TONUMBER" <sip:+TONUMBER at xxx.example.at <sip:+TONUMBER at xxx.example.at>>
 
v=0
o=FreeSWITCH 1601512094 1601512095 IN IP4 xxx.xxx.xxx.xxx
s=FreeSWITCH
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
a=msid-semantic: WMS pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF
m=audio 17052 RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-mux
a=rtcp:17052 IN IP4 xxx.xxx.xxx.xxx
a=ice-ufrag:czL8C9GwWgQdfSIK
a=ice-pwd:gICzT6c1eiFHGhUJGvI8vevi
a=candidate:2023834489 1 udp 659136 xxx.xxx.xxx.xxx 17052 typ host generation 0
a=end-of-candidates
a=ssrc:1804998514 cname:Fy91tlEMiigNDyLm
a=ssrc:1804998514 msid:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF a0
a=ssrc:1804998514 mslabel:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF
a=ssrc:1804998514 label:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cFa0
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:aoXrZjxmNQmdkxShUkmMO526J+5laS1UL8ZWiKyS
 
I compared it with your examples, and noticed that i have an ip-address in contact-header, you on the other hand have a domain. 
Do you think that could be the problem?
 
regards, 
 
 
Von: David Horton <daveh at drachtio.org <mailto:daveh at drachtio.org>>
Gesendet: Mittwoch, 23. September 2020 19:24
An: Stefan Kainz <s.kainz at wnt.at <mailto:s.kainz at wnt.at>>
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
Betreff: Re: [Freeswitch-users] Microsoft Teams integration
 
In my implementation I am using a drachtio server (https://drachtio <https://drachtio/>.org), more specifically configured as part of the jambonz CPaaS platform I build (https://docs.jambonz.org/teams/ <https://docs.jambonz.org/teams/>)
 
A Record-Route header only makes sense if you are creating a sip proxy, AFAIC.  In my implementation I built a B2BUA so on requests I send to Teams there is no RR.  Of course, on responses I preserve their RR headers.
 
For your scenario, I think I passed on the trace but just in case here is a link to it again: https://gist.github.com/davehorton/9ea3d79e8eeda3cbb5e2e87209dcfe63 <https://gist.github.com/davehorton/9ea3d79e8eeda3cbb5e2e87209dcfe63>
 
One question: for the domain you are putting in the Contact header of your 200 OK response, does that resolve to a single server or multiple?  I have found in the latter case Teams will gladly send a mid-call request to a different server that heretofore was not involved in the call at all.  On the server that initially handled the call, it will simply look like Teams is not being responsive..
 
Dave
 
On Sep 23, 2020, at 3:56 AM, Stefan Kainz <s.kainz at wnt.at <mailto:s.kainz at wnt.at>> wrote:
 
Thank you for your Answer!
 
May I ask what kind of SBC you are using?
We also use Audiocodes SBC’s, which are working great, but even those send Record-Route Headers.
 
Maybe I need to explain the szenario a little better…
In the teams client I dial an external number ( my mobile phone for example )
I pick up, and then push hold in the teams client. ( So far so good )
When I try to unhold, of course also in teams, it becomes silent on both ends. ( even the moh stops )
 
And then as I said I see a few OK Messages being retransmitted to the Microsoft servers.
 
Its funny that you say you don’t need Record-Route Headers at all, because without setting those, I cant even make incoming calls work.
With those headers, everything works perfectly except for the thing explained above.
Even when trying to get incoming calls back from hold on teams. ( PSTN – freeswitch – teams, hold / unhold on teams )
 
I also found a tutorial from kamailio on how to connect it to teams.
There they also state that the record-route headers a needed.
https://skalatan.de/en/blog/kamailio-sbc-teams <https://skalatan.de/en/blog/kamailio-sbc-teams>
 
Thank you again David!
 
Regards,
Stefan
 
 
From: David Horton <daveh at drachtio.org <mailto:daveh at drachtio.org>> 
Sent: Montag, 21. September 2020 22:36
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
Cc: Stefan Kainz <s.kainz at wnt.at <mailto:s.kainz at wnt.at>>
Subject: Re: [Freeswitch-users] Microsoft Teams integration
 
It has not been my experience that Microsoft Teams direct routing requires a Record-Route header on requests / responses you send to it (though their requests will have an RR and you must Route accordingly)
 
Here is a sip trace showing a successful on-hold / off-hold scenario from one of my servers (not a freeswitch server).  This is an on-hold triggered from the MS teams client, not sure if you are trying to do the reverse.
 
One thing: make sure the Contact header in your INVITEs resolve to a single server, otherwise (if you have mutiple A records for the DNS name) you will find that MS Teams will gladly respond to a reINVITE to a completely different server than you sent from — this will manifest as not seeing the 200 OK to a re-INVITE you send, or not seeing the ACK when you respond 200 OK to a re-INVITE.
 
daveh
 
One thing to make sure of is that 
 
On Sep 20, 2020, at 12:05 PM, Stefan Kainz <s.kainz at wnt.at <mailto:s.kainz at wnt.at>> wrote:
 
Hello, 
 
we're trying to use Freeswitch as a Microsoft Teams direct routing SBC.
 
So far everything works perfectly, except getting the call back from hold on an outbound call. 
The Problem is that Microsoft expects a Record-Route Header in every Request and every Response.
I have been able to set/rewrite this Record-Route header with sip_h_ and sip_rh_ in most szenarios, but like i said - unhold on an outgoing call does not work.
 
The hold is initiated with a reinvite with a=inactive in sdp, which does work. 
However, the unhold - also a reinvite with a=sendrecv in sdp does not work correctly.
In the sip-trace i see that the OK ( in response to the INVITE for unhold ) gets retransmitted a couple of times, because the ACK from microsoft isn't received. 
 
This is due to the missing Record-Route Header in the OK Message.
I have seen this behavior on inbound calls too, when i didnt set the Record-Route Header.
 
Now my question:
Is there any way to add the Record-Route Header to every Message on a sip-profile or gateway? 
Or if not, is it possible to handle a reinvite in dialplan so i could set the header manually?
Maybe there is somebody who already has this working that could point me in the right direction.
 
I know freeswitch isn't a SBC, but it would be nice to be able to use freeswitch as a Teams-SBC without Kamailio or something similar.
 
Any help is very much appreciated 🙂
 
regards, 
Stefan
 
 
 
 
 
 
 
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From s.kainz at wnt.at  Thu Oct  1 15:50:37 2020
From: s.kainz at wnt.at (Stefan Kainz)
Date: Thu, 1 Oct 2020 15:50:37 +0000
Subject: [Freeswitch-users] Microsoft Teams integration
In-Reply-To: <137C0D75-CF83-4A16-ABE4-0AC1F11D5D0D@drachtio.org>
References: <VI1P190MB04944EF9D1570CDE35F3E78C863D0@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <D89498C5-DB0D-4C06-ADAC-66E09FBCCBEE@drachtio.org>
 <VI1P190MB04949434E780D5D125633E4386380@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <A94AB550-C012-416A-BACB-0009121B0761@drachtio.org>
 <VI1P190MB0494E51F6504798137148DEB86300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <A384C846-2B00-456F-9ACC-73ED703FBD39@drachtio.org>
 <VI1P190MB049482B1DCA37658CB9A87C586300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <CD20FD48-EB37-4E4A-866C-F5555ECA742D@drachtio.org>
 <VI1P190MB049472C0AE2452F2F7D3E69886300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <137C0D75-CF83-4A16-ABE4-0AC1F11D5D0D@drachtio.org>
Message-ID: <VI1P190MB04943246E0F046989D701A9A86300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>

Good point…
I also just tried domain / domain_name settings in vars.xml, which should be the hostname of the freeswitch-instance but … doesn’t work …

Seems freeswitch doesn’t want a FQDN In Contact Header …

Regards,

From: David Horton <daveh at drachtio.org>
Sent: Donnerstag, 1. Oktober 2020 17:43
To: Stefan Kainz <s.kainz at wnt.at>
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Microsoft Teams integration

good…just a point: putting it in /etc/hosts won’t be enough — it needs to be resolvable by Microsoft, so there need to be DNS records propagated

On Oct 1, 2020, at 11:21 AM, Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>> wrote:

Ok, our fqdn does resolve to just one ip. No multiple entries.
Thank you so far David, I’ll try to get the fqdn in contact header and if I cant do it, maybe send another mail to the list.

Thank you very much!

Regards,

From: David Horton <daveh at drachtio.org<mailto:daveh at drachtio.org>>
Sent: Donnerstag, 1. Oktober 2020 17:03
To: Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>>
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Microsoft Teams integration

I’m sure there is a way to do it in Freeswitch, but I myself don’t know how as I’ve never needed to do that.  Maybe others can chime in..

It may be obvious, but I should have mentioned that the DNS name in the Contact header must resolve to your server either via SRV or A records.  Secondly, you do not want to have multiple A records for the DNS name (as for instance if you had a bank of freeswitch servers) because if you do Microsoft will happily send a mid-call reINVITE (or BYE) to a different server than sent it the INVITE in the first place.

On Oct 1, 2020, at 10:20 AM, Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>> wrote:

Ok, then i might have found whats causing all my problems.
Would you happen to know how I achive this in freeswitch?

Ive tried a couple of things already, but none seem to work.
The gateway has “contact-host” set correctly, I also tried sip_contact_host …

Thank you very much!

Regards,

From: David Horton <daveh at drachtio.org<mailto:daveh at drachtio.org>>
Sent: Donnerstag, 1. Oktober 2020 14:20
To: Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>>
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Microsoft Teams integration

Yes, you must have a domain in the Contact header.  That’s another ridiculous requirement Microsoft came up with for this integration...

On Oct 1, 2020, at 1:30 AM, Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>> wrote:

Im following your advice now, and dont add record-route headers.
Im trying an outbound call ( teams -> pstn ) and it disconnects soon after freeswitch sends the OK.

Here is the OK:

send 1695 bytes to tls/[52.114.75.24]:10176 at 07:12:26.210854:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bKbc16542e;rport=10176
Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>
FROM: Stefan Kainz<sip:+FROMNUMBER at sip.pstnhub.microsoft.com:5061;user=phone>;tag=b88e3a220f48472cb452fec5fe46579e
To: <sip:+TONUMBER at xxx.example.at:5061;user=phone>;tag=9FFZm2y4cBmrS
CALL-ID: 6a933a0ac454552081ca931b3daf42de
CSEQ: 1 INVITE
Contact: <sip:+TONUMBER at xxx.xxx.xxx.xxx:5061;transport=tls>
User-Agent: TTG01 1.0
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY
Require: timer
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Session-Expires: 3600;refresher=uac
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 801
P-Asserted-Identity: "+TONUMBER" <sip:+TONUMBER at xxx.example.at>

v=0
o=FreeSWITCH 1601512094 1601512095 IN IP4 xxx.xxx.xxx.xxx
s=FreeSWITCH
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
a=msid-semantic: WMS pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF
m=audio 17052 RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-mux
a=rtcp:17052 IN IP4 xxx.xxx.xxx.xxx
a=ice-ufrag:czL8C9GwWgQdfSIK
a=ice-pwd:gICzT6c1eiFHGhUJGvI8vevi
a=candidate:2023834489 1 udp 659136 xxx.xxx.xxx.xxx 17052 typ host generation 0
a=end-of-candidates
a=ssrc:1804998514 cname:Fy91tlEMiigNDyLm
a=ssrc:1804998514 msid:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF a0
a=ssrc:1804998514 mslabel:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cF
a=ssrc:1804998514 label:pGvDw5lMt7NEFur7TOXy5dyI4XJtq8cFa0
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:aoXrZjxmNQmdkxShUkmMO526J+5laS1UL8ZWiKyS

I compared it with your examples, and noticed that i have an ip-address in contact-header, you on the other hand have a domain.
Do you think that could be the problem?

regards,


________________________________
Von: David Horton <daveh at drachtio.org<mailto:daveh at drachtio.org>>
Gesendet: Mittwoch, 23. September 2020 19:24
An: Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>>
Cc: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Betreff: Re: [Freeswitch-users] Microsoft Teams integration

In my implementation I am using a drachtio server (https://drachtio<https://drachtio/>.org), more specifically configured as part of the jambonz CPaaS platform I build (https://docs.jambonz.org/teams/)

A Record-Route header only makes sense if you are creating a sip proxy, AFAIC.  In my implementation I built a B2BUA so on requests I send to Teams there is no RR.  Of course, on responses I preserve their RR headers.

For your scenario, I think I passed on the trace but just in case here is a link to it again: https://gist.github.com/davehorton/9ea3d79e8eeda3cbb5e2e87209dcfe63

One question: for the domain you are putting in the Contact header of your 200 OK response, does that resolve to a single server or multiple?  I have found in the latter case Teams will gladly send a mid-call request to a different server that heretofore was not involved in the call at all.  On the server that initially handled the call, it will simply look like Teams is not being responsive..

Dave

On Sep 23, 2020, at 3:56 AM, Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>> wrote:

Thank you for your Answer!

May I ask what kind of SBC you are using?
We also use Audiocodes SBC’s, which are working great, but even those send Record-Route Headers.

Maybe I need to explain the szenario a little better…
In the teams client I dial an external number ( my mobile phone for example )
I pick up, and then push hold in the teams client. ( So far so good )
When I try to unhold, of course also in teams, it becomes silent on both ends. ( even the moh stops )

And then as I said I see a few OK Messages being retransmitted to the Microsoft servers.

Its funny that you say you don’t need Record-Route Headers at all, because without setting those, I cant even make incoming calls work.
With those headers, everything works perfectly except for the thing explained above.
Even when trying to get incoming calls back from hold on teams. ( PSTN – freeswitch – teams, hold / unhold on teams )

I also found a tutorial from kamailio on how to connect it to teams.
There they also state that the record-route headers a needed.
https://skalatan.de/en/blog/kamailio-sbc-teams

Thank you again David!

Regards,
Stefan


From: David Horton <daveh at drachtio.org<mailto:daveh at drachtio.org>>
Sent: Montag, 21. September 2020 22:36
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Cc: Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>>
Subject: Re: [Freeswitch-users] Microsoft Teams integration

It has not been my experience that Microsoft Teams direct routing requires a Record-Route header on requests / responses you send to it (though their requests will have an RR and you must Route accordingly)

Here is a sip trace showing a successful on-hold / off-hold scenario from one of my servers (not a freeswitch server).  This is an on-hold triggered from the MS teams client, not sure if you are trying to do the reverse.

One thing: make sure the Contact header in your INVITEs resolve to a single server, otherwise (if you have mutiple A records for the DNS name) you will find that MS Teams will gladly respond to a reINVITE to a completely different server than you sent from — this will manifest as not seeing the 200 OK to a re-INVITE you send, or not seeing the ACK when you respond 200 OK to a re-INVITE.

daveh

One thing to make sure of is that

On Sep 20, 2020, at 12:05 PM, Stefan Kainz <s.kainz at wnt.at<mailto:s.kainz at wnt.at>> wrote:

Hello,

we're trying to use Freeswitch as a Microsoft Teams direct routing SBC.

So far everything works perfectly, except getting the call back from hold on an outbound call.
The Problem is that Microsoft expects a Record-Route Header in every Request and every Response.
I have been able to set/rewrite this Record-Route header with sip_h_ and sip_rh_ in most szenarios, but like i said - unhold on an outgoing call does not work.

The hold is initiated with a reinvite with a=inactive in sdp, which does work.
However, the unhold - also a reinvite with a=sendrecv in sdp does not work correctly.
In the sip-trace i see that the OK ( in response to the INVITE for unhold ) gets retransmitted a couple of times, because the ACK from microsoft isn't received.

This is due to the missing Record-Route Header in the OK Message.
I have seen this behavior on inbound calls too, when i didnt set the Record-Route Header.

Now my question:
Is there any way to add the Record-Route Header to every Message on a sip-profile or gateway?
Or if not, is it possible to handle a reinvite in dialplan so i could set the header manually?
Maybe there is somebody who already has this working that could point me in the right direction.

I know freeswitch isn't a SBC, but it would be nice to be able to use freeswitch as a Teams-SBC without Kamailio or something similar.

Any help is very much appreciated 🙂

regards,
Stefan







_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com<https://signalwire.com/>
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community<https://signalwire.community/>

Professional FreeSWITCH Services
sales at freeswitch.com<mailto:sales at freeswitch.com>
https://freeswitch.com<https://freeswitch.com/>

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com<https://cluecon.com/>

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com<https://freeswitch.com/>

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From s.safarov at gmail.com  Thu Oct  1 22:18:26 2020
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 2 Oct 2020 01:18:26 +0300
Subject: [Freeswitch-users] Clone freeswitch-contrib.git
In-Reply-To: <4d6b0526ff0956a065e5199bb7ca8db5@soft-gator.com>
References: <4d6b0526ff0956a065e5199bb7ca8db5@soft-gator.com>
Message-ID: <CAHtxdDeLuQaS-Dudd3Lhko4N1e9r+tpnJO120sngsMVoYeFwhQ@mail.gmail.com>

FreeSwitch repo relocated to https://github.com/signalwire/freeswitch

On Thu, Oct 1, 2020 at 11:19 PM Ing Sergio Basurto J. <
sbasurto at soft-gator.com> wrote:

> Hello,
>
> Anyone knows how can I clone contrib repo?, because when I ran:
>
> git clone https://freeswitch.org/stash/scm/fs/freeswitch-contrib.git
>
> Cloning into 'freeswitch-contrib'...
> Username for 'https://freeswitch.org':
> Password for 'https://freeswitch.org':
> fatal: Authentication failed for '
> https://freeswitch.org/stash/scm/fs/freeswitch-contrib.git/'
>
> Ask for username and password, there is a default username and password?
>
> I just want the module mod_odbc_query, am I missing something?
>
>
> Any help will be appreciated!!
>
> Best Regards,
> --
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From krice at freeswitch.org  Thu Oct  1 23:43:22 2020
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 1 Oct 2020 18:43:22 -0500
Subject: [Freeswitch-users] Clone freeswitch-contrib.git
In-Reply-To: <CAHtxdDeLuQaS-Dudd3Lhko4N1e9r+tpnJO120sngsMVoYeFwhQ@mail.gmail.com>
References: <CAHtxdDeLuQaS-Dudd3Lhko4N1e9r+tpnJO120sngsMVoYeFwhQ@mail.gmail.com>
Message-ID: <2ED1EACF-B6B8-4743-999A-725FD862F0E9@freeswitch.org>

thats not the contrib repo. theres a ton ofnuseful code in the contrib repo

Sent from my iPhone

> On Oct 1, 2020, at 17:19, Sergey Safarov <s.safarov at gmail.com> wrote:
> 
> 
> FreeSwitch repo relocated to https://github.com/signalwire/freeswitch
> 
>> On Thu, Oct 1, 2020 at 11:19 PM Ing Sergio Basurto J. <sbasurto at soft-gator.com> wrote:
>> Hello,
>> 
>> Anyone knows how can I clone contrib repo?, because when I ran:
>> 
>> git clone https://freeswitch.org/stash/scm/fs/freeswitch-contrib.git
>> 
>> Cloning into 'freeswitch-contrib'...
>> Username for 'https://freeswitch.org': 
>> Password for 'https://freeswitch.org': 
>> fatal: Authentication failed for 'https://freeswitch.org/stash/scm/fs/freeswitch-contrib.git/'
>> 
>> Ask for username and password, there is a default username and password?
>> 
>> I just want the module mod_odbc_query, am I missing something?
>> 
>> 
>> 
>> Any help will be appreciated!!
>> 
>> Best Regards,
>> 
>> -- 
>> 
>> 
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>> Build your next product on our scalable cloud platform.
>> 
>> Join our online community to chat in real time https://signalwire.community
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From covici at ccs.covici.com  Fri Oct  2 10:08:25 2020
From: covici at ccs.covici.com (John Covici)
Date: Fri, 02 Oct 2020 06:08:25 -0400
Subject: [Freeswitch-users] Clone freeswitch-contrib.git
In-Reply-To: <2ED1EACF-B6B8-4743-999A-725FD862F0E9@freeswitch.org>
References: <CAHtxdDeLuQaS-Dudd3Lhko4N1e9r+tpnJO120sngsMVoYeFwhQ@mail.gmail.com>
 <2ED1EACF-B6B8-4743-999A-725FD862F0E9@freeswitch.org>
Message-ID: <m31rih7xeu.wl-covici@ccs.covici.com>

So, where is the contrib repo?  Might be quite useful.

On Thu, 01 Oct 2020 19:43:22 -0400,
Ken Rice wrote:
> 
> [1  <multipart/alternative (7bit)>]
> [1.1  <text/plain; utf-8 (quoted-printable)>]
> thats not the contrib repo. theres a ton ofnuseful code in the contrib repo
> 
> Sent from my iPhone
> 
> > On Oct 1, 2020, at 17:19, Sergey Safarov <s.safarov at gmail.com> wrote:
> > 
> > 
> > FreeSwitch repo relocated to https://github.com/signalwire/freeswitch
> > 
> >> On Thu, Oct 1, 2020 at 11:19 PM Ing Sergio Basurto J. <sbasurto at soft-gator.com> wrote:
> >> Hello,
> >> 
> >> Anyone knows how can I clone contrib repo?, because when I ran:
> >> 
> >> git clone https://freeswitch.org/stash/scm/fs/freeswitch-contrib.git
> >> 
> >> Cloning into 'freeswitch-contrib'...
> >> Username for 'https://freeswitch.org': 
> >> Password for 'https://freeswitch.org': 
> >> fatal: Authentication failed for 'https://freeswitch.org/stash/scm/fs/freeswitch-contrib.git/'
> >> 
> >> Ask for username and password, there is a default username and password?
> >> 
> >> I just want the module mod_odbc_query, am I missing something?
> >> 
> >> 
> >> 
> >> Any help will be appreciated!!
> >> 
> >> Best Regards,
> >> 
> >> -- 
> >> 
> >> 
> >> _________________________________________________________________________
> >> 
> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> >> Build your next product on our scalable cloud platform.
> >> 
> >> Join our online community to chat in real time https://signalwire.community
> >> 
> >> Professional FreeSWITCH Services
> >> sales at freeswitch.com
> >> https://freeswitch.com
> >> 
> >> Official FreeSWITCH Sites
> >> https://freeswitch.com/oss
> >> https://freeswitch.org/confluence
> >> https://cluecon.com
> >> 
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> https://freeswitch.com
> > _________________________________________________________________________
> > 
> > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> > Build your next product on our scalable cloud platform.
> > 
> > Join our online community to chat in real time https://signalwire.community
> > 
> > Professional FreeSWITCH Services
> > sales at freeswitch.com
> > https://freeswitch.com
> > 
> > Official FreeSWITCH Sites
> > https://freeswitch.com/oss
> > https://freeswitch.org/confluence
> > https://cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > https://freeswitch.com
> [1.2  <text/html; utf-8 (quoted-printable)>]
> [2  <text/plain; utf-8 (base64)>]
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici wb2una
         covici at ccs.covici.com


From nathan at robotics.net  Fri Oct  2 10:32:13 2020
From: nathan at robotics.net (Nathan Stratton)
Date: Fri, 2 Oct 2020 06:32:13 -0400
Subject: [Freeswitch-users] Force a reINVITE?
Message-ID: <CAHubkyPfvzCv_jmwZ5vxcnwcx3yQb5LnTSaHUbjbh2+hTuaaFA@mail.gmail.com>

I am starting a call with OPUS only on the B leg, the call starts fine with
inbound-late-negotiation and OPUS is sent back on the A leg. However, the
call is then transferred to a user with OPUS and VP8 video with the new
INVITE, but nothing ever is sent back to the original A leg to update VP8.
Is it possible to send a reINVITE back so I can update the SDP with VP8 on
the original A leg?

><>
nathan stratton
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From andywolk at gmail.com  Fri Oct  2 10:43:39 2020
From: andywolk at gmail.com (Andrey Volk)
Date: Fri, 2 Oct 2020 14:43:39 +0400
Subject: [Freeswitch-users] Clone freeswitch-contrib.git
In-Reply-To: <m31rih7xeu.wl-covici@ccs.covici.com>
References: <CAHtxdDeLuQaS-Dudd3Lhko4N1e9r+tpnJO120sngsMVoYeFwhQ@mail.gmail.com>
 <2ED1EACF-B6B8-4743-999A-725FD862F0E9@freeswitch.org>
 <m31rih7xeu.wl-covici@ccs.covici.com>
Message-ID: <CAD55zR0xkbQSgJrnZnn9S8=Hyb00ULqMEQZSM4h-vWzYR=Eu5w@mail.gmail.com>

https://github.com/freeswitch/freeswitch-contrib

пт, 2 окт. 2020 г. в 14:38, John Covici <covici at ccs.covici.com>:

> So, where is the contrib repo?  Might be quite useful.
>
> On Thu, 01 Oct 2020 19:43:22 -0400,
> Ken Rice wrote:
> >
> > [1  <multipart/alternative (7bit)>]
> > [1.1  <text/plain; utf-8 (quoted-printable)>]
> > thats not the contrib repo. theres a ton ofnuseful code in the contrib
> repo
> >
> > Sent from my iPhone
> >
> > > On Oct 1, 2020, at 17:19, Sergey Safarov <s.safarov at gmail.com> wrote:
> > >
> > > 
> > > FreeSwitch repo relocated to https://github.com/signalwire/freeswitch
> > >
> > >> On Thu, Oct 1, 2020 at 11:19 PM Ing Sergio Basurto J. <
> sbasurto at soft-gator.com> wrote:
> > >> Hello,
> > >>
> > >> Anyone knows how can I clone contrib repo?, because when I ran:
> > >>
> > >> git clone https://freeswitch.org/stash/scm/fs/freeswitch-contrib.git
> > >>
> > >> Cloning into 'freeswitch-contrib'...
> > >> Username for 'https://freeswitch.org':
> > >> Password for 'https://freeswitch.org':
> > >> fatal: Authentication failed for '
> https://freeswitch.org/stash/scm/fs/freeswitch-contrib.git/'
> > >>
> > >> Ask for username and password, there is a default username and
> password?
> > >>
> > >> I just want the module mod_odbc_query, am I missing something?
> > >>
> > >>
> > >>
> > >> Any help will be appreciated!!
> > >>
> > >> Best Regards,
> > >>
> > >> --
> > >>
> > >>
> > >>
> _________________________________________________________________________
> > >>
> > >> The FreeSWITCH project is sponsored by SignalWire
> https://signalwire.com
> > >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> > >> Build your next product on our scalable cloud platform.
> > >>
> > >> Join our online community to chat in real time
> https://signalwire.community
> > >>
> > >> Professional FreeSWITCH Services
> > >> sales at freeswitch.com
> > >> https://freeswitch.com
> > >>
> > >> Official FreeSWITCH Sites
> > >> https://freeswitch.com/oss
> > >> https://freeswitch.org/confluence
> > >> https://cluecon.com
> > >>
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> https://freeswitch.com
> > >
> _________________________________________________________________________
> > >
> > > The FreeSWITCH project is sponsored by SignalWire
> https://signalwire.com
> > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> > > Build your next product on our scalable cloud platform.
> > >
> > > Join our online community to chat in real time
> https://signalwire.community
> > >
> > > Professional FreeSWITCH Services
> > > sales at freeswitch.com
> > > https://freeswitch.com
> > >
> > > Official FreeSWITCH Sites
> > > https://freeswitch.com/oss
> > > https://freeswitch.org/confluence
> > > https://cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > https://freeswitch.com
> > [1.2  <text/html; utf-8 (quoted-printable)>]
> > [2  <text/plain; utf-8 (base64)>]
> > _________________________________________________________________________
> >
> > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> > Build your next product on our scalable cloud platform.
> >
> > Join our online community to chat in real time
> https://signalwire.community
> >
> > Professional FreeSWITCH Services
> > sales at freeswitch.com
> > https://freeswitch.com
> >
> > Official FreeSWITCH Sites
> > https://freeswitch.com/oss
> > https://freeswitch.org/confluence
> > https://cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > https://freeswitch.com
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>          John Covici wb2una
>          covici at ccs.covici.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From cmrienzo at gmail.com  Fri Oct  2 13:59:00 2020
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Fri, 2 Oct 2020 09:59:00 -0400
Subject: [Freeswitch-users] Clone freeswitch-contrib.git
In-Reply-To: <m31rih7xeu.wl-covici@ccs.covici.com>
References: <CAHtxdDeLuQaS-Dudd3Lhko4N1e9r+tpnJO120sngsMVoYeFwhQ@mail.gmail.com>
 <2ED1EACF-B6B8-4743-999A-725FD862F0E9@freeswitch.org>
 <m31rih7xeu.wl-covici@ccs.covici.com>
Message-ID: <CAD-qRkhZNGHeQCL3BT3aWQ=E=Ur-2BNt0qwC=kuJuoTkZc=tXw@mail.gmail.com>

We've moved it to https://github.com/freeswitch/freeswitch-contrib

Chris


On Fri, Oct 2, 2020 at 6:56 AM John Covici <covici at ccs.covici.com> wrote:

> So, where is the contrib repo?  Might be quite useful.
>
> On Thu, 01 Oct 2020 19:43:22 -0400,
> Ken Rice wrote:
> >
> > [1  <multipart/alternative (7bit)>]
> > [1.1  <text/plain; utf-8 (quoted-printable)>]
> > thats not the contrib repo. theres a ton ofnuseful code in the contrib
> repo
> >
> > Sent from my iPhone
> >
> > > On Oct 1, 2020, at 17:19, Sergey Safarov <s.safarov at gmail.com> wrote:
> > >
> > > 
> > > FreeSwitch repo relocated to https://github.com/signalwire/freeswitch
> > >
> > >> On Thu, Oct 1, 2020 at 11:19 PM Ing Sergio Basurto J. <
> sbasurto at soft-gator.com> wrote:
> > >> Hello,
> > >>
> > >> Anyone knows how can I clone contrib repo?, because when I ran:
> > >>
> > >> git clone https://freeswitch.org/stash/scm/fs/freeswitch-contrib.git
> > >>
> > >> Cloning into 'freeswitch-contrib'...
> > >> Username for 'https://freeswitch.org':
> > >> Password for 'https://freeswitch.org':
> > >> fatal: Authentication failed for '
> https://freeswitch.org/stash/scm/fs/freeswitch-contrib.git/'
> > >>
> > >> Ask for username and password, there is a default username and
> password?
> > >>
> > >> I just want the module mod_odbc_query, am I missing something?
> > >>
> > >>
> > >>
> > >> Any help will be appreciated!!
> > >>
> > >> Best Regards,
> > >>
> > >> --
> > >>
> > >>
> > >>
> _________________________________________________________________________
> > >>
> > >> The FreeSWITCH project is sponsored by SignalWire
> https://signalwire.com
> > >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> > >> Build your next product on our scalable cloud platform.
> > >>
> > >> Join our online community to chat in real time
> https://signalwire.community
> > >>
> > >> Professional FreeSWITCH Services
> > >> sales at freeswitch.com
> > >> https://freeswitch.com
> > >>
> > >> Official FreeSWITCH Sites
> > >> https://freeswitch.com/oss
> > >> https://freeswitch.org/confluence
> > >> https://cluecon.com
> > >>
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> https://freeswitch.com
> > >
> _________________________________________________________________________
> > >
> > > The FreeSWITCH project is sponsored by SignalWire
> https://signalwire.com
> > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> > > Build your next product on our scalable cloud platform.
> > >
> > > Join our online community to chat in real time
> https://signalwire.community
> > >
> > > Professional FreeSWITCH Services
> > > sales at freeswitch.com
> > > https://freeswitch.com
> > >
> > > Official FreeSWITCH Sites
> > > https://freeswitch.com/oss
> > > https://freeswitch.org/confluence
> > > https://cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > https://freeswitch.com
> > [1.2  <text/html; utf-8 (quoted-printable)>]
> > [2  <text/plain; utf-8 (base64)>]
> > _________________________________________________________________________
> >
> > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> > Build your next product on our scalable cloud platform.
> >
> > Join our online community to chat in real time
> https://signalwire.community
> >
> > Professional FreeSWITCH Services
> > sales at freeswitch.com
> > https://freeswitch.com
> >
> > Official FreeSWITCH Sites
> > https://freeswitch.com/oss
> > https://freeswitch.org/confluence
> > https://cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > https://freeswitch.com
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>          John Covici wb2una
>          covici at ccs.covici.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From sbasurto at soft-gator.com  Fri Oct  2 16:31:33 2020
From: sbasurto at soft-gator.com (Ing Sergio Basurto J.)
Date: Fri, 02 Oct 2020 11:31:33 -0500
Subject: [Freeswitch-users] Clone freeswitch-contrib.git
In-Reply-To: <CAD-qRkhZNGHeQCL3BT3aWQ=E=Ur-2BNt0qwC=kuJuoTkZc=tXw@mail.gmail.com>
References: <CAHtxdDeLuQaS-Dudd3Lhko4N1e9r+tpnJO120sngsMVoYeFwhQ@mail.gmail.com>
 <2ED1EACF-B6B8-4743-999A-725FD862F0E9@freeswitch.org>
 <m31rih7xeu.wl-covici@ccs.covici.com>
 <CAD-qRkhZNGHeQCL3BT3aWQ=E=Ur-2BNt0qwC=kuJuoTkZc=tXw@mail.gmail.com>
Message-ID: <768fecbfbf61a3d348a0bd3da3be6c6f@soft-gator.com>

Thank you very much to all for the response. 

I can get the contrib successfully in the link you pointed out. 

It would be nice if the link was up to date in documentation
(https://freeswitch.org/confluence/display/FREESWITCH/Tips+For+Using+Git)


Best Regards, 

El 2020-10-02 08:59, Christopher Rienzo escribió:

> We've moved it to https://github.com/freeswitch/freeswitch-contrib 
> 
> Chris 
> 
> On Fri, Oct 2, 2020 at 6:56 AM John Covici <covici at ccs.covici.com> wrote: 
> 
>> So, where is the contrib repo?  Might be quite useful.
>> 
>> On Thu, 01 Oct 2020 19:43:22 -0400,
>> Ken Rice wrote:
>>> 
>>> [1  <multipart/alternative (7bit)>]
>>> [1.1  <text/plain; utf-8 (quoted-printable)>]
>>> thats not the contrib repo. theres a ton ofnuseful code in the contrib repo
>>> 
>>> Sent from my iPhone
>>> 
>>>> On Oct 1, 2020, at 17:19, Sergey Safarov <s.safarov at gmail.com> wrote:
>>>> 
>>>> 
>>>> FreeSwitch repo relocated to https://github.com/signalwire/freeswitch
>>>> 
>>>>> On Thu, Oct 1, 2020 at 11:19 PM Ing Sergio Basurto J. <sbasurto at soft-gator.com> wrote:
>>>>> Hello,
>>>>> 
>>>>> Anyone knows how can I clone contrib repo?, because when I ran:
>>>>> 
>>>>> git clone https://freeswitch.org/stash/scm/fs/freeswitch-contrib.git
>>>>> 
>>>>> Cloning into 'freeswitch-contrib'...
>>>>> Username for 'https://freeswitch.org': 
>>>>> Password for 'https://freeswitch.org': 
>>>>> fatal: Authentication failed for 'https://freeswitch.org/stash/scm/fs/freeswitch-contrib.git/'
>>>>> 
>>>>> Ask for username and password, there is a default username and password?
>>>>> 
>>>>> I just want the module mod_odbc_query, am I missing something?
>>>>> 
>>>>> 
>>>>> 
>>>>> Any help will be appreciated!!
>>>>> 
>>>>> Best Regards,
>>>>> 
>>>>> -- 
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> 
>>>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>>>>> Build your next product on our scalable cloud platform.
>>>>> 
>>>>> Join our online community to chat in real time https://signalwire.community
>>>>> 
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>> _________________________________________________________________________
>>>> 
>>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>>>> Build your next product on our scalable cloud platform.
>>>> 
>>>> Join our online community to chat in real time https://signalwire.community
>>>> 
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>> 
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>> [1.2  <text/html; utf-8 (quoted-printable)>]
>>> [2  <text/plain; utf-8 (base64)>]
>>> _________________________________________________________________________
>>> 
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>>> Build your next product on our scalable cloud platform.
>>> 
>>> Join our online community to chat in real time https://signalwire.community
>>> 
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>> 
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>> 
>> -- 
>> Your life is like a penny.  You're going to lose it.  The question is:
>> How do
>> you spend it?
>> 
>> John Covici wb2una
>> covici at ccs.covici.com
>> 
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>> Build your next product on our scalable cloud platform.
>> 
>> Join our online community to chat in real time https://signalwire.community
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
> 
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

--
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From olle at zaark.com  Sat Oct  3 14:17:00 2020
From: olle at zaark.com (olle at zaark.com)
Date: Sat, 3 Oct 2020 16:17:00 +0200
Subject: [Freeswitch-users] http_cache and Amazon S3
Message-ID: <023501d6998f$dcf0f400$96d2dc00$@zaark.com>

Hi we have problems with getting S3 access working after upgrading to
1.10.5.

 

This configuration used to work fine in 1.10.3

 

  <profiles>

    <!-- amazon s3 security credentials -->

    <profile name="s3">

      <aws-s3>

        <!-- 20 character key identifier -->

        <access-key-id><![CDATA[.TOKEN.]]></access-key-id>

        <!-- 40 character secret -->

        <secret-access-key><![CDATA[..SECRET-TOKEN.]]></secret-access-key>

      </aws-s3>

      <!-- optional list of domains that this profile will automatically be
applied to -->

      <!-- if you wish to apply the s3 credentials to a domain not listed
here, then use

           {profile=s3}http://foo.s3... -->

      <domains>

 

        <domain name="bucket.s3.amazonaws.com"/>

      </domains>

    </profile>

 

And in dialplan we access with:

<action application="playback"
data="https://${bucket}.s3.amazonaws.com/file.wav

 

We have tried adding region and base_domain (should not be needed) but we
get 403 access error from S3 

 

We can manually access same object using aws cli

 

Does anybody have a sample config and dial plan examples that work with
1.10.5

 

BR/Olle

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From s.safarov at gmail.com  Sat Oct  3 15:00:49 2020
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 3 Oct 2020 18:00:49 +0300
Subject: [Freeswitch-users] http_cache and Amazon S3
In-Reply-To: <023501d6998f$dcf0f400$96d2dc00$@zaark.com>
References: <023501d6998f$dcf0f400$96d2dc00$@zaark.com>
Message-ID: <CAHtxdDfdm8i-5WRAeU+2RGX3xfsTPHet8s5vZBu3pEwuzaPNRQ@mail.gmail.com>

using "git bisect" command you can identify the commit brokes feature.
Then need to revert this commit.

Then you can also report the issue to FreeSwitch bug tracker

Sergey

On Sat, Oct 3, 2020 at 5:54 PM <olle at zaark.com> wrote:

> Hi we have problems with getting S3 access working after upgrading to
> 1.10.5.
>
>
>
> This configuration used to work fine in 1.10.3
>
>
>
>   <profiles>
>
>     <!-- amazon s3 security credentials -->
>
>     <profile name="s3">
>
>       <aws-s3>
>
>         <!-- 20 character key identifier -->
>
>         <access-key-id><![CDATA[…TOKEN…]]></access-key-id>
>
>         <!-- 40 character secret -->
>
>         <secret-access-key><![CDATA[..SECRET-TOKEN…]]></secret-access-key>
>
>       </aws-s3>
>
>       <!-- optional list of domains that this profile will automatically
> be applied to -->
>
>       <!-- if you wish to apply the s3 credentials to a domain not listed
> here, then use
>
>            {profile=s3}http://foo.s3... -->
>
>       <domains>
>
>
>
>         <domain name="bucket.s3.amazonaws.com"/>
>
>       </domains>
>
>     </profile>
>
>
>
> And in dialplan we access with:
>
> <action application="playback" data="https://${bucket}.
> s3.amazonaws.com/file.wav
>
>
>
> We have tried adding region and base_domain (should not be needed) but we
> get 403 access error from S3
>
>
>
> We can manually access same object using aws cli
>
>
>
> Does anybody have a sample config and dial plan examples that work with
> 1.10.5
>
>
>
> BR/Olle
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From s.kainz at wnt.at  Sun Oct  4 16:10:00 2020
From: s.kainz at wnt.at (Stefan Kainz)
Date: Sun, 4 Oct 2020 16:10:00 +0000
Subject: [Freeswitch-users] Microsoft Teams integration
In-Reply-To: <mailman.1680.1601564691.1288.freeswitch-users@lists.freeswitch.org>
References: <VI1P190MB04944EF9D1570CDE35F3E78C863D0@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <D89498C5-DB0D-4C06-ADAC-66E09FBCCBEE@drachtio.org>
 <VI1P190MB04949434E780D5D125633E4386380@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <A94AB550-C012-416A-BACB-0009121B0761@drachtio.org>
 <VI1P190MB0494E51F6504798137148DEB86300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>
 <A384C846-2B00-456F-9ACC-73ED703FBD39@drachtio.org>
 <VI1P190MB049482B1DCA37658CB9A87C586300@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>,
 <mailman.1680.1601564691.1288.freeswitch-users@lists.freeswitch.org>
Message-ID: <VI1P190MB0494EC31BA1F4F81273A5730860F0@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>

That was it! Thank you!
Everything is now working as expected.

Crazy. The amount of time is spent modifiing sip-headers and record-route headers, and all it took was one simple setting in the profile.

Someone should write like a wiki article for connecting fs to teams 🙂

thank you both very much!
have a nice sunday!

regards,
________________________________
Von: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> im Auftrag von Matteo via FreeSWITCH-users <freeswitch-users at lists.freeswitch.org>
Gesendet: Donnerstag, 1. Oktober 2020 18:09
An: freeswitch-users <freeswitch-users at lists.freeswitch.org>
Betreff: Re: [Freeswitch-users] Microsoft Teams integration

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
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From michal.bielicki at seventhsignal.de  Sun Oct  4 17:31:51 2020
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Sun, 04 Oct 2020 17:31:51 +0000
Subject: [Freeswitch-users] Microsoft Teams integration
In-Reply-To: <VI1P190MB0494EC31BA1F4F81273A5730860F0@vi1p190mb0494.eurp190.prod.outlook.com>
References: <VI1P190MB04944EF9D1570CDE35F3E78C863D0@vi1p190mb0494.eurp190.prod.outlook.com>
 <D89498C5-DB0D-4C06-ADAC-66E09FBCCBEE@drachtio.org>
 <VI1P190MB04949434E780D5D125633E4386380@vi1p190mb0494.eurp190.prod.outlook.com>
 <A94AB550-C012-416A-BACB-0009121B0761@drachtio.org>
 <VI1P190MB0494E51F6504798137148DEB86300@vi1p190mb0494.eurp190.prod.outlook.com>
 <A384C846-2B00-456F-9ACC-73ED703FBD39@drachtio.org>
 <VI1P190MB049482B1DCA37658CB9A87C586300@vi1p190mb0494.eurp190.prod.outlook.com>
 <VI1P190MB0494EC31BA1F4F81273A5730860F0@vi1p190mb0494.eurp190.prod.outlook.com>
Message-ID: <em0c8cc195-0047-456b-bd6c-ccd1e2afa78d@michal-norisbook-266.local>

Which setting was it in the end ?

------ Original Message ------
From: "Stefan Kainz" <s.kainz at wnt.at>
To: "freeswitch-users" <freeswitch-users at lists.freeswitch.org>; "Matteo" 
<mbrancaleoni at voismart.it>; "David Horton" <daveh at drachtio.org>
Sent: 04/10/2020 18:10:00
Subject: Re: [Freeswitch-users] Microsoft Teams integration

>That was it! Thank you!
>Everything is now working as expected.
>
>Crazy. The amount of time is spent modifiing sip-headers and 
>record-route headers, and all it took was one simple setting in the 
>profile.
>
>Someone should write like a wiki article for connecting fs to teams 🙂
>
>thank you both very much!
>have a nice sunday!
>
>regards,
>--------------------------------------------------------------------------------
>Von: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> 
>im Auftrag von Matteo via FreeSWITCH-users 
><freeswitch-users at lists.freeswitch.org>
>Gesendet: Donnerstag, 1. Oktober 2020 18:09
>An: freeswitch-users <freeswitch-users at lists.freeswitch.org>
>Betreff: Re: [Freeswitch-users] Microsoft Teams integration
>
>_________________________________________________________________________
>
>The FreeSWITCH project is sponsored by SignalWire 
>https://signalwire.com <https://signalwire.com>
>Enhance your FreeSWITCH install with disruptive priced SMS and PSTN 
>services.
>Build your next product on our scalable cloud platform.
>
>Join our online community to chat in real time 
>https://signalwire.community <https://signalwire.community>
>
>Professional FreeSWITCH Services
>sales at freeswitch.com
>https://freeswitch.com
>
>Official FreeSWITCH Sites
>https://freeswitch.com/oss
>https://freeswitch.org/confluence
>https://cluecon.com
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>https://freeswitch.com
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From s.kainz at wnt.at  Mon Oct  5 07:27:25 2020
From: s.kainz at wnt.at (Stefan Kainz)
Date: Mon, 5 Oct 2020 07:27:25 +0000
Subject: [Freeswitch-users] Microsoft Teams integration
In-Reply-To: <mailman.1756.1601832747.1288.freeswitch-users@lists.freeswitch.org>
References: <VI1P190MB04944EF9D1570CDE35F3E78C863D0@vi1p190mb0494.eurp190.prod.outlook.com>
 <D89498C5-DB0D-4C06-ADAC-66E09FBCCBEE@drachtio.org>
 <VI1P190MB04949434E780D5D125633E4386380@vi1p190mb0494.eurp190.prod.outlook.com>
 <A94AB550-C012-416A-BACB-0009121B0761@drachtio.org>
 <VI1P190MB0494E51F6504798137148DEB86300@vi1p190mb0494.eurp190.prod.outlook.com>
 <A384C846-2B00-456F-9ACC-73ED703FBD39@drachtio.org>
 <VI1P190MB049482B1DCA37658CB9A87C586300@vi1p190mb0494.eurp190.prod.outlook.com>
 <VI1P190MB0494EC31BA1F4F81273A5730860F0@vi1p190mb0494.eurp190.prod.outlook.com>,
 <mailman.1756.1601832747.1288.freeswitch-users@lists.freeswitch.org>
Message-ID: <VI1P190MB049407531246DF55DDE38F52860C0@VI1P190MB0494.EURP190.PROD.OUTLOOK.COM>

Hi,

there were a few critical things, here are the ones that caused me some headaches:

1) Gateway needs "contact-in-ping" set to true, otherwise Teams will reject OPTIONS
( Which does exist since freeswitch 1.10.3 as i understand, so pretty new. )

1.1) It takes a little time for Microsoft Teams to recognize the sbc to be active in their SBC Overview.

2) "sip-ip" and "ext-sip-ip" have to be set to the sbc-domain you use in Teams.
( Which is sort of strange, because in my profile configs it says "do not use domains", so ... we will see if that causes any trouble. Hasn't so far. )

3) TLS - we tried letsencrypt for a very long time, but it didnt work, so we got our certificate from godaddy, which works like a charm.
( I've read that others got it to work with letsencrypt, so it might just be my incompetence )

4) Debugging is sort of hard to do since communication with teams is encrypted.
Use "sofia profile xxx siptrace on" in fs_cli. Once you see the decrypted messages, Microsoft sometimes has useful Error-messages in their sip-responses.

We now have a fully compatible multi-tenant setup, which is sort of cool.
For this, you need at least 4 defined Gateways per customer ( at least one for the provider side, and three for teams, because they have three proxies ) but we used mod_lua to load the configuration dynamically, so thats managable.
But i can remember that there are some limitations when it comes to gateways.
Something like, you can have a maximum of 255 gateways.
Please correct me if im wrong here.

regards,
________________________________
Von: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> im Auftrag von Michal Bielicki via FreeSWITCH-users <freeswitch-users at lists.freeswitch.org>
Gesendet: Sonntag, 4. Oktober 2020 20:09
An: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>; freeswitch-users <freeswitch-users at lists.freeswitch.org>; Matteo <mbrancaleoni at voismart.it>; David Horton <daveh at drachtio.org>
Betreff: Re: [Freeswitch-users] Microsoft Teams integration

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
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From harshita19012 at iiitd.ac.in  Mon Oct  5 04:42:41 2020
From: harshita19012 at iiitd.ac.in (Harshita Pandey)
Date: Mon, 5 Oct 2020 10:12:41 +0530
Subject: [Freeswitch-users] Gateway Configuration
Message-ID: <CA+by-=WZkCCAa1rhUkeXmo=aS83822jMfW+n7jhSog-sTSvBJQ@mail.gmail.com>

Hi,

I have installed freeswitch on a Virtual machine (Ubuntu 20.04). When I am
trying to add a gateway, it shows FAIL_WAIT in sofia status and also gives
the following error on the freeswitch command line.
 2020-10-05 04:40:01.528063 [NOTICE] sofia_reg.c:453 Registering
GatewayIIITD                                            2020-10-05
04:40:33.528067 [ERR] sofia_reg.c:2469 GatewayIIITD Failed Registration
with status Request Timeout [408]. failure #6

     2020-10-05 04:40:33.568078 [WARNING] sofia_reg.c:510 GatewayIIITD
Failed Registration [408], setting retry to 180 seconds.

Much help needed.
Thanks in advance.
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From botelist at gmail.com  Tue Oct  6 08:17:24 2020
From: botelist at gmail.com (Bote Man)
Date: Tue, 6 Oct 2020 04:17:24 -0400
Subject: [Freeswitch-users] Gateway Configuration
In-Reply-To: <CA+by-=WZkCCAa1rhUkeXmo=aS83822jMfW+n7jhSog-sTSvBJQ@mail.gmail.com>
References: <CA+by-=WZkCCAa1rhUkeXmo=aS83822jMfW+n7jhSog-sTSvBJQ@mail.gmail.com>
Message-ID: <00b301d69bb9$1fac2a00$5f047e00$@gmail.com>

Can your FreeSWITCH machine ping the gateway?

 

Is the gateway configured properly with the correct IP number or DNS name and port number?

 

Is there a firewall rule blocking access?

 

That’s all that I can think of without more specific information.

 

All the best.

 

 

--- 

John Boteler 

BnC Group U.S.A. 

 

 

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Harshita Pandey
Sent: Monday, 5 October, 2020 00:43
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Gateway Configuration

 

Hi,

 

I have installed freeswitch on a Virtual machine (Ubuntu 20.04). When I am trying to add a gateway, it shows FAIL_WAIT in sofia status and also gives the following error on the freeswitch command line.

 2020-10-05 04:40:01.528063 [NOTICE] sofia_reg.c:453 Registering GatewayIIITD                                            2020-10-05 04:40:33.528067 [ERR] sofia_reg.c:2469 GatewayIIITD Failed Registration with status Request Timeout [408]. failure #6                                                                                                                2020-10-05 04:40:33.568078 [WARNING] sofia_reg.c:510 GatewayIIITD Failed Registration [408], setting retry to 180 seconds. 

 

Much help needed.

Thanks in advance.

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From david.villasmil.work at gmail.com  Tue Oct  6 08:23:28 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 6 Oct 2020 09:23:28 +0100
Subject: [Freeswitch-users] Gateway Configuration
In-Reply-To: <CA+by-=WZkCCAa1rhUkeXmo=aS83822jMfW+n7jhSog-sTSvBJQ@mail.gmail.com>
References: <CA+by-=WZkCCAa1rhUkeXmo=aS83822jMfW+n7jhSog-sTSvBJQ@mail.gmail.com>
Message-ID: <CAFGRPVqFL9ROf2heMB2Uwd1ZH5Pk102ehWnz8D5Uf7j2xCJ-oQ@mail.gmail.com>

It looks like the gateway on which you are trying to register is not
responding. Get a trace by executing “sofia global siptrace on” and reload
the gateway. Make sure you can ping the iPad of the gateway and that the
firewall (if any) is open for FS.

On Tue, 6 Oct 2020 at 07:54, Harshita Pandey <harshita19012 at iiitd.ac.in>
wrote:

> Hi,
>
> I have installed freeswitch on a Virtual machine (Ubuntu 20.04). When I am
> trying to add a gateway, it shows FAIL_WAIT in sofia status and also gives
> the following error on the freeswitch command line.
>  2020-10-05 04:40:01.528063 [NOTICE] sofia_reg.c:453 Registering
> GatewayIIITD                                            2020-10-05
> 04:40:33.528067 [ERR] sofia_reg.c:2469 GatewayIIITD Failed Registration
> with status Request Timeout [408]. failure #6
>
>      2020-10-05 04:40:33.568078 [WARNING] sofia_reg.c:510 GatewayIIITD
> Failed Registration [408], setting retry to 180 seconds.
>
> Much help needed.
> Thanks in advance.
>
>
> _________________________________________________________________________
>
>
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
>
> Build your next product on our scalable cloud platform.
>
>
>
> Join our online community to chat in real time
> https://signalwire.community
>
>
>
> Professional FreeSWITCH Services
>
> sales at freeswitch.com
>
> https://freeswitch.com
>
>
>
> Official FreeSWITCH Sites
>
> https://freeswitch.com/oss
>
> https://freeswitch.org/confluence
>
> https://cluecon.com
>
>
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From nathan at robotics.net  Tue Oct  6 09:43:46 2020
From: nathan at robotics.net (Nathan Stratton)
Date: Tue, 6 Oct 2020 05:43:46 -0400
Subject: [Freeswitch-users] Why is transfer not resulting in a reINVITE?
Message-ID: <CAHubkyO3b=Zbw=j8E221Scb5_JYbU-psvUbce1y57m2exe5-4w@mail.gmail.com>

I am starting a call with OPUS only on the B leg, the call starts fine with
inbound-late-negotiation, and OPUS is sent back on the A leg. However, the
call is then transferred to a user with OPUS and VP8 video with the new
INVITE, but nothing ever is sent back to the original A leg to update VP8.
Is it possible to send a reINVITE back so I can update the SDP with VP8 on
the original A leg?

><>
nathan stratton
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From shahnawaj.khan1990 at gmail.com  Tue Oct  6 13:41:32 2020
From: shahnawaj.khan1990 at gmail.com (Shahnawaj Khan)
Date: Tue, 6 Oct 2020 19:11:32 +0530
Subject: [Freeswitch-users] Freeswitch unable to redirect to contact
	received in 302 Moved Temporarily
Message-ID: <CAGmow12rRwDgA6LaKb7gWJAKpFPa+mZ48m-LEdoRKOf96poFGw@mail.gmail.com>

Hi,

I am running a test scenario where freeswitch is getting "302 Moved
Temporarily" in response to an invite generated for b-leg. Response
also contains the contact for redirection but freeswitch is unable to
process the response. It is neither sending ACK for the initiated
transaction nor generating the new invite for the contact received in
response. my dialplan looks like this.

<extension name="302">
            <condition field="destination_number" expression="^(5010)$">
                    <action application="bridge"
data="sofia/3clogic_external/5010@<IP>:6060"/>
            </condition>
        </extension>

I don't want to manually handle the 302 response. I just need FS to
forward the call. Please let me know if anything needs to be added in
dialplan or some configuration needs to be updated.

I have also attached the console logs of the freeswitch with siptrace.

Thanks & Regards,
Shahnawaz
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From botelist at gmail.com  Wed Oct  7 08:03:31 2020
From: botelist at gmail.com (Bote Man)
Date: Wed, 7 Oct 2020 04:03:31 -0400
Subject: [Freeswitch-users] Freeswitch unable to redirect to
	contact	received in 302 Moved Temporarily
In-Reply-To: <CAGmow12rRwDgA6LaKb7gWJAKpFPa+mZ48m-LEdoRKOf96poFGw@mail.gmail.com>
References: <CAGmow12rRwDgA6LaKb7gWJAKpFPa+mZ48m-LEdoRKOf96poFGw@mail.gmail.com>
Message-ID: <004101d69c80$592aea60$0b80bf20$@gmail.com>

Perhaps this is by design. 

A mailing list message from 2016 advises:
"But bear in mind that the client might not follow the 302. FreeSWITCH for example either terminates the call on 302 or returns to the dialplan so you can validate the new destination, so that you don't (as an example) have someone redirecting you to a premium rate number without the caller's knowledge."

Perhaps you might be required to set some combination of
hangup_after_bridge=false
or
continue_on_fail=true

and continue processing the call? I don't know what else you could do at that point.


 --- 
John Boteler 
BnC Group U.S.A. 



-----Original Message-----
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Shahnawaj Khan
Sent: Tuesday, 6 October, 2020 09:42
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Freeswitch unable to redirect to contact received in 302 Moved Temporarily

Hi,

I am running a test scenario where freeswitch is getting "302 Moved Temporarily" in response to an invite generated for b-leg. Response also contains the contact for redirection but freeswitch is unable to process the response. It is neither sending ACK for the initiated transaction nor generating the new invite for the contact received in response. my dialplan looks like this.

<extension name="302">
            <condition field="destination_number" expression="^(5010)$">
                    <action application="bridge"
data="sofia/3clogic_external/5010@<IP>:6060"/>
            </condition>
        </extension>

I don't want to manually handle the 302 response. I just need FS to forward the call. Please let me know if anything needs to be added in dialplan or some configuration needs to be updated.

I have also attached the console logs of the freeswitch with siptrace.

Thanks & Regards,
Shahnawaz



From harshita19012 at iiitd.ac.in  Thu Oct  8 11:01:45 2020
From: harshita19012 at iiitd.ac.in (Harshita Pandey)
Date: Thu, 8 Oct 2020 16:31:45 +0530
Subject: [Freeswitch-users] Port Required to communicate with Gateway
Message-ID: <CA+by-=WB_zzT1NMS6-AKFAjjNs3k1Hv5GYaRqOi3_MbyTCoO7A@mail.gmail.com>

Hi,

I have installed freeswitch on a Virtual Machine, and I am not able to
configure the gateway for some reason. Can someone tell me which ports are
required and need to be opened on the virtual machine to communicate with
Gateway in freeswitch?
Since all the ports are not opened in the Virtual Machine, this could help
me open up those ports and configure Gateway for freeswitch


Much help is needed.
It's Urgent
Thanks in advance.
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From covici at ccs.covici.com  Thu Oct  8 12:13:38 2020
From: covici at ccs.covici.com (John Covici)
Date: Thu, 08 Oct 2020 08:13:38 -0400
Subject: [Freeswitch-users] Port Required to communicate with Gateway
In-Reply-To: <CA+by-=WB_zzT1NMS6-AKFAjjNs3k1Hv5GYaRqOi3_MbyTCoO7A@mail.gmail.com>
References: <CA+by-=WB_zzT1NMS6-AKFAjjNs3k1Hv5GYaRqOi3_MbyTCoO7A@mail.gmail.com>
Message-ID: <m3o8lc6hl9.wl-covici@ccs.covici.com>


On Thu, 08 Oct 2020 07:01:45 -0400,
Harshita Pandey wrote:
> 
> [1  <multipart/alternative (7bit)>]
> [1.1  <text/plain; UTF-8 (7bit)>]
> Hi,
> 
> I have installed freeswitch on a Virtual Machine, and I am not able to
> configure the gateway for some reason. Can someone tell me which ports are
> required and need to be opened on the virtual machine to communicate with
> Gateway in freeswitch?
> Since all the ports are not opened in the Virtual Machine, this could help
> me open up those ports and configure Gateway for freeswitch
> 

Depending on your configuration, you may need 5060 through 5080.  But
you need to know the ports, each freeswitch profile must specify the
ip address and port it listens on.

I hope that helps.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici wb2una
         covici at ccs.covici.com


From botelist at gmail.com  Thu Oct  8 12:13:20 2020
From: botelist at gmail.com (Bote Man)
Date: Thu, 8 Oct 2020 08:13:20 -0400
Subject: [Freeswitch-users] Port Required to communicate with Gateway
In-Reply-To: <CA+by-=WB_zzT1NMS6-AKFAjjNs3k1Hv5GYaRqOi3_MbyTCoO7A@mail.gmail.com>
References: <CA+by-=WB_zzT1NMS6-AKFAjjNs3k1Hv5GYaRqOi3_MbyTCoO7A@mail.gmail.com>
Message-ID: <009801d69d6c$6a077230$3e165690$@gmail.com>

For the SIP control channel the conventional port is 5060 for UDP, or 5061 for TLS. But these can be changed in each config file under 

${conf_dir}/sip_profiles

If you run more than one SIP profile on a single IP address, each profile must use a different port, of course.

 

The RTP media ports can be configured in

${conf_dir}/autoload_configs/switch.conf.xml: 

 

<param name="rtp-start-port" value="16384"/>

<param name="rtp-end-port" value="32768"/>

 

You must restart FreeSWITCH after changing these values.

 

These articles on the wiki might help you further:

https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2

 

https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal

 

Hope this helps.

 

 

--- 

John Boteler 

BnC Group U.S.A. 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Harshita Pandey
Sent: Thursday, 8 October, 2020 07:02
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Port Required to communicate with Gateway

 

Hi,

 

I have installed freeswitch on a Virtual Machine, and I am not able to configure the gateway for some reason. Can someone tell me which ports are required and need to be opened on the virtual machine to communicate with Gateway in freeswitch? 

Since all the ports are not opened in the Virtual Machine, this could help me open up those ports and configure Gateway for freeswitch

 

 

Much help is needed. 

It's Urgent

Thanks in advance.

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From joeygo at gmail.com  Wed Oct  7 15:06:10 2020
From: joeygo at gmail.com (Joey Golan)
Date: Wed, 7 Oct 2020 18:06:10 +0300
Subject: [Freeswitch-users] Bridge 2 conferences
References: <76a5ca71-822b-411d-8b5c-a50f2005d05b@Spark>
Message-ID: <6535c437-be51-4221-bf1c-9a7470788818@Spark>

Hello,
I have 2 conferences with 50 users on each one of them and one speaker total.
I would like to bridge between them so all the 99 will hear the speaker. How can I achieve that?

Thanks,
Joey.
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From rangeare.01 at gmail.com  Thu Oct  8 12:47:19 2020
From: rangeare.01 at gmail.com (Jay Desai)
Date: Thu, 8 Oct 2020 18:17:19 +0530
Subject: [Freeswitch-users] Port Required to communicate with Gateway
In-Reply-To: <CA+by-=WB_zzT1NMS6-AKFAjjNs3k1Hv5GYaRqOi3_MbyTCoO7A@mail.gmail.com>
References: <CA+by-=WB_zzT1NMS6-AKFAjjNs3k1Hv5GYaRqOi3_MbyTCoO7A@mail.gmail.com>
Message-ID: <CAGvKJUWdUSvXja99N5XN=fR6AVY9t26tfT6SWxrL9MOCFpA6Kg@mail.gmail.com>

Freeswitch uses different profiles for outgoing and incoming calls.
The default port for an internal profile is 5060 and external profile uses
5080.
Can you provide more details about the gateway and what you are trying to
do?

On Thu, Oct 8, 2020 at 4:58 PM Harshita Pandey <harshita19012 at iiitd.ac.in>
wrote:

> Hi,
>
> I have installed freeswitch on a Virtual Machine, and I am not able to
> configure the gateway for some reason. Can someone tell me which ports are
> required and need to be opened on the virtual machine to communicate with
> Gateway in freeswitch?
> Since all the ports are not opened in the Virtual Machine, this could help
> me open up those ports and configure Gateway for freeswitch
>
>
> Much help is needed.
> It's Urgent
> Thanks in advance.
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From harshita19012 at iiitd.ac.in  Thu Oct  8 14:33:37 2020
From: harshita19012 at iiitd.ac.in (Harshita Pandey)
Date: Thu, 8 Oct 2020 20:03:37 +0530
Subject: [Freeswitch-users] Port Required to communicate with Gateway
In-Reply-To: <m3o8lc6hl9.wl-covici@ccs.covici.com>
References: <CA+by-=WB_zzT1NMS6-AKFAjjNs3k1Hv5GYaRqOi3_MbyTCoO7A@mail.gmail.com>
 <m3o8lc6hl9.wl-covici@ccs.covici.com>
Message-ID: <CA+by-=UN5WYLELXb_DZOZRxFdV+hCeG0DY0EGEAJaDHa-K8Cqg@mail.gmail.com>

How do I find The IP address and port to which the freeswitch I have
installed is listening? Actually, I tried running the command "sudo netstat
nltp" and it is showing various ports in which freeswitch is running.

On Thu, Oct 8, 2020 at 5:43 PM John Covici <covici at ccs.covici.com> wrote:

>
> On Thu, 08 Oct 2020 07:01:45 -0400,
> Harshita Pandey wrote:
> >
> > [1  <multipart/alternative (7bit)>]
> > [1.1  <text/plain; UTF-8 (7bit)>]
> > Hi,
> >
> > I have installed freeswitch on a Virtual Machine, and I am not able to
> > configure the gateway for some reason. Can someone tell me which ports
> are
> > required and need to be opened on the virtual machine to communicate with
> > Gateway in freeswitch?
> > Since all the ports are not opened in the Virtual Machine, this could
> help
> > me open up those ports and configure Gateway for freeswitch
> >
>
> Depending on your configuration, you may need 5060 through 5080.  But
> you need to know the ports, each freeswitch profile must specify the
> ip address and port it listens on.
>
> I hope that helps.
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>          John Covici wb2una
>          covici at ccs.covici.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From harshita19012 at iiitd.ac.in  Thu Oct  8 14:38:49 2020
From: harshita19012 at iiitd.ac.in (Harshita Pandey)
Date: Thu, 8 Oct 2020 20:08:49 +0530
Subject: [Freeswitch-users] Port Required to communicate with Gateway
In-Reply-To: <CAGvKJUWdUSvXja99N5XN=fR6AVY9t26tfT6SWxrL9MOCFpA6Kg@mail.gmail.com>
References: <CA+by-=WB_zzT1NMS6-AKFAjjNs3k1Hv5GYaRqOi3_MbyTCoO7A@mail.gmail.com>
 <CAGvKJUWdUSvXja99N5XN=fR6AVY9t26tfT6SWxrL9MOCFpA6Kg@mail.gmail.com>
Message-ID: <CA+by-=Ve-XpzY1wnEOy+MUPowTXXbAXMxV0Q1pFXwiJHgndFLg@mail.gmail.com>

Okay. I am actually trying to configure Gateway so that I can start making
calls through a number. The virtual machine in which I have installed
freeswitch has only port 22 opened, so wanted to know the other ports which
need to be opened to configure the Gateway correctly.

I get the following errors:
MySIP Failed Registration with status Forbidden [403]. failure #1still
getting the same error
OR
MySIP failed Registration error [503]. retrying in 30 seconds.

According to what I have understood this is because some ports are not
opened in the VM

On Thu, Oct 8, 2020 at 7:52 PM Jay Desai <rangeare.01 at gmail.com> wrote:

> Freeswitch uses different profiles for outgoing and incoming calls.
> The default port for an internal profile is 5060 and external profile uses
> 5080.
> Can you provide more details about the gateway and what you are trying to
> do?
>
> On Thu, Oct 8, 2020 at 4:58 PM Harshita Pandey <harshita19012 at iiitd.ac.in>
> wrote:
>
>> Hi,
>>
>> I have installed freeswitch on a Virtual Machine, and I am not able to
>> configure the gateway for some reason. Can someone tell me which ports are
>> required and need to be opened on the virtual machine to communicate with
>> Gateway in freeswitch?
>> Since all the ports are not opened in the Virtual Machine, this could
>> help me open up those ports and configure Gateway for freeswitch
>>
>>
>> Much help is needed.
>> It's Urgent
>> Thanks in advance.
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From makafre at gmail.com  Thu Oct  8 14:37:59 2020
From: makafre at gmail.com (Frederic Jean)
Date: Thu, 08 Oct 2020 14:37:59 +0000
Subject: [Freeswitch-users] Defunct processes from starting a script
Message-ID: <em44b00752-a749-4146-ac68-09be1d032b59@fred-pc>


Dear community,

Newbie here; I am currently experiencing aleatory defunct processes by 
starting a Perl script this way:

<action application="set" data= 
"ivr_path=/usr/local/freeswitch/scripts/myscript.pl"/>
<action application="socket" data="127.0.0.1:9080 sync full"/>

Are there any better ways to have it executed and avoid these?

Regards
Fred
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From per at wgtwo.com  Thu Oct  8 15:25:06 2020
From: per at wgtwo.com (Per Modin)
Date: Thu, 8 Oct 2020 17:25:06 +0200
Subject: [Freeswitch-users] Port Required to communicate with Gateway
In-Reply-To: <CA+by-=WB_zzT1NMS6-AKFAjjNs3k1Hv5GYaRqOi3_MbyTCoO7A@mail.gmail.com>
References: <CA+by-=WB_zzT1NMS6-AKFAjjNs3k1Hv5GYaRqOi3_MbyTCoO7A@mail.gmail.com>
Message-ID: <20201008152506.ep3uthp3uflrwrii@wg2modin>

On 2020-10-08 16:31 (Thu), Harshita Pandey wrote:
> Can someone tell me which ports are required and need to be opened on
> the virtual machine to communicate with Gateway in freeswitch?

I've snipped this from our dockerfile, not all might be needed thou.

ports:
- "16384:16384"
- "1719:1719"
- "1720:1720"
- "3478:3478"
- "3479:3479"
- "5002:5002"
- "5003:5003"
- "5060:5060/tcp"
- "5060:5060/udp"
- "5066:5066"
- "5070:5070"
- "5080:5080/tcp"
- "5080:5080/udp"
- "7443:7443"
- "8021:8021"
- "64535-64555:64535-64555/udp"

Best,
Per.
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From botelist at gmail.com  Thu Oct  8 16:48:07 2020
From: botelist at gmail.com (Bote Man)
Date: Thu, 8 Oct 2020 12:48:07 -0400
Subject: [Freeswitch-users] Defunct processes from starting a script
In-Reply-To: <em44b00752-a749-4146-ac68-09be1d032b59@fred-pc>
References: <em44b00752-a749-4146-ac68-09be1d032b59@fred-pc>
Message-ID: <002601d69d92$ccec1ba0$66c452e0$@gmail.com>

I do not know what your script does, and that determines the best approach to solve the problem.

 

In the only case with which I have experience, I run a perl script as a stand-alone linux process using the normal perl interpreter. This script then connects to FreeSWITCH via ESL (the Event Socket Library) and listens for certain FreeSWITCH events, then reacts to them. This removes the script from the telephony processing loop in the dialplan and eliminates any possible blocking of telephony processing in the dialplan (which seems to be a common problem with many script implementations when called from the dialplan).

 

This method requires compiling the ESL.pm module which is part of the FS source; this module provides the connection from perl to FS. Install it using these instructions:

 

https://freeswitch.org/confluence/display/FREESWITCH/Event+Socket+Library#EventSocketLibrary-Installation

 

There are perl examples on this wiki page:

https://freeswitch.org/confluence/display/FREESWITCH/Perl+ESL

 

That’s not much, but I hope it helps.

 

 

--- 

John Boteler 

BnC Group U.S.A. 

 

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Frederic Jean
Sent: Thursday, 8 October, 2020 10:38
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Defunct processes from starting a script

 

 

Dear community,

 

Newbie here; I am currently experiencing aleatory defunct processes by starting a Perl script this way:

 

<action application="set" data= "ivr_path=/usr/local/freeswitch/scripts/myscript.pl"/>

<action application="socket" data="127.0.0.1:9080 sync full"/>

 

Are there any better ways to have it executed and avoid these?

 

Regards

Fred

 

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From rangeare.01 at gmail.com  Thu Oct  8 20:46:10 2020
From: rangeare.01 at gmail.com (Jay Desai)
Date: Fri, 9 Oct 2020 02:16:10 +0530
Subject: [Freeswitch-users] Port Required to communicate with Gateway
In-Reply-To: <CA+by-=Ve-XpzY1wnEOy+MUPowTXXbAXMxV0Q1pFXwiJHgndFLg@mail.gmail.com>
References: <CA+by-=WB_zzT1NMS6-AKFAjjNs3k1Hv5GYaRqOi3_MbyTCoO7A@mail.gmail.com>
 <CAGvKJUWdUSvXja99N5XN=fR6AVY9t26tfT6SWxrL9MOCFpA6Kg@mail.gmail.com>
 <CA+by-=Ve-XpzY1wnEOy+MUPowTXXbAXMxV0Q1pFXwiJHgndFLg@mail.gmail.com>
Message-ID: <CAGvKJUV+AMThxzT+ngevqC7hZx30LaNChdD7n0=GPaVB6c1YJA@mail.gmail.com>

I suspect this to be something related to your sip configuration. If you
are getting 4XX or 5XX response from gateway means the ports are open (you
don't have to open ports for outbound connections unless they are
explicitly blocked by any firewall or program.)

Share your sip trace by executing *sofia global sip trace on* or by using
the *tcpdump* command.

On Thu, Oct 8, 2020 at 8:44 PM Harshita Pandey <harshita19012 at iiitd.ac.in>
wrote:

> Okay. I am actually trying to configure Gateway so that I can start making
> calls through a number. The virtual machine in which I have installed
> freeswitch has only port 22 opened, so wanted to know the other ports which
> need to be opened to configure the Gateway correctly.
>
> I get the following errors:
> MySIP Failed Registration with status Forbidden [403]. failure #1still
> getting the same error
> OR
> MySIP failed Registration error [503]. retrying in 30 seconds.
>
> According to what I have understood this is because some ports are not
> opened in the VM
>
> On Thu, Oct 8, 2020 at 7:52 PM Jay Desai <rangeare.01 at gmail.com> wrote:
>
>> Freeswitch uses different profiles for outgoing and incoming calls.
>> The default port for an internal profile is 5060 and external profile
>> uses 5080.
>> Can you provide more details about the gateway and what you are trying to
>> do?
>>
>> On Thu, Oct 8, 2020 at 4:58 PM Harshita Pandey <harshita19012 at iiitd.ac.in>
>> wrote:
>>
>>> Hi,
>>>
>>> I have installed freeswitch on a Virtual Machine, and I am not able to
>>> configure the gateway for some reason. Can someone tell me which ports are
>>> required and need to be opened on the virtual machine to communicate with
>>> Gateway in freeswitch?
>>> Since all the ports are not opened in the Virtual Machine, this could
>>> help me open up those ports and configure Gateway for freeswitch
>>>
>>>
>>> Much help is needed.
>>> It's Urgent
>>> Thanks in advance.
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From davidswalkabout at gmail.com  Thu Oct  8 22:26:13 2020
From: davidswalkabout at gmail.com (David P)
Date: Fri, 9 Oct 2020 11:26:13 +1300
Subject: [Freeswitch-users] RECORD_STEREO and record_sample_rate aren't
	working with conference
Message-ID: <CAE23tru_7TRWKQ9qrxyRVJriVEjnOFNEYtaeHzDRtOt097zAFw@mail.gmail.com>

We're using FSv10.5 on Debian 10 with verto/Opus (i.e. no codec explicitly
configured), and we're using a conference. We've tried these settings but
they're having no effect on the recorded mp4s:

<action application="set" data="RECORD_STEREO=true"/>
<action application="set" data="record_sample_rate=44100"/>

Instead:
1) The mp4s have the same audio in both channels, contrary to
https://freeswitch.org/confluence/display/FREESWITCH/RECORD_STEREO

2) The sampling rate is 8kHz, contrary to
https://freeswitch.org/confluence/display/FREESWITCH/record_sample_rate

FWIW:
a) We noticed that mp4s from a year ago when we were using FS 1.8 also have
the same audio on both channels.

b) We also have this, and I don't recall why and can't find a description
in confluence:
<action application="set" data="record_concat_video=true"/>

Is it that these are known not to work with conferences, or is there some
setting we might have overlooked?
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From brian at freeswitch.com  Fri Oct  9 00:06:52 2020
From: brian at freeswitch.com (Brian West)
Date: Thu, 8 Oct 2020 19:06:52 -0500
Subject: [Freeswitch-users] RECORD_STEREO and record_sample_rate aren't
 working with conference
In-Reply-To: <CAE23tru_7TRWKQ9qrxyRVJriVEjnOFNEYtaeHzDRtOt097zAFw@mail.gmail.com>
References: <CAE23tru_7TRWKQ9qrxyRVJriVEjnOFNEYtaeHzDRtOt097zAFw@mail.gmail.com>
Message-ID: <CAEJMVkAVUW7=TZ_x9SvFxHCUuZhPhMVtppL8cw3ekOtHVZJ+EA@mail.gmail.com>

Yah that's probably true, you should file issues on github please.

On Thu, Oct 8, 2020 at 5:59 PM David P <davidswalkabout at gmail.com> wrote:

> We're using FSv10.5 on Debian 10 with verto/Opus (i.e. no codec explicitly
> configured), and we're using a conference. We've tried these settings but
> they're having no effect on the recorded mp4s:
>
> <action application="set" data="RECORD_STEREO=true"/>
> <action application="set" data="record_sample_rate=44100"/>
>
> Instead:
> 1) The mp4s have the same audio in both channels, contrary to
> https://freeswitch.org/confluence/display/FREESWITCH/RECORD_STEREO
>
> 2) The sampling rate is 8kHz, contrary to
> https://freeswitch.org/confluence/display/FREESWITCH/record_sample_rate
>
> FWIW:
> a) We noticed that mp4s from a year ago when we were using FS 1.8 also
> have the same audio on both channels.
>
> b) We also have this, and I don't recall why and can't find a description
> in confluence:
> <action application="set" data="record_concat_video=true"/>
>
> Is it that these are known not to work with conferences, or is there some
> setting we might have overlooked?
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>

Email: brian at freeswitch.com

Mobile: 918-424-9378

Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>

[image: https://www.facebook.com/signalwireinc?src=email]
<https://www.facebook.com/freeswitch> [image:
https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
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From svanherwaarden at precisionag.org  Fri Oct  9 07:44:29 2020
From: svanherwaarden at precisionag.org (Sam van Herwaarden)
Date: Fri, 9 Oct 2020 09:44:29 +0200
Subject: [Freeswitch-users] IVR UI/UX design
Message-ID: <CAAtWTekEPFw4AxxigLfd0d9nzMrW+1-WFFT5LUvd3u2Fo_VNYg@mail.gmail.com>

Hi all,

My organization runs a number of IVR systems for information delivery, and
one issue we often run into is that users find the interface a bit awkward
or hard to navigate. We have experimented quite a bit internally with
adjusting the structures of menus, ordering, phrasing etc. But we feel like
there must be some knowledge/experience out there and that it shouldn't be
necessary to rediscover most of this ourselves.

Does anyone have recommendations for resources with info on best practices
for IVR design? Or contacts in organizations/companies that have a lot of
experience with this?

Kind regards,
Sam
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From botelist at gmail.com  Fri Oct  9 13:22:22 2020
From: botelist at gmail.com (Bote Man)
Date: Fri, 9 Oct 2020 09:22:22 -0400
Subject: [Freeswitch-users] IVR UI/UX design
In-Reply-To: <CAAtWTekEPFw4AxxigLfd0d9nzMrW+1-WFFT5LUvd3u2Fo_VNYg@mail.gmail.com>
References: <CAAtWTekEPFw4AxxigLfd0d9nzMrW+1-WFFT5LUvd3u2Fo_VNYg@mail.gmail.com>
Message-ID: <009901d69e3f$3943a0f0$abcae2d0$@gmail.com>

I don’t know the answer, but there’s a good possibility that a fine lady who records professional IVR prompts knows where to point you:

 

Allison at theivrvoice.com <mailto:Allison at theivrvoice.com> 

 

 

--- 

John Boteler 

BnC Group U.S.A. 

 

 

 

 

From: Sam van Herwaarden
Sent: Friday, 9 October, 2020 03:44
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] IVR UI/UX design

 

Hi all,

 

My organization runs a number of IVR systems for information delivery, and one issue we often run into is that users find the interface a bit awkward or hard to navigate. We have experimented quite a bit internally with adjusting the structures of menus, ordering, phrasing etc. But we feel like there must be some knowledge/experience out there and that it shouldn't be necessary to rediscover most of this ourselves.

 

Does anyone have recommendations for resources with info on best practices for IVR design? Or contacts in organizations/companies that have a lot of experience with this?

 

Kind regards,

Sam

 

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From makafre at gmail.com  Fri Oct  9 14:11:38 2020
From: makafre at gmail.com (Frederic Jean)
Date: Fri, 09 Oct 2020 14:11:38 +0000
Subject: [Freeswitch-users] IVR UI/UX design
In-Reply-To: <009901d69e3f$3943a0f0$abcae2d0$@gmail.com>
References: <CAAtWTekEPFw4AxxigLfd0d9nzMrW+1-WFFT5LUvd3u2Fo_VNYg@mail.gmail.com>
 <009901d69e3f$3943a0f0$abcae2d0$@gmail.com>
Message-ID: <em36deb13a-aa38-4bb3-a5e6-f03845a08ea2@fred-pc>


Generally speaking:

- The more options in a menu the more chance your caller will need to 
have a menu repeated;
- Always tell most used options first so that the caller doesn't stay 
long in the IVR;
- Use 3x3 or 4x4 matrix - e.g. 3 levels deep w/3 options each at a 
maximum;
- Always provide a key to repeat the menu and optimize the waiting time 
for key input so that callers have enough time;
- Use the same voice across the menu options;
- Optimize each sentence - they must be short but very informative;
- Be consistent; e.g. same way of telling the options across the menus;
- Hang up after 3 wrong options - this also avoids lingering calls

On 2020-10-09 09:22:22, "Bote Man" <botelist at gmail.com> wrote:

I don’t know the answer, but there’s a good possibility that a fine lady 
who records professional IVR prompts knows where to point you:



Allison at theivrvoice.com





---

John Boteler

BnC Group U.S.A.









From: Sam van Herwaarden
Sent: Friday, 9 October, 2020 03:44
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] IVR UI/UX design



Hi all,



My organization runs a number of IVR systems for information delivery, 
and one issue we often run into is that users find the interface a bit 
awkward or hard to navigate. We have experimented quite a bit internally 
with adjusting the structures of menus, ordering, phrasing etc. But we 
feel like there must be some knowledge/experience out there and that it 
shouldn't be necessary to rediscover most of this ourselves.



Does anyone have recommendations for resources with info on best 
practices for IVR design? Or contacts in organizations/companies that 
have a lot of experience with this?



Kind regards,

Sam


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From makafre at gmail.com  Fri Oct  9 16:04:10 2020
From: makafre at gmail.com (Frederic Jean)
Date: Fri, 09 Oct 2020 16:04:10 +0000
Subject: [Freeswitch-users] Defunct processes from starting a script
In-Reply-To: <002601d69d92$ccec1ba0$66c452e0$@gmail.com>
References: <em44b00752-a749-4146-ac68-09be1d032b59@fred-pc>
 <002601d69d92$ccec1ba0$66c452e0$@gmail.com>
Message-ID: <emc2f9886a-0fa7-4322-b1bb-a6b27efb1a66@fred-pc>


Thank you very much for the insight John.

- Fred

On 2020-10-08 12:48:07, "Bote Man" <botelist at gmail.com> wrote:

I do not know what your script does, and that determines the best 
approach to solve the problem.



In the only case with which I have experience, I run a perl script as a 
stand-alone linux process using the normal perl interpreter. This script 
then connects to FreeSWITCH via ESL (the Event Socket Library) and 
listens for certain FreeSWITCH events, then reacts to them. This removes 
the script from the telephony processing loop in the dialplan and 
eliminates any possible blocking of telephony processing in the dialplan 
(which seems to be a common problem with many script implementations 
when called from the dialplan).



This method requires compiling the ESL.pm module which is part of the FS 
source; this module provides the connection from perl to FS. Install it 
using these instructions:



https://freeswitch.org/confluence/display/FREESWITCH/Event+Socket+Library#EventSocketLibrary-Installation



There are perl examples on this wiki page:

https://freeswitch.org/confluence/display/FREESWITCH/Perl+ESL



That’s not much, but I hope it helps.





---

John Boteler

BnC Group U.S.A.









From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> 
On Behalf Of Frederic Jean
Sent: Thursday, 8 October, 2020 10:38
To:freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Defunct processes from starting a script





Dear community,



Newbie here; I am currently experiencing aleatory defunct processes by 
starting a Perl script this way:



<action application="set" data= 
"ivr_path=/usr/local/freeswitch/scripts/myscript.pl"/>

<action application="socket" data="127.0.0.1:9080 sync full"/>



Are there any better ways to have it executed and avoid these?



Regards

Fred


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From davidswalkabout at gmail.com  Fri Oct  9 20:43:51 2020
From: davidswalkabout at gmail.com (David P)
Date: Sat, 10 Oct 2020 09:43:51 +1300
Subject: [Freeswitch-users] RECORD_STEREO and record_sample_rate aren't
	working with conference
In-Reply-To: <mailman.1868.1602202055.1288.freeswitch-users@lists.freeswitch.org>
References: <mailman.1868.1602202055.1288.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAE23trtg1kCoy1FEVapAWQqn9xrK3X+0JT5JZDNA8cuBpSd1cg@mail.gmail.com>

Sure, I filed https://github.com/signalwire/freeswitch/issues/895

In the meantime, we're urgently looking for a workaround, and we're
considering introducing a 2nd FS between our current FS and the peer
destination (Asterisk). We'd remove transcoding from the conference config
of the 1st FS.

In the 1st FS we need to change the sofia gateway path from this:

<action application="conference_set_auto_outcall"

data="['conference_member_flags=endconf,jitterbuffer_msec=5p:100p']sofia/gateway/asterisk/${destination_number}"/>

and I'm guessing the path should be " sofia/gateway/freeswitch/" (with a
corresponding sip_profiles/internal/freeswitch.xml) But googling "sofia
gateway freeswitch" doesn't reveal any promising matches.

Is that right?

From: Brian West <brian at freeswitch.com>
>
> Yah that's probably true, you should file issues on github please.
>
> On Thu, Oct 8, 2020 at 5:59 PM David P <davidswalkabout at gmail.com> wrote:
>
>> We're using FSv10.5 on Debian 10 with verto/Opus (i.e. no codec
>> explicitly configured), and we're using a conference. We've tried these
>> settings but they're having no effect on the recorded mp4s:
>>
>> <action application="set" data="RECORD_STEREO=true"/>
>> <action application="set" data="record_sample_rate=44100"/>
>>
>> Instead:
>> 1) The mp4s have the same audio in both channels, contrary to
>> https://freeswitch.org/confluence/display/FREESWITCH/RECORD_STEREO
>>
>> 2) The sampling rate is 8kHz, contrary to
>> https://freeswitch.org/confluence/display/FREESWITCH/record_sample_rate
>>
>> FWIW:
>> a) We noticed that mp4s from a year ago when we were using FS 1.8 also
>> have the same audio on both channels.
>>
>> b) We also have this, and I don't recall why and can't find a description
>> in confluence:
>> <action application="set" data="record_concat_video=true"/>
>>
>> Is it that these are known not to work with conferences, or is there some
>> setting we might have overlooked?
>>
>
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From harshita19012 at iiitd.ac.in  Sat Oct 10 05:06:22 2020
From: harshita19012 at iiitd.ac.in (Harshita Pandey)
Date: Sat, 10 Oct 2020 10:36:22 +0530
Subject: [Freeswitch-users] Port Required to communicate with Gateway
In-Reply-To: <CA+by-=WB_zzT1NMS6-AKFAjjNs3k1Hv5GYaRqOi3_MbyTCoO7A@mail.gmail.com>
References: <CA+by-=WB_zzT1NMS6-AKFAjjNs3k1Hv5GYaRqOi3_MbyTCoO7A@mail.gmail.com>
Message-ID: <CA+by-=Xf9OS+YW3DXL_Rt2tw95UZc-piP177dHpmOQXYsL1Qzw@mail.gmail.com>

I made some changes and now when I run sofia status, it shows either
FAIL_WAIT or sometimes it shows UNREGED. What may be the reasons for these ?

On Thu, Oct 8, 2020 at 4:31 PM Harshita Pandey <harshita19012 at iiitd.ac.in>
wrote:

> Hi,
>
> I have installed freeswitch on a Virtual Machine, and I am not able to
> configure the gateway for some reason. Can someone tell me which ports are
> required and need to be opened on the virtual machine to communicate with
> Gateway in freeswitch?
> Since all the ports are not opened in the Virtual Machine, this could help
> me open up those ports and configure Gateway for freeswitch
>
>
> Much help is needed.
> It's Urgent
> Thanks in advance.
>
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From botelist at gmail.com  Sat Oct 10 06:30:11 2020
From: botelist at gmail.com (Bote Man)
Date: Sat, 10 Oct 2020 02:30:11 -0400
Subject: [Freeswitch-users] RECORD_STEREO and record_sample_rate
	aren't	working with conference
In-Reply-To: <CAE23trtg1kCoy1FEVapAWQqn9xrK3X+0JT5JZDNA8cuBpSd1cg@mail.gmail.com>
References: <mailman.1868.1602202055.1288.freeswitch-users@lists.freeswitch.org>
 <CAE23trtg1kCoy1FEVapAWQqn9xrK3X+0JT5JZDNA8cuBpSd1cg@mail.gmail.com>
Message-ID: <00eb01d69ece$cf7262c0$6e572840$@gmail.com>

You create as many SIP profiles or gateways as you need. You can name the files anything you wish because the important bits are contained inside the files, typically near the top. 

 

The typical profile in the Vanilla config files contains these lines:

 

  <gateways>

    <X-PRE-PROCESS cmd="include" data="internal/*.xml"/>

  </gateways> 

 

which tells FS to scan the “internal” subdirectory for all XML files. Inside one of those would be your Asterisk file that contains a line like:

 

<gateway name="asterisk"> 

 

which appears in your current dial string. That string in the gateway name= is what tells FS which gateway to use when you specify it in the dial string.

 

Since your Asterisk SIP profile works, just copy that, set the owner and permissions appropriately, and edit the contents to point to your 2nd FreeSWITCH box IP and port number and whatever other parameters need to change.

 

 

--- 

John Boteler 

BnC Group U.S.A. 

 

 

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of David P
Sent: Friday, 9 October, 2020 16:44
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] RECORD_STEREO and record_sample_rate aren't working with conference

 

Sure, I filed https://github.com/signalwire/freeswitch/issues/895

 

In the meantime, we're urgently looking for a workaround, and we're considering introducing a 2nd FS between our current FS and the peer destination (Asterisk). We'd remove transcoding from the conference config of the 1st FS.

 

In the 1st FS we need to change the sofia gateway path from this:

 

<action application="conference_set_auto_outcall"
        data="['conference_member_flags=endconf,jitterbuffer_msec=5p:100p']sofia/gateway/asterisk/${destination_number}"/>

 

and I'm guessing the path should be " sofia/gateway/freeswitch/" (with a corresponding sip_profiles/internal/freeswitch.xml) But googling "sofia gateway freeswitch" doesn't reveal any promising matches.

 

Is that right?

 

From: Brian West <brian at freeswitch.com <mailto:brian at freeswitch.com> >

Yah that's probably true, you should file issues on github please.

 

On Thu, Oct 8, 2020 at 5:59 PM David P <davidswalkabout at gmail.com <mailto:davidswalkabout at gmail.com> > wrote:

We're using FSv10.5 on Debian 10 with verto/Opus (i.e. no codec explicitly configured), and we're using a conference. We've tried these settings but they're having no effect on the recorded mp4s:

 

<action application="set" data="RECORD_STEREO=true"/>
<action application="set" data="record_sample_rate=44100"/>

Instead:

1) The mp4s have the same audio in both channels, contrary to https://freeswitch.org/confluence/display/FREESWITCH/RECORD_STEREO

 

2) The sampling rate is 8kHz, contrary to https://freeswitch.org/confluence/display/FREESWITCH/record_sample_rate

FWIW:

a) We noticed that mp4s from a year ago when we were using FS 1.8 also have the same audio on both channels.

 

b) We also have this, and I don't recall why and can't find a description in confluence:
<action application="set" data="record_concat_video=true"/>

 

Is it that these are known not to work with conferences, or is there some setting we might have overlooked?

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From botelist at gmail.com  Sat Oct 10 06:30:11 2020
From: botelist at gmail.com (Bote Man)
Date: Sat, 10 Oct 2020 02:30:11 -0400
Subject: [Freeswitch-users] Port Required to communicate with Gateway
In-Reply-To: <CA+by-=Xf9OS+YW3DXL_Rt2tw95UZc-piP177dHpmOQXYsL1Qzw@mail.gmail.com>
References: <CA+by-=WB_zzT1NMS6-AKFAjjNs3k1Hv5GYaRqOi3_MbyTCoO7A@mail.gmail.com>
 <CA+by-=Xf9OS+YW3DXL_Rt2tw95UZc-piP177dHpmOQXYsL1Qzw@mail.gmail.com>
Message-ID: <00e601d69ece$ceccd3f0$6c667bd0$@gmail.com>

Something is not working, but it’s impossible to say what without more specific information.

 

I’m guessing that UNREGED means that you have configured a gateway to register with some other switch or service, but FreeSWITCH can not reach it for some reason.

 

Bote

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Harshita Pandey
Sent: Saturday, 10 October, 2020 01:06
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Port Required to communicate with Gateway

 

I made some changes and now when I run sofia status, it shows either FAIL_WAIT or sometimes it shows UNREGED. What may be the reasons for these ?

 

On Thu, Oct 8, 2020 at 4:31 PM Harshita Pandey <harshita19012 at iiitd.ac.in <mailto:harshita19012 at iiitd.ac.in> > wrote:

Hi,

 

I have installed freeswitch on a Virtual Machine, and I am not able to configure the gateway for some reason. Can someone tell me which ports are required and need to be opened on the virtual machine to communicate with Gateway in freeswitch? 

Since all the ports are not opened in the Virtual Machine, this could help me open up those ports and configure Gateway for freeswitch

 

 

Much help is needed. 

It's Urgent

Thanks in advance.

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From ziv at ip-com.co.il  Sun Oct 11 10:27:12 2020
From: ziv at ip-com.co.il (ziv at ip-com.co.il)
Date: Sun, 11 Oct 2020 13:27:12 +0300
Subject: [Freeswitch-users] Freeswitch CPU consumption on AWS EC2
Message-ID: <043801d69fb9$157ce0c0$4076a240$@ip-com.co.il>

Hi,

  

We run FreeSWITCH on Amazon AWS EC2, using the c5.xlarge machine size, meaning 4 CPUs with hyper-threading, sums up to 8 vCpu.

In time of high-traffic, the system is reaching 100% CPU at about 400 concurrent sessions, which is typically 200 calls (2 sessions per call).

RTP is sent as G-711. SIP signaling is this test is UDP (no TCP or other protocols).

These numbers are after reducing the log level and moving the core-dsn to memory.

The dialplan in this test is very simple.

ulimit values are as recommended in FreeSWITCH documentation.

On thread level, we see many threads that consume <1.5% CPU. Several threads consume more, apparently the signaling.

  

Question:

Do these numbers make sense? We expected to see much more traffic before the CPU is congested.

Are there simple tricks to reduce the CPU usage (as we did with changing the log level)

Is there a way to analyze the CPU consumption so we can try and improve?

  

Thanks

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From rangeare.01 at gmail.com  Mon Oct 12 06:38:28 2020
From: rangeare.01 at gmail.com (Jay Desai)
Date: Mon, 12 Oct 2020 12:08:28 +0530
Subject: [Freeswitch-users] Port Required to communicate with Gateway
In-Reply-To: <CA+by-=Xf9OS+YW3DXL_Rt2tw95UZc-piP177dHpmOQXYsL1Qzw@mail.gmail.com>
References: <CA+by-=WB_zzT1NMS6-AKFAjjNs3k1Hv5GYaRqOi3_MbyTCoO7A@mail.gmail.com>
 <CA+by-=Xf9OS+YW3DXL_Rt2tw95UZc-piP177dHpmOQXYsL1Qzw@mail.gmail.com>
Message-ID: <CAGvKJUUra2Kjkmygh5137tkOOjioSnfeMWt9kEaj+UD3+=JZqA@mail.gmail.com>

Can you give remote access?

On Sat, Oct 10, 2020 at 10:55 AM Harshita Pandey <harshita19012 at iiitd.ac.in>
wrote:

> I made some changes and now when I run sofia status, it shows either
> FAIL_WAIT or sometimes it shows UNREGED. What may be the reasons for these ?
>
> On Thu, Oct 8, 2020 at 4:31 PM Harshita Pandey <harshita19012 at iiitd.ac.in>
> wrote:
>
>> Hi,
>>
>> I have installed freeswitch on a Virtual Machine, and I am not able to
>> configure the gateway for some reason. Can someone tell me which ports are
>> required and need to be opened on the virtual machine to communicate with
>> Gateway in freeswitch?
>> Since all the ports are not opened in the Virtual Machine, this could
>> help me open up those ports and configure Gateway for freeswitch
>>
>>
>> Much help is needed.
>> It's Urgent
>> Thanks in advance.
>>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From s.safarov at gmail.com  Mon Oct 12 07:30:06 2020
From: s.safarov at gmail.com (Sergey Safarov)
Date: Mon, 12 Oct 2020 10:30:06 +0300
Subject: [Freeswitch-users] Freeswitch CPU consumption on AWS EC2
In-Reply-To: <043801d69fb9$157ce0c0$4076a240$@ip-com.co.il>
References: <043801d69fb9$157ce0c0$4076a240$@ip-com.co.il>
Message-ID: <CAHtxdDdK=0skEn5R0Ko7cWy=XA0B1cmQqM8TEUy7FENtCnY_bg@mail.gmail.com>

my test results using sipp script +  FreeSWITCH docker container + PCMU
on c5.4xlarge instance only audio stream. Each call duration is 200 seconds.
Tested Kazoo all-in-one install. So, here some overhead exit for Elang
components.

According to load test on c5.4xlarge Amazon instance after 282 seconds is
reached 1258 simultaneous calls at 6 calls per second rate. During the test
all calls and registrations are successful. During the test CPU load is
reached 79% value, memory usage 7Gb, and 150 mb/s total network usage over
all docker containers on "all-in-one" Kazoo configuration.

Sergey

On Sun, Oct 11, 2020 at 1:51 PM <ziv at ip-com.co.il> wrote:

> Hi,
>
>
>
> We run FreeSWITCH on Amazon AWS EC2, using the c5.xlarge machine size,
> meaning 4 CPUs with hyper-threading, sums up to 8 vCpu.
>
> In time of high-traffic, the system is reaching 100% CPU at about 400
> concurrent sessions, which is typically 200 calls (2 sessions per call).
>
> RTP is sent as G-711. SIP signaling is this test is UDP (no TCP or other
> protocols).
>
> These numbers are after reducing the log level and moving the core-dsn to
> memory.
>
> The dialplan in this test is very simple.
>
> ulimit values are as recommended in FreeSWITCH documentation.
>
> On thread level, we see many threads that consume <1.5% CPU. Several
> threads consume more, apparently the signaling.
>
>
>
> Question:
>
> Do these numbers make sense? We expected to see much more traffic before
> the CPU is congested.
>
> Are there simple tricks to reduce the CPU usage (as we did with changing
> the log level)
>
> Is there a way to analyze the CPU consumption so we can try and improve?
>
>
>
> Thanks
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From ziv at ip-com.co.il  Mon Oct 12 08:34:29 2020
From: ziv at ip-com.co.il (ziv at ip-com.co.il)
Date: Mon, 12 Oct 2020 11:34:29 +0300
Subject: [Freeswitch-users] Freeswitch CPU consumption on AWS EC2
In-Reply-To: <CAHtxdDdK=0skEn5R0Ko7cWy=XA0B1cmQqM8TEUy7FENtCnY_bg@mail.gmail.com>
References: <043801d69fb9$157ce0c0$4076a240$@ip-com.co.il>
 <CAHtxdDdK=0skEn5R0Ko7cWy=XA0B1cmQqM8TEUy7FENtCnY_bg@mail.gmail.com>
Message-ID: <058501d6a072$81228f50$8367adf0$@ip-com.co.il>

Assuming it’s 1258 one-session calls, and remembering that c5.4xlarge has 4 times more CPU than c5.xlarge, and taking into account that my test was on 30 seconds call vs. 200 seconds call in your test, it seems that our findings are similar.

This makes the FreeSWITCH deployment costly. There must be a way to reduce CPU consumption.

 �

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Sergey Safarov
Sent: Monday, October 12, 2020 10:30 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Freeswitch CPU consumption on AWS EC2

 �

my test results using sipp script �+  �FreeSWITCH docker container �+ PCMU on �c5.4xlarge instance only audio stream. Each call duration is 200 seconds.

Tested Kazoo all-in-one install. So, here some overhead exit for Elang components.

According to load test on c5.4xlarge Amazon instance after 282 seconds is reached 1258 simultaneous calls at 6 calls per second rate. During the test all calls and registrations are successful. During the test CPU load is reached 79% value, memory usage 7Gb, and 150 mb/s total network usage over all docker containers on "all-in-one" Kazoo configuration.

 �

Sergey

 �

On Sun, Oct 11, 2020 at 1:51 PM <ziv at ip-com.co.il <mailto:ziv at ip-com.co.il> > wrote:

Hi,

 �

We run FreeSWITCH on Amazon AWS EC2, using the c5.xlarge machine size, meaning 4 CPUs with hyper-threading, sums up to 8 vCpu.

In time of high-traffic, the system is reaching 100% CPU at about 400 concurrent sessions, which is typically 200 calls (2 sessions per call).

RTP is sent as G-711. SIP signaling is this test is UDP (no TCP or other protocols).

These numbers are after reducing the log level and moving the core-dsn to memory.

The dialplan in this test is very simple.

ulimit values are as recommended in FreeSWITCH documentation.

On thread level, we see many threads that consume <1.5% CPU. Several threads consume more, apparently the signaling.

 �

Question:

Do these numbers make sense? We expected to see much more traffic before the CPU is congested.

Are there simple tricks to reduce the CPU usage (as we did with changing the log level)

Is there a way to analyze the CPU consumption so we can try and improve?

 �

Thanks

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales at freeswitch.com <mailto:sales at freeswitch.com> 
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org> 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com

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From s.safarov at gmail.com  Mon Oct 12 10:48:17 2020
From: s.safarov at gmail.com (Sergey Safarov)
Date: Mon, 12 Oct 2020 13:48:17 +0300
Subject: [Freeswitch-users] Freeswitch CPU consumption on AWS EC2
In-Reply-To: <058501d6a072$81228f50$8367adf0$@ip-com.co.il>
References: <043801d69fb9$157ce0c0$4076a240$@ip-com.co.il>
 <CAHtxdDdK=0skEn5R0Ko7cWy=XA0B1cmQqM8TEUy7FENtCnY_bg@mail.gmail.com>
 <058501d6a072$81228f50$8367adf0$@ip-com.co.il>
Message-ID: <CAHtxdDcZ5W_K3oDUKECRGHZUKvfm1+BZj=gsztxQvjZL23pK3Q@mail.gmail.com>

You can double decrease AWS cost by using "reserved" instances or "spot"
instances.
Also, important call duration. Think reals calls longer, I think about 1-2
mins average.
I expect, if you increase call duration two times (decrease CPS two times),
then you will increase the number of calls.

Also important to evaluate how complex your dialplan.
Is used codec conversion or not.

All of this may impact total calls handled by FreeSwitch.

Sergey

On Mon, Oct 12, 2020 at 12:00 PM <ziv at ip-com.co.il> wrote:

> Assuming it’s 1258 one-session calls, and remembering that c5.4xlarge has
> 4 times more CPU than c5.xlarge, and taking into account that my test was
> on 30 seconds call vs. 200 seconds call in your test, it seems that our
> findings are similar.
>
> This makes the FreeSWITCH deployment costly. There must be a way to reduce
> CPU consumption.
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Sergey Safarov
> *Sent:* Monday, October 12, 2020 10:30 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Freeswitch CPU consumption on AWS EC2
>
>
>
> my test results using sipp script +  FreeSWITCH docker container + PCMU
> on c5.4xlarge instance only audio stream. Each call duration is 200 seconds.
>
> Tested Kazoo all-in-one install. So, here some overhead exit for Elang
> components.
>
> According to load test on c5.4xlarge Amazon instance after 282 seconds is
> reached 1258 simultaneous calls at 6 calls per second rate. During the test
> all calls and registrations are successful. During the test CPU load is
> reached 79% value, memory usage 7Gb, and 150 mb/s total network usage over
> all docker containers on "all-in-one" Kazoo configuration.
>
>
>
> Sergey
>
>
>
> On Sun, Oct 11, 2020 at 1:51 PM <ziv at ip-com.co.il> wrote:
>
> Hi,
>
>
>
> We run FreeSWITCH on Amazon AWS EC2, using the c5.xlarge machine size,
> meaning 4 CPUs with hyper-threading, sums up to 8 vCpu.
>
> In time of high-traffic, the system is reaching 100% CPU at about 400
> concurrent sessions, which is typically 200 calls (2 sessions per call).
>
> RTP is sent as G-711. SIP signaling is this test is UDP (no TCP or other
> protocols).
>
> These numbers are after reducing the log level and moving the core-dsn to
> memory.
>
> The dialplan in this test is very simple.
>
> ulimit values are as recommended in FreeSWITCH documentation.
>
> On thread level, we see many threads that consume <1.5% CPU. Several
> threads consume more, apparently the signaling.
>
>
>
> Question:
>
> Do these numbers make sense? We expected to see much more traffic before
> the CPU is congested.
>
> Are there simple tricks to reduce the CPU usage (as we did with changing
> the log level)
>
> Is there a way to analyze the CPU consumption so we can try and improve?
>
>
>
> Thanks
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From sdevoy at bizfocused.com  Mon Oct 12 21:50:37 2020
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Mon, 12 Oct 2020 21:50:37 +0000
Subject: [Freeswitch-users] SUBSCRIBE -  Bad  Event
Message-ID: <BL0PR0102MB34258901D7CA3942AFE79AD7BB070@BL0PR0102MB3425.prod.exchangelabs.com>

Hi,

I am getting 489 Bad Event for my users "SUBSCRIBE" packets. SUBSCRIBE is clearly listed in ALLOW. 2 Sample packets below.

Any ideas?

Thanks,
Sean

recv 524 bytes from tcp/[73.555.94.74]:1024 at 17:52:51.791847:
------------------------------------------------------------------------
SUBSCRIBE sip:214 at clientdomain.com<sip:214 at fs-esta2.bizfocused.com> SIP/2.0
Via: SIP/2.0/TCP 192.168.2.124:5075;branch=z9hG4bK-1d541664;rport
From: "214" <sip:214 at clientdomain.com<sip:214 at fs-esta2.bizfocused.com>>;tag=59cf9549d7610470
To: "214" <sip:214 at clientdomain.com<sip:214 at fs-esta2.bizfocused.com>>
Call-ID: b36ae378-cf9d3789 at 192.168.2.124<mailto:b36ae378-cf9d3789 at 192.168.2.124>
CSeq: 1001 SUBSCRIBE
Max-Forwards: 70
Contact: "214" <sip:214 at 192.168.2.124:5075;transport=tcp>
Accept: application/x-broadworks-callpark-info+xml
Expires: 1800
Event: x-broadworks-callpark
User-Agent: Cisco/SPA504G-7.6.2e
Content-Length: 0

send 717 bytes to tcp/[73.555.94.74]:1024 at 17:52:51.791962:
------------------------------------------------------------------------
SIP/2.0 489 Bad Event
Via: SIP/2.0/TCP 192.168.2.124:5075;branch=z9hG4bK-1d541664;rport=1024;received=73.555.94.74
From: "214" <sip:214 at clientdomain.com<sip:214 at fs-esta2.bizfocused.com>>;tag=59cf9549d7610470
To: "214" <sip:214 at clientdomain.com<sip:214 at fs-esta2.bizfocused.com>>;tag=279QBZZaKS7cr
Call-ID: b36ae378-cf9d3789 at 192.168.2.124<mailto:b36ae378-cf9d3789 at 192.168.2.124>
CSeq: 1001 SUBSCRIBE
User-Agent: FreeSWITCH-mod_sofia/1.10.2-release-14-f7bdd3845a~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0
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From shaun at sysconfig.cloud  Tue Oct 13 08:03:26 2020
From: shaun at sysconfig.cloud (Shaun Stokes)
Date: Tue, 13 Oct 2020 08:03:26 +0000
Subject: [Freeswitch-users] Freeswitch CPU consumption on AWS EC2
In-Reply-To: <058501d6a072$81228f50$8367adf0$@ip-com.co.il>
References: <043801d69fb9$157ce0c0$4076a240$@ip-com.co.il>
 <CAHtxdDdK=0skEn5R0Ko7cWy=XA0B1cmQqM8TEUy7FENtCnY_bg@mail.gmail.com>,
 <058501d6a072$81228f50$8367adf0$@ip-com.co.il>
Message-ID: <CWLP123MB23716ABC179D34D3B322AD35B7040@CWLP123MB2371.GBRP123.PROD.OUTLOOK.COM>

Are you using the call centre module? If so, we've ran into issues with the FS SQLite thread (single child process) becoming CPU bound, you can fix it by using an RDBMS such as PGSQL which is multi-threaded.

________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of ziv at ip-com.co.il <ziv at ip-com.co.il>
Sent: 12 October 2020 10:34
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Freeswitch CPU consumption on AWS EC2


Assuming it’s 1258 one-session calls, and remembering that c5.4xlarge has 4 times more CPU than c5.xlarge, and taking into account that my test was on 30 seconds call vs. 200 seconds call in your test, it seems that our findings are similar.

This makes the FreeSWITCH deployment costly. There must be a way to reduce CPU consumption.



From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Sergey Safarov
Sent: Monday, October 12, 2020 10:30 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Freeswitch CPU consumption on AWS EC2



my test results using sipp script +  FreeSWITCH docker container + PCMU on c5.4xlarge instance only audio stream. Each call duration is 200 seconds.

Tested Kazoo all-in-one install. So, here some overhead exit for Elang components.

According to load test on c5.4xlarge Amazon instance after 282 seconds is reached 1258 simultaneous calls at 6 calls per second rate. During the test all calls and registrations are successful. During the test CPU load is reached 79% value, memory usage 7Gb, and 150 mb/s total network usage over all docker containers on "all-in-one" Kazoo configuration.



Sergey



On Sun, Oct 11, 2020 at 1:51 PM <ziv at ip-com.co.il<mailto:ziv at ip-com.co.il>> wrote:

Hi,



We run FreeSWITCH on Amazon AWS EC2, using the c5.xlarge machine size, meaning 4 CPUs with hyper-threading, sums up to 8 vCpu.

In time of high-traffic, the system is reaching 100% CPU at about 400 concurrent sessions, which is typically 200 calls (2 sessions per call).

RTP is sent as G-711. SIP signaling is this test is UDP (no TCP or other protocols).

These numbers are after reducing the log level and moving the core-dsn to memory.

The dialplan in this test is very simple.

ulimit values are as recommended in FreeSWITCH documentation.

On thread level, we see many threads that consume <1.5% CPU. Several threads consume more, apparently the signaling.



Question:

Do these numbers make sense? We expected to see much more traffic before the CPU is congested.

Are there simple tricks to reduce the CPU usage (as we did with changing the log level)

Is there a way to analyze the CPU consumption so we can try and improve?



Thanks

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales at freeswitch.com<mailto:sales at freeswitch.com>
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
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From botelist at gmail.com  Tue Oct 13 08:23:48 2020
From: botelist at gmail.com (Bote Man)
Date: Tue, 13 Oct 2020 04:23:48 -0400
Subject: [Freeswitch-users] SUBSCRIBE -  Bad  Event
In-Reply-To: <BL0PR0102MB34258901D7CA3942AFE79AD7BB070@BL0PR0102MB3425.prod.exchangelabs.com>
References: <BL0PR0102MB34258901D7CA3942AFE79AD7BB070@BL0PR0102MB3425.prod.exchangelabs.com>
Message-ID: <00e901d6a13a$2d1d5340$8757f9c0$@gmail.com>

I had that with some Poly(com) VVX501 phones. 

 

The solution was to set every occurrence of .regevent="0" so
reg.1.regevent="0", reg.2.regevent="0", and so on. 

 

>From their docs:

voIpProt.SIP.regevent 

0 (default) - The phone is not subscribed to registration state change
notifications for all phone lines.
1 - The phone is subscribed to registration state change notifications for
all phone lines.

This parameter is overridden by the per-phone parameter reg.x.regevent. 

I did not troubleshoot it, I simply set it to 0 and moved on. I have no idea
what the equivalent would be on a Cisco phone, but if you see a related
setting maybe it's worth toggling it to see what the result is.

 

Hope this helps.

 

--- 

John Boteler 

BnC Group U.S.A. 

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On
Behalf Of Sean Devoy
Sent: Monday, 12 October, 2020 17:51
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] SUBSCRIBE - Bad Event

 

Hi,

 

I am getting 489 Bad Event for my users "SUBSCRIBE" packets. SUBSCRIBE is
clearly listed in ALLOW. 2 Sample packets below.

 

Any ideas?

 

Thanks,

Sean

 

recv 524 bytes from tcp/[73.555.94.74]:1024 at 17:52:51.791847:

------------------------------------------------------------------------

SUBSCRIBE sip:214 at clientdomain.com <sip:214 at fs-esta2.bizfocused.com>
SIP/2.0

Via: SIP/2.0/TCP 192.168.2.124:5075;branch=z9hG4bK-1d541664;rport

From: "214" <sip:214 at clientdomain.com <sip:214 at fs-esta2.bizfocused.com>
>;tag=59cf9549d7610470

To: "214" <sip:214 at clientdomain.com <sip:214 at fs-esta2.bizfocused.com> >

Call-ID: b36ae378-cf9d3789 at 192.168.2.124
<mailto:b36ae378-cf9d3789 at 192.168.2.124> 

CSeq: 1001 SUBSCRIBE

Max-Forwards: 70

Contact: "214" <sip:214 at 192.168.2.124:5075;transport=tcp>

Accept: application/x-broadworks-callpark-info+xml

Expires: 1800

Event: x-broadworks-callpark

User-Agent: Cisco/SPA504G-7.6.2e

Content-Length: 0

 

send 717 bytes to tcp/[73.555.94.74]:1024 at 17:52:51.791962:

------------------------------------------------------------------------

SIP/2.0 489 Bad Event

Via: SIP/2.0/TCP
192.168.2.124:5075;branch=z9hG4bK-1d541664;rport=1024;received=73.555.94.74

From: "214" <sip:214 at clientdomain.com <sip:214 at fs-esta2.bizfocused.com>
>;tag=59cf9549d7610470

To: "214" <sip:214 at clientdomain.com <sip:214 at fs-esta2.bizfocused.com>
>;tag=279QBZZaKS7cr

Call-ID: b36ae378-cf9d3789 at 192.168.2.124
<mailto:b36ae378-cf9d3789 at 192.168.2.124> 

CSeq: 1001 SUBSCRIBE

User-Agent: FreeSWITCH-mod_sofia/1.10.2-release-14-f7bdd3845a~64bit

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE

Supported: timer, path, replaces

Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer

Content-Length: 0

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From abalashov at evaristesys.com  Tue Oct 13 02:09:27 2020
From: abalashov at evaristesys.com (Alex Balashov)
Date: Mon, 12 Oct 2020 22:09:27 -0400
Subject: [Freeswitch-users] SUBSCRIBE - Bad Event
In-Reply-To: <BL0PR0102MB34258901D7CA3942AFE79AD7BB070@BL0PR0102MB3425.prod.exchangelabs.com>
References: <BL0PR0102MB34258901D7CA3942AFE79AD7BB070@BL0PR0102MB3425.prod.exchangelabs.com>
Message-ID: <1787e69a-5d60-1ca8-29f6-edfcb85443f7@evaristesys.com>

Yeah, but it's the event package itself that matters:

    Event: x-broadworks-callpark

FS doesn't understand it, which is little surprise given that it seems 
proprietary. It does not appear to be IANA-registered.

-- Alex

-- 
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/


From martin.paterson at technologywithin.com  Tue Oct 13 08:54:40 2020
From: martin.paterson at technologywithin.com (Martin Paterson)
Date: Tue, 13 Oct 2020 08:54:40 +0000
Subject: [Freeswitch-users] SUBSCRIBE -  Bad  Event
In-Reply-To: <BL0PR0102MB34258901D7CA3942AFE79AD7BB070@BL0PR0102MB3425.prod.exchangelabs.com>
References: <BL0PR0102MB34258901D7CA3942AFE79AD7BB070@BL0PR0102MB3425.prod.exchangelabs.com>
Message-ID: <DB7PR10MB2377D1E48A043C178523F71695040@DB7PR10MB2377.EURPRD10.PROD.OUTLOOK.COM>

The event you're subscribing to is in the header:
Event: x-broadworks-callpark

That's related to the Cisco BroadWorks product. FreeSWITCH won't know about those events.

Martin.



Martin Paterson
Development Team
Phone: 0207 953 8840
Email: martin.paterson at technologywithin.com
Chevron Business Park, Limekiln Lane, Southampton, Hampshire, SO45 2QL
Registered Office: CP House, Otterspool Way, Watford, WD25 8JJ, U.K 
​Registered in England No: 5964349 | VAT Number: GB 902 5369 37 
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Sean Devoy
Sent: 12 October 2020 22:51
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] SUBSCRIBE - Bad Event

Hi,

I am getting 489 Bad Event for my users "SUBSCRIBE" packets. SUBSCRIBE is clearly listed in ALLOW. 2 Sample packets below.

Any ideas?

Thanks,
Sean

recv 524 bytes from tcp/[73.555.94.74]:1024 at 17:52:51.791847:
------------------------------------------------------------------------
SUBSCRIBE sip:214 at clientdomain.com<sip:214 at fs-esta2.bizfocused.com> SIP/2.0
Via: SIP/2.0/TCP 192.168.2.124:5075;branch=z9hG4bK-1d541664;rport
From: "214" <sip:214 at clientdomain.com<sip:214 at fs-esta2.bizfocused.com>>;tag=59cf9549d7610470
To: "214" <sip:214 at clientdomain.com<sip:214 at fs-esta2.bizfocused.com>>
Call-ID: b36ae378-cf9d3789 at 192.168.2.124<mailto:b36ae378-cf9d3789 at 192.168.2.124>
CSeq: 1001 SUBSCRIBE
Max-Forwards: 70
Contact: "214" <sip:214 at 192.168.2.124:5075;transport=tcp>
Accept: application/x-broadworks-callpark-info+xml
Expires: 1800
Event: x-broadworks-callpark
User-Agent: Cisco/SPA504G-7.6.2e
Content-Length: 0

send 717 bytes to tcp/[73.555.94.74]:1024 at 17:52:51.791962:
------------------------------------------------------------------------
SIP/2.0 489 Bad Event
Via: SIP/2.0/TCP 192.168.2.124:5075;branch=z9hG4bK-1d541664;rport=1024;received=73.555.94.74
From: "214" <sip:214 at clientdomain.com<sip:214 at fs-esta2.bizfocused.com>>;tag=59cf9549d7610470
To: "214" <sip:214 at clientdomain.com<sip:214 at fs-esta2.bizfocused.com>>;tag=279QBZZaKS7cr
Call-ID: b36ae378-cf9d3789 at 192.168.2.124<mailto:b36ae378-cf9d3789 at 192.168.2.124>
CSeq: 1001 SUBSCRIBE
User-Agent: FreeSWITCH-mod_sofia/1.10.2-release-14-f7bdd3845a~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0
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From sdevoy at bizfocused.com  Tue Oct 13 15:33:43 2020
From: sdevoy at bizfocused.com (Sean Devoy)
Date: Tue, 13 Oct 2020 15:33:43 +0000
Subject: [Freeswitch-users] SUBSCRIBE -  Bad  Event
In-Reply-To: <DB7PR10MB2377D1E48A043C178523F71695040@DB7PR10MB2377.EURPRD10.PROD.OUTLOOK.COM>
References: <BL0PR0102MB34258901D7CA3942AFE79AD7BB070@BL0PR0102MB3425.prod.exchangelabs.com>
 <DB7PR10MB2377D1E48A043C178523F71695040@DB7PR10MB2377.EURPRD10.PROD.OUTLOOK.COM>
Message-ID: <BL0PR0102MB34255455B9BC21B23013A300BB040@BL0PR0102MB3425.prod.exchangelabs.com>

Thanks Martin and Alex.
I saw that and misread it as broadsoft.
I will find a way to disable it on the client.

Sean

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Martin Paterson
Sent: Tuesday, October 13, 2020 4:55 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] SUBSCRIBE - Bad Event

The event you’re subscribing to is in the header:
Event: x-broadworks-callpark

That’s related to the Cisco BroadWorks product. FreeSWITCH won’t know about those events.

Martin.


[cid:image001.png at 01D6A154.B3CD4F70]
Development Team​
[cid:image002.png at 01D6A154.B3CD4F70]<https://technologywithin.com/>
Phone:
0207 953 8840<tel:0207%20953%208840>
Email:
martin.paterson at technologywithin.com<mailto:martin.paterson at technologywithin.com>
Chevron Business Park, Limekiln Lane
,
Southampton
,
Hampshire
,
SO45 2QL
[cid:image003.png at 01D6A154.B3CD4F70]<https://en-gb.facebook.com/we.are.technologywithin>
[cid:image004.png at 01D6A154.B3CD4F70]<https://www.linkedin.com/company/technologywithin>
[cid:image005.png at 01D6A154.B3CD4F70]<https://twitter.com/@techwithin>
[cid:image006.png at 01D6A154.B3CD4F70]<https://www.instagram.com/technologywithin/>
[cid:image007.png at 01D6A154.B3CD4F70]<https://zoom.us/webinar/register/6916019886550/WN_HjuT4R-uTD636gPHq18iRA>
Registered Office: CP House, Otterspool Way, Watford, WD25 8JJ, U.K
​Registered in England No: 5964349 | VAT Number: GB 902 5369 37
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> On Behalf Of Sean Devoy
Sent: 12 October 2020 22:51
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: [Freeswitch-users] SUBSCRIBE - Bad Event

Hi,

I am getting 489 Bad Event for my users “SUBSCRIBE” packets. SUBSCRIBE is clearly listed in ALLOW. 2 Sample packets below.

Any ideas?

Thanks,
Sean

recv 524 bytes from tcp/[73.555.94.74]:1024 at 17:52:51.791847:
------------------------------------------------------------------------
SUBSCRIBE sip:214 at clientdomain.com<sip:214 at fs-esta2.bizfocused.com> SIP/2.0
Via: SIP/2.0/TCP 192.168.2.124:5075;branch=z9hG4bK-1d541664;rport
From: "214" <sip:214 at clientdomain.com<sip:214 at fs-esta2.bizfocused.com>>;tag=59cf9549d7610470
To: "214" <sip:214 at clientdomain.com<sip:214 at fs-esta2.bizfocused.com>>
Call-ID: b36ae378-cf9d3789 at 192.168.2.124<mailto:b36ae378-cf9d3789 at 192.168.2.124>
CSeq: 1001 SUBSCRIBE
Max-Forwards: 70
Contact: "214" <sip:214 at 192.168.2.124:5075;transport=tcp>
Accept: application/x-broadworks-callpark-info+xml
Expires: 1800
Event: x-broadworks-callpark
User-Agent: Cisco/SPA504G-7.6.2e
Content-Length: 0

send 717 bytes to tcp/[73.555.94.74]:1024 at 17:52:51.791962:
------------------------------------------------------------------------
SIP/2.0 489 Bad Event
Via: SIP/2.0/TCP 192.168.2.124:5075;branch=z9hG4bK-1d541664;rport=1024;received=73.555.94.74
From: "214" <sip:214 at clientdomain.com<sip:214 at fs-esta2.bizfocused.com>>;tag=59cf9549d7610470
To: "214" <sip:214 at clientdomain.com<sip:214 at fs-esta2.bizfocused.com>>;tag=279QBZZaKS7cr
Call-ID: b36ae378-cf9d3789 at 192.168.2.124<mailto:b36ae378-cf9d3789 at 192.168.2.124>
CSeq: 1001 SUBSCRIBE
User-Agent: FreeSWITCH-mod_sofia/1.10.2-release-14-f7bdd3845a~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0
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From mike at freeswitch.org  Tue Oct 13 17:03:09 2020
From: mike at freeswitch.org (Mike Jerris)
Date: Tue, 13 Oct 2020 13:03:09 -0400
Subject: [Freeswitch-users] SUBSCRIBE -  Bad  Event
In-Reply-To: <BL0PR0102MB34255455B9BC21B23013A300BB040@BL0PR0102MB3425.prod.exchangelabs.com>
References: <BL0PR0102MB34258901D7CA3942AFE79AD7BB070@BL0PR0102MB3425.prod.exchangelabs.com>
 <DB7PR10MB2377D1E48A043C178523F71695040@DB7PR10MB2377.EURPRD10.PROD.OUTLOOK.COM>
 <BL0PR0102MB34255455B9BC21B23013A300BB040@BL0PR0102MB3425.prod.exchangelabs.com>
Message-ID: <DBC1F448-5565-48F0-AF37-0FB954A61B45@freeswitch.org>

Freeswitch doesn’t support this event type.  It is proprietary


> On Oct 13, 2020, at 11:33 AM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> 
> Thanks Martin and Alex.
> I saw that and misread it as broadsoft.
> I will find a way to disable it on the client.
>  
> Sean
>  
> From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>> On Behalf Of Martin Paterson
> Sent: Tuesday, October 13, 2020 4:55 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: Re: [Freeswitch-users] SUBSCRIBE - Bad Event
>  
> The event you’re subscribing to is in the header:
> Event: x-broadworks-callpark
>  
> That’s related to the Cisco BroadWorks product. FreeSWITCH won’t know about those events.
>  
> Martin.
>  
>  
> <image001.png>
> Development Team​
> <image002.png> <https://technologywithin.com/>
> Phone: 
> 0207 953 8840 <tel:0207%20953%208840>
> Email: 
> martin.paterson at technologywithin.com <mailto:martin.paterson at technologywithin.com>
> Chevron Business Park, Limekiln Lane
> , 
> Southampton
> , 
> Hampshire
> , 
> SO45 2QL
> <image003.png> <https://en-gb.facebook.com/we.are.technologywithin>	
> <image004.png> <https://www.linkedin.com/company/technologywithin>	
> <image005.png> <https://twitter.com/@techwithin>	
> <image006.png> <https://www.instagram.com/technologywithin/>
> <image007.png> <https://zoom.us/webinar/register/6916019886550/WN_HjuT4R-uTD636gPHq18iRA>
> Registered Office: CP House, Otterspool Way, Watford, WD25 8JJ, U.K 
> ​Registered in England No: 5964349 | VAT Number: GB 902 5369 37 
> From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>> On Behalf Of Sean Devoy
> Sent: 12 October 2020 22:51
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: [Freeswitch-users] SUBSCRIBE - Bad Event
>  
> Hi,
>  
> I am getting 489 Bad Event for my users “SUBSCRIBE” packets. SUBSCRIBE is clearly listed in ALLOW. 2 Sample packets below.
>  
> Any ideas?
>  
> Thanks,
> Sean
>  
> recv 524 bytes from tcp/[73.555.94.74]:1024 at 17:52:51.791847:
> ------------------------------------------------------------------------
> SUBSCRIBE sip:214 at clientdomain.com <sip:214 at fs-esta2.bizfocused.com> SIP/2.0
> Via: SIP/2.0/TCP 192.168.2.124:5075;branch=z9hG4bK-1d541664;rport
> From: "214" <sip:214 at clientdomain.com <sip:214 at fs-esta2.bizfocused.com>>;tag=59cf9549d7610470
> To: "214" <sip:214 at clientdomain.com <sip:214 at fs-esta2.bizfocused.com>>
> Call-ID: b36ae378-cf9d3789 at 192.168.2.124 <mailto:b36ae378-cf9d3789 at 192.168.2.124>
> CSeq: 1001 SUBSCRIBE
> Max-Forwards: 70
> Contact: "214" <sip:214 at 192.168.2.124:5075;transport=tcp <sip:214 at 192.168.2.124:5075;transport=tcp>>
> Accept: application/x-broadworks-callpark-info+xml
> Expires: 1800
> Event: x-broadworks-callpark
> User-Agent: Cisco/SPA504G-7.6.2e
> Content-Length: 0
>  
> send 717 bytes to tcp/[73.555.94.74]:1024 at 17:52:51.791962:
> ------------------------------------------------------------------------
> SIP/2.0 489 Bad Event
> Via: SIP/2.0/TCP 192.168.2.124:5075;branch=z9hG4bK-1d541664;rport=1024;received=73.555.94.74
> From: "214" <sip:214 at clientdomain.com <sip:214 at fs-esta2.bizfocused.com>>;tag=59cf9549d7610470
> To: "214" <sip:214 at clientdomain.com <sip:214 at fs-esta2.bizfocused.com>>;tag=279QBZZaKS7cr
> Call-ID: b36ae378-cf9d3789 at 192.168.2.124 <mailto:b36ae378-cf9d3789 at 192.168.2.124>
> CSeq: 1001 SUBSCRIBE
> User-Agent: FreeSWITCH-mod_sofia/1.10.2-release-14-f7bdd3845a~64bit
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, path, replaces
> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
> Content-Length: 

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From shahnawaj.khan1990 at gmail.com  Tue Oct 13 19:01:56 2020
From: shahnawaj.khan1990 at gmail.com (Shahnawaj Khan)
Date: Wed, 14 Oct 2020 00:31:56 +0530
Subject: [Freeswitch-users] Freeswitch skipping silence packets in recording
	file
Message-ID: <CAGmow13uDf+oV0qj3o8JeROH+H1W8=dtpNdbu=k-DcxMuV275g@mail.gmail.com>

Hello All,,

I am working on freeswitch 1.10.2 and using record_session to record
the media of the calls. I am running a scenario where we have silence
on both ends of the call for a few minutes. Assuming We have the call
talk time of 10 mins and the call contains 1 minutes of silence. The
recording file size is around 9 minutes and 30 seconds i.e. it
contains 30 seconds of silence but also skipped for about 30 sec.
I haven't configured the "silence_hits" value for the record_session.
I am unable to find the default value of silence_hit associated with
record_session or it may be due to some other reason.

Thanks & Regards,
Shahnawaz


From david.villasmil.work at gmail.com  Tue Oct 13 22:13:34 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 13 Oct 2020 23:13:34 +0100
Subject: [Freeswitch-users] FS on a VM only uses one cpu?
Message-ID: <CAFGRPVptR7vuz+t31W-YVQJjHifpuoAntmV4VUqirRR9ujPXaA@mail.gmail.com>

Hello all,

I compiled 1.10 on a debian-buster VM, and from htop it looks it's only
utilizing one CPU.
Anyone seen this?

[image: Screenshot 2020-10-13 at 23.11.13.png]

I'm just running test calls, fs receives and playback a wav file. This is
at 700 channels.

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From david.villasmil.work at gmail.com  Tue Oct 13 22:14:25 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 13 Oct 2020 23:14:25 +0100
Subject: [Freeswitch-users] FS on a VM only uses one cpu?
In-Reply-To: <CAFGRPVptR7vuz+t31W-YVQJjHifpuoAntmV4VUqirRR9ujPXaA@mail.gmail.com>
References: <CAFGRPVptR7vuz+t31W-YVQJjHifpuoAntmV4VUqirRR9ujPXaA@mail.gmail.com>
Message-ID: <CAFGRPVp8ry2Bxn-b2b3gx0z7GJ3DcHcno_dBf+EBo-JVJNdc_Q@mail.gmail.com>

BTW sqlite is on a ram disk.

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337


On Tue, Oct 13, 2020 at 11:13 PM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello all,
>
> I compiled 1.10 on a debian-buster VM, and from htop it looks it's only
> utilizing one CPU.
> Anyone seen this?
>
> [image: Screenshot 2020-10-13 at 23.11.13.png]
>
> I'm just running test calls, fs receives and playback a wav file. This is
> at 700 channels.
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
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From shaun at sysconfig.cloud  Wed Oct 14 06:10:32 2020
From: shaun at sysconfig.cloud (Shaun Stokes)
Date: Wed, 14 Oct 2020 06:10:32 +0000
Subject: [Freeswitch-users] FS on a VM only uses one cpu?
In-Reply-To: <CAFGRPVp8ry2Bxn-b2b3gx0z7GJ3DcHcno_dBf+EBo-JVJNdc_Q@mail.gmail.com>
References: <CAFGRPVptR7vuz+t31W-YVQJjHifpuoAntmV4VUqirRR9ujPXaA@mail.gmail.com>,
 <CAFGRPVp8ry2Bxn-b2b3gx0z7GJ3DcHcno_dBf+EBo-JVJNdc_Q@mail.gmail.com>
Message-ID: <CWLP123MB237156BD0110CE252B6A8E62B7050@CWLP123MB2371.GBRP123.PROD.OUTLOOK.COM>

Hi David,

We also use FS on a RAM Disk, it works great with 100's of concurrent calls on all the databases except for the call centre module which quickly becomes CPU bound.

Were you using the call centre module in your test? If so, I'd suggest moving it to an external RDBMS such as PGSQL (could be running on the same VM).

FS is multi-threaded but there are certain processes which FS creates a dedicated child process for (one thread) such as SQLite (one process per database). If you can find out which FS process is using 100% CPU on one thread then you should be able to identify the bottleneck.

Similar issue here: https://lists.freeswitch.org/pipermail/freeswitch-users/2012-October/088895.html

Thanks,
Shaun
________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of David Villasmil <david.villasmil.work at gmail.com>
Sent: 14 October 2020 00:14
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FS on a VM only uses one cpu?

BTW sqlite is on a ram disk.

Regards,

David Villasmil
email: david.villasmil.work at gmail.com<mailto:david.villasmil.work at gmail.com>
phone: +34669448337


On Tue, Oct 13, 2020 at 11:13 PM David Villasmil <david.villasmil.work at gmail.com<mailto:david.villasmil.work at gmail.com>> wrote:
Hello all,

I compiled 1.10 on a debian-buster VM, and from htop it looks it's only utilizing one CPU.
Anyone seen this?

[Screenshot 2020-10-13 at 23.11.13.png]

I'm just running test calls, fs receives and playback a wav file. This is at 700 channels.

Regards,

David Villasmil
email: david.villasmil.work at gmail.com<mailto:david.villasmil.work at gmail.com>
phone: +34669448337
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From david.villasmil.work at gmail.com  Wed Oct 14 09:35:02 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 14 Oct 2020 10:35:02 +0100
Subject: [Freeswitch-users] FS on a VM only uses one cpu?
In-Reply-To: <CWLP123MB237156BD0110CE252B6A8E62B7050@CWLP123MB2371.GBRP123.PROD.OUTLOOK.COM>
References: <CAFGRPVptR7vuz+t31W-YVQJjHifpuoAntmV4VUqirRR9ujPXaA@mail.gmail.com>
 <CAFGRPVp8ry2Bxn-b2b3gx0z7GJ3DcHcno_dBf+EBo-JVJNdc_Q@mail.gmail.com>
 <CWLP123MB237156BD0110CE252B6A8E62B7050@CWLP123MB2371.GBRP123.PROD.OUTLOOK.COM>
Message-ID: <CAFGRPVr8xBm-Dgag0WqeXv66o-E8kSbwkjpUSBitO3PntAHszg@mail.gmail.com>

Hello Shaun,

Thanks for replying. I’m not using mod_callcenter at all. I’m simply
playing back a file. I wonder if this is cpu bounding. I guess I can test
this by doing echo instead.

I also added another profile, since I read on a thread it might help (also
read that’s not true), but that’s not it. All of my metal boxes work just
fine at ~3000 channels, so I’m thinking it’s related somehow to the
underlying VM.

Thanks again.

On Wed, 14 Oct 2020 at 07:29, Shaun Stokes <shaun at sysconfig.cloud> wrote:

>
>
>
>
>
>
>
>
>
>
>
>
>
>
> Hi David,
>
>
>
>
>
>
>
>
>
>
>
> We also use FS on a RAM Disk, it works great with 100's of concurrent
> calls on all the databases except for the call centre module which quickly
> becomes CPU bound.
>
>
>
>
>
>
>
>
>
>
>
> Were you using the call centre module in your test? If so, I'd suggest
> moving it to an external RDBMS such as PGSQL (could be
>
> running on the same VM).
>
>
>
>
>
>
>
>
>
>
>
>
>
> FS is multi-threaded but there are certain processes which FS creates a
> dedicated child process for (one thread) such as SQLite (one process per
> database). If you can find out which FS process is using 100% CPU on one
> thread then you should be able to identify
>
> the bottleneck.
>
>
>
>
>
>
>
>
>
>
>
> Similar issue here:
> https://lists.freeswitch.org/pipermail/freeswitch-users/2012-October/088895.html
>
>
>
>
>
>
>
>
>
>
>
> Thanks,
>
>
>
>
> Shaun
>
>
>
>
> ------------------------------
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org>
> on behalf of David Villasmil <david.villasmil.work at gmail.com>
>
>
> *Sent:* 14 October 2020 00:14
>
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
>
> *Subject:* Re: [Freeswitch-users] FS on a VM only uses one cpu?
>
>
>
>
>
>
>
>
> BTW sqlite is on a ram disk.
>
>
>
>
>
>
>
>
>
>
> Regards,
>
>
>
>
>
>
>
> David Villasmil
>
> email: david.villasmil.work at gmail.com
>
>
> phone: +34669448337
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> On Tue, Oct 13, 2020 at 11:13 PM David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>
>
>
>
>
>
>
> Hello all,
>
>
>
>
>
>
>
>
>
> I compiled 1.10 on a debian-buster VM, and from htop it looks it's only
> utilizing one CPU.
>
>
> Anyone seen this?
>
>
>
>
>
>
>
>
>
> [image: Screenshot 2020-10-13 at 23.11.13.png]
>
>
>
>
>
>
>
>
>
>
>
> I'm just running test calls, fs receives and playback a wav file. This is
> at 700 channels.
>
>
>
>
>
>
>
>
>
>
>
>
> Regards,
>
>
>
>
>
>
>
> David Villasmil
>
> email: david.villasmil.work at gmail.com
>
>
> phone: +34669448337
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
>
>
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
>
> Build your next product on our scalable cloud platform.
>
>
>
> Join our online community to chat in real time
> https://signalwire.community
>
>
>
> Professional FreeSWITCH Services
>
> sales at freeswitch.com
>
> https://freeswitch.com
>
>
>
> Official FreeSWITCH Sites
>
> https://freeswitch.com/oss
>
> https://freeswitch.org/confluence
>
> https://cluecon.com
>
>
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From gmaruzz at gmail.com  Wed Oct 14 12:10:30 2020
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 14 Oct 2020 14:10:30 +0200
Subject: [Freeswitch-users] FS on a VM only uses one cpu?
In-Reply-To: <CAFGRPVr8xBm-Dgag0WqeXv66o-E8kSbwkjpUSBitO3PntAHszg@mail.gmail.com>
References: <CAFGRPVptR7vuz+t31W-YVQJjHifpuoAntmV4VUqirRR9ujPXaA@mail.gmail.com>
 <CAFGRPVp8ry2Bxn-b2b3gx0z7GJ3DcHcno_dBf+EBo-JVJNdc_Q@mail.gmail.com>
 <CWLP123MB237156BD0110CE252B6A8E62B7050@CWLP123MB2371.GBRP123.PROD.OUTLOOK.COM>
 <CAFGRPVr8xBm-Dgag0WqeXv66o-E8kSbwkjpUSBitO3PntAHszg@mail.gmail.com>
Message-ID: <CALXCt0qCamdeRgUybSU_f3u1p5A8xDL1wHp3f3GUNmWojAP2VQ@mail.gmail.com>

On Wed, Oct 14, 2020 at 11:35 AM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Thanks for replying. I’m not using mod_callcenter at all. I’m simply
> playing back a file. I wonder if this is cpu bounding. I guess I can test
> this by doing echo instead.
>
>
to eliminate cpu load from reading/decoding file, use moh (read just one
time, then broadcast samples to all calls, no cpu, I believe less than echo
in case of many concurrent calls)

-giovanni

I also added another profile, since I read on a thread it might help (also
> read that’s not true), but that’s not it. All of my metal boxes work just
> fine at ~3000 channels, so I’m thinking it’s related somehow to the
> underlying VM.
>
> Thanks again.
>
> On Wed, 14 Oct 2020 at 07:29, Shaun Stokes <shaun at sysconfig.cloud> wrote:
>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Hi David,
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> We also use FS on a RAM Disk, it works great with 100's of concurrent
>> calls on all the databases except for the call centre module which quickly
>> becomes CPU bound.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Were you using the call centre module in your test? If so, I'd suggest
>> moving it to an external RDBMS such as PGSQL (could be
>>
>> running on the same VM).
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> FS is multi-threaded but there are certain processes which FS creates a
>> dedicated child process for (one thread) such as SQLite (one process per
>> database). If you can find out which FS process is using 100% CPU on one
>> thread then you should be able to identify
>>
>> the bottleneck.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Similar issue here:
>> https://lists.freeswitch.org/pipermail/freeswitch-users/2012-October/088895.html
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Thanks,
>>
>>
>>
>>
>> Shaun
>>
>>
>>
>>
>> ------------------------------
>>
>>
>> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org>
>> on behalf of David Villasmil <david.villasmil.work at gmail.com>
>>
>>
>> *Sent:* 14 October 2020 00:14
>>
>>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>
>>
>> *Subject:* Re: [Freeswitch-users] FS on a VM only uses one cpu?
>>
>>
>>
>>
>>
>>
>>
>>
>> BTW sqlite is on a ram disk.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Regards,
>>
>>
>>
>>
>>
>>
>>
>> David Villasmil
>>
>> email: david.villasmil.work at gmail.com
>>
>>
>> phone: +34669448337
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On Tue, Oct 13, 2020 at 11:13 PM David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>
>>
>>
>>
>>
>>
>>
>> Hello all,
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> I compiled 1.10 on a debian-buster VM, and from htop it looks it's only
>> utilizing one CPU.
>>
>>
>> Anyone seen this?
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> [image: Screenshot 2020-10-13 at 23.11.13.png]
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> I'm just running test calls, fs receives and playback a wav file. This is
>> at 700 channels.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> Regards,
>>
>>
>>
>>
>>
>>
>>
>> David Villasmil
>>
>> email: david.villasmil.work at gmail.com
>>
>>
>> phone: +34669448337
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>>
>>
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>>
>> Build your next product on our scalable cloud platform.
>>
>>
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>>
>>
>> Professional FreeSWITCH Services
>>
>> sales at freeswitch.com
>>
>> https://freeswitch.com
>>
>>
>>
>> Official FreeSWITCH Sites
>>
>> https://freeswitch.com/oss
>>
>> https://freeswitch.org/confluence
>>
>> https://cluecon.com
>>
>>
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> https://freeswitch.com
>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From nathan at robotics.net  Wed Oct 14 12:40:25 2020
From: nathan at robotics.net (Nathan Stratton)
Date: Wed, 14 Oct 2020 08:40:25 -0400
Subject: [Freeswitch-users] Play video file to conference
Message-ID: <CAHubkyM6EN_vrKY+jJmVyTma2CADw79HPoxKov=VabUVmCC-Nw@mail.gmail.com>

I have a broadcast only conference room setup where all users who dial in
get audio / video, but can't send it. I would like to play a video file to
it, but when I try it I only get audio.

conference 4600 play /root/spacex.mp4

I can broadcast to a specific user and get audio and video:

uuid_broadcast 9f857c4a-0294-4446-8864-e92a8ac0433c /root/spacex.mp4

><>
nathan stratton
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From prashant.lamba at gmail.com  Wed Oct 14 14:32:11 2020
From: prashant.lamba at gmail.com (Prashant Lamba)
Date: Wed, 14 Oct 2020 20:02:11 +0530
Subject: [Freeswitch-users] IVR UI/UX design
In-Reply-To: <em36deb13a-aa38-4bb3-a5e6-f03845a08ea2@fred-pc>
References: <CAAtWTekEPFw4AxxigLfd0d9nzMrW+1-WFFT5LUvd3u2Fo_VNYg@mail.gmail.com>
 <009901d69e3f$3943a0f0$abcae2d0$@gmail.com>
 <em36deb13a-aa38-4bb3-a5e6-f03845a08ea2@fred-pc>
Message-ID: <CAPHrvXF3SGr+uUV+VTxBiDvoPMBtkpd6=2YZ4ZWbBSn5mTo8ag@mail.gmail.com>

++ Vote for Frederic Jean.

One more: Make sure you always use the standard '0' or '9' as the xfer to
operator agent, if you have one.

Prashant

On Fri, Oct 9, 2020 at 7:49 PM Frederic Jean <makafre at gmail.com> wrote:

>
> Generally speaking:
>
> - The more options in a menu the more chance your caller will need to have
> a menu repeated;
> - Always tell most used options first so that the caller doesn't stay long
> in the IVR;
> - Use 3x3 or 4x4 matrix - e.g. 3 levels deep w/3 options each at a maximum;
> - Always provide a key to repeat the menu and optimize the waiting time
> for key input so that callers have enough time;
> - Use the same voice across the menu options;
> - Optimize each sentence - they must be short but very informative;
> - Be consistent; e.g. same way of telling the options across the menus;
> - Hang up after 3 wrong options - this also avoids lingering calls
>
> On 2020-10-09 09:22:22, "Bote Man" <botelist at gmail.com> wrote:
>
> I don’t know the answer, but there’s a good possibility that a fine lady
> who records professional IVR prompts knows where to point you:
>
>
>
> Allison at theivrvoice.com
>
>
>
>
>
> ---
>
> John Boteler
>
> BnC Group U.S.A.
>
>
>
>
>
>
>
>
>
> *From:* Sam van Herwaarden
> *Sent:* Friday, 9 October, 2020 03:44
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] IVR UI/UX design
>
>
>
> Hi all,
>
>
>
> My organization runs a number of IVR systems for information delivery, and
> one issue we often run into is that users find the interface a bit awkward
> or hard to navigate. We have experimented quite a bit internally with
> adjusting the structures of menus, ordering, phrasing etc. But we feel like
> there must be some knowledge/experience out there and that it shouldn't be
> necessary to rediscover most of this ourselves.
>
>
>
> Does anyone have recommendations for resources with info on best practices
> for IVR design? Or contacts in organizations/companies that have a lot of
> experience with this?
>
>
>
> Kind regards,
>
> Sam
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From avi at avimarcus.net  Wed Oct 14 14:52:44 2020
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 14 Oct 2020 14:52:44 +0000
Subject: [Freeswitch-users] IVR UI/UX design
In-Reply-To: <em36deb13a-aa38-4bb3-a5e6-f03845a08ea2@fred-pc>
References: <CAAtWTekEPFw4AxxigLfd0d9nzMrW+1-WFFT5LUvd3u2Fo_VNYg@mail.gmail.com>
 <009901d69e3f$3943a0f0$abcae2d0$@gmail.com>
 <em36deb13a-aa38-4bb3-a5e6-f03845a08ea2@fred-pc>
Message-ID: <0100017527996013-d3cca8fd-f55a-41e1-948b-37512abc7fef-000000@email.amazonses.com>

As an example: when I call a place for the first time, I have to hear the
whole menu to see which option sounds closest to what I need.
Of course by that time I've forgotten which digit it is, and I need to
listen again.

On Fri, Oct 9, 2020, 5:12 PM Frederic Jean <makafre at gmail.com> wrote:

>
> - The more options in a menu the more chance your caller will need to have
> a menu repeated;
> - Always provide a key to repeat the menu and optimize the waiting time
> for key input so that callers have enough time;
>
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From nathan at robotics.net  Wed Oct 14 15:57:59 2020
From: nathan at robotics.net (Nathan Stratton)
Date: Wed, 14 Oct 2020 11:57:59 -0400
Subject: [Freeswitch-users] $500 bounty RTMP to conference room
Message-ID: <CAHubkyNPZ4Rc=cZCy7co1U_1vbOLOGm41ghaHXyJcG4sW=OF8g@mail.gmail.com>

Trying to broadcast from vlc or a sling box with rtmp to freeswitch
conference room. I have mod_rtmp installed and why I try to steam from vlc
I get:

2020-10-14 11:56:44.516563 [NOTICE] mod_rtmp.c:905 New RTMP session
[88e00c1b-6b3a-4bac-83f5-bc12b879884b]
2020-10-14 11:56:44.516563 [INFO] rtmp_tcp.c:234 Rtmp connection from
50.210.153.253:36864
2020-10-14 11:56:44.516563 [DEBUG] rtmp.c:882 Sent handshake response
2020-10-14 11:56:44.536531 [DEBUG] rtmp.c:907 Done with handshake
2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
ts=0 stream_id=0x0] len=155
2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for
connect
2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x1
stream_id=0x0] len=4
2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x5
stream_id=0x0] len=4
2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x6
stream_id=0x0] len=5
2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4
stream_id=0x0] len=6
2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
stream_id=0x0] len=201
2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
stream_id=0x0] len=61
2020-10-14 11:56:44.556571 [NOTICE] rtmp_sig.c:122 Sent connect reply
2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=2 type=0x1
ts=0 stream_id=0x0] len=4
2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1089 SET CHUNKSIZE=512
2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
ts=0 stream_id=0x0] len=29
2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for
releaseStream
2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for
"releaseStream"
2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
ts=0 stream_id=0x0] len=25
2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for
FCPublish
2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for
"FCPublish"
2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
ts=0 stream_id=0x0] len=25
2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for
createStream
2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
stream_id=0x0] len=29
2020-10-14 11:56:44.636529 [INFO] rtmp_sig.c:137 Replied to createStream (0)
2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:1081 [chunk_stream=8 type=0x14
ts=0 stream_id=0x1] len=30
2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:192 [amfnumber=8] Got INVOKE for
publish
2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4
stream_id=0x0] len=6
2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=5 type=0x14
stream_id=0x1] len=138
2020-10-14 11:56:44.696580 [INFO] rtmp_sig.c:290 Got publish on stream 1.
2020-10-14 11:56:45.216562 [DEBUG] rtmp.c:1081 [chunk_stream=4 type=0x12
ts=0 stream_id=0x1] len=319
2020-10-14 11:56:45.656516 [DEBUG] rtmp_tcp.c:243 Closing socket
2020-10-14 11:56:51.796530 [NOTICE] mod_rtmp.c:1027 RTMP session ended
[88e00c1b-6b3a-4bac-83f5-bc12b879884b]

Anyone able to help me? $500 via Paypal.

><>
nathan stratton
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From botelist at gmail.com  Wed Oct 14 16:55:57 2020
From: botelist at gmail.com (Bote Man)
Date: Wed, 14 Oct 2020 12:55:57 -0400
Subject: [Freeswitch-users] moh music on hold
Message-ID: <019701d6a24a$e3a92100$aafb6300$@gmail.com>

This is very interesting!

 

Giovanni, do you mean that FS reads once the target of the “hold_music” variable?

 

Or specifically “local_stream://moh” as an argument to local_stream?


I’ll update the wiki with this useful info.

 

Thanks.

 

 

--- 

John Boteler 

BnC Group U.S.A. 

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Giovanni Maruzzelli
Sent: Wednesday, 14 October, 2020 08:11
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FS on a VM only uses one cpu?

 

On Wed, Oct 14, 2020 at 11:35 AM David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com> > wrote:

Thanks for replying. I’m not using mod_callcenter at all. I’m simply playing back a file. I wonder if this is cpu bounding. I guess I can test this by doing echo instead.

 

 

to eliminate cpu load from reading/decoding file, use moh (read just one time, then broadcast samples to all calls, no cpu, I believe less than echo in case of many concurrent calls)

 

-giovanni

 

I also added another profile, since I read on a thread it might help (also read that’s not true), but that’s not it. All of my metal boxes work just fine at ~3000 channels, so I’m thinking it’s related somehow to the underlying VM.

 

Thanks again.

 

On Wed, 14 Oct 2020 at 07:29, Shaun Stokes <shaun at sysconfig.cloud <mailto:shaun at sysconfig.cloud> > wrote:











 



Hi David,

 






 



We also use FS on a RAM Disk, it works great with 100's of concurrent calls on all the databases except for the call centre module which quickly becomes CPU bound.

 






 



Were you using the call centre module in your test? If so, I'd suggest moving it to an external RDBMS such as PGSQL (could be

running on the same VM).

 






 



FS is multi-threaded but there are certain processes which FS creates a dedicated child process for (one thread) such as SQLite (one process per database). If you can find out which FS process is using 100% CPU on one thread then you should be able to identify

the bottleneck.

 






 



Similar issue here: https://lists.freeswitch.org/pipermail/freeswitch-users/2012-October/088895.html

 






 



Thanks,

 



Shaun

 

 

  _____  

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> > on behalf of David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com> >


Sent: 14 October 2020 00:14


To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >


Subject: Re: [Freeswitch-users] FS on a VM only uses one cpu?

 

 

 

 

BTW sqlite is on a ram disk.




 

 

 

Regards,

 

 



David Villasmil

email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com> 

 

phone: +34669448337

 

 

 






 






 

On Tue, Oct 13, 2020 at 11:13 PM David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com> > wrote:

 

 

 

Hello all,

 

 

 

I compiled 1.10 on a debian-buster VM, and from htop it looks it's only utilizing one CPU.


Anyone seen this?

 

 

 

 



 

 

 



I'm just running test calls, fs receives and playback a wav file. This is at 700 channels.

 

 

 

 

 

Regards,

 

 



David Villasmil

email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com> 

 

phone: +34669448337

 

 

 

 

 

 

 

 

 

 

 





_________________________________________________________________________



The FreeSWITCH project is sponsored by SignalWire https://signalwire.com

Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.

Build your next product on our scalable cloud platform.



Join our online community to chat in real time https://signalwire.community



Professional FreeSWITCH Services

sales at freeswitch.com <mailto:sales at freeswitch.com> 

https://freeswitch.com



Official FreeSWITCH Sites

https://freeswitch.com/oss

https://freeswitch.org/confluence

https://cluecon.com



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org> 

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

https://freeswitch.com

-- 

Regards,

 

David Villasmil

email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com> 

phone: +34669448337

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales at freeswitch.com <mailto:sales at freeswitch.com> 
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org> 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com




 

-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18

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From mike at freeswitch.org  Wed Oct 14 16:56:46 2020
From: mike at freeswitch.org (Mike Jerris)
Date: Wed, 14 Oct 2020 12:56:46 -0400
Subject: [Freeswitch-users] $500 bounty RTMP to conference room
In-Reply-To: <CAHubkyNPZ4Rc=cZCy7co1U_1vbOLOGm41ghaHXyJcG4sW=OF8g@mail.gmail.com>
References: <CAHubkyNPZ4Rc=cZCy7co1U_1vbOLOGm41ghaHXyJcG4sW=OF8g@mail.gmail.com>
Message-ID: <5EFA741A-5934-46DD-98D5-9370E039BF4B@freeswitch.org>

Use mod_av to record/playback rtmp://

mod_rtmp is an endpoint module. Its a different animal that really has no use now that webrtc is available almost everywhere.

> On Oct 14, 2020, at 11:57 AM, Nathan Stratton <nathan at robotics.net> wrote:
> 
> 
> Trying to broadcast from vlc or a sling box with rtmp to freeswitch conference room. I have mod_rtmp installed and why I try to steam from vlc I get:
> 
> 2020-10-14 11:56:44.516563 [NOTICE] mod_rtmp.c:905 New RTMP session [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
> 2020-10-14 11:56:44.516563 [INFO] rtmp_tcp.c:234 Rtmp connection from 50.210.153.253:36864 <http://50.210.153.253:36864/>
> 2020-10-14 11:56:44.516563 [DEBUG] rtmp.c:882 Sent handshake response
> 2020-10-14 11:56:44.536531 [DEBUG] rtmp.c:907 Done with handshake
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=155
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for connect
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x1 stream_id=0x0] len=4 
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x5 stream_id=0x0] len=4 
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x6 stream_id=0x0] len=5 
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4 stream_id=0x0] len=6 
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14 stream_id=0x0] len=201 
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14 stream_id=0x0] len=61 
> 2020-10-14 11:56:44.556571 [NOTICE] rtmp_sig.c:122 Sent connect reply
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=2 type=0x1 ts=0 stream_id=0x0] len=4
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1089 SET CHUNKSIZE=512
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=29
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for releaseStream
> 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for "releaseStream"
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=25
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for FCPublish
> 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for "FCPublish"
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=25
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for createStream
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14 stream_id=0x0] len=29 
> 2020-10-14 11:56:44.636529 [INFO] rtmp_sig.c:137 Replied to createStream (0)
> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:1081 [chunk_stream=8 type=0x14 ts=0 stream_id=0x1] len=30
> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:192 [amfnumber=8] Got INVOKE for publish
> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4 stream_id=0x0] len=6 
> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=5 type=0x14 stream_id=0x1] len=138 
> 2020-10-14 11:56:44.696580 [INFO] rtmp_sig.c:290 Got publish on stream 1.
> 2020-10-14 11:56:45.216562 [DEBUG] rtmp.c:1081 [chunk_stream=4 type=0x12 ts=0 stream_id=0x1] len=319
> 2020-10-14 11:56:45.656516 [DEBUG] rtmp_tcp.c:243 Closing socket
> 2020-10-14 11:56:51.796530 [NOTICE] mod_rtmp.c:1027 RTMP session ended [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
> 
> Anyone able to help me? $500 via Paypal. 
> 
> ><>
> nathan stratto

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From nathan at robotics.net  Wed Oct 14 18:15:33 2020
From: nathan at robotics.net (Nathan Stratton)
Date: Wed, 14 Oct 2020 14:15:33 -0400
Subject: [Freeswitch-users] $500 bounty RTMP to conference room
In-Reply-To: <5EFA741A-5934-46DD-98D5-9370E039BF4B@freeswitch.org>
References: <CAHubkyNPZ4Rc=cZCy7co1U_1vbOLOGm41ghaHXyJcG4sW=OF8g@mail.gmail.com>
 <5EFA741A-5934-46DD-98D5-9370E039BF4B@freeswitch.org>
Message-ID: <CAHubkyOv-c8DG-3aQHN87a+_QU0o3vsSQY-wD_SmhhF7O6oCYA@mail.gmail.com>

Got it, I can't find much on mod_av and nothing on how to have mod_av
accept rtmp stream and add it to a conference. Not a big fan of RTMP, but
that is what slingstudio supports.

><>
nathan stratton


On Wed, Oct 14, 2020 at 12:57 PM Mike Jerris <mike at freeswitch.org> wrote:

> Use mod_av to record/playback rtmp://
>
> mod_rtmp is an endpoint module. Its a different animal that really has no
> use now that webrtc is available almost everywhere.
>
> On Oct 14, 2020, at 11:57 AM, Nathan Stratton <nathan at robotics.net> wrote:
>
>
> Trying to broadcast from vlc or a sling box with rtmp to freeswitch
> conference room. I have mod_rtmp installed and why I try to steam from vlc
> I get:
>
> 2020-10-14 11:56:44.516563 [NOTICE] mod_rtmp.c:905 New RTMP session
> [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
> 2020-10-14 11:56:44.516563 [INFO] rtmp_tcp.c:234 Rtmp connection from
> 50.210.153.253:36864
> 2020-10-14 11:56:44.516563 [DEBUG] rtmp.c:882 Sent handshake response
> 2020-10-14 11:56:44.536531 [DEBUG] rtmp.c:907 Done with handshake
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
> ts=0 stream_id=0x0] len=155
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for
> connect
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x1
> stream_id=0x0] len=4
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x5
> stream_id=0x0] len=4
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x6
> stream_id=0x0] len=5
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4
> stream_id=0x0] len=6
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
> stream_id=0x0] len=201
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
> stream_id=0x0] len=61
> 2020-10-14 11:56:44.556571 [NOTICE] rtmp_sig.c:122 Sent connect reply
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=2 type=0x1
> ts=0 stream_id=0x0] len=4
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1089 SET CHUNKSIZE=512
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
> ts=0 stream_id=0x0] len=29
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for
> releaseStream
> 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for
> "releaseStream"
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
> ts=0 stream_id=0x0] len=25
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for
> FCPublish
> 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for
> "FCPublish"
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
> ts=0 stream_id=0x0] len=25
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for
> createStream
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
> stream_id=0x0] len=29
> 2020-10-14 11:56:44.636529 [INFO] rtmp_sig.c:137 Replied to createStream
> (0)
> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:1081 [chunk_stream=8 type=0x14
> ts=0 stream_id=0x1] len=30
> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:192 [amfnumber=8] Got INVOKE for
> publish
> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4
> stream_id=0x0] len=6
> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=5 type=0x14
> stream_id=0x1] len=138
> 2020-10-14 11:56:44.696580 [INFO] rtmp_sig.c:290 Got publish on stream 1.
> 2020-10-14 11:56:45.216562 [DEBUG] rtmp.c:1081 [chunk_stream=4 type=0x12
> ts=0 stream_id=0x1] len=319
> 2020-10-14 11:56:45.656516 [DEBUG] rtmp_tcp.c:243 Closing socket
> 2020-10-14 11:56:51.796530 [NOTICE] mod_rtmp.c:1027 RTMP session ended
> [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
>
> Anyone able to help me? $500 via Paypal.
>
> ><>
> nathan stratto
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From shahnawaj.khan1990 at gmail.com  Wed Oct 14 18:36:09 2020
From: shahnawaj.khan1990 at gmail.com (Shahnawaj Khan)
Date: Thu, 15 Oct 2020 00:06:09 +0530
Subject: [Freeswitch-users] Freeswitch skipping silence packets in
	recording file
In-Reply-To: <CAGmow13uDf+oV0qj3o8JeROH+H1W8=dtpNdbu=k-DcxMuV275g@mail.gmail.com>
References: <CAGmow13uDf+oV0qj3o8JeROH+H1W8=dtpNdbu=k-DcxMuV275g@mail.gmail.com>
Message-ID: <CAGmow12C3jC7Dhydfc0i4kg=1+gYzd-g78s1tri5yYL4P02ZLg@mail.gmail.com>

Hello All,

I am still facing the issue where the duration of the recording is
less than the duration of the call for the mentioned use case. Please
let me know if we can force freeswitch to not trim the silence in call
recording.

Thanks & Regards,
Shahnawaz

On Wed, Oct 14, 2020 at 12:31 AM Shahnawaj Khan
<shahnawaj.khan1990 at gmail.com> wrote:
>
> Hello All,,
>
> I am working on freeswitch 1.10.2 and using record_session to record
> the media of the calls. I am running a scenario where we have silence
> on both ends of the call for a few minutes. Assuming We have the call
> talk time of 10 mins and the call contains 1 minutes of silence. The
> recording file size is around 9 minutes and 30 seconds i.e. it
> contains 30 seconds of silence but also skipped for about 30 sec.
> I haven't configured the "silence_hits" value for the record_session.
> I am unable to find the default value of silence_hit associated with
> record_session or it may be due to some other reason.
>
> Thanks & Regards,
> Shahnawaz


From brian at freeswitch.com  Wed Oct 14 20:31:34 2020
From: brian at freeswitch.com (Brian West)
Date: Wed, 14 Oct 2020 15:31:34 -0500
Subject: [Freeswitch-users] $500 bounty RTMP to conference room
In-Reply-To: <CAHubkyOv-c8DG-3aQHN87a+_QU0o3vsSQY-wD_SmhhF7O6oCYA@mail.gmail.com>
References: <CAHubkyNPZ4Rc=cZCy7co1U_1vbOLOGm41ghaHXyJcG4sW=OF8g@mail.gmail.com>
 <5EFA741A-5934-46DD-98D5-9370E039BF4B@freeswitch.org>
 <CAHubkyOv-c8DG-3aQHN87a+_QU0o3vsSQY-wD_SmhhF7O6oCYA@mail.gmail.com>
Message-ID: <CAEJMVkAtHChwf=YKrZMRO5yBp47e1i5+10i_F6jTOw14a-qbjg@mail.gmail.com>

conference play rtmp://blah?

On Wed, Oct 14, 2020 at 2:13 PM Nathan Stratton <nathan at robotics.net> wrote:

> Got it, I can't find much on mod_av and nothing on how to have mod_av
> accept rtmp stream and add it to a conference. Not a big fan of RTMP, but
> that is what slingstudio supports.
>
> ><>
> nathan stratton
>
>
> On Wed, Oct 14, 2020 at 12:57 PM Mike Jerris <mike at freeswitch.org> wrote:
>
>> Use mod_av to record/playback rtmp://
>>
>> mod_rtmp is an endpoint module. Its a different animal that really has no
>> use now that webrtc is available almost everywhere.
>>
>> On Oct 14, 2020, at 11:57 AM, Nathan Stratton <nathan at robotics.net>
>> wrote:
>>
>>
>> Trying to broadcast from vlc or a sling box with rtmp to freeswitch
>> conference room. I have mod_rtmp installed and why I try to steam from vlc
>> I get:
>>
>> 2020-10-14 11:56:44.516563 [NOTICE] mod_rtmp.c:905 New RTMP session
>> [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
>> 2020-10-14 11:56:44.516563 [INFO] rtmp_tcp.c:234 Rtmp connection from
>> 50.210.153.253:36864
>> 2020-10-14 11:56:44.516563 [DEBUG] rtmp.c:882 Sent handshake response
>> 2020-10-14 11:56:44.536531 [DEBUG] rtmp.c:907 Done with handshake
>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
>> ts=0 stream_id=0x0] len=155
>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>> for connect
>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x1
>> stream_id=0x0] len=4
>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x5
>> stream_id=0x0] len=4
>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x6
>> stream_id=0x0] len=5
>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4
>> stream_id=0x0] len=6
>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
>> stream_id=0x0] len=201
>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
>> stream_id=0x0] len=61
>> 2020-10-14 11:56:44.556571 [NOTICE] rtmp_sig.c:122 Sent connect reply
>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=2 type=0x1
>> ts=0 stream_id=0x0] len=4
>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1089 SET CHUNKSIZE=512
>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
>> ts=0 stream_id=0x0] len=29
>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>> for releaseStream
>> 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for
>> "releaseStream"
>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
>> ts=0 stream_id=0x0] len=25
>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>> for FCPublish
>> 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for
>> "FCPublish"
>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
>> ts=0 stream_id=0x0] len=25
>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>> for createStream
>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
>> stream_id=0x0] len=29
>> 2020-10-14 11:56:44.636529 [INFO] rtmp_sig.c:137 Replied to createStream
>> (0)
>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:1081 [chunk_stream=8 type=0x14
>> ts=0 stream_id=0x1] len=30
>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:192 [amfnumber=8] Got INVOKE
>> for publish
>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4
>> stream_id=0x0] len=6
>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=5 type=0x14
>> stream_id=0x1] len=138
>> 2020-10-14 11:56:44.696580 [INFO] rtmp_sig.c:290 Got publish on stream 1.
>> 2020-10-14 11:56:45.216562 [DEBUG] rtmp.c:1081 [chunk_stream=4 type=0x12
>> ts=0 stream_id=0x1] len=319
>> 2020-10-14 11:56:45.656516 [DEBUG] rtmp_tcp.c:243 Closing socket
>> 2020-10-14 11:56:51.796530 [NOTICE] mod_rtmp.c:1027 RTMP session ended
>> [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
>>
>> Anyone able to help me? $500 via Paypal.
>>
>> ><>
>> nathan stratto
>>
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>

Email: brian at freeswitch.com

Mobile: 918-424-9378

Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>

[image: https://www.facebook.com/signalwireinc?src=email]
<https://www.facebook.com/freeswitch> [image:
https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
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From s.safarov at gmail.com  Wed Oct 14 20:33:24 2020
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 14 Oct 2020 23:33:24 +0300
Subject: [Freeswitch-users] $500 bounty RTMP to conference room
In-Reply-To: <CAHubkyOv-c8DG-3aQHN87a+_QU0o3vsSQY-wD_SmhhF7O6oCYA@mail.gmail.com>
References: <CAHubkyNPZ4Rc=cZCy7co1U_1vbOLOGm41ghaHXyJcG4sW=OF8g@mail.gmail.com>
 <5EFA741A-5934-46DD-98D5-9370E039BF4B@freeswitch.org>
 <CAHubkyOv-c8DG-3aQHN87a+_QU0o3vsSQY-wD_SmhhF7O6oCYA@mail.gmail.com>
Message-ID: <CAHtxdDeszrOj4yyp6Bz6R4UZNfJ7U3W_MUP0EZyY=51JApSfeg@mail.gmail.com>

You can try mod_vlc.
I have tested mod_vlc several months ago and found some functions do not
work as expected.

But you can try.


On Wed, Oct 14, 2020 at 9:57 PM Nathan Stratton <nathan at robotics.net> wrote:

> Got it, I can't find much on mod_av and nothing on how to have mod_av
> accept rtmp stream and add it to a conference. Not a big fan of RTMP, but
> that is what slingstudio supports.
>
> ><>
> nathan stratton
>
>
> On Wed, Oct 14, 2020 at 12:57 PM Mike Jerris <mike at freeswitch.org> wrote:
>
>> Use mod_av to record/playback rtmp://
>>
>> mod_rtmp is an endpoint module. Its a different animal that really has no
>> use now that webrtc is available almost everywhere.
>>
>> On Oct 14, 2020, at 11:57 AM, Nathan Stratton <nathan at robotics.net>
>> wrote:
>>
>>
>> Trying to broadcast from vlc or a sling box with rtmp to freeswitch
>> conference room. I have mod_rtmp installed and why I try to steam from vlc
>> I get:
>>
>> 2020-10-14 11:56:44.516563 [NOTICE] mod_rtmp.c:905 New RTMP session
>> [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
>> 2020-10-14 11:56:44.516563 [INFO] rtmp_tcp.c:234 Rtmp connection from
>> 50.210.153.253:36864
>> 2020-10-14 11:56:44.516563 [DEBUG] rtmp.c:882 Sent handshake response
>> 2020-10-14 11:56:44.536531 [DEBUG] rtmp.c:907 Done with handshake
>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
>> ts=0 stream_id=0x0] len=155
>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>> for connect
>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x1
>> stream_id=0x0] len=4
>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x5
>> stream_id=0x0] len=4
>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x6
>> stream_id=0x0] len=5
>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4
>> stream_id=0x0] len=6
>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
>> stream_id=0x0] len=201
>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
>> stream_id=0x0] len=61
>> 2020-10-14 11:56:44.556571 [NOTICE] rtmp_sig.c:122 Sent connect reply
>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=2 type=0x1
>> ts=0 stream_id=0x0] len=4
>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1089 SET CHUNKSIZE=512
>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
>> ts=0 stream_id=0x0] len=29
>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>> for releaseStream
>> 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for
>> "releaseStream"
>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
>> ts=0 stream_id=0x0] len=25
>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>> for FCPublish
>> 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for
>> "FCPublish"
>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
>> ts=0 stream_id=0x0] len=25
>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>> for createStream
>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
>> stream_id=0x0] len=29
>> 2020-10-14 11:56:44.636529 [INFO] rtmp_sig.c:137 Replied to createStream
>> (0)
>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:1081 [chunk_stream=8 type=0x14
>> ts=0 stream_id=0x1] len=30
>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:192 [amfnumber=8] Got INVOKE
>> for publish
>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4
>> stream_id=0x0] len=6
>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=5 type=0x14
>> stream_id=0x1] len=138
>> 2020-10-14 11:56:44.696580 [INFO] rtmp_sig.c:290 Got publish on stream 1.
>> 2020-10-14 11:56:45.216562 [DEBUG] rtmp.c:1081 [chunk_stream=4 type=0x12
>> ts=0 stream_id=0x1] len=319
>> 2020-10-14 11:56:45.656516 [DEBUG] rtmp_tcp.c:243 Closing socket
>> 2020-10-14 11:56:51.796530 [NOTICE] mod_rtmp.c:1027 RTMP session ended
>> [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
>>
>> Anyone able to help me? $500 via Paypal.
>>
>> ><>
>> nathan stratto
>>
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From krice at freeswitch.org  Wed Oct 14 21:30:52 2020
From: krice at freeswitch.org (krice at freeswitch.org)
Date: Wed, 14 Oct 2020 16:30:52 -0500
Subject: [Freeswitch-users] $500 bounty RTMP to conference room
In-Reply-To: <CAHubkyOv-c8DG-3aQHN87a+_QU0o3vsSQY-wD_SmhhF7O6oCYA@mail.gmail.com>
References: <CAHubkyNPZ4Rc=cZCy7co1U_1vbOLOGm41ghaHXyJcG4sW=OF8g@mail.gmail.com>
 <5EFA741A-5934-46DD-98D5-9370E039BF4B@freeswitch.org>
 <CAHubkyOv-c8DG-3aQHN87a+_QU0o3vsSQY-wD_SmhhF7O6oCYA@mail.gmail.com>
Message-ID: <009101d6a271$4c1161a0$e43424e0$@freeswitch.org>

Conference CONFNAME play rtmp://whatever/stream/name 

just have mod_av installed also you can conference record rtmp://… too broadcast the conf room out 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Nathan Stratton
Sent: Wednesday, October 14, 2020 1:16 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] $500 bounty RTMP to conference room

 

Got it, I can't find much on mod_av and nothing on how to have mod_av accept rtmp stream and add it to a conference. Not a big fan of RTMP, but that is what slingstudio supports.



><>
nathan stratton

 

 

On Wed, Oct 14, 2020 at 12:57 PM Mike Jerris <mike at freeswitch.org <mailto:mike at freeswitch.org> > wrote:

Use mod_av to record/playback rtmp://

 

mod_rtmp is an endpoint module. Its a different animal that really has no use now that webrtc is available almost everywhere.





On Oct 14, 2020, at 11:57 AM, Nathan Stratton <nathan at robotics.net <mailto:nathan at robotics.net> > wrote:

 




Trying to broadcast from vlc or a sling box with rtmp to freeswitch conference room. I have mod_rtmp installed and why I try to steam from vlc I get:

 

2020-10-14 11:56:44.516563 [NOTICE] mod_rtmp.c:905 New RTMP session [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
2020-10-14 11:56:44.516563 [INFO] rtmp_tcp.c:234 Rtmp connection from 50.210.153.253:36864 <http://50.210.153.253:36864/> 
2020-10-14 11:56:44.516563 [DEBUG] rtmp.c:882 Sent handshake response
2020-10-14 11:56:44.536531 [DEBUG] rtmp.c:907 Done with handshake
2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=155
2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for connect
2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x1 stream_id=0x0] len=4 
2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x5 stream_id=0x0] len=4 
2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x6 stream_id=0x0] len=5 
2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4 stream_id=0x0] len=6 
2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14 stream_id=0x0] len=201 
2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14 stream_id=0x0] len=61 
2020-10-14 11:56:44.556571 [NOTICE] rtmp_sig.c:122 Sent connect reply
2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=2 type=0x1 ts=0 stream_id=0x0] len=4
2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1089 SET CHUNKSIZE=512
2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=29
2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for releaseStream
2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for "releaseStream"
2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=25
2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for FCPublish
2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for "FCPublish"
2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=25
2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for createStream
2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14 stream_id=0x0] len=29 
2020-10-14 11:56:44.636529 [INFO] rtmp_sig.c:137 Replied to createStream (0)
2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:1081 [chunk_stream=8 type=0x14 ts=0 stream_id=0x1] len=30
2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:192 [amfnumber=8] Got INVOKE for publish
2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4 stream_id=0x0] len=6 
2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=5 type=0x14 stream_id=0x1] len=138 
2020-10-14 11:56:44.696580 [INFO] rtmp_sig.c:290 Got publish on stream 1.
2020-10-14 11:56:45.216562 [DEBUG] rtmp.c:1081 [chunk_stream=4 type=0x12 ts=0 stream_id=0x1] len=319
2020-10-14 11:56:45.656516 [DEBUG] rtmp_tcp.c:243 Closing socket
2020-10-14 11:56:51.796530 [NOTICE] mod_rtmp.c:1027 RTMP session ended [88e00c1b-6b3a-4bac-83f5-bc12b879884b]

 

Anyone able to help me? $500 via Paypal. 


><>
nathan stratto

 

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales at freeswitch.com <mailto:sales at freeswitch.com> 
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org> 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com

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From davidswalkabout at gmail.com  Thu Oct 15 02:39:33 2020
From: davidswalkabout at gmail.com (David P)
Date: Thu, 15 Oct 2020 15:39:33 +1300
Subject: [Freeswitch-users] action application="capture_video"
Message-ID: <CAE23trv6tedC9DybWM-uDU3U+n7umzEU78B8m_8WSgx-S0RBtA@mail.gmail.com>

I just came across this intriguing snippet:

<action application="capture_video"
data="#transcode{vcodec=h264,acodec=mpga}:std{access=file,mux=mp4,dst=/tmp/1.mp4}"/>

I can't find any documentation for capture_video; does it allow setting the
sample rate? Can it be used more than once in a dialplan to create an mp4
at the conference rate and a wav at a higher rate?
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From davidswalkabout at gmail.com  Thu Oct 15 02:42:41 2020
From: davidswalkabout at gmail.com (David P)
Date: Thu, 15 Oct 2020 15:42:41 +1300
Subject: [Freeswitch-users] Streaming out of conference via rtmp
Message-ID: <CAE23trsgMAoEY7hBi+5z+hvY-fN0WRgbAFhBe165QsGgdfA60Q@mail.gmail.com>

In response to the thread " $500 bounty RTMP to conference room", try
something like this:

<action application="record_session"
data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
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From svanherwaarden at precisionag.org  Thu Oct 15 04:28:36 2020
From: svanherwaarden at precisionag.org (Sam van Herwaarden)
Date: Thu, 15 Oct 2020 06:28:36 +0200
Subject: [Freeswitch-users] IVR UI/UX design
In-Reply-To: <0100017527996013-d3cca8fd-f55a-41e1-948b-37512abc7fef-000000@email.amazonses.com>
References: <CAAtWTekEPFw4AxxigLfd0d9nzMrW+1-WFFT5LUvd3u2Fo_VNYg@mail.gmail.com>
 <009901d69e3f$3943a0f0$abcae2d0$@gmail.com>
 <em36deb13a-aa38-4bb3-a5e6-f03845a08ea2@fred-pc>
 <0100017527996013-d3cca8fd-f55a-41e1-948b-37512abc7fef-000000@email.amazonses.com>
Message-ID: <CAAtWTem20xj4se+FN_+nYvf1=sF1588PWeKN2KSO3OcPaS4jTg@mail.gmail.com>

Thanks, all, for the useful input!

On Wed, Oct 14, 2020 at 4:53 PM Avi Marcus <avi at avimarcus.net> wrote:

> As an example: when I call a place for the first time, I have to hear the
> whole menu to see which option sounds closest to what I need.
> Of course by that time I've forgotten which digit it is, and I need to
> listen again.
>
> On Fri, Oct 9, 2020, 5:12 PM Frederic Jean <makafre at gmail.com> wrote:
>
>>
>> - The more options in a menu the more chance your caller will need to
>> have a menu repeated;
>> - Always provide a key to repeat the menu and optimize the waiting time
>> for key input so that callers have enough time;
>>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From nathan at robotics.net  Thu Oct 15 13:37:23 2020
From: nathan at robotics.net (Nathan Stratton)
Date: Thu, 15 Oct 2020 09:37:23 -0400
Subject: [Freeswitch-users] $500 bounty RTMP to conference room
In-Reply-To: <009101d6a271$4c1161a0$e43424e0$@freeswitch.org>
References: <CAHubkyNPZ4Rc=cZCy7co1U_1vbOLOGm41ghaHXyJcG4sW=OF8g@mail.gmail.com>
 <5EFA741A-5934-46DD-98D5-9370E039BF4B@freeswitch.org>
 <CAHubkyOv-c8DG-3aQHN87a+_QU0o3vsSQY-wD_SmhhF7O6oCYA@mail.gmail.com>
 <009101d6a271$4c1161a0$e43424e0$@freeswitch.org>
Message-ID: <CAHubkyODvjWv6+RA1gV0U8V7+VthGpE_P0PzfFeemaMYxzZcfQ@mail.gmail.com>

On Wed, Oct 14, 2020 at 5:31 PM <krice at freeswitch.org> wrote:

> Conference CONFNAME play rtmp://whatever/stream/name
>
> just have mod_av installed also you can conference record rtmp://… too
> broadcast the conf room out
>

 Hmm, and that will have it listen for RTPM from a source broadcasted to
that IP? I am getting file not found, what I need is for it to listen for
RTPM from the broadcast side that is being sent to 10.0.0.51.


freeswitch at troc.staging.vocinity.com> conference 4600 play rtmp://10.0.0.51/
-ERR (play) File: rtmp://10.0.0.51/ not found.

2020-10-15 09:36:24.876534 [DEBUG] avformat.c:1665 sample rate: 8000,
channels: 1
2020-10-15 09:36:24.876534 [WARNING] avformat.c:1116 Could not open input
file '10.0.0.51/' (error 'No such file or directory')
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From nathan at robotics.net  Thu Oct 15 13:41:15 2020
From: nathan at robotics.net (Nathan Stratton)
Date: Thu, 15 Oct 2020 09:41:15 -0400
Subject: [Freeswitch-users] Streaming out of conference via rtmp
In-Reply-To: <CAE23trsgMAoEY7hBi+5z+hvY-fN0WRgbAFhBe165QsGgdfA60Q@mail.gmail.com>
References: <CAE23trsgMAoEY7hBi+5z+hvY-fN0WRgbAFhBe165QsGgdfA60Q@mail.gmail.com>
Message-ID: <CAHubkyNoJVBr=3vJmJ5L=o1e8MfA+_emh3suF4+pb+gvNgU9aw@mail.gmail.com>

On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout at gmail.com> wrote:

> In response to the thread " $500 bounty RTMP to conference room", try
> something like this:
>
> <action application="record_session"
> data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
>

So I am not trying to send rtmp, I am trying to do I think what mod_rtmp
was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server.

-Nathan
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From nathan at robotics.net  Thu Oct 15 13:43:27 2020
From: nathan at robotics.net (Nathan Stratton)
Date: Thu, 15 Oct 2020 09:43:27 -0400
Subject: [Freeswitch-users] $500 bounty RTMP to conference room
In-Reply-To: <5EFA741A-5934-46DD-98D5-9370E039BF4B@freeswitch.org>
References: <CAHubkyNPZ4Rc=cZCy7co1U_1vbOLOGm41ghaHXyJcG4sW=OF8g@mail.gmail.com>
 <5EFA741A-5934-46DD-98D5-9370E039BF4B@freeswitch.org>
Message-ID: <CAHubkyP0+zVUPP1EXDcYSx6hG8fYAex_XXkRhnEew8_B_2C93A@mail.gmail.com>

On Wed, Oct 14, 2020 at 12:57 PM Mike Jerris <mike at freeswitch.org> wrote:

> Use mod_av to record/playback rtmp://
>
> mod_rtmp is an endpoint module. Its a different animal that really has no
> use now that webrtc is available almost everywhere
>

I was trying it because it looked like it was the only thing that would
listen for an RTMP stream on FreeSWITCH.
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From nneul at mst.edu  Thu Oct 15 14:03:47 2020
From: nneul at mst.edu (Nathan Neulinger)
Date: Thu, 15 Oct 2020 09:03:47 -0500
Subject: [Freeswitch-users] quality/stability of freeswitch under docker
Message-ID: <9fc9aca3-7cf0-7999-a085-b2777b6cb6a1@mst.edu>

I've got a production environment that's been running w/ a source build on ubuntu 16.04 for several years. I'm in the 
process of upgrading this to ubuntu 20, which also appears to be working fine. Setup is dual hosts with keepalived for 
failover, pointing to a separate 3-node mariadb based galera cluster.

However, during this process I've been considering whether it would be a bit cleaner to move to a docker based 
deployment to have freeswitch inside the container use the more fully "supported" debian packages.

Anyone have any thoughts on the general stability/performance/etc. of running FS under docker vs source build? Any 
differences? Main areas of concern I have are the network interactions, and I was thinking about using the 'host 
network' mode to address that.


I'm looking at using https://github.com/signalwire/freeswitch/blob/master/docker/master/Dockerfile modified to reference 
buster instead of jesse. (Can't see doing an updated deployment on 8 if 10 has production-ready packages.)


Any feedback/experiences/etc. would be appreciated.


-- Nathan
------------------------------------------------------------
Nathan Neulinger                            nneul at mst.edu
Missouri S&T Information Technology         (573) 341-6679
System Administrator - Architect            (573) 612-1412
System and Desktop Infrastructure Team Manager

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From david.villasmil.work at gmail.com  Thu Oct 15 14:42:31 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 15 Oct 2020 15:42:31 +0100
Subject: [Freeswitch-users] Streaming out of conference via rtmp
In-Reply-To: <CAHubkyNoJVBr=3vJmJ5L=o1e8MfA+_emh3suF4+pb+gvNgU9aw@mail.gmail.com>
References: <CAE23trsgMAoEY7hBi+5z+hvY-fN0WRgbAFhBe165QsGgdfA60Q@mail.gmail.com>
 <CAHubkyNoJVBr=3vJmJ5L=o1e8MfA+_emh3suF4+pb+gvNgU9aw@mail.gmail.com>
Message-ID: <CAFGRPVpxtmi+Hq3iDK+CpLN-GvUPky6-M_VbxXEMJB96ozuC0w@mail.gmail.com>

What are you trying to accomplish exactly?

On Thu, 15 Oct 2020 at 15:13, Nathan Stratton <nathan at robotics.net> wrote:

> On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout at gmail.com>
> wrote:
>
>> In response to the thread "
>>
>>
>>
>> $500 bounty RTMP to conference room", try something like this:
>>
>> <action application="record_session"
>> data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
>>
>
> So I am not trying to send rtmp, I am trying to do I think what mod_rtmp
> was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server.
>
> -Nathan
>
>
> _________________________________________________________________________
>
>
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
>
> Build your next product on our scalable cloud platform.
>
>
>
> Join our online community to chat in real time
> https://signalwire.community
>
>
>
> Professional FreeSWITCH Services
>
> sales at freeswitch.com
>
> https://freeswitch.com
>
>
>
> Official FreeSWITCH Sites
>
> https://freeswitch.com/oss
>
> https://freeswitch.org/confluence
>
> https://cluecon.com
>
>
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From nathan at robotics.net  Thu Oct 15 14:57:25 2020
From: nathan at robotics.net (Nathan Stratton)
Date: Thu, 15 Oct 2020 10:57:25 -0400
Subject: [Freeswitch-users] Streaming out of conference via rtmp
In-Reply-To: <CAFGRPVpxtmi+Hq3iDK+CpLN-GvUPky6-M_VbxXEMJB96ozuC0w@mail.gmail.com>
References: <CAE23trsgMAoEY7hBi+5z+hvY-fN0WRgbAFhBe165QsGgdfA60Q@mail.gmail.com>
 <CAHubkyNoJVBr=3vJmJ5L=o1e8MfA+_emh3suF4+pb+gvNgU9aw@mail.gmail.com>
 <CAFGRPVpxtmi+Hq3iDK+CpLN-GvUPky6-M_VbxXEMJB96ozuC0w@mail.gmail.com>
Message-ID: <CAHubkyNuCHUfLZARNWM25=YbyRuUnXyvyCGAFVeGWqW3WahEDQ@mail.gmail.com>

Sure, so we have a number of users that dial into a room, say 4600, they
receive audio and video from the moderator but don't send anything
themselves because of mute|vmute. My customer wants to instead of calling
in as a moderator, to instead send content to the room via SlingStudio like
they do with Facebook and Youtube. Sling Studio sends a stream as
rtmp://{my ip}/{room} I was hoping I could do that with mod_rtmp but was
not able to make it work. So I am looking for another way to send video
from the RTMP server to the FreeSWITCH room. I have found SIP to RTMP
software, but I am trying to do the opposite, I need to get from RTMP to
SIP room.

  <extension name="Simple add-caller conf: member">
    <condition field="destination_number" expression="^(46\d{2})$">
      <action application="answer"/>
      <action application="conference" data="$1 at default+flags{mute|vmute}"/>
    </condition>
  </extension>

  <extension name="Simple add-caller conf: moderator">
     <condition field="destination_number" expression="^\*(46\d{2}$)">
       <action application="answer"/>
       <action application="digit_action_set_realm" data="moderator"/>
       <action application="conference" data="$1 at simple+flags{moderator}"/>
     </condition>
  </extension>

><>
nathan stratton


On Thu, Oct 15, 2020 at 10:43 AM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> What are you trying to accomplish exactly?
>
> On Thu, 15 Oct 2020 at 15:13, Nathan Stratton <nathan at robotics.net> wrote:
>
>> On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout at gmail.com>
>> wrote:
>>
>>> In response to the thread "
>>>
>>>
>>>
>>> $500 bounty RTMP to conference room", try something like this:
>>>
>>> <action application="record_session"
>>> data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
>>>
>>
>> So I am not trying to send rtmp, I am trying to do I think what mod_rtmp
>> was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server.
>>
>> -Nathan
>>
>>
>> _________________________________________________________________________
>>
>>
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>>
>> Build your next product on our scalable cloud platform.
>>
>>
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>>
>>
>> Professional FreeSWITCH Services
>>
>> sales at freeswitch.com
>>
>> https://freeswitch.com
>>
>>
>>
>> Official FreeSWITCH Sites
>>
>> https://freeswitch.com/oss
>>
>> https://freeswitch.org/confluence
>>
>> https://cluecon.com
>>
>>
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> https://freeswitch.com
>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From david.villasmil.work at gmail.com  Thu Oct 15 15:44:14 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 15 Oct 2020 16:44:14 +0100
Subject: [Freeswitch-users] Streaming out of conference via rtmp
In-Reply-To: <CAHubkyNuCHUfLZARNWM25=YbyRuUnXyvyCGAFVeGWqW3WahEDQ@mail.gmail.com>
References: <CAE23trsgMAoEY7hBi+5z+hvY-fN0WRgbAFhBe165QsGgdfA60Q@mail.gmail.com>
 <CAHubkyNoJVBr=3vJmJ5L=o1e8MfA+_emh3suF4+pb+gvNgU9aw@mail.gmail.com>
 <CAFGRPVpxtmi+Hq3iDK+CpLN-GvUPky6-M_VbxXEMJB96ozuC0w@mail.gmail.com>
 <CAHubkyNuCHUfLZARNWM25=YbyRuUnXyvyCGAFVeGWqW3WahEDQ@mail.gmail.com>
Message-ID: <CAFGRPVodHT4FKZ7MTZbqgS2ESr=smM9Mo5TWZeWBV8A7B=g8qA@mail.gmail.com>

There’s a mod_vlc, never used it but might be what you’re looking for. Have
you seen it?

On Thu, 15 Oct 2020 at 16:41, Nathan Stratton <nathan at robotics.net> wrote:

> Sure, so we have a number of users that dial into a room, say 4600, they
> receive audio and video from the moderator but don't send anything
> themselves because of mute|vmute. My customer wants to instead of calling
> in as a moderator, to instead send content to the room via SlingStudio like
> they do with Facebook and Youtube. Sling Studio sends a stream as
> rtmp://{my ip}/{room} I was hoping I could do that with mod_rtmp but was
> not able to make it work. So I am looking for another way to send video
> from the RTMP server to the FreeSWITCH room. I have found SIP to RTMP
> software, but I am trying to do the opposite, I need to get from RTMP to
> SIP room.
>
>   <extension name="Simple add-caller conf: member">
>     <condition field="destination_number" expression="^(46\d{2})$">
>       <action application="answer"/>
>       <action application="conference" data="$1 at default
> +flags{mute|vmute}"/>
>     </condition>
>   </extension>
>
>   <extension name="Simple add-caller conf: moderator">
>      <condition field="destination_number" expression="^\*(46\d{2}$)">
>        <action application="answer"/>
>        <action application="digit_action_set_realm" data="moderator"/>
>        <action application="conference" data="$1 at simple
> +flags{moderator}"/>
>      </condition>
>   </extension>
>
> ><>
>
> nathan stratton
>
>
> On Thu, Oct 15, 2020 at 10:43 AM David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> What are you trying to accomplish exactly?
>>
>> On Thu, 15 Oct 2020 at 15:13, Nathan Stratton <nathan at robotics.net>
>> wrote:
>>
>>> On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout at gmail.com>
>>> wrote:
>>>
>>>> In response to the thread "
>>>>
>>>>
>>>>
>>>> $500 bounty RTMP to conference room", try something like this:
>>>>
>>>> <action application="record_session"
>>>> data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
>>>>
>>>
>>> So I am not trying to send rtmp, I am trying to do I think what mod_rtmp
>>> was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server.
>>>
>>> -Nathan
>>>
>>>
>>> _________________________________________________________________________
>>>
>>>
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>>
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>>
>>> Build your next product on our scalable cloud platform.
>>>
>>>
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>>
>>>
>>> Professional FreeSWITCH Services
>>>
>>> sales at freeswitch.com
>>>
>>> https://freeswitch.com
>>>
>>>
>>>
>>> Official FreeSWITCH Sites
>>>
>>> https://freeswitch.com/oss
>>>
>>> https://freeswitch.org/confluence
>>>
>>> https://cluecon.com
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> https://freeswitch.com
>>
>> --
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>>
>>
>> _________________________________________________________________________
>>
>>
>>
>>
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>
>>
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>>
>>
>> Build your next product on our scalable cloud platform.
>>
>>
>>
>>
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>>
>>
>>
>>
>> Professional FreeSWITCH Services
>>
>>
>> sales at freeswitch.com
>>
>>
>> https://freeswitch.com
>>
>>
>>
>>
>>
>> Official FreeSWITCH Sites
>>
>>
>> https://freeswitch.com/oss
>>
>>
>> https://freeswitch.org/confluence
>>
>>
>> https://cluecon.com
>>
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>>
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>>
>> https://freeswitch.com
>
>
>
> _________________________________________________________________________
>
>
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
>
> Build your next product on our scalable cloud platform.
>
>
>
> Join our online community to chat in real time
> https://signalwire.community
>
>
>
> Professional FreeSWITCH Services
>
> sales at freeswitch.com
>
> https://freeswitch.com
>
>
>
> Official FreeSWITCH Sites
>
> https://freeswitch.com/oss
>
> https://freeswitch.org/confluence
>
> https://cluecon.com
>
>
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From sagarmalam at gmail.com  Thu Oct 15 16:12:07 2020
From: sagarmalam at gmail.com (sagar malam)
Date: Thu, 15 Oct 2020 21:42:07 +0530
Subject: [Freeswitch-users] Streaming out of conference via rtmp
In-Reply-To: <CAE23trsgMAoEY7hBi+5z+hvY-fN0WRgbAFhBe165QsGgdfA60Q@mail.gmail.com>
References: <CAE23trsgMAoEY7hBi+5z+hvY-fN0WRgbAFhBe165QsGgdfA60Q@mail.gmail.com>
Message-ID: <CAEZuF3srA6LECqZFntLmh-6eBv85ubuwrxGLw8cyDZRvfsoxCQ@mail.gmail.com>

Hi
I can help u
Please confirm if bounty is still open

On Thu, 15 Oct 2020, 8:59 am David P, <davidswalkabout at gmail.com> wrote:

> In response to the thread " $500 bounty RTMP to conference room", try
> something like this:
>
> <action application="record_session"
> data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From jurijs.ivolga at gmail.com  Thu Oct 15 16:24:36 2020
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Thu, 15 Oct 2020 19:24:36 +0300
Subject: [Freeswitch-users] quality/stability of freeswitch under docker
In-Reply-To: <9fc9aca3-7cf0-7999-a085-b2777b6cb6a1@mst.edu>
References: <9fc9aca3-7cf0-7999-a085-b2777b6cb6a1@mst.edu>
Message-ID: <CAOrA2UZ=i8E0gt5L1COkJd3BJPKPpPAXnTtWN45jVuxOzhRLug@mail.gmail.com>

Hi,


I run in production for several years Freeswitch in docker and had never
any problem with docker environment. I do like better custom bridge for
networking when you can assign separate IP per container, but this is
mainly because of particular requirements. Moving to docker is definitely
worth it.

With kind regards,

On Thu, 15. Oct 2020 at 17:47, Nathan Neulinger <nneul at mst.edu> wrote:

> I've got a production environment that's been running w/ a source build on
> ubuntu 16.04 for several years. I'm in the process of upgrading this to
> ubuntu 20, which also appears to be working fine. Setup is dual hosts with
> keepalived for failover, pointing to a separate 3-node mariadb based galera
> cluster.
>
> However, during this process I've been considering whether it would be a
> bit cleaner to move to a docker based deployment to have freeswitch inside
> the container use the more fully "supported" debian packages.
>
> Anyone have any thoughts on the general stability/performance/etc. of
> running FS under docker vs source build? Any differences? Main areas of
> concern I have are the network interactions, and I was thinking about using
> the 'host network' mode to address that.
>
>
> I'm looking at using
> https://github.com/signalwire/freeswitch/blob/master/docker/master/Dockerfile
> modified to reference buster instead of jesse. (Can't see doing an updated
> deployment on 8 if 10 has production-ready packages.)
>
>
> Any feedback/experiences/etc. would be appreciated.
>
>
> -- Nathan
> ------------------------------------------------------------
> Nathan Neulinger                            nneul at mst.edu
> Missouri S&T Information Technology         (573) 341-6679
> System Administrator - Architect            (573) 612-1412
> System and Desktop Infrastructure Team Manager
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Jurijs
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From krice at freeswitch.org  Thu Oct 15 16:31:33 2020
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 15 Oct 2020 11:31:33 -0500
Subject: [Freeswitch-users] Streaming out of conference via rtmp
In-Reply-To: <CAHubkyNoJVBr=3vJmJ5L=o1e8MfA+_emh3suF4+pb+gvNgU9aw@mail.gmail.com>
References: <CAHubkyNoJVBr=3vJmJ5L=o1e8MfA+_emh3suF4+pb+gvNgU9aw@mail.gmail.com>
Message-ID: <983C2779-D99D-4858-9E1A-6CDC079762E2@freeswitch.org>

Mod_rtmp was intended as a softphone endpoint written in flash. dont use it. litterally load mod_av then via esl, fs_cli etc  “conference <confname> [play|record]  rtmp://{rest/of/url}”

its really that simple. play will pkay the rtmp stream into the conference like any other media file. record will shoot a media stream out to what ever rtmp reflector you are using auch as youtube or facebook or your own private rtmp server.

Sent from my iPhone

> On Oct 15, 2020, at 08:42, Nathan Stratton <nathan at robotics.net> wrote:
> 
> 
>> On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout at gmail.com> wrote:
> 
>> In response to the thread " $500 bounty RTMP to conference room", try something like this:
>>  
>> <action application="record_session" data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
> 
> So I am not trying to send rtmp, I am trying to do I think what mod_rtmp was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server. 
> 
> -Nathan
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From nathan at robotics.net  Thu Oct 15 16:41:55 2020
From: nathan at robotics.net (Nathan Stratton)
Date: Thu, 15 Oct 2020 12:41:55 -0400
Subject: [Freeswitch-users] Streaming out of conference via rtmp
In-Reply-To: <983C2779-D99D-4858-9E1A-6CDC079762E2@freeswitch.org>
References: <CAHubkyNoJVBr=3vJmJ5L=o1e8MfA+_emh3suF4+pb+gvNgU9aw@mail.gmail.com>
 <983C2779-D99D-4858-9E1A-6CDC079762E2@freeswitch.org>
Message-ID: <CAHubkyNcFfWq_CzXER-bxn3-4CCRR5hkEo-FBpKpEJ6Jtm6=WA@mail.gmail.com>

Thanks, so yes, I want to play not record, but when I try to play it's
looking for a file, it's not listening on port 1935 for a RTMP stream from
the server. I need to ACCEPT a RTMP steam like mod_rtmp did, I understand
why mod_rtmp is not working but is there any other way to listen from RTMP
on port 1935 and send it into a conference?


freeswitch at troc.staging.vocinity.com> conference 4600 play rtmp://localhost
-ERR (play) File: rtmp://localhost not found.

2020-10-15 12:38:12.636530 [ERR] avformat.c:1603 Invalid Format


So the issue is that play rtmp:// does not listen for an incoming RTMP
stream.
><>
nathan stratton


On Thu, Oct 15, 2020 at 12:32 PM Ken Rice <krice at freeswitch.org> wrote:

> Mod_rtmp was intended as a softphone endpoint written in flash. dont use
> it. litterally load mod_av then via esl, fs_cli etc  “conference <confname>
> [play|record]  rtmp://{rest/of/url}”
>
> its really that simple. play will pkay the rtmp stream into the conference
> like any other media file. record will shoot a media stream out to what
> ever rtmp reflector you are using auch as youtube or facebook or your own
> private rtmp server.
>
> Sent from my iPhone
>
> On Oct 15, 2020, at 08:42, Nathan Stratton <nathan at robotics.net> wrote:
>
> 
> On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout at gmail.com>
> wrote:
>
>> In response to the thread " $500 bounty RTMP to conference room", try
>> something like this:
>>
>> <action application="record_session"
>> data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
>>
>
> So I am not trying to send rtmp, I am trying to do I think what mod_rtmp
> was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server.
>
> -Nathan
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From s.safarov at gmail.com  Thu Oct 15 19:01:26 2020
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 15 Oct 2020 22:01:26 +0300
Subject: [Freeswitch-users] quality/stability of freeswitch under docker
In-Reply-To: <CAOrA2UZ=i8E0gt5L1COkJd3BJPKPpPAXnTtWN45jVuxOzhRLug@mail.gmail.com>
References: <9fc9aca3-7cf0-7999-a085-b2777b6cb6a1@mst.edu>
 <CAOrA2UZ=i8E0gt5L1COkJd3BJPKPpPAXnTtWN45jVuxOzhRLug@mail.gmail.com>
Message-ID: <CAHtxdDfbiiD553kT5stRQWp8Ej3tQnv1293gSLr+Ong3twvPvQ@mail.gmail.com>

required you bridge network, or other network driver without NAT.
Without docker NAT is mandatory.

Also, I have issues with randomly not generated ACK messages on v1.10
branch.
So now I use 1.8.5 in docker - no issues.

On Thu, Oct 15, 2020 at 7:55 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
wrote:

> Hi,
>
>
> I run in production for several years Freeswitch in docker and had never
> any problem with docker environment. I do like better custom bridge for
> networking when you can assign separate IP per container, but this is
> mainly because of particular requirements. Moving to docker is definitely
> worth it.
>
> With kind regards,
>
> On Thu, 15. Oct 2020 at 17:47, Nathan Neulinger <nneul at mst.edu> wrote:
>
>> I've got a production environment that's been running w/ a source build
>> on ubuntu 16.04 for several years. I'm in the process of upgrading this to
>> ubuntu 20, which also appears to be working fine. Setup is dual hosts with
>> keepalived for failover, pointing to a separate 3-node mariadb based galera
>> cluster.
>>
>> However, during this process I've been considering whether it would be a
>> bit cleaner to move to a docker based deployment to have freeswitch inside
>> the container use the more fully "supported" debian packages.
>>
>> Anyone have any thoughts on the general stability/performance/etc. of
>> running FS under docker vs source build? Any differences? Main areas of
>> concern I have are the network interactions, and I was thinking about using
>> the 'host network' mode to address that.
>>
>>
>> I'm looking at using
>> https://github.com/signalwire/freeswitch/blob/master/docker/master/Dockerfile
>> modified to reference buster instead of jesse. (Can't see doing an updated
>> deployment on 8 if 10 has production-ready packages.)
>>
>>
>> Any feedback/experiences/etc. would be appreciated.
>>
>>
>> -- Nathan
>> ------------------------------------------------------------
>> Nathan Neulinger                            nneul at mst.edu
>> Missouri S&T Information Technology         (573) 341-6679
>> System Administrator - Architect            (573) 612-1412
>> System and Desktop Infrastructure Team Manager
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> --
> Jurijs
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From krice at freeswitch.org  Fri Oct 16 00:43:57 2020
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 15 Oct 2020 19:43:57 -0500
Subject: [Freeswitch-users] Streaming out of conference via rtmp
In-Reply-To: <CAHubkyNcFfWq_CzXER-bxn3-4CCRR5hkEo-FBpKpEJ6Jtm6=WA@mail.gmail.com>
References: <CAHubkyNcFfWq_CzXER-bxn3-4CCRR5hkEo-FBpKpEJ6Jtm6=WA@mail.gmail.com>
Message-ID: <5A4EE73B-2C64-478F-8D20-0E0BFD5F56AD@freeswitch.org>

these is to my knowledge no way to listen for a stream to randomly start up. This would require designing a new module to handle the listening service.

Sent from my iPhone

> On Oct 15, 2020, at 11:43, Nathan Stratton <nathan at robotics.net> wrote:
> 
> 
> Thanks, so yes, I want to play not record, but when I try to play it's looking for a file, it's not listening on port 1935 for a RTMP stream from the server. I need to ACCEPT a RTMP steam like mod_rtmp did, I understand why mod_rtmp is not working but is there any other way to listen from RTMP on port 1935 and send it into a conference? 
> 
> 
> freeswitch at troc.staging.vocinity.com> conference 4600 play rtmp://localhost
> -ERR (play) File: rtmp://localhost not found.
> 
> 2020-10-15 12:38:12.636530 [ERR] avformat.c:1603 Invalid Format
> 
> 
> So the issue is that play rtmp:// does not listen for an incoming RTMP stream.
> ><>
> nathan stratton
> 
> 
>> On Thu, Oct 15, 2020 at 12:32 PM Ken Rice <krice at freeswitch.org> wrote:
>> Mod_rtmp was intended as a softphone endpoint written in flash. dont use it. litterally load mod_av then via esl, fs_cli etc  “conference <confname> [play|record]  rtmp://{rest/of/url}”
>> 
>> its really that simple. play will pkay the rtmp stream into the conference like any other media file. record will shoot a media stream out to what ever rtmp reflector you are using auch as youtube or facebook or your own private rtmp server.
>> 
>> Sent from my iPhone
>> 
>>>> On Oct 15, 2020, at 08:42, Nathan Stratton <nathan at robotics.net> wrote:
>>>> 
>>> 
>>>> On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout at gmail.com> wrote:
>>> 
>>>> In response to the thread " $500 bounty RTMP to conference room", try something like this:
>>>>  
>>>> <action application="record_session" data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
>>> 
>>> So I am not trying to send rtmp, I am trying to do I think what mod_rtmp was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server. 
>>> 
>>> -Nathan
>>> _________________________________________________________________________
>>> 
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>>> Build your next product on our scalable cloud platform.
>>> 
>>> Join our online community to chat in real time https://signalwire.community
>>> 
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>> 
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>> Build your next product on our scalable cloud platform.
>> 
>> Join our online community to chat in real time https://signalwire.community
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From cmrienzo at gmail.com  Fri Oct 16 02:35:28 2020
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Thu, 15 Oct 2020 22:35:28 -0400
Subject: [Freeswitch-users] Streaming out of conference via rtmp
In-Reply-To: <5A4EE73B-2C64-478F-8D20-0E0BFD5F56AD@freeswitch.org>
References: <CAHubkyNcFfWq_CzXER-bxn3-4CCRR5hkEo-FBpKpEJ6Jtm6=WA@mail.gmail.com>
 <5A4EE73B-2C64-478F-8D20-0E0BFD5F56AD@freeswitch.org>
Message-ID: <CAD-qRkjLu0dKb4s80YJMHtN0rOHq0+fX2Tw9A6VQm1wfmC-wNw@mail.gmail.com>

You could probably set up ffmpeg as a server to listen for the stream and
then pull it on FS.

On Thu, Oct 15, 2020 at 9:28 PM Ken Rice <krice at freeswitch.org> wrote:

> these is to my knowledge no way to listen for a stream to randomly start
> up. This would require designing a new module to handle the listening
> service.
>
> Sent from my iPhone
>
> On Oct 15, 2020, at 11:43, Nathan Stratton <nathan at robotics.net> wrote:
>
> 
> Thanks, so yes, I want to play not record, but when I try to play it's
> looking for a file, it's not listening on port 1935 for a RTMP stream from
> the server. I need to ACCEPT a RTMP steam like mod_rtmp did, I understand
> why mod_rtmp is not working but is there any other way to listen from RTMP
> on port 1935 and send it into a conference?
>
>
> freeswitch at troc.staging.vocinity.com> conference 4600 play
> rtmp://localhost
> -ERR (play) File: rtmp://localhost not found.
>
> 2020-10-15 12:38:12.636530 [ERR] avformat.c:1603 Invalid Format
>
>
> So the issue is that play rtmp:// does not listen for an incoming RTMP
> stream.
> ><>
> nathan stratton
>
>
> On Thu, Oct 15, 2020 at 12:32 PM Ken Rice <krice at freeswitch.org> wrote:
>
>> Mod_rtmp was intended as a softphone endpoint written in flash. dont use
>> it. litterally load mod_av then via esl, fs_cli etc  “conference <confname>
>> [play|record]  rtmp://{rest/of/url}”
>>
>> its really that simple. play will pkay the rtmp stream into the
>> conference like any other media file. record will shoot a media stream out
>> to what ever rtmp reflector you are using auch as youtube or facebook or
>> your own private rtmp server.
>>
>> Sent from my iPhone
>>
>> On Oct 15, 2020, at 08:42, Nathan Stratton <nathan at robotics.net> wrote:
>>
>> 
>> On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout at gmail.com>
>> wrote:
>>
>>> In response to the thread " $500 bounty RTMP to conference room", try
>>> something like this:
>>>
>>> <action application="record_session"
>>> data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
>>>
>>
>> So I am not trying to send rtmp, I am trying to do I think what mod_rtmp
>> was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server.
>>
>> -Nathan
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From nathan at robotics.net  Fri Oct 16 09:14:28 2020
From: nathan at robotics.net (Nathan Stratton)
Date: Fri, 16 Oct 2020 05:14:28 -0400
Subject: [Freeswitch-users] Streaming out of conference via rtmp
In-Reply-To: <5A4EE73B-2C64-478F-8D20-0E0BFD5F56AD@freeswitch.org>
References: <CAHubkyNcFfWq_CzXER-bxn3-4CCRR5hkEo-FBpKpEJ6Jtm6=WA@mail.gmail.com>
 <5A4EE73B-2C64-478F-8D20-0E0BFD5F56AD@freeswitch.org>
Message-ID: <CAHubkyNZQyRoiBvDeww41LuxQZWq6V06xCqo1a_G06xBHXYxQA@mail.gmail.com>

You think it's worth modifying mod_rtmp? It seems to at least accept a RTMP
stream.

><>
nathan stratton


On Thu, Oct 15, 2020 at 8:44 PM Ken Rice <krice at freeswitch.org> wrote:

> these is to my knowledge no way to listen for a stream to randomly start
> up. This would require designing a new module to handle the listening
> service.
>
> Sent from my iPhone
>
> On Oct 15, 2020, at 11:43, Nathan Stratton <nathan at robotics.net> wrote:
>
> 
> Thanks, so yes, I want to play not record, but when I try to play it's
> looking for a file, it's not listening on port 1935 for a RTMP stream from
> the server. I need to ACCEPT a RTMP steam like mod_rtmp did, I understand
> why mod_rtmp is not working but is there any other way to listen from RTMP
> on port 1935 and send it into a conference?
>
>
> freeswitch at troc.staging.vocinity.com> conference 4600 play
> rtmp://localhost
> -ERR (play) File: rtmp://localhost not found.
>
> 2020-10-15 12:38:12.636530 [ERR] avformat.c:1603 Invalid Format
>
>
> So the issue is that play rtmp:// does not listen for an incoming RTMP
> stream.
> ><>
> nathan stratton
>
>
> On Thu, Oct 15, 2020 at 12:32 PM Ken Rice <krice at freeswitch.org> wrote:
>
>> Mod_rtmp was intended as a softphone endpoint written in flash. dont use
>> it. litterally load mod_av then via esl, fs_cli etc  “conference <confname>
>> [play|record]  rtmp://{rest/of/url}”
>>
>> its really that simple. play will pkay the rtmp stream into the
>> conference like any other media file. record will shoot a media stream out
>> to what ever rtmp reflector you are using auch as youtube or facebook or
>> your own private rtmp server.
>>
>> Sent from my iPhone
>>
>> On Oct 15, 2020, at 08:42, Nathan Stratton <nathan at robotics.net> wrote:
>>
>> 
>> On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout at gmail.com>
>> wrote:
>>
>>> In response to the thread " $500 bounty RTMP to conference room", try
>>> something like this:
>>>
>>> <action application="record_session"
>>> data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
>>>
>>
>> So I am not trying to send rtmp, I am trying to do I think what mod_rtmp
>> was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server.
>>
>> -Nathan
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From nathan at robotics.net  Fri Oct 16 09:15:11 2020
From: nathan at robotics.net (Nathan Stratton)
Date: Fri, 16 Oct 2020 05:15:11 -0400
Subject: [Freeswitch-users] Streaming out of conference via rtmp
In-Reply-To: <CAD-qRkjLu0dKb4s80YJMHtN0rOHq0+fX2Tw9A6VQm1wfmC-wNw@mail.gmail.com>
References: <CAHubkyNcFfWq_CzXER-bxn3-4CCRR5hkEo-FBpKpEJ6Jtm6=WA@mail.gmail.com>
 <5A4EE73B-2C64-478F-8D20-0E0BFD5F56AD@freeswitch.org>
 <CAD-qRkjLu0dKb4s80YJMHtN0rOHq0+fX2Tw9A6VQm1wfmC-wNw@mail.gmail.com>
Message-ID: <CAHubkyO4TPEkpBUxMEns=h6ELsnB_5zdEv9nmVvaBhw-R3Orsg@mail.gmail.com>

I can make ffmpeg listen for RTMP, how would I get that into a conference
room?

><>
nathan stratton


On Thu, Oct 15, 2020 at 10:36 PM Christopher Rienzo <cmrienzo at gmail.com>
wrote:

> You could probably set up ffmpeg as a server to listen for the stream and
> then pull it on FS.
>
> On Thu, Oct 15, 2020 at 9:28 PM Ken Rice <krice at freeswitch.org> wrote:
>
>> these is to my knowledge no way to listen for a stream to randomly start
>> up. This would require designing a new module to handle the listening
>> service.
>>
>> Sent from my iPhone
>>
>> On Oct 15, 2020, at 11:43, Nathan Stratton <nathan at robotics.net> wrote:
>>
>> 
>> Thanks, so yes, I want to play not record, but when I try to play it's
>> looking for a file, it's not listening on port 1935 for a RTMP stream from
>> the server. I need to ACCEPT a RTMP steam like mod_rtmp did, I understand
>> why mod_rtmp is not working but is there any other way to listen from RTMP
>> on port 1935 and send it into a conference?
>>
>>
>> freeswitch at troc.staging.vocinity.com> conference 4600 play
>> rtmp://localhost
>> -ERR (play) File: rtmp://localhost not found.
>>
>> 2020-10-15 12:38:12.636530 [ERR] avformat.c:1603 Invalid Format
>>
>>
>> So the issue is that play rtmp:// does not listen for an incoming RTMP
>> stream.
>> ><>
>> nathan stratton
>>
>>
>> On Thu, Oct 15, 2020 at 12:32 PM Ken Rice <krice at freeswitch.org> wrote:
>>
>>> Mod_rtmp was intended as a softphone endpoint written in flash. dont use
>>> it. litterally load mod_av then via esl, fs_cli etc  “conference <confname>
>>> [play|record]  rtmp://{rest/of/url}”
>>>
>>> its really that simple. play will pkay the rtmp stream into the
>>> conference like any other media file. record will shoot a media stream out
>>> to what ever rtmp reflector you are using auch as youtube or facebook or
>>> your own private rtmp server.
>>>
>>> Sent from my iPhone
>>>
>>> On Oct 15, 2020, at 08:42, Nathan Stratton <nathan at robotics.net> wrote:
>>>
>>> 
>>> On Wed, Oct 14, 2020 at 10:43 PM David P <davidswalkabout at gmail.com>
>>> wrote:
>>>
>>>> In response to the thread " $500 bounty RTMP to conference room", try
>>>> something like this:
>>>>
>>>> <action application="record_session"
>>>> data="{modname=mod_av,vw=320,vh=240}rtmp://${ipOfDestination}:${portOfDestination}/${queryPathForDestination}"/>
>>>>
>>>
>>> So I am not trying to send rtmp, I am trying to do I think what mod_rtmp
>>> was doing and LISTEN for rtmp that is broadcast to the FreeSWITCH server.
>>>
>>> -Nathan
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From nathan at robotics.net  Fri Oct 16 11:22:28 2020
From: nathan at robotics.net (Nathan Stratton)
Date: Fri, 16 Oct 2020 07:22:28 -0400
Subject: [Freeswitch-users] $500 bounty RTMP to conference room
In-Reply-To: <009101d6a271$4c1161a0$e43424e0$@freeswitch.org>
References: <CAHubkyNPZ4Rc=cZCy7co1U_1vbOLOGm41ghaHXyJcG4sW=OF8g@mail.gmail.com>
 <5EFA741A-5934-46DD-98D5-9370E039BF4B@freeswitch.org>
 <CAHubkyOv-c8DG-3aQHN87a+_QU0o3vsSQY-wD_SmhhF7O6oCYA@mail.gmail.com>
 <009101d6a271$4c1161a0$e43424e0$@freeswitch.org>
Message-ID: <CAHubkyNxcpbEkwXk9O7NX0UzgaA9=2o-BVijBj+WXnA99NdSLg@mail.gmail.com>

So the big part of what I was missing was something to receive the RTMP
stream, using nginx I now can receive RTMP on 1935 and if I connect to
nginx on 1935 I can play it. So now I just need this to work:


freeswitch at troc.staging.vocinity.com> conference 4600 play rtmp://
10.0.0.51:1935/4600

-ERR (play) File: rtmp://10.0.0.51:1935/4600 not found.

2020-10-16 07:18:35.507992 [DEBUG] avformat.c:1665 sample rate: 16000,
channels: 1
2020-10-16 07:18:35.507992 [WARNING] avformat.c:1116 Could not open input
file '10.0.0.51:1935/4600' (error 'Protocol not found')


However, if I try mplayer with the same destination it works:

mplayer rtmp://10.0.0.51:1935/4600



mod_av is installed:

freeswitch at troc.staging.vocinity.com> load mod_av

+OK Reloading XML
-ERR [Module already loaded]

2020-10-16 07:22:04.967994 [WARNING] switch_loadable_module.c:1855 Module
mod_av Already Loaded!
2020-10-16 07:22:04.967994 [INFO] switch_time.c:1430 Timezone reloaded 0
definitions



><>
nathan stratton


On Wed, Oct 14, 2020 at 5:31 PM <krice at freeswitch.org> wrote:

> Conference CONFNAME play rtmp://whatever/stream/name
>
> just have mod_av installed also you can conference record rtmp://… too
> broadcast the conf room out
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Nathan Stratton
> *Sent:* Wednesday, October 14, 2020 1:16 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] $500 bounty RTMP to conference room
>
>
>
> Got it, I can't find much on mod_av and nothing on how to have mod_av
> accept rtmp stream and add it to a conference. Not a big fan of RTMP, but
> that is what slingstudio supports.
>
>
> ><>
> nathan stratton
>
>
>
>
>
> On Wed, Oct 14, 2020 at 12:57 PM Mike Jerris <mike at freeswitch.org> wrote:
>
> Use mod_av to record/playback rtmp://
>
>
>
> mod_rtmp is an endpoint module. Its a different animal that really has no
> use now that webrtc is available almost everywhere.
>
>
>
> On Oct 14, 2020, at 11:57 AM, Nathan Stratton <nathan at robotics.net> wrote:
>
>
>
>
> Trying to broadcast from vlc or a sling box with rtmp to freeswitch
> conference room. I have mod_rtmp installed and why I try to steam from vlc
> I get:
>
>
>
> 2020-10-14 11:56:44.516563 [NOTICE] mod_rtmp.c:905 New RTMP session
> [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
> 2020-10-14 11:56:44.516563 [INFO] rtmp_tcp.c:234 Rtmp connection from
> 50.210.153.253:36864
> 2020-10-14 11:56:44.516563 [DEBUG] rtmp.c:882 Sent handshake response
> 2020-10-14 11:56:44.536531 [DEBUG] rtmp.c:907 Done with handshake
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
> ts=0 stream_id=0x0] len=155
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for
> connect
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x1
> stream_id=0x0] len=4
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x5
> stream_id=0x0] len=4
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x6
> stream_id=0x0] len=5
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4
> stream_id=0x0] len=6
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
> stream_id=0x0] len=201
> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
> stream_id=0x0] len=61
> 2020-10-14 11:56:44.556571 [NOTICE] rtmp_sig.c:122 Sent connect reply
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=2 type=0x1
> ts=0 stream_id=0x0] len=4
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1089 SET CHUNKSIZE=512
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
> ts=0 stream_id=0x0] len=29
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for
> releaseStream
> 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for
> "releaseStream"
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
> ts=0 stream_id=0x0] len=25
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for
> FCPublish
> 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for
> "FCPublish"
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
> ts=0 stream_id=0x0] len=25
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for
> createStream
> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
> stream_id=0x0] len=29
> 2020-10-14 11:56:44.636529 [INFO] rtmp_sig.c:137 Replied to createStream
> (0)
> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:1081 [chunk_stream=8 type=0x14
> ts=0 stream_id=0x1] len=30
> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:192 [amfnumber=8] Got INVOKE for
> publish
> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4
> stream_id=0x0] len=6
> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=5 type=0x14
> stream_id=0x1] len=138
> 2020-10-14 11:56:44.696580 [INFO] rtmp_sig.c:290 Got publish on stream 1.
> 2020-10-14 11:56:45.216562 [DEBUG] rtmp.c:1081 [chunk_stream=4 type=0x12
> ts=0 stream_id=0x1] len=319
> 2020-10-14 11:56:45.656516 [DEBUG] rtmp_tcp.c:243 Closing socket
> 2020-10-14 11:56:51.796530 [NOTICE] mod_rtmp.c:1027 RTMP session ended
> [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
>
>
>
> Anyone able to help me? $500 via Paypal.
>
>
> ><>
> nathan stratto
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From nathan at robotics.net  Fri Oct 16 11:23:26 2020
From: nathan at robotics.net (Nathan Stratton)
Date: Fri, 16 Oct 2020 07:23:26 -0400
Subject: [Freeswitch-users] $500 bounty RTMP to conference room
In-Reply-To: <CAHtxdDeszrOj4yyp6Bz6R4UZNfJ7U3W_MUP0EZyY=51JApSfeg@mail.gmail.com>
References: <CAHubkyNPZ4Rc=cZCy7co1U_1vbOLOGm41ghaHXyJcG4sW=OF8g@mail.gmail.com>
 <5EFA741A-5934-46DD-98D5-9370E039BF4B@freeswitch.org>
 <CAHubkyOv-c8DG-3aQHN87a+_QU0o3vsSQY-wD_SmhhF7O6oCYA@mail.gmail.com>
 <CAHtxdDeszrOj4yyp6Bz6R4UZNfJ7U3W_MUP0EZyY=51JApSfeg@mail.gmail.com>
Message-ID: <CAHubkyOO=49S8FjHktrMbkhbqaJ+wh97bG42wP0G13TiG-BNng@mail.gmail.com>

Wanted to try mod_vlc, but ran into:

root at troc:~# apt-get install freeswitch-mod-vlc
Reading package lists... Done
Building dependency tree
Reading state information... Done
Some packages could not be installed. This may mean that you have
requested an impossible situation or if you are using the unstable
distribution that some required packages have not yet been created
or been moved out of Incoming.
The following information may help to resolve the situation:

The following packages have unmet dependencies:
 freeswitch-mod-vlc : Depends: vlc-nox but it is not installable
E: Unable to correct problems, you have held broken packages.
root at troc:~#

><>
nathan stratton


On Wed, Oct 14, 2020 at 4:34 PM Sergey Safarov <s.safarov at gmail.com> wrote:

> You can try mod_vlc.
> I have tested mod_vlc several months ago and found some functions do not
> work as expected.
>
> But you can try.
>
>
> On Wed, Oct 14, 2020 at 9:57 PM Nathan Stratton <nathan at robotics.net>
> wrote:
>
>> Got it, I can't find much on mod_av and nothing on how to have mod_av
>> accept rtmp stream and add it to a conference. Not a big fan of RTMP, but
>> that is what slingstudio supports.
>>
>> ><>
>> nathan stratton
>>
>>
>> On Wed, Oct 14, 2020 at 12:57 PM Mike Jerris <mike at freeswitch.org> wrote:
>>
>>> Use mod_av to record/playback rtmp://
>>>
>>> mod_rtmp is an endpoint module. Its a different animal that really has
>>> no use now that webrtc is available almost everywhere.
>>>
>>> On Oct 14, 2020, at 11:57 AM, Nathan Stratton <nathan at robotics.net>
>>> wrote:
>>>
>>>
>>> Trying to broadcast from vlc or a sling box with rtmp to freeswitch
>>> conference room. I have mod_rtmp installed and why I try to steam from vlc
>>> I get:
>>>
>>> 2020-10-14 11:56:44.516563 [NOTICE] mod_rtmp.c:905 New RTMP session
>>> [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
>>> 2020-10-14 11:56:44.516563 [INFO] rtmp_tcp.c:234 Rtmp connection from
>>> 50.210.153.253:36864
>>> 2020-10-14 11:56:44.516563 [DEBUG] rtmp.c:882 Sent handshake response
>>> 2020-10-14 11:56:44.536531 [DEBUG] rtmp.c:907 Done with handshake
>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
>>> ts=0 stream_id=0x0] len=155
>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>>> for connect
>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x1
>>> stream_id=0x0] len=4
>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x5
>>> stream_id=0x0] len=4
>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x6
>>> stream_id=0x0] len=5
>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4
>>> stream_id=0x0] len=6
>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
>>> stream_id=0x0] len=201
>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
>>> stream_id=0x0] len=61
>>> 2020-10-14 11:56:44.556571 [NOTICE] rtmp_sig.c:122 Sent connect reply
>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=2 type=0x1
>>> ts=0 stream_id=0x0] len=4
>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1089 SET CHUNKSIZE=512
>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
>>> ts=0 stream_id=0x0] len=29
>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>>> for releaseStream
>>> 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for
>>> "releaseStream"
>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
>>> ts=0 stream_id=0x0] len=25
>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>>> for FCPublish
>>> 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for
>>> "FCPublish"
>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
>>> ts=0 stream_id=0x0] len=25
>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>>> for createStream
>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
>>> stream_id=0x0] len=29
>>> 2020-10-14 11:56:44.636529 [INFO] rtmp_sig.c:137 Replied to createStream
>>> (0)
>>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:1081 [chunk_stream=8 type=0x14
>>> ts=0 stream_id=0x1] len=30
>>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:192 [amfnumber=8] Got INVOKE
>>> for publish
>>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4
>>> stream_id=0x0] len=6
>>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=5 type=0x14
>>> stream_id=0x1] len=138
>>> 2020-10-14 11:56:44.696580 [INFO] rtmp_sig.c:290 Got publish on stream 1.
>>> 2020-10-14 11:56:45.216562 [DEBUG] rtmp.c:1081 [chunk_stream=4 type=0x12
>>> ts=0 stream_id=0x1] len=319
>>> 2020-10-14 11:56:45.656516 [DEBUG] rtmp_tcp.c:243 Closing socket
>>> 2020-10-14 11:56:51.796530 [NOTICE] mod_rtmp.c:1027 RTMP session ended
>>> [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
>>>
>>> Anyone able to help me? $500 via Paypal.
>>>
>>> ><>
>>> nathan stratto
>>>
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From darren at aleph-com.net  Wed Oct 14 20:42:20 2020
From: darren at aleph-com.net (Darren Wiebe)
Date: Wed, 14 Oct 2020 14:42:20 -0600
Subject: [Freeswitch-users] $500 bounty RTMP to conference room
In-Reply-To: <CAHtxdDeszrOj4yyp6Bz6R4UZNfJ7U3W_MUP0EZyY=51JApSfeg@mail.gmail.com>
References: <CAHubkyNPZ4Rc=cZCy7co1U_1vbOLOGm41ghaHXyJcG4sW=OF8g@mail.gmail.com>
 <5EFA741A-5934-46DD-98D5-9370E039BF4B@freeswitch.org>
 <CAHubkyOv-c8DG-3aQHN87a+_QU0o3vsSQY-wD_SmhhF7O6oCYA@mail.gmail.com>
 <CAHtxdDeszrOj4yyp6Bz6R4UZNfJ7U3W_MUP0EZyY=51JApSfeg@mail.gmail.com>
Message-ID: <CAGzU91NpH3KRan2JOWqEP52cz9BSJsVa0qr_ZiLxDEKf3yh3-A@mail.gmail.com>

This might be overly complicated but I've used ffmpeg to convert RTMP to
icecast and then used mod_shout or mod_vlc to play the stream into a
conference bridge.

Darren


On Wed, Oct 14, 2020 at 2:34 PM Sergey Safarov <s.safarov at gmail.com> wrote:

> You can try mod_vlc.
> I have tested mod_vlc several months ago and found some functions do not
> work as expected.
>
> But you can try.
>
>
> On Wed, Oct 14, 2020 at 9:57 PM Nathan Stratton <nathan at robotics.net>
> wrote:
>
>> Got it, I can't find much on mod_av and nothing on how to have mod_av
>> accept rtmp stream and add it to a conference. Not a big fan of RTMP, but
>> that is what slingstudio supports.
>>
>> ><>
>> nathan stratton
>>
>>
>> On Wed, Oct 14, 2020 at 12:57 PM Mike Jerris <mike at freeswitch.org> wrote:
>>
>>> Use mod_av to record/playback rtmp://
>>>
>>> mod_rtmp is an endpoint module. Its a different animal that really has
>>> no use now that webrtc is available almost everywhere.
>>>
>>> On Oct 14, 2020, at 11:57 AM, Nathan Stratton <nathan at robotics.net>
>>> wrote:
>>>
>>>
>>> Trying to broadcast from vlc or a sling box with rtmp to freeswitch
>>> conference room. I have mod_rtmp installed and why I try to steam from vlc
>>> I get:
>>>
>>> 2020-10-14 11:56:44.516563 [NOTICE] mod_rtmp.c:905 New RTMP session
>>> [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
>>> 2020-10-14 11:56:44.516563 [INFO] rtmp_tcp.c:234 Rtmp connection from
>>> 50.210.153.253:36864
>>> 2020-10-14 11:56:44.516563 [DEBUG] rtmp.c:882 Sent handshake response
>>> 2020-10-14 11:56:44.536531 [DEBUG] rtmp.c:907 Done with handshake
>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
>>> ts=0 stream_id=0x0] len=155
>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>>> for connect
>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x1
>>> stream_id=0x0] len=4
>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x5
>>> stream_id=0x0] len=4
>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x6
>>> stream_id=0x0] len=5
>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4
>>> stream_id=0x0] len=6
>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
>>> stream_id=0x0] len=201
>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
>>> stream_id=0x0] len=61
>>> 2020-10-14 11:56:44.556571 [NOTICE] rtmp_sig.c:122 Sent connect reply
>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=2 type=0x1
>>> ts=0 stream_id=0x0] len=4
>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1089 SET CHUNKSIZE=512
>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
>>> ts=0 stream_id=0x0] len=29
>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>>> for releaseStream
>>> 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for
>>> "releaseStream"
>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
>>> ts=0 stream_id=0x0] len=25
>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>>> for FCPublish
>>> 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for
>>> "FCPublish"
>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14
>>> ts=0 stream_id=0x0] len=25
>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>>> for createStream
>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
>>> stream_id=0x0] len=29
>>> 2020-10-14 11:56:44.636529 [INFO] rtmp_sig.c:137 Replied to createStream
>>> (0)
>>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:1081 [chunk_stream=8 type=0x14
>>> ts=0 stream_id=0x1] len=30
>>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:192 [amfnumber=8] Got INVOKE
>>> for publish
>>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4
>>> stream_id=0x0] len=6
>>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=5 type=0x14
>>> stream_id=0x1] len=138
>>> 2020-10-14 11:56:44.696580 [INFO] rtmp_sig.c:290 Got publish on stream 1.
>>> 2020-10-14 11:56:45.216562 [DEBUG] rtmp.c:1081 [chunk_stream=4 type=0x12
>>> ts=0 stream_id=0x1] len=319
>>> 2020-10-14 11:56:45.656516 [DEBUG] rtmp_tcp.c:243 Closing socket
>>> 2020-10-14 11:56:51.796530 [NOTICE] mod_rtmp.c:1027 RTMP session ended
>>> [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
>>>
>>> Anyone able to help me? $500 via Paypal.
>>>
>>> ><>
>>> nathan stratto
>>>
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From cpservicespb at gmail.com  Fri Oct 16 09:32:23 2020
From: cpservicespb at gmail.com (CpServiceSPb)
Date: Fri, 16 Oct 2020 12:32:23 +0300
Subject: [Freeswitch-users] Multiple trunks failover schema setting up !
Message-ID: <CAN4xyZOhWWOTo_Kkt=p72VFR4KgtroVcxvW346RxBOC9366n1g@mail.gmail.com>

Hi.
I have 3 external SIP trunks for making external calls, provided by SIP
telephony carriers:
SIP carrier1,
SIP carrier2,
SIP carrier3.

If SIP carrier1 trunk is ready to making callings then the outboud calling
outside of FS is making using the trunk, otherwise, for example, if the
trunk state is unknown/invalid or disabled by carrier that is unavailable
or already ringing/busy then attempting of calling will be made using SIP
carrier2 and if the same situation with SIP carrier2 trunk then attempting
of calling will be made using SIP carrier3 trunk and if SIP carrier3 trunk
has state which do not allow the calling voice message to caller will be
send back
What is the best way to make the following schema ?
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From nathan at robotics.net  Fri Oct 16 18:58:49 2020
From: nathan at robotics.net (Nathan Stratton)
Date: Fri, 16 Oct 2020 14:58:49 -0400
Subject: [Freeswitch-users] $500 bounty RTMP to conference room
In-Reply-To: <CAGzU91NpH3KRan2JOWqEP52cz9BSJsVa0qr_ZiLxDEKf3yh3-A@mail.gmail.com>
References: <CAHubkyNPZ4Rc=cZCy7co1U_1vbOLOGm41ghaHXyJcG4sW=OF8g@mail.gmail.com>
 <5EFA741A-5934-46DD-98D5-9370E039BF4B@freeswitch.org>
 <CAHubkyOv-c8DG-3aQHN87a+_QU0o3vsSQY-wD_SmhhF7O6oCYA@mail.gmail.com>
 <CAHtxdDeszrOj4yyp6Bz6R4UZNfJ7U3W_MUP0EZyY=51JApSfeg@mail.gmail.com>
 <CAGzU91NpH3KRan2JOWqEP52cz9BSJsVa0qr_ZiLxDEKf3yh3-A@mail.gmail.com>
Message-ID: <CAHubkyNx_3o-Wt6xqDULcq0mVv5Uhb_563gRW6o=u0aGyAUYHg@mail.gmail.com>

freeswitch at troc.staging.vocinity.com> conference 4600 play vlc://rtmp://
10.0.0.51:1935/4600
+OK (play) Playing file vlc://rtmp://10.0.0.51:1935/4600

2020-10-16 14:47:52.214273 [DEBUG] mod_vlc.c:838 VLC attempt to open rtmp://
10.0.0.51:1935/4600 read
2020-10-16 14:47:52.214273 [DEBUG] mod_vlc.c:863 VLC open rtmp://
10.0.0.51:1935/4600 for reading
2020-10-16 14:47:52.214273 [NOTICE] mod_vlc.c:881 VLC Path is unknown type
rtmp://10.0.0.51:1935/4600
VLC: PulseAudio server connection failure: Connection refused
2020-10-16 14:47:52.214273 [DEBUG] mod_vlc.c:242 Got a
libvlc_MediaStateChanged callback. New state: 1
Starting connection attempt to 10.0.0.51 port 1935
Successfully connected to 10.0.0.51 port 1935
Reinit context to 1280x720, pix_fmt: yuv420p
2020-10-16 14:47:57.334303 [DEBUG] mod_vlc.c:242 Got a
libvlc_MediaStateChanged callback. New state: 3
VLC: buffer deadlock prevented
VLC: parent window not available
VLC: parent window not available
VLC: window not available
freeswitch at troc.staging.vocinity.com>
root at troc:/usr/local/freeswitch/mod#

I get audio for a second (no video) and then it crashes.


><>
nathan stratton


On Fri, Oct 16, 2020 at 2:19 PM Darren Wiebe <darren at aleph-com.net> wrote:

> This might be overly complicated but I've used ffmpeg to convert RTMP to
> icecast and then used mod_shout or mod_vlc to play the stream into a
> conference bridge.
>
> Darren
>
>
> On Wed, Oct 14, 2020 at 2:34 PM Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> You can try mod_vlc.
>> I have tested mod_vlc several months ago and found some functions do not
>> work as expected.
>>
>> But you can try.
>>
>>
>> On Wed, Oct 14, 2020 at 9:57 PM Nathan Stratton <nathan at robotics.net>
>> wrote:
>>
>>> Got it, I can't find much on mod_av and nothing on how to have mod_av
>>> accept rtmp stream and add it to a conference. Not a big fan of RTMP, but
>>> that is what slingstudio supports.
>>>
>>> ><>
>>> nathan stratton
>>>
>>>
>>> On Wed, Oct 14, 2020 at 12:57 PM Mike Jerris <mike at freeswitch.org>
>>> wrote:
>>>
>>>> Use mod_av to record/playback rtmp://
>>>>
>>>> mod_rtmp is an endpoint module. Its a different animal that really has
>>>> no use now that webrtc is available almost everywhere.
>>>>
>>>> On Oct 14, 2020, at 11:57 AM, Nathan Stratton <nathan at robotics.net>
>>>> wrote:
>>>>
>>>>
>>>> Trying to broadcast from vlc or a sling box with rtmp to freeswitch
>>>> conference room. I have mod_rtmp installed and why I try to steam from vlc
>>>> I get:
>>>>
>>>> 2020-10-14 11:56:44.516563 [NOTICE] mod_rtmp.c:905 New RTMP session
>>>> [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
>>>> 2020-10-14 11:56:44.516563 [INFO] rtmp_tcp.c:234 Rtmp connection from
>>>> 50.210.153.253:36864
>>>> 2020-10-14 11:56:44.516563 [DEBUG] rtmp.c:882 Sent handshake response
>>>> 2020-10-14 11:56:44.536531 [DEBUG] rtmp.c:907 Done with handshake
>>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:1081 [chunk_stream=3
>>>> type=0x14 ts=0 stream_id=0x0] len=155
>>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>>>> for connect
>>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x1
>>>> stream_id=0x0] len=4
>>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x5
>>>> stream_id=0x0] len=4
>>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x6
>>>> stream_id=0x0] len=5
>>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4
>>>> stream_id=0x0] len=6
>>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
>>>> stream_id=0x0] len=201
>>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
>>>> stream_id=0x0] len=61
>>>> 2020-10-14 11:56:44.556571 [NOTICE] rtmp_sig.c:122 Sent connect reply
>>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=2 type=0x1
>>>> ts=0 stream_id=0x0] len=4
>>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1089 SET CHUNKSIZE=512
>>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3
>>>> type=0x14 ts=0 stream_id=0x0] len=29
>>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>>>> for releaseStream
>>>> 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for
>>>> "releaseStream"
>>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3
>>>> type=0x14 ts=0 stream_id=0x0] len=25
>>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>>>> for FCPublish
>>>> 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for
>>>> "FCPublish"
>>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3
>>>> type=0x14 ts=0 stream_id=0x0] len=25
>>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>>>> for createStream
>>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
>>>> stream_id=0x0] len=29
>>>> 2020-10-14 11:56:44.636529 [INFO] rtmp_sig.c:137 Replied to
>>>> createStream (0)
>>>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:1081 [chunk_stream=8
>>>> type=0x14 ts=0 stream_id=0x1] len=30
>>>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:192 [amfnumber=8] Got INVOKE
>>>> for publish
>>>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4
>>>> stream_id=0x0] len=6
>>>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=5 type=0x14
>>>> stream_id=0x1] len=138
>>>> 2020-10-14 11:56:44.696580 [INFO] rtmp_sig.c:290 Got publish on stream
>>>> 1.
>>>> 2020-10-14 11:56:45.216562 [DEBUG] rtmp.c:1081 [chunk_stream=4
>>>> type=0x12 ts=0 stream_id=0x1] len=319
>>>> 2020-10-14 11:56:45.656516 [DEBUG] rtmp_tcp.c:243 Closing socket
>>>> 2020-10-14 11:56:51.796530 [NOTICE] mod_rtmp.c:1027 RTMP session ended
>>>> [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
>>>>
>>>> Anyone able to help me? $500 via Paypal.
>>>>
>>>> ><>
>>>> nathan stratto
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From brandon at cryy.com  Fri Oct 16 23:58:46 2020
From: brandon at cryy.com (Brandon Armstead)
Date: Fri, 16 Oct 2020 16:58:46 -0700
Subject: [Freeswitch-users] $500 bounty RTMP to conference room
In-Reply-To: <CAHubkyNx_3o-Wt6xqDULcq0mVv5Uhb_563gRW6o=u0aGyAUYHg@mail.gmail.com>
References: <CAHubkyNPZ4Rc=cZCy7co1U_1vbOLOGm41ghaHXyJcG4sW=OF8g@mail.gmail.com>
 <5EFA741A-5934-46DD-98D5-9370E039BF4B@freeswitch.org>
 <CAHubkyOv-c8DG-3aQHN87a+_QU0o3vsSQY-wD_SmhhF7O6oCYA@mail.gmail.com>
 <CAHtxdDeszrOj4yyp6Bz6R4UZNfJ7U3W_MUP0EZyY=51JApSfeg@mail.gmail.com>
 <CAGzU91NpH3KRan2JOWqEP52cz9BSJsVa0qr_ZiLxDEKf3yh3-A@mail.gmail.com>
 <CAHubkyNx_3o-Wt6xqDULcq0mVv5Uhb_563gRW6o=u0aGyAUYHg@mail.gmail.com>
Message-ID: <CABAX3EpEi9nTxoGyovBv8udr5uU0pu114gH7frcmVQ9d0PHZxw@mail.gmail.com>

Freeswitch crashes? Is there a core dump?

On Fri, Oct 16, 2020 at 12:00 PM Nathan Stratton <nathan at robotics.net>
wrote:

>
> freeswitch at troc.staging.vocinity.com> conference 4600 play vlc://rtmp://
> 10.0.0.51:1935/4600
> +OK (play) Playing file vlc://rtmp://10.0.0.51:1935/4600
>
> 2020-10-16 14:47:52.214273 [DEBUG] mod_vlc.c:838 VLC attempt to open
> rtmp://10.0.0.51:1935/4600 read
> 2020-10-16 14:47:52.214273 [DEBUG] mod_vlc.c:863 VLC open rtmp://
> 10.0.0.51:1935/4600 for reading
> 2020-10-16 14:47:52.214273 [NOTICE] mod_vlc.c:881 VLC Path is unknown type
> rtmp://10.0.0.51:1935/4600
> VLC: PulseAudio server connection failure: Connection refused
> 2020-10-16 14:47:52.214273 [DEBUG] mod_vlc.c:242 Got a
> libvlc_MediaStateChanged callback. New state: 1
> Starting connection attempt to 10.0.0.51 port 1935
> Successfully connected to 10.0.0.51 port 1935
> Reinit context to 1280x720, pix_fmt: yuv420p
> 2020-10-16 14:47:57.334303 [DEBUG] mod_vlc.c:242 Got a
> libvlc_MediaStateChanged callback. New state: 3
> VLC: buffer deadlock prevented
> VLC: parent window not available
> VLC: parent window not available
> VLC: window not available
> freeswitch at troc.staging.vocinity.com>
> root at troc:/usr/local/freeswitch/mod#
>
> I get audio for a second (no video) and then it crashes.
>
>
> ><>
> nathan stratton
>
>
> On Fri, Oct 16, 2020 at 2:19 PM Darren Wiebe <darren at aleph-com.net> wrote:
>
>> This might be overly complicated but I've used ffmpeg to convert RTMP to
>> icecast and then used mod_shout or mod_vlc to play the stream into a
>> conference bridge.
>>
>> Darren
>>
>>
>> On Wed, Oct 14, 2020 at 2:34 PM Sergey Safarov <s.safarov at gmail.com>
>> wrote:
>>
>>> You can try mod_vlc.
>>> I have tested mod_vlc several months ago and found some functions do not
>>> work as expected.
>>>
>>> But you can try.
>>>
>>>
>>> On Wed, Oct 14, 2020 at 9:57 PM Nathan Stratton <nathan at robotics.net>
>>> wrote:
>>>
>>>> Got it, I can't find much on mod_av and nothing on how to have mod_av
>>>> accept rtmp stream and add it to a conference. Not a big fan of RTMP, but
>>>> that is what slingstudio supports.
>>>>
>>>> ><>
>>>> nathan stratton
>>>>
>>>>
>>>> On Wed, Oct 14, 2020 at 12:57 PM Mike Jerris <mike at freeswitch.org>
>>>> wrote:
>>>>
>>>>> Use mod_av to record/playback rtmp://
>>>>>
>>>>> mod_rtmp is an endpoint module. Its a different animal that really has
>>>>> no use now that webrtc is available almost everywhere.
>>>>>
>>>>> On Oct 14, 2020, at 11:57 AM, Nathan Stratton <nathan at robotics.net>
>>>>> wrote:
>>>>>
>>>>>
>>>>> Trying to broadcast from vlc or a sling box with rtmp to freeswitch
>>>>> conference room. I have mod_rtmp installed and why I try to steam from vlc
>>>>> I get:
>>>>>
>>>>> 2020-10-14 11:56:44.516563 [NOTICE] mod_rtmp.c:905 New RTMP session
>>>>> [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
>>>>> 2020-10-14 11:56:44.516563 [INFO] rtmp_tcp.c:234 Rtmp connection from
>>>>> 50.210.153.253:36864
>>>>> 2020-10-14 11:56:44.516563 [DEBUG] rtmp.c:882 Sent handshake response
>>>>> 2020-10-14 11:56:44.536531 [DEBUG] rtmp.c:907 Done with handshake
>>>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:1081 [chunk_stream=3
>>>>> type=0x14 ts=0 stream_id=0x0] len=155
>>>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>>>>> for connect
>>>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x1
>>>>> stream_id=0x0] len=4
>>>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x5
>>>>> stream_id=0x0] len=4
>>>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x6
>>>>> stream_id=0x0] len=5
>>>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4
>>>>> stream_id=0x0] len=6
>>>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
>>>>> stream_id=0x0] len=201
>>>>> 2020-10-14 11:56:44.556571 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
>>>>> stream_id=0x0] len=61
>>>>> 2020-10-14 11:56:44.556571 [NOTICE] rtmp_sig.c:122 Sent connect reply
>>>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=2
>>>>> type=0x1 ts=0 stream_id=0x0] len=4
>>>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1089 SET CHUNKSIZE=512
>>>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3
>>>>> type=0x14 ts=0 stream_id=0x0] len=29
>>>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>>>>> for releaseStream
>>>>> 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for
>>>>> "releaseStream"
>>>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3
>>>>> type=0x14 ts=0 stream_id=0x0] len=25
>>>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>>>>> for FCPublish
>>>>> 2020-10-14 11:56:44.636529 [WARNING] rtmp.c:198 Unhandled invoke for
>>>>> "FCPublish"
>>>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:1081 [chunk_stream=3
>>>>> type=0x14 ts=0 stream_id=0x0] len=25
>>>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE
>>>>> for createStream
>>>>> 2020-10-14 11:56:44.636529 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14
>>>>> stream_id=0x0] len=29
>>>>> 2020-10-14 11:56:44.636529 [INFO] rtmp_sig.c:137 Replied to
>>>>> createStream (0)
>>>>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:1081 [chunk_stream=8
>>>>> type=0x14 ts=0 stream_id=0x1] len=30
>>>>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:192 [amfnumber=8] Got INVOKE
>>>>> for publish
>>>>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4
>>>>> stream_id=0x0] len=6
>>>>> 2020-10-14 11:56:44.696580 [DEBUG] rtmp.c:656 [amfnumber=5 type=0x14
>>>>> stream_id=0x1] len=138
>>>>> 2020-10-14 11:56:44.696580 [INFO] rtmp_sig.c:290 Got publish on stream
>>>>> 1.
>>>>> 2020-10-14 11:56:45.216562 [DEBUG] rtmp.c:1081 [chunk_stream=4
>>>>> type=0x12 ts=0 stream_id=0x1] len=319
>>>>> 2020-10-14 11:56:45.656516 [DEBUG] rtmp_tcp.c:243 Closing socket
>>>>> 2020-10-14 11:56:51.796530 [NOTICE] mod_rtmp.c:1027 RTMP session ended
>>>>> [88e00c1b-6b3a-4bac-83f5-bc12b879884b]
>>>>>
>>>>> Anyone able to help me? $500 via Paypal.
>>>>>
>>>>> ><>
>>>>> nathan stratto
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From social at bohboh.info  Sat Oct 17 23:28:29 2020
From: social at bohboh.info (Social Boh)
Date: Sat, 17 Oct 2020 18:28:29 -0500
Subject: Verto Communicator error
Message-ID: <433beac7-3d1a-c20b-cab5-ebfe03f46634@bohboh.info>

Hello,

this is the problem:

bower bootstrap-material-design#~0.3.0       not-cached 
https://github.com/FezVrasta/bootstrap-material-design.git#~0.3.0
bower bootstrap-material-design#~0.3.0          resolve 
https://github.com/FezVrasta/bootstrap-material-design.git#~0.3.0
bower bootstrap-material-design#~0.3.0     ENORESTARGET No tag found 
that was able to satisfy ~0.3.0

the new location for this tool is: 
https://github.com/mdbootstrap/bootstrap-material-design

Where can i change the new link?

Thank you

Regards

-- 
---
I'm SoCIaL, MayBe



From harshita19012 at iiitd.ac.in  Mon Oct 19 07:20:40 2020
From: harshita19012 at iiitd.ac.in (Harshita Pandey)
Date: Mon, 19 Oct 2020 12:50:40 +0530
Subject: [Freeswitch-users] Port Required to communicate with Gateway
In-Reply-To: <CAGvKJUV+AMThxzT+ngevqC7hZx30LaNChdD7n0=GPaVB6c1YJA@mail.gmail.com>
References: <CA+by-=WB_zzT1NMS6-AKFAjjNs3k1Hv5GYaRqOi3_MbyTCoO7A@mail.gmail.com>
 <CAGvKJUWdUSvXja99N5XN=fR6AVY9t26tfT6SWxrL9MOCFpA6Kg@mail.gmail.com>
 <CA+by-=Ve-XpzY1wnEOy+MUPowTXXbAXMxV0Q1pFXwiJHgndFLg@mail.gmail.com>
 <CAGvKJUV+AMThxzT+ngevqC7hZx30LaNChdD7n0=GPaVB6c1YJA@mail.gmail.com>
Message-ID: <CA+by-=VWygJCuf+F55DtD_hepxbaruF-a0a8EtSLV4rXBmBQ2g@mail.gmail.com>

I executed the sofia global siptrace on command and get the following error

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 sip-ip [192.168.2.215]
                                                              2020-10-19
07:18:34.576508 [DEBUG] sofia.c:4597 ext-rtp-ip [103.25.231.107]
                                                   2020-10-19
07:18:34.576508 [DEBUG] sofia.c:4597 ext-sip-ip [103.25.231.107]
                                                   2020-10-19
07:18:34.576508 [DEBUG] sofia.c:4597 rtp-timeout-sec [300]
                                                   2020-10-19
07:18:34.576508 [WARNING] sofia.c:5192 rtp-timeout-sec deprecated use
media_timeout variable.                                 2020-10-19
07:18:34.576508 [DEBUG] sofia.c:4597 rtp-hold-timeout-sec [1800]
                                                   2020-10-19
07:18:34.576508 [WARNING] sofia.c:5199 rtp-hold-timeout-sec deprecated use
media_hold_timeout variable.                       2020-10-19
07:18:34.576508 [DEBUG] sofia.c:4597 tls [false]
                                                   2020-10-19
07:18:34.576508 [DEBUG] sofia.c:4597 tls-only [false]
                                                  2020-10-19
07:18:34.576508 [DEBUG] sofia.c:4597 tls-bind-params [transport=tls]
                                                   2020-10-19
07:18:34.576508 [DEBUG] sofia.c:4597 tls-sip-port [5081]
                                                   2020-10-19
07:18:34.576508 [DEBUG] sofia.c:4597 tls-passphrase []
                                                   2020-10-19
07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-date [true]
                                                  2020-10-19
07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-policy [none]
                                                  2020-10-19
07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-depth [2]
                                                  2020-10-19
07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-in-subjects []
                                                   2020-10-19
07:18:34.576508 [DEBUG] sofia.c:4597 tls-version [tlsv1,tlsv1.1,tlsv1.2]
                                                   2020-10-19
07:18:34.576508 [INFO] sofia.c:5993 Setting MAX Auth Validity to 0 Attempts
                                                  2020-10-19
07:18:34.576508 [NOTICE] sofia.c:6160 Started Profile external
[sofia_reg_external]                                           2020-10-19
07:18:34.576508 [DEBUG] sofia.c:3134 Creating agent for external
                                                   2020-10-19
07:18:34.576508 [DEBUG] sofia.c:3253 Created agent for external
                                                  2020-10-19
07:18:34.576508 [DEBUG] sofia.c:3302 Set params for external
                                                   2020-10-19
07:18:34.576508 [DEBUG] sofia.c:3349 Activated db for external
                                                   2020-10-19
07:18:34.576508 [INFO] switch_core_sqldb.c:1870 sofia:external Starting SQL
thread.                                           2020-10-19
07:18:34.576508 [DEBUG] sofia.c:3387 Starting thread for external
                                                  2020-10-19
07:18:34.576508 [DEBUG] sofia.c:3034 Launching worker thread for external
                                                  2020-10-19
07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway 'MySIP' to profile
'external'                                         2020-10-19
07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway 'MySIP1' to profile
'external'                                        2020-10-19
07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway 'example.com' to
profile 'external'                                   2020-10-19
07:18:35.576510 [NOTICE] sofia_reg.c:453 Registering MySIP1
                                                  2020-10-19
07:18:35.576510 [NOTICE] sofia_reg.c:453 Registering MySIP
                                                   2020-10-19
07:18:35.756510 [ERR] sofia_reg.c:2469 MySIP1 Failed Registration with
status Service Unavailable [503]. failure #1           2020-10-19
07:18:35.756510 [ERR] sofia_reg.c:2469 MySIP Failed Registration with
status Service Unavailable [503]. failure #1            2020-10-19
07:18:36.576508 [WARNING] sofia_reg.c:510 MySIP1 Failed Registration [503],
setting retry to 5 seconds.                       2020-10-19
07:18:36.576508 [WARNING] sofia_reg.c:510 MySIP Failed Registration [503],
setting retry to 30 seconds.                       2020-10-19
07:18:42.596508 [NOTICE] sofia_reg.c:453 Registering MySIP1
                                                  2020-10-19
07:18:42.956509 [ERR] sofia_reg.c:2469 MySIP1 Failed Registration with
status Service Unavailable [503]. failure #2           2020-10-19
07:18:43.596508 [WARNING] sofia_reg.c:510 MySIP1 Failed Registration [503],
setting retry to 5 seconds.                       2020-10-19
07:18:49.596509 [NOTICE] sofia_reg.c:453 Registering MySIP1
                                                  2020-10-19
07:18:49.956507 [ERR] sofia_reg.c:2469 MySIP1 Failed Registration with
status Service Unavailable [503]. failure #3           2020-10-19
07:18:50.596510 [WARNING] sofia_reg.c:510 MySIP1 Failed Registration [503],
setting retry to 5 seconds.

On Fri, Oct 9, 2020 at 2:52 AM Jay Desai <rangeare.01 at gmail.com> wrote:

> I suspect this to be something related to your sip configuration. If you
> are getting 4XX or 5XX response from gateway means the ports are open (you
> don't have to open ports for outbound connections unless they are
> explicitly blocked by any firewall or program.)
>
> Share your sip trace by executing *sofia global sip trace on* or by using
> the *tcpdump* command.
>
> On Thu, Oct 8, 2020 at 8:44 PM Harshita Pandey <harshita19012 at iiitd.ac.in>
> wrote:
>
>> Okay. I am actually trying to configure Gateway so that I can start
>> making calls through a number. The virtual machine in which I have
>> installed freeswitch has only port 22 opened, so wanted to know the other
>> ports which need to be opened to configure the Gateway correctly.
>>
>> I get the following errors:
>> MySIP Failed Registration with status Forbidden [403]. failure #1still
>> getting the same error
>> OR
>> MySIP failed Registration error [503]. retrying in 30 seconds.
>>
>> According to what I have understood this is because some ports are not
>> opened in the VM
>>
>> On Thu, Oct 8, 2020 at 7:52 PM Jay Desai <rangeare.01 at gmail.com> wrote:
>>
>>> Freeswitch uses different profiles for outgoing and incoming calls.
>>> The default port for an internal profile is 5060 and external profile
>>> uses 5080.
>>> Can you provide more details about the gateway and what you are trying
>>> to do?
>>>
>>> On Thu, Oct 8, 2020 at 4:58 PM Harshita Pandey <
>>> harshita19012 at iiitd.ac.in> wrote:
>>>
>>>> Hi,
>>>>
>>>> I have installed freeswitch on a Virtual Machine, and I am not able to
>>>> configure the gateway for some reason. Can someone tell me which ports are
>>>> required and need to be opened on the virtual machine to communicate with
>>>> Gateway in freeswitch?
>>>> Since all the ports are not opened in the Virtual Machine, this could
>>>> help me open up those ports and configure Gateway for freeswitch
>>>>
>>>>
>>>> Much help is needed.
>>>> It's Urgent
>>>> Thanks in advance.
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From harshita19012 at iiitd.ac.in  Mon Oct 19 07:22:31 2020
From: harshita19012 at iiitd.ac.in (Harshita Pandey)
Date: Mon, 19 Oct 2020 12:52:31 +0530
Subject: [Freeswitch-users] Port Required to communicate with Gateway
In-Reply-To: <CA+by-=VWygJCuf+F55DtD_hepxbaruF-a0a8EtSLV4rXBmBQ2g@mail.gmail.com>
References: <CA+by-=WB_zzT1NMS6-AKFAjjNs3k1Hv5GYaRqOi3_MbyTCoO7A@mail.gmail.com>
 <CAGvKJUWdUSvXja99N5XN=fR6AVY9t26tfT6SWxrL9MOCFpA6Kg@mail.gmail.com>
 <CA+by-=Ve-XpzY1wnEOy+MUPowTXXbAXMxV0Q1pFXwiJHgndFLg@mail.gmail.com>
 <CAGvKJUV+AMThxzT+ngevqC7hZx30LaNChdD7n0=GPaVB6c1YJA@mail.gmail.com>
 <CA+by-=VWygJCuf+F55DtD_hepxbaruF-a0a8EtSLV4rXBmBQ2g@mail.gmail.com>
Message-ID: <CA+by-=Upttv+rx29_ZZikUfZLdcwZ3i7_ExnfHesRCou=FUE+w@mail.gmail.com>

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 sip-ip [192.168.2.215]

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 ext-rtp-ip [103.25.231.107]

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 ext-sip-ip [103.25.231.107]

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 rtp-timeout-sec [300]

2020-10-19 07:18:34.576508 [WARNING] sofia.c:5192 rtp-timeout-sec
deprecated use media_timeout variable.
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 rtp-hold-timeout-sec [1800]

2020-10-19 07:18:34.576508 [WARNING] sofia.c:5199 rtp-hold-timeout-sec
deprecated use media_hold_timeout variable.
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls [false]

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-only [false]

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-bind-params
[transport=tls]
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-sip-port [5081]

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-passphrase []

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-date [true]

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-policy [none]

 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-depth [2]

 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-in-subjects []

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-version
[tlsv1,tlsv1.1,tlsv1.2]

2020-10-19 07:18:34.576508 [INFO] sofia.c:5993 Setting MAX Auth Validity to
0 Attempts
2020-10-19 07:18:34.576508 [NOTICE] sofia.c:6160 Started Profile external
[sofia_reg_external]
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3134 Creating agent for external

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3253 Created agent for external

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3302 Set params for external

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3349 Activated db for external

2020-10-19 07:18:34.576508 [INFO] switch_core_sqldb.c:1870 sofia:external
Starting SQL thread.
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3387 Starting thread for
external
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3034 Launching worker thread for
external
 2020-10-19 07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway 'MySIP'
to profile 'external'
2020-10-19 07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway 'MySIP1'
to profile 'external'
2020-10-19 07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway '
example.com' to profile 'external'
2020-10-19 07:18:35.576510 [NOTICE] sofia_reg.c:453 Registering MySIP1

2020-10-19 07:18:35.576510 [NOTICE] sofia_reg.c:453 Registering MySIP

2020-10-19 07:18:35.756510 [ERR] sofia_reg.c:2469 MySIP1 Failed
Registration with status Service Unavailable [503]. failure #1
2020-10-19 07:18:35.756510 [ERR] sofia_reg.c:2469 MySIP Failed Registration
with status Service Unavailable [503]. failure #1
2020-10-19 07:18:36.576508 [WARNING] sofia_reg.c:510 MySIP1 Failed
Registration [503], setting retry to 5 seconds.
2020-10-19 07:18:36.576508 [WARNING] sofia_reg.c:510 MySIP Failed
Registration [503], setting retry to 30 seconds.
2020-10-19 07:18:42.596508 [NOTICE] sofia_reg.c:453 Registering MySIP1

2020-10-19 07:18:42.956509 [ERR] sofia_reg.c:2469 MySIP1 Failed
Registration with status Service Unavailable [503]. failure #2
2020-10-19 07:18:43.596508 [WARNING] sofia_reg.c:510 MySIP1 Failed
Registration [503], setting retry to 5 seconds.
2020-10-19 07:18:49.596509 [NOTICE] sofia_reg.c:453 Registering MySIP1

2020-10-19 07:18:49.956507 [ERR] sofia_reg.c:2469 MySIP1 Failed
Registration with status Service Unavailable [503]. failure #3
2020-10-19 07:18:50.596510 [WARNING] sofia_reg.c:510 MySIP1 Failed
Registration [503], setting retry to 5 seconds.

On Mon, Oct 19, 2020 at 12:50 PM Harshita Pandey <harshita19012 at iiitd.ac.in>
wrote:

> I executed the sofia global siptrace on command and get the following error
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 sip-ip [192.168.2.215]
>                                                               2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:4597 ext-rtp-ip [103.25.231.107]
>                                                    2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:4597 ext-sip-ip [103.25.231.107]
>                                                    2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:4597 rtp-timeout-sec [300]
>                                                    2020-10-19
> 07:18:34.576508 [WARNING] sofia.c:5192 rtp-timeout-sec deprecated use
> media_timeout variable.                                 2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:4597 rtp-hold-timeout-sec [1800]
>                                                    2020-10-19
> 07:18:34.576508 [WARNING] sofia.c:5199 rtp-hold-timeout-sec deprecated use
> media_hold_timeout variable.                       2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:4597 tls [false]
>                                                    2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:4597 tls-only [false]
>                                                   2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:4597 tls-bind-params [transport=tls]
>                                                    2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:4597 tls-sip-port [5081]
>                                                    2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:4597 tls-passphrase []
>                                                    2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-date [true]
>                                                   2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-policy [none]
>                                                   2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-depth [2]
>                                                   2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-in-subjects []
>                                                    2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:4597 tls-version [tlsv1,tlsv1.1,tlsv1.2]
>                                                    2020-10-19
> 07:18:34.576508 [INFO] sofia.c:5993 Setting MAX Auth Validity to 0 Attempts
>                                                   2020-10-19
> 07:18:34.576508 [NOTICE] sofia.c:6160 Started Profile external
> [sofia_reg_external]                                           2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:3134 Creating agent for external
>                                                    2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:3253 Created agent for external
>                                                   2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:3302 Set params for external
>                                                    2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:3349 Activated db for external
>                                                    2020-10-19
> 07:18:34.576508 [INFO] switch_core_sqldb.c:1870 sofia:external Starting SQL
> thread.                                           2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:3387 Starting thread for external
>                                                   2020-10-19
> 07:18:34.576508 [DEBUG] sofia.c:3034 Launching worker thread for external
>                                                   2020-10-19
> 07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway 'MySIP' to profile
> 'external'                                         2020-10-19
> 07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway 'MySIP1' to profile
> 'external'                                        2020-10-19
> 07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway 'example.com' to
> profile 'external'                                   2020-10-19
> 07:18:35.576510 [NOTICE] sofia_reg.c:453 Registering MySIP1
>                                                   2020-10-19
> 07:18:35.576510 [NOTICE] sofia_reg.c:453 Registering MySIP
>                                                    2020-10-19
> 07:18:35.756510 [ERR] sofia_reg.c:2469 MySIP1 Failed Registration with
> status Service Unavailable [503]. failure #1           2020-10-19
> 07:18:35.756510 [ERR] sofia_reg.c:2469 MySIP Failed Registration with
> status Service Unavailable [503]. failure #1            2020-10-19
> 07:18:36.576508 [WARNING] sofia_reg.c:510 MySIP1 Failed Registration [503],
> setting retry to 5 seconds.                       2020-10-19
> 07:18:36.576508 [WARNING] sofia_reg.c:510 MySIP Failed Registration [503],
> setting retry to 30 seconds.                       2020-10-19
> 07:18:42.596508 [NOTICE] sofia_reg.c:453 Registering MySIP1
>                                                   2020-10-19
> 07:18:42.956509 [ERR] sofia_reg.c:2469 MySIP1 Failed Registration with
> status Service Unavailable [503]. failure #2           2020-10-19
> 07:18:43.596508 [WARNING] sofia_reg.c:510 MySIP1 Failed Registration [503],
> setting retry to 5 seconds.                       2020-10-19
> 07:18:49.596509 [NOTICE] sofia_reg.c:453 Registering MySIP1
>                                                   2020-10-19
> 07:18:49.956507 [ERR] sofia_reg.c:2469 MySIP1 Failed Registration with
> status Service Unavailable [503]. failure #3           2020-10-19
> 07:18:50.596510 [WARNING] sofia_reg.c:510 MySIP1 Failed Registration [503],
> setting retry to 5 seconds.
>
> On Fri, Oct 9, 2020 at 2:52 AM Jay Desai <rangeare.01 at gmail.com> wrote:
>
>> I suspect this to be something related to your sip configuration. If you
>> are getting 4XX or 5XX response from gateway means the ports are open (you
>> don't have to open ports for outbound connections unless they are
>> explicitly blocked by any firewall or program.)
>>
>> Share your sip trace by executing *sofia global sip trace on* or by
>> using the *tcpdump* command.
>>
>> On Thu, Oct 8, 2020 at 8:44 PM Harshita Pandey <harshita19012 at iiitd.ac.in>
>> wrote:
>>
>>> Okay. I am actually trying to configure Gateway so that I can start
>>> making calls through a number. The virtual machine in which I have
>>> installed freeswitch has only port 22 opened, so wanted to know the other
>>> ports which need to be opened to configure the Gateway correctly.
>>>
>>> I get the following errors:
>>> MySIP Failed Registration with status Forbidden [403]. failure #1still
>>> getting the same error
>>> OR
>>> MySIP failed Registration error [503]. retrying in 30 seconds.
>>>
>>> According to what I have understood this is because some ports are not
>>> opened in the VM
>>>
>>> On Thu, Oct 8, 2020 at 7:52 PM Jay Desai <rangeare.01 at gmail.com> wrote:
>>>
>>>> Freeswitch uses different profiles for outgoing and incoming calls.
>>>> The default port for an internal profile is 5060 and external profile
>>>> uses 5080.
>>>> Can you provide more details about the gateway and what you are trying
>>>> to do?
>>>>
>>>> On Thu, Oct 8, 2020 at 4:58 PM Harshita Pandey <
>>>> harshita19012 at iiitd.ac.in> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> I have installed freeswitch on a Virtual Machine, and I am not able to
>>>>> configure the gateway for some reason. Can someone tell me which ports are
>>>>> required and need to be opened on the virtual machine to communicate with
>>>>> Gateway in freeswitch?
>>>>> Since all the ports are not opened in the Virtual Machine, this could
>>>>> help me open up those ports and configure Gateway for freeswitch
>>>>>
>>>>>
>>>>> Much help is needed.
>>>>> It's Urgent
>>>>> Thanks in advance.
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
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From harshita19012 at iiitd.ac.in  Mon Oct 19 08:47:11 2020
From: harshita19012 at iiitd.ac.in (Harshita Pandey)
Date: Mon, 19 Oct 2020 14:17:11 +0530
Subject: [Freeswitch-users] Gateway Configuration on Virtual Machine
Message-ID: <CA+by-=Ux7FwgJSfz+Ppzp1PXaoWFSC0K0nwDvRfQwqYUOHDcMA@mail.gmail.com>

Hi all,
I have installed freeswitch on Virtual Machine. However, when I try to do
the gateway configuration, it gives an error.* Are some additional changes
required to be made in the Virtual machine *while installing freeswitch as
compared to installing freeswitch on the local machine?

I get the following error message

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 sip-ip [192.168.2.215]

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 ext-rtp-ip [103.25.231.107]

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 ext-sip-ip [103.25.231.107]

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 rtp-timeout-sec [300]

2020-10-19 07:18:34.576508 [WARNING] sofia.c:5192 rtp-timeout-sec
deprecated use media_timeout variable.
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 rtp-hold-timeout-sec [1800]

2020-10-19 07:18:34.576508 [WARNING] sofia.c:5199 rtp-hold-timeout-sec
deprecated use media_hold_timeout variable.
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls [false]

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-only [false]

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-bind-params
[transport=tls]
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-sip-port [5081]

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-passphrase []

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-date [true]

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-policy [none]

 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-depth [2]

 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-in-subjects []

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-version
[tlsv1,tlsv1.1,tlsv1.2]

2020-10-19 07:18:34.576508 [INFO] sofia.c:5993 Setting MAX Auth Validity to
0 Attempts
2020-10-19 07:18:34.576508 [NOTICE] sofia.c:6160 Started Profile external
[sofia_reg_external]
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3134 Creating agent for external

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3253 Created agent for external

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3302 Set params for external

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3349 Activated db for external

2020-10-19 07:18:34.576508 [INFO] switch_core_sqldb.c:1870 sofia:external
Starting SQL thread.
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3387 Starting thread for
external
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3034 Launching worker thread for
external
 2020-10-19 07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway 'MySIP'
to profile 'external'
2020-10-19 07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway 'MySIP1'
to profile 'external'
2020-10-19 07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway '
example.com' to profile 'external'
2020-10-19 07:18:35.576510 [NOTICE] sofia_reg.c:453 Registering MySIP1

2020-10-19 07:18:35.576510 [NOTICE] sofia_reg.c:453 Registering MySIP

2020-10-19 07:18:35.756510 [ERR] sofia_reg.c:2469 MySIP1 Failed
Registration with status Service Unavailable [503]. failure #1
2020-10-19 07:18:35.756510 [ERR] sofia_reg.c:2469 MySIP Failed Registration
with status Service Unavailable [503]. failure #1
2020-10-19 07:18:36.576508 [WARNING] sofia_reg.c:510 MySIP1 Failed
Registration [503], setting retry to 5 seconds.
2020-10-19 07:18:36.576508 [WARNING] sofia_reg.c:510 MySIP Failed
Registration [503], setting retry to 30 seconds.
2020-10-19 07:18:42.596508 [NOTICE] sofia_reg.c:453 Registering MySIP1

2020-10-19 07:18:42.956509 [ERR] sofia_reg.c:2469 MySIP1 Failed
Registration with status Service Unavailable [503]. failure #2
2020-10-19 07:18:43.596508 [WARNING] sofia_reg.c:510 MySIP1 Failed
Registration [503], setting retry to 5 seconds.
2020-10-19 07:18:49.596509 [NOTICE] sofia_reg.c:453 Registering MySIP1

2020-10-19 07:18:49.956507 [ERR] sofia_reg.c:2469 MySIP1 Failed
Registration with status Service Unavailable [503]. failure #3
2020-10-19 07:18:50.596510 [WARNING] sofia_reg.c:510 MySIP1 Failed
Registration [503], setting retry to 5 seconds.


Thanks in advance.
Regards
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From martin.paterson at technologywithin.com  Mon Oct 19 15:55:08 2020
From: martin.paterson at technologywithin.com (Martin Paterson)
Date: Mon, 19 Oct 2020 15:55:08 +0000
Subject: [Freeswitch-users] Gateway Configuration on Virtual Machine
In-Reply-To: <CA+by-=Ux7FwgJSfz+Ppzp1PXaoWFSC0K0nwDvRfQwqYUOHDcMA@mail.gmail.com>
References: <CA+by-=Ux7FwgJSfz+Ppzp1PXaoWFSC0K0nwDvRfQwqYUOHDcMA@mail.gmail.com>
Message-ID: <AM8PR10MB4131B1A8C8B1F82CDD6A82B1951E0@AM8PR10MB4131.EURPRD10.PROD.OUTLOOK.COM>

Nothing different required for a VM setup.

The error below (503 Service Unavailable) looks like it’s coming from the gateway you’re talking to, so check the gateway config on FreeSWITCH – are you definitely pointing to a SIP gateway and not something else, for example. Check that the gateway you’re talking to (if it’s yours) is properly set up and running. You could try registering to the gateway directly with a SIP phone and see what happens.

Martin.


Martin Paterson
Development Team
Phone: 0207 953 8840
Email: martin.paterson at technologywithin.com
Chevron Business Park, Limekiln Lane, Southampton, Hampshire, SO45 2QL
Registered Office: CP House, Otterspool Way, Watford, WD25 8JJ, U.K 
​Registered in England No: 5964349 | VAT Number: GB 902 5369 37 
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Harshita Pandey
Sent: 19 October 2020 09:47
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Gateway Configuration on Virtual Machine

Hi all,
I have installed freeswitch on Virtual Machine. However, when I try to do the gateway configuration, it gives an error. Are some additional changes required to be made in the Virtual machine while installing freeswitch as compared to installing freeswitch on the local machine?

I get the following error message

2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 sip-ip [192.168.2.215]
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 ext-rtp-ip [103.25.231.107]
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 ext-sip-ip [103.25.231.107]
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 rtp-timeout-sec [300]
2020-10-19 07:18:34.576508 [WARNING] sofia.c:5192 rtp-timeout-sec deprecated use media_timeout variable.
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 rtp-hold-timeout-sec [1800]
2020-10-19 07:18:34.576508 [WARNING] sofia.c:5199 rtp-hold-timeout-sec deprecated use media_hold_timeout variable.
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls [false]
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-only [false]
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-bind-params [transport=tls]
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-sip-port [5081]
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-passphrase []
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-date [true]
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-policy [none]
 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-depth [2]
 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-in-subjects []
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-version [tlsv1,tlsv1.1,tlsv1.2]
2020-10-19 07:18:34.576508 [INFO] sofia.c:5993 Setting MAX Auth Validity to 0 Attempts
2020-10-19 07:18:34.576508 [NOTICE] sofia.c:6160 Started Profile external [sofia_reg_external]
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3134 Creating agent for external
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3253 Created agent for external
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3302 Set params for external
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3349 Activated db for external
2020-10-19 07:18:34.576508 [INFO] switch_core_sqldb.c:1870 sofia:external Starting SQL thread.
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3387 Starting thread for external
2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3034 Launching worker thread for external
 2020-10-19 07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway 'MySIP' to profile 'external'
2020-10-19 07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway 'MySIP1' to profile 'external'
2020-10-19 07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway 'example.com<http://example.com/>' to profile 'external'
2020-10-19 07:18:35.576510 [NOTICE] sofia_reg.c:453 Registering MySIP1
2020-10-19 07:18:35.576510 [NOTICE] sofia_reg.c:453 Registering MySIP
2020-10-19 07:18:35.756510 [ERR] sofia_reg.c:2469 MySIP1 Failed Registration with status Service Unavailable [503]. failure #1
2020-10-19 07:18:35.756510 [ERR] sofia_reg.c:2469 MySIP Failed Registration with status Service Unavailable [503]. failure #1
2020-10-19 07:18:36.576508 [WARNING] sofia_reg.c:510 MySIP1 Failed Registration [503], setting retry to 5 seconds.
2020-10-19 07:18:36.576508 [WARNING] sofia_reg.c:510 MySIP Failed Registration [503], setting retry to 30 seconds.
2020-10-19 07:18:42.596508 [NOTICE] sofia_reg.c:453 Registering MySIP1
2020-10-19 07:18:42.956509 [ERR] sofia_reg.c:2469 MySIP1 Failed Registration with status Service Unavailable [503]. failure #2
2020-10-19 07:18:43.596508 [WARNING] sofia_reg.c:510 MySIP1 Failed Registration [503], setting retry to 5 seconds.
2020-10-19 07:18:49.596509 [NOTICE] sofia_reg.c:453 Registering MySIP1
2020-10-19 07:18:49.956507 [ERR] sofia_reg.c:2469 MySIP1 Failed Registration with status Service Unavailable [503]. failure #3
2020-10-19 07:18:50.596510 [WARNING] sofia_reg.c:510 MySIP1 Failed Registration [503], setting retry to 5 seconds.


Thanks in advance.
Regards

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From s.safarov at gmail.com  Tue Oct 20 06:17:32 2020
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 20 Oct 2020 09:17:32 +0300
Subject: [Freeswitch-users] exired lettencript cert
Message-ID: <CAHtxdDf3Cnw2+bNAfX6f60RRur=DSAtq3DErm7PKOsPdinpLbQ@mail.gmail.com>

some of the "files.freeswitch.org" servers have expired certificates.
Please update the certificate on all servers.

Details

[safarov at safarov-dell esrp-packer]$ openssl s_client -showcerts -connect
files.freeswitch.org:443
CONNECTED(00000003)
depth=2 O = Digital Signature Trust Co., CN = DST Root CA X3
verify return:1
depth=1 C = US, O = Let's Encrypt, CN = Let's Encrypt Authority X3
verify return:1
depth=0 CN = files.freeswitch.org
verify error:num=10:certificate has expired
notAfter=Oct 17 04:58:08 2020 GMT
verify return:1
depth=0 CN = files.freeswitch.org
notAfter=Oct 17 04:58:08 2020 GMT
verify return:1
---
Certificate chain
 0 s:CN = files.freeswitch.org
   i:C = US, O = Let's Encrypt, CN = Let's Encrypt Authority X3
-----BEGIN CERTIFICATE-----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-----END CERTIFICATE-----
 1 s:C = US, O = Let's Encrypt, CN = Let's Encrypt Authority X3
   i:O = Digital Signature Trust Co., CN = DST Root CA X3
-----BEGIN CERTIFICATE-----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-----END CERTIFICATE-----
---
Server certificate
subject=CN = files.freeswitch.org

issuer=C = US, O = Let's Encrypt, CN = Let's Encrypt Authority X3

---
No client certificate CA names sent
Peer signing digest: SHA256
Peer signature type: RSA
Server Temp Key: ECDH, P-256, 256 bits
---
SSL handshake has read 3289 bytes and written 458 bytes
Verification error: certificate has expired
---
New, TLSv1.2, Cipher is ECDHE-RSA-AES256-GCM-SHA384
Server public key is 2048 bit
Secure Renegotiation IS supported
Compression: NONE
Expansion: NONE
No ALPN negotiated
SSL-Session:
    Protocol  : TLSv1.2
    Cipher    : ECDHE-RSA-AES256-GCM-SHA384
    Session-ID:
52162023FF88BE2656B82283B18706FEB2A75670892C7E4FBF2402FB5FB6C95D
    Session-ID-ctx:
    Master-Key:
E19ADB76D51F6EDBA1207396E41F0C96615F1C064B567385429A62A362B7AD21DBAC1BDF5675B8F9AAD56D5BA32705F0
    PSK identity: None
    PSK identity hint: None
    SRP username: None
    TLS session ticket lifetime hint: 300 (seconds)
    TLS session ticket:
    0000 - 02 86 95 bd 73 2e a3 99-ef b7 a8 d1 4d 56 c7 ea
....s.......MV..
    0010 - b0 99 68 a5 22 42 7f 11-4c e3 93 ea de 82 3a 4d
..h."B..L.....:M
    0020 - 4a 36 94 65 f5 78 e8 8d-e4 0a 59 70 b5 77 2c 2f
J6.e.x....Yp.w,/
    0030 - 10 3c b1 73 c8 f1 0f 5c-1d a3 17 69 70 94 89 e4
.<.s...\...ip...
    0040 - 3c 70 4b f3 64 42 fb 5a-b7 99 06 1a e5 2c e3 f8
<pK.dB.Z.....,..
    0050 - 2f 5d 66 34 df ff 04 e1-e6 b7 e9 3f 87 71 4c 28
/]f4.......?.qL(
    0060 - 48 27 e4 07 0a 95 6f c5-50 54 e1 b3 08 bd ef b5
H'....o.PT......
    0070 - 64 f5 63 a4 ed d0 9b f2-dc f8 9a e0 77 e5 6c 45
d.c.........w.lE
    0080 - 34 3b 76 1c ca ca df 6f-df 6d 3b a7 e6 84 e9 53
4;v....o.m;....S
    0090 - 14 d1 ba 11 24 40 a7 5c-e4 ec 9d e5 79 95 39 7b
....$@.\....y.9{
    00a0 - 76 2c 2f 11 5e 6f 8f b2-e0 59 ac 94 07 45 0f 8d
v,/.^o...Y...E..
    00b0 - 18 6e 0b 5c 3b 04 d6 95-f7 33 d4 ed 66 b3 08 34
.n.\;....3..f..4
    00c0 - 7f 51 9d 8b 52 a6 c1 7e-67 45 00 00 47 59 80 c4
.Q..R..~gE..GY..

    Start Time: 1603174347
    Timeout   : 7200 (sec)
    Verify return code: 10 (certificate has expired)
    Extended master secret: no
---

HTTP/1.1 400 Bad Request
Date: Tue, 20 Oct 2020 06:12:37 GMT
Server: Apache/2.4.25 (Debian)
Strict-Transport-Security: max-age=15552000; includeSubDomains; preload
Content-Length: 313
Connection: close
Content-Type: text/html; charset=iso-8859-1

<!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML 2.0//EN">
<html><head>
<title>400 Bad Request</title>
</head><body>
<h1>Bad Request</h1>
<p>Your browser sent a request that this server could not understand.<br />
</p>
<hr>
<address>Apache/2.4.25 (Debian) Server at files.freeswitch.org Port
443</address>
</body></html>
closed
[safarov at safarov-dell esrp-packer]$

Sergey
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From kokoska.rokoska at post.cz  Tue Oct 20 06:57:41 2020
From: kokoska.rokoska at post.cz (kokoska.rokoska)
Date: Tue, 20 Oct 2020 08:57:41 +0200
Subject: [Freeswitch-users] ring_ready on answered call to play message
Message-ID: <16b661e7-2cec-89ed-1d99-966b68104bf9@post.cz>

Hello,

I need to hang up the b-leg of the answered call, play the message to 
a-leg, and dial a new b-leg after a while when message is still playing. 
How to achieve this, please?

I tried to use ring_ready, but it won't start playing until the new 
b-leg starts ringing.
Any recommendations so I don't have to manage sessions manually?

Everything will be clearer from the following dialplan. I need the 
message to start playing right at ring_ready and not when the callee2 
starts ringing:

<action application='set' data='hangup_after_bridge=false'/>
<action application='set' data='execute_on_answer=sched_hangup +10 
normal_clearing bleg'/>
<action application='bridge' data='sofia/gateway/server/callee1'/>
<action application='set' data='ringback=local_stream://message'/>
<action application='ring_ready'/>
<action application='sleep' data='10000'/>
<action application='set' data='ignore_early_media=true'/>
<action application='bridge' data='sofia/gateway/server/callee2'/>
<action application='hangup'/>

Best regards,

kokoska.rokoska


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From andywolk at gmail.com  Tue Oct 20 09:45:33 2020
From: andywolk at gmail.com (Andrey Volk)
Date: Tue, 20 Oct 2020 13:45:33 +0400
Subject: [Freeswitch-users] exired lettencript cert
In-Reply-To: <CAHtxdDf3Cnw2+bNAfX6f60RRur=DSAtq3DErm7PKOsPdinpLbQ@mail.gmail.com>
References: <CAHtxdDf3Cnw2+bNAfX6f60RRur=DSAtq3DErm7PKOsPdinpLbQ@mail.gmail.com>
Message-ID: <CAD55zR34YRnv=7PtgUkCVb5u12aRJ=03BdMU4B0ZRnS0oSieGQ@mail.gmail.com>

Thanks. Should be fine now.

вт, 20 окт. 2020 г. в 10:43, Sergey Safarov <s.safarov at gmail.com>:

> some of the "files.freeswitch.org" servers have expired certificates.
> Please update the certificate on all servers.
>
> Details
>
> [safarov at safarov-dell esrp-packer]$ openssl s_client -showcerts -connect
> files.freeswitch.org:443
> CONNECTED(00000003)
> depth=2 O = Digital Signature Trust Co., CN = DST Root CA X3
> verify return:1
> depth=1 C = US, O = Let's Encrypt, CN = Let's Encrypt Authority X3
> verify return:1
> depth=0 CN = files.freeswitch.org
> verify error:num=10:certificate has expired
> notAfter=Oct 17 04:58:08 2020 GMT
> verify return:1
> depth=0 CN = files.freeswitch.org
> notAfter=Oct 17 04:58:08 2020 GMT
> verify return:1
> ---
> Certificate chain
>  0 s:CN = files.freeswitch.org
>    i:C = US, O = Let's Encrypt, CN = Let's Encrypt Authority X3
> -----BEGIN CERTIFICATE-----
> MIIFejCCBGKgAwIBAgISA6WIL9TQrycry+w8B8l5HdFkMA0GCSqGSIb3DQEBCwUA
> MEoxCzAJBgNVBAYTAlVTMRYwFAYDVQQKEw1MZXQncyBFbmNyeXB0MSMwIQYDVQQD
> ExpMZXQncyBFbmNyeXB0IEF1dGhvcml0eSBYMzAeFw0yMDA3MTkwNDU4MDhaFw0y
> MDEwMTcwNDU4MDhaMB8xHTAbBgNVBAMTFGZpbGVzLmZyZWVzd2l0Y2gub3JnMIIB
> IjANBgkqhkiG9w0BAQEFAAOCAQ8AMIIBCgKCAQEA2kvIqeeTRraRcWEWGcS+Ozzw
> DEpR68LbqVDsAu4GUmkyTzCrNZGQcA1vl+iyRLlA6aVp2HDPaG35iSQUSoTWD/w+
> kdEUnIlz1EWMmkbBGqy+HNo6R9atlScIABZRUzY6KWDkuaxxk84Fuu+ZsfXIeJUl
> CEURSOjKxpKDiAOIZiCQYj43km14a0JBObIO2hHqdYw0QHn+sIvWVa7kDGG5WqyQ
> WSNUrxKQE3D5SufucKrOEx0Ktgs9NnboMgRjn+9bJefaT/vLPXqTvnKy3kNBwAZo
> E5Na73es8FDtTiDIrGAkWZQNqrD5DHM08ERYHwBzqqtsgcGn5mdBqfvOUSJH3wID
> AQABo4ICgzCCAn8wDgYDVR0PAQH/BAQDAgWgMB0GA1UdJQQWMBQGCCsGAQUFBwMB
> BggrBgEFBQcDAjAMBgNVHRMBAf8EAjAAMB0GA1UdDgQWBBTWzYsuNS1t+dCXV+s7
> xpqtK8k/OTAfBgNVHSMEGDAWgBSoSmpjBH3duubRObemRWXv86jsoTBvBggrBgEF
> BQcBAQRjMGEwLgYIKwYBBQUHMAGGImh0dHA6Ly9vY3NwLmludC14My5sZXRzZW5j
> cnlwdC5vcmcwLwYIKwYBBQUHMAKGI2h0dHA6Ly9jZXJ0LmludC14My5sZXRzZW5j
> cnlwdC5vcmcvMDoGA1UdEQQzMDGCGWZpbGVzLXN5bmMuZnJlZXN3aXRjaC5vcmeC
> FGZpbGVzLmZyZWVzd2l0Y2gub3JnMEwGA1UdIARFMEMwCAYGZ4EMAQIBMDcGCysG
> AQQBgt8TAQEBMCgwJgYIKwYBBQUHAgEWGmh0dHA6Ly9jcHMubGV0c2VuY3J5cHQu
> b3JnMIIBAwYKKwYBBAHWeQIEAgSB9ASB8QDvAHYAsh4FzIuizYogTodm+Su5iiUg
> Z2va+nDnsklTLe+LkF4AAAFzZaatSwAABAMARzBFAiEAhSZUY/ZugqtLtgSnR8Q6
> 2d4H77VHpOp8P3mLsf1pctECIFFh0zMgkbRrC0Vt2DVaprFHs1gWwAk1IbnRJuct
> XTRhAHUAb1N2rDHwMRnYmQCkURX/dxUcEdkCwQApBo2yCJo32RMAAAFzZaatdwAA
> BAMARjBEAiATkNcbjpzgXBhIIMg8F5SuR99wBY9C9mPkoMFVVGnH1QIgEuau5oTS
> UfIlnngJHyZdt9DIyMDywUbejQVjFtvuuewwDQYJKoZIhvcNAQELBQADggEBAIAN
> sFwzLeERpXDtzbID2UJ00BvopM8skXSRN1tQ5wu07IAfNdghLnT+DxuS81A5KUPt
> nkFscKwv1++EnX6+iueZlEnA6er1EeFySbCCrk3024RN8UoAwaCaqRfLA/gdxhPo
> 8e+IJAoL5W5lakXwSPPQhmYch1yk7SJKiWd/svRmK67BK7WM1qy0nFxxBTXscJU1
> Y05rApNIdVm+4E1y8AyGWfiZUuFumitkHjgd46nre+MwrMpF0bIykAjbn3CyzgI5
> 8o+XhbTMTudRORPD3/aK2NtERzZnE24pWuSraDeVtSlkD7+lfBHQvcXrwteTWcpF
> /uSqRuA4JSv+SiLqLJU=
> -----END CERTIFICATE-----
>  1 s:C = US, O = Let's Encrypt, CN = Let's Encrypt Authority X3
>    i:O = Digital Signature Trust Co., CN = DST Root CA X3
> -----BEGIN CERTIFICATE-----
> MIIEkjCCA3qgAwIBAgIQCgFBQgAAAVOFc2oLheynCDANBgkqhkiG9w0BAQsFADA/
> MSQwIgYDVQQKExtEaWdpdGFsIFNpZ25hdHVyZSBUcnVzdCBDby4xFzAVBgNVBAMT
> DkRTVCBSb290IENBIFgzMB4XDTE2MDMxNzE2NDA0NloXDTIxMDMxNzE2NDA0Nlow
> SjELMAkGA1UEBhMCVVMxFjAUBgNVBAoTDUxldCdzIEVuY3J5cHQxIzAhBgNVBAMT
> GkxldCdzIEVuY3J5cHQgQXV0aG9yaXR5IFgzMIIBIjANBgkqhkiG9w0BAQEFAAOC
> AQ8AMIIBCgKCAQEAnNMM8FrlLke3cl03g7NoYzDq1zUmGSXhvb418XCSL7e4S0EF
> q6meNQhY7LEqxGiHC6PjdeTm86dicbp5gWAf15Gan/PQeGdxyGkOlZHP/uaZ6WA8
> SMx+yk13EiSdRxta67nsHjcAHJyse6cF6s5K671B5TaYucv9bTyWaN8jKkKQDIZ0
> Z8h/pZq4UmEUEz9l6YKHy9v6Dlb2honzhT+Xhq+w3Brvaw2VFn3EK6BlspkENnWA
> a6xK8xuQSXgvopZPKiAlKQTGdMDQMc2PMTiVFrqoM7hD8bEfwzB/onkxEz0tNvjj
> /PIzark5McWvxI0NHWQWM6r6hCm21AvA2H3DkwIDAQABo4IBfTCCAXkwEgYDVR0T
> AQH/BAgwBgEB/wIBADAOBgNVHQ8BAf8EBAMCAYYwfwYIKwYBBQUHAQEEczBxMDIG
> CCsGAQUFBzABhiZodHRwOi8vaXNyZy50cnVzdGlkLm9jc3AuaWRlbnRydXN0LmNv
> bTA7BggrBgEFBQcwAoYvaHR0cDovL2FwcHMuaWRlbnRydXN0LmNvbS9yb290cy9k
> c3Ryb290Y2F4My5wN2MwHwYDVR0jBBgwFoAUxKexpHsscfrb4UuQdf/EFWCFiRAw
> VAYDVR0gBE0wSzAIBgZngQwBAgEwPwYLKwYBBAGC3xMBAQEwMDAuBggrBgEFBQcC
> ARYiaHR0cDovL2Nwcy5yb290LXgxLmxldHNlbmNyeXB0Lm9yZzA8BgNVHR8ENTAz
> MDGgL6AthitodHRwOi8vY3JsLmlkZW50cnVzdC5jb20vRFNUUk9PVENBWDNDUkwu
> Y3JsMB0GA1UdDgQWBBSoSmpjBH3duubRObemRWXv86jsoTANBgkqhkiG9w0BAQsF
> AAOCAQEA3TPXEfNjWDjdGBX7CVW+dla5cEilaUcne8IkCJLxWh9KEik3JHRRHGJo
> uM2VcGfl96S8TihRzZvoroed6ti6WqEBmtzw3Wodatg+VyOeph4EYpr/1wXKtx8/
> wApIvJSwtmVi4MFU5aMqrSDE6ea73Mj2tcMyo5jMd6jmeWUHK8so/joWUoHOUgwu
> X4Po1QYz+3dszkDqMp4fklxBwXRsW10KXzPMTZ+sOPAveyxindmjkW8lGy+QsRlG
> PfZ+G6Z6h7mjem0Y+iWlkYcV4PIWL1iwBi8saCbGS5jN2p8M+X+Q7UNKEkROb3N6
> KOqkqm57TH2H3eDJAkSnh6/DNFu0Qg==
> -----END CERTIFICATE-----
> ---
> Server certificate
> subject=CN = files.freeswitch.org
>
> issuer=C = US, O = Let's Encrypt, CN = Let's Encrypt Authority X3
>
> ---
> No client certificate CA names sent
> Peer signing digest: SHA256
> Peer signature type: RSA
> Server Temp Key: ECDH, P-256, 256 bits
> ---
> SSL handshake has read 3289 bytes and written 458 bytes
> Verification error: certificate has expired
> ---
> New, TLSv1.2, Cipher is ECDHE-RSA-AES256-GCM-SHA384
> Server public key is 2048 bit
> Secure Renegotiation IS supported
> Compression: NONE
> Expansion: NONE
> No ALPN negotiated
> SSL-Session:
>     Protocol  : TLSv1.2
>     Cipher    : ECDHE-RSA-AES256-GCM-SHA384
>     Session-ID:
> 52162023FF88BE2656B82283B18706FEB2A75670892C7E4FBF2402FB5FB6C95D
>     Session-ID-ctx:
>     Master-Key:
> E19ADB76D51F6EDBA1207396E41F0C96615F1C064B567385429A62A362B7AD21DBAC1BDF5675B8F9AAD56D5BA32705F0
>     PSK identity: None
>     PSK identity hint: None
>     SRP username: None
>     TLS session ticket lifetime hint: 300 (seconds)
>     TLS session ticket:
>     0000 - 02 86 95 bd 73 2e a3 99-ef b7 a8 d1 4d 56 c7 ea
> ....s.......MV..
>     0010 - b0 99 68 a5 22 42 7f 11-4c e3 93 ea de 82 3a 4d
> ..h."B..L.....:M
>     0020 - 4a 36 94 65 f5 78 e8 8d-e4 0a 59 70 b5 77 2c 2f
> J6.e.x....Yp.w,/
>     0030 - 10 3c b1 73 c8 f1 0f 5c-1d a3 17 69 70 94 89 e4
> .<.s...\...ip...
>     0040 - 3c 70 4b f3 64 42 fb 5a-b7 99 06 1a e5 2c e3 f8
> <pK.dB.Z.....,..
>     0050 - 2f 5d 66 34 df ff 04 e1-e6 b7 e9 3f 87 71 4c 28
> /]f4.......?.qL(
>     0060 - 48 27 e4 07 0a 95 6f c5-50 54 e1 b3 08 bd ef b5
> H'....o.PT......
>     0070 - 64 f5 63 a4 ed d0 9b f2-dc f8 9a e0 77 e5 6c 45
> d.c.........w.lE
>     0080 - 34 3b 76 1c ca ca df 6f-df 6d 3b a7 e6 84 e9 53
> 4;v....o.m;....S
>     0090 - 14 d1 ba 11 24 40 a7 5c-e4 ec 9d e5 79 95 39 7b
> ....$@.\....y.9{
>     00a0 - 76 2c 2f 11 5e 6f 8f b2-e0 59 ac 94 07 45 0f 8d
> v,/.^o...Y...E..
>     00b0 - 18 6e 0b 5c 3b 04 d6 95-f7 33 d4 ed 66 b3 08 34
> .n.\;....3..f..4
>     00c0 - 7f 51 9d 8b 52 a6 c1 7e-67 45 00 00 47 59 80 c4
> .Q..R..~gE..GY..
>
>     Start Time: 1603174347
>     Timeout   : 7200 (sec)
>     Verify return code: 10 (certificate has expired)
>     Extended master secret: no
> ---
>
> HTTP/1.1 400 Bad Request
> Date: Tue, 20 Oct 2020 06:12:37 GMT
> Server: Apache/2.4.25 (Debian)
> Strict-Transport-Security: max-age=15552000; includeSubDomains; preload
> Content-Length: 313
> Connection: close
> Content-Type: text/html; charset=iso-8859-1
>
> <!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML 2.0//EN">
> <html><head>
> <title>400 Bad Request</title>
> </head><body>
> <h1>Bad Request</h1>
> <p>Your browser sent a request that this server could not understand.<br />
> </p>
> <hr>
> <address>Apache/2.4.25 (Debian) Server at files.freeswitch.org Port
> 443</address>
> </body></html>
> closed
> [safarov at safarov-dell esrp-packer]$
>
> Sergey
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From harshita19012 at iiitd.ac.in  Thu Oct 22 15:55:18 2020
From: harshita19012 at iiitd.ac.in (Harshita Pandey)
Date: Thu, 22 Oct 2020 21:25:18 +0530
Subject: [Freeswitch-users] Gateway Configuration on Virtual Machine
In-Reply-To: <AM8PR10MB4131B1A8C8B1F82CDD6A82B1951E0@AM8PR10MB4131.EURPRD10.PROD.OUTLOOK.COM>
References: <CA+by-=Ux7FwgJSfz+Ppzp1PXaoWFSC0K0nwDvRfQwqYUOHDcMA@mail.gmail.com>
 <AM8PR10MB4131B1A8C8B1F82CDD6A82B1951E0@AM8PR10MB4131.EURPRD10.PROD.OUTLOOK.COM>
Message-ID: <CA+by-=XS6UDr7XswU+jGMaFxp=cYJYzsZqE3E8vnz8dGAkLLPA@mail.gmail.com>

Hi,
I am not even able to connect to a softphone. When I had installed the
freeswitch on a local Machine I was able to connect it to a softphone, but
now I have installed freeswitch on a Virtual Machine and that too on a
server.

I have used the following credentials:

Username: 1001
Domain: The IP Address I get on executing eval${local_ip_v4} on the
freeswitch command line
Password: 1234 (the default password).

Where am I doing wrong, or is it that I need to add something else as well,
while installing freeswitch on server or VM while connecting to a
softphone. I have been using a Linphone as a softphone.

Thanks in advance for your help


On Mon, Oct 19, 2020 at 9:26 PM Martin Paterson <
martin.paterson at technologywithin.com> wrote:

> Nothing different required for a VM setup.
>
>
>
> The error below (503 Service Unavailable) looks like it’s coming from the
> gateway you’re talking to, so check the gateway config on FreeSWITCH – are
> you definitely pointing to a SIP gateway and not something else, for
> example. Check that the gateway you’re talking to (if it’s yours) is
> properly set up and running. You could try registering to the gateway
> directly with a SIP phone and see what happens.
>
>
>
> Martin.
>
>
> Development Team​
> <https://technologywithin.com/>
> Phone:  *0207 953 8840* <0207%20953%208840>
> Email:  *martin.paterson at technologywithin.com*
> <martin.paterson at technologywithin.com>
> Chevron Business Park, Limekiln Lane ,  Southampton ,  Hampshire ,
> SO45 2QL
> <https://en-gb.facebook.com/we.are.technologywithin>
> <https://www.linkedin.com/company/technologywithin>
> <https://twitter.com/@techwithin>
> <https://www.instagram.com/technologywithin/>
> <https://zoom.us/webinar/register/6916019886550/WN_HjuT4R-uTD636gPHq18iRA>
> Registered Office: CP House, Otterspool Way, Watford, WD25 8JJ, U.K
> ​Registered in England No: 5964349 | VAT Number: GB 902 5369 37
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Harshita Pandey
> *Sent:* 19 October 2020 09:47
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Gateway Configuration on Virtual Machine
>
>
>
> Hi all,
>
> I have installed freeswitch on Virtual Machine. However, when I try to do
> the gateway configuration, it gives an error.* Are some additional
> changes required to be made in the Virtual machine *while installing
> freeswitch as compared to installing freeswitch on the local machine?
>
>
>
> I get the following error message
>
>
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 sip-ip [192.168.2.215]
>
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 ext-rtp-ip
> [103.25.231.107]
>
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 ext-sip-ip
> [103.25.231.107]
>
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 rtp-timeout-sec [300]
>
>
> 2020-10-19 07:18:34.576508 [WARNING] sofia.c:5192 rtp-timeout-sec
> deprecated use media_timeout variable.
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 rtp-hold-timeout-sec
> [1800]
>
> 2020-10-19 07:18:34.576508 [WARNING] sofia.c:5199 rtp-hold-timeout-sec
> deprecated use media_hold_timeout variable.
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls [false]
>
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-only [false]
>
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-bind-params
> [transport=tls]
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-sip-port [5081]
>
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-passphrase []
>
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-date [true]
>
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-policy [none]
>
>
>  2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-depth [2]
>
>
>  2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-in-subjects []
>
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-version
> [tlsv1,tlsv1.1,tlsv1.2]
>
>
> 2020-10-19 07:18:34.576508 [INFO] sofia.c:5993 Setting MAX Auth Validity
> to 0 Attempts
>
> 2020-10-19 07:18:34.576508 [NOTICE] sofia.c:6160 Started Profile external
> [sofia_reg_external]
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3134 Creating agent for
> external
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3253 Created agent for external
>
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3302 Set params for external
>
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3349 Activated db for external
>
>
> 2020-10-19 07:18:34.576508 [INFO] switch_core_sqldb.c:1870 sofia:external
> Starting SQL thread.
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3387 Starting thread for
> external
>
> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3034 Launching worker thread
> for external
>
>  2020-10-19 07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway
> 'MySIP' to profile 'external'
>
> 2020-10-19 07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway
> 'MySIP1' to profile 'external'
>
> 2020-10-19 07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway '
> example.com' to profile 'external'
>
> 2020-10-19 07:18:35.576510 [NOTICE] sofia_reg.c:453 Registering MySIP1
>
>
> 2020-10-19 07:18:35.576510 [NOTICE] sofia_reg.c:453 Registering MySIP
>
>
> 2020-10-19 07:18:35.756510 [ERR] sofia_reg.c:2469 MySIP1 Failed
> Registration with status Service Unavailable [503]. failure #1
>
> 2020-10-19 07:18:35.756510 [ERR] sofia_reg.c:2469 MySIP Failed
> Registration with status Service Unavailable [503]. failure #1
>
> 2020-10-19 07:18:36.576508 [WARNING] sofia_reg.c:510 MySIP1 Failed
> Registration [503], setting retry to 5 seconds.
>
> 2020-10-19 07:18:36.576508 [WARNING] sofia_reg.c:510 MySIP Failed
> Registration [503], setting retry to 30 seconds.
>
> 2020-10-19 07:18:42.596508 [NOTICE] sofia_reg.c:453 Registering MySIP1
>
>
> 2020-10-19 07:18:42.956509 [ERR] sofia_reg.c:2469 MySIP1 Failed
> Registration with status Service Unavailable [503]. failure #2
>
> 2020-10-19 07:18:43.596508 [WARNING] sofia_reg.c:510 MySIP1 Failed
> Registration [503], setting retry to 5 seconds.
>
> 2020-10-19 07:18:49.596509 [NOTICE] sofia_reg.c:453 Registering MySIP1
>
>
> 2020-10-19 07:18:49.956507 [ERR] sofia_reg.c:2469 MySIP1 Failed
> Registration with status Service Unavailable [503]. failure #3
>
> 2020-10-19 07:18:50.596510 [WARNING] sofia_reg.c:510 MySIP1 Failed
> Registration [503], setting retry to 5 seconds.
>
>
>
>
>
> Thanks in advance.
>
> Regards
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From harshita19012 at iiitd.ac.in  Thu Oct 22 15:56:26 2020
From: harshita19012 at iiitd.ac.in (Harshita Pandey)
Date: Thu, 22 Oct 2020 21:26:26 +0530
Subject: [Freeswitch-users] Not able to connect to a Softphone
Message-ID: <CA+by-=XU0aMDdjctvTcjGSW_Xs12=XZMQA15F9DOcByzH8x7yw@mail.gmail.com>

Hi,
I am not even able to connect to a softphone. When I had installed the
freeswitch on a local Machine I was able to connect it to a softphone, but
now I have installed freeswitch on a Virtual Machine and that too on a
server.

I have used the following credentials:

Username: 1001
Domain: The IP Address I get on executing eval${local_ip_v4} on the
freeswitch command line
Password: 1234 (the default password).

Where am I doing wrong, or is it that I need to add something else as well,
while installing freeswitch on server or VM while connecting to a
softphone. I have been using a Linphone as a softphone.

Thanks in advance for your help
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From gregor at infomedia.si  Thu Oct 22 16:36:02 2020
From: gregor at infomedia.si (Gregor Nanger)
Date: Thu, 22 Oct 2020 18:36:02 +0200
Subject: [Freeswitch-users] Gateway Configuration on Virtual Machine
In-Reply-To: <CA+by-=XS6UDr7XswU+jGMaFxp=cYJYzsZqE3E8vnz8dGAkLLPA@mail.gmail.com>
References: <CA+by-=Ux7FwgJSfz+Ppzp1PXaoWFSC0K0nwDvRfQwqYUOHDcMA@mail.gmail.com>
 <AM8PR10MB4131B1A8C8B1F82CDD6A82B1951E0@AM8PR10MB4131.EURPRD10.PROD.OUTLOOK.COM>
 <CA+by-=XS6UDr7XswU+jGMaFxp=cYJYzsZqE3E8vnz8dGAkLLPA@mail.gmail.com>
Message-ID: <CAC7ttFEfytsDY7D+oWx+5CogW-bhOwybbGwLK4GFx4T6qNV46w@mail.gmail.com>

I think with newer version of FS, it is blocked registration, if you do not
change default password in vars.xml

On Thu, Oct 22, 2020, 17:56 Harshita Pandey <harshita19012 at iiitd.ac.in>
wrote:

> Hi,
> I am not even able to connect to a softphone. When I had installed the
> freeswitch on a local Machine I was able to connect it to a softphone, but
> now I have installed freeswitch on a Virtual Machine and that too on a
> server.
>
> I have used the following credentials:
>
> Username: 1001
> Domain: The IP Address I get on executing eval${local_ip_v4} on the
> freeswitch command line
> Password: 1234 (the default password).
>
> Where am I doing wrong, or is it that I need to add something else as
> well, while installing freeswitch on server or VM while connecting to a
> softphone. I have been using a Linphone as a softphone.
>
> Thanks in advance for your help
>
>
> On Mon, Oct 19, 2020 at 9:26 PM Martin Paterson <
> martin.paterson at technologywithin.com> wrote:
>
>> Nothing different required for a VM setup.
>>
>>
>>
>> The error below (503 Service Unavailable) looks like it’s coming from the
>> gateway you’re talking to, so check the gateway config on FreeSWITCH – are
>> you definitely pointing to a SIP gateway and not something else, for
>> example. Check that the gateway you’re talking to (if it’s yours) is
>> properly set up and running. You could try registering to the gateway
>> directly with a SIP phone and see what happens.
>>
>>
>>
>> Martin.
>>
>>
>> Development Team​
>> <https://technologywithin.com/>
>> Phone:  *0207 953 8840* <0207%20953%208840>
>> Email:  *martin.paterson at technologywithin.com*
>> <martin.paterson at technologywithin.com>
>> Chevron Business Park, Limekiln Lane ,  Southampton ,  Hampshire ,
>> SO45 2QL
>> <https://en-gb.facebook.com/we.are.technologywithin>
>> <https://www.linkedin.com/company/technologywithin>
>> <https://twitter.com/@techwithin>
>> <https://www.instagram.com/technologywithin/>
>> <https://zoom.us/webinar/register/6916019886550/WN_HjuT4R-uTD636gPHq18iRA>
>> Registered Office: CP House, Otterspool Way, Watford, WD25 8JJ, U.K
>> ​Registered in England No: 5964349 | VAT Number: GB 902 5369 37
>>
>> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
>> Behalf Of *Harshita Pandey
>> *Sent:* 19 October 2020 09:47
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* [Freeswitch-users] Gateway Configuration on Virtual Machine
>>
>>
>>
>> Hi all,
>>
>> I have installed freeswitch on Virtual Machine. However, when I try to do
>> the gateway configuration, it gives an error.* Are some additional
>> changes required to be made in the Virtual machine *while installing
>> freeswitch as compared to installing freeswitch on the local machine?
>>
>>
>>
>> I get the following error message
>>
>>
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 sip-ip [192.168.2.215]
>>
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 ext-rtp-ip
>> [103.25.231.107]
>>
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 ext-sip-ip
>> [103.25.231.107]
>>
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 rtp-timeout-sec [300]
>>
>>
>> 2020-10-19 07:18:34.576508 [WARNING] sofia.c:5192 rtp-timeout-sec
>> deprecated use media_timeout variable.
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 rtp-hold-timeout-sec
>> [1800]
>>
>> 2020-10-19 07:18:34.576508 [WARNING] sofia.c:5199 rtp-hold-timeout-sec
>> deprecated use media_hold_timeout variable.
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls [false]
>>
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-only [false]
>>
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-bind-params
>> [transport=tls]
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-sip-port [5081]
>>
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-passphrase []
>>
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-date [true]
>>
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-policy [none]
>>
>>
>>  2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-depth [2]
>>
>>
>>  2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-verify-in-subjects
>> []
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:4597 tls-version
>> [tlsv1,tlsv1.1,tlsv1.2]
>>
>>
>> 2020-10-19 07:18:34.576508 [INFO] sofia.c:5993 Setting MAX Auth Validity
>> to 0 Attempts
>>
>> 2020-10-19 07:18:34.576508 [NOTICE] sofia.c:6160 Started Profile external
>> [sofia_reg_external]
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3134 Creating agent for
>> external
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3253 Created agent for
>> external
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3302 Set params for external
>>
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3349 Activated db for external
>>
>>
>> 2020-10-19 07:18:34.576508 [INFO] switch_core_sqldb.c:1870 sofia:external
>> Starting SQL thread.
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3387 Starting thread for
>> external
>>
>> 2020-10-19 07:18:34.576508 [DEBUG] sofia.c:3034 Launching worker thread
>> for external
>>
>>  2020-10-19 07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway
>> 'MySIP' to profile 'external'
>>
>> 2020-10-19 07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway
>> 'MySIP1' to profile 'external'
>>
>> 2020-10-19 07:18:34.576508 [NOTICE] sofia_reg.c:3426 Added gateway '
>> example.com' to profile 'external'
>>
>> 2020-10-19 07:18:35.576510 [NOTICE] sofia_reg.c:453 Registering MySIP1
>>
>>
>> 2020-10-19 07:18:35.576510 [NOTICE] sofia_reg.c:453 Registering MySIP
>>
>>
>> 2020-10-19 07:18:35.756510 [ERR] sofia_reg.c:2469 MySIP1 Failed
>> Registration with status Service Unavailable [503]. failure #1
>>
>> 2020-10-19 07:18:35.756510 [ERR] sofia_reg.c:2469 MySIP Failed
>> Registration with status Service Unavailable [503]. failure #1
>>
>> 2020-10-19 07:18:36.576508 [WARNING] sofia_reg.c:510 MySIP1 Failed
>> Registration [503], setting retry to 5 seconds.
>>
>> 2020-10-19 07:18:36.576508 [WARNING] sofia_reg.c:510 MySIP Failed
>> Registration [503], setting retry to 30 seconds.
>>
>> 2020-10-19 07:18:42.596508 [NOTICE] sofia_reg.c:453 Registering MySIP1
>>
>>
>> 2020-10-19 07:18:42.956509 [ERR] sofia_reg.c:2469 MySIP1 Failed
>> Registration with status Service Unavailable [503]. failure #2
>>
>> 2020-10-19 07:18:43.596508 [WARNING] sofia_reg.c:510 MySIP1 Failed
>> Registration [503], setting retry to 5 seconds.
>>
>> 2020-10-19 07:18:49.596509 [NOTICE] sofia_reg.c:453 Registering MySIP1
>>
>>
>> 2020-10-19 07:18:49.956507 [ERR] sofia_reg.c:2469 MySIP1 Failed
>> Registration with status Service Unavailable [503]. failure #3
>>
>> 2020-10-19 07:18:50.596510 [WARNING] sofia_reg.c:510 MySIP1 Failed
>> Registration [503], setting retry to 5 seconds.
>>
>>
>>
>>
>>
>> Thanks in advance.
>>
>> Regards
>>
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From mario_fs at mgtech.com  Thu Oct 22 17:36:13 2020
From: mario_fs at mgtech.com (mario_fs)
Date: Thu, 22 Oct 2020 10:36:13 -0700
Subject: [Freeswitch-users] How to bridge to external using URL,
	which is correct?
Message-ID: <CE71C9EE-A381-4043-9905-99D9B64F621A@mgtech.com>

I am trying to bridge an incoming call to an external number(s). I currently have this working for years to normal cell phones. I want to change it to call my ITSP instead of a mobile number but having problems. I have ready the wiki and many posts but they have many variations. Have tried with registered and unregistered. I am missing something.

The question is what is the correct/best format, I can register the outbound # since it’s my account, the inbound is already registered.
Thanks very much! Mario G

 I tried #1 below and it does not work. 

1. sofia/gateway/${dial_gateway}/sip:number at itsp.com <sip:number at itsp.com>. <- NOGO

2. WIki has these but I am really puzzled but the use of “internal": 
data="sofia/internal/9998881111 at sip.yourprovider.com”
OR
 data="{sip_auth_username=foo,sip_auth_password=bar}sofia/internal/9998881111 at sip.yourprovider.com”/>

Wiki also shows this example which I use to call cell #, but does not work for directions ITSP call:
sofia/gateway/asterlink/18005551212

And found this:
action application="bridge" data="sofia/gateway/providerA/$1
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From david.villasmil.work at gmail.com  Thu Oct 22 21:31:45 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 22 Oct 2020 22:31:45 +0100
Subject: [Freeswitch-users] How to bridge to external using URL,
	which is correct?
In-Reply-To: <CE71C9EE-A381-4043-9905-99D9B64F621A@mgtech.com>
References: <CE71C9EE-A381-4043-9905-99D9B64F621A@mgtech.com>
Message-ID: <CAFGRPVpq0=6p7AfDZq84JHunQVp2qMeCdHvJj9wFAxVGMPG=+Q@mail.gmail.com>

I’m not really sure I understand what you’re trying to do.

If you already have a gateway pointing to your service provider, i.e.: it
is registered on your provider, simply doing

sofia/gateway/GATEWAYNAME/+1234567890

Should work. That’s just saying “call +1234567890 using the gateway named
GATEWAYNAME which is the xml gateway you created which registers on the
service provider.

About using internal/external and calling directly like

sofia/internal/+1234567890 at SERVICEPROVIDER_IP

Is just saying send a call to ip SERVICEPROVIDER_IP with phone number
+1234567890 using the internal profile, which is bound to whatever port it
is using (usually 5060). This one doesn’t do authentication. (Unless you
add it like you said, setting user/pass on the bridge, but I’ve never done
it that way)

On Thu, 22 Oct 2020 at 18:40, mario_fs <mario_fs at mgtech.com> wrote:

> I am trying to bridge an incoming call to an external number(s). I
> currently have this working for years to normal cell phones. I want to
> change it to call my ITSP instead of a mobile number but having problems. I
> have ready the wiki and many posts but they have many variations. Have
> tried with registered and unregistered. I am missing something.
>
> The question is what is the correct/best format, I can register the
> outbound # since it’s my account, the inbound is already registered.
> Thanks very much! Mario G
>
>  I tried #1 below and it does not work.
>
> 1. sofia/gateway/${dial_gateway}/sip:number at itsp.com. <- NOGO
>
> 2. WIki has these but I am really puzzled but the use of “internal":
> data="sofia/internal/9998881111 at sip.yourprovider.com”
> OR
>  data="{sip_auth_username=foo,
> sip_auth_password=bar}sofia/internal/9998881111 at sip.yourprovider.com”/>
>
> Wiki also shows this example which I use to call cell #, but does not work
> for directions ITSP call:
> sofia/gateway/asterlink/18005551212
>
> And found this:
> action application="bridge" data="sofia/gateway/providerA/$1
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From mario_fs at mgtech.com  Fri Oct 23 17:10:41 2020
From: mario_fs at mgtech.com (mario_fs)
Date: Fri, 23 Oct 2020 10:10:41 -0700
Subject: [Freeswitch-users] How to bridge to external using URL,
 which is correct?
In-Reply-To: <CAFGRPVpq0=6p7AfDZq84JHunQVp2qMeCdHvJj9wFAxVGMPG=+Q@mail.gmail.com>
References: <CE71C9EE-A381-4043-9905-99D9B64F621A@mgtech.com>
 <CAFGRPVpq0=6p7AfDZq84JHunQVp2qMeCdHvJj9wFAxVGMPG=+Q@mail.gmail.com>
Message-ID: <6B2F60E8-3588-4B1F-A9C5-F19E381EC892@mgtech.com>

Thank you. I did wind up using the first one, but may try the second one to see if original callerid will be passed. FYI here is what I am doing:

1 Call inbound to FS.
2 Bridge to local phones, iPads, and an 18 second delay to external unregistered ITSP extension.
3 The unregistered extension at the ITSP fowards to 2 other registered extensions on same account which ring cell phones.

The only bummer is that callerid is not forwarded even though I am providing it in the bridge. The only way to get caller ID to pass is to use VPN and register iPhones as local extensions but can’t keep VPN on permanently.

I setup VPN per app on IO but it will not work with Bria. I am changing from calling the 2 mobile numbers directly because sometimes mobile VM is tripped because of 30 second max ringing instead of allowing FS to handle it and we miss messages. Now the timeout is handled via the bridge parameters.

Mario G 

> On Oct 22, 2020, at 2:31 PM, David Villasmil <david.villasmil.work at gmail.com> wrote:
> 
> I’m not really sure I understand what you’re trying to do.
> 
> If you already have a gateway pointing to your service provider, i.e.: it is registered on your provider, simply doing
> 
> sofia/gateway/GATEWAYNAME/+1234567890 
> 
> Should work. That’s just saying “call +1234567890 using the gateway named GATEWAYNAME which is the xml gateway you created which registers on the service provider.
> 
> About using internal/external and calling directly like 
> 
> sofia/internal/+1234567890 at SERVICEPROVIDER_IP
> 
> Is just saying send a call to ip SERVICEPROVIDER_IP with phone number +1234567890 using the internal profile, which is bound to whatever port it is using (usually 5060). This one doesn’t do authentication. (Unless you add it like you said, setting user/pass on the bridge, but I’ve never done it that way)
> 
> On Thu, 22 Oct 2020 at 18:40, mario_fs <mario_fs at mgtech.com <mailto:mario_fs at mgtech.com>> wrote:
> I am trying to bridge an incoming call to an external number(s). I currently have this working for years to normal cell phones. I want to change it to call my ITSP instead of a mobile number but having problems. I have ready the wiki and many posts but they have many variations. Have tried with registered and unregistered. I am missing something.
> 
> The question is what is the correct/best format, I can register the outbound # since it’s my account, the inbound is already registered.
> Thanks very much! Mario G
> 
>  I tried #1 below and it does not work. 
> 
> 1. sofia/gateway/${dial_gateway}/sip:number at itsp.com <>. <- NOGO
> 
> 2. WIki has these but I am really puzzled but the use of “internal": 
> data="sofia/internal/9998881111 at sip.yourprovider.com <mailto:sofia/internal/9998881111 at sip.yourprovider.com>”
> OR
>  data="{sip_auth_username=foo,sip_auth_password=bar}sofia/internal/9998881111 at sip.yourprovider.com <mailto:sip_auth_password=bar%7Dsofia/internal/9998881111 at sip.yourprovider.com>”/>
> 
> Wiki also shows this example which I use to call cell #, but does not work for directions ITSP call:
> sofia/gateway/asterlink/18005551212
> 
> And found this:
> action application="bridge" data="sofia/gateway/providerA/$1
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>-- 
> Regards,
> 
> David Villasmil
> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
> phone: +34669448337
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From david.villasmil.work at gmail.com  Fri Oct 23 17:57:19 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 23 Oct 2020 18:57:19 +0100
Subject: [Freeswitch-users] How to bridge to external using URL,
	which is correct?
In-Reply-To: <6B2F60E8-3588-4B1F-A9C5-F19E381EC892@mgtech.com>
References: <CE71C9EE-A381-4043-9905-99D9B64F621A@mgtech.com>
 <CAFGRPVpq0=6p7AfDZq84JHunQVp2qMeCdHvJj9wFAxVGMPG=+Q@mail.gmail.com>
 <6B2F60E8-3588-4B1F-A9C5-F19E381EC892@mgtech.com>
Message-ID: <CAFGRPVqaf8a+Xvu7YdN5+tr21A2UOxHwokOYtOh=8W33pXyBKA@mail.gmail.com>

You need to set in the gateway configuration

<param name=“caller-id-in-from” value=”true”/>



On Fri, 23 Oct 2020 at 18:10, mario_fs <mario_fs at mgtech.com> wrote:

> Thank you. I did wind up using the first one, but may try the second one
> to see if original callerid will be passed. FYI here is what I am doing:
>
> 1 Call inbound to FS.
> 2 Bridge to local phones, iPads, and an 18 second delay to external
> unregistered ITSP extension.
> 3 The unregistered extension at the ITSP fowards to 2 other registered
> extensions on same account which ring cell phones.
>
> The only bummer is that callerid is not forwarded even though I am
> providing it in the bridge. The only way to get caller ID to pass is to use
> VPN and register iPhones as local extensions but can’t keep VPN on
> permanently.
>
> I setup VPN per app on IO but it will not work with Bria. I am changing
> from calling the 2 mobile numbers directly because sometimes mobile VM is
> tripped because of 30 second max ringing instead of allowing FS to handle
> it and we miss messages. Now the timeout is handled via the bridge
> parameters.
>
> Mario G
>
> On Oct 22, 2020, at 2:31 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
> I’m not really sure I understand what you’re trying to do.
>
> If you already have a gateway pointing to your service provider, i.e.: it
> is registered on your provider, simply doing
>
> sofia/gateway/GATEWAYNAME/+1234567890
>
> Should work. That’s just saying “call +1234567890 using the gateway named
> GATEWAYNAME which is the xml gateway you created which registers on the
> service provider.
>
> About using internal/external and calling directly like
>
> sofia/internal/+1234567890 at SERVICEPROVIDER_IP
>
> Is just saying send a call to ip SERVICEPROVIDER_IP with phone number
> +1234567890 using the internal profile, which is bound to whatever port it
> is using (usually 5060). This one doesn’t do authentication. (Unless you
> add it like you said, setting user/pass on the bridge, but I’ve never done
> it that way)
>
> On Thu, 22 Oct 2020 at 18:40, mario_fs <mario_fs at mgtech.com> wrote:
>
>> I am trying to bridge an incoming call to an external number(s). I
>> currently have this working for years to normal cell phones. I want to
>> change it to call my ITSP instead of a mobile number but having problems. I
>> have ready the wiki and many posts but they have many variations. Have
>> tried with registered and unregistered. I am missing something.
>>
>> The question is what is the correct/best format, I can register the
>> outbound # since it’s my account, the inbound is already registered.
>> Thanks very much! Mario G
>>
>>  I tried #1 below and it does not work.
>>
>> 1. sofia/gateway/${dial_gateway}/sip:number at itsp.com. <- NOGO
>>
>> 2. WIki has these but I am really puzzled but the use of “internal":
>> data="sofia/internal/9998881111 at sip.yourprovider.com”
>> OR
>>  data="{sip_auth_username=foo,
>> sip_auth_password=bar}sofia/internal/9998881111 at sip.yourprovider.com”/>
>>
>> Wiki also shows this example which I use to call cell #, but does not
>> work for directions ITSP call:
>> sofia/gateway/asterlink/18005551212
>>
>> And found this:
>> action application="bridge" data="sofia/gateway/providerA/$1
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From martyn at magiccow.co.uk  Sat Oct 24 08:41:01 2020
From: martyn at magiccow.co.uk (Martyn Davies)
Date: Sat, 24 Oct 2020 09:41:01 +0100
Subject: [Freeswitch-users] Lost ACKs and BYE (ack timeout)
Message-ID: <CA++4AS6uAW5uWf8SscQujZ=td4DuntJe=9dN7_t1a3g_=So4JA@mail.gmail.com>

I have Freeswitch running in an Alpine VM with a Sangoma gateway
downstream.  Until recently I was using Freeswitch 16 and it was working
fine, but due to an OS upgrade, it seemed reasonable to upgrade to
Freeswitch 18 (the version that comes with that Alpine release). However,
now upstream calls from the Sangoma fail.  Specifically, when FS sends the
200 OK (for the Invite), the Sangoma now does not reply with ACK, and so FS
tears down the call after 30 sec.

I tried the 'connectile dysfunction' parameter, but this did not seem to
have any positive effect.  In fact, it was perhaps worse because the
Register (from Sangoma to FS) seems to get stuck in a loop, with Sangoma
sending the request over and over back to back.

I would be interested to hear if this rings a bell with anyone?  Are there
any crucial differences in base config between 16 and 18 that would change
the behaviour?

Regards,
Martyn
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From flavio at voffice.com.br  Sat Oct 24 09:15:58 2020
From: flavio at voffice.com.br (Flavio Goncalves)
Date: Sat, 24 Oct 2020 06:15:58 -0300
Subject: [Freeswitch-users] Lost ACKs and BYE (ack timeout)
In-Reply-To: <CA++4AS6uAW5uWf8SscQujZ=td4DuntJe=9dN7_t1a3g_=So4JA@mail.gmail.com>
References: <CA++4AS6uAW5uWf8SscQujZ=td4DuntJe=9dN7_t1a3g_=So4JA@mail.gmail.com>
Message-ID: <CAJkFROxV8Vk0X_hmgr1z1e3+OBMVFbgnNfOMbq1N80UfHxO=+g@mail.gmail.com>

ACKs are sent to what you have in Contact and Record-Route Headers of the
200OK. Check if your Contact is not pointing somewhere else.

Flavio E. Goncalves


Em sáb., 24 de out. de 2020 às 05:52, Martyn Davies <martyn at magiccow.co.uk>
escreveu:

> I have Freeswitch running in an Alpine VM with a Sangoma gateway
> downstream.  Until recently I was using Freeswitch 16 and it was working
> fine, but due to an OS upgrade, it seemed reasonable to upgrade to
> Freeswitch 18 (the version that comes with that Alpine release). However,
> now upstream calls from the Sangoma fail.  Specifically, when FS sends the
> 200 OK (for the Invite), the Sangoma now does not reply with ACK, and so FS
> tears down the call after 30 sec.
>
> I tried the 'connectile dysfunction' parameter, but this did not seem to
> have any positive effect.  In fact, it was perhaps worse because the
> Register (from Sangoma to FS) seems to get stuck in a loop, with Sangoma
> sending the request over and over back to back.
>
> I would be interested to hear if this rings a bell with anyone?  Are there
> any crucial differences in base config between 16 and 18 that would change
> the behaviour?
>
> Regards,
> Martyn
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From mario_fs at mgtech.com  Mon Oct 26 20:25:54 2020
From: mario_fs at mgtech.com (mario_fs)
Date: Mon, 26 Oct 2020 13:25:54 -0700
Subject: [Freeswitch-users] How to bridge to external using URL,
 which is correct?
In-Reply-To: <CAFGRPVqaf8a+Xvu7YdN5+tr21A2UOxHwokOYtOh=8W33pXyBKA@mail.gmail.com>
References: <CE71C9EE-A381-4043-9905-99D9B64F621A@mgtech.com>
 <CAFGRPVpq0=6p7AfDZq84JHunQVp2qMeCdHvJj9wFAxVGMPG=+Q@mail.gmail.com>
 <6B2F60E8-3588-4B1F-A9C5-F19E381EC892@mgtech.com>
 <CAFGRPVqaf8a+Xvu7YdN5+tr21A2UOxHwokOYtOh=8W33pXyBKA@mail.gmail.com>
Message-ID: <538E0D02-0A2E-4CF0-BACE-820145372427@mgtech.com>

Thanks again David for the last tip. It did not work but I am posting how I got it to work in case this info is useful to others:

I could not get the callerid passed no way using a variety of methods using a gateway. The param mentioned did not work. Years ago, passing callerid through worked but due to spoofing I think ITSPs are eliminating this capability.

I gave up and just for kicks tried INTERNAL, and got it to work! I tried to pass sip auth name, password and realm and I kept getting not authorized even though I used the correct info. Then, I remembered my ITSP callcentric.com had info that stated if a URI came in from a non-callcentric account use “in.callcentric.com <http://in.callcentric.com/>”. Since there is no gateway using internal, I removed the sip authorization, add “in." to the url, and bingo, the call went through, forwarded to push server, rang both iPhones, and passed the callerid! One caveat though, I had people call me to test and I always got the number not always the name. For instance, had someone call from Home Depot, I got the number but no name. Calls from landlines worked for both. Will need to look into this further but at least now the callerid is back when Freeswitch bridges a call to our iPhones.

Hope this helps someone someday. I worked years on this, all January, and finally all the last 3 days. My line is a simple:

sofia/internal/12345678901130 at in.callcentric.com <mailto:sofia/internal/12345678901130 at in.callcentric.com>
The number is the account number followed by 3 digit extension (130).

Mario G

> On Oct 23, 2020, at 10:57 AM, David Villasmil <david.villasmil.work at gmail.com> wrote:
> 
> You need to set in the gateway configuration 
> 
> <param name=“caller-id-in-from” value=”true”/>
> 
> 
> 
> On Fri, 23 Oct 2020 at 18:10, mario_fs <mario_fs at mgtech.com <mailto:mario_fs at mgtech.com>> wrote:
> Thank you. I did wind up using the first one, but may try the second one to see if original callerid will be passed. FYI here is what I am doing:
> 
> 1 Call inbound to FS.
> 2 Bridge to local phones, iPads, and an 18 second delay to external unregistered ITSP extension.
> 3 The unregistered extension at the ITSP fowards to 2 other registered extensions on same account which ring cell phones.
> 
> The only bummer is that callerid is not forwarded even though I am providing it in the bridge. The only way to get caller ID to pass is to use VPN and register iPhones as local extensions but can’t keep VPN on permanently.
> 
> I setup VPN per app on IO but it will not work with Bria. I am changing from calling the 2 mobile numbers directly because sometimes mobile VM is tripped because of 30 second max ringing instead of allowing FS to handle it and we miss messages. Now the timeout is handled via the bridge parameters.
> 
> Mario G 
> 
>> On Oct 22, 2020, at 2:31 PM, David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
>> 
>> I’m not really sure I understand what you’re trying to do.
>> 
>> If you already have a gateway pointing to your service provider, i.e.: it is registered on your provider, simply doing
>> 
>> sofia/gateway/GATEWAYNAME/+1234567890 
>> 
>> Should work. That’s just saying “call +1234567890 using the gateway named GATEWAYNAME which is the xml gateway you created which registers on the service provider.
>> 
>> About using internal/external and calling directly like 
>> 
>> sofia/internal/+1234567890 at SERVICEPROVIDER_IP
>> 
>> Is just saying send a call to ip SERVICEPROVIDER_IP with phone number +1234567890 using the internal profile, which is bound to whatever port it is using (usually 5060). This one doesn’t do authentication. (Unless you add it like you said, setting user/pass on the bridge, but I’ve never done it that way)
>> 
>> On Thu, 22 Oct 2020 at 18:40, mario_fs <mario_fs at mgtech.com <mailto:mario_fs at mgtech.com>> wrote:
>> I am trying to bridge an incoming call to an external number(s). I currently have this working for years to normal cell phones. I want to change it to call my ITSP instead of a mobile number but having problems. I have ready the wiki and many posts but they have many variations. Have tried with registered and unregistered. I am missing something.
>> 
>> The question is what is the correct/best format, I can register the outbound # since it’s my account, the inbound is already registered.
>> Thanks very much! Mario G
>> 
>>  I tried #1 below and it does not work. 
>> 
>> 1. sofia/gateway/${dial_gateway}/sip:number at itsp.com <>. <- NOGO
>> 
>> 2. WIki has these but I am really puzzled but the use of “internal": 
>> data="sofia/internal/9998881111 at sip.yourprovider.com <mailto:sofia/internal/9998881111 at sip.yourprovider.com>”
>> OR
>>  data="{sip_auth_username=foo,sip_auth_password=bar}sofia/internal/9998881111 at sip.yourprovider.com <mailto:sip_auth_password=bar%7Dsofia/internal/9998881111 at sip.yourprovider.com>”/>
>> 
>> Wiki also shows this example which I use to call cell #, but does not work for directions ITSP call:
>> sofia/gateway/asterlink/18005551212
>> 
>> And found this:
>> action application="bridge" data="sofia/gateway/providerA/$1
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>> Build your next product on our scalable cloud platform.
>> 
>> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>> https://cluecon.com <https://cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> https://freeswitch.com <https://freeswitch.com/>-- 
>> Regards,
>> 
>> David Villasmil
>> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
>> phone: +34669448337
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>> Build your next product on our scalable cloud platform.
>> 
>> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>> https://cluecon.com <https://cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> https://freeswitch.com <https://freeswitch.com/>
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>-- 
> Regards,
> 
> David Villasmil
> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
> phone: +34669448337
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From vladislaus at gmail.com  Tue Oct 27 16:16:40 2020
From: vladislaus at gmail.com (Andres Gomez)
Date: Tue, 27 Oct 2020 11:16:40 -0500
Subject: [Freeswitch-users] Installation Fs 1.10.5 problem
Message-ID: <CAHzfA-ifxpeBiYx-GueiOC=_CWXiFWUg0CThFvh0vAXWd-V1EQ@mail.gmail.com>

Hi all.

I have a problem after installing standard vanilla. It doesn't start with
fs_cli, and when I try to run it manually I see the following:


2020-10-27 10:54:00.042884 [NOTICE] sofia.c:6195 Started Profile
external-ipv6 [sofia_reg_external-ipv6]
2020-10-27 10:54:00.043067 [ERR] sofia.c:5083 Invalid ext-rtp-ip
2020-10-27 10:54:00.043074 [ERR] sofia.c:5180 Invalid ext-sip-ip
2020-10-27 10:54:00.043080 [NOTICE] switch_event.c:464 Subclass reservation
deleted for mod_sofia.c:sofia::notify_refer
2020-10-27 10:54:00.043084 [NOTICE] switch_event.c:464 Subclass reservation
deleted for mod_sofia.c:sofia::notify_watched_header
2020-10-27 10:54:00.043087 [NOTICE] switch_event.c:464 Subclass reservation
deleted for mod_sofia.c:sofia::unregister
2020-10-27 10:54:00.043089 [NOTICE] switch_event.c:464 Subclass reservation
deleted for mod_sofia.c:sofia::profile_start
2020-10-27 10:54:00.043092 [NOTICE] switch_event.c:464 Subclass reservation
deleted for mod_sofia.c:sofia::reinvite
2020-10-27 10:54:00.043095 [NOTICE] switch_event.c:464 Subclass reservation
deleted for mod_sofia.c:sofia::replaced
2020-10-27 10:54:00.043098 [NOTICE] switch_event.c:464 Subclass reservation
deleted for mod_sofia.c:sofia::transferor
2020-10-27 10:54:00.043100 [NOTICE] switch_event.c:464 Subclass reservation
deleted for mod_sofia.c:sofia::transferee
2020-10-27 10:54:00.043103 [NOTICE] switch_event.c:464 Subclass reservation
deleted for mod_sofia.c:sofia::error
2020-10-27 10:54:00.043105 [NOTICE] switch_event.c:464 Subclass reservation
deleted for mod_sofia.c:sofia::intercepted
2020-10-27 10:54:00.043108 [NOTICE] switch_event.c:464 Subclass reservation
deleted for mod_sofia.c:sofia::gateway_state
2020-10-27 10:54:00.043111 [NOTICE] switch_event.c:464 Subclass reservation
deleted for mod_sofia.c:sofia::sip_user_state
2020-10-27 10:54:00.043113 [NOTICE] switch_event.c:464 Subclass reservation
deleted for mod_sofia.c:sofia::gateway_delete
2020-10-27 10:54:00.043117 [NOTICE] switch_event.c:464 Subclass reservation
deleted for mod_sofia.c:sofia::expire
2020-10-27 10:54:00.043120 [NOTICE] switch_event.c:464 Subclass reservation
deleted for mod_sofia.c:sofia::register_attempt
*2020-10-27 10:54:00.043123 [NOTICE] switch_event.c:464 Subclass
reservation deleted for mod_sofia.c:sofia::register_failureSegmentation
fault (core dumped)*


I´m installing with Debian 10 and :.
https://freeswitch.org/confluence/display/FREESWITCH/Debian+10+Buster

Regards
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From david.villasmil.work at gmail.com  Tue Oct 27 19:01:37 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 27 Oct 2020 19:01:37 +0000
Subject: [Freeswitch-users] Installation Fs 1.10.5 problem
In-Reply-To: <CAHzfA-ifxpeBiYx-GueiOC=_CWXiFWUg0CThFvh0vAXWd-V1EQ@mail.gmail.com>
References: <CAHzfA-ifxpeBiYx-GueiOC=_CWXiFWUg0CThFvh0vAXWd-V1EQ@mail.gmail.com>
Message-ID: <CAFGRPVphfDf6+XGua-6SqULWxz3Ri7y7dgAHAyRbHscp0tDpbw@mail.gmail.com>

Have you tried setting

2020-10-27 10:54:00.043067 [ERR] sofia.c:5083 Invalid ext-rtp-ip
2020-10-27 10:54:00.043074 [ERR] sofia.c:5180 Invalid ext-sip-ip

Manually and try again?

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337


On Tue, Oct 27, 2020 at 4:17 PM Andres Gomez <vladislaus at gmail.com> wrote:

> Hi all.
>
> I have a problem after installing standard vanilla. It doesn't start with
> fs_cli, and when I try to run it manually I see the following:
>
>
> 2020-10-27 10:54:00.042884 [NOTICE] sofia.c:6195 Started Profile
> external-ipv6 [sofia_reg_external-ipv6]
> 2020-10-27 10:54:00.043067 [ERR] sofia.c:5083 Invalid ext-rtp-ip
> 2020-10-27 10:54:00.043074 [ERR] sofia.c:5180 Invalid ext-sip-ip
> 2020-10-27 10:54:00.043080 [NOTICE] switch_event.c:464 Subclass
> reservation deleted for mod_sofia.c:sofia::notify_refer
> 2020-10-27 10:54:00.043084 [NOTICE] switch_event.c:464 Subclass
> reservation deleted for mod_sofia.c:sofia::notify_watched_header
> 2020-10-27 10:54:00.043087 [NOTICE] switch_event.c:464 Subclass
> reservation deleted for mod_sofia.c:sofia::unregister
> 2020-10-27 10:54:00.043089 [NOTICE] switch_event.c:464 Subclass
> reservation deleted for mod_sofia.c:sofia::profile_start
> 2020-10-27 10:54:00.043092 [NOTICE] switch_event.c:464 Subclass
> reservation deleted for mod_sofia.c:sofia::reinvite
> 2020-10-27 10:54:00.043095 [NOTICE] switch_event.c:464 Subclass
> reservation deleted for mod_sofia.c:sofia::replaced
> 2020-10-27 10:54:00.043098 [NOTICE] switch_event.c:464 Subclass
> reservation deleted for mod_sofia.c:sofia::transferor
> 2020-10-27 10:54:00.043100 [NOTICE] switch_event.c:464 Subclass
> reservation deleted for mod_sofia.c:sofia::transferee
> 2020-10-27 10:54:00.043103 [NOTICE] switch_event.c:464 Subclass
> reservation deleted for mod_sofia.c:sofia::error
> 2020-10-27 10:54:00.043105 [NOTICE] switch_event.c:464 Subclass
> reservation deleted for mod_sofia.c:sofia::intercepted
> 2020-10-27 10:54:00.043108 [NOTICE] switch_event.c:464 Subclass
> reservation deleted for mod_sofia.c:sofia::gateway_state
> 2020-10-27 10:54:00.043111 [NOTICE] switch_event.c:464 Subclass
> reservation deleted for mod_sofia.c:sofia::sip_user_state
> 2020-10-27 10:54:00.043113 [NOTICE] switch_event.c:464 Subclass
> reservation deleted for mod_sofia.c:sofia::gateway_delete
> 2020-10-27 10:54:00.043117 [NOTICE] switch_event.c:464 Subclass
> reservation deleted for mod_sofia.c:sofia::expire
> 2020-10-27 10:54:00.043120 [NOTICE] switch_event.c:464 Subclass
> reservation deleted for mod_sofia.c:sofia::register_attempt
> *2020-10-27 10:54:00.043123 [NOTICE] switch_event.c:464 Subclass
> reservation deleted for mod_sofia.c:sofia::register_failureSegmentation
> fault (core dumped)*
>
>
> I´m installing with Debian 10 and :.
> https://freeswitch.org/confluence/display/FREESWITCH/Debian+10+Buster
>
> Regards
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From shahnawaj.khan1990 at gmail.com  Tue Oct 27 19:03:59 2020
From: shahnawaj.khan1990 at gmail.com (Shahnawaj Khan)
Date: Wed, 28 Oct 2020 00:33:59 +0530
Subject: [Freeswitch-users] Recording file is bigger than the call duration
Message-ID: <CAGmow11Co0LCHjpm0NejaR8wtMycWbO4n_B=TWiB6Ot+MzPtUQ@mail.gmail.com>

Hi,

I am working on freeswitch 1.10.2 and using record_session to record
the media of the calls. I found a case where call recording was
greater than the call duration by 4 seconds. I was running freeswitch
in debug logs but nothing seems off in the logs. But I found some
error logs in xml cdr and unable to find an explanation for those in
any of the confluence pages. Please let me know if someone has the
idea of following error logs.
    <error-log>
        <error-period>
          <start>1603359922800633</start>
          <stop>1603359925820633</stop>
          <flaws>2</flaws>
          <consecutive-flaws>1</consecutive-flaws>
          <duration-msec>3020</duration-msec>
        </error-period>
        <error-period>
          <start>1603359908820633</start>
          <stop>1603359911840633</stop>
          <flaws>3</flaws>
          <consecutive-flaws>1</consecutive-flaws>
          <duration-msec>3020</duration-msec>
        </error-period>
        <error-period>
          <start>1603359757800632</start>
          <stop>1603359760820633</stop>
          <flaws>3</flaws>
          <consecutive-flaws>1</consecutive-flaws>
          <duration-msec>3020</duration-msec>
        </error-period>
        <error-period>
          <start>1603359751800633</start>
          <stop>1603359754820633</stop>
          <flaws>3</flaws>
          <consecutive-flaws>1</consecutive-flaws>
          <duration-msec>3020</duration-msec>
        </error-period>
        <error-period>
          <start>1603359742800633</start>
          <stop>1603359746000634</stop>
          <flaws>6</flaws>
          <consecutive-flaws>2</consecutive-flaws>
          <duration-msec>3200</duration-msec>
        </error-period>
        <error-period>
          <start>1603359601780633</start>
          <stop>1603359604800633</stop>
          <flaws>3</flaws>
          <consecutive-flaws>1</consecutive-flaws>
          <duration-msec>3020</duration-msec>
        </error-period>
        <error-period>
          <start>1603359566800632</start>
          <stop>1603359569820633</stop>
          <flaws>3</flaws>
          <consecutive-flaws>1</consecutive-flaws>
          <duration-msec>3020</duration-msec>
        </error-period>
        <error-period>
          <start>1603359497800634</start>
          <stop>1603359500820633</stop>
          <flaws>3</flaws>
          <consecutive-flaws>1</consecutive-flaws>
          <duration-msec>3019</duration-msec>
        </error-period>
      </error-log>

Thanks & Regards,
Shahnawaz


From brian at freeswitch.com  Tue Oct 27 19:57:38 2020
From: brian at freeswitch.com (Brian West)
Date: Tue, 27 Oct 2020 14:57:38 -0500
Subject: [Freeswitch-users] Installation Fs 1.10.5 problem
In-Reply-To: <CAFGRPVphfDf6+XGua-6SqULWxz3Ri7y7dgAHAyRbHscp0tDpbw@mail.gmail.com>
References: <CAHzfA-ifxpeBiYx-GueiOC=_CWXiFWUg0CThFvh0vAXWd-V1EQ@mail.gmail.com>
 <CAFGRPVphfDf6+XGua-6SqULWxz3Ri7y7dgAHAyRbHscp0tDpbw@mail.gmail.com>
Message-ID: <CAEJMVkDtKuLnS1=dvj7domPVK7TGQ02_VnKGQcpko9V6QN-m_A@mail.gmail.com>

This is because those are set to an empty space, and that causes Sofia to
crash on the second try.  The first one fails but attempting to load
mod_sofia again :boom:

On Tue, Oct 27, 2020 at 2:44 PM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Have you tried setting
>
> 2020-10-27 10:54:00.043067 [ERR] sofia.c:5083 Invalid ext-rtp-ip
> 2020-10-27 10:54:00.043074 [ERR] sofia.c:5180 Invalid ext-sip-ip
>
> Manually and try again?
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
>
> On Tue, Oct 27, 2020 at 4:17 PM Andres Gomez <vladislaus at gmail.com> wrote:
>
>> Hi all.
>>
>> I have a problem after installing standard vanilla. It doesn't start
>> with fs_cli, and when I try to run it manually I see the following:
>>
>>
>> 2020-10-27 10:54:00.042884 [NOTICE] sofia.c:6195 Started Profile
>> external-ipv6 [sofia_reg_external-ipv6]
>> 2020-10-27 10:54:00.043067 [ERR] sofia.c:5083 Invalid ext-rtp-ip
>> 2020-10-27 10:54:00.043074 [ERR] sofia.c:5180 Invalid ext-sip-ip
>> 2020-10-27 10:54:00.043080 [NOTICE] switch_event.c:464 Subclass
>> reservation deleted for mod_sofia.c:sofia::notify_refer
>> 2020-10-27 10:54:00.043084 [NOTICE] switch_event.c:464 Subclass
>> reservation deleted for mod_sofia.c:sofia::notify_watched_header
>> 2020-10-27 10:54:00.043087 [NOTICE] switch_event.c:464 Subclass
>> reservation deleted for mod_sofia.c:sofia::unregister
>> 2020-10-27 10:54:00.043089 [NOTICE] switch_event.c:464 Subclass
>> reservation deleted for mod_sofia.c:sofia::profile_start
>> 2020-10-27 10:54:00.043092 [NOTICE] switch_event.c:464 Subclass
>> reservation deleted for mod_sofia.c:sofia::reinvite
>> 2020-10-27 10:54:00.043095 [NOTICE] switch_event.c:464 Subclass
>> reservation deleted for mod_sofia.c:sofia::replaced
>> 2020-10-27 10:54:00.043098 [NOTICE] switch_event.c:464 Subclass
>> reservation deleted for mod_sofia.c:sofia::transferor
>> 2020-10-27 10:54:00.043100 [NOTICE] switch_event.c:464 Subclass
>> reservation deleted for mod_sofia.c:sofia::transferee
>> 2020-10-27 10:54:00.043103 [NOTICE] switch_event.c:464 Subclass
>> reservation deleted for mod_sofia.c:sofia::error
>> 2020-10-27 10:54:00.043105 [NOTICE] switch_event.c:464 Subclass
>> reservation deleted for mod_sofia.c:sofia::intercepted
>> 2020-10-27 10:54:00.043108 [NOTICE] switch_event.c:464 Subclass
>> reservation deleted for mod_sofia.c:sofia::gateway_state
>> 2020-10-27 10:54:00.043111 [NOTICE] switch_event.c:464 Subclass
>> reservation deleted for mod_sofia.c:sofia::sip_user_state
>> 2020-10-27 10:54:00.043113 [NOTICE] switch_event.c:464 Subclass
>> reservation deleted for mod_sofia.c:sofia::gateway_delete
>> 2020-10-27 10:54:00.043117 [NOTICE] switch_event.c:464 Subclass
>> reservation deleted for mod_sofia.c:sofia::expire
>> 2020-10-27 10:54:00.043120 [NOTICE] switch_event.c:464 Subclass
>> reservation deleted for mod_sofia.c:sofia::register_attempt
>> *2020-10-27 10:54:00.043123 [NOTICE] switch_event.c:464 Subclass
>> reservation deleted for mod_sofia.c:sofia::register_failureSegmentation
>> fault (core dumped)*
>>
>>
>> I´m installing with Debian 10 and :.
>> https://freeswitch.org/confluence/display/FREESWITCH/Debian+10+Buster
>>
>> Regards
>>
>>
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>

Email: brian at freeswitch.com

Mobile: 918-424-9378

Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>

[image: https://www.facebook.com/signalwireinc?src=email]
<https://www.facebook.com/freeswitch> [image:
https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
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From toraritte at gmail.com  Wed Oct 28 13:23:18 2020
From: toraritte at gmail.com (=?UTF-8?Q?Guly=C3=A1s_Attila?=)
Date: Wed, 28 Oct 2020 13:23:18 +0000
Subject: [Freeswitch-users] FreeSWITCH patch: `uuid_fileman` to preserve
	pitch on playback speed change
Message-ID: <CADnELph-kWQ__oQKVXaKBYdQCyq74GV5k9NWc9ZxZnVz2u8uUw@mail.gmail.com>

Good Morning,

Would it be possible to merge this pull request, and add it to the
next FreeSWITCH release?
https://github.com/signalwire/freeswitch/pull/244

Tried to get a response on Slack and by commenting in the pull
request, but no joy. I totally understand that all the devs are busy,
and that this is probably one of the lowest priority issues out there,
but here's some context why I am being a pest about this:

Our Access News phone service is for visually and print impaired
individuals who are unable to consume conventional print materials. We
just replaced an old, on-premises system with aFreeSWITCH solution,
and it has been a dream to work with, except this one flaw. Non-visual
users rely heavily on changing the playback speed to efficiently
browse through vast amounts of audio content, and the current
behaviour is very disruptive, especially because many legally blind
folks have other disabilities as well (such as cognitive and hearing
impairments).

Please let us know if there's anything we could do to facilitate this
process. Thank you!

Appreciatively,
Attila Gulyas  |  IT/Program Assistant
Email: agulyas at societyfortheblind.org
Phone: (916) 889-7510
Access News helpdesk: (916) 889-7519

Access News service:
(800) 665-4667
(916) 732-4000
https://societyfortheblind.org/programs-services/access-news/

Society for the Blind
1238 S Street
Sacramento, CA 95811

SFTB Main Phone: (916) 452-8271
Fax: (916) 492-2483
www.societyfortheblind.org

Our mission is to empower individuals living with low vision or
blindness to discover, develop and achieve their full potential.


From tom at tomlynn.com  Fri Oct 30 02:48:31 2020
From: tom at tomlynn.com (Tom Lynn)
Date: Thu, 29 Oct 2020 19:48:31 -0700
Subject: [Freeswitch-users] Conference with max-members =2 allows a third
	caller
Message-ID: <CAERdjiYNnGpt73HdMe0Gu56ooLMrnU1CG3oM5RONLTUAAPia1A@mail.gmail.com>

I've set a new conference profile, to which I've added max-members and set
it equal to 2. I've assigned the new profile to a conference, but the limit
does not seem to be enforced. The conference allows a 3rd participant to
join. Is there anything more I need to do to limit the number of callers
besides setting max-members = X ?
Thanks
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From sarkmobilesolutions at gmail.com  Thu Oct 22 05:55:14 2020
From: sarkmobilesolutions at gmail.com (Sark Mobiles)
Date: Thu, 22 Oct 2020 11:25:14 +0530
Subject: [Freeswitch-users] Fwd: v1.10 - Able to answer call but no Audio
In-Reply-To: <CAJuiBy-MbrR3vKPtKt+2m97U3BkX6OMWREXuoYKEHB73GypXQQ@mail.gmail.com>
References: <CAJuiBy-MbrR3vKPtKt+2m97U3BkX6OMWREXuoYKEHB73GypXQQ@mail.gmail.com>
Message-ID: <CAJuiBy8-9OXvwOao_0TfvwocSH16wXGqj0N7f0o2ayPhk7VGiA@mail.gmail.com>

Hello,

We installed and configured Freeswitch on CentOS as per the steps provided
in below link,

https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7#CentOS7andRHEL7-CentOS7andRHEL7-Stable

This is the status -
* Users are getting registered
* When User A calls User B, B gets a ringtone and is able to answer the call
* A sees that the call is connected but there is no audio for either A or B.

netstat output
---------------------

tcp        0      0 <SVR_IP>:5060        0.0.0.0:*
LISTEN      10503/freeswitch
tcp        0      0 <SVR_IP>:8081        0.0.0.0:*
LISTEN      10503/freeswitch
tcp        0      0 <SVR_IP>:8082        0.0.0.0:*
LISTEN      10503/freeswitch
tcp        0      0 0.0.0.0:8021         0.0.0.0:*
LISTEN      10503/freeswitch
tcp        0      0 <SVR_IP>:5080        0.0.0.0:*
LISTEN      10503/freeswitch


Log output

---------------

2020-10-21 08:32:12.323087 [NOTICE] switch_ivr_originate.c:3794
Channel [sofia/internal/1001@<SVR_IP>] has been answered
2020-10-21 08:32:12.323087 [DEBUG] switch_channel.c:3865
(sofia/internal/1001@<SVR_IP>) Callstate Change EARLY -> ACTIVE
2020-10-21 08:32:12.323087 [DEBUG] sofia.c:7326 Channel
sofia/internal/1001@<SVR_IP> entering state [completed][200]
2020-10-21 08:32:12.323087 [DEBUG] switch_ivr_originate.c:3852
Originate Resulted in Success:
[sofia/internal/1000 at 192.168.43.32:18486]
2020-10-21 08:32:12.323087 [DEBUG] switch_ivr_originate.c:3852
Originate Resulted in Success:
[sofia/internal/1000 at 192.168.43.32:18486]
2020-10-21 08:32:12.323087 [DEBUG] switch_ivr_bridge.c:1793
(sofia/internal/1000 at 192.168.43.32:18486) State Change
CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2020-10-21 08:32:12.323087 [DEBUG] switch_core_state_machine.c:585
(sofia/internal/1000 at 192.168.43.32:18486) Running State Change
CS_EXCHANGE_MEDIA (Cur 2 Tot 6)
2020-10-21 08:32:12.323087 [DEBUG] switch_core_state_machine.c:654
(sofia/internal/1000 at 192.168.43.32:18486) State EXCHANGE_MEDIA
2020-10-21 08:32:12.323087 [DEBUG] mod_sofia.c:656 SOFIA EXCHANGE_MEDIA
2020-10-21 08:32:12.723235 [DEBUG] sofia.c:7326 Channel
sofia/internal/1001@<SVR_IP> entering state [ready][200]
2020-10-21 08:32:16.123236 [DEBUG] switch_rtp.c:2448 RTCP packet not written
2020-10-21 08:32:20.143147 [DEBUG] switch_rtp.c:2448 RTCP packet not written
2020-10-21 08:32:24.163187 [DEBUG] switch_rtp.c:2448 RTCP packet not written

This keeps repeating - "RTCP packet not written"


Please let us know what we might be doing wrong. Appreciate your help.

Regards,

Krishnan
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From rroch at tutanota.de  Mon Oct 19 12:41:44 2020
From: rroch at tutanota.de (Rachel Roch)
Date: Mon, 19 Oct 2020 14:41:44 +0200 (CEST)
Subject: No audio on SRTP/ZRTP
Message-ID: <MK-sIj---3-2@tutanota.de>

Hi,

I have achieved a functional Freeswitch configuration in that I can call both Freeswitch hosted services (e.g. talking clock) and make outbound VoIP calls through upstream provider.

However this is only achievable using Linphone in Transport TLS + Encryption None config.

The moment I switch Linphone Encryption to TLS or ZRTP, I get complete loss of audio (for both Freeswitch hosted and outbound VoIP).

I have tried expanding the list of ciphers in "sip_tls_ciphers" and "rtp_sdes_suites" (both in vars.xml) but that makes no difference.

Evidently SSL certs are correctly setup because otherwise Transport TLS would not work.

I'm not quite sure where to go from here ?

Laura


From n5d9xq3ti233xiyif2vp at protonmail.ch  Sun Oct 18 11:52:52 2020
From: n5d9xq3ti233xiyif2vp at protonmail.ch (Laura Smith)
Date: Sun, 18 Oct 2020 11:52:52 +0000
Subject: sofia.c:3901 ERROR: unsupported transport
Message-ID: <zrTDZBQAC5nm00Ksrf_Ft5qOb5ZWL_2ZMLrhTJEiSFigu_LCMl3urj4tV_qmS4-vrvKKi3rQfWgX5K43cneRmGG-yeSK49R59nfNQXgEPgA=@protonmail.ch>

Hi,

Any ideas what this obscure message means ?   sofia.c:3901 ERROR: unsupported transport.  It shows up when running a "sofia profile external rescan" (and originating calls doesn't work either).

My provider sip_profile was copied over from a known good other machine. So I don't know what sofia is complaining about ?

<include>
        <gateway name="foobar">
           <param name="auth-username" value="none"/>
           <param name="password" value="none"/>
           <param name="extension" value="default"/>
           <param name="proxy" value="foobar.example.com"/>
           <param name="expire-seconds" value="300"/>
           <param name="register" value="false" />
           <param name="register-transport" value="tls"/>
           <param name="ping" value="false" />
         </gateway>
</include>


From mourad at bilog.fr  Tue Oct 20 09:44:40 2020
From: mourad at bilog.fr (Mourad Hedfi)
Date: Tue, 20 Oct 2020 11:44:40 +0200
Subject: [Freeswitch-users] Configure Freeswith with ovh SIP TRUNK
Message-ID: <CAJkBiSY+=MToB7EsLkSxpnMFfbSHPQEP_Krf2JpawWRJoPg69A@mail.gmail.com>

Hello everyone,

We have installed FreeSwitch with BigBlueButton to allow users to access
BBB meetings with their phone.

Alos, we bought a SIP TRUNK (5 Channels) from OVH and we have configured
the system through this
https://docs.ovh.com/fr/voip/freeswitch-configuration-et-utilisation/

But, only one person can access the meeting.

When we add the extensions, we can't access the meeting and the dial fails.

For that, can anyone tell me if I must have distinct phone numbers of the
SIP TRUNK number to create the extensions ?

What's the role of those extension and are they obligatory ?

We really need your help. Many thanks in advance.

Best regards.
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From makafre at gmail.com  Wed Oct 21 14:28:34 2020
From: makafre at gmail.com (Frederic Jean)
Date: Wed, 21 Oct 2020 14:28:34 +0000
Subject: [Freeswitch-users] mod_conference: recommended values for comfort
	noise and energy level?
Message-ID: <em4629e301-07e3-427d-9851-25f030b45c34@fred-pc>


Hello group,

We have setup a new conference service and it works well, except for 
some complaints here and there about the voice quality;

I was then wondering what are the recommended values for comfort noise 
and energy level as per your experience for conferences?

My feeling is that comfort noise should be disabled but I am unsure 
about the energy level; is this a try and see parameter, or should it 
ideally simply be set to 0 or a small value for a conference call 
system?

Presently:

<profile name="confmodreception">
       <param name="domain" value="$${domain}"/>
       <param name="rate" value="16000"/>
       <param name="ivr-dtmf-timeout" value="2000"/>
       <param name="interval" value="20"/>
       <param name="caller-controls" value="reception"/>
       <param name="moderator-controls" value="reception"/>
       <param name="energy-level" value="200"/>
       <param name="alone-sound" value="conference/conf-alone.wav"/>
       <!-- <param name="moh-sound" value="silence"/> -->
       <param name="comfort-noise" value="true"/>
       <param name="suppress-events" 
value="start-talking,stop-talking,energy-level,volume-level,gain-level,energy-level-member,play-file,floor-change,play-file-done,play-file-member"/>
       <param name="enter-sound" 
value="tone_stream://%(200,0,500,600,700)"/>
       <param name="exit-sound" 
value="tone_stream://%(500,0,300,200,100,50,25)"/>
       <param name="muted-sound" 
value="/usr/local/freeswitch/sounds/fr/ca/june/conference/16000/conf-muted.wav"/>
       <param name="unmuted-sound" 
value="/usr/local/freeswitch/sounds/fr/ca/june/conference/16000/conf-unmuted.wav"/>
       <param name="terminate-on-silence" value="120"/>
       <param name="locked-sound" 
value="/usr/local/freeswitch/sounds/fr/ca/june/conference/16000/conf-locked.wav"/>
       <param name="is-locked-sound" 
value="/usr/local/freeswitch/sounds/fr/ca/june/conference/16000/conf-is-locked.wav"/>
       <param name="is-unlocked-sound" 
value="/usr/local/freeswitch/sounds/fr/ca/june/conference/16000/conf-is-unlocked.wav"/>
     </profile>


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From cris.espinosa at ringcentral.com  Tue Oct 27 18:55:28 2020
From: cris.espinosa at ringcentral.com (Cris Espinosa)
Date: Tue, 27 Oct 2020 18:55:28 +0000
Subject: sqlite db
Message-ID: <SG2PR03MB30031833248B7EDB288A7778FD160@SG2PR03MB3003.apcprd03.prod.outlook.com>


Hello,

https://lists.freeswitch.org/pipermail/freeswitch-users/2018-January/128600.html<https://urldefense.com/v3/__https://lists.freeswitch.org/pipermail/freeswitch-users/2018-January/128600.html__;!!J-1DKIBqn-Pi!T1HTvq_hHdBanmAvN7dD1p-dHkqvaKFB2JTMwE5XhyYPRFeZgCQhBA_6ZWUAYEwaGa5A7g$>

I'm following up on a thread from a couple of years ago where Antonio asked Geoff if we were using wss endpoints in our freeswitch setup. The answer to that question is yes. I realize this was a couple of years ago, but we are still having unexplained crashes with our freeswitch instances and switching to mysql did not seem to correct the issue.

Is there some concern with wss endpoints and sqlite? The version this issue was occuring on was 1.6.20.

As a side note, we are currently working on migrating this setup to 1.10.5 and possibly moving back to sqlite


Cris Espinosa
COMMUNICATE
COLLABORATE
CONNECT
SoftWare Engineer


[http://go.ringcentral.com/rs/ringcentral/images/newlogoforsig.png]

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From alaghu.sundram at orchestrate.com  Thu Oct 29 09:38:59 2020
From: alaghu.sundram at orchestrate.com (Alaghu Sundram)
Date: Thu, 29 Oct 2020 15:08:59 +0530
Subject: [Freeswitch-users] Regarding register issue
Message-ID: <C2241BCE25F644C79890AC436BE10C57@AdminstratorPC>

Hi,

I register vertility  through freeswitch  
  I got message as "expire time in 60 by request of proxy sip: sofia_reg. C: 2458 Changing expire time in 60 by request of proxy "
  I also found error registering other VoIP Registration failed. "Too much hope list" 
 
  Thanks 
  R.Alaghu Sundram
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From chad at apartmentlines.com  Sat Oct 31 00:37:25 2020
From: chad at apartmentlines.com (Chad Phillips)
Date: Fri, 30 Oct 2020 17:37:25 -0700
Subject: [Freeswitch-users] US local DID provider recommendations
Message-ID: <CAJ0sghRMG9aWF7gMSZuOz2goh9EfNKM6Auhqkhf4gT-U6m8+rg@mail.gmail.com>

I'm on the hunt for a secondary DID provider, and would love any
recommendations.

Requirements:

 - Local DIDs in the continental US, with good coverage of the various
NPA-NXXs
 - Per-minute billing, not per trunk
 - Some kind of server/IP/host failover configuration capability, in case
my primary switch is down
 - Instant number provisioning
 - Supports SIP Redirects (surprisingly, 2 of my 3 current providers do not)

Chad
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From david.villasmil.work at gmail.com  Sat Oct 31 10:54:54 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 31 Oct 2020 10:54:54 +0000
Subject: [Freeswitch-users] Regarding register issue
In-Reply-To: <C2241BCE25F644C79890AC436BE10C57@AdminstratorPC>
References: <C2241BCE25F644C79890AC436BE10C57@AdminstratorPC>
Message-ID: <CAFGRPVpEmmu8fPe6JqzNKM2RsEU=75WZVqWYwgjmgmY8XXnqDw@mail.gmail.com>

Check the proxy side. It looks like it doesn’t think the REGISTER is for
itself, so it forwards it (to itself) and a loop happens.
On the proxy, make sure whatever fs is sending in the URI, is configured as
an alias.

On Fri, 30 Oct 2020 at 16:46, Alaghu Sundram <alaghu.sundram at orchestrate.com>
wrote:

> Hi,
>
> I register vertility  through freeswitch
>   I got message as "expire time in 60 by request of proxy sip: sofia_reg.
> C: 2458 Changing expire time in 60 by request of proxy "
>   I also found error registering other VoIP Registration failed. "Too much
> hope list"
>   Thanks
>   R.Alaghu Sundram
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From dion at openlogic.com.au  Sat Oct 31 14:54:07 2020
From: dion at openlogic.com.au (Dion Phillips)
Date: Sat, 31 Oct 2020 22:54:07 +0800
Subject: [Freeswitch-users] Second incoming call terminates after 32 seconds
Message-ID: <3f22a582-454d-d482-4712-813ba4b4f61d@openlogic.com.au>

Hi

I have Freeswitch setup on a cloud server and Grandstream phones in the 
office connected successfully to the switch. I have never had any issues 
with NAT causing the phones or Freeswitch to lose their connection. The 
phones have "keep-alive" set so are always sending "OPTIONS" messages to 
Freeswitch to keep the ports open. I don't use the default 5060 port for 
the internal profile on the Freeswitch side.

The office has a Fortigate firewall and a Opnsense box that is used to 
connect the office to a DC cloud server.

The issue is that when a second call comes into the office, it will 
terminate after 32 seconds. There are only 2 voip lines so max 2 calls 
at a time. This only occurs if the second call is inbound. What is even 
more weird is that if the second call is answered by the phone that is 
already on a call, then this does not happen. If the second call is 
outbound, there is also no issue.

I have done a sip trace and the calls progress correctly from CALL SETUP 
-> IN CALL -> COMPLETED. The logfile however has an ORIGINATOR_CANCEL 
message when the call is terminated. I have two phones at my house 
connected to the same PBX and when I use them to test, I cannot get the 
call to drop which suggest to me there is something in the office that 
is dropping packets but only on the second call.

If the firewall was an issue, why would the first incoming call work and 
all outgoing calls work also? I have tried creating a rule on the 
firewall to allow all traffic from the Freeswitch IP but that make no 
difference. If the SIP trunk was the issue why would the home phones work.

Can anyone give me some pointers on what I should be looking for in a 
SIP trace or tcpdump or loglfile or tea leaves?

Thanks
Dion.