From alex at freeswitch.com  Mon Oct  1 08:28:53 2018
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Mon, 1 Oct 2018 17:28:53 +0900
Subject: [Freeswitch-users] RTP problems when using "soft" timer
In-Reply-To: <CADZpXoXcmUpgKvo2U=VVRzRTPy0jPFZZq_-N+8WR4j7XOvpnaw@mail.gmail.com>
References: <CADZpXoXcmUpgKvo2U=VVRzRTPy0jPFZZq_-N+8WR4j7XOvpnaw@mail.gmail.com>
Message-ID: <CAGJLwFj3od3Af4hDVN31ywL6gw8fRXMxuKU0EVxCzkG2dvnO6g@mail.gmail.com>

Hi,

Try without virtualisation first, it may screw your timers.

Regards,

Alex

On Fri, Sep 28, 2018 at 12:44 AM Antonio Modesto <modesto916 at gmail.com>
wrote:

> Hi everyone,
>
> I have had some problems lately with poor call quality when using FXO
> channels (mod_khomp) talking to sip phones. Using wireshark I noticed that
> the RTP stream sent from FS to my SIP device had some packets with wrong
> timestamps. After changing the rtp timer from soft to posix, the problem
> was solved. Do you guys know if there is any side effect in using the posix
> timer?
>
> *I am running FS 1.6.20 on a XenServer VM with Debian (Kernel 3.16.0). I
> can't upgrade to FS 1.8 because Khomp still does not support it.
>
> --
> Antônio Modesto
> _________________________________________________________________________
> 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



-- 
Alex Sibyakin | Support Engineer
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: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.
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From gmaruzz at gmail.com  Mon Oct  1 08:59:29 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 1 Oct 2018 10:59:29 +0200
Subject: [Freeswitch-users] Video Upgrade/Downgrade and Transcoding
In-Reply-To: <CAOCjumG8pyCS=eMvtpGp+3OeowsYFfk+yK5z9M_pRiOPbiDmKQ@mail.gmail.com>
References: <CAOCjumG8pyCS=eMvtpGp+3OeowsYFfk+yK5z9M_pRiOPbiDmKQ@mail.gmail.com>
Message-ID: <CALXCt0q9cs7=rMuBri+tEV+WEV5NNwfvJjfRTMoSPf5zS0QO2Q@mail.gmail.com>

Hello  Carsten,

Is the sending of video codecs to audio only endpoints causing some
problems, or is only useless but no problems?

-giovanni


On Wed, Sep 26, 2018, 14:56 Carsten Bock <carsten at ng-voice.com> wrote:

> Hi,
>
> I am using FreeSwitch Release:
> FreeSWITCH (Version 1.9.0 -n20180903T123919Z-1~jessie+1git 5dd4451
> 2018-08-31 19:05:39Z 64bit) is ready
>
> I have the following, rather simple scenario, where I can't figure
> out, how to configure it properly. I use FreeSwitch as Transcoding
> B2B-UA/SBC. All the logic is done on a Kamailio in Front of
> FreeSwitch. Thus my dialplan is rather simple:
>
>   <context name="public">
>     <extension name="other">
>       <condition>
>         <action application="bridge"
> data="sofia/internal/${destination_number}@${loadbalancer}"/>
>       </condition>
>     </extension>
>   </context>
>
> In order to provide transcoding, I have set the following in my
> internal.xml profile:
>
>     <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
>     <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
>
>     <param name="disable-transcoding" value="false"/>
>     <param name="inherit_codec" value="false"/>
>
>     <!--set to 'greedy' if you want your codec list to take precedence -->
>     <param name="inbound-codec-negotiation" value="generous"/>
>
>     <!-- Let calls hit the dialplan before selecting codec for the a-leg
> -->
>     <param name="inbound-late-negotiation" value="true"/>
>
> In vars.xml, I have the following:
>
>   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA"/>
>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,PCMU,PCMA"/>
>
>   <X-PRE-PROCESS cmd="set" data="media_mix_inbound_outbound_codecs=true"/>
>   <X-PRE-PROCESS cmd="set" data="rtp_pass_codecs_on_stream_change=true"/>
>   <X-PRE-PROCESS cmd="set" data="sip_renegotiate_codec_on_reinvite=true"/>
>
> This works like a charm for all Voice-Calls, the B-Party gets an offer
> including all the codecs from the list, e.g. A-Party supports only
> G722, B-Party supports only PCMU, so FreeSwitch does the transcoding.
>
> However, it leads to issues, when it comes to Video. As soon as I add
> H264 and VP8 to the codecs list (global_codec_pref /
> outbound_codec_prefs), the call to my B-Party includes the offer for
> Video as well, even if the A-Party did not offer Video initially. Is
> there an easy way to solve this? What are the right settings for such
> scenario?
>
> I tried the following in my dialplan:
>
>       <!-- Here we check for Video to add it back into the SDP if we want
> -->
>       <condition field="${switch_r_sdp}" expression="/(.*)(m=video
> \d+)(.*)/s" break="never">
>         <anti-action application="export"
> data="nolocal:absolute_codec_string=G722,PCMA,PCMU"/>
>       </condition>
>
> This would remove the Video from the initial invite, if the A-Party
> did not offer Video initially. However, if I then enable Video at a
> later stage (Re-INVITE with m=video), FreeSwitch will not forward the
> new stream to the B-Party and then instead just send a re-INVITE with
> Audio only to the B-Party.
>
> Where did I go wrong? Any ideas?
>
> Thanks,
> Carsten
>
> --
>
> Carsten Bock
> CEO (Geschäftsführer)
>
> ng-voice GmbH
> Millerntorplatz 1
> 20359 Hamburg / Germany
>
> http://www.ng-voice.com
> mailto:carsten at ng-voice.com
>
> Office +49 40 5247593-40
> Fax +49 40 5247593-99
>
> Sitz der Gesellschaft: Hamburg
> Registergericht: Amtsgericht Hamburg, HRB 120189
> Geschäftsführer: Carsten Bock
> Ust-ID: DE279344284
>
> Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
> http://www.ng-voice.com/imprint/
>
> _________________________________________________________________________
> 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 bipin at xbipin.com  Tue Oct  2 05:38:54 2018
From: bipin at xbipin.com (Bipin Patel)
Date: Tue, 2 Oct 2018 09:38:54 +0400
Subject: [Freeswitch-users] keep-alive RE-INVITE dropped calls
Message-ID: <fe4b716e-2336-d39f-3f74-a253608caad8@xbipin.com>

hi,

im suffering call drops from carrier in roughly 15mins and i reported 
this to the carrier and they replied with the below:
If the duration of the call was a multiple of 15 minutes, please make 
sure that you can properly respond to the keep-alive RE-INVITE that we 
sends every 15 minutes.

i have no clue on how this works in FS so can any1 point me the right 
direction on how to sole this, we use FS to route calls from our switch 
to carrier

-- 
Regards,
Bipin


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From admin at smallunix.net  Tue Oct  2 09:09:14 2018
From: admin at smallunix.net (Andrea Mazzeo)
Date: Tue, 2 Oct 2018 11:09:14 +0200
Subject: [Freeswitch-users] zRTP + OPUS one-way audio issue
In-Reply-To: <CABms_cU_wasa_mZf-pvVykLFkNc+4+PSVcMrGZ9+JmOXfGd9Ng@mail.gmail.com>
References: <CABms_cU_wasa_mZf-pvVykLFkNc+4+PSVcMrGZ9+JmOXfGd9Ng@mail.gmail.com>
Message-ID: <CABms_cVpkrN9OORfqKRrLBKftzji_gpd0p4pm=0JQV=ZRcskyA@mail.gmail.com>

Dear All,

I opened a bug for this:
https://freeswitch.org/jira/browse/FS-11402

I would like to offer a bounty, but it's not easy for me to evaluate the
amount of needed work.

Please if someone is interested let me know or better share a bid.

Thank you,
Andrea

Il giorno ven 21 set 2018 alle ore 12:47 Andrea Mazzeo <admin at smallunix.net>
ha scritto:

> Hi,
>
> I'm facing an one-way audio issue with zRTP while using OPUS. zRTP is set
> in passthru mode.
> Called party can hear Calling party but not vice versa.
>
> Full log:
> https://pastebin.freeswitch.org/view/639adfe9
>
> This issue is happening *only* with dynamic payloads codecs.
> Everything is working fine if I use G722 or G729.
>
> Scenario Leg-A (207) -> FS -> Leg-B (213)
>
> Checking FS logs, I see
>
> Leg-A's SDP:
> a=rtpmap:103 opus/48000/2
>
> Leg-B's SDP:
> a=rtpmap:102 opus/48000/2
>
> Actually having different payloads should not be an issue.
> I opened a case to Acrobits, they said the issue is in the actual RTP
> traffic sent to Leg-A from FS:
>
> From the client log Leg-A
> Sending RTP packet #200 192.168.128.158:59176 > 31.102.111.134:28560,
> len=116, really=116,
> data=80679E55DF67E34A291167D899830CC03C124F353DA3C069574410EC0333125F
> This is using 103, correct as agreed on this side of the call
>
> Received RTP packet #200 31.102.111.134:28560 > 192.168.128.158:59176,
> len=90,
> data=80663D3B22CC8CB414490864A230F49B6EFF42E7BCA9EB64B864374B1377965C
> This is using 102, which is wrong for this side of the call.
>
> Seems that after negotiated payload 102 with Leg-B, FS is trying to use it
> on Leg-A, where it should be used 102 instead.
>
> Any idea?
>
> FreeSWITCH Version 1.8.1-2-4f54cff~64bit (-2-4f54cff 64bit)
> OS.: Debian 8.11
> Linux pbx-186 3.16.0-4-amd64 #1 SMP Debian 3.16.51-3 (2017-12-13) x86_64
> GNU/Linux
>
> Thank you,
> Andrea Mazzeo
>
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From sagarmalam at gmail.com  Tue Oct  2 11:05:48 2018
From: sagarmalam at gmail.com (sagar malam)
Date: Tue, 2 Oct 2018 16:35:48 +0530
Subject: [Freeswitch-users] Codec negotiation issue
Message-ID: <CAEZuF3uvOe9MmrH+LjMuJ3MDMCPXZqL_0DFp5uwkaq9OcDKv3A@mail.gmail.com>

I am facing an issue related to codec negotiation.Let me explain scenario
 inbound/outbound codec pref list : g722,g729,g711
 late negotiation enabled
 mix inbound outbound codec is enabled
 When call hits FS, i give instant ringback to call which mean g722 codec
will be negotiated for early media
Now when callee answers call with g729
FS does transcoding from g722: g729
Instead it should select g729 on both legs and tell aleg to update codec
from g722 to g729 to avoid transcoding.
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From mike at jerris.com  Tue Oct  2 20:05:12 2018
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Oct 2018 16:05:12 -0400
Subject: [Freeswitch-users] keep-alive RE-INVITE dropped calls
In-Reply-To: <fe4b716e-2336-d39f-3f74-a253608caad8@xbipin.com>
References: <fe4b716e-2336-d39f-3f74-a253608caad8@xbipin.com>
Message-ID: <3E13D811-B7BD-48AB-A2F9-885E8B94425F@jerris.com>

Sounds like a nat issue that their re-invite isnt getting to us.  Review what you are sending in sip signaling to make sure they are the correct addresses and nothing in the network is blocking that request.

Mike


> On Oct 2, 2018, at 1:38 AM, Bipin Patel <bipin at xbipin.com> wrote:
> 
> hi,
> 
> im suffering call drops from carrier in roughly 15mins and i reported this to the carrier and they replied with the below:
> If the duration of the call was a multiple of 15 minutes, please make sure that you can properly respond to the keep-alive RE-INVITE that we sends every 15 minutes.
> 
> i have no clue on how this works in FS so can any1 point me the right direction on how to sole this, we use FS to route calls from our switch to carrier
> 
> -- 
> Regards,
> Bipin
> 

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From mike at jerris.com  Tue Oct  2 20:06:11 2018
From: mike at jerris.com (Michael Jerris)
Date: Tue, 2 Oct 2018 16:06:11 -0400
Subject: [Freeswitch-users] Codec negotiation issue
In-Reply-To: <CAEZuF3uvOe9MmrH+LjMuJ3MDMCPXZqL_0DFp5uwkaq9OcDKv3A@mail.gmail.com>
References: <CAEZuF3uvOe9MmrH+LjMuJ3MDMCPXZqL_0DFp5uwkaq9OcDKv3A@mail.gmail.com>
Message-ID: <9E9891C3-7017-46B1-B5AD-738CDF7DB9A5@jerris.com>

https://freeswitch.org/confluence/display/FREESWITCH/inherit_codec <https://freeswitch.org/confluence/display/FREESWITCH/inherit_codec>


> On Oct 2, 2018, at 7:05 AM, sagar malam <sagarmalam at gmail.com> wrote:
> 
> I am facing an issue related to codec negotiation.Let me explain scenario
>  inbound/outbound codec pref list : g722,g729,g711
>  late negotiation enabled
>  mix inbound outbound codec is enabled
>  When call hits FS, i give instant ringback to call which mean g722 codec will be negotiated for early media
> Now when callee answers call with g729
> FS does transcoding from g722: g729
> Instead it should select g729 on both legs and tell aleg to update codec from g722 to g729 to avoid transcoding.

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From steveayre at gmail.com  Tue Oct  2 21:11:49 2018
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 2 Oct 2018 22:11:49 +0100
Subject: [Freeswitch-users] keep-alive RE-INVITE dropped calls
In-Reply-To: <fe4b716e-2336-d39f-3f74-a253608caad8@xbipin.com>
References: <fe4b716e-2336-d39f-3f74-a253608caad8@xbipin.com>
Message-ID: <CAFiqYukmyy1NYzcifNo+WCJxFkkyGGkLMt6_qGQ-Y3bzZrU9Mg@mail.gmail.com>

This will be referring to SIP session timers, which FS does support.

Try collecting a SIP packet capture to see what packets are being sent and
what the reINVITE packet gets as a response.

On Tue, 2 Oct 2018 at 20:40, Bipin Patel <bipin at xbipin.com> wrote:

> hi,
>
> im suffering call drops from carrier in roughly 15mins and i reported this
> to the carrier and they replied with the below:
> If the duration of the call was a multiple of 15 minutes, please make sure
> that you can properly respond to the keep-alive RE-INVITE that we sends
> every 15 minutes.
>
> i have no clue on how this works in FS so can any1 point me the right
> direction on how to sole this, we use FS to route calls from our switch to
> carrier
>
> --
> Regards,
> Bipin
>
>
> _________________________________________________________________________
> 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 bipin at xbipin.com  Wed Oct  3 08:36:15 2018
From: bipin at xbipin.com (Bipin Patel)
Date: Wed, 03 Oct 2018 12:36:15 +0400
Subject: [Freeswitch-users] keep-alive RE-INVITE dropped calls
In-Reply-To: <3E13D811-B7BD-48AB-A2F9-885E8B94425F@jerris.com>
References: <fe4b716e-2336-d39f-3f74-a253608caad8@xbipin.com>
 <3E13D811-B7BD-48AB-A2F9-885E8B94425F@jerris.com>
Message-ID: <c036e697c2994e70ffebdd429f31e1b4@xbipin.com>

thanks, managed to fix this, problem was carrier was sending the re
invite but it wasn't reaching FS even though i had ports open and both
my server and carrier server were able to talk to each other for
outgoing calls, my FS is on a public IP so NAT isnt an issue and i had
to explicitly add a allow rule for FS external SIP port and strangely
the re invite started reaching FS and FS started responding, my other
problem is i want FS to just reply to the re invite after which the
carrier sends a ack but dont want FS to renegotiate codecs because the
carrier re invite has SDP with the same codec that the call is running
on so codecs will not change but they just send the re invite to prevent
fraud which needs to be answered to prevent call being dropped. 

is there a way to make FS respond to the re invite but not try and
renegotiate codecs coz currently i see messages such as renegotiating
codecs and then no change. 

On 03-10-2018 00:05, Michael Jerris wrote:

> Sounds like a nat issue that their re-invite isnt getting to us.  Review what you are sending in sip signaling to make sure they are the correct addresses and nothing in the network is blocking that request. 
> 
> Mike 
> 
>> On Oct 2, 2018, at 1:38 AM, Bipin Patel <bipin at xbipin.com> wrote: 
>> hi,
>> 
>> im suffering call drops from carrier in roughly 15mins and i reported this to the carrier and they replied with the below:
>> If the duration of the call was a multiple of 15 minutes, please make sure that you can properly respond to the keep-alive RE-INVITE that we sends every 15 minutes.
>> 
>> i have no clue on how this works in FS so can any1 point me the right direction on how to sole this, we use FS to route calls from our switch to carrier
>> 
>> -- 
>> Regards,
>> Bipin
> 
> _________________________________________________________________________
> 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 sagarmalam at gmail.com  Wed Oct  3 11:12:38 2018
From: sagarmalam at gmail.com (sagar malam)
Date: Wed, 3 Oct 2018 16:42:38 +0530
Subject: [Freeswitch-users] Codec negotiation issue
In-Reply-To: <9E9891C3-7017-46B1-B5AD-738CDF7DB9A5@jerris.com>
References: <CAEZuF3uvOe9MmrH+LjMuJ3MDMCPXZqL_0DFp5uwkaq9OcDKv3A@mail.gmail.com>
 <9E9891C3-7017-46B1-B5AD-738CDF7DB9A5@jerris.com>
Message-ID: <CAEZuF3tXU_zexmYXWzVbf-4pvj52qivnvEi4oZCNwCSEj0KdDw@mail.gmail.com>

Thanks.

I will test and get back

On Wed, Oct 3, 2018 at 3:13 AM Michael Jerris <mike at jerris.com> wrote:

> https://freeswitch.org/confluence/display/FREESWITCH/inherit_codec
>
>
> On Oct 2, 2018, at 7:05 AM, sagar malam <sagarmalam at gmail.com> wrote:
>
> I am facing an issue related to codec negotiation.Let me explain scenario
>  inbound/outbound codec pref list : g722,g729,g711
>  late negotiation enabled
>  mix inbound outbound codec is enabled
>  When call hits FS, i give instant ringback to call which mean g722 codec
> will be negotiated for early media
> Now when callee answers call with g729
> FS does transcoding from g722: g729
> Instead it should select g729 on both legs and tell aleg to update codec
> from g722 to g729 to avoid transcoding.
>
>
> _________________________________________________________________________
> 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  3 18:05:17 2018
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 3 Oct 2018 18:05:17 +0000
Subject: [Freeswitch-users] npa_nxx_company_ocn database?
Message-ID: <010001663b1aff12-10340d0f-fd77-4f99-92f5-8335df3a43eb-000000@email.amazonses.com>

mod_lcr mentions using npa_nxx_company_ocn to determine inter/intra-state
or inter/intra-lata calls.

Is there an updated database somewhere? I have npa_nxx_company_ocn from
someone from years ago.

Thanks!

-Avi Marcus
1-718-989-9485 (USA)
02-372-1570 (Israel)
020-3298-2875 (UK)
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From avi at avimarcus.net  Wed Oct  3 18:16:06 2018
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 3 Oct 2018 18:16:06 +0000
Subject: [Freeswitch-users] npa_nxx_company_ocn database?
In-Reply-To: <CALNZuwkvnXOzs-ieKp+g6VoEdbDD8oFxUeKxc1nFfcjuk9xrxw@mail.gmail.com>
References: <CALNZuwkvnXOzs-ieKp+g6VoEdbDD8oFxUeKxc1nFfcjuk9xrxw@mail.gmail.com>
Message-ID: <010001663b24e6c7-43cbf94f-029c-44c4-9893-d1f747639d8e-000000@email.amazonses.com>

Ah, it was linked from mod_cidlookup page:
http://files.freeswitch.org/npa-nxx-companytype-ocn.csv
But `curl --head` tells me: Last-Modified: Wed, 08 Jul 2009 01:19:32 GMT

Seems my carriers only provide intra or interstate, no LATA calculations
necessary.

Anyone have an updated npa/nxx per state list?

Do any npa's cross state lines? Perhaps I only need the much smaller NPA
list. According to that 2009 data, none do.

-Avi Marcus
1-718-989-9485 (USA)
02-372-1570 (Israel)
020-3298-2875 (UK)


On Wed, Oct 3, 2018 at 9:04 PM Avi Marcus <avi at avimarcus.net> wrote:

> mod_lcr mentions using npa_nxx_company_ocn to determine inter/intra-state
> or inter/intra-lata calls.
>
> Is there an updated database somewhere? I have npa_nxx_company_ocn from
> someone from years ago.
>
> Thanks!
>
> -Avi Marcus
> 1-718-989-9485 (USA)
> 02-372-1570 (Israel)
> 020-3298-2875 (UK)
>
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From michael at mailworks.org  Wed Oct  3 18:32:42 2018
From: michael at mailworks.org (Michael Avers)
Date: Wed, 03 Oct 2018 11:32:42 -0700
Subject: [Freeswitch-users] npa_nxx_company_ocn database?
In-Reply-To: <010001663b24e6c7-43cbf94f-029c-44c4-9893-d1f747639d8e-000000@email.amazonses.com>
References: <CALNZuwkvnXOzs-ieKp+g6VoEdbDD8oFxUeKxc1nFfcjuk9xrxw@mail.gmail.com>
 <010001663b24e6c7-43cbf94f-029c-44c4-9893-d1f747639d8e-000000@email.amazonses.com>
Message-ID: <1538591562.1075751.1529594912.2D841D24@webmail.messagingengine.com>

You need these: 

https://www.telcodata.us/data-downloads

Mike


On Wed, Oct 3, 2018, at 11:16 AM, Avi Marcus wrote:
> Ah, it was linked from mod_cidlookup page:
> http://files.freeswitch.org/npa-nxx-companytype-ocn.csv> But `curl --head` tells me: Last-Modified: Wed, 08 Jul 2009
> 01:19:32 GMT> 
> Seems my carriers only provide intra or interstate, no LATA
> calculations necessary.> 
> Anyone have an updated npa/nxx per state list?
> 
> Do any npa's cross state lines? Perhaps I only need the much smaller
> NPA list. According to that 2009 data, none do.>  
> -Avi Marcus
> 1-718-989-9485 (USA)
> 02-372-1570 (Israel)
> 020-3298-2875 (UK)
> 
> 
> On Wed, Oct 3, 2018 at 9:04 PM Avi Marcus <avi at avimarcus.net> wrote:
>> mod_lcr mentions using npa_nxx_company_ocn to determine inter/intra-
>> state or inter/intra-lata calls.>> 
>> Is there an updated database somewhere? I have npa_nxx_company_ocn
>> from someone from years ago.>> 
>> Thanks!
>> 
>> -Avi Marcus
>> 1-718-989-9485 (USA)
>> 02-372-1570 (Israel)
>> 020-3298-2875 (UK)
> ___________________________________________________________________-
> ________> 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 nneul at mst.edu  Wed Oct  3 18:51:12 2018
From: nneul at mst.edu (Nathan Neulinger)
Date: Wed, 3 Oct 2018 13:51:12 -0500
Subject: [Freeswitch-users] npa_nxx_company_ocn database?
In-Reply-To: <1538591562.1075751.1529594912.2D841D24@webmail.messagingengine.com>
References: <CALNZuwkvnXOzs-ieKp+g6VoEdbDD8oFxUeKxc1nFfcjuk9xrxw@mail.gmail.com>
 <010001663b24e6c7-43cbf94f-029c-44c4-9893-d1f747639d8e-000000@email.amazonses.com>
 <1538591562.1075751.1529594912.2D841D24@webmail.messagingengine.com>
Message-ID: <c827c34c-c566-9da6-9c22-acee7a35f111@mst.edu>

Another source:

https://www.nationalnanpa.com/reports/reports_cocodes_assign.html

-- Nathan

------------------------------------------------------------------------------------------------------------------------
*From:* Michael Avers <michael at mailworks.org>
*Sent:* Wed, Oct 3, 2018 1:32 PM CDT
*To:* freeswitch-users at lists.freeswitch.org
*Subject:* [Freeswitch-users] npa_nxx_company_ocn database?

> You need these:
>
> https://www.telcodata.us/data-downloads
>
> Mike
>
>
> On Wed, Oct 3, 2018, at 11:16 AM, Avi Marcus wrote:
>> Ah, it was linked from mod_cidlookup page: http://files.freeswitch.org/npa-nxx-companytype-ocn.csv
>> But `curl --head` tells me: Last-Modified: Wed, 08 Jul 2009 01:19:32 GMT
>>
>> Seems my carriers only provide intra or interstate, no LATA calculations necessary.
>>
>> Anyone have an updated npa/nxx per state list?
>>
>> Do any npa's cross state lines? Perhaps I only need the much smaller NPA list. According to that 2009 data, none do.
>>
>> -Avi Marcus
>> 1-718-989-9485 (USA)
>> 02-372-1570 (Israel)
>> 020-3298-2875 (UK)
>>
>>
>> On Wed, Oct 3, 2018 at 9:04 PM Avi Marcus <avi at avimarcus.net <mailto:avi at avimarcus.net>> wrote:
>>
>>     mod_lcr mentions using npa_nxx_company_ocn to determine inter/intra-state or inter/intra-lata calls.
>>
>>     Is there an updated database somewhere? I have npa_nxx_company_ocn from someone from years ago.
>>
>>     Thanks!
>>
>>     -Avi Marcus
>>     1-718-989-9485 (USA)
>>     02-372-1570 (Israel)
>>     020-3298-2875 (UK)
>>
>> ___________________________________________________________________________
>> 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
>
>
> _________________________________________________________________________
> 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

-- 
------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect

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From krice at tollfreegateway.com  Wed Oct  3 19:17:12 2018
From: krice at tollfreegateway.com (krice at tollfreegateway.com)
Date: Wed, 3 Oct 2018 14:17:12 -0500
Subject: [Freeswitch-users] npa_nxx_company_ocn database?
In-Reply-To: <010001663b24e6c7-43cbf94f-029c-44c4-9893-d1f747639d8e-000000@email.amazonses.com>
References: <CALNZuwkvnXOzs-ieKp+g6VoEdbDD8oFxUeKxc1nFfcjuk9xrxw@mail.gmail.com>
 <010001663b24e6c7-43cbf94f-029c-44c4-9893-d1f747639d8e-000000@email.amazonses.com>
Message-ID: <1bc901d45b4d$b02aee80$1080cb80$@tollfreegateway.com>

NPAs don’t cross state lines, but LATAs do cross state lines.

 

Any data such as that on the FreeSWITCH website would be sample data. The official sources for that data would be NANPA… you can get something close from telcodata.us

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Avi Marcus
Sent: Wednesday, October 3, 2018 1:16 PM
To: FreeSWITCH Users Help <FreeSWITCH-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] npa_nxx_company_ocn database?

 

Ah, it was linked from mod_cidlookup page: http://files.freeswitch.org/npa-nxx-companytype-ocn.csv

But `curl --head` tells me: Last-Modified: Wed, 08 Jul 2009 01:19:32 GMT

 

Seems my carriers only provide intra or interstate, no LATA calculations necessary.

 

Anyone have an updated npa/nxx per state list?

 

Do any npa's cross state lines? Perhaps I only need the much smaller NPA list. According to that 2009 data, none do.

 

-Avi Marcus
1-718-989-9485 (USA)
02-372-1570 (Israel)
020-3298-2875 (UK)

 

 

On Wed, Oct 3, 2018 at 9:04 PM Avi Marcus <avi at avimarcus.net <mailto:avi at avimarcus.net> > wrote:

mod_lcr mentions using npa_nxx_company_ocn to determine inter/intra-state or inter/intra-lata calls.

 

Is there an updated database somewhere? I have npa_nxx_company_ocn from someone from years ago.

 

Thanks!




-Avi Marcus
1-718-989-9485 (USA)
02-372-1570 (Israel)
020-3298-2875 (UK)

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From sos at sokhapkin.dyndns.org  Wed Oct  3 20:40:18 2018
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 03 Oct 2018 16:40:18 -0400
Subject: [Freeswitch-users] npa_nxx_company_ocn database?
In-Reply-To: <1538591562.1075751.1529594912.2D841D24@webmail.messagingengine.com>
References: <CALNZuwkvnXOzs-ieKp+g6VoEdbDD8oFxUeKxc1nFfcjuk9xrxw@mail.gmail.com>
 <010001663b24e6c7-43cbf94f-029c-44c4-9893-d1f747639d8e-000000@email.amazonses.com>
 <1538591562.1075751.1529594912.2D841D24@webmail.messagingengine.com>
Message-ID: <2789979.j5c4PI2N8u@mobile5>

It well worth to pay $15 every few months or once a year to have the up to 
date list. How often - it depends on your needs.

On Wednesday, October 3, 2018 2:32:42 PM EDT Michael Avers wrote:

You need these: 



https://www.telcodata.us/data-downloads



Mike




On Wed, Oct 3, 2018, at 11:16 AM, Avi Marcus wrote:

Ah, it was linked from mod_cidlookup page: http://files.freeswitch.org/npa-nxx-companytype-ocn.csv

But `curl --head` tells me: Last-Modified: Wed, 08 Jul 2009 01:19:32 GMT



Seems my carriers only provide intra or interstate, no LATA calculations 
necessary.



Anyone have an updated npa/nxx per state list?



Do any npa's cross state lines? Perhaps I only need the much smaller NPA list. 
According to that 2009 data, none do.

 

-Avi Marcus

1-718-989-9485 (USA)

02-372-1570 (Israel)

020-3298-2875 (UK)





On Wed, Oct 3, 2018 at 9:04 PM Avi Marcus <avi at avimarcus.net> wrote:

mod_lcr mentions using npa_nxx_company_ocn to determine inter/intra-state or 
inter/intra-lata calls.



Is there an updated database somewhere? I have npa_nxx_company_ocn from 
someone from years ago.



Thanks!



-Avi Marcus

1-718-989-9485 (USA)

02-372-1570 (Israel)

020-3298-2875 (UK)

_________________________________________________________________________

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 Lee.barnard at voip-unlimited.net  Thu Oct  4 08:55:51 2018
From: Lee.barnard at voip-unlimited.net (Lee Barnard)
Date: Thu, 4 Oct 2018 08:55:51 +0000
Subject: [Freeswitch-users] Call Group Assistance
Message-ID: <B85E41B33FFF6B4093A51EBC185297CE3DC2DA3F@serverk.it-unlimited.local>

Hi Guys,

Having an issue with a call group.

Issue:
Have 3 users

  1.  1 External
  2.  2 Internal

Call rings to all 3 endpoints, however, when rejecting call on the external entity it clears call on all legs.
If call is rejected on either internal entities continues to ring on all other entities.

Can anyone give a pointer as to what to try to get the functionality the same as the internals across the external entities.
NOTE: You can see some artefacts of things tried in the dial-string.

Configuration is built from LUA script and appears in debug as below:
2018-10-03 12:32:52.430641 [NOTICE] switch_cpp.cpp:1365 Call Group Request
2018-10-03 12:32:52.430641 [NOTICE] switch_cpp.cpp:1365 Extension: 2000 and Domain: domain.co.uk
2018-10-03 12:32:52.430641 [NOTICE] switch_cpp.cpp:1365 Query Result:
2018-10-03 12:32:52.430641 [NOTICE] switch_cpp.cpp:1365 Extension: 2001 and Domain: domain.co.uk
2018-10-03 12:32:52.430641 [NOTICE] switch_cpp.cpp:1365 Query Result: 123456789
2018-10-03 12:32:52.430641 [NOTICE] switch_cpp.cpp:1365 Action Huntgroup Returns:
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<section name="directory">
<domain name="domain.co.uk">
<groups>
<group name="groupname">
<users>
<user id="123456789">
<params>
<param name="dial-string" value="{hangup_after_bridge=true,fail_on_single_reject=false}sofia/internal/123456789 at domain.co.uk"/>
</params>
<variables>
<variable name="user_context" value="default"/>
</variables>
</user>
<user id="2000" type="pointer"/>
<user id="2001" type="pointer"/>
</users>
</group>
</groups>
</domain>
</section>
</document>
2018-10-03 12:32:52.430641 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0xc6bb90 released.
2ffa1d17-2ae3-4920-8f48-23798644ed25 EXECUTE sofia/internal/987654321 at IPADDRESS bridge([hangup_after_bridge=true,fail_on_single_reject=false]sofia/internal/123456789 at domain.co.uk,user/2000 at domain.co.uk,user/2001 at domain.co.uk<mailto:sofia/internal/123456789 at domain.co.uk,user/2000 at domain.co.uk,user/2001 at domain.co.uk>)


Happy to share any additional information required – please be gentle, I am fairly newish to freeswitch.

Thanks in advance
Lee

Kind regards,

Lee Barnard     Voice Systems Engineer

T: 01202612000      F: 01202612111   E: Lee.barnard at voip-unlimited.net

Online ordering now live!
Visit: portal.voip-unlimited.net/ordering<https://portal.voip-unlimited.net/ordering> or speak to your Account Manager today.
________________________________

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[Facebook]<https://www.facebook.com/pages/VoIP-Unlimited/286623001407365>[cid:image8dda1d.PNG at aaf8f1d3.4fabb9b6]<https://plus.google.com/114885723213491901544>[cid:imagec180ba.PNG at 3cf76013.479eeb80]<https://www.linkedin.com/company/voip-unlimited>[cid:image418cf6.PNG at 78887a23.4285e855]<https://twitter.com/voipunlimited>[cid:image3d111d.PNG at 9a6f9dd3.4dac0962]<https://www.youtube.com/watch?v=TmD1Eg3uCwY>






[VU Guard DDoS Mitigation]<http://www.voip-unlimited.net/services/channel-ddos>





6 Albany Business Park, Cabot Lane Poole, Dorset, UK, BH17 7BX.

This email including any attachments is intended only for the addressee named above and it may contain confidential or privileged information. If you are not the intended recipient please notify the sender and note that the contents must not be disclosed to anyone else. Voip Unlimited will not be liable for any error in transmission. You should carry out your own virus checks before opening any attachments. Opinions, conclusions and other information that do not relate to the official business of Voip Unlimited are neither given nor endorsed by it.

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From albert.hsueh at cloudigit.com  Thu Oct  4 10:01:31 2018
From: albert.hsueh at cloudigit.com (=?UTF-8?B?6Jab5YWJ5a6P?=)
Date: Thu, 4 Oct 2018 18:01:31 +0800
Subject: [Freeswitch-users] Is it possible add new item in sip message
	header?
Message-ID: <CABHWxvzw+xP0bmwDg3CA-yT2GvT8PsRga726vm9w9tCNZYhvng@mail.gmail.com>

Hi All,

We run a mobile app to *register* FreeSWITCH 1.6.20 via sip. This app will
check value of server field in message header but the packet sent from
FreeSWITCH doesn't have this field.
We would like to know is it possible to add filed "Server" to message
header by adjusting FreeSWITCH configuration?

We tried this by adjusting dialplan (
https://freeswitch.org/confluence/display/FREESWITCH/Sofia+SIP+Stack)
        <action application="set"><![CDATA[sip_h_Server=xxxxxx]]></action>

it works in invite stage, but can not work in register stage. The app
checks the filed "Server" in register stage.

Thanks in advance.

-- 
Sincerely

Albert Hsueh
System Engineer
Cloudpe Corporation
02-8712 5955 ext. 831
www.cloudpe.com
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From joes.mailing.lists at gmail.com  Thu Oct  4 12:30:35 2018
From: joes.mailing.lists at gmail.com (.)
Date: Thu, 4 Oct 2018 14:30:35 +0200
Subject: [Freeswitch-users] Incorrect example for 'originate' with socket
	and async full
Message-ID: <CAHUGhbKq2FEPJ23mfjvfDBSkM+16yz5F6OCFanN_5tsKRoHj-g@mail.gmail.com>

Hi,

I wanted to report an error/typo in the confluence wiki related to
'originate', and its use together with the socket application.

Apologies, I don't have access to the wiki, but thought this was worth
passing on, so here it is:

   -
   https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-originate

The wiki example for ring_ready also includes the socket application and is
written as:

   -

   originate {return_ring_ready=true}sofia/gateway/someprovider/919246461929
&socket(127.0.0.1:8082 async full)


however this hangs forever if used as written. Eventually, after several
hours of hair tearing and googling, I stumbled on this thread:

   -
   https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-3591/FS-3591.html

Indeed this is the problem.  The correct syntax for the socket component is:

   -

   &socket('127.0.0.1:8082 async full')

   - Specifically single quotes around the command so that it is treated as
      a single string.

After making this change there is no more hanging on the outbound calls and
all events come through in a timely fashion.

Cheers,
Joe
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From gmaruzz at gmail.com  Thu Oct  4 14:54:13 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 4 Oct 2018 16:54:13 +0200
Subject: [Freeswitch-users] Astricon: OpenSource Soiree
Message-ID: <CALXCt0oxZosG+6Zzy8N0oD3QOvSi7hm--tZTd0jKE2p8uOqHww@mail.gmail.com>

Hello friends and collegues,

A nice get together has been organized for evening Wednesday 10th at 7pm

The idea is to have drinks and chitchat about opensource telephony and
webrtc, and all opensource project founders, members, and practitioners are
invited.

Please join me, as distinguished participants too!

Check it out at  http://party.officering.com

(no animals has been harmed in this mail)

-giovanni

-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From morfair at gmail.com  Thu Oct  4 15:28:23 2018
From: morfair at gmail.com (morfair at gmail.com)
Date: Thu, 4 Oct 2018 18:28:23 +0300
Subject: [Freeswitch-users] Wrong response SIP port
Message-ID: <e1577ca6-0e33-cb81-57ef-8d2c3f42d526@gmail.com>

I have external profile on port 5080. When FS get INVITE on 5080 from 
remote 5060 port, FS response on 5080 port too instead 5060

20.20.20.20 - fs

10.10.10.10 - remote sip

18:23:49.675349 IP 10.10.10.10.5060 > 20.20.20.20.5080: SIP: INVITE 
sip:7001 at 20.20.20.20:5080;user=phone SIP/2.0
18:23:49.678953 IP 20.20.20.20.5080 > 10.10.10.10.5080: UDP, length 363


How to fix whis? FS should response on originate udp port.



From avi at avimarcus.net  Thu Oct  4 16:15:04 2018
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 4 Oct 2018 16:15:04 +0000
Subject: [Freeswitch-users] npa_nxx_company_ocn database?
In-Reply-To: <1bc901d45b4d$b02aee80$1080cb80$@tollfreegateway.com>
References: <CALNZuwkvnXOzs-ieKp+g6VoEdbDD8oFxUeKxc1nFfcjuk9xrxw@mail.gmail.com>
 <010001663b24e6c7-43cbf94f-029c-44c4-9893-d1f747639d8e-000000@email.amazonses.com>
 <1bc901d45b4d$b02aee80$1080cb80$@tollfreegateway.com>
Message-ID: <010001663fdc758c-d236b676-730d-4144-9adf-9dcc78f63819-000000@email.amazonses.com>

I just needed NPA to state for inter or intrastate.

I imported from neustar and I have json data now like this. Enjoy, anyone
that needs it:

https://github.com/avimar/north-america-phone/blob/master/data/area-codes-us.json
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From mike at jerris.com  Thu Oct  4 18:18:50 2018
From: mike at jerris.com (Michael Jerris)
Date: Thu, 4 Oct 2018 14:18:50 -0400
Subject: [Freeswitch-users] Wrong response SIP port
In-Reply-To: <e1577ca6-0e33-cb81-57ef-8d2c3f42d526@gmail.com>
References: <e1577ca6-0e33-cb81-57ef-8d2c3f42d526@gmail.com>
Message-ID: <FD32268A-01D2-4B9A-9213-BFE739D9AD61@jerris.com>

I would need to see a full and UNMODIFIED sip trace to properly answer this question.

> On Oct 4, 2018, at 11:28 AM, morfair at gmail.com wrote:
> 
> I have external profile on port 5080. When FS get INVITE on 5080 from remote 5060 port, FS response on 5080 port too instead 5060
> 
> 20.20.20.20 - fs
> 
> 10.10.10.10 - remote sip
> 
> 18:23:49.675349 IP 10.10.10.10.5060 > 20.20.20.20.5080: SIP: INVITE sip:7001 at 20.20.20.20:5080;user=phone SIP/2.0
> 18:23:49.678953 IP 20.20.20.20.5080 > 10.10.10.10.5080: UDP, length 363
> 
> 
> How to fix whis? FS should response on originate udp port.
> 



From david.villasmil.work at gmail.com  Thu Oct  4 18:43:49 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 4 Oct 2018 19:43:49 +0100
Subject: [Freeswitch-users] Wrong response SIP port
In-Reply-To: <e1577ca6-0e33-cb81-57ef-8d2c3f42d526@gmail.com>
References: <e1577ca6-0e33-cb81-57ef-8d2c3f42d526@gmail.com>
Message-ID: <CAFGRPVob2q+rY99wc2mAxPFaXs2XvTsVv9Ys0kSFyn-4cwEGZQ@mail.gmail.com>

Can you provide the complete trace?

Regards,

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

ᐧ

On Thu, Oct 4, 2018 at 6:26 PM morfair at gmail.com <morfair at gmail.com> wrote:

> I have external profile on port 5080. When FS get INVITE on 5080 from
> remote 5060 port, FS response on 5080 port too instead 5060
>
> 20.20.20.20 - fs
>
> 10.10.10.10 - remote sip
>
> 18:23:49.675349 IP 10.10.10.10.5060 > 20.20.20.20.5080: SIP: INVITE
> sip:7001 at 20.20.20.20:5080;user=phone SIP/2.0
> 18:23:49.678953 IP 20.20.20.20.5080 > 10.10.10.10.5080: UDP, length 363
>
>
> How to fix whis? FS should response on originate udp port.
>
>
> _________________________________________________________________________
> 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 wsimon at stratusvideo.com  Fri Oct  5 14:27:58 2018
From: wsimon at stratusvideo.com (William Simon)
Date: Fri, 5 Oct 2018 14:27:58 +0000
Subject: [Freeswitch-users] Accepting an "optional" SRTP offer (crypto in
 RTP/AVP) and establishing SRTP
Message-ID: <2AE1EF96-BD07-4E78-A638-D9D53FF8FF1E@stratusvideo.com>

We are working with an endpoint that offers optional SRTP in the non-RFC-compliant way of an a=crypto attribute within RTP/AVP.

I have told FreeSWITCH to allow this with rtp_allow_crypto_in_avp=true at the right place in the dialplan.

Immediately after that in the dialplan I have to reject SRTP by using rtp_secure_media=forbidden, otherwise the call setup still fails. By setting the value to forbidden, the call does proceed unencrypted.

We want FreeSWITCH to proceed with media encryption. Setting rtp_secure_media to any other value results in FreeSWITCH rejecting the offer like this:

v=0
o=FreeSWITCH 1538660754 1538660755 IN IP4 192.168.100.104
s=FreeSWITCH
c=IN IP4 192.168.100.104
t=0 0
m=audio 0 RTP/AVP 19
m=video 0 RTP/AVP 19

Is there anything else I can do to force SRTP in the answer?







“The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”

From asilva at wirelessmundi.com  Fri Oct  5 14:42:45 2018
From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=)
Date: Fri, 5 Oct 2018 16:42:45 +0200
Subject: [Freeswitch-users] Multiple concurrent Verto calls in browser
In-Reply-To: <CAC7ttFFdnQOveF=Vw1DjR2yz-X-=ci5arqYPx-_WzqaRb8hoBg@mail.gmail.com>
References: <1521446608.822385.1307854600.6BAD2C6E@webmail.messagingengine.com>
 <CAC7ttFF642Rn6GziOYRiaU58gptnb-0JMa1oyBqyOqXmBQu-TQ@mail.gmail.com>
 <7B8D444E-7087-450C-96B8-AC7755BAB74D@gmail.com>
 <CAC7ttFH7LcMjRON0xY15rMZH7KHJBgwfAkBxPkia6Pf7C_Rk_A@mail.gmail.com>
 <1522307633.2221439.1319939472.62B616C1@webmail.messagingengine.com>
 <CAC7ttFFdnQOveF=Vw1DjR2yz-X-=ci5arqYPx-_WzqaRb8hoBg@mail.gmail.com>
Message-ID: <8cce328e-5c31-f47d-38fb-783da1de6ac9@wirelessmundi.com>

Hi,

I've done the same, keep an array of active calls, but i'm facing a 
weird issue, when a new incoming call arrives, in chrome, i stop listen 
to the active call (i can talk to the other end),  to fixed i just put 
hold and remove hold and i  have sound again...

I firefox this doesn't happen... any idea what could be the issue? did 
it happen to you?


On 29/03/2018 09:32, Gregor Nanger wrote:
> We are dealing same way as multiline phone. We are showing to user in 
> GUI waiting call (can be many of them) and it is up to user what he 
> wants to do. If he picks up  call and one is active, we put active on 
> hold. There can be only one active at a time and others are on hold 
> (if they were pickedup) or on queuee.
>
> 2018-03-29 9:13 GMT+02:00 Michael Avers <michael at mailworks.org 
> <mailto:michael at mailworks.org>>:
>
>     How do you handle the state of the multiple calls? Say user is on
>     a call, and a new incoming call comes in and they want to answer
>     it - do you place the currently active call on hold before you
>     answer the new one?
>
>     Thanks
>     Mike
>
>
>     On Wed, Mar 21, 2018, at 3:36 AM, Gregor Nanger wrote:
>>     @Tihomir
>>     Sorry, to advanced question for me :-) We just insert new video
>>     canvas for each call and hide, since we only use audio and keep
>>     own array list with call guid and states as we want.
>>
>>     Gregor
>>
>>
>>     2018-03-21 7:19 GMT+01:00 Tihomir Culjaga <tculjaga at gmail.com
>>     <mailto:tculjaga at gmail.com>>:
>>
>>         @Michael,
>>         We did the same thing as Gregor, implemented verto api in our
>>         own web interface... a single wss is enough :)
>>
>>         @Gregor, sorry to hijack the mail ;)
>>
>>         But did you happen to be able to set googDscp in
>>         RTCPeerConnection and chrome actually to mark rtp packets?
>>         My chrome (latest) on win10 keeps ignoring it :(.
>>
>>         Tihomir.
>>
>>
>>
>>         Sent from my iPhone
>>
>>         On 19 Mar 2018, at 09:56, Gregor Nanger <gregor at infomedia.si
>>         <mailto:gregor at infomedia.si>> wrote:
>>>         Hi,
>>>
>>>         we are using such scenario, but it is not based on Verto
>>>         Communicator. We took core libraries (javascript) and
>>>         implement GUI on our own. Everything what happens on verto
>>>         triggers event  and you can implement own logic and
>>>         callstate for each call in array. For example, when user
>>>         picks up  a call, put active on hold via verto command.
>>>
>>>         And for info, you can issue as many calls you want through
>>>         one verto connection.
>>>
>>>
>>>
>>>
>>>         2018-03-19 9:03 GMT+01:00 Michael Avers
>>>         <michael at mailworks.org <mailto:michael at mailworks.org>>:
>>>
>>>             Hello,
>>>
>>>             I'm using Verto Communicator as a base for building a
>>>             simple app to make/receive calls - and it works great.
>>>
>>>             I'm trying to add the ability to manage multiple
>>>             concurrent calls.
>>>
>>>             My first thought is to create a new Verto service (in
>>>             Angular) that maintains call state and reference to
>>>             currently active call and otherwise keeps all calls in
>>>             an array. Then to switch calls it'd place the currently
>>>             active one on hold and then unhold any of the other
>>>             calls in the array. Does this make sense?
>>>
>>>             Any tips on handling multiple calls with Verto? Do I
>>>             need a separate websocket connection for each? (they're
>>>             all using the same proxy and credentials).
>>>
>>>             Thanks!
>>>             Mike
>>>
>>>             _________________________________________________________________________
>>>             Professional FreeSWITCH Consulting Services:
>>>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>             http://www.freeswitchsolutions.com
>>>             <http://www.freeswitchsolutions.com>
>>>
>>>             Official FreeSWITCH Sites
>>>             http://www.freeswitch.org
>>>             http://confluence.freeswitch.org
>>>             <http://confluence.freeswitch.org>
>>>             http://www.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>
>>>             http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>         -- 
>>>         Gregor Nanger
>>>
>>>         	*CTO*
>>>         t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>>>         • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>>>         • www.infomedia.si <http://www.infomedia.si>
>>>
>>>         _________________________________________________________________________
>>>         Professional FreeSWITCH Consulting Services:
>>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>         http://www.freeswitchsolutions.com
>>>         <http://www.freeswitchsolutions.com>
>>>
>>>         Official FreeSWITCH Sites
>>>         http://www.freeswitch.org
>>>         http://confluence.freeswitch.org
>>>         <http://confluence.freeswitch.org>
>>>         http://www.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>
>>>         http://www.freeswitch.org
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>         <http://www.freeswitchsolutions.com>
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://confluence.freeswitch.org
>>         <http://confluence.freeswitch.org>
>>         http://www.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>
>>         http://www.freeswitch.org
>>
>>
>>
>>
>>     -- 
>>     Gregor Nanger
>>
>>     	*CTO*
>>     t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>>     • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>>     • www.infomedia.si <http://www.infomedia.si>
>>
>>     ___________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>     <http://www.freeswitchsolutions.com>
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://confluence.freeswitch.org <http://confluence.freeswitch.org>
>>     http://www.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>
>>     http://www.freeswitch.org
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com>
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org <http://confluence.freeswitch.org>
>     http://www.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>
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Gregor Nanger
> 	*CTO*
> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
> • www.infomedia.si <http://www.infomedia.si>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.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
> http://www.freeswitch.org

-- 
Saludos / Regards / Cumprimentos
António Silva

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From steveayre at gmail.com  Fri Oct  5 15:10:29 2018
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 5 Oct 2018 16:10:29 +0100
Subject: [Freeswitch-users] Wrong response SIP port
In-Reply-To: <FD32268A-01D2-4B9A-9213-BFE739D9AD61@jerris.com>
References: <e1577ca6-0e33-cb81-57ef-8d2c3f42d526@gmail.com>
 <FD32268A-01D2-4B9A-9213-BFE739D9AD61@jerris.com>
Message-ID: <CAFiqYunxPFEjJxOTOBcOJ91r1iUhF8ch_=Zo=dsBugNtkvP8YQ@mail.gmail.com>

morfair, for clarification sip responses do -not- necessarily go to the
originating port, Via Route Contact headers etc can mean other ports are
used. It could be 10.10.10.10 told 20.20.20.20 to respond on 5080. Hence
the request to see the actual packets.

On Fri, 5 Oct 2018 at 00:02, Michael Jerris <mike at jerris.com> wrote:

> I would need to see a full and UNMODIFIED sip trace to properly answer
> this question.
>
> > On Oct 4, 2018, at 11:28 AM, morfair at gmail.com wrote:
> >
> > I have external profile on port 5080. When FS get INVITE on 5080 from
> remote 5060 port, FS response on 5080 port too instead 5060
> >
> > 20.20.20.20 - fs
> >
> > 10.10.10.10 - remote sip
> >
> > 18:23:49.675349 IP 10.10.10.10.5060 > 20.20.20.20.5080: SIP: INVITE
> sip:7001 at 20.20.20.20:5080;user=phone SIP/2.0
> > 18:23:49.678953 IP 20.20.20.20.5080 > 10.10.10.10.5080: UDP, length 363
> >
> >
> > How to fix whis? FS should response on originate udp port.
> >
>
>
> _________________________________________________________________________
> 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 steveayre at gmail.com  Fri Oct  5 15:23:38 2018
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 5 Oct 2018 16:23:38 +0100
Subject: [Freeswitch-users] Is it possible add new item in sip message
	header?
In-Reply-To: <CABHWxvzw+xP0bmwDg3CA-yT2GvT8PsRga726vm9w9tCNZYhvng@mail.gmail.com>
References: <CABHWxvzw+xP0bmwDg3CA-yT2GvT8PsRga726vm9w9tCNZYhvng@mail.gmail.com>
Message-ID: <CAFiqYum-eyojm7n+AQJuBsS+6F2_=SzH07WC-MRh+4qqjB1A0g@mail.gmail.com>

Any specific reason the app needs to check that header? It couldn't be
considered secure since anyone could insert that header.

It's similar to the user-agent which is set in the sofia profile, if
implemented it'd likely be implemented in the same way which'd mean a small
change to the mod_sofia code. I'd file a Jira.

On Thu, 4 Oct 2018 at 17:28, 薛光宏 <albert.hsueh at cloudigit.com> wrote:

> Hi All,
>
> We run a mobile app to *register* FreeSWITCH 1.6.20 via sip. This app
> will check value of server field in message header but the packet sent from
> FreeSWITCH doesn't have this field.
> We would like to know is it possible to add filed "Server" to message
> header by adjusting FreeSWITCH configuration?
>
> We tried this by adjusting dialplan (
> https://freeswitch.org/confluence/display/FREESWITCH/Sofia+SIP+Stack)
>         <action application="set"><![CDATA[sip_h_Server=xxxxxx]]></action>
>
> it works in invite stage, but can not work in register stage. The app
> checks the filed "Server" in register stage.
>
> Thanks in advance.
>
> --
> Sincerely
>
> Albert Hsueh
> System Engineer
> Cloudpe Corporation
> 02-8712 5955 ext. 831
> www.cloudpe.com
> _________________________________________________________________________
> 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 mbodbg at gmx.net  Fri Oct  5 16:05:37 2018
From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=)
Date: Fri, 5 Oct 2018 18:05:37 +0200
Subject: [Freeswitch-users] *.tmp.xml.fsxml Files
Message-ID: <4C8F4B22-30F8-4DAD-BD64-3FE6DEA08252@gmx.net>

Hallo All,

I found an old Post discussing <uuid>.tmp.xml.fsxml files in /var/log/freeswitch.log

http://lists.freeswitch.org/pipermail/freeswitch-users/2016-October/122997.html <http://lists.freeswitch.org/pipermail/freeswitch-users/2016-October/122997.html>

Unfortunately there is no real answer on if they can be disabled or why they are written.

We have one server on Version 1.6.20-37-987c9b9~64bit which is writing about 30+ of those files per minute. So maybe something is going wrong here. Are there any further Details on those files available?

Thanks and regards

Markus

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From mike at jerris.com  Fri Oct  5 19:34:29 2018
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Oct 2018 15:34:29 -0400
Subject: [Freeswitch-users] Is it possible add new item in sip message
 header?
In-Reply-To: <CABHWxvzw+xP0bmwDg3CA-yT2GvT8PsRga726vm9w9tCNZYhvng@mail.gmail.com>
References: <CABHWxvzw+xP0bmwDg3CA-yT2GvT8PsRga726vm9w9tCNZYhvng@mail.gmail.com>
Message-ID: <4EF56976-EE84-4921-A04B-0ED666DC08FE@jerris.com>

There is no way to do this without code modification to add this feature.


> On Oct 4, 2018, at 6:01 AM, 薛光宏 <albert.hsueh at cloudigit.com> wrote:
> 
> Hi All,
> 
> We run a mobile app to register FreeSWITCH 1.6.20 via sip. This app will check value of server field in message header but the packet sent from FreeSWITCH doesn't have this field. 
> We would like to know is it possible to add filed "Server" to message header by adjusting FreeSWITCH configuration?
> 
> We tried this by adjusting dialplan (https://freeswitch.org/confluence/display/FREESWITCH/Sofia+SIP+Stack <https://freeswitch.org/confluence/display/FREESWITCH/Sofia+SIP+Stack>)
>         <action application="set"><![CDATA[sip_h_Server=xxxxxx]]></action>
> 
> it works in invite stage, but can not work in register stage. The app checks the filed "Server" in register stage.
> 

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From mike at jerris.com  Fri Oct  5 19:35:16 2018
From: mike at jerris.com (Michael Jerris)
Date: Fri, 5 Oct 2018 15:35:16 -0400
Subject: [Freeswitch-users] *.tmp.xml.fsxml Files
In-Reply-To: <4C8F4B22-30F8-4DAD-BD64-3FE6DEA08252@gmx.net>
References: <4C8F4B22-30F8-4DAD-BD64-3FE6DEA08252@gmx.net>
Message-ID: <8235F8C9-BF94-4D83-80CB-3D9111B2A6CB@jerris.com>

Sounds like xml_curl debug files.

> On Oct 5, 2018, at 12:05 PM, Markus Bönke <mbodbg at gmx.net> wrote:
> 
> Hallo All,
> 
> I found an old Post discussing <uuid>.tmp.xml.fsxml files in /var/log/freeswitch.log
> 
> http://lists.freeswitch.org/pipermail/freeswitch-users/2016-October/122997.html <http://lists.freeswitch.org/pipermail/freeswitch-users/2016-October/122997.html>
> 
> Unfortunately there is no real answer on if they can be disabled or why they are written.
> 
> We have one server on Version 1.6.20-37-987c9b9~64bit which is writing about 30+ of those files per minute. So maybe something is going wrong here. Are there any further Details on those files available?

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From mario_fs at mgtech.com  Fri Oct  5 20:25:27 2018
From: mario_fs at mgtech.com (Mario)
Date: Fri, 5 Oct 2018 13:25:27 -0700
Subject: [Freeswitch-users] Incorrect example for 'originate' with
 socket and async full
In-Reply-To: <CAHUGhbKq2FEPJ23mfjvfDBSkM+16yz5F6OCFanN_5tsKRoHj-g@mail.gmail.com>
References: <CAHUGhbKq2FEPJ23mfjvfDBSkM+16yz5F6OCFanN_5tsKRoHj-g@mail.gmail.com>
Message-ID: <7337B160-A5A0-4B70-8F69-926E133242AE@mgtech.com>

Fixed.
Mario G

> On Oct 4, 2018, at 5:30 AM, . <joes.mailing.lists at gmail.com> wrote:
> 
> Hi,
> 
> I wanted to report an error/typo in the confluence wiki related to 'originate', and its use together with the socket application.  
> 
> Apologies, I don't have access to the wiki, but thought this was worth passing on, so here it is:
> https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-originate <https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-originate>
> The wiki example for ring_ready also includes the socket application and is written as:
> originate {return_ring_ready=true}sofia/gateway/someprovider/919246461929 &socket(127.0.0.1:8082 <http://127.0.0.1:8082/> async full)
> however this hangs forever if used as written. Eventually, after several hours of hair tearing and googling, I stumbled on this thread:
> https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-3591/FS-3591.html <https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-3591/FS-3591.html>
> Indeed this is the problem.  The correct syntax for the socket component is:
> &socket('127.0.0.1:8082 <http://127.0.0.1:8082/> async full')
> Specifically single quotes around the command so that it is treated as a single string.
> After making this change there is no more hanging on the outbound calls and all events come through in a timely fashion.
> 
> Cheers,
> Joe
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> _________________________________________________________________________
> 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 agoulis at opensips.org  Fri Oct  5 22:24:06 2018
From: agoulis at opensips.org (Alex Goulis)
Date: Fri, 05 Oct 2018 17:24:06 -0500
Subject: [Freeswitch-users] mod_spy, does it only work for registered users?
Message-ID: <E1g8YW5-0004Dy-KP@mail.opensips.org>

Hi all...
Does the userspy application only target registered users or can you use it to spy calls to let's say calls to a DID?

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From andrew at zeta.digital-domain.net  Sat Oct  6 01:04:00 2018
From: andrew at zeta.digital-domain.net (Andrew Clayton)
Date: Sat, 6 Oct 2018 02:04:00 +0100
Subject: [Freeswitch-users] How to update the caller id when bridging?
Message-ID: <20181006020400.3324ed18@kappa.digital-domain.net>

So this is a question based around the FreeSWITCH C API.

I have two scenarios involving two endpoints, A & B.

Scenario 1

FS calls B using switch_ivr_originate(), it then calls A (say a
Linphone client) and creates a new switch_caller_profile_t for it using
switch_caller_profile_new() and sets the ->caller_id_number field to e.g
1234 to indicate a call from B and then does a switch_ivr_originate().

A rings and Linphone shows the call coming from 1234 at ... OK, that's all
fine. The calls are bridged together with switch_ivr_uuid_bridge(A, B)

Scenario 2

FS calls A (again lets say a Linphone client) using
switch_ivr_originate(), A rings and shows the call coming from
0000... at ..., that's fine, I didn't set any CID.

Now A is parked and is off-hook.

FS then calls B, A & B are then bridged together with
switch_ivr_uuid_bridge(A, B), now of course Linphone still shows the
call as from  0000... at ... I'd like it show for example 1234 at ... like
above.

Now I've tried setting numerous fields in the switch_caller_profile_t
structure for A just before doing the bridge, but to no avail.

So does what I'm trying to do make sense?

Is it even possible?

Cheers,
Andrew


From patel.jenish.k at gmail.com  Sat Oct  6 06:53:29 2018
From: patel.jenish.k at gmail.com (Jenish Patel)
Date: Sat, 6 Oct 2018 12:23:29 +0530
Subject: [Freeswitch-users] execute_on_answer does not work when group dial
 is used for bridging to multiple extensions
Message-ID: <CALbqRWgWsLVpU6hg6ZbnHtGkwknCX4cVzET8OSd4MGyf+jKzJg@mail.gmail.com>

Hi everyone,

I am trying to use the bind_digit_action on the b-leg , and as wiki
suggested I am using execute_on_answer to assign the bind_digit on the
b-leg. This works like charm in case of the normal simple bridge. But in
case of the Ring-group like bridge when I bridge the multiple users at the
same time, it does not get executed on the channel answer.

I see in the logs that in working case when b-leg answered , the
execute_on_answer gets executed. But in ring group case it does not get
executed at all.

Am I missing something here ? Like is there any other param needs to be set
to use the execute_on_answer in the ring group like structure.

And if it is not possible, can you suggest me any other way to use
bind_digit on b-leg from dialplan itself.

Below is my dialplan for ring group.




<include>

   <extension name="Ring_all">
      <condition field="destination_number" expression="3301">
      <action application="export" data="dialed_extension=$1"/>

<!-- Feature Code settings -->
      <!-- bind_meta_app can have these args <key> [a|b|ab] [a|b|o|s] <app>
-->
        <action application="bind_digit_action"
data="features,*1,exec:valet_park::my_lot auto in 8001 8999,both,self"/>
        <action application="export"
data="nolocal:execute_on_answer_11=features,*1,exec:valet_park::my_lot auto
in 8001 8999,both,self"/>

        <action application="bind_digit_action"
data="features,*2,exec:execute_extension,START_RECORDING XML
features,both,self"/>
        <action application="export"
data="nolocal:execute_on_answer_12=features,*2,exec:execute_extension,START_RECORDING
XML features,both,self"/>

        <action application="bind_digit_action"
data="features,*3,exec:execute_extension,att_xfer XML features,both,self"/>
        <action application="export"
data="nolocal:execute_on_answer_13=bind_digit_action
features,*3,exec:execute_extension,att_xfer XML features,both,self"/>

        <action application="bind_digit_action"
data="features,*4,exec:execute_extension,blind_xfer XML
features,both,self"/>
        <action application="export"
data="nolocal:execute_on_answer_14=bind_digit_action
features,*4,exec:execute_extension,blind_xfer XML features,both,self"/>

        <action application="bind_digit_action"
data="features,*51,exec:execute_extension,START_RECORDING XML
features,aleg,self"/>
        <action application="export"
data="nolocal:execute_on_answer_15=features,*51,exec:execute_extension,START_RECORDING
XML features,aleg,self"/>

        <action application="bind_digit_action"
data="features,*52,exec:execute_extension,STOP_RECORDING XML
features,aleg,self"/>
        <action application="export"
data="nolocal:execute_on_answer_16=features,*52,exec:execute_extension,STOP_RECORDING
XML features,aleg,self"/>

<!-- FEature Code set -->


<!-- Setting Variabls -->
                        <action application="set"
data="ringback=/home/jenish/moh.wav"/>
                        <action application="set"
data="transfer_ringback=/home/jenish.moh.wav"/>
                        <action application="set"
data="call_timeout=300"/>
                        <action application="set"
data="hangup_after_bridge=true"/>
                        <action application="set"
data="continue_on_fail=true"/>
                        <action application="set" data="REC_TYPE=GROUP"/>
                        <action application="set" data="REC_STATUS=Y"/>
                        <action application="export"
data="effective_caller_id_name=Department_group_call"/>


                        <action application="record_session"
data="/usr/local/freeswitch/recordings/1000/GROUP/${strftime(%Y-%m-%d-%H-%M-%S)}_from_${caller_id_number}_to_${destination_number}__${uuid}.wav"/>
                        <action application="set"
data="REC_PATH=/usr/local/freeswitch/recordings/1000/GROUP/${strftime(%Y-%m-%d-%H-%M-%S)}_from_${caller_id_number}_to_${destination_number}__${uuid}.wav"/>

                        <action application="set"
data="export_vars=call_direction,original_source,REC_PATH,REC_TYPE,REC_STATUS"/>

                        <action application="bridge"
data="[absolute_codec_string=^^:PCMA at 8000h@20i at 64000b
]${sofia_contact(internal/1000101@
${domain_name}},[absolute_codec_string=^^:PCMU at 8000h@20i at 64000b
]${sofia_contact(internal/1000100@
${domain_name}},[absolute_codec_string=^^:PCMA at 8000h@20i at 64000b:PCMU at 8000h
@20i at 64000b]${sofia_contact(internal/1000102@
${domain_name}},[absolute_codec_string=^^:PCMA at 8000h@20i at 64000b:PCMU at 8000h
@20i at 64000b]${sofia_contact(internal/1000103@${domain_name}}"/>

                        <action application="answer"/>
                        <action application="sleep" data="1000"/>
                        <action application="bridge"
data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>

</include>


-- 

Thanks & Regards

Jenish Patel
+91 9033987576
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From albert.hsueh at cloudigit.com  Mon Oct  8 01:30:36 2018
From: albert.hsueh at cloudigit.com (=?UTF-8?B?6Jab5YWJ5a6P?=)
Date: Mon, 8 Oct 2018 09:30:36 +0800
Subject: [Freeswitch-users] Is it possible add new item in sip message
	header?
In-Reply-To: <CAFiqYum-eyojm7n+AQJuBsS+6F2_=SzH07WC-MRh+4qqjB1A0g@mail.gmail.com>
References: <CABHWxvzw+xP0bmwDg3CA-yT2GvT8PsRga726vm9w9tCNZYhvng@mail.gmail.com>
 <CAFiqYum-eyojm7n+AQJuBsS+6F2_=SzH07WC-MRh+4qqjB1A0g@mail.gmail.com>
Message-ID: <CABHWxvwRysqdMd2YFDGjSEZQnz1_z86scZqApHV99XEOzXy5HQ@mail.gmail.com>

Hi

Thanks for reply.
The app is for school. We think the developer wish this mobile app only
connect to specific sip server so they do this kind of check.

Sincerley
Albert

Steven Ayre <steveayre at gmail.com> 於 2018年10月6日 週六 上午3:33寫道：

> Any specific reason the app needs to check that header? It couldn't be
> considered secure since anyone could insert that header.
>
> It's similar to the user-agent which is set in the sofia profile, if
> implemented it'd likely be implemented in the same way which'd mean a small
> change to the mod_sofia code. I'd file a Jira.
>
> On Thu, 4 Oct 2018 at 17:28, 薛光宏 <albert.hsueh at cloudigit.com> wrote:
>
>> Hi All,
>>
>> We run a mobile app to *register* FreeSWITCH 1.6.20 via sip. This app
>> will check value of server field in message header but the packet sent from
>> FreeSWITCH doesn't have this field.
>> We would like to know is it possible to add filed "Server" to message
>> header by adjusting FreeSWITCH configuration?
>>
>> We tried this by adjusting dialplan (
>> https://freeswitch.org/confluence/display/FREESWITCH/Sofia+SIP+Stack)
>>         <action application="set"><![CDATA[sip_h_Server=xxxxxx]]></action>
>>
>> it works in invite stage, but can not work in register stage. The app
>> checks the filed "Server" in register stage.
>>
>> Thanks in advance.
>>
>> --
>> Sincerely
>>
>> Albert Hsueh
>> System Engineer
>> Cloudpe Corporation
>> 02-8712 5955 ext. 831
>> www.cloudpe.com
>> _________________________________________________________________________
>> 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
>
> _________________________________________________________________________
> 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

Albert Hsueh
System Engineer
Cloudpe Corporation
02-8712 5955 ext. 831
www.cloudpe.com
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From albert.hsueh at cloudigit.com  Mon Oct  8 01:32:22 2018
From: albert.hsueh at cloudigit.com (=?UTF-8?B?6Jab5YWJ5a6P?=)
Date: Mon, 8 Oct 2018 09:32:22 +0800
Subject: [Freeswitch-users] Is it possible add new item in sip message
	header?
In-Reply-To: <4EF56976-EE84-4921-A04B-0ED666DC08FE@jerris.com>
References: <CABHWxvzw+xP0bmwDg3CA-yT2GvT8PsRga726vm9w9tCNZYhvng@mail.gmail.com>
 <4EF56976-EE84-4921-A04B-0ED666DC08FE@jerris.com>
Message-ID: <CABHWxvzb=megjcHw0vnUrLM7-n6cJq8KCo_J0bE+JNQTck_zww@mail.gmail.com>

Hi

Thanks for reply.
We will consider stop looking for that.

Sincerely
Albert

Michael Jerris <mike at jerris.com> 於 2018年10月6日 週六 上午4:31寫道：

> There is no way to do this without code modification to add this feature.
>
>
> On Oct 4, 2018, at 6:01 AM, 薛光宏 <albert.hsueh at cloudigit.com> wrote:
>
> Hi All,
>
> We run a mobile app to *register* FreeSWITCH 1.6.20 via sip. This app
> will check value of server field in message header but the packet sent from
> FreeSWITCH doesn't have this field.
> We would like to know is it possible to add filed "Server" to message
> header by adjusting FreeSWITCH configuration?
>
> We tried this by adjusting dialplan (
> https://freeswitch.org/confluence/display/FREESWITCH/Sofia+SIP+Stack)
>         <action application="set"><![CDATA[sip_h_Server=xxxxxx]]></action>
>
> it works in invite stage, but can not work in register stage. The app
> checks the filed "Server" in register stage.
>
>
> _________________________________________________________________________
> 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

Albert Hsueh
System Engineer
Cloudpe Corporation
02-8712 5955 ext. 831
www.cloudpe.com
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From brian at freeswitch.com  Mon Oct  8 10:13:46 2018
From: brian at freeswitch.com (Brian West)
Date: Mon, 8 Oct 2018 05:13:46 -0500
Subject: [Freeswitch-users] Incorrect example for 'originate' with
 socket and async full
In-Reply-To: <7337B160-A5A0-4B70-8F69-926E133242AE@mgtech.com>
References: <CAHUGhbKq2FEPJ23mfjvfDBSkM+16yz5F6OCFanN_5tsKRoHj-g@mail.gmail.com>
 <7337B160-A5A0-4B70-8F69-926E133242AE@mgtech.com>
Message-ID: <CAEJMVkDchaK2AutTfzxVKZ=b-XG8SX_711ODP7ZU=0fKKmij+g@mail.gmail.com>

The &app notation shouldn’t be used anymore the inline dialplan is the
appropriate way to do this moving forward.

/b

On Fri, Oct 5, 2018 at 6:41 PM Mario <mario_fs at mgtech.com> wrote:

> Fixed.
> Mario G
>
>
> On Oct 4, 2018, at 5:30 AM, . <joes.mailing.lists at gmail.com> wrote:
>
> Hi,
>
> I wanted to report an error/typo in the confluence wiki related to
> 'originate', and its use together with the socket application.
>
> Apologies, I don't have access to the wiki, but thought this was worth
> passing on, so here it is:
>
>    -
>    https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-originate
>
> The wiki example for ring_ready also includes the socket application and
> is written as:
>
>    -
>
>    originate {return_ring_ready=true}sofia/gateway/someprovider/919246461929 &socket(127.0.0.1:8082 async full)
>
>
> however this hangs forever if used as written. Eventually, after several
> hours of hair tearing and googling, I stumbled on this thread:
>
>    -
>    https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-3591/FS-3591.html
>
> Indeed this is the problem.  The correct syntax for the socket component
> is:
>
>    -
>
>    &socket('127.0.0.1:8082 async full')
>
>    - Specifically single quotes around the command so that it is treated
>       as a single string.
>
> After making this change there is no more hanging on the outbound calls
> and all events come through in a timely fashion.
>
> Cheers,
> Joe
>
>
>
>
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> 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
>
>
> _________________________________________________________________________
> 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 michael at mailworks.org  Mon Oct  8 14:21:49 2018
From: michael at mailworks.org (Michael Avers)
Date: Mon, 08 Oct 2018 07:21:49 -0700
Subject: [Freeswitch-users] Incorrect example for 'originate' with
 socket and async full
In-Reply-To: <CAEJMVkDchaK2AutTfzxVKZ=b-XG8SX_711ODP7ZU=0fKKmij+g@mail.gmail.com>
References: <CAHUGhbKq2FEPJ23mfjvfDBSkM+16yz5F6OCFanN_5tsKRoHj-g@mail.gmail.com>
 <7337B160-A5A0-4B70-8F69-926E133242AE@mgtech.com>
 <CAEJMVkDchaK2AutTfzxVKZ=b-XG8SX_711ODP7ZU=0fKKmij+g@mail.gmail.com>
Message-ID: <1539008509.884661.1534620552.4EF2A2AC@webmail.messagingengine.com>

Hi Brian can you please clarify what you mean by online dialplan? Why
shouldn't &app be used anymore?
Thank you
Mike


On Mon, Oct 8, 2018, at 3:13 AM, Brian West wrote:
> The &app notation shouldn’t be used anymore the inline dialplan is the
> appropriate way to do this moving forward.> 
> /b
> 
> On Fri, Oct 5, 2018 at 6:41 PM Mario <mario_fs at mgtech.com> wrote:
>> Fixed.
>> Mario G
>> 
>> 
>>> On Oct 4, 2018, at 5:30 AM, . <joes.mailing.lists at gmail.com> wrote:>>> 
>>> Hi,
>>> 
>>> I wanted to report an error/typo in the confluence wiki related to
>>> 'originate', and its use together with the socket application.>>> 
>>> Apologies, I don't have access to the wiki, but thought this was
>>> worth passing on, so here it is:>>>  * https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-originate>>> The wiki example for ring_ready also includes the socket application
>>> and is written as:>>>  * originate
>>>    {return_ring_ready=true}sofia/gateway/someprovider/919246461929
>>>    &socket(127.0.0.1:8082[1] async full)>>> however this hangs forever if used as written. Eventually, after
>>> several hours of hair tearing and googling, I stumbled on this
>>> thread:>>>  * https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-3591/FS-3591.html>>> Indeed this is the problem.  The correct syntax for the socket
>>> component is:>>>  * &socket('127.0.0.1:8082[2] async full')
>>>    * Specifically single quotes around the command so that it is
>>>      treated as a single string.>>> After making this change there is no more hanging on the outbound
>>> calls and all events come through in a timely fashion.>>> 
>>> Cheers,
>>> Joe
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> ___________________________________________________________________-
>>> ______>>> 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
>> 
>> __________________________________________________________________-
>> _______>>  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[3]> Email: brian at freeswitch.com


> Mobile: 918-424-9378


> Website: https://www.FreeSWITCH.com[4]


> https://www.facebook.com/signalwireinc?src=email
> https://twitter.com/freeswitch> ___________________________________________________________________-
> ________> 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


Links:

  1. http://127.0.0.1:8082/
  2. http://127.0.0.1:8082/
  3. https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g
  4. https://www.freeswitch.com/
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From sagarmalam at gmail.com  Mon Oct  8 15:03:46 2018
From: sagarmalam at gmail.com (sagar malam)
Date: Mon, 8 Oct 2018 20:33:46 +0530
Subject: [Freeswitch-users] No RTP after unholding call
Message-ID: <CAEZuF3tCfuD3hnH9SYNM94DXKMMHszCJJtgCWniidrn4aJEKuA@mail.gmail.com>

Caller A and Caller B are on call.Caller A puts call on hold for 2
mins.During these 2 mins FS does not send any RTP to firewall due to which
firewall closes RTP port of caller A due to udp inactivity
can we configure FS to send something like empty RTP to caller A ?
I did not find any settings in wiki

Thanks in advance
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From sagarmalam at gmail.com  Mon Oct  8 15:05:12 2018
From: sagarmalam at gmail.com (sagar malam)
Date: Mon, 8 Oct 2018 20:35:12 +0530
Subject: [Freeswitch-users] Codec negotiation issue
In-Reply-To: <CAEZuF3tXU_zexmYXWzVbf-4pvj52qivnvEi4oZCNwCSEj0KdDw@mail.gmail.com>
References: <CAEZuF3uvOe9MmrH+LjMuJ3MDMCPXZqL_0DFp5uwkaq9OcDKv3A@mail.gmail.com>
 <9E9891C3-7017-46B1-B5AD-738CDF7DB9A5@jerris.com>
 <CAEZuF3tXU_zexmYXWzVbf-4pvj52qivnvEi4oZCNwCSEj0KdDw@mail.gmail.com>
Message-ID: <CAEZuF3t_M-ym8pfQ9PWHBk0UzGkb8EvAzsiv7ha8xOMTCG=pGw@mail.gmail.com>

inherit_codec setting works.

Thanks for the help

On Wed, Oct 3, 2018 at 4:42 PM sagar malam <sagarmalam at gmail.com> wrote:

> Thanks.
>
> I will test and get back
>
>
> On Wed, Oct 3, 2018 at 3:13 AM Michael Jerris <mike at jerris.com> wrote:
>
>> https://freeswitch.org/confluence/display/FREESWITCH/inherit_codec
>>
>>
>> On Oct 2, 2018, at 7:05 AM, sagar malam <sagarmalam at gmail.com> wrote:
>>
>> I am facing an issue related to codec negotiation.Let me explain scenario
>>  inbound/outbound codec pref list : g722,g729,g711
>>  late negotiation enabled
>>  mix inbound outbound codec is enabled
>>  When call hits FS, i give instant ringback to call which mean g722 codec
>> will be negotiated for early media
>> Now when callee answers call with g729
>> FS does transcoding from g722: g729
>> Instead it should select g729 on both legs and tell aleg to update codec
>> from g722 to g729 to avoid transcoding.
>>
>>
>> _________________________________________________________________________
>> 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 brian at freeswitch.com  Mon Oct  8 18:33:35 2018
From: brian at freeswitch.com (Brian West)
Date: Mon, 8 Oct 2018 14:33:35 -0400
Subject: [Freeswitch-users] Incorrect example for 'originate' with
 socket and async full
In-Reply-To: <1539008509.884661.1534620552.4EF2A2AC@webmail.messagingengine.com>
References: <CAHUGhbKq2FEPJ23mfjvfDBSkM+16yz5F6OCFanN_5tsKRoHj-g@mail.gmail.com>
 <7337B160-A5A0-4B70-8F69-926E133242AE@mgtech.com>
 <CAEJMVkDchaK2AutTfzxVKZ=b-XG8SX_711ODP7ZU=0fKKmij+g@mail.gmail.com>
 <1539008509.884661.1534620552.4EF2A2AC@webmail.messagingengine.com>
Message-ID: <CAEJMVkALhsdbXWscfAfp2GFBg-f1xRV1n0baY00jxP+P==zB_A@mail.gmail.com>

Because the &app method has been deprecated for a while now and the way you
should accomplish this is via inline dialplan

https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+Inline+Dialplan

Its actualy more powerful.

On Mon, Oct 8, 2018 at 11:05 AM Michael Avers <michael at mailworks.org> wrote:

> Hi Brian can you please clarify what you mean by online dialplan? Why
> shouldn't &app be used anymore?
>
> Thank you
> Mike
>
>
> On Mon, Oct 8, 2018, at 3:13 AM, Brian West wrote:
>
> The &app notation shouldn’t be used anymore the inline dialplan is the
> appropriate way to do this moving forward.
>
> /b
>
> On Fri, Oct 5, 2018 at 6:41 PM Mario <mario_fs at mgtech.com> wrote:
>
> Fixed.
> Mario G
>
>
> On Oct 4, 2018, at 5:30 AM, . <joes.mailing.lists at gmail.com> wrote:
>
> Hi,
>
> I wanted to report an error/typo in the confluence wiki related to
> 'originate', and its use together with the socket application.
>
> Apologies, I don't have access to the wiki, but thought this was worth
> passing on, so here it is:
>
>    -
>    https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-originate
>
> The wiki example for ring_ready also includes the socket application and
> is written as:
>
>    -
>
>    originate {return_ring_ready=true}sofia/gateway/someprovider/919246461929 &socket(127.0.0.1:8082 async full)
>
>
> however this hangs forever if used as written. Eventually, after several
> hours of hair tearing and googling, I stumbled on this thread:
>
>    -
>    https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-3591/FS-3591.html
>
> Indeed this is the problem.  The correct syntax for the socket component
> is:
>
>    -
>
>    &socket('127.0.0.1:8082 async full')
>
>    - Specifically single quotes around the command so that it is treated
>       as a single string.
>
> After making this change there is no more hanging on the outbound calls
> and all events come through in a timely fashion.
>
> Cheers,
> Joe
>
>
>
>
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> 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
>
>
> _________________________________________________________________________
> 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>
> *_________________________________________________________________________*
> 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
>
>
> _________________________________________________________________________
> 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 BJordan at E-Teleco.com  Mon Oct  8 19:42:01 2018
From: BJordan at E-Teleco.com (Branden Jordan)
Date: Mon, 8 Oct 2018 19:42:01 +0000
Subject: [Freeswitch-users] No RTP after unholding call
In-Reply-To: <CAEZuF3tCfuD3hnH9SYNM94DXKMMHszCJJtgCWniidrn4aJEKuA@mail.gmail.com>
References: <CAEZuF3tCfuD3hnH9SYNM94DXKMMHszCJJtgCWniidrn4aJEKuA@mail.gmail.com>
Message-ID: <890b49783c12468cbf0b4f6cfdad99ec@MDF-EXCH1.MDF-Holdings.local>

Maybe comfort noise will help you achieve what you are trying to do here?

https://freeswitch.org/confluence/display/FREESWITCH/bridge_generate_comfort_noise
https://freeswitch.org/confluence/display/FREESWITCH/send_silence_when_idle

Thanks,
Branden

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of sagar malam
Sent: Monday, October 08, 2018 8:04 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] No RTP after unholding call

Caller A and Caller B are on call.Caller A puts call on hold for 2 mins.During these 2 mins FS does not send any RTP to firewall due to which firewall closes RTP port of caller A due to udp inactivity
can we configure FS to send something like empty RTP to caller A ?
I did not find any settings in wiki

Thanks in advance
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From BJordan at E-Teleco.com  Mon Oct  8 23:50:56 2018
From: BJordan at E-Teleco.com (Branden Jordan)
Date: Mon, 8 Oct 2018 23:50:56 +0000
Subject: [Freeswitch-users] No RTP after unholding call
In-Reply-To: <CAEZuF3tCfuD3hnH9SYNM94DXKMMHszCJJtgCWniidrn4aJEKuA@mail.gmail.com>
References: <CAEZuF3tCfuD3hnH9SYNM94DXKMMHszCJJtgCWniidrn4aJEKuA@mail.gmail.com>
Message-ID: <ba88ecfa19e2459d8ba808d6114ff236@MDF-EXCH1.MDF-Holdings.local>

Maybe comfort noise will help you achieve what you are trying to do here?

https://freeswitch.org/confluence/display/FREESWITCH/bridge_generate_comfort_noise
https://freeswitch.org/confluence/display/FREESWITCH/send_silence_when_idle

Thanks,
Branden


From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of sagar malam
Sent: Monday, October 08, 2018 8:04 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] No RTP after unholding call

Caller A and Caller B are on call.Caller A puts call on hold for 2 mins.During these 2 mins FS does not send any RTP to firewall due to which firewall closes RTP port of caller A due to udp inactivity
can we configure FS to send something like empty RTP to caller A ?
I did not find any settings in wiki

Thanks in advance
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From joe at expert.net  Tue Oct  9 02:07:45 2018
From: joe at expert.net (Joseph Barrero)
Date: Mon, 8 Oct 2018 21:07:45 -0500
Subject: [Freeswitch-users] VP9 Error in Version 1.8.2
Message-ID: <CAGAa33tnWfCK0dHVpr6PZOW8-uHCDa=f07j537sB4i2L-Z8LdA@mail.gmail.com>

I'm receiving the following error message in freeswitch.log ever since
updating to version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git
a98a958 2018-09-26 17:55:25Z 64bit) on Debian 8.11 (amd-64):

[ERR] switch_vpx.c:529 VPX encoder WebM Project VP9 Encoder v1.6.0 codec
init error: [8:Profile > 1 not supported in this build configuration]


The error is present when I'm using WebRTC to connect to FreeSWITCH on
either Chrome or Firefox.  Any advice on how to fix is greatly appreciated.


- Joe
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From ksrigo at gmail.com  Tue Oct  9 06:20:57 2018
From: ksrigo at gmail.com (Srigo Kanapathipillai)
Date: Tue, 09 Oct 2018 08:20:57 +0200
Subject: [Freeswitch-users] No RTP after unholding call
In-Reply-To: <CAEZuF3tCfuD3hnH9SYNM94DXKMMHszCJJtgCWniidrn4aJEKuA@mail.gmail.com>
References: <CAEZuF3tCfuD3hnH9SYNM94DXKMMHszCJJtgCWniidrn4aJEKuA@mail.gmail.com>
Message-ID: <bdcbdfac-cfbf-42a2-91ee-2648784ff1eb@gmail.com>

Try this:

<action application="set" data="hold_music=silence" />

Or

<action application="set" data="hold_music=/sounds/holdmusic.wav" />


⁣Srigo Kana​

On 8 Oct 2018, 17:43, at 17:43, sagar malam <sagarmalam at gmail.com> wrote:
>Caller A and Caller B are on call.Caller A puts call on hold for 2
>mins.During these 2 mins FS does not send any RTP to firewall due to
>which
>firewall closes RTP port of caller A due to udp inactivity
>can we configure FS to send something like empty RTP to caller A ?
>I did not find any settings in wiki
>
>Thanks in advance
>
>
>------------------------------------------------------------------------
>
>_________________________________________________________________________
>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 sagarmalam at gmail.com  Tue Oct  9 11:52:41 2018
From: sagarmalam at gmail.com (sagar malam)
Date: Tue, 9 Oct 2018 17:22:41 +0530
Subject: [Freeswitch-users] Shared presence not working with callcenter
	application of Freeswitch
Message-ID: <CAEZuF3vgN4OdaRha=edmCkhd_Ac-ZLThs6_uXx2LT3FGdVgfdQ@mail.gmail.com>

I am facing issue related to presence/ shared presence for agents in
callcenter(mod_callcenter).
Suppose we are dialling a user 1001 at example.com and it is a shared user in
two phones. Shared presence works perfectly if we directly dial extension
1001.

But if same extension is configured as agent of callcenter and call is
bridged to agent(extension) through callcenter, shared presence does not
work. Freeswitch does not generate notify for agent state change.

Is this expected behaviour ? If not then what can fix this ?

Thanks in advance
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From mbodbg at gmx.net  Tue Oct  9 15:52:06 2018
From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=)
Date: Tue, 9 Oct 2018 17:52:06 +0200
Subject: [Freeswitch-users] Sometimes "hanging" channels
Message-ID: <87D934A1-4A37-4709-93B5-96FA0659ABBE@gmx.net>

Hello All,

we are running freeswitch 1.6.20, calls are controlled via ESL, CDRs are written with mod_xml_cdrl. Sometimes we see „hanging“ channels. In such a case the CDR via mod_xml_cdr is written and the last log entry for such a call is "Locked, Waiting on external entities“.
...
freeswitch.log.2018-10-09-10-19-07.1:db2908e3-c06b-4293-88b1-e18cffb66263 2018-10-09 10:18:15.197345 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/anonymous at anonymous.invalid) State Change CS_REPORTING -> CS_DESTROY
freeswitch.log.2018-10-09-10-19-07.1:db2908e3-c06b-4293-88b1-e18cffb66263 2018-10-09 10:18:15.197345 [DEBUG] switch_core_session.c:1665 Session 166111 (sofia/internal/anonymous at anonymous.invalid) Locked, Waiting on external entities
...
How can I proceed to further analyze the problem ? In the last log line I also see the session number (Session 166111) - is there a way to find out on which external entity it is waiting?

Thanks and regards

Markus 
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From mbodbg at gmx.net  Tue Oct  9 16:37:29 2018
From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=)
Date: Tue, 9 Oct 2018 18:37:29 +0200
Subject: [Freeswitch-users] *.tmp.xml.fsxml Files
In-Reply-To: <8235F8C9-BF94-4D83-80CB-3D9111B2A6CB@jerris.com>
References: <4C8F4B22-30F8-4DAD-BD64-3FE6DEA08252@gmx.net>
 <8235F8C9-BF94-4D83-80CB-3D9111B2A6CB@jerris.com>
Message-ID: <752F9A18-3B2B-4783-9D44-85237BE840D8@gmx.net>

The files were indeed generated when requests from mod_xml_curl got not answered properly.

Thanks

Markus
> Am 05.10.2018 um 21:35 schrieb Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>:
> 
> Sounds like xml_curl debug files.
> 
>> On Oct 5, 2018, at 12:05 PM, Markus Bönke <mbodbg at gmx.net <mailto:mbodbg at gmx.net>> wrote:
>> 
>> Hallo All,
>> 
>> I found an old Post discussing <uuid>.tmp.xml.fsxml files in /var/log/freeswitch.log
>> 
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2016-October/122997.html <http://lists.freeswitch.org/pipermail/freeswitch-users/2016-October/122997.html>
>> 
>> Unfortunately there is no real answer on if they can be disabled or why they are written.
>> 
>> We have one server on Version 1.6.20-37-987c9b9~64bit which is writing about 30+ of those files per minute. So maybe something is going wrong here. Are there any further Details on those files available?
> 
> _________________________________________________________________________
> 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
> 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 JHChinn at TheNavisWay.com  Tue Oct  9 18:49:26 2018
From: JHChinn at TheNavisWay.com (Jerry Chinn)
Date: Tue, 9 Oct 2018 18:49:26 +0000
Subject: [Freeswitch-users] SNMP v3
Message-ID: <5e2c8e24ef3b4138bea3ef402bf48a88@hil-vs-exdag02.buehner-fry.com>

Anyone know if SNMP v3 is supported in FS 1.6.20?
I can't find anything recent on SNMP.
Mod_snmp documentation is from 2005.

Jerry Chinn
Telecom VoIP Specialist
NAVIS More Performance. More Profit.
tel 541-330-3562
www.TheNavisWay.com<http://www.thenavisway.com/>
Facebook<https://www.facebook.com/theNAVISway/> | Twitter<https://twitter.com/NAVISway> | LinkedIn<https://www.linkedin.com/company/navisway> | Blog<https://www.thenavisway.com/blog>

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From shaun.stokes at itec-support.co.uk  Tue Oct  9 20:32:53 2018
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Tue, 9 Oct 2018 20:32:53 +0000
Subject: [Freeswitch-users] Shared presence not working with
	callcenter	application of Freeswitch
In-Reply-To: <CAEZuF3vgN4OdaRha=edmCkhd_Ac-ZLThs6_uXx2LT3FGdVgfdQ@mail.gmail.com>
References: <CAEZuF3vgN4OdaRha=edmCkhd_Ac-ZLThs6_uXx2LT3FGdVgfdQ@mail.gmail.com>
Message-ID: <LNXP265MB1180E1DFD30C173B4DFC6759EBE70@LNXP265MB1180.GBRP265.PROD.OUTLOOK.COM>

You need to set the presence_id variable of your agent when you bridge to the agent in the agent contact field.

Shaun

Get Outlook for iOS<https://aka.ms/o0ukef>
________________________________
From: 20110170300n behalf of
Sent: Tuesday, October 9, 2018 19:48
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Shared presence not working with callcenter application of Freeswitch

I am facing issue related to presence/ shared presence for agents in callcenter(mod_callcenter).
Suppose we are dialling a user 1001 at example.com<mailto:1001 at example.com> and it is a shared user in two phones. Shared presence works perfectly if we directly dial extension 1001.

But if same extension is configured as agent of callcenter and call is bridged to agent(extension) through callcenter, shared presence does not work. Freeswitch does not generate notify for agent state change.

Is this expected behaviour ? If not then what can fix this ?

Thanks in advance
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From social at bohboh.info  Tue Oct  9 20:37:32 2018
From: social at bohboh.info (Social Boh)
Date: Tue, 9 Oct 2018 15:37:32 -0500
Subject: [Freeswitch-users] SNMP v3
In-Reply-To: <5e2c8e24ef3b4138bea3ef402bf48a88@hil-vs-exdag02.buehner-fry.com>
References: <5e2c8e24ef3b4138bea3ef402bf48a88@hil-vs-exdag02.buehner-fry.com>
Message-ID: <eb076238-770d-8218-0c20-211ebdb8be19@bohboh.info>

I tested mod_snmp last week with versión 2 and works fine.

Regards

---
I'm SoCIaL, MayBe

El 09/10/2018 a las 13:49, Jerry Chinn escribió:
>
> Anyone know if SNMP v3 is supported in FS 1.6.20?
>
> I can’t find anything recent on SNMP.
>
> Mod_snmp documentation is from 2005.
>
> *Jerry Chinn*
>
> *Telecom VoIP Specialist*
>
> *NAVIS *More Performance. More Profit.
>
> tel 541-330-3562
>
> www.TheNavisWay.com <http://www.thenavisway.com/>
>
> Facebook <https://www.facebook.com/theNAVISway/> | Twitter 
> <https://twitter.com/NAVISway> | LinkedIn 
> <https://www.linkedin.com/company/navisway> | Blog 
> <https://www.thenavisway.com/blog>
>
>
>
> _________________________________________________________________________
> 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 me at nevian.org  Tue Oct  9 21:34:57 2018
From: me at nevian.org (Serge S. Yuriev)
Date: Wed, 10 Oct 2018 00:34:57 +0300
Subject: [Freeswitch-users] SNMP v3
In-Reply-To: <5e2c8e24ef3b4138bea3ef402bf48a88@hil-vs-exdag02.buehner-fry.com>
References: <5e2c8e24ef3b4138bea3ef402bf48a88@hil-vs-exdag02.buehner-fry.com>
Message-ID: <8421751539120897@sas1-0a6c2e2b59d7.qloud-c.yandex.net>

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From krice at freeswitch.org  Tue Oct  9 22:37:50 2018
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 9 Oct 2018 17:37:50 -0500
Subject: [Freeswitch-users] SNMP v3
In-Reply-To: <eb076238-770d-8218-0c20-211ebdb8be19@bohboh.info>
References: <5e2c8e24ef3b4138bea3ef402bf48a88@hil-vs-exdag02.buehner-fry.com>
 <eb076238-770d-8218-0c20-211ebdb8be19@bohboh.info>
Message-ID: <CEF784EF-1AB4-4058-9639-CEE0145C5476@freeswitch.org>

freeswitch depends in net-snmp for the actual snmp daemon functions so v2 or v3 needs to be supported there. 

mod_snmp is the module for freeswitch to feed data over to snmpd. mod_snmp can not be queried directly by snmp on the wire

Sent from my iPhone

> On Oct 9, 2018, at 15:37, Social Boh <social at bohboh.info> wrote:
> 
> I tested mod_snmp last week with versión 2 and works fine.
> Regards
> ---
> I'm SoCIaL, MayBe
>> El 09/10/2018 a las 13:49, Jerry Chinn escribió:
>> Anyone know if SNMP v3 is supported in FS 1.6.20?
>> I can’t find anything recent on SNMP.
>> Mod_snmp documentation is from 2005.
>>  
>> Jerry Chinn
>> Telecom VoIP Specialist
>> NAVIS More Performance. More Profit.
>> tel 541-330-3562   
>> www.TheNavisWay.com
>> Facebook | Twitter | LinkedIn | Blog
>>  
>> 
>> 
>> _________________________________________________________________________
>> 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
> 
> _________________________________________________________________________
> 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 brian at freeswitch.com  Wed Oct 10 03:06:30 2018
From: brian at freeswitch.com (Brian West)
Date: Tue, 9 Oct 2018 23:06:30 -0400
Subject: [Freeswitch-users] VP9 Error in Version 1.8.2
In-Reply-To: <CAGAa33tnWfCK0dHVpr6PZOW8-uHCDa=f07j537sB4i2L-Z8LdA@mail.gmail.com>
References: <CAGAa33tnWfCK0dHVpr6PZOW8-uHCDa=f07j537sB4i2L-Z8LdA@mail.gmail.com>
Message-ID: <CAEJMVkDQy8mPOkQrRc=f=StewbxxVGdLRefm8sVjYuE6d=5U1Q@mail.gmail.com>

vp9 is not supported.

On Tue, Oct 9, 2018 at 2:42 PM Joseph Barrero <joe at expert.net> wrote:

> I'm receiving the following error message in freeswitch.log ever since
> updating to version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git
> a98a958 2018-09-26 17:55:25Z 64bit) on Debian 8.11 (amd-64):
>
> [ERR] switch_vpx.c:529 VPX encoder WebM Project VP9 Encoder v1.6.0 codec
> init error: [8:Profile > 1 not supported in this build configuration]
>
>
> The error is present when I'm using WebRTC to connect to FreeSWITCH on
> either Chrome or Firefox.  Any advice on how to fix is greatly appreciated.
>
>
> - Joe
> _________________________________________________________________________
> 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 sagarmalam at gmail.com  Wed Oct 10 09:29:32 2018
From: sagarmalam at gmail.com (sagar malam)
Date: Wed, 10 Oct 2018 14:59:32 +0530
Subject: [Freeswitch-users] Shared presence not working with callcenter
 application of Freeswitch
In-Reply-To: <LNXP265MB1180E1DFD30C173B4DFC6759EBE70@LNXP265MB1180.GBRP265.PROD.OUTLOOK.COM>
References: <CAEZuF3vgN4OdaRha=edmCkhd_Ac-ZLThs6_uXx2LT3FGdVgfdQ@mail.gmail.com>
 <LNXP265MB1180E1DFD30C173B4DFC6759EBE70@LNXP265MB1180.GBRP265.PROD.OUTLOOK.COM>
Message-ID: <CAEZuF3tbp1tPH9Bt7jDqiVeUAzBToBsD3Ah9mNS2RF+o7B+C9A@mail.gmail.com>

I have already tried that setting along with sip_invite_domain.But without
any success.Further when i enable SLA and Presence debug , i dont see any
SLA / presence logs which appears in case of one to one calls.

On Wed, Oct 10, 2018 at 4:54 AM Shaun Stokes <
shaun.stokes at itec-support.co.uk> wrote:

> You need to set the presence_id variable of your agent when you bridge to
> the agent in the agent contact field.
>
> Shaun
>
> Get Outlook for iOS <https://aka.ms/o0ukef>
> ------------------------------
> *From:* 20110170300n behalf of
> *Sent:* Tuesday, October 9, 2018 19:48
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Shared presence not working with callcenter
> application of Freeswitch
>
> I am facing issue related to presence/ shared presence for agents in
> callcenter(mod_callcenter).
> Suppose we are dialling a user 1001 at example.com and it is a shared user
> in two phones. Shared presence works perfectly if we directly dial
> extension 1001.
>
> But if same extension is configured as agent of callcenter and call is
> bridged to agent(extension) through callcenter, shared presence does not
> work. Freeswitch does not generate notify for agent state change.
>
> Is this expected behaviour ? If not then what can fix this ?
>
> Thanks in advance
> _________________________________________________________________________
> 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 steveayre at gmail.com  Wed Oct 10 12:03:53 2018
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 10 Oct 2018 13:03:53 +0100
Subject: [Freeswitch-users] Sometimes "hanging" channels
In-Reply-To: <87D934A1-4A37-4709-93B5-96FA0659ABBE@gmx.net>
References: <87D934A1-4A37-4709-93B5-96FA0659ABBE@gmx.net>
Message-ID: <CAFiqYun3i3NDvqR4pkvfDCquQNMALXXAA-fp2EPm6bXx0VuVmg@mail.gmail.com>

If you take a gcore it'll generate a core dump. You could then dig into it
with gdb to find the thread for that channel and see what lock it's waiting
on.
However the gcore will pause the process for a while, so it will impact on
any other calls on that box.

Have you tried reproducing it on 1.8?

On Tue, 9 Oct 2018 at 20:36, Markus Bönke <mbodbg at gmx.net> wrote:

> Hello All,
>
> we are running freeswitch 1.6.20, calls are controlled via ESL, CDRs are
> written with mod_xml_cdrl. Sometimes we see „hanging“ channels. In such a
> case the CDR via mod_xml_cdr is written and the last log entry for such a
> call is "Locked, Waiting on external entities“.
> ...
> freeswitch.log.2018-10-09-10-19-07.1:db2908e3-c06b-4293-88b1-e18cffb66263
> 2018-10-09 10:18:15.197345 [DEBUG] switch_core_state_machine.c:610 (
> sofia/internal/anonymous at anonymous.invalid) State Change CS_REPORTING ->
> CS_DESTROY
> freeswitch.log.2018-10-09-10-19-07.1:db2908e3-c06b-4293-88b1-e18cffb66263
> 2018-10-09 10:18:15.197345 [DEBUG] switch_core_session.c:1665 Session
> 166111 (sofia/internal/anonymous at anonymous.invalid) Locked, Waiting on
> external entities
> ...
> How can I proceed to further analyze the problem ? In the last log line I
> also see the session number (Session 166111) - is there a way to find out
> on which external entity it is waiting?
>
> Thanks and regards
>
> Markus
> _________________________________________________________________________
> 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 wsimon at stratusvideo.com  Wed Oct 10 13:04:07 2018
From: wsimon at stratusvideo.com (William Simon)
Date: Wed, 10 Oct 2018 13:04:07 +0000
Subject: [Freeswitch-users] Accepting an "optional" SRTP offer (crypto
 in RTP/AVP) and establishing SRTP
In-Reply-To: <2AE1EF96-BD07-4E78-A638-D9D53FF8FF1E@stratusvideo.com>
References: <2AE1EF96-BD07-4E78-A638-D9D53FF8FF1E@stratusvideo.com>
Message-ID: <5821E934-2105-4B27-A925-74605B34B769@stratusvideo.com>

Can anyone offer insight into this matter?

Endpoint offers RTP/AVP with crypto. We want Freeswitch to respond to the RTP/AVP and agree to the crypto and do SRTP.


> On Oct 5, 2018, at 10:27 AM, William Simon <wsimon at stratusvideo.com> wrote:
>
> We are working with an endpoint that offers optional SRTP in the non-RFC-compliant way of an a=crypto attribute within RTP/AVP.
>
> I have told FreeSWITCH to allow this with rtp_allow_crypto_in_avp=true at the right place in the dialplan.
>
> Immediately after that in the dialplan I have to reject SRTP by using rtp_secure_media=forbidden, otherwise the call setup still fails. By setting the value to forbidden, the call does proceed unencrypted.
>
> We want FreeSWITCH to proceed with media encryption. Setting rtp_secure_media to any other value results in FreeSWITCH rejecting the offer like this:
>
> v=0
> o=FreeSWITCH 1538660754 1538660755 IN IP4 192.168.100.104
> s=FreeSWITCH
> c=IN IP4 192.168.100.104
> t=0 0
> m=audio 0 RTP/AVP 19
> m=video 0 RTP/AVP 19
>
> Is there anything else I can do to force SRTP in the answer?
>



“The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”

From mbodbg at gmx.net  Wed Oct 10 13:54:10 2018
From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=)
Date: Wed, 10 Oct 2018 15:54:10 +0200
Subject: [Freeswitch-users] Sometimes "hanging" channels
In-Reply-To: <CAFiqYun3i3NDvqR4pkvfDCquQNMALXXAA-fp2EPm6bXx0VuVmg@mail.gmail.com>
References: <87D934A1-4A37-4709-93B5-96FA0659ABBE@gmx.net>
 <CAFiqYun3i3NDvqR4pkvfDCquQNMALXXAA-fp2EPm6bXx0VuVmg@mail.gmail.com>
Message-ID: <1E3CBAFE-0B7B-445C-AAF2-EF4C074392B7@gmx.net>

So far I found out that the channel was kept open, because our ESL application never received a CHANNEL_HANGUP and CHANNEL_HANGUP_COMPLETE event. I’m running an ESL trace with tshark now. If it happens again I can see if those events are really not send by FS or they get lost in our app. If it turns out that FS sometimes is not sending the events we will upgrade to 1.8.2.

Thanks and regards

Markus


> Am 10.10.2018 um 14:03 schrieb Steven Ayre <steveayre at gmail.com>:
> 
> If you take a gcore it'll generate a core dump. You could then dig into it with gdb to find the thread for that channel and see what lock it's waiting on.
> However the gcore will pause the process for a while, so it will impact on any other calls on that box.
> 
> Have you tried reproducing it on 1.8?
> 
> On Tue, 9 Oct 2018 at 20:36, Markus Bönke <mbodbg at gmx.net <mailto:mbodbg at gmx.net>> wrote:
> Hello All,
> 
> we are running freeswitch 1.6.20, calls are controlled via ESL, CDRs are written with mod_xml_cdrl. Sometimes we see „hanging“ channels. In such a case the CDR via mod_xml_cdr is written and the last log entry for such a call is "Locked, Waiting on external entities“.
> ...
> freeswitch.log.2018-10-09-10-19-07.1:db2908e3-c06b-4293-88b1-e18cffb66263 2018-10-09 10:18:15.197345 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/anonymous at anonymous.invalid <mailto:sofia/internal/anonymous at anonymous.invalid>) State Change CS_REPORTING -> CS_DESTROY
> freeswitch.log.2018-10-09-10-19-07.1:db2908e3-c06b-4293-88b1-e18cffb66263 2018-10-09 10:18:15.197345 [DEBUG] switch_core_session.c:1665 Session 166111 (sofia/internal/anonymous at anonymous.invalid <mailto:sofia/internal/anonymous at anonymous.invalid>) Locked, Waiting on external entities
> ...
> How can I proceed to further analyze the problem ? In the last log line I also see the session number (Session 166111) - is there a way to find out on which external entity it is waiting?
> 
> Thanks and regards
> 
> Markus 
> _________________________________________________________________________
> 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/>_________________________________________________________________________
> 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
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> https://freeswitch.com

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From andrew.keil at visytel.com  Wed Oct 10 22:53:13 2018
From: andrew.keil at visytel.com (Andrew Keil)
Date: Wed, 10 Oct 2018 22:53:13 +0000
Subject: [Freeswitch-users] Re- DTMF 2833 issue
Message-ID: <ME1PR01MB1538D19D7BEB2A2C925A4983F3E00@ME1PR01MB1538.ausprd01.prod.outlook.com>

To FreeSWITCH Users,

I have just noticed when making calls internationally to the UK from Australia that the following SDP sometimes happens when the call lands on FreeSWITCH:

d5f14e47-2156-4236-be16-4eef1b7e9d3d 2018-10-04 06:59:28.819816 [DEBUG] sofia.c:7301 Remote SDP:
d5f14e47-2156-4236-be16-4eef1b7e9d3d v=0
d5f14e47-2156-4236-be16-4eef1b7e9d3d o=sbc-uk-mr-dh05a 451499 139728 IN IP4 XXX.XXX.XXX.XXX
d5f14e47-2156-4236-be16-4eef1b7e9d3d s=sip call
d5f14e47-2156-4236-be16-4eef1b7e9d3d c=IN IP4 XXX.XXX.XXX.XXX
d5f14e47-2156-4236-be16-4eef1b7e9d3d t=0 0
d5f14e47-2156-4236-be16-4eef1b7e9d3d m=audio 62754 RTP/AVP 8
d5f14e47-2156-4236-be16-4eef1b7e9d3d a=rtpmap:8 PCMA/8000
d5f14e47-2156-4236-be16-4eef1b7e9d3d a=ptime:20

This then creates the following message around call answer:

d5f14e47-2156-4236-be16-4eef1b7e9d3d 2018-10-04 06:59:28.839799 [DEBUG] switch_core_media.c:5766 No 2833 in SDP. Liberal DTMF mode adding 101 as telephone-event.

>From this point on DTMF is NOT detected by the FreeSWITCH IVR service.  I assume I would then need to use mod_dptools: start_dtmf to detect the DTMF tones within the channel.  Which I am yet to test.

However my main question is whether there is a channel variable that would enable me to check whether 2833 support is not inside the SDP?

Regards,

Andrew Keil


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From alex at freeswitch.com  Thu Oct 11 01:56:53 2018
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Thu, 11 Oct 2018 10:56:53 +0900
Subject: [Freeswitch-users] Accepting an "optional" SRTP offer (crypto
 in RTP/AVP) and establishing SRTP
In-Reply-To: <5821E934-2105-4B27-A925-74605B34B769@stratusvideo.com>
References: <2AE1EF96-BD07-4E78-A638-D9D53FF8FF1E@stratusvideo.com>
 <5821E934-2105-4B27-A925-74605B34B769@stratusvideo.com>
Message-ID: <CAGJLwFjP_6vUsqMPYdme9SZB-6YE+yUwjx4Gw8B02eFD1kdS-g@mail.gmail.com>

Take a close look to default.xml of vanilla dialplan. There are some
examples of SDP parsing here, you can use them in condition to detect your
special case. To enforce SRTP you just need to set rtp_secure_media. Don't
forget to reread documentation on the last one:

https://freeswitch.org/confluence/display/FREESWITCH/rtp_secure_media

Alex

On Thu, Oct 11, 2018 at 12:54 AM William Simon <wsimon at stratusvideo.com>
wrote:

> Can anyone offer insight into this matter?
>
> Endpoint offers RTP/AVP with crypto. We want Freeswitch to respond to the
> RTP/AVP and agree to the crypto and do SRTP.
>
>
> > On Oct 5, 2018, at 10:27 AM, William Simon <wsimon at stratusvideo.com>
> wrote:
> >
> > We are working with an endpoint that offers optional SRTP in the
> non-RFC-compliant way of an a=crypto attribute within RTP/AVP.
> >
> > I have told FreeSWITCH to allow this with rtp_allow_crypto_in_avp=true
> at the right place in the dialplan.
> >
> > Immediately after that in the dialplan I have to reject SRTP by using
> rtp_secure_media=forbidden, otherwise the call setup still fails. By
> setting the value to forbidden, the call does proceed unencrypted.
> >
> > We want FreeSWITCH to proceed with media encryption. Setting
> rtp_secure_media to any other value results in FreeSWITCH rejecting the
> offer like this:
> >
> > v=0
> > o=FreeSWITCH 1538660754 1538660755 IN IP4 192.168.100.104
> > s=FreeSWITCH
> > c=IN IP4 192.168.100.104
> > t=0 0
> > m=audio 0 RTP/AVP 19
> > m=video 0 RTP/AVP 19
> >
> > Is there anything else I can do to force SRTP in the answer?
> >
>
>
>
> “The information transmitted is intended only for the person or entity to
> which it is addressed and may contain proprietary, business-confidential
> and/or privileged material. If you are not the intended recipient of this
> message you are hereby notified that any use, review, retransmission,
> dissemination, distribution, reproduction or any action taken in reliance
> upon this message is prohibited. If you received this in error, please
> contact the sender and delete the material from any computer.”
> _________________________________________________________________________
> 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



-- 
Alex Sibyakin | Support Engineer
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: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.
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From julien.terrasson at gmail.com  Fri Oct 12 10:37:51 2018
From: julien.terrasson at gmail.com (Julien Terrasson)
Date: Fri, 12 Oct 2018 12:37:51 +0200
Subject: [Freeswitch-users] Freeswitch 1.8.2 : Debian Package : sometimes
 crash when executing nolocal:execute_on_answer lua script.
Message-ID: <CAFi=dz9cjZnfEM5vFzX1+NGYxUstp_f7s05EW5CEpipRsy-i8A@mail.gmail.com>

I'm trying to use freeswitch as a transcription gateway :

The goal is to allow a PSTN subscriber (A_party) to call another (B_party)
through freeswitch : freeswitch being used to record and transcript the
call.

This is the scenario i'm trying to acheive :

1/ A_party call freeswitch through a PSTN gateway.

2/ freeswitch execute a lua script (IVRPrompt8.lua) that :
* query the service database
* Initialize few sessions variables
* Play A_party a vocal prompt.
* Attempt to bridge B_party.

3/ When B_party answer, a lua script is called : catchBAnswer2.lua
This script is used to initialise several session variable and play legal
warning to both party.

<extension name="experimental_dialplan">
      <condition field="destination_number" expression="^(\+33XXXXXXXXX)$">

<!-- Play IVR to get the destination number and service parameters -->
<action inline="true" application="lua" data="IVRPrompt8.lua" />

<!-- Set channel record settings -->
<action application="export" data="RECORD_READ_ONLY=true"/>
<action application="export" data="RECORD_STEREO=false"/>

<!-- Free the "#" terminator from other use-->
<action application="set" data="ivr_menu_terminator=none"/>

<action application="export" data="nolocal:execute_on_answer=lua
catchBAnswer2.lua"/>

<!-- Finally bridge the call -->
<action application="bridge"
data="sofia/gateway/ippi_33XXXXXXXXX/${destination_number}"/>

</condition>
    </extension>

Sometimes the scenario works just fine, but most of the time it crash when
calling catchBAnswer2.lua, as show the last console logs (DEBUG LEVEL):

2018-10-12 09:42:26.812449 [NOTICE] sofia.c:7304 Pre-Answer
sofia/external5090/0665199963!
2018-10-12 09:42:26.812449 [DEBUG] switch_channel.c:3482
(sofia/external5090/0665199963) Callstate Change RINGING -> EARLY
2018-10-12 09:42:26.812449 [DEBUG] sofia.c:7291 Channel
sofia/external5090/0665199963 entering state [ready][200]
2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:5478 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:5533 Audio Codec
Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:5394 Set
telephone-event payload to 101 at 8000
2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:3781 Set Codec
sofia/external5090/0665199963 PCMA/8000 20 ms 160 samples 64000 bits 1
channels
2018-10-12 09:42:26.812449 [DEBUG] switch_core_codec.c:111
sofia/external5090/0665199963 Original read codec set to PCMA:8
2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:5737 Set
telephone-event payload to 101 at 8000
2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:5795
sofia/external5090/0665199963 Set 2833 dtmf send payload to 101 recv
payload to 101
2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:8511 AUDIO RTP
[sofia/external5090/0665199963] 94.23.42.139 port 29498 -> 194.169.214.61
port 59684 codec: 8 ms: 20
2018-10-12 09:42:26.812449 [DEBUG] switch_rtp.c:4300 Starting timer [soft]
160 bytes per 20ms
2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:8815
sofia/external5090/0665199963 Set 2833 dtmf send payload to 101
2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:8822
sofia/external5090/0665199963 Set 2833 dtmf receive payload to 101
2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:8845
sofia/external5090/0665199963 Set rtp dtmf delay to 40
2018-10-12 09:42:26.812449 [NOTICE] sofia.c:8429 Channel
[sofia/external5090/0665199963] has been answered
EXECUTE sofia/external5090/0665199963 lua(catchBAnswer2.lua)

** CORE DUMP **

I have no clues why this is happening, so i included the backtrack enclosed
to have feedbacks (https://freeswitch.org/jira/browse/FS-11456).

Can somebody have a look and give a more precise idea on what is making
freeswitch to crash ?

I can then setup monitoring and collect additional traces if needed (SIP,
RTP, etc..).

J. Terrasson
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From marquore at gmail.com  Wed Oct 10 19:35:50 2018
From: marquore at gmail.com (Marc Quore)
Date: Wed, 10 Oct 2018 22:35:50 +0300
Subject: [Freeswitch-users] [Missing local host] in proxy media mode
Message-ID: <CAMBkyc1ETujbQLgYEyv+-URcV2OwCuWW0Dh2k+b6U2XjUkAMig@mail.gmail.com>

Hi, All

I've got FreeSwitch 1.8.2 running in media proxy mode:

<param name="inbound-proxy-media" value="true"/>
<param name="inbound-late-negotiation" value="true"/>

Dialplan is configured to bridge incoming calls to remote gateway and most
of calls completing successfully, however some of them fail with
INCOMPATIBLE_DESTINATION cause. For failed calls everything goes normally
until bridged connection gets early media from callee:

[NOTICE] sofia.c:7304 Pre-Answer sofia/external/2222 at 2.2.2.2!
[INFO] switch_ivr_originate.c:3747 Sending early media
[DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PCMU/0
0 ms 960 samples 384000 bits 1 channels
[DEBUG] switch_core_codec.c:111 sofia/internal/1111 at 1.1.1.1 Original read
codec set to PCMU:0
[DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
1111 at 1.1.1.1] 1.1.1.1:18792->1.1.1.1:18792 codec: 0 ms: 20
[ERR] switch_core_media.c:9510 AUDIO RTP REPORTS ERROR: *[Missing local
host]*
[NOTICE] switch_core_media.c:9511 Hangup sofia/internal/1111 at 1.1.1.1
[CS_EXECUTE] [INCOMPATIBLE_DESTINATION]

Here's a dump for this call:

[image: image.png]

Healthy call on the same machine:

[DEBUG] switch_core_media.c:3781 Set Codec sofia/vpn/2222 at 2.2.2.2 PROXY/0 0
ms 160 samples 0 bits 1 channels
[DEBUG] switch_core_codec.c:111 sofia/internal/2222 at 2.2.2.2 Original read
codec set to PROXY:0
[DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
2222 at 2.2.2.2] 2.2.2.2:60818->2.2.2.2:60818 codec: 0 ms: 20

[DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PROXY/0
0 ms 160 samples 0 bits 1 channels
[DEBUG] switch_core_codec.c:111 sofia/external/1111 at 1.1.1.1 Original read
codec set to PROXY:0
[DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/external/
1111 at 1.1.1.1] 1.1.1.1:31412->1.1.1.1:31412 codec: 0 ms: 20

So it looks like proxified media connection cannot be established for some
calls. It shouldn't be a codec problem as call parties are configured with
common codecs, FS is bond to local interface which is running with no
errors. But I have no idea what "local host is missing" in my case so I'll
be appreciated for any hint.

FreeSWITCH Version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git a98a958
2018-09-26 17:55:25Z 64bit)
CentOS Linux release 7.5.1804 (Core)

Best,
Marc
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From mickael at winlux.fr  Thu Oct 11 15:32:33 2018
From: mickael at winlux.fr (Mickael Hubert)
Date: Thu, 11 Oct 2018 17:32:33 +0200
Subject: [Freeswitch-users] Multi gateways in profile
Message-ID: <CA+fRVSUkFuJRgkUAQCxr=3p6fj-joQR_yQD=c=YD4WRu8r=P-g@mail.gmail.com>

Hi guys,
I want to identify source of inbounds calls.
In my external profile I have two gateways:

- One with REGISTER method
- One without REGISTER method (IP to IP), only iptables filtering

<gateways>
      <gateway name="GW1">
        <param name="from-domain" value="**********"/>
        <param name="caller-id-in-from" value="true"/>
        <param name="proxy" value="1.1.1.1"/>
        <param name="register" value="true"/>
        <param name="username" value="**************"/>
        <param name="password" value="****************"/>
        <param name="expire-seconds" value="3600"/>
        <param name="retry-seconds" value="30"/>
        <param name="ping" value="25"/>
      </gateway>
      <gateway name="GW2">
        <param name="from-domain" value="*********"/>
        <param name="caller-id-in-from" value="true"/>
        <param name="proxy" value="2.2.2.2"/>
        <param name="register" value="false"/>
        <param name="retry-seconds" value="30"/>
        <param name="ping" value="25"/>
      </gateway>
    </gateways>

In my dialplan I have this conf:

<condition field="${sip_gateway}"  expression="GW1" break="never">
   <action application="set" data="number_dest=${destination_number}" />
   <action application="execute_extension" data="to-private XML from-gw1"/>
</condition>
<condition field="${sip_gateway}"  expression="GW2" break="never">
   <action application="set" data="number_dest=${destination_number}" />
   <action application="execute_extension" data="to-private XML from-gw2"/>
</condition>

When a call arrives throught GW1 (REGISTERED), ${sip_gateway} is populate
with "GW1" perfect !. But when another call arrives throught GW2 (not
REGISTERED), ${sip_gateway} is empty.

Please, how can I identify the source gateway where calls arrive ? maybe
it's not a good way to do that ?

thanks in advance
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From mike at jerris.com  Fri Oct 12 21:53:34 2018
From: mike at jerris.com (Michael Jerris)
Date: Fri, 12 Oct 2018 17:53:34 -0400
Subject: [Freeswitch-users] mod_spy,
 does it only work for registered users?
In-Reply-To: <E1g8YW5-0004Dy-KP@mail.opensips.org>
References: <E1g8YW5-0004Dy-KP@mail.opensips.org>
Message-ID: <A4270737-380F-4E7B-9EA7-B252AEB1619F@jerris.com>

After talking to you about this, I just took a look. No its not based on registered users, but authed users will probably more reliably set the right variables to figure out the right user to match to.  Trick is to have the vars set to be able to look them up.

    username[0] = switch_event_get_header(event, "Caller-Username");                                                                                           
    domain[0] = switch_event_get_header(event, "variable_domain_name");                                                                                        
                                                                                                                                                                
    username[1] = switch_event_get_header(event, "variable_dialed_user");                                                                                      
    domain[1] = switch_event_get_header(event, "variable_dialed_domain");                                                                                      
                                                                                                                                                                
    username[2] = switch_event_get_header(event, "variable_user_name");                                                                                        
    domain[2] = switch_event_get_header(event, "variable_domain_name");                                                                                        
                                                                                                                                                                
    username[3] = switch_event_get_header(event, "variable_sip_to_user");                                                                                      
    domain[3] = switch_event_get_header(event, "variable_domain_name");                                                                                        
                                                                                                                                                                
    username[4] = switch_event_get_header(event, "variable_verto_user");                                                                                       
    domain[4] = switch_event_get_header(event, "variable_verto_host");                                                                                         

It has to match one of those pairs.


> On Oct 5, 2018, at 6:24 PM, Alex Goulis <agoulis at opensips.org> wrote:
> 
> Hi all...
> 
> Does the userspy application only target registered users or can you use it to spy calls to let's say calls to a DID?
> 
> 
> _________________________________________________________________________
> 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 mike at jerris.com  Fri Oct 12 21:56:37 2018
From: mike at jerris.com (Michael Jerris)
Date: Fri, 12 Oct 2018 17:56:37 -0400
Subject: [Freeswitch-users] Shared presence not working with callcenter
 application of Freeswitch
In-Reply-To: <CAEZuF3tbp1tPH9Bt7jDqiVeUAzBToBsD3Ah9mNS2RF+o7B+C9A@mail.gmail.com>
References: <CAEZuF3vgN4OdaRha=edmCkhd_Ac-ZLThs6_uXx2LT3FGdVgfdQ@mail.gmail.com>
 <LNXP265MB1180E1DFD30C173B4DFC6759EBE70@LNXP265MB1180.GBRP265.PROD.OUTLOOK.COM>
 <CAEZuF3tbp1tPH9Bt7jDqiVeUAzBToBsD3Ah9mNS2RF+o7B+C9A@mail.gmail.com>
Message-ID: <6ABFDA54-2016-495F-BB0D-DCFB55874429@jerris.com>

Is manage presense enabled?

> On Oct 10, 2018, at 5:29 AM, sagar malam <sagarmalam at gmail.com> wrote:
> 
> I have already tried that setting along with sip_invite_domain.But without any success.Further when i enable SLA and Presence debug , i dont see any SLA / presence logs which appears in case of one to one calls.
> 
> On Wed, Oct 10, 2018 at 4:54 AM Shaun Stokes <shaun.stokes at itec-support.co.uk <mailto:shaun.stokes at itec-support.co.uk>> wrote:
> You need to set the presence_id variable of your agent when you bridge to the agent in the agent contact field.
> 
> Shaun
> 
> Get Outlook for iOS <https://aka.ms/o0ukef>
> From: 20110170300n behalf of 
> Sent: Tuesday, October 9, 2018 19:48
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] Shared presence not working with callcenter application of Freeswitch
>  
> I am facing issue related to presence/ shared presence for agents in callcenter(mod_callcenter).
> Suppose we are dialling a user 1001 at example.com <mailto:1001 at example.com> and it is a shared user in two phones. Shared presence works perfectly if we directly dial extension 1001.
> 
> But if same extension is configured as agent of callcenter and call is bridged to agent(extension) through callcenter, shared presence does not work. Freeswitch does not generate notify for agent state change. 
> 
> Is this expected behaviour ? If not then what can fix this ? 
> 
> Thanks in advance

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From mike at jerris.com  Fri Oct 12 21:59:06 2018
From: mike at jerris.com (Michael Jerris)
Date: Fri, 12 Oct 2018 17:59:06 -0400
Subject: [Freeswitch-users] Re- DTMF 2833 issue
In-Reply-To: <ME1PR01MB1538D19D7BEB2A2C925A4983F3E00@ME1PR01MB1538.ausprd01.prod.outlook.com>
References: <ME1PR01MB1538D19D7BEB2A2C925A4983F3E00@ME1PR01MB1538.ausprd01.prod.outlook.com>
Message-ID: <1BC124F9-C601-43C2-92D0-BF68194198BB@jerris.com>

There are vars with the full SDP, you can do a regex condition.  Check the info app to see all the vars you have at your disposal.  There may be a var for the te as well to show the pt.

Mike

> On Oct 10, 2018, at 6:53 PM, Andrew Keil <andrew.keil at visytel.com> wrote:
> 
> To FreeSWITCH Users,
>  
> I have just noticed when making calls internationally to the UK from Australia that the following SDP sometimes happens when the call lands on FreeSWITCH:
>  
> d5f14e47-2156-4236-be16-4eef1b7e9d3d 2018-10-04 06:59:28.819816 [DEBUG] sofia.c:7301 Remote SDP:
> d5f14e47-2156-4236-be16-4eef1b7e9d3d v=0
> d5f14e47-2156-4236-be16-4eef1b7e9d3d o=sbc-uk-mr-dh05a 451499 139728 IN IP4 XXX.XXX.XXX.XXX
> d5f14e47-2156-4236-be16-4eef1b7e9d3d s=sip call
> d5f14e47-2156-4236-be16-4eef1b7e9d3d c=IN IP4 XXX.XXX.XXX.XXX
> d5f14e47-2156-4236-be16-4eef1b7e9d3d t=0 0
> d5f14e47-2156-4236-be16-4eef1b7e9d3d m=audio 62754 RTP/AVP 8
> d5f14e47-2156-4236-be16-4eef1b7e9d3d a=rtpmap:8 PCMA/8000
> d5f14e47-2156-4236-be16-4eef1b7e9d3d a=ptime:20
>  
> This then creates the following message around call answer: 
>  
> d5f14e47-2156-4236-be16-4eef1b7e9d3d 2018-10-04 06:59:28.839799 [DEBUG] switch_core_media.c:5766 No 2833 in SDP. Liberal DTMF mode adding 101 as telephone-event.
>  
> From this point on DTMF is NOT detected by the FreeSWITCH IVR service.  I assume I would then need to use mod_dptools: start_dtmf to detect the DTMF tones within the channel.  Which I am yet to test.
>  
> However my main question is whether there is a channel variable that would enable me to check whether 2833 support is not inside the SDP?
>  
> Regards,
>  
> Andrew Keil 
>  

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From mike at jerris.com  Fri Oct 12 22:00:04 2018
From: mike at jerris.com (Michael Jerris)
Date: Fri, 12 Oct 2018 18:00:04 -0400
Subject: [Freeswitch-users] Accepting an "optional" SRTP offer (crypto
 in RTP/AVP) and establishing SRTP
In-Reply-To: <2AE1EF96-BD07-4E78-A638-D9D53FF8FF1E@stratusvideo.com>
References: <2AE1EF96-BD07-4E78-A638-D9D53FF8FF1E@stratusvideo.com>
Message-ID: <C85C798A-BE0B-4EE2-8300-6051CE2103FC@jerris.com>

Are you trying this on 1.8.2 version?

> On Oct 5, 2018, at 10:27 AM, William Simon <wsimon at stratusvideo.com> wrote:
> 
> We are working with an endpoint that offers optional SRTP in the non-RFC-compliant way of an a=crypto attribute within RTP/AVP.
> 
> I have told FreeSWITCH to allow this with rtp_allow_crypto_in_avp=true at the right place in the dialplan.
> 
> Immediately after that in the dialplan I have to reject SRTP by using rtp_secure_media=forbidden, otherwise the call setup still fails. By setting the value to forbidden, the call does proceed unencrypted.
> 
> We want FreeSWITCH to proceed with media encryption. Setting rtp_secure_media to any other value results in FreeSWITCH rejecting the offer like this:
> 
> v=0
> o=FreeSWITCH 1538660754 1538660755 IN IP4 192.168.100.104
> s=FreeSWITCH
> c=IN IP4 192.168.100.104
> t=0 0
> m=audio 0 RTP/AVP 19
> m=video 0 RTP/AVP 19
> 
> Is there anything else I can do to force SRTP in the answer?
> 
> 



From mike at jerris.com  Fri Oct 12 22:01:20 2018
From: mike at jerris.com (Michael Jerris)
Date: Fri, 12 Oct 2018 18:01:20 -0400
Subject: [Freeswitch-users] Freeswitch 1.8.2 : Debian Package :
 sometimes crash when executing nolocal:execute_on_answer lua script.
In-Reply-To: <CAFi=dz9cjZnfEM5vFzX1+NGYxUstp_f7s05EW5CEpipRsy-i8A@mail.gmail.com>
References: <CAFi=dz9cjZnfEM5vFzX1+NGYxUstp_f7s05EW5CEpipRsy-i8A@mail.gmail.com>
Message-ID: <162C6063-F6EC-4897-9BEA-A419702A013B@jerris.com>

In order to look at this we can’t look at the core file directly, you’ll need to generate a backtrace from it so we can see.

https://freeswitch.org/confluence/display/FREESWITCH/Debugging#Debugging-GettingaBacktrace <https://freeswitch.org/confluence/display/FREESWITCH/Debugging#Debugging-GettingaBacktrace> 

Please update the bug when you have this info.


> On Oct 12, 2018, at 6:37 AM, Julien Terrasson <julien.terrasson at gmail.com> wrote:
> 
> I'm trying to use freeswitch as a transcription gateway : 
> 
> The goal is to allow a PSTN subscriber (A_party) to call another (B_party) through freeswitch : freeswitch being used to record and transcript the call. 
> 
> This is the scenario i'm trying to acheive : 
> 
> 1/ A_party call freeswitch through a PSTN gateway. 
> 
> 2/ freeswitch execute a lua script (IVRPrompt8.lua) that : 
> * query the service database 
> * Initialize few sessions variables 
> * Play A_party a vocal prompt. 
> * Attempt to bridge B_party. 
> 
> 3/ When B_party answer, a lua script is called : catchBAnswer2.lua 
> This script is used to initialise several session variable and play legal warning to both party. 
> 
> <extension name="experimental_dialplan"> 
>       <condition field="destination_number" expression="^(\+33XXXXXXXXX)$"> 
> 
> <!-- Play IVR to get the destination number and service parameters --> 
> <action inline="true" application="lua" data="IVRPrompt8.lua" /> 
> 
> <!-- Set channel record settings --> 
> <action application="export" data="RECORD_READ_ONLY=true"/> 
> <action application="export" data="RECORD_STEREO=false"/> 
> 
> <!-- Free the "#" terminator from other use--> 
> <action application="set" data="ivr_menu_terminator=none"/> 
> 
> <action application="export" data="nolocal:execute_on_answer=lua catchBAnswer2.lua"/> 
> 
> <!-- Finally bridge the call --> 
> <action application="bridge" data="sofia/gateway/ippi_33XXXXXXXXX/${destination_number}"/> 
> 
> </condition> 
>     </extension> 
> 
> Sometimes the scenario works just fine, but most of the time it crash when calling catchBAnswer2.lua, as show the last console logs (DEBUG LEVEL): 
> 
> 2018-10-12 09:42:26.812449 [NOTICE] sofia.c:7304 Pre-Answer sofia/external5090/0665199963! 
> 2018-10-12 09:42:26.812449 [DEBUG] switch_channel.c:3482 (sofia/external5090/0665199963) Callstate Change RINGING -> EARLY 
> 2018-10-12 09:42:26.812449 [DEBUG] sofia.c:7291 Channel sofia/external5090/0665199963 entering state [ready][200] 
> 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:5478 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 
> 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:5533 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 
> 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:5394 Set telephone-event payload to 101 at 8000 <mailto:101 at 8000> 
> 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:3781 Set Codec sofia/external5090/0665199963 PCMA/8000 20 ms 160 samples 64000 bits 1 channels 
> 2018-10-12 09:42:26.812449 [DEBUG] switch_core_codec.c:111 sofia/external5090/0665199963 Original read codec set to PCMA:8 
> 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:5737 Set telephone-event payload to 101 at 8000 <mailto:101 at 8000> 
> 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:5795 sofia/external5090/0665199963 Set 2833 dtmf send payload to 101 recv payload to 101 
> 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:8511 AUDIO RTP [sofia/external5090/0665199963] 94.23.42.139 port 29498 -> 194.169.214.61 port 59684 codec: 8 ms: 20 
> 2018-10-12 09:42:26.812449 [DEBUG] switch_rtp.c:4300 Starting timer [soft] 160 bytes per 20ms 
> 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:8815 sofia/external5090/0665199963 Set 2833 dtmf send payload to 101 
> 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:8822 sofia/external5090/0665199963 Set 2833 dtmf receive payload to 101 
> 2018-10-12 09:42:26.812449 [DEBUG] switch_core_media.c:8845 sofia/external5090/0665199963 Set rtp dtmf delay to 40 
> 2018-10-12 09:42:26.812449 [NOTICE] sofia.c:8429 Channel [sofia/external5090/0665199963] has been answered 
> EXECUTE sofia/external5090/0665199963 lua(catchBAnswer2.lua) 
> 
> ** CORE DUMP ** 
> 
> I have no clues why this is happening, so i included the backtrack enclosed to have feedbacks (https://freeswitch.org/jira/browse/FS-11456 <https://freeswitch.org/jira/browse/FS-11456>).
> 
> Can somebody have a look and give a more precise idea on what is making freeswitch to crash ? 
> 
> I can then setup monitoring and collect additional traces if needed (SIP, RTP, etc..).
> 
> J. Terrasson
> _________________________________________________________________________
> 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 darshanmody at avaya.com  Thu Oct 11 09:16:15 2018
From: darshanmody at avaya.com (Mody, Darshan (Darshan))
Date: Thu, 11 Oct 2018 09:16:15 +0000
Subject: [Freeswitch-users] Freeswitch is not handling 180 after 200 OK
Message-ID: <25D2EC755404B4409F263AC6D050FEBB2E6D6A6E@AZ-FFEXMB03.global.avaya.com>

Hi

We are running UDP traffic. During heavy load FS receives sometimes receives 180 response after 200 OK already received on the same call-id (dialog). However FS does not discard the same and starts processing 180 even though it has already received 200 OK on the same call-id (dialog).

Freeswitch should be discarding the 180 response after it gets 200 OK response from far-end.

In our case the FS core state machine even though it gets 200 OK waits for 200 Ok again.

Thanks
Darshan
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From brian at freeswitch.com  Fri Oct 12 22:07:51 2018
From: brian at freeswitch.com (Brian West)
Date: Fri, 12 Oct 2018 18:07:51 -0400
Subject: [Freeswitch-users] [Missing local host] in proxy media mode
In-Reply-To: <CAMBkyc1ETujbQLgYEyv+-URcV2OwCuWW0Dh2k+b6U2XjUkAMig@mail.gmail.com>
References: <CAMBkyc1ETujbQLgYEyv+-URcV2OwCuWW0Dh2k+b6U2XjUkAMig@mail.gmail.com>
Message-ID: <CAEJMVkB9sggz6y4RfPvWTYFuxOEpkvvM2Htmsnav8nfxRZzRog@mail.gmail.com>

Could you elaborate on why you're using Proxy Media for?
/b

On Fri, Oct 12, 2018 at 5:37 PM Marc Quore <marquore at gmail.com> wrote:

> Hi, All
>
> I've got FreeSwitch 1.8.2 running in media proxy mode:
>
> <param name="inbound-proxy-media" value="true"/>
> <param name="inbound-late-negotiation" value="true"/>
>
> Dialplan is configured to bridge incoming calls to remote gateway and most
> of calls completing successfully, however some of them fail with
> INCOMPATIBLE_DESTINATION cause. For failed calls everything goes normally
> until bridged connection gets early media from callee:
>
> [NOTICE] sofia.c:7304 Pre-Answer sofia/external/2222 at 2.2.2.2!
> [INFO] switch_ivr_originate.c:3747 Sending early media
> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PCMU/0
> 0 ms 960 samples 384000 bits 1 channels
> [DEBUG] switch_core_codec.c:111 sofia/internal/1111 at 1.1.1.1 Original read
> codec set to PCMU:0
> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
> 1111 at 1.1.1.1] 1.1.1.1:18792->1.1.1.1:18792 codec: 0 ms: 20
> [ERR] switch_core_media.c:9510 AUDIO RTP REPORTS ERROR: *[Missing local
> host]*
> [NOTICE] switch_core_media.c:9511 Hangup sofia/internal/1111 at 1.1.1.1
> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
>
> Here's a dump for this call:
>
> [image: image.png]
>
> Healthy call on the same machine:
>
> [DEBUG] switch_core_media.c:3781 Set Codec sofia/vpn/2222 at 2.2.2.2 PROXY/0
> 0 ms 160 samples 0 bits 1 channels
> [DEBUG] switch_core_codec.c:111 sofia/internal/2222 at 2.2.2.2 Original read
> codec set to PROXY:0
> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
> 2222 at 2.2.2.2] 2.2.2.2:60818->2.2.2.2:60818 codec: 0 ms: 20
>
> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PROXY/0
> 0 ms 160 samples 0 bits 1 channels
> [DEBUG] switch_core_codec.c:111 sofia/external/1111 at 1.1.1.1 Original read
> codec set to PROXY:0
> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/external/
> 1111 at 1.1.1.1] 1.1.1.1:31412->1.1.1.1:31412 codec: 0 ms: 20
>
> So it looks like proxified media connection cannot be established for some
> calls. It shouldn't be a codec problem as call parties are configured with
> common codecs, FS is bond to local interface which is running with no
> errors. But I have no idea what "local host is missing" in my case so I'll
> be appreciated for any hint.
>
> FreeSWITCH Version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git
> a98a958 2018-09-26 17:55:25Z 64bit)
> CentOS Linux release 7.5.1804 (Core)
>
> Best,
> Marc
> _________________________________________________________________________
> 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 brian at freeswitch.com  Fri Oct 12 22:08:40 2018
From: brian at freeswitch.com (Brian West)
Date: Fri, 12 Oct 2018 18:08:40 -0400
Subject: [Freeswitch-users] Multi gateways in profile
In-Reply-To: <CA+fRVSUkFuJRgkUAQCxr=3p6fj-joQR_yQD=c=YD4WRu8r=P-g@mail.gmail.com>
References: <CA+fRVSUkFuJRgkUAQCxr=3p6fj-joQR_yQD=c=YD4WRu8r=P-g@mail.gmail.com>
Message-ID: <CAEJMVkA4ErVRN3BaBzY25o8EXa=-VxBNT5RrdCCXkZR+96HKKA@mail.gmail.com>

You'll have to look closer, unless the remote calls you back at the
registered contact, or includes the params that it registered with there is
no way to match it.
/b


On Fri, Oct 12, 2018 at 5:38 PM Mickael Hubert <mickael at winlux.fr> wrote:

> Hi guys,
> I want to identify source of inbounds calls.
> In my external profile I have two gateways:
>
> - One with REGISTER method
> - One without REGISTER method (IP to IP), only iptables filtering
>
> <gateways>
>       <gateway name="GW1">
>         <param name="from-domain" value="**********"/>
>         <param name="caller-id-in-from" value="true"/>
>         <param name="proxy" value="1.1.1.1"/>
>         <param name="register" value="true"/>
>         <param name="username" value="**************"/>
>         <param name="password" value="****************"/>
>         <param name="expire-seconds" value="3600"/>
>         <param name="retry-seconds" value="30"/>
>         <param name="ping" value="25"/>
>       </gateway>
>       <gateway name="GW2">
>         <param name="from-domain" value="*********"/>
>         <param name="caller-id-in-from" value="true"/>
>         <param name="proxy" value="2.2.2.2"/>
>         <param name="register" value="false"/>
>         <param name="retry-seconds" value="30"/>
>         <param name="ping" value="25"/>
>       </gateway>
>     </gateways>
>
> In my dialplan I have this conf:
>
> <condition field="${sip_gateway}"  expression="GW1" break="never">
>    <action application="set" data="number_dest=${destination_number}" />
>    <action application="execute_extension" data="to-private XML from-gw1"/>
> </condition>
> <condition field="${sip_gateway}"  expression="GW2" break="never">
>    <action application="set" data="number_dest=${destination_number}" />
>    <action application="execute_extension" data="to-private XML
> from-gw2"/>
> </condition>
>
> When a call arrives throught GW1 (REGISTERED), ${sip_gateway} is populate
> with "GW1" perfect !. But when another call arrives throught GW2 (not
> REGISTERED), ${sip_gateway} is empty.
>
> Please, how can I identify the source gateway where calls arrive ? maybe
> it's not a good way to do that ?
>
> thanks in advance
> _________________________________________________________________________
> 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 agoulis at opensips.org  Fri Oct 12 23:40:30 2018
From: agoulis at opensips.org (Alex Goulis)
Date: Fri, 12 Oct 2018 17:40:30 -0600
Subject: [Freeswitch-users] mod_spy,
 does it only work for registered users?
In-Reply-To: <A4270737-380F-4E7B-9EA7-B252AEB1619F@jerris.com>
Message-ID: <E1gB72t-0001fn-QS@mail.opensips.org>

Man thanks for digging deeper... 
unfortunately I have an identical match to Pairs 0-3 in my tests.   Ie, they all have the same user/domain in them. And they all match the user/domain I set when invoking userspy.
The events after the invite show them populated as well. 
I'm pretty sure the eavesdrop trigger happens before I can manually override them with dialplan on the incoming call to be targeted.
Happy to test any scenario you can further think of though.
Alex


Sent via the Samsung Galaxy Note8, an AT&T 4G LTE smartphone
-------- Original message --------From: Michael Jerris <mike at jerris.com> Date: 10/12/18  3:53 PM  (GMT-07:00) To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> Subject: Re: [Freeswitch-users] mod_spy, does it only work for registered users? 
After talking to you about this, I just took a look. No its not based on registered users, but authed users will probably more reliably set the right variables to figure out the right user to match to.  Trick is to have the vars set to be able to look them up.
    username[0] = switch_event_get_header(event, "Caller-Username");                                                                                                domain[0] = switch_event_get_header(event, "variable_domain_name");                                                                                                                                                                                                                                                         
    username[1] = switch_event_get_header(event, "variable_dialed_user");                                                                                           domain[1] = switch_event_get_header(event, "variable_dialed_domain");                                                                                                                                                                                                                                                       
    username[2] = switch_event_get_header(event, "variable_user_name");                                                                                             domain[2] = switch_event_get_header(event, "variable_domain_name");                                                                                                                                                                                                                                                         
    username[3] = switch_event_get_header(event, "variable_sip_to_user");                                                                                           domain[3] = switch_event_get_header(event, "variable_domain_name");                                                                                                                                                                                                                                                         
    username[4] = switch_event_get_header(event, "variable_verto_user");                                                                                            domain[4] = switch_event_get_header(event, "variable_verto_host");                                                                                          
It has to match one of those pairs.

On Oct 5, 2018, at 6:24 PM, Alex Goulis <agoulis at opensips.org> wrote:
Hi all...
Does the userspy application only target registered users or can you use it to spy calls to let's say calls to a DID?

_________________________________________________________________________
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From marquore at gmail.com  Fri Oct 12 23:41:09 2018
From: marquore at gmail.com (Marc Quore)
Date: Sat, 13 Oct 2018 02:41:09 +0300
Subject: [Freeswitch-users] Multi gateways in profile
In-Reply-To: <CA+fRVSUkFuJRgkUAQCxr=3p6fj-joQR_yQD=c=YD4WRu8r=P-g@mail.gmail.com>
References: <CA+fRVSUkFuJRgkUAQCxr=3p6fj-joQR_yQD=c=YD4WRu8r=P-g@mail.gmail.com>
Message-ID: <CAMBkyc3tMojX-3cR49mj_ak-NmhEaNNB8H0BTf6TCd1+Nx-HAQ@mail.gmail.com>

 https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration

[image: image.png]
Hope this helps.


сб, 13 окт. 2018 г. в 1:43, Mickael Hubert <mickael at winlux.fr>:

> Hi guys,
> I want to identify source of inbounds calls.
> In my external profile I have two gateways:
>
> - One with REGISTER method
> - One without REGISTER method (IP to IP), only iptables filtering
>
> <gateways>
>       <gateway name="GW1">
>         <param name="from-domain" value="**********"/>
>         <param name="caller-id-in-from" value="true"/>
>         <param name="proxy" value="1.1.1.1"/>
>         <param name="register" value="true"/>
>         <param name="username" value="**************"/>
>         <param name="password" value="****************"/>
>         <param name="expire-seconds" value="3600"/>
>         <param name="retry-seconds" value="30"/>
>         <param name="ping" value="25"/>
>       </gateway>
>       <gateway name="GW2">
>         <param name="from-domain" value="*********"/>
>         <param name="caller-id-in-from" value="true"/>
>         <param name="proxy" value="2.2.2.2"/>
>         <param name="register" value="false"/>
>         <param name="retry-seconds" value="30"/>
>         <param name="ping" value="25"/>
>       </gateway>
>     </gateways>
>
> In my dialplan I have this conf:
>
> <condition field="${sip_gateway}"  expression="GW1" break="never">
>    <action application="set" data="number_dest=${destination_number}" />
>    <action application="execute_extension" data="to-private XML from-gw1"/>
> </condition>
> <condition field="${sip_gateway}"  expression="GW2" break="never">
>    <action application="set" data="number_dest=${destination_number}" />
>    <action application="execute_extension" data="to-private XML
> from-gw2"/>
> </condition>
>
> When a call arrives throught GW1 (REGISTERED), ${sip_gateway} is populate
> with "GW1" perfect !. But when another call arrives throught GW2 (not
> REGISTERED), ${sip_gateway} is empty.
>
> Please, how can I identify the source gateway where calls arrive ? maybe
> it's not a good way to do that ?
>
> thanks in advance
> _________________________________________________________________________
> 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
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From tculjaga at gmail.com  Sat Oct 13 11:46:24 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sat, 13 Oct 2018 13:46:24 +0200
Subject: [Freeswitch-users] Multi gateways in profile
In-Reply-To: <CA+fRVSUkFuJRgkUAQCxr=3p6fj-joQR_yQD=c=YD4WRu8r=P-g@mail.gmail.com>
References: <CA+fRVSUkFuJRgkUAQCxr=3p6fj-joQR_yQD=c=YD4WRu8r=P-g@mail.gmail.com>
Message-ID: <CAJoKKdWu6rsYAjgoQ7j9X7xZN4qtytCyS1EeqN=Gy=WUZGiyUg@mail.gmail.com>

it works only if the gateway does outbound registration (for GW1 in your
case). If you want track inbound calls for non registering gateways, then
you need to match them by remote IP.

On Sat, 13 Oct 2018 at 01:17, Mickael Hubert <mickael at winlux.fr> wrote:

> Hi guys,
> I want to identify source of inbounds calls.
> In my external profile I have two gateways:
>
> - One with REGISTER method
> - One without REGISTER method (IP to IP), only iptables filtering
>
> <gateways>
>       <gateway name="GW1">
>         <param name="from-domain" value="**********"/>
>         <param name="caller-id-in-from" value="true"/>
>         <param name="proxy" value="1.1.1.1"/>
>         <param name="register" value="true"/>
>         <param name="username" value="**************"/>
>         <param name="password" value="****************"/>
>         <param name="expire-seconds" value="3600"/>
>         <param name="retry-seconds" value="30"/>
>         <param name="ping" value="25"/>
>       </gateway>
>       <gateway name="GW2">
>         <param name="from-domain" value="*********"/>
>         <param name="caller-id-in-from" value="true"/>
>         <param name="proxy" value="2.2.2.2"/>
>         <param name="register" value="false"/>
>         <param name="retry-seconds" value="30"/>
>         <param name="ping" value="25"/>
>       </gateway>
>     </gateways>
>
> In my dialplan I have this conf:
>
> <condition field="${sip_gateway}"  expression="GW1" break="never">
>    <action application="set" data="number_dest=${destination_number}" />
>    <action application="execute_extension" data="to-private XML from-gw1"/>
> </condition>
> <condition field="${sip_gateway}"  expression="GW2" break="never">
>    <action application="set" data="number_dest=${destination_number}" />
>    <action application="execute_extension" data="to-private XML
> from-gw2"/>
> </condition>
>
> When a call arrives throught GW1 (REGISTERED), ${sip_gateway} is populate
> with "GW1" perfect !. But when another call arrives throught GW2 (not
> REGISTERED), ${sip_gateway} is empty.
>
> Please, how can I identify the source gateway where calls arrive ? maybe
> it's not a good way to do that ?
>
> thanks in advance
> _________________________________________________________________________
> 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 steveayre at gmail.com  Sat Oct 13 12:42:46 2018
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 13 Oct 2018 13:42:46 +0100
Subject: [Freeswitch-users] [Missing local host] in proxy media mode
In-Reply-To: <CAMBkyc1ETujbQLgYEyv+-URcV2OwCuWW0Dh2k+b6U2XjUkAMig@mail.gmail.com>
References: <CAMBkyc1ETujbQLgYEyv+-URcV2OwCuWW0Dh2k+b6U2XjUkAMig@mail.gmail.com>
Message-ID: <CAFiqYumUHTHcLD5Xa6LO_RyH3SJh3ey6vEk9ytSUqDoBbYVavA@mail.gmail.com>

Are you sure you need proxy media?

FS has three media modes - the default, bypass, and proxy. In proxy mode
you lose a lot of the FreeSWITCH functionality as you cannot interact with
the media. It's generally only necessary if you need to support a codec FS
does not support.

On Fri, 12 Oct 2018 at 23:32, Marc Quore <marquore at gmail.com> wrote:

> Hi, All
>
> I've got FreeSwitch 1.8.2 running in media proxy mode:
>
> <param name="inbound-proxy-media" value="true"/>
> <param name="inbound-late-negotiation" value="true"/>
>
> Dialplan is configured to bridge incoming calls to remote gateway and most
> of calls completing successfully, however some of them fail with
> INCOMPATIBLE_DESTINATION cause. For failed calls everything goes normally
> until bridged connection gets early media from callee:
>
> [NOTICE] sofia.c:7304 Pre-Answer sofia/external/2222 at 2.2.2.2!
> [INFO] switch_ivr_originate.c:3747 Sending early media
> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PCMU/0
> 0 ms 960 samples 384000 bits 1 channels
> [DEBUG] switch_core_codec.c:111 sofia/internal/1111 at 1.1.1.1 Original read
> codec set to PCMU:0
> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
> 1111 at 1.1.1.1] 1.1.1.1:18792->1.1.1.1:18792 codec: 0 ms: 20
> [ERR] switch_core_media.c:9510 AUDIO RTP REPORTS ERROR: *[Missing local
> host]*
> [NOTICE] switch_core_media.c:9511 Hangup sofia/internal/1111 at 1.1.1.1
> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
>
> Here's a dump for this call:
>
> [image: image.png]
>
> Healthy call on the same machine:
>
> [DEBUG] switch_core_media.c:3781 Set Codec sofia/vpn/2222 at 2.2.2.2 PROXY/0
> 0 ms 160 samples 0 bits 1 channels
> [DEBUG] switch_core_codec.c:111 sofia/internal/2222 at 2.2.2.2 Original read
> codec set to PROXY:0
> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
> 2222 at 2.2.2.2] 2.2.2.2:60818->2.2.2.2:60818 codec: 0 ms: 20
>
> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PROXY/0
> 0 ms 160 samples 0 bits 1 channels
> [DEBUG] switch_core_codec.c:111 sofia/external/1111 at 1.1.1.1 Original read
> codec set to PROXY:0
> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/external/
> 1111 at 1.1.1.1] 1.1.1.1:31412->1.1.1.1:31412 codec: 0 ms: 20
>
> So it looks like proxified media connection cannot be established for some
> calls. It shouldn't be a codec problem as call parties are configured with
> common codecs, FS is bond to local interface which is running with no
> errors. But I have no idea what "local host is missing" in my case so I'll
> be appreciated for any hint.
>
> FreeSWITCH Version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git
> a98a958 2018-09-26 17:55:25Z 64bit)
> CentOS Linux release 7.5.1804 (Core)
>
> Best,
> Marc
> _________________________________________________________________________
> 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 steveayre at gmail.com  Sat Oct 13 12:49:47 2018
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 13 Oct 2018 13:49:47 +0100
Subject: [Freeswitch-users] Re- DTMF 2833 issue
In-Reply-To: <1BC124F9-C601-43C2-92D0-BF68194198BB@jerris.com>
References: <ME1PR01MB1538D19D7BEB2A2C925A4983F3E00@ME1PR01MB1538.ausprd01.prod.outlook.com>
 <1BC124F9-C601-43C2-92D0-BF68194198BB@jerris.com>
Message-ID: <CAFiqYunKvWbiWOdyGiW_yeLLbSWGKLBM6DXNDBQuv8HuBjK-TQ@mail.gmail.com>

Try something like this:

  <extension name="inband-dtmf" continue="true">
    <condition field="${switch_r_sdp}" expression="a=rtpmap:\d+
telephone-event\/\d+">
      <action application="start_dtmf"/>
    </condition>
  </extension>


I think the wiki used to have an example, but I don't see it on confluence.

On Sat, 13 Oct 2018 at 00:15, Michael Jerris <mike at jerris.com> wrote:

> There are vars with the full SDP, you can do a regex condition.  Check the
> info app to see all the vars you have at your disposal.  There may be a var
> for the te as well to show the pt.
>
> Mike
>
> On Oct 10, 2018, at 6:53 PM, Andrew Keil <andrew.keil at visytel.com> wrote:
>
> To FreeSWITCH Users,
>
> I have just noticed when making calls internationally to the UK from
> Australia that the following SDP sometimes happens when the call lands on
> FreeSWITCH:
>
> d5f14e47-2156-4236-be16-4eef1b7e9d3d 2018-10-04 06:59:28.819816 [DEBUG]
> sofia.c:7301 Remote SDP:
> d5f14e47-2156-4236-be16-4eef1b7e9d3d v=0
> d5f14e47-2156-4236-be16-4eef1b7e9d3d o=sbc-uk-mr-dh05a 451499 139728 IN
> IP4 XXX.XXX.XXX.XXX
> d5f14e47-2156-4236-be16-4eef1b7e9d3d s=sip call
> d5f14e47-2156-4236-be16-4eef1b7e9d3d c=IN IP4 XXX.XXX.XXX.XXX
> d5f14e47-2156-4236-be16-4eef1b7e9d3d t=0 0
> d5f14e47-2156-4236-be16-4eef1b7e9d3d m=audio 62754 RTP/AVP 8
> d5f14e47-2156-4236-be16-4eef1b7e9d3d a=rtpmap:8 PCMA/8000
> d5f14e47-2156-4236-be16-4eef1b7e9d3d a=ptime:20
>
> This then creates the following message around call answer:
>
> d5f14e47-2156-4236-be16-4eef1b7e9d3d 2018-10-04 06:59:28.839799 [DEBUG]
> switch_core_media.c:5766 *No 2833 in SDP. Liberal DTMF mode adding 101 as
> telephone-event.*
>
> From this point on DTMF is NOT detected by the FreeSWITCH IVR service.  I
> assume I would then need to use mod_dptools: start_dtmf to detect the DTMF
> tones within the channel.  Which I am yet to test.
>
> However my main question is whether there is a channel variable that would
> enable me to check whether 2833 support is not inside the SDP?
>
> Regards,
>
> Andrew Keil
>
>
>
> _________________________________________________________________________
> 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
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From steveayre at gmail.com  Sat Oct 13 15:14:26 2018
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 13 Oct 2018 16:14:26 +0100
Subject: [Freeswitch-users] Is it possible add new item in sip message
	header?
In-Reply-To: <CABHWxvwRysqdMd2YFDGjSEZQnz1_z86scZqApHV99XEOzXy5HQ@mail.gmail.com>
References: <CABHWxvzw+xP0bmwDg3CA-yT2GvT8PsRga726vm9w9tCNZYhvng@mail.gmail.com>
 <CAFiqYum-eyojm7n+AQJuBsS+6F2_=SzH07WC-MRh+4qqjB1A0g@mail.gmail.com>
 <CABHWxvwRysqdMd2YFDGjSEZQnz1_z86scZqApHV99XEOzXy5HQ@mail.gmail.com>
Message-ID: <CAFiqYukpkz4sUMnNnYqVhKo_jddjwm_wEzUHAsjXCzzft=r85w@mail.gmail.com>

That's my point though, anyone can put any value there they like (not in FS
without a code change but you could in a SIP proxy or custom UA). The only
way to limit it to a specific server securely would be via some kind of
authentication or encryption.

On Mon, 8 Oct 2018 at 15:51, 薛光宏 <albert.hsueh at cloudigit.com> wrote:

> Hi
>
> Thanks for reply.
> The app is for school. We think the developer wish this mobile app only
> connect to specific sip server so they do this kind of check.
>
> Sincerley
> Albert
>
> Steven Ayre <steveayre at gmail.com> 於 2018年10月6日 週六 上午3:33寫道：
>
>> Any specific reason the app needs to check that header? It couldn't be
>> considered secure since anyone could insert that header.
>>
>> It's similar to the user-agent which is set in the sofia profile, if
>> implemented it'd likely be implemented in the same way which'd mean a small
>> change to the mod_sofia code. I'd file a Jira.
>>
>> On Thu, 4 Oct 2018 at 17:28, 薛光宏 <albert.hsueh at cloudigit.com> wrote:
>>
>>> Hi All,
>>>
>>> We run a mobile app to *register* FreeSWITCH 1.6.20 via sip. This app
>>> will check value of server field in message header but the packet sent from
>>> FreeSWITCH doesn't have this field.
>>> We would like to know is it possible to add filed "Server" to message
>>> header by adjusting FreeSWITCH configuration?
>>>
>>> We tried this by adjusting dialplan (
>>> https://freeswitch.org/confluence/display/FREESWITCH/Sofia+SIP+Stack)
>>>         <action
>>> application="set"><![CDATA[sip_h_Server=xxxxxx]]></action>
>>>
>>> it works in invite stage, but can not work in register stage. The app
>>> checks the filed "Server" in register stage.
>>>
>>> Thanks in advance.
>>>
>>> --
>>> Sincerely
>>>
>>> Albert Hsueh
>>> System Engineer
>>> Cloudpe Corporation
>>> 02-8712 5955 ext. 831
>>> www.cloudpe.com
>>> _________________________________________________________________________
>>> 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
>>
>> _________________________________________________________________________
>> 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
>
> Albert Hsueh
> System Engineer
> Cloudpe Corporation
> 02-8712 5955 ext. 831
> www.cloudpe.com
> _________________________________________________________________________
> 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 ksrigo at gmail.com  Sat Oct 13 16:11:40 2018
From: ksrigo at gmail.com (Srigo Kanapathipillai)
Date: Sat, 13 Oct 2018 18:11:40 +0200
Subject: [Freeswitch-users] Multi gateways in profile
In-Reply-To: <CA+fRVSUkFuJRgkUAQCxr=3p6fj-joQR_yQD=c=YD4WRu8r=P-g@mail.gmail.com>
References: <CA+fRVSUkFuJRgkUAQCxr=3p6fj-joQR_yQD=c=YD4WRu8r=P-g@mail.gmail.com>
Message-ID: <f7699f6b-56aa-4f82-8841-1c1258cef39b@gmail.com>

Hi Mickael,

<gateway> 
..params...
<variables> 
<variable name="inbound_var_name" value="this is inbound" direction="inbound"/>
<variable name="outbound_var_name" value="this is outbound" direction="outbound"/> 
<variable name="both_var_name" value="this on any direction"/> 
</variables> 
</gateway>

Did you try to set variables on your gateway and use it in your dialplan?

⁣Srigo Kana​

On 13 Oct 2018, 00:31, at 00:31, Mickael Hubert <mickael at winlux.fr> wrote:
>Hi guys,
>I want to identify source of inbounds calls.
>In my external profile I have two gateways:
>
>- One with REGISTER method
>- One without REGISTER method (IP to IP), only iptables filtering
>
><gateways>
>      <gateway name="GW1">
>        <param name="from-domain" value="**********"/>
>        <param name="caller-id-in-from" value="true"/>
>        <param name="proxy" value="1.1.1.1"/>
>        <param name="register" value="true"/>
>        <param name="username" value="**************"/>
>        <param name="password" value="****************"/>
>        <param name="expire-seconds" value="3600"/>
>        <param name="retry-seconds" value="30"/>
>        <param name="ping" value="25"/>
>      </gateway>
>      <gateway name="GW2">
>        <param name="from-domain" value="*********"/>
>        <param name="caller-id-in-from" value="true"/>
>        <param name="proxy" value="2.2.2.2"/>
>        <param name="register" value="false"/>
>        <param name="retry-seconds" value="30"/>
>        <param name="ping" value="25"/>
>      </gateway>
>    </gateways>
>
>In my dialplan I have this conf:
>
><condition field="${sip_gateway}"  expression="GW1" break="never">
>  <action application="set" data="number_dest=${destination_number}" />
><action application="execute_extension" data="to-private XML
>from-gw1"/>
></condition>
><condition field="${sip_gateway}"  expression="GW2" break="never">
>  <action application="set" data="number_dest=${destination_number}" />
><action application="execute_extension" data="to-private XML
>from-gw2"/>
></condition>
>
>When a call arrives throught GW1 (REGISTERED), ${sip_gateway} is
>populate
>with "GW1" perfect !. But when another call arrives throught GW2 (not
>REGISTERED), ${sip_gateway} is empty.
>
>Please, how can I identify the source gateway where calls arrive ?
>maybe
>it's not a good way to do that ?
>
>thanks in advance
>
>
>------------------------------------------------------------------------
>
>_________________________________________________________________________
>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  Sat Oct 13 16:59:18 2018
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 13 Oct 2018 11:59:18 -0500
Subject: [Freeswitch-users] Freeswitch is not handling 180 after 200 OK
In-Reply-To: <25D2EC755404B4409F263AC6D050FEBB2E6D6A6E@AZ-FFEXMB03.global.avaya.com>
References: <25D2EC755404B4409F263AC6D050FEBB2E6D6A6E@AZ-FFEXMB03.global.avaya.com>
Message-ID: <196AF0AC-43CF-4C9E-A53C-10396AF214DB@freeswitch.org>

if you think you have found a bug, it should be reported to Jira complete with debug level logging. for sip related issues sofia global siptrace should be enabled and inline. and the logs should be attached as a .txt file

see https://freeswitch.org/confluence for more information on reporting bugs

Sent from my iPhone

> On Oct 11, 2018, at 04:16, Mody, Darshan (Darshan) <darshanmody at avaya.com> wrote:
> 
> Hi
>  
> We are running UDP traffic. During heavy load FS receives sometimes receives 180 response after 200 OK already received on the same call-id (dialog). However FS does not discard the same and starts processing 180 even though it has already received 200 OK on the same call-id (dialog).
>  
> Freeswitch should be discarding the 180 response after it gets 200 OK response from far-end.
>  
> In our case the FS core state machine even though it gets 200 OK waits for 200 Ok again.
>  
> Thanks
> Darshan
> _________________________________________________________________________
> 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  Sat Oct 13 17:24:51 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 13 Oct 2018 18:24:51 +0100
Subject: [Freeswitch-users] [Missing local host] in proxy media mode
In-Reply-To: <CAEJMVkB9sggz6y4RfPvWTYFuxOEpkvvM2Htmsnav8nfxRZzRog@mail.gmail.com>
References: <CAMBkyc1ETujbQLgYEyv+-URcV2OwCuWW0Dh2k+b6U2XjUkAMig@mail.gmail.com>
 <CAEJMVkB9sggz6y4RfPvWTYFuxOEpkvvM2Htmsnav8nfxRZzRog@mail.gmail.com>
Message-ID: <CAFGRPVpoOY1LWnM1Gv97sg6qWKsFKxnQdUmQypncEe45-AVDJA@mail.gmail.com>

http://lists.freeswitch.org/pipermail/freeswitch-users/2016-August/121866.html

On Sat, Oct 13, 2018, 18:19 Brian West <brian at freeswitch.com> wrote:

> Could you elaborate on why you're using Proxy Media for?
> /b
>
> On Fri, Oct 12, 2018 at 5:37 PM Marc Quore <marquore at gmail.com> wrote:
>
>> Hi, All
>>
>> I've got FreeSwitch 1.8.2 running in media proxy mode:
>>
>> <param name="inbound-proxy-media" value="true"/>
>> <param name="inbound-late-negotiation" value="true"/>
>>
>> Dialplan is configured to bridge incoming calls to remote gateway and
>> most of calls completing successfully, however some of them fail with
>> INCOMPATIBLE_DESTINATION cause. For failed calls everything goes normally
>> until bridged connection gets early media from callee:
>>
>> [NOTICE] sofia.c:7304 Pre-Answer sofia/external/2222 at 2.2.2.2!
>> [INFO] switch_ivr_originate.c:3747 Sending early media
>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PCMU/0
>> 0 ms 960 samples 384000 bits 1 channels
>> [DEBUG] switch_core_codec.c:111 sofia/internal/1111 at 1.1.1.1 Original
>> read codec set to PCMU:0
>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
>> 1111 at 1.1.1.1] 1.1.1.1:18792->1.1.1.1:18792 codec: 0 ms: 20
>> [ERR] switch_core_media.c:9510 AUDIO RTP REPORTS ERROR: *[Missing local
>> host]*
>> [NOTICE] switch_core_media.c:9511 Hangup sofia/internal/1111 at 1.1.1.1
>> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
>>
>> Here's a dump for this call:
>>
>> [image: image.png]
>>
>> Healthy call on the same machine:
>>
>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/vpn/2222 at 2.2.2.2 PROXY/0
>> 0 ms 160 samples 0 bits 1 channels
>> [DEBUG] switch_core_codec.c:111 sofia/internal/2222 at 2.2.2.2 Original
>> read codec set to PROXY:0
>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
>> 2222 at 2.2.2.2] 2.2.2.2:60818->2.2.2.2:60818 codec: 0 ms: 20
>>
>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PROXY/0
>> 0 ms 160 samples 0 bits 1 channels
>> [DEBUG] switch_core_codec.c:111 sofia/external/1111 at 1.1.1.1 Original
>> read codec set to PROXY:0
>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/external/
>> 1111 at 1.1.1.1] 1.1.1.1:31412->1.1.1.1:31412 codec: 0 ms: 20
>>
>> So it looks like proxified media connection cannot be established for
>> some calls. It shouldn't be a codec problem as call parties are configured
>> with common codecs, FS is bond to local interface which is running with no
>> errors. But I have no idea what "local host is missing" in my case so I'll
>> be appreciated for any hint.
>>
>> FreeSWITCH Version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git
>> a98a958 2018-09-26 17:55:25Z 64bit)
>> CentOS Linux release 7.5.1804 (Core)
>>
>> Best,
>> Marc
>> _________________________________________________________________________
>> 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>
> _________________________________________________________________________
> 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 mickael at winlux.fr  Sat Oct 13 20:11:19 2018
From: mickael at winlux.fr (Mickael Hubert)
Date: Sat, 13 Oct 2018 22:11:19 +0200
Subject: [Freeswitch-users] Multi gateways in profile
In-Reply-To: <f7699f6b-56aa-4f82-8841-1c1258cef39b@gmail.com>
References: <CA+fRVSUkFuJRgkUAQCxr=3p6fj-joQR_yQD=c=YD4WRu8r=P-g@mail.gmail.com>
 <f7699f6b-56aa-4f82-8841-1c1258cef39b@gmail.com>
Message-ID: <CA+fRVSXA5-bzKqNgmrRaq=FFBqe06iPYYgv0rc48chwY9_PZ1A@mail.gmail.com>

Hi
Thanks a lot !

I think the best way (maybe only way) it's to match with from IP.
It's sure when inbound call arrives from GW2, any variables in gateway
configuration aren't populate.

But how Can I do math with IP please ?

++


Le sam. 13 oct. 2018 18:26, Srigo Kanapathipillai <ksrigo at gmail.com> a
écrit :

> Hi Mickael,
>
> <gateway>
> ..params...
> <variables>
> <variable name="inbound_var_name" value="this is inbound"
> direction="inbound"/>
> <variable name="outbound_var_name" value="this is outbound"
> direction="outbound"/>
> <variable name="both_var_name" value="this on any direction"/>
> </variables>
> </gateway>
>
> Did you try to set variables on your gateway and use it in your dialplan?
>
> Srigo Kana
> On 13 Oct 2018, at 00:31, Mickael Hubert <mickael at winlux.fr> wrote:
>>
>> Hi guys,
>> I want to identify source of inbounds calls.
>> In my external profile I have two gateways:
>>
>> - One with REGISTER method
>> - One without REGISTER method (IP to IP), only iptables filtering
>>
>> <gateways>
>>       <gateway name="GW1">
>>         <param name="from-domain" value="**********"/>
>>         <param name="caller-id-in-from" value="true"/>
>>         <param name="proxy" value="1.1.1.1"/>
>>         <param name="register" value="true"/>
>>         <param name="username" value="**************"/>
>>         <param name="password" value="****************"/>
>>         <param name="expire-seconds" value="3600"/>
>>         <param name="retry-seconds" value="30"/>
>>         <param name="ping" value="25"/>
>>       </gateway>
>>       <gateway name="GW2">
>>         <param name="from-domain" value="*********"/>
>>         <param name="caller-id-in-from" value="true"/>
>>         <param name="proxy" value="2.2.2.2"/>
>>         <param name="register" value="false"/>
>>         <param name="retry-seconds" value="30"/>
>>         <param name="ping" value="25"/>
>>       </gateway>
>>     </gateways>
>>
>> In my dialplan I have this conf:
>>
>> <condition field="${sip_gateway}"  expression="GW1" break="never">
>>    <action application="set" data="number_dest=${destination_number}" />
>>    <action application="execute_extension" data="to-private XML
>> from-gw1"/>
>> </condition>
>> <condition field="${sip_gateway}"  expression="GW2" break="never">
>>    <action application="set" data="number_dest=${destination_number}" />
>>    <action application="execute_extension" data="to-private XML
>> from-gw2"/>
>> </condition>
>>
>> When a call arrives throught GW1 (REGISTERED), ${sip_gateway} is populate
>> with "GW1" perfect !. But when another call arrives throught GW2 (not
>> REGISTERED), ${sip_gateway} is empty.
>>
>> Please, how can I identify the source gateway where calls arrive ? maybe
>> it's not a good way to do that ?
>>
>> thanks in advance
>>
>> ------------------------------
>>
>> 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
>>
>> _________________________________________________________________________
> 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  Sat Oct 13 21:38:48 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 13 Oct 2018 22:38:48 +0100
Subject: [Freeswitch-users] Freeswitch is not handling 180 after 200 OK
In-Reply-To: <25D2EC755404B4409F263AC6D050FEBB2E6D6A6E@AZ-FFEXMB03.global.avaya.com>
References: <25D2EC755404B4409F263AC6D050FEBB2E6D6A6E@AZ-FFEXMB03.global.avaya.com>
Message-ID: <CAFGRPVrqqmr63XzfyYeWW7iK2GXpJafRj6eve8-oi+TbmjQYww@mail.gmail.com>

Well, technically speaking, the remote end should _not_ send a 180 after a
200. But then it seems like it's traditionally FS' job to adapt to faulty
gws.

Take a look at
https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/6587394
you may find some answers there.

Good luck.

On Thursday, October 11, 2018, Mody, Darshan (Darshan) <
darshanmody at avaya.com> wrote:

> Hi
>
>
>
> We are running UDP traffic. During heavy load FS receives sometimes
> receives 180 response after 200 OK already received on the same call-id
> (dialog). However FS does not discard the same and starts processing 180
> even though it has already received 200 OK on the same call-id (dialog).
>
>
>
> Freeswitch should be discarding the 180 response after it gets 200 OK
> response from far-end.
>
>
>
> In our case the FS core state machine even though it gets 200 OK waits for
> 200 Ok again.
>
>
>
> Thanks
>
> Darshan
>


-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From brian at freeswitch.com  Sat Oct 13 23:30:26 2018
From: brian at freeswitch.com (Brian West)
Date: Sat, 13 Oct 2018 18:30:26 -0500
Subject: [Freeswitch-users] Multi gateways in profile
In-Reply-To: <f7699f6b-56aa-4f82-8841-1c1258cef39b@gmail.com>
References: <CA+fRVSUkFuJRgkUAQCxr=3p6fj-joQR_yQD=c=YD4WRu8r=P-g@mail.gmail.com>
 <f7699f6b-56aa-4f82-8841-1c1258cef39b@gmail.com>
Message-ID: <CAEJMVkDEYW1WuXDmbouo-j-KiBanY67yWTNvXUQy7pCaE_QeYQ@mail.gmail.com>

Will NOT work unless the remote calls back the exact registered contact or
pass back the params on the contact.

On Sat, Oct 13, 2018 at 11:26 AM Srigo Kanapathipillai <ksrigo at gmail.com>
wrote:

> Hi Mickael,
>
> <gateway>
> ..params...
> <variables>
> <variable name="inbound_var_name" value="this is inbound"
> direction="inbound"/>
> <variable name="outbound_var_name" value="this is outbound"
> direction="outbound"/>
> <variable name="both_var_name" value="this on any direction"/>
> </variables>
> </gateway>
>
> Did you try to set variables on your gateway and use it in your dialplan?
>
> Srigo Kana
> On 13 Oct 2018, at 00:31, Mickael Hubert <mickael at winlux.fr> wrote:
>>
>> Hi guys,
>> I want to identify source of inbounds calls.
>> In my external profile I have two gateways:
>>
>> - One with REGISTER method
>> - One without REGISTER method (IP to IP), only iptables filtering
>>
>> <gateways>
>>       <gateway name="GW1">
>>         <param name="from-domain" value="**********"/>
>>         <param name="caller-id-in-from" value="true"/>
>>         <param name="proxy" value="1.1.1.1"/>
>>         <param name="register" value="true"/>
>>         <param name="username" value="**************"/>
>>         <param name="password" value="****************"/>
>>         <param name="expire-seconds" value="3600"/>
>>         <param name="retry-seconds" value="30"/>
>>         <param name="ping" value="25"/>
>>       </gateway>
>>       <gateway name="GW2">
>>         <param name="from-domain" value="*********"/>
>>         <param name="caller-id-in-from" value="true"/>
>>         <param name="proxy" value="2.2.2.2"/>
>>         <param name="register" value="false"/>
>>         <param name="retry-seconds" value="30"/>
>>         <param name="ping" value="25"/>
>>       </gateway>
>>     </gateways>
>>
>> In my dialplan I have this conf:
>>
>> <condition field="${sip_gateway}"  expression="GW1" break="never">
>>    <action application="set" data="number_dest=${destination_number}" />
>>    <action application="execute_extension" data="to-private XML
>> from-gw1"/>
>> </condition>
>> <condition field="${sip_gateway}"  expression="GW2" break="never">
>>    <action application="set" data="number_dest=${destination_number}" />
>>    <action application="execute_extension" data="to-private XML
>> from-gw2"/>
>> </condition>
>>
>> When a call arrives throught GW1 (REGISTERED), ${sip_gateway} is populate
>> with "GW1" perfect !. But when another call arrives throught GW2 (not
>> REGISTERED), ${sip_gateway} is empty.
>>
>> Please, how can I identify the source gateway where calls arrive ? maybe
>> it's not a good way to do that ?
>>
>> thanks in advance
>>
>> ------------------------------
>>
>> 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
>>
>> _________________________________________________________________________
> 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 vma at vallimamod.org  Sun Oct 14 17:38:03 2018
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Sun, 14 Oct 2018 19:38:03 +0200
Subject: [Freeswitch-users] Multi gateways in profile
In-Reply-To: <CA+fRVSXA5-bzKqNgmrRaq=FFBqe06iPYYgv0rc48chwY9_PZ1A@mail.gmail.com>
References: <CA+fRVSUkFuJRgkUAQCxr=3p6fj-joQR_yQD=c=YD4WRu8r=P-g@mail.gmail.com>
 <f7699f6b-56aa-4f82-8841-1c1258cef39b@gmail.com>
 <CA+fRVSXA5-bzKqNgmrRaq=FFBqe06iPYYgv0rc48chwY9_PZ1A@mail.gmail.com>
Message-ID: <C2ADA4BE-996C-4995-8D75-92C7F5BDA0CC@vallimamod.org>

Hi,

For ip addr matching: <condition field="network_addr" expression="^192\.168\.1\.1$"/>

Alternatively, If you want some variables populated on incoming call from a given IP the same way as with registered gateways, you can define a directory user entry (with "cidr" param to allow ip-auth) and define the your variables there. For example:

    <user id="GW2" cidr="2.2.2.2/32">
        <variables>
            <variable name="accountcode" value="GW2"/>
            <variable name="call_limit" value="10"/>
            <variable name="max_ringtime" value="45"/>
        </variables>
    </user>

Hope this helps!


Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sip.solutions
linkedin.com/in/vallimamod
.

> On 13 Oct 2018, at 22:11, Mickael Hubert <mickael at winlux.fr> wrote:
> 
> Hi
> Thanks a lot !
> 
> I think the best way (maybe only way) it's to match with from IP.
> It's sure when inbound call arrives from GW2, any variables in gateway configuration aren't populate.
> 
> But how Can I do math with IP please ?
> 
> ++
> 
> 
> Le sam. 13 oct. 2018 18:26, Srigo Kanapathipillai <ksrigo at gmail.com> a écrit :
> Hi Mickael,
> 
> <gateway> 
> ..params...
> <variables> 
> <variable name="inbound_var_name" value="this is inbound" direction="inbound"/>
> <variable name="outbound_var_name" value="this is outbound" direction="outbound"/> 
> <variable name="both_var_name" value="this on any direction"/> 
> </variables> 
> </gateway>
> 
> Did you try to set variables on your gateway and use it in your dialplan?
> 
> Srigo Kana
> On 13 Oct 2018, at 00:31, Mickael Hubert <mickael at winlux.fr> wrote:
> Hi guys,
> I want to identify source of inbounds calls. 
> In my external profile I have two gateways:
> 
> - One with REGISTER method
> - One without REGISTER method (IP to IP), only iptables filtering
> 
> <gateways>
>       <gateway name="GW1">
>         <param name="from-domain" value="**********"/>
>         <param name="caller-id-in-from" value="true"/>
>         <param name="proxy" value="1.1.1.1"/>
>         <param name="register" value="true"/>
>         <param name="username" value="**************"/>
>         <param name="password" value="****************"/>
>         <param name="expire-seconds" value="3600"/>
>         <param name="retry-seconds" value="30"/>
>         <param name="ping" value="25"/>
>       </gateway>
>       <gateway name="GW2">
>         <param name="from-domain" value="*********"/>
>         <param name="caller-id-in-from" value="true"/>
>         <param name="proxy" value="2.2.2.2"/>
>         <param name="register" value="false"/>
>         <param name="retry-seconds" value="30"/>
>         <param name="ping" value="25"/>
>       </gateway>
>     </gateways>
> 
> In my dialplan I have this conf:
> 
> <condition field="${sip_gateway}"  expression="GW1" break="never">
>    <action application="set" data="number_dest=${destination_number}" />
>    <action application="execute_extension" data="to-private XML from-gw1"/>
> </condition>
> <condition field="${sip_gateway}"  expression="GW2" break="never">
>    <action application="set" data="number_dest=${destination_number}" />
>    <action application="execute_extension" data="to-private XML from-gw2"/>
> </condition>
> 
> When a call arrives throught GW1 (REGISTERED), ${sip_gateway} is populate with "GW1" perfect !. But when another call arrives throught GW2 (not REGISTERED), ${sip_gateway} is empty.
> 
> Please, how can I identify the source gateway where calls arrive ? maybe it's not a good way to do that ?
> 
> thanks in advance
> 
> 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
> _________________________________________________________________________
> 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
> _________________________________________________________________________
> 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 mickael at winlux.fr  Sun Oct 14 17:59:18 2018
From: mickael at winlux.fr (Mickael Hubert)
Date: Sun, 14 Oct 2018 19:59:18 +0200
Subject: [Freeswitch-users] Multi gateways in profile
In-Reply-To: <CAEJMVkDEYW1WuXDmbouo-j-KiBanY67yWTNvXUQy7pCaE_QeYQ@mail.gmail.com>
References: <CA+fRVSUkFuJRgkUAQCxr=3p6fj-joQR_yQD=c=YD4WRu8r=P-g@mail.gmail.com>
 <f7699f6b-56aa-4f82-8841-1c1258cef39b@gmail.com>
 <CAEJMVkDEYW1WuXDmbouo-j-KiBanY67yWTNvXUQy7pCaE_QeYQ@mail.gmail.com>
Message-ID: <CA+fRVSXeXz_Bj3WanBzWnws1t976Ztvecqo-ph83HtRV=5592w@mail.gmail.com>

Hi
even if i check by source IP ?

Ex:

<condition field="network_addr" expression="^192\.168\.1\.1$"/>





Le dim. 14 oct. 2018 18:34, Brian West <brian at freeswitch.com> a écrit :

> Will NOT work unless the remote calls back the exact registered contact or
> pass back the params on the contact.
>
> On Sat, Oct 13, 2018 at 11:26 AM Srigo Kanapathipillai <ksrigo at gmail.com>
> wrote:
>
>> Hi Mickael,
>>
>> <gateway>
>> ..params...
>> <variables>
>> <variable name="inbound_var_name" value="this is inbound"
>> direction="inbound"/>
>> <variable name="outbound_var_name" value="this is outbound"
>> direction="outbound"/>
>> <variable name="both_var_name" value="this on any direction"/>
>> </variables>
>> </gateway>
>>
>> Did you try to set variables on your gateway and use it in your dialplan?
>>
>> Srigo Kana
>> On 13 Oct 2018, at 00:31, Mickael Hubert <mickael at winlux.fr> wrote:
>>>
>>> Hi guys,
>>> I want to identify source of inbounds calls.
>>> In my external profile I have two gateways:
>>>
>>> - One with REGISTER method
>>> - One without REGISTER method (IP to IP), only iptables filtering
>>>
>>> <gateways>
>>>       <gateway name="GW1">
>>>         <param name="from-domain" value="**********"/>
>>>         <param name="caller-id-in-from" value="true"/>
>>>         <param name="proxy" value="1.1.1.1"/>
>>>         <param name="register" value="true"/>
>>>         <param name="username" value="**************"/>
>>>         <param name="password" value="****************"/>
>>>         <param name="expire-seconds" value="3600"/>
>>>         <param name="retry-seconds" value="30"/>
>>>         <param name="ping" value="25"/>
>>>       </gateway>
>>>       <gateway name="GW2">
>>>         <param name="from-domain" value="*********"/>
>>>         <param name="caller-id-in-from" value="true"/>
>>>         <param name="proxy" value="2.2.2.2"/>
>>>         <param name="register" value="false"/>
>>>         <param name="retry-seconds" value="30"/>
>>>         <param name="ping" value="25"/>
>>>       </gateway>
>>>     </gateways>
>>>
>>> In my dialplan I have this conf:
>>>
>>> <condition field="${sip_gateway}"  expression="GW1" break="never">
>>>    <action application="set" data="number_dest=${destination_number}" />
>>>    <action application="execute_extension" data="to-private XML
>>> from-gw1"/>
>>> </condition>
>>> <condition field="${sip_gateway}"  expression="GW2" break="never">
>>>    <action application="set" data="number_dest=${destination_number}" />
>>>    <action application="execute_extension" data="to-private XML
>>> from-gw2"/>
>>> </condition>
>>>
>>> When a call arrives throught GW1 (REGISTERED), ${sip_gateway} is
>>> populate with "GW1" perfect !. But when another call arrives throught GW2
>>> (not REGISTERED), ${sip_gateway} is empty.
>>>
>>> Please, how can I identify the source gateway where calls arrive ? maybe
>>> it's not a good way to do that ?
>>>
>>> thanks in advance
>>>
>>> ------------------------------
>>>
>>> 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
>>>
>>> _________________________________________________________________________
>> 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>
> _________________________________________________________________________
> 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 brian at freeswitch.com  Sun Oct 14 18:06:56 2018
From: brian at freeswitch.com (Brian West)
Date: Sun, 14 Oct 2018 13:06:56 -0500
Subject: [Freeswitch-users] Freeswitch is not handling 180 after 200 OK
In-Reply-To: <CAFGRPVrqqmr63XzfyYeWW7iK2GXpJafRj6eve8-oi+TbmjQYww@mail.gmail.com>
References: <25D2EC755404B4409F263AC6D050FEBB2E6D6A6E@AZ-FFEXMB03.global.avaya.com>
 <CAFGRPVrqqmr63XzfyYeWW7iK2GXpJafRj6eve8-oi+TbmjQYww@mail.gmail.com>
Message-ID: <CAEJMVkCtAs_XNa26FRMRNfY01_HC-ep2w2izqbzgQxrWQ_a__g@mail.gmail.com>

This is already filed :) . And the order has little to do with it.

/b


On Sun, Oct 14, 2018 at 11:33 AM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Well, technically speaking, the remote end should _not_ send a 180 after a
> 200. But then it seems like it's traditionally FS' job to adapt to faulty
> gws.
>
> Take a look at
> https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/6587394
> you may find some answers there.
>
> Good luck.
>
> On Thursday, October 11, 2018, Mody, Darshan (Darshan) <
> darshanmody at avaya.com> wrote:
>
>> Hi
>>
>>
>>
>> We are running UDP traffic. During heavy load FS receives sometimes
>> receives 180 response after 200 OK already received on the same call-id
>> (dialog). However FS does not discard the same and starts processing 180
>> even though it has already received 200 OK on the same call-id (dialog).
>>
>>
>>
>> Freeswitch should be discarding the 180 response after it gets 200 OK
>> response from far-end.
>>
>>
>>
>> In our case the FS core state machine even though it gets 200 OK waits
>> for 200 Ok again.
>>
>>
>>
>> Thanks
>>
>> Darshan
>>
>
>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
> _________________________________________________________________________
> 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 mike at jerris.com  Sun Oct 14 21:36:06 2018
From: mike at jerris.com (Michael Jerris)
Date: Sun, 14 Oct 2018 16:36:06 -0500
Subject: [Freeswitch-users] mod_spy,
 does it only work for registered users?
In-Reply-To: <E1gB72t-0001fn-QS@mail.opensips.org>
References: <E1gB72t-0001fn-QS@mail.opensips.org>
Message-ID: <D2A08A40-D9D9-489E-BA5E-63DBAF123CF8@jerris.com>

Can you show me exactly how you are overriding them in dialplan?  This should work.

> On Oct 12, 2018, at 6:40 PM, Alex Goulis <agoulis at opensips.org> wrote:
> 
> Man thanks for digging deeper... 
> 
> unfortunately I have an identical match to Pairs 0-3 in my tests.   Ie, they all have the same user/domain in them. And they all match the user/domain I set when invoking userspy.
> 
> The events after the invite show them populated as well. 
> 
> I'm pretty sure the eavesdrop trigger happens before I can manually override them with dialplan on the incoming call to be targeted.
> 
> Happy to test any scenario you can further think of though.
> 
> Alex
> 
> 
> 
> Sent via the Samsung Galaxy Note8, an AT&T 4G LTE smartphone
> 
> -------- Original message --------
> From: Michael Jerris <mike at jerris.com>
> Date: 10/12/18 3:53 PM (GMT-07:00)
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] mod_spy, does it only work for registered users?
> 
> After talking to you about this, I just took a look. No its not based on registered users, but authed users will probably more reliably set the right variables to figure out the right user to match to.  Trick is to have the vars set to be able to look them up.
> 
>     username[0] = switch_event_get_header(event, "Caller-Username");                                                                                           
>     domain[0] = switch_event_get_header(event, "variable_domain_name");                                                                                        
>                                                                                                                                                                 
>     username[1] = switch_event_get_header(event, "variable_dialed_user");                                                                                      
>     domain[1] = switch_event_get_header(event, "variable_dialed_domain");                                                                                      
>                                                                                                                                                                 
>     username[2] = switch_event_get_header(event, "variable_user_name");                                                                                        
>     domain[2] = switch_event_get_header(event, "variable_domain_name");                                                                                        
>                                                                                                                                                                 
>     username[3] = switch_event_get_header(event, "variable_sip_to_user");                                                                                      
>     domain[3] = switch_event_get_header(event, "variable_domain_name");                                                                                        
>                                                                                                                                                                 
>     username[4] = switch_event_get_header(event, "variable_verto_user");                                                                                       
>     domain[4] = switch_event_get_header(event, "variable_verto_host");                                                                                         
> 
> It has to match one of those pairs.
> 
> 
>> On Oct 5, 2018, at 6:24 PM, Alex Goulis <agoulis at opensips.org <mailto:agoulis at opensips.org>> wrote:
>> 
>> Hi all...
>> 
>> Does the userspy application only target registered users or can you use it to spy calls to let's say calls to a DID?

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From mike at jerris.com  Sun Oct 14 21:40:39 2018
From: mike at jerris.com (Michael Jerris)
Date: Sun, 14 Oct 2018 16:40:39 -0500
Subject: [Freeswitch-users] [Missing local host] in proxy media mode
In-Reply-To: <CAEJMVkB9sggz6y4RfPvWTYFuxOEpkvvM2Htmsnav8nfxRZzRog@mail.gmail.com>
References: <CAMBkyc1ETujbQLgYEyv+-URcV2OwCuWW0Dh2k+b6U2XjUkAMig@mail.gmail.com>
 <CAEJMVkB9sggz6y4RfPvWTYFuxOEpkvvM2Htmsnav8nfxRZzRog@mail.gmail.com>
Message-ID: <187AA53B-CE2E-416D-996E-AD6D8F12043B@jerris.com>

I’ll expand on Brian’s comment.  Proxy media should NEVER be used.  There is one exception to this rule, and that is when you are trying to pass codecs that freeswitch does not know about, and even that case has a better workaround than to use proxy media.  If you are using proxy media you should stop, if you are not sure how to do so you should discuss on the mailing list.  Proxy media mode is NOT the way to avoid transcoding, that can be done properly without this setting and this setting should never be used.

Mike

> On Oct 12, 2018, at 5:07 PM, Brian West <brian at freeswitch.com> wrote:
> 
> Could you elaborate on why you're using Proxy Media for?
> /b
> 
> On Fri, Oct 12, 2018 at 5:37 PM Marc Quore <marquore at gmail.com <mailto:marquore at gmail.com>> wrote:
> Hi, All
> 
> I've got FreeSwitch 1.8.2 running in media proxy mode:
> 
> <param name="inbound-proxy-media" value="true"/>
> <param name="inbound-late-negotiation" value="true"/>
> 
> Dialplan is configured to bridge incoming calls to remote gateway and most of calls completing successfully, however some of them fail with INCOMPATIBLE_DESTINATION cause. For failed calls everything goes normally until bridged connection gets early media from callee:
> 
> [NOTICE] sofia.c:7304 Pre-Answer sofia/external/2222 at 2.2.2.2 <mailto:2222 at 2.2.2.2>!
> [INFO] switch_ivr_originate.c:3747 Sending early media
> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 <mailto:1111 at 1.1.1.1> PCMU/0 0 ms 960 samples 384000 bits 1 channels
> [DEBUG] switch_core_codec.c:111 sofia/internal/1111 at 1.1.1.1 <mailto:1111 at 1.1.1.1> Original read codec set to PCMU:0
> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/1111 at 1.1.1.1 <mailto:1111 at 1.1.1.1>] 1.1.1.1:18792->1.1.1.1:18792 <http://1.1.1.1:18792/> codec: 0 ms: 20
> [ERR] switch_core_media.c:9510 AUDIO RTP REPORTS ERROR: [Missing local host]
> [NOTICE] switch_core_media.c:9511 Hangup sofia/internal/1111 at 1.1.1.1 <mailto:1111 at 1.1.1.1> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
> 
> Here's a dump for this call:
> 
> <image.png>
> 
> Healthy call on the same machine:
> 
> [DEBUG] switch_core_media.c:3781 Set Codec sofia/vpn/2222 at 2.2.2.2 <mailto:2222 at 2.2.2.2> PROXY/0 0 ms 160 samples 0 bits 1 channels
> [DEBUG] switch_core_codec.c:111 sofia/internal/2222 at 2.2.2.2 <mailto:2222 at 2.2.2.2> Original read codec set to PROXY:0
> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/2222 at 2.2.2.2 <mailto:2222 at 2.2.2.2>] 2.2.2.2:60818->2.2.2.2:60818 <http://2.2.2.2:60818/> codec: 0 ms: 20
> 
> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 <mailto:1111 at 1.1.1.1> PROXY/0 0 ms 160 samples 0 bits 1 channels
> [DEBUG] switch_core_codec.c:111 sofia/external/1111 at 1.1.1.1 <mailto:1111 at 1.1.1.1> Original read codec set to PROXY:0
> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/external/1111 at 1.1.1.1 <mailto:1111 at 1.1.1.1>] 1.1.1.1:31412->1.1.1.1:31412 <http://1.1.1.1:31412/> codec: 0 ms: 20
> 
> So it looks like proxified media connection cannot be established for some calls. It shouldn't be a codec problem as call parties are configured with common codecs, FS is bond to local interface which is running with no errors. But I have no idea what "local host is missing" in my case so I'll be appreciated for any hint.
> 
> FreeSWITCH Version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git a98a958 2018-09-26 17:55:25Z 64bit)
> CentOS Linux release 7.5.1804 (Core)
> 
> Best,
> Marc
> _________________________________________________________________________
> 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/>
> 
> -- 
> 
> Brian West | Co-founder and Developer
> Need Commercial support? email sales at freeswitch.com <mailto: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 <mailto:brian at freeswitch.com>
> Mobile: 918-424-9378
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>  <https://www.facebook.com/freeswitch>  <https://twitter.com/freeswitch>
> _________________________________________________________________________
> 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 marquore at gmail.com  Sun Oct 14 21:48:07 2018
From: marquore at gmail.com (Marc Quore)
Date: Mon, 15 Oct 2018 00:48:07 +0300
Subject: [Freeswitch-users] [Missing local host] in proxy media mode
In-Reply-To: <CAEJMVkB9sggz6y4RfPvWTYFuxOEpkvvM2Htmsnav8nfxRZzRog@mail.gmail.com>
References: <CAMBkyc1ETujbQLgYEyv+-URcV2OwCuWW0Dh2k+b6U2XjUkAMig@mail.gmail.com>
 <CAEJMVkB9sggz6y4RfPvWTYFuxOEpkvvM2Htmsnav8nfxRZzRog@mail.gmail.com>
Message-ID: <CAMBkyc0LSNjTST4Qj6xq2HMWXzjMVpMRFForaiGEOt_nhYADmw@mail.gmail.com>

Hello Brian,

I was under impression that media proxying mode is optimal choice if we
don't touch media, as in our case.

Regards,
Marc


сб, 13 окт. 2018 г. в 21:29, Brian West <brian at freeswitch.com>:

> Could you elaborate on why you're using Proxy Media for?
> /b
>
> On Fri, Oct 12, 2018 at 5:37 PM Marc Quore <marquore at gmail.com> wrote:
>
>> Hi, All
>>
>> I've got FreeSwitch 1.8.2 running in media proxy mode:
>>
>> <param name="inbound-proxy-media" value="true"/>
>> <param name="inbound-late-negotiation" value="true"/>
>>
>> Dialplan is configured to bridge incoming calls to remote gateway and
>> most of calls completing successfully, however some of them fail with
>> INCOMPATIBLE_DESTINATION cause. For failed calls everything goes normally
>> until bridged connection gets early media from callee:
>>
>> [NOTICE] sofia.c:7304 Pre-Answer sofia/external/2222 at 2.2.2.2!
>> [INFO] switch_ivr_originate.c:3747 Sending early media
>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PCMU/0
>> 0 ms 960 samples 384000 bits 1 channels
>> [DEBUG] switch_core_codec.c:111 sofia/internal/1111 at 1.1.1.1 Original
>> read codec set to PCMU:0
>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
>> 1111 at 1.1.1.1] 1.1.1.1:18792->1.1.1.1:18792 codec: 0 ms: 20
>> [ERR] switch_core_media.c:9510 AUDIO RTP REPORTS ERROR: *[Missing local
>> host]*
>> [NOTICE] switch_core_media.c:9511 Hangup sofia/internal/1111 at 1.1.1.1
>> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
>>
>> Here's a dump for this call:
>>
>> [image: image.png]
>>
>> Healthy call on the same machine:
>>
>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/vpn/2222 at 2.2.2.2 PROXY/0
>> 0 ms 160 samples 0 bits 1 channels
>> [DEBUG] switch_core_codec.c:111 sofia/internal/2222 at 2.2.2.2 Original
>> read codec set to PROXY:0
>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
>> 2222 at 2.2.2.2] 2.2.2.2:60818->2.2.2.2:60818 codec: 0 ms: 20
>>
>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PROXY/0
>> 0 ms 160 samples 0 bits 1 channels
>> [DEBUG] switch_core_codec.c:111 sofia/external/1111 at 1.1.1.1 Original
>> read codec set to PROXY:0
>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/external/
>> 1111 at 1.1.1.1] 1.1.1.1:31412->1.1.1.1:31412 codec: 0 ms: 20
>>
>> So it looks like proxified media connection cannot be established for
>> some calls. It shouldn't be a codec problem as call parties are configured
>> with common codecs, FS is bond to local interface which is running with no
>> errors. But I have no idea what "local host is missing" in my case so I'll
>> be appreciated for any hint.
>>
>> FreeSWITCH Version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git
>> a98a958 2018-09-26 17:55:25Z 64bit)
>> CentOS Linux release 7.5.1804 (Core)
>>
>> Best,
>> Marc
>> _________________________________________________________________________
>> 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>
> _________________________________________________________________________
> 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 marquore at gmail.com  Sun Oct 14 22:04:14 2018
From: marquore at gmail.com (Marc Quore)
Date: Mon, 15 Oct 2018 01:04:14 +0300
Subject: [Freeswitch-users] [Missing local host] in proxy media mode
In-Reply-To: <CAFGRPVpoOY1LWnM1Gv97sg6qWKsFKxnQdUmQypncEe45-AVDJA@mail.gmail.com>
References: <CAMBkyc1ETujbQLgYEyv+-URcV2OwCuWW0Dh2k+b6U2XjUkAMig@mail.gmail.com>
 <CAEJMVkB9sggz6y4RfPvWTYFuxOEpkvvM2Htmsnav8nfxRZzRog@mail.gmail.com>
 <CAFGRPVpoOY1LWnM1Gv97sg6qWKsFKxnQdUmQypncEe45-AVDJA@mail.gmail.com>
Message-ID: <CAMBkyc1w8=KxdPyUwVEegKJ3xGQBPmxZLySHwNfTy8_Ro2FOqw@mail.gmail.com>

Hi David, I've seen this thank you.

Regards,
Marc


вс, 14 окт. 2018 г. в 20:50, David Villasmil <david.villasmil.work at gmail.com
>:

>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2016-August/121866.html
>
> On Sat, Oct 13, 2018, 18:19 Brian West <brian at freeswitch.com> wrote:
>
>> Could you elaborate on why you're using Proxy Media for?
>> /b
>>
>> On Fri, Oct 12, 2018 at 5:37 PM Marc Quore <marquore at gmail.com> wrote:
>>
>>> Hi, All
>>>
>>> I've got FreeSwitch 1.8.2 running in media proxy mode:
>>>
>>> <param name="inbound-proxy-media" value="true"/>
>>> <param name="inbound-late-negotiation" value="true"/>
>>>
>>> Dialplan is configured to bridge incoming calls to remote gateway and
>>> most of calls completing successfully, however some of them fail with
>>> INCOMPATIBLE_DESTINATION cause. For failed calls everything goes normally
>>> until bridged connection gets early media from callee:
>>>
>>> [NOTICE] sofia.c:7304 Pre-Answer sofia/external/2222 at 2.2.2.2!
>>> [INFO] switch_ivr_originate.c:3747 Sending early media
>>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PCMU/0
>>> 0 ms 960 samples 384000 bits 1 channels
>>> [DEBUG] switch_core_codec.c:111 sofia/internal/1111 at 1.1.1.1 Original
>>> read codec set to PCMU:0
>>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
>>> 1111 at 1.1.1.1] 1.1.1.1:18792->1.1.1.1:18792 codec: 0 ms: 20
>>> [ERR] switch_core_media.c:9510 AUDIO RTP REPORTS ERROR: *[Missing local
>>> host]*
>>> [NOTICE] switch_core_media.c:9511 Hangup sofia/internal/1111 at 1.1.1.1
>>> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
>>>
>>> Here's a dump for this call:
>>>
>>> [image: image.png]
>>>
>>> Healthy call on the same machine:
>>>
>>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/vpn/2222 at 2.2.2.2 PROXY/0
>>> 0 ms 160 samples 0 bits 1 channels
>>> [DEBUG] switch_core_codec.c:111 sofia/internal/2222 at 2.2.2.2 Original
>>> read codec set to PROXY:0
>>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
>>> 2222 at 2.2.2.2] 2.2.2.2:60818->2.2.2.2:60818 codec: 0 ms: 20
>>>
>>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PROXY/0
>>> 0 ms 160 samples 0 bits 1 channels
>>> [DEBUG] switch_core_codec.c:111 sofia/external/1111 at 1.1.1.1 Original
>>> read codec set to PROXY:0
>>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/external/
>>> 1111 at 1.1.1.1] 1.1.1.1:31412->1.1.1.1:31412 codec: 0 ms: 20
>>>
>>> So it looks like proxified media connection cannot be established for
>>> some calls. It shouldn't be a codec problem as call parties are configured
>>> with common codecs, FS is bond to local interface which is running with no
>>> errors. But I have no idea what "local host is missing" in my case so I'll
>>> be appreciated for any hint.
>>>
>>> FreeSWITCH Version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git
>>> a98a958 2018-09-26 17:55:25Z 64bit)
>>> CentOS Linux release 7.5.1804 (Core)
>>>
>>> Best,
>>> Marc
>>> _________________________________________________________________________
>>> 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>
>> _________________________________________________________________________
>> 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
>
> _________________________________________________________________________
> 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 andrew.keil at visytel.com  Sun Oct 14 22:58:21 2018
From: andrew.keil at visytel.com (Andrew Keil)
Date: Sun, 14 Oct 2018 22:58:21 +0000
Subject: [Freeswitch-users] Re- DTMF 2833 issue
In-Reply-To: <CAFiqYunKvWbiWOdyGiW_yeLLbSWGKLBM6DXNDBQuv8HuBjK-TQ@mail.gmail.com>
References: <ME1PR01MB1538D19D7BEB2A2C925A4983F3E00@ME1PR01MB1538.ausprd01.prod.outlook.com>
 <1BC124F9-C601-43C2-92D0-BF68194198BB@jerris.com>
 <CAFiqYunKvWbiWOdyGiW_yeLLbSWGKLBM6DXNDBQuv8HuBjK-TQ@mail.gmail.com>
Message-ID: <ME1PR01MB1538D8E058358B6C9849688CF3FC0@ME1PR01MB1538.ausprd01.prod.outlook.com>

Steven & Mike,

Much appreciated.  I will give this a try.

Andrew

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Steven Ayre
Sent: Saturday, 13 October 2018 11:50 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Re- DTMF 2833 issue

Try something like this:


  <extension name="inband-dtmf" continue="true">

    <condition field="${switch_r_sdp}" expression="a=rtpmap:\d+

telephone-event\/\d+">

      <action application="start_dtmf"/>

    </condition>

  </extension>


I think the wiki used to have an example, but I don't see it on confluence.

On Sat, 13 Oct 2018 at 00:15, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
There are vars with the full SDP, you can do a regex condition.  Check the info app to see all the vars you have at your disposal.  There may be a var for the te as well to show the pt.

Mike


On Oct 10, 2018, at 6:53 PM, Andrew Keil <andrew.keil at visytel.com<mailto:andrew.keil at visytel.com>> wrote:

To FreeSWITCH Users,

I have just noticed when making calls internationally to the UK from Australia that the following SDP sometimes happens when the call lands on FreeSWITCH:

d5f14e47-2156-4236-be16-4eef1b7e9d3d 2018-10-04 06:59:28.819816 [DEBUG] sofia.c:7301 Remote SDP:
d5f14e47-2156-4236-be16-4eef1b7e9d3d v=0
d5f14e47-2156-4236-be16-4eef1b7e9d3d o=sbc-uk-mr-dh05a 451499 139728 IN IP4 XXX.XXX.XXX.XXX
d5f14e47-2156-4236-be16-4eef1b7e9d3d s=sip call
d5f14e47-2156-4236-be16-4eef1b7e9d3d c=IN IP4 XXX.XXX.XXX.XXX
d5f14e47-2156-4236-be16-4eef1b7e9d3d t=0 0
d5f14e47-2156-4236-be16-4eef1b7e9d3d m=audio 62754 RTP/AVP 8
d5f14e47-2156-4236-be16-4eef1b7e9d3d a=rtpmap:8 PCMA/8000
d5f14e47-2156-4236-be16-4eef1b7e9d3d a=ptime:20

This then creates the following message around call answer:

d5f14e47-2156-4236-be16-4eef1b7e9d3d 2018-10-04 06:59:28.839799 [DEBUG] switch_core_media.c:5766 No 2833 in SDP. Liberal DTMF mode adding 101 as telephone-event.

From this point on DTMF is NOT detected by the FreeSWITCH IVR service.  I assume I would then need to use mod_dptools: start_dtmf to detect the DTMF tones within the channel.  Which I am yet to test.

However my main question is whether there is a channel variable that would enable me to check whether 2833 support is not inside the SDP?

Regards,

Andrew Keil


_________________________________________________________________________
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 gregor at infomedia.si  Mon Oct 15 09:52:20 2018
From: gregor at infomedia.si (Gregor Nanger)
Date: Mon, 15 Oct 2018 11:52:20 +0200
Subject: [Freeswitch-users] [Missing local host] in proxy media mode
In-Reply-To: <187AA53B-CE2E-416D-996E-AD6D8F12043B@jerris.com>
References: <CAMBkyc1ETujbQLgYEyv+-URcV2OwCuWW0Dh2k+b6U2XjUkAMig@mail.gmail.com>
 <CAEJMVkB9sggz6y4RfPvWTYFuxOEpkvvM2Htmsnav8nfxRZzRog@mail.gmail.com>
 <187AA53B-CE2E-416D-996E-AD6D8F12043B@jerris.com>
Message-ID: <CAC7ttFGWWWicu3-uYT7pNaqkBotfd5GwdiB9eSLTe-VJd3bAWg@mail.gmail.com>

Michael, can you please explain your comment a little bit more:

"There is one exception to this rule, and that is when you are trying to
pass codecs that freeswitch does not know about, and even that case has a
better workaround than to use proxy media"

We are using proxy, because FS on windows can't support g729 and
transcoding. What is better workaround than use proxy media?

Gregor

V V ned., 14. okt. 2018 ob 23:42 je oseba Michael Jerris <mike at jerris.com>
napisala:

> I’ll expand on Brian’s comment.  Proxy media should NEVER be used.  There
> is one exception to this rule, and that is when you are trying to pass
> codecs that freeswitch does not know about, and even that case has a better
> workaround than to use proxy media.  If you are using proxy media you
> should stop, if you are not sure how to do so you should discuss on the
> mailing list.  Proxy media mode is NOT the way to avoid transcoding, that
> can be done properly without this setting and this setting should never be
> used.
>
> Mike
>
> On Oct 12, 2018, at 5:07 PM, Brian West <brian at freeswitch.com> wrote:
>
> Could you elaborate on why you're using Proxy Media for?
> /b
>
> On Fri, Oct 12, 2018 at 5:37 PM Marc Quore <marquore at gmail.com> wrote:
>
>> Hi, All
>>
>> I've got FreeSwitch 1.8.2 running in media proxy mode:
>>
>> <param name="inbound-proxy-media" value="true"/>
>> <param name="inbound-late-negotiation" value="true"/>
>>
>> Dialplan is configured to bridge incoming calls to remote gateway and
>> most of calls completing successfully, however some of them fail with
>> INCOMPATIBLE_DESTINATION cause. For failed calls everything goes normally
>> until bridged connection gets early media from callee:
>>
>> [NOTICE] sofia.c:7304 Pre-Answer sofia/external/2222 at 2.2.2.2!
>> [INFO] switch_ivr_originate.c:3747 Sending early media
>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PCMU/0
>> 0 ms 960 samples 384000 bits 1 channels
>> [DEBUG] switch_core_codec.c:111 sofia/internal/1111 at 1.1.1.1 Original
>> read codec set to PCMU:0
>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
>> 1111 at 1.1.1.1] 1.1.1.1:18792->1.1.1.1:18792 codec: 0 ms: 20
>> [ERR] switch_core_media.c:9510 AUDIO RTP REPORTS ERROR: *[Missing local
>> host]*
>> [NOTICE] switch_core_media.c:9511 Hangup sofia/internal/1111 at 1.1.1.1
>> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
>>
>> Here's a dump for this call:
>>
>> <image.png>
>>
>> Healthy call on the same machine:
>>
>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/vpn/2222 at 2.2.2.2 PROXY/0
>> 0 ms 160 samples 0 bits 1 channels
>> [DEBUG] switch_core_codec.c:111 sofia/internal/2222 at 2.2.2.2 Original
>> read codec set to PROXY:0
>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
>> 2222 at 2.2.2.2] 2.2.2.2:60818->2.2.2.2:60818 codec: 0 ms: 20
>>
>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PROXY/0
>> 0 ms 160 samples 0 bits 1 channels
>> [DEBUG] switch_core_codec.c:111 sofia/external/1111 at 1.1.1.1 Original
>> read codec set to PROXY:0
>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/external/
>> 1111 at 1.1.1.1] 1.1.1.1:31412->1.1.1.1:31412 codec: 0 ms: 20
>>
>> So it looks like proxified media connection cannot be established for
>> some calls. It shouldn't be a codec problem as call parties are configured
>> with common codecs, FS is bond to local interface which is running with no
>> errors. But I have no idea what "local host is missing" in my case so I'll
>> be appreciated for any hint.
>>
>> FreeSWITCH Version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git
>> a98a958 2018-09-26 17:55:25Z 64bit)
>> CentOS Linux release 7.5.1804 (Core)
>>
>> Best,
>> Marc
>> _________________________________________________________________________
>> 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>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
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> https://freeswitch.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
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> https://freeswitch.com



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
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From mickael at winlux.fr  Mon Oct 15 09:55:47 2018
From: mickael at winlux.fr (Mickael Hubert)
Date: Mon, 15 Oct 2018 11:55:47 +0200
Subject: [Freeswitch-users] Multi gateways in profile
In-Reply-To: <C2ADA4BE-996C-4995-8D75-92C7F5BDA0CC@vallimamod.org>
References: <CA+fRVSUkFuJRgkUAQCxr=3p6fj-joQR_yQD=c=YD4WRu8r=P-g@mail.gmail.com>
 <f7699f6b-56aa-4f82-8841-1c1258cef39b@gmail.com>
 <CA+fRVSXA5-bzKqNgmrRaq=FFBqe06iPYYgv0rc48chwY9_PZ1A@mail.gmail.com>
 <C2ADA4BE-996C-4995-8D75-92C7F5BDA0CC@vallimamod.org>
Message-ID: <CA+fRVSVv+yz=UYpDcDfaDC5UhQQ1UxgYutt200CuXR+BjEEE9A@mail.gmail.com>

Thanks Valli ;) and thanks all too !
it works like a charm !

I can easily mix filter by gateway and by IP ;)

++

Le dim. 14 oct. 2018 à 23:34, Vallimamod Abdullah <vma at vallimamod.org> a
écrit :

> Hi,
>
> For ip addr matching: <condition field="network_addr"
> expression="^192\.168\.1\.1$"/>
>
> Alternatively, If you want some variables populated on incoming call from
> a given IP the same way as with registered gateways, you can define a
> directory user entry (with "cidr" param to allow ip-auth) and define the
> your variables there. For example:
>
>     <user id="GW2" cidr="2.2.2.2/32">
>         <variables>
>             <variable name="accountcode" value="GW2"/>
>             <variable name="call_limit" value="10"/>
>             <variable name="max_ringtime" value="45"/>
>         </variables>
>     </user>
>
> Hope this helps!
>
>
> Best Regards,
> --
> Vallimamod Abdullah
> SIP Solutions
> vma at sip.solutions
> linkedin.com/in/vallimamod
> .
>
> > On 13 Oct 2018, at 22:11, Mickael Hubert <mickael at winlux.fr> wrote:
> >
> > Hi
> > Thanks a lot !
> >
> > I think the best way (maybe only way) it's to match with from IP.
> > It's sure when inbound call arrives from GW2, any variables in gateway
> configuration aren't populate.
> >
> > But how Can I do math with IP please ?
> >
> > ++
> >
> >
> > Le sam. 13 oct. 2018 18:26, Srigo Kanapathipillai <ksrigo at gmail.com> a
> écrit :
> > Hi Mickael,
> >
> > <gateway>
> > ..params...
> > <variables>
> > <variable name="inbound_var_name" value="this is inbound"
> direction="inbound"/>
> > <variable name="outbound_var_name" value="this is outbound"
> direction="outbound"/>
> > <variable name="both_var_name" value="this on any direction"/>
> > </variables>
> > </gateway>
> >
> > Did you try to set variables on your gateway and use it in your dialplan?
> >
> > Srigo Kana
> > On 13 Oct 2018, at 00:31, Mickael Hubert <mickael at winlux.fr> wrote:
> > Hi guys,
> > I want to identify source of inbounds calls.
> > In my external profile I have two gateways:
> >
> > - One with REGISTER method
> > - One without REGISTER method (IP to IP), only iptables filtering
> >
> > <gateways>
> >       <gateway name="GW1">
> >         <param name="from-domain" value="**********"/>
> >         <param name="caller-id-in-from" value="true"/>
> >         <param name="proxy" value="1.1.1.1"/>
> >         <param name="register" value="true"/>
> >         <param name="username" value="**************"/>
> >         <param name="password" value="****************"/>
> >         <param name="expire-seconds" value="3600"/>
> >         <param name="retry-seconds" value="30"/>
> >         <param name="ping" value="25"/>
> >       </gateway>
> >       <gateway name="GW2">
> >         <param name="from-domain" value="*********"/>
> >         <param name="caller-id-in-from" value="true"/>
> >         <param name="proxy" value="2.2.2.2"/>
> >         <param name="register" value="false"/>
> >         <param name="retry-seconds" value="30"/>
> >         <param name="ping" value="25"/>
> >       </gateway>
> >     </gateways>
> >
> > In my dialplan I have this conf:
> >
> > <condition field="${sip_gateway}"  expression="GW1" break="never">
> >    <action application="set" data="number_dest=${destination_number}" />
> >    <action application="execute_extension" data="to-private XML
> from-gw1"/>
> > </condition>
> > <condition field="${sip_gateway}"  expression="GW2" break="never">
> >    <action application="set" data="number_dest=${destination_number}" />
> >    <action application="execute_extension" data="to-private XML
> from-gw2"/>
> > </condition>
> >
> > When a call arrives throught GW1 (REGISTERED), ${sip_gateway} is
> populate with "GW1" perfect !. But when another call arrives throught GW2
> (not REGISTERED), ${sip_gateway} is empty.
> >
> > Please, how can I identify the source gateway where calls arrive ? maybe
> it's not a good way to do that ?
> >
> > thanks in advance
> >
> > 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
> > _________________________________________________________________________
> > 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
> > _________________________________________________________________________
> > 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
>
>
> _________________________________________________________________________
> 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
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From vma at vallimamod.org  Mon Oct 15 12:26:10 2018
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Mon, 15 Oct 2018 14:26:10 +0200
Subject: [Freeswitch-users] Sometimes "hanging" channels
In-Reply-To: <1E3CBAFE-0B7B-445C-AAF2-EF4C074392B7@gmx.net>
References: <87D934A1-4A37-4709-93B5-96FA0659ABBE@gmx.net>
 <CAFiqYun3i3NDvqR4pkvfDCquQNMALXXAA-fp2EPm6bXx0VuVmg@mail.gmail.com>
 <1E3CBAFE-0B7B-445C-AAF2-EF4C074392B7@gmx.net>
Message-ID: <C27A69BA-D5E4-4B30-8053-E53C34E585FA@vallimamod.org>

Hi,

The message "Locked, Waiting on external entities" is not alarming by itself as it is always logged on hangup. But it should be followed by:

[NOTICE] switch_core_session.c:1683 Session XXX (sofia/internal/YYY) Ended
[NOTICE] switch_core_session.c:1687 Close Channel sofia/internal/YYY [CS_DESTROY]

If it is not the case then, you have an app or a module that is holding a lock on the session and preventing its release.

And this happens after the reporting state i.e. after the hangup_complete event and CDR posting. So there may be other causes for your missing events.



Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sip.solutions
linkedin.com/in/vallimamod
.


> On 10 Oct 2018, at 15:54, Markus Bönke <mbodbg at gmx.net> wrote:
> 
> So far I found out that the channel was kept open, because our ESL application never received a CHANNEL_HANGUP and CHANNEL_HANGUP_COMPLETE event. I’m running an ESL trace with tshark now. If it happens again I can see if those events are really not send by FS or they get lost in our app. If it turns out that FS sometimes is not sending the events we will upgrade to 1.8.2.
> 
> Thanks and regards
> 
> Markus
> 
> 
>> Am 10.10.2018 um 14:03 schrieb Steven Ayre <steveayre at gmail.com>:
>> 
>> If you take a gcore it'll generate a core dump. You could then dig into it with gdb to find the thread for that channel and see what lock it's waiting on.
>> However the gcore will pause the process for a while, so it will impact on any other calls on that box.
>> 
>> Have you tried reproducing it on 1.8?
>> 
>> On Tue, 9 Oct 2018 at 20:36, Markus Bönke <mbodbg at gmx.net> wrote:
>> Hello All,
>> 
>> we are running freeswitch 1.6.20, calls are controlled via ESL, CDRs are written with mod_xml_cdrl. Sometimes we see „hanging“ channels. In such a case the CDR via mod_xml_cdr is written and the last log entry for such a call is "Locked, Waiting on external entities“.
>> ...
>> freeswitch.log.2018-10-09-10-19-07.1:db2908e3-c06b-4293-88b1-e18cffb66263 2018-10-09 10:18:15.197345 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/anonymous at anonymous.invalid) State Change CS_REPORTING -> CS_DESTROY
>> freeswitch.log.2018-10-09-10-19-07.1:db2908e3-c06b-4293-88b1-e18cffb66263 2018-10-09 10:18:15.197345 [DEBUG] switch_core_session.c:1665 Session 166111 (sofia/internal/anonymous at anonymous.invalid) Locked, Waiting on external entities
>> ...
>> How can I proceed to further analyze the problem ? In the last log line I also see the session number (Session 166111) - is there a way to find out on which external entity it is waiting?
>> 
>> Thanks and regards
>> 
>> Markus 
>> _________________________________________________________________________
>> 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
>> _________________________________________________________________________
>> 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
> 
> _________________________________________________________________________
> 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 vma at vallimamod.org  Mon Oct 15 12:58:14 2018
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Mon, 15 Oct 2018 14:58:14 +0200
Subject: [Freeswitch-users] How to update the caller id when bridging?
In-Reply-To: <20181006020400.3324ed18@kappa.digital-domain.net>
References: <20181006020400.3324ed18@kappa.digital-domain.net>
Message-ID: <A9F49B6F-79D3-4444-A186-EE33BA36874A@vallimamod.org>

Hi,

Have a look at sofia_send_callee_id() and sofia_update_callee_id() funcs in sofia.c.
I think they are doing what you want to achieve.



Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sip.solutions
linkedin.com/in/vallimamod
.


> On 6 Oct 2018, at 03:04, Andrew Clayton <andrew at zeta.digital-domain.net> wrote:
> 
> So this is a question based around the FreeSWITCH C API.
> 
> I have two scenarios involving two endpoints, A & B.
> 
> Scenario 1
> 
> FS calls B using switch_ivr_originate(), it then calls A (say a
> Linphone client) and creates a new switch_caller_profile_t for it using
> switch_caller_profile_new() and sets the ->caller_id_number field to e.g
> 1234 to indicate a call from B and then does a switch_ivr_originate().
> 
> A rings and Linphone shows the call coming from 1234 at ... OK, that's all
> fine. The calls are bridged together with switch_ivr_uuid_bridge(A, B)
> 
> Scenario 2
> 
> FS calls A (again lets say a Linphone client) using
> switch_ivr_originate(), A rings and shows the call coming from
> 0000... at ..., that's fine, I didn't set any CID.
> 
> Now A is parked and is off-hook.
> 
> FS then calls B, A & B are then bridged together with
> switch_ivr_uuid_bridge(A, B), now of course Linphone still shows the
> call as from  0000... at ... I'd like it show for example 1234 at ... like
> above.
> 
> Now I've tried setting numerous fields in the switch_caller_profile_t
> structure for A just before doing the bridge, but to no avail.
> 
> So does what I'm trying to do make sense?
> 
> Is it even possible?
> 
> Cheers,
> Andrew
> 
> _________________________________________________________________________
> 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



Bien cordialement,
-- 
Vallimamod Abdullah
SIP Solutions
Conseil & Développement de plateformes et services VOIP
vma at sip.solutions
+33 6 62 60 68 97
linkedin.com/in/vallimamod
.



From andrew at zeta.digital-domain.net  Mon Oct 15 14:21:50 2018
From: andrew at zeta.digital-domain.net (Andrew Clayton)
Date: Mon, 15 Oct 2018 15:21:50 +0100
Subject: [Freeswitch-users] How to update the caller id when bridging?
In-Reply-To: <A9F49B6F-79D3-4444-A186-EE33BA36874A@vallimamod.org>
References: <20181006020400.3324ed18@kappa.digital-domain.net>
 <A9F49B6F-79D3-4444-A186-EE33BA36874A@vallimamod.org>
Message-ID: <20181015152150.0ddb31a8@kappa.digital-domain.net>

On Mon, 15 Oct 2018 14:58:14 +0200
Vallimamod Abdullah <vma at vallimamod.org> wrote:

> Hi,
> 
> Have a look at sofia_send_callee_id() and sofia_update_callee_id() funcs in sofia.c.
> I think they are doing what you want to achieve.

Thanks for the pointer, I'll take a look.

Cheers,
Andrew



From covici at ccs.covici.com  Mon Oct 15 14:55:43 2018
From: covici at ccs.covici.com (John Covici)
Date: Mon, 15 Oct 2018 10:55:43 -0400
Subject: [Freeswitch-users] How to update the caller id when bridging?
In-Reply-To: <A9F49B6F-79D3-4444-A186-EE33BA36874A@vallimamod.org>
References: <20181006020400.3324ed18@kappa.digital-domain.net>
 <A9F49B6F-79D3-4444-A186-EE33BA36874A@vallimamod.org>
Message-ID: <m37eijntsg.wl-covici@ccs.covici.com>

Anyway to do this in dialplan?

On Mon, 15 Oct 2018 08:58:14 -0400,
Vallimamod Abdullah wrote:
> 
> Hi,
> 
> Have a look at sofia_send_callee_id() and sofia_update_callee_id() funcs in sofia.c.
> I think they are doing what you want to achieve.
> 
> 
> 
> Best Regards,
> -- 
> Vallimamod Abdullah
> SIP Solutions
> vma at sip.solutions
> linkedin.com/in/vallimamod
> .
> 
> 
> > On 6 Oct 2018, at 03:04, Andrew Clayton <andrew at zeta.digital-domain.net> wrote:
> > 
> > So this is a question based around the FreeSWITCH C API.
> > 
> > I have two scenarios involving two endpoints, A & B.
> > 
> > Scenario 1
> > 
> > FS calls B using switch_ivr_originate(), it then calls A (say a
> > Linphone client) and creates a new switch_caller_profile_t for it using
> > switch_caller_profile_new() and sets the ->caller_id_number field to e.g
> > 1234 to indicate a call from B and then does a switch_ivr_originate().
> > 
> > A rings and Linphone shows the call coming from 1234 at ... OK, that's all
> > fine. The calls are bridged together with switch_ivr_uuid_bridge(A, B)
> > 
> > Scenario 2
> > 
> > FS calls A (again lets say a Linphone client) using
> > switch_ivr_originate(), A rings and shows the call coming from
> > 0000... at ..., that's fine, I didn't set any CID.
> > 
> > Now A is parked and is off-hook.
> > 
> > FS then calls B, A & B are then bridged together with
> > switch_ivr_uuid_bridge(A, B), now of course Linphone still shows the
> > call as from  0000... at ... I'd like it show for example 1234 at ... like
> > above.
> > 
> > Now I've tried setting numerous fields in the switch_caller_profile_t
> > structure for A just before doing the bridge, but to no avail.
> > 
> > So does what I'm trying to do make sense?
> > 
> > Is it even possible?
> > 
> > Cheers,
> > Andrew
> > 
> > _________________________________________________________________________
> > 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
> 
> 
> 
> Bien cordialement,
> -- 
> Vallimamod Abdullah
> SIP Solutions
> Conseil & Développement de plateformes et services VOIP
> vma at sip.solutions
> +33 6 62 60 68 97
> linkedin.com/in/vallimamod
> .
> 
> 
> _________________________________________________________________________
> 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 wsimon at stratusvideo.com  Mon Oct 15 15:04:00 2018
From: wsimon at stratusvideo.com (William Simon)
Date: Mon, 15 Oct 2018 15:04:00 +0000
Subject: [Freeswitch-users] Accepting an "optional" SRTP offer (crypto
 in RTP/AVP) and establishing SRTP
In-Reply-To: <C85C798A-BE0B-4EE2-8300-6051CE2103FC@jerris.com>
References: <2AE1EF96-BD07-4E78-A638-D9D53FF8FF1E@stratusvideo.com>
 <C85C798A-BE0B-4EE2-8300-6051CE2103FC@jerris.com>
Message-ID: <BC79BB9D-4CC4-4E8E-84D0-11483A3975C7@stratusvideo.com>

We are on 1.6.20.

> On Oct 12, 2018, at 6:00 PM, Michael Jerris <mike at jerris.com> wrote:
>
> Are you trying this on 1.8.2 version?
>
>> On Oct 5, 2018, at 10:27 AM, William Simon <wsimon at stratusvideo.com> wrote:
>>
>> We are working with an endpoint that offers optional SRTP in the non-RFC-compliant way of an a=crypto attribute within RTP/AVP.
>>
>> I have told FreeSWITCH to allow this with rtp_allow_crypto_in_avp=true at the right place in the dialplan.
>>
>> Immediately after that in the dialplan I have to reject SRTP by using rtp_secure_media=forbidden, otherwise the call setup still fails. By setting the value to forbidden, the call does proceed unencrypted.
>>
>> We want FreeSWITCH to proceed with media encryption. Setting rtp_secure_media to any other value results in FreeSWITCH rejecting the offer like this:
>>
>> v=0
>> o=FreeSWITCH 1538660754 1538660755 IN IP4 192.168.100.104
>> s=FreeSWITCH
>> c=IN IP4 192.168.100.104
>> t=0 0
>> m=audio 0 RTP/AVP 19
>> m=video 0 RTP/AVP 19
>>
>> Is there anything else I can do to force SRTP in the answer?
>>
>>
>
>
> _________________________________________________________________________
> 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 information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”

From wsimon at stratusvideo.com  Mon Oct 15 15:05:09 2018
From: wsimon at stratusvideo.com (William Simon)
Date: Mon, 15 Oct 2018 15:05:09 +0000
Subject: [Freeswitch-users] Accepting an "optional" SRTP offer (crypto
 in RTP/AVP) and establishing SRTP
In-Reply-To: <CAGJLwFjP_6vUsqMPYdme9SZB-6YE+yUwjx4Gw8B02eFD1kdS-g@mail.gmail.com>
References: <2AE1EF96-BD07-4E78-A638-D9D53FF8FF1E@stratusvideo.com>
 <5821E934-2105-4B27-A925-74605B34B769@stratusvideo.com>
 <CAGJLwFjP_6vUsqMPYdme9SZB-6YE+yUwjx4Gw8B02eFD1kdS-g@mail.gmail.com>
Message-ID: <D734FCE0-636C-4D11-965B-2068451FDF9E@stratusvideo.com>

Unfortunately, that doesn't work. It gives the result I described: refusal of all media with m=audio 0 and m=video 0.


On Oct 10, 2018, at 9:56 PM, Alexey Sibyakin <alex at freeswitch.com<mailto:alex at freeswitch.com>> wrote:

Take a close look to default.xml of vanilla dialplan. There are some examples of SDP parsing here, you can use them in condition to detect your special case. To enforce SRTP you just need to set rtp_secure_media. Don't forget to reread documentation on the last one:

https://freeswitch.org/confluence/display/FREESWITCH/rtp_secure_media

Alex

On Thu, Oct 11, 2018 at 12:54 AM William Simon <wsimon at stratusvideo.com<mailto:wsimon at stratusvideo.com>> wrote:
Can anyone offer insight into this matter?

Endpoint offers RTP/AVP with crypto. We want Freeswitch to respond to the RTP/AVP and agree to the crypto and do SRTP.


> On Oct 5, 2018, at 10:27 AM, William Simon <wsimon at stratusvideo.com<mailto:wsimon at stratusvideo.com>> wrote:
>
> We are working with an endpoint that offers optional SRTP in the non-RFC-compliant way of an a=crypto attribute within RTP/AVP.
>
> I have told FreeSWITCH to allow this with rtp_allow_crypto_in_avp=true at the right place in the dialplan.
>
> Immediately after that in the dialplan I have to reject SRTP by using rtp_secure_media=forbidden, otherwise the call setup still fails. By setting the value to forbidden, the call does proceed unencrypted.
>
> We want FreeSWITCH to proceed with media encryption. Setting rtp_secure_media to any other value results in FreeSWITCH rejecting the offer like this:
>
> v=0
> o=FreeSWITCH 1538660754 1538660755 IN IP4 192.168.100.104
> s=FreeSWITCH
> c=IN IP4 192.168.100.104
> t=0 0
> m=audio 0 RTP/AVP 19
> m=video 0 RTP/AVP 19
>
> Is there anything else I can do to force SRTP in the answer?
>



“The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”
_________________________________________________________________________
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>
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com<https://freeswitch.com/>


--
Alex Sibyakin | Support Engineer
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: alex at freeswitch.com<mailto:alex at freeswitch.com>
Website: https://www.FreeSWITCH.com<https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com<mailto:sales at freeswitch.com> for details.

_________________________________________________________________________
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
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com



“The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”
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From ksrigo at gmail.com  Mon Oct 15 16:53:33 2018
From: ksrigo at gmail.com (Srigo Kanapathipillai)
Date: Mon, 15 Oct 2018 18:53:33 +0200
Subject: [Freeswitch-users] Multi gateways in profile
In-Reply-To: <CA+fRVSXA5-bzKqNgmrRaq=FFBqe06iPYYgv0rc48chwY9_PZ1A@mail.gmail.com>
References: <CA+fRVSUkFuJRgkUAQCxr=3p6fj-joQR_yQD=c=YD4WRu8r=P-g@mail.gmail.com>
 <f7699f6b-56aa-4f82-8841-1c1258cef39b@gmail.com>
 <CA+fRVSXA5-bzKqNgmrRaq=FFBqe06iPYYgv0rc48chwY9_PZ1A@mail.gmail.com>
Message-ID: <6416dd6b-6cee-4feb-bfe2-f55a80337ae6@gmail.com>

Why you can't set the same variable in both gateways profile:

In gw1:

<variable name="incoming_gw" value="gw1"/>  


In gw2:

<variable name="incoming_gw" value="gw2"/>


And check in your dialplan if incoming_gw is set to gw1 OR gw2?


⁣Srigo Kana​

On 14 Oct 2018, 18:33, at 18:33, Mickael Hubert <mickael at winlux.fr> wrote:
>Hi
>Thanks a lot !
>
>I think the best way (maybe only way) it's to match with from IP.
>It's sure when inbound call arrives from GW2, any variables in gateway
>configuration aren't populate.
>
>But how Can I do math with IP please ?
>
>++
>
>
>Le sam. 13 oct. 2018 18:26, Srigo Kanapathipillai <ksrigo at gmail.com> a
>écrit :
>
>> Hi Mickael,
>>
>> <gateway>
>> ..params...
>> <variables>
>> <variable name="inbound_var_name" value="this is inbound"
>> direction="inbound"/>
>> <variable name="outbound_var_name" value="this is outbound"
>> direction="outbound"/>
>> <variable name="both_var_name" value="this on any direction"/>
>> </variables>
>> </gateway>
>>
>> Did you try to set variables on your gateway and use it in your
>dialplan?
>>
>> Srigo Kana
>> On 13 Oct 2018, at 00:31, Mickael Hubert <mickael at winlux.fr> wrote:
>>>
>>> Hi guys,
>>> I want to identify source of inbounds calls.
>>> In my external profile I have two gateways:
>>>
>>> - One with REGISTER method
>>> - One without REGISTER method (IP to IP), only iptables filtering
>>>
>>> <gateways>
>>>       <gateway name="GW1">
>>>         <param name="from-domain" value="**********"/>
>>>         <param name="caller-id-in-from" value="true"/>
>>>         <param name="proxy" value="1.1.1.1"/>
>>>         <param name="register" value="true"/>
>>>         <param name="username" value="**************"/>
>>>         <param name="password" value="****************"/>
>>>         <param name="expire-seconds" value="3600"/>
>>>         <param name="retry-seconds" value="30"/>
>>>         <param name="ping" value="25"/>
>>>       </gateway>
>>>       <gateway name="GW2">
>>>         <param name="from-domain" value="*********"/>
>>>         <param name="caller-id-in-from" value="true"/>
>>>         <param name="proxy" value="2.2.2.2"/>
>>>         <param name="register" value="false"/>
>>>         <param name="retry-seconds" value="30"/>
>>>         <param name="ping" value="25"/>
>>>       </gateway>
>>>     </gateways>
>>>
>>> In my dialplan I have this conf:
>>>
>>> <condition field="${sip_gateway}"  expression="GW1" break="never">
>>>    <action application="set"
>data="number_dest=${destination_number}" />
>>>    <action application="execute_extension" data="to-private XML
>>> from-gw1"/>
>>> </condition>
>>> <condition field="${sip_gateway}"  expression="GW2" break="never">
>>>    <action application="set"
>data="number_dest=${destination_number}" />
>>>    <action application="execute_extension" data="to-private XML
>>> from-gw2"/>
>>> </condition>
>>>
>>> When a call arrives throught GW1 (REGISTERED), ${sip_gateway} is
>populate
>>> with "GW1" perfect !. But when another call arrives throught GW2
>(not
>>> REGISTERED), ${sip_gateway} is empty.
>>>
>>> Please, how can I identify the source gateway where calls arrive ?
>maybe
>>> it's not a good way to do that ?
>>>
>>> thanks in advance
>>>
>>> ------------------------------
>>>
>>> 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
>>>
>>>
>_________________________________________________________________________
>> 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
>
>
>------------------------------------------------------------------------
>
>_________________________________________________________________________
>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
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From brian at freeswitch.com  Mon Oct 15 17:31:33 2018
From: brian at freeswitch.com (Brian West)
Date: Mon, 15 Oct 2018 12:31:33 -0500
Subject: [Freeswitch-users] How to update the caller id when bridging?
In-Reply-To: <m37eijntsg.wl-covici@ccs.covici.com>
References: <20181006020400.3324ed18@kappa.digital-domain.net>
 <A9F49B6F-79D3-4444-A186-EE33BA36874A@vallimamod.org>
 <m37eijntsg.wl-covici@ccs.covici.com>
Message-ID: <CAEJMVkAFx3sQRdGzDVcG+0j1dCQXbafNoSUf_bD7wpSNag1pQQ@mail.gmail.com>

The updates won't get sent unless the user agents are on the safe list, so
i'd review the code.

/b


On Mon, Oct 15, 2018 at 11:51 AM John Covici <covici at ccs.covici.com> wrote:

> Anyway to do this in dialplan?
>
> On Mon, 15 Oct 2018 08:58:14 -0400,
> Vallimamod Abdullah wrote:
> >
> > Hi,
> >
> > Have a look at sofia_send_callee_id() and sofia_update_callee_id() funcs
> in sofia.c.
> > I think they are doing what you want to achieve.
> >
> >
> >
> > Best Regards,
> > --
> > Vallimamod Abdullah
> > SIP Solutions
> > vma at sip.solutions
> > linkedin.com/in/vallimamod
> > .
> >
> >
> > > On 6 Oct 2018, at 03:04, Andrew Clayton <
> andrew at zeta.digital-domain.net> wrote:
> > >
> > > So this is a question based around the FreeSWITCH C API.
> > >
> > > I have two scenarios involving two endpoints, A & B.
> > >
> > > Scenario 1
> > >
> > > FS calls B using switch_ivr_originate(), it then calls A (say a
> > > Linphone client) and creates a new switch_caller_profile_t for it using
> > > switch_caller_profile_new() and sets the ->caller_id_number field to
> e.g
> > > 1234 to indicate a call from B and then does a switch_ivr_originate().
> > >
> > > A rings and Linphone shows the call coming from 1234 at ... OK, that's
> all
> > > fine. The calls are bridged together with switch_ivr_uuid_bridge(A, B)
> > >
> > > Scenario 2
> > >
> > > FS calls A (again lets say a Linphone client) using
> > > switch_ivr_originate(), A rings and shows the call coming from
> > > 0000... at ..., that's fine, I didn't set any CID.
> > >
> > > Now A is parked and is off-hook.
> > >
> > > FS then calls B, A & B are then bridged together with
> > > switch_ivr_uuid_bridge(A, B), now of course Linphone still shows the
> > > call as from  0000... at ... I'd like it show for example 1234 at ... like
> > > above.
> > >
> > > Now I've tried setting numerous fields in the switch_caller_profile_t
> > > structure for A just before doing the bridge, but to no avail.
> > >
> > > So does what I'm trying to do make sense?
> > >
> > > Is it even possible?
> > >
> > > Cheers,
> > > Andrew
> > >
> > >
> _________________________________________________________________________
> > > 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
> >
> >
> >
> > Bien cordialement,
> > --
> > Vallimamod Abdullah
> > SIP Solutions
> > Conseil & Développement de plateformes et services VOIP
> > vma at sip.solutions
> > +33 6 62 60 68 97
> > linkedin.com/in/vallimamod
> > .
> >
> >
> > _________________________________________________________________________
> > 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
>
> _________________________________________________________________________
> 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:
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From steveayre at gmail.com  Mon Oct 15 20:58:36 2018
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 15 Oct 2018 21:58:36 +0100
Subject: [Freeswitch-users] [Missing local host] in proxy media mode
In-Reply-To: <CAC7ttFGWWWicu3-uYT7pNaqkBotfd5GwdiB9eSLTe-VJd3bAWg@mail.gmail.com>
References: <CAMBkyc1ETujbQLgYEyv+-URcV2OwCuWW0Dh2k+b6U2XjUkAMig@mail.gmail.com>
 <CAEJMVkB9sggz6y4RfPvWTYFuxOEpkvvM2Htmsnav8nfxRZzRog@mail.gmail.com>
 <187AA53B-CE2E-416D-996E-AD6D8F12043B@jerris.com>
 <CAC7ttFGWWWicu3-uYT7pNaqkBotfd5GwdiB9eSLTe-VJd3bAWg@mail.gmail.com>
Message-ID: <CAFiqYu=G2GWb4yLV1Nib6v3RBeVirteHc_8wiM9kGhTQr8fcUA@mail.gmail.com>

mod_g729 will allow passthrough g729 on windows, which is all proxy media
allows. only mod_com_g729 is limited to linux.

On Mon, 15 Oct 2018 at 13:22, Gregor Nanger <gregor at infomedia.si> wrote:

> Michael, can you please explain your comment a little bit more:
>
> "There is one exception to this rule, and that is when you are trying to
> pass codecs that freeswitch does not know about, and even that case has a
> better workaround than to use proxy media"
>
> We are using proxy, because FS on windows can't support g729 and
> transcoding. What is better workaround than use proxy media?
>
> Gregor
>
> V V ned., 14. okt. 2018 ob 23:42 je oseba Michael Jerris <mike at jerris.com>
> napisala:
>
>> I’ll expand on Brian’s comment.  Proxy media should NEVER be used.  There
>> is one exception to this rule, and that is when you are trying to pass
>> codecs that freeswitch does not know about, and even that case has a better
>> workaround than to use proxy media.  If you are using proxy media you
>> should stop, if you are not sure how to do so you should discuss on the
>> mailing list.  Proxy media mode is NOT the way to avoid transcoding, that
>> can be done properly without this setting and this setting should never be
>> used.
>>
>> Mike
>>
>> On Oct 12, 2018, at 5:07 PM, Brian West <brian at freeswitch.com> wrote:
>>
>> Could you elaborate on why you're using Proxy Media for?
>> /b
>>
>> On Fri, Oct 12, 2018 at 5:37 PM Marc Quore <marquore at gmail.com> wrote:
>>
>>> Hi, All
>>>
>>> I've got FreeSwitch 1.8.2 running in media proxy mode:
>>>
>>> <param name="inbound-proxy-media" value="true"/>
>>> <param name="inbound-late-negotiation" value="true"/>
>>>
>>> Dialplan is configured to bridge incoming calls to remote gateway and
>>> most of calls completing successfully, however some of them fail with
>>> INCOMPATIBLE_DESTINATION cause. For failed calls everything goes normally
>>> until bridged connection gets early media from callee:
>>>
>>> [NOTICE] sofia.c:7304 Pre-Answer sofia/external/2222 at 2.2.2.2!
>>> [INFO] switch_ivr_originate.c:3747 Sending early media
>>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PCMU/0
>>> 0 ms 960 samples 384000 bits 1 channels
>>> [DEBUG] switch_core_codec.c:111 sofia/internal/1111 at 1.1.1.1 Original
>>> read codec set to PCMU:0
>>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
>>> 1111 at 1.1.1.1] 1.1.1.1:18792->1.1.1.1:18792 codec: 0 ms: 20
>>> [ERR] switch_core_media.c:9510 AUDIO RTP REPORTS ERROR: *[Missing local
>>> host]*
>>> [NOTICE] switch_core_media.c:9511 Hangup sofia/internal/1111 at 1.1.1.1
>>> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
>>>
>>> Here's a dump for this call:
>>>
>>> <image.png>
>>>
>>> Healthy call on the same machine:
>>>
>>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/vpn/2222 at 2.2.2.2 PROXY/0
>>> 0 ms 160 samples 0 bits 1 channels
>>> [DEBUG] switch_core_codec.c:111 sofia/internal/2222 at 2.2.2.2 Original
>>> read codec set to PROXY:0
>>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
>>> 2222 at 2.2.2.2] 2.2.2.2:60818->2.2.2.2:60818 codec: 0 ms: 20
>>>
>>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PROXY/0
>>> 0 ms 160 samples 0 bits 1 channels
>>> [DEBUG] switch_core_codec.c:111 sofia/external/1111 at 1.1.1.1 Original
>>> read codec set to PROXY:0
>>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/external/
>>> 1111 at 1.1.1.1] 1.1.1.1:31412->1.1.1.1:31412 codec: 0 ms: 20
>>>
>>> So it looks like proxified media connection cannot be established for
>>> some calls. It shouldn't be a codec problem as call parties are configured
>>> with common codecs, FS is bond to local interface which is running with no
>>> errors. But I have no idea what "local host is missing" in my case so I'll
>>> be appreciated for any hint.
>>>
>>> FreeSWITCH Version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git
>>> a98a958 2018-09-26 17:55:25Z 64bit)
>>> CentOS Linux release 7.5.1804 (Core)
>>>
>>> Best,
>>> Marc
>>> _________________________________________________________________________
>>> 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>
>> _________________________________________________________________________
>> 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
>>
>>
>> _________________________________________________________________________
>> 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
>
>
>
> --
> Gregor Nanger
>
> *CTO*
> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
> • www.infomedia.si
> _________________________________________________________________________
> 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 gregor at infomedia.si  Mon Oct 15 22:31:45 2018
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 16 Oct 2018 00:31:45 +0200
Subject: [Freeswitch-users] [Missing local host] in proxy media mode
In-Reply-To: <CAFiqYu=G2GWb4yLV1Nib6v3RBeVirteHc_8wiM9kGhTQr8fcUA@mail.gmail.com>
References: <CAMBkyc1ETujbQLgYEyv+-URcV2OwCuWW0Dh2k+b6U2XjUkAMig@mail.gmail.com>
 <CAEJMVkB9sggz6y4RfPvWTYFuxOEpkvvM2Htmsnav8nfxRZzRog@mail.gmail.com>
 <187AA53B-CE2E-416D-996E-AD6D8F12043B@jerris.com>
 <CAC7ttFGWWWicu3-uYT7pNaqkBotfd5GwdiB9eSLTe-VJd3bAWg@mail.gmail.com>
 <CAFiqYu=G2GWb4yLV1Nib6v3RBeVirteHc_8wiM9kGhTQr8fcUA@mail.gmail.com>
Message-ID: <CAC7ttFFVcCWRKkQJrvbnbAx-ifgh7KsYPGJXqsGRjUgq=T_dDw@mail.gmail.com>

Yes, I am using mod_g729 and proxy media mode and it works. Just wanted to
know, why should I avoid proxy media mode at all? In scenario where I use
proxy media, I don't want to use any FS processing on channel, just to flow
from one ip to another.

V V tor., 16. okt. 2018 ob 00:17 je oseba Steven Ayre <steveayre at gmail.com>
napisala:

> mod_g729 will allow passthrough g729 on windows, which is all proxy media
> allows. only mod_com_g729 is limited to linux.
>
> On Mon, 15 Oct 2018 at 13:22, Gregor Nanger <gregor at infomedia.si> wrote:
>
>> Michael, can you please explain your comment a little bit more:
>>
>> "There is one exception to this rule, and that is when you are trying to
>> pass codecs that freeswitch does not know about, and even that case has a
>> better workaround than to use proxy media"
>>
>> We are using proxy, because FS on windows can't support g729 and
>> transcoding. What is better workaround than use proxy media?
>>
>> Gregor
>>
>> V V ned., 14. okt. 2018 ob 23:42 je oseba Michael Jerris <mike at jerris.com>
>> napisala:
>>
>>> I’ll expand on Brian’s comment.  Proxy media should NEVER be used.
>>> There is one exception to this rule, and that is when you are trying to
>>> pass codecs that freeswitch does not know about, and even that case has a
>>> better workaround than to use proxy media.  If you are using proxy media
>>> you should stop, if you are not sure how to do so you should discuss on the
>>> mailing list.  Proxy media mode is NOT the way to avoid transcoding, that
>>> can be done properly without this setting and this setting should never be
>>> used.
>>>
>>> Mike
>>>
>>> On Oct 12, 2018, at 5:07 PM, Brian West <brian at freeswitch.com> wrote:
>>>
>>> Could you elaborate on why you're using Proxy Media for?
>>> /b
>>>
>>> On Fri, Oct 12, 2018 at 5:37 PM Marc Quore <marquore at gmail.com> wrote:
>>>
>>>> Hi, All
>>>>
>>>> I've got FreeSwitch 1.8.2 running in media proxy mode:
>>>>
>>>> <param name="inbound-proxy-media" value="true"/>
>>>> <param name="inbound-late-negotiation" value="true"/>
>>>>
>>>> Dialplan is configured to bridge incoming calls to remote gateway and
>>>> most of calls completing successfully, however some of them fail with
>>>> INCOMPATIBLE_DESTINATION cause. For failed calls everything goes normally
>>>> until bridged connection gets early media from callee:
>>>>
>>>> [NOTICE] sofia.c:7304 Pre-Answer sofia/external/2222 at 2.2.2.2!
>>>> [INFO] switch_ivr_originate.c:3747 Sending early media
>>>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PCMU/0
>>>> 0 ms 960 samples 384000 bits 1 channels
>>>> [DEBUG] switch_core_codec.c:111 sofia/internal/1111 at 1.1.1.1 Original
>>>> read codec set to PCMU:0
>>>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
>>>> 1111 at 1.1.1.1] 1.1.1.1:18792->1.1.1.1:18792 codec: 0 ms: 20
>>>> [ERR] switch_core_media.c:9510 AUDIO RTP REPORTS ERROR: *[Missing
>>>> local host]*
>>>> [NOTICE] switch_core_media.c:9511 Hangup sofia/internal/1111 at 1.1.1.1
>>>> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
>>>>
>>>> Here's a dump for this call:
>>>>
>>>> <image.png>
>>>>
>>>> Healthy call on the same machine:
>>>>
>>>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/vpn/2222 at 2.2.2.2 PROXY/0
>>>> 0 ms 160 samples 0 bits 1 channels
>>>> [DEBUG] switch_core_codec.c:111 sofia/internal/2222 at 2.2.2.2 Original
>>>> read codec set to PROXY:0
>>>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
>>>> 2222 at 2.2.2.2] 2.2.2.2:60818->2.2.2.2:60818 codec: 0 ms: 20
>>>>
>>>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PROXY/0
>>>> 0 ms 160 samples 0 bits 1 channels
>>>> [DEBUG] switch_core_codec.c:111 sofia/external/1111 at 1.1.1.1 Original
>>>> read codec set to PROXY:0
>>>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/external/
>>>> 1111 at 1.1.1.1] 1.1.1.1:31412->1.1.1.1:31412 codec: 0 ms: 20
>>>>
>>>> So it looks like proxified media connection cannot be established for
>>>> some calls. It shouldn't be a codec problem as call parties are configured
>>>> with common codecs, FS is bond to local interface which is running with no
>>>> errors. But I have no idea what "local host is missing" in my case so I'll
>>>> be appreciated for any hint.
>>>>
>>>> FreeSWITCH Version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git
>>>> a98a958 2018-09-26 17:55:25Z 64bit)
>>>> CentOS Linux release 7.5.1804 (Core)
>>>>
>>>> Best,
>>>> Marc
>>>>
>>>> _________________________________________________________________________
>>>> 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>
>>> _________________________________________________________________________
>>> 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
>>>
>>>
>>> _________________________________________________________________________
>>> 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
>>
>>
>>
>> --
>> Gregor Nanger
>>
>> *CTO*
>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>> • www.infomedia.si
>> _________________________________________________________________________
>> 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
>
> _________________________________________________________________________
> 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



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
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From shaun.stokes at itec-support.co.uk  Tue Oct 16 05:13:10 2018
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Tue, 16 Oct 2018 05:13:10 +0000
Subject: [Freeswitch-users] How to update the caller id when bridging?
In-Reply-To: <CAEJMVkAFx3sQRdGzDVcG+0j1dCQXbafNoSUf_bD7wpSNag1pQQ@mail.gmail.com>
References: <20181006020400.3324ed18@kappa.digital-domain.net>
 <A9F49B6F-79D3-4444-A186-EE33BA36874A@vallimamod.org>
 <m37eijntsg.wl-covici@ccs.covici.com>,
 <CAEJMVkAFx3sQRdGzDVcG+0j1dCQXbafNoSUf_bD7wpSNag1pQQ@mail.gmail.com>
Message-ID: <LNXP265MB11802515DD41E60BBF96917EEBFE0@LNXP265MB1180.GBRP265.PROD.OUTLOOK.COM>

We’ve found that other PBXs we’ve tested this on send CLI updates using another INVITE which works more consistently across various user agents than using an UPDATE which requires custom code per user agent.

We have a JIRA for this issue open for Bria as we’ve had some trouble with this, but perhaps if the user agent isn’t on the safe list or doesn’t support SIP UPDATE FreeSWITCH could revert to sending another INVITE instead.

Thoughts?

Thanks,
Shaun

Get Outlook for iOS<https://aka.ms/o0ukef>
________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Brian West <brian at freeswitch.com>
Sent: Monday, October 15, 2018 20:50
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How to update the caller id when bridging?

The updates won't get sent unless the user agents are on the safe list, so i'd review the code.

/b


On Mon, Oct 15, 2018 at 11:51 AM John Covici <covici at ccs.covici.com<mailto:covici at ccs.covici.com>> wrote:
Anyway to do this in dialplan?

On Mon, 15 Oct 2018 08:58:14 -0400,
Vallimamod Abdullah wrote:
>
> Hi,
>
> Have a look at sofia_send_callee_id() and sofia_update_callee_id() funcs in sofia.c.
> I think they are doing what you want to achieve.
>
>
>
> Best Regards,
> --
> Vallimamod Abdullah
> SIP Solutions
> vma at sip.solutions
> linkedin.com/in/vallimamod<http://linkedin.com/in/vallimamod>
> .
>
>
> > On 6 Oct 2018, at 03:04, Andrew Clayton <andrew at zeta.digital-domain.net<mailto:andrew at zeta.digital-domain.net>> wrote:
> >
> > So this is a question based around the FreeSWITCH C API.
> >
> > I have two scenarios involving two endpoints, A & B.
> >
> > Scenario 1
> >
> > FS calls B using switch_ivr_originate(), it then calls A (say a
> > Linphone client) and creates a new switch_caller_profile_t for it using
> > switch_caller_profile_new() and sets the ->caller_id_number field to e.g
> > 1234 to indicate a call from B and then does a switch_ivr_originate().
> >
> > A rings and Linphone shows the call coming from 1234 at ... OK, that's all
> > fine. The calls are bridged together with switch_ivr_uuid_bridge(A, B)
> >
> > Scenario 2
> >
> > FS calls A (again lets say a Linphone client) using
> > switch_ivr_originate(), A rings and shows the call coming from
> > 0000... at ..., that's fine, I didn't set any CID.
> >
> > Now A is parked and is off-hook.
> >
> > FS then calls B, A & B are then bridged together with
> > switch_ivr_uuid_bridge(A, B), now of course Linphone still shows the
> > call as from  0000... at ... I'd like it show for example 1234 at ... like
> > above.
> >
> > Now I've tried setting numerous fields in the switch_caller_profile_t
> > structure for A just before doing the bridge, but to no avail.
> >
> > So does what I'm trying to do make sense?
> >
> > Is it even possible?
> >
> > Cheers,
> > Andrew
> >
> > _________________________________________________________________________
> > 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
>
>
>
> Bien cordialement,
> --
> Vallimamod Abdullah
> SIP Solutions
> Conseil & Développement de plateformes et services VOIP
> vma at sip.solutions
> +33 6 62 60 68 97
> linkedin.com/in/vallimamod<http://linkedin.com/in/vallimamod>
> .
>
>
> _________________________________________________________________________
> 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
>

--
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<mailto:covici at ccs.covici.com>

_________________________________________________________________________
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


--
[https://hipchat.freeswitch.org/files/1/9111/w0eGOzyOVyZQdMg/email_logo.png]

Brian West | Co-founder and Developer

Need Commercial support? emailsales at freeswitch.com<mailto: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<mailto:brian at freeswitch.com>

Mobile: 918-424-9378

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

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From mickael at winlux.fr  Tue Oct 16 12:18:16 2018
From: mickael at winlux.fr (Mickael Hubert)
Date: Tue, 16 Oct 2018 14:18:16 +0200
Subject: [Freeswitch-users] Multi gateways in profile
In-Reply-To: <6416dd6b-6cee-4feb-bfe2-f55a80337ae6@gmail.com>
References: <CA+fRVSUkFuJRgkUAQCxr=3p6fj-joQR_yQD=c=YD4WRu8r=P-g@mail.gmail.com>
 <f7699f6b-56aa-4f82-8841-1c1258cef39b@gmail.com>
 <CA+fRVSXA5-bzKqNgmrRaq=FFBqe06iPYYgv0rc48chwY9_PZ1A@mail.gmail.com>
 <6416dd6b-6cee-4feb-bfe2-f55a80337ae6@gmail.com>
Message-ID: <CA+fRVSUQFB9a4urdqAvoeiw1L+iotJq8oHQ1kcTZ+KfGgy8kUw@mail.gmail.com>

Hi Srigo,
for my case, I have gateway without REGISTER (it's IP peering)

In this case, when a incoming call arrives into FS, it does not find the
good gateway and all variables into gateway conf are not populate

++

Le lun. 15 oct. 2018 à 18:55, Srigo Kanapathipillai <ksrigo at gmail.com> a
écrit :

> Why you can't set the same variable in both gateways profile:
>
> In gw1:
>
> <variable name="incoming_gw" value="gw1"/>
>
>
> In gw2:
>
> <variable name="incoming_gw" value="gw2"/>
>
>
> And check in your dialplan if incoming_gw is set to gw1 OR gw2?
>
>
> Srigo Kana
> On 14 Oct 2018, at 18:33, Mickael Hubert <mickael at winlux.fr> wrote:
>>
>> Hi
>> Thanks a lot !
>>
>> I think the best way (maybe only way) it's to match with from IP.
>> It's sure when inbound call arrives from GW2, any variables in gateway
>> configuration aren't populate.
>>
>> But how Can I do math with IP please ?
>>
>> ++
>>
>>
>> Le sam. 13 oct. 2018 18:26, Srigo Kanapathipillai <ksrigo at gmail.com> a
>> écrit :
>>
>>> Hi Mickael,
>>>
>>> <gateway>
>>> ..params...
>>> <variables>
>>> <variable name="inbound_var_name" value="this is inbound"
>>> direction="inbound"/>
>>> <variable name="outbound_var_name" value="this is outbound"
>>> direction="outbound"/>
>>> <variable name="both_var_name" value="this on any direction"/>
>>> </variables>
>>> </gateway>
>>>
>>> Did you try to set variables on your gateway and use it in your dialplan?
>>>
>>> Srigo Kana
>>> On 13 Oct 2018, at 00:31, Mickael Hubert <mickael at winlux.fr> wrote:
>>>>
>>>> Hi guys,
>>>> I want to identify source of inbounds calls.
>>>> In my external profile I have two gateways:
>>>>
>>>> - One with REGISTER method
>>>> - One without REGISTER method (IP to IP), only iptables filtering
>>>>
>>>> <gateways>
>>>>       <gateway name="GW1">
>>>>         <param name="from-domain" value="**********"/>
>>>>         <param name="caller-id-in-from" value="true"/>
>>>>         <param name="proxy" value="1.1.1.1"/>
>>>>         <param name="register" value="true"/>
>>>>         <param name="username" value="**************"/>
>>>>         <param name="password" value="****************"/>
>>>>         <param name="expire-seconds" value="3600"/>
>>>>         <param name="retry-seconds" value="30"/>
>>>>         <param name="ping" value="25"/>
>>>>       </gateway>
>>>>       <gateway name="GW2">
>>>>         <param name="from-domain" value="*********"/>
>>>>         <param name="caller-id-in-from" value="true"/>
>>>>         <param name="proxy" value="2.2.2.2"/>
>>>>         <param name="register" value="false"/>
>>>>         <param name="retry-seconds" value="30"/>
>>>>         <param name="ping" value="25"/>
>>>>       </gateway>
>>>>     </gateways>
>>>>
>>>> In my dialplan I have this conf:
>>>>
>>>> <condition field="${sip_gateway}"  expression="GW1" break="never">
>>>>    <action application="set" data="number_dest=${destination_number}" />
>>>>    <action application="execute_extension" data="to-private XML
>>>> from-gw1"/>
>>>> </condition>
>>>> <condition field="${sip_gateway}"  expression="GW2" break="never">
>>>>    <action application="set" data="number_dest=${destination_number}" />
>>>>    <action application="execute_extension" data="to-private XML
>>>> from-gw2"/>
>>>> </condition>
>>>>
>>>> When a call arrives throught GW1 (REGISTERED), ${sip_gateway} is
>>>> populate with "GW1" perfect !. But when another call arrives throught GW2
>>>> (not REGISTERED), ${sip_gateway} is empty.
>>>>
>>>> Please, how can I identify the source gateway where calls arrive ?
>>>> maybe it's not a good way to do that ?
>>>>
>>>> thanks in advance
>>>>
>>>> ------------------------------
>>>>
>>>> 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
>>>>
>>>>
>>> _________________________________________________________________________
>>> 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
>>
>> ------------------------------
>>
>> 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
>>
>> _________________________________________________________________________
> 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  Tue Oct 16 13:39:10 2018
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 16 Oct 2018 08:39:10 -0500
Subject: [Freeswitch-users] [Missing local host] in proxy media mode
In-Reply-To: <CAC7ttFFVcCWRKkQJrvbnbAx-ifgh7KsYPGJXqsGRjUgq=T_dDw@mail.gmail.com>
References: <CAMBkyc1ETujbQLgYEyv+-URcV2OwCuWW0Dh2k+b6U2XjUkAMig@mail.gmail.com>
 <CAEJMVkB9sggz6y4RfPvWTYFuxOEpkvvM2Htmsnav8nfxRZzRog@mail.gmail.com>
 <187AA53B-CE2E-416D-996E-AD6D8F12043B@jerris.com>
 <CAC7ttFGWWWicu3-uYT7pNaqkBotfd5GwdiB9eSLTe-VJd3bAWg@mail.gmail.com>
 <CAFiqYu=G2GWb4yLV1Nib6v3RBeVirteHc_8wiM9kGhTQr8fcUA@mail.gmail.com>
 <CAC7ttFFVcCWRKkQJrvbnbAx-ifgh7KsYPGJXqsGRjUgq=T_dDw@mail.gmail.com>
Message-ID: <206001d46555$9f155d30$dd401790$@freeswitch.org>

Proxy media mode is a special mode that may or may not work the way you think it does, like bypass media mode, depending on your use cases it may cause more problems than its worth as a large number of things are bypassed disabled… there is no advantage for using proxy media mode unless you are just trying to use a new completely unsupported codec.  Proxy media mode is no faster than the default standard media mode when you are not transcoding.

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Gregor Nanger
Sent: Monday, October 15, 2018 5:32 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] [Missing local host] in proxy media mode

 

Yes, I am using mod_g729 and proxy media mode and it works. Just wanted to know, why should I avoid proxy media mode at all? In scenario where I use proxy media, I don't want to use any FS processing on channel, just to flow from one ip to another.

 

V V tor., 16. okt. 2018 ob 00:17 je oseba Steven Ayre <steveayre at gmail.com <mailto:steveayre at gmail.com> > napisala:

mod_g729 will allow passthrough g729 on windows, which is all proxy media allows. only mod_com_g729 is limited to linux.

 

On Mon, 15 Oct 2018 at 13:22, Gregor Nanger <gregor at infomedia.si <mailto:gregor at infomedia.si> > wrote:

Michael, can you please explain your comment a little bit more:

 

"There is one exception to this rule, and that is when you are trying to pass codecs that freeswitch does not know about, and even that case has a better workaround than to use proxy media"

 

We are using proxy, because FS on windows can't support g729 and transcoding. What is better workaround than use proxy media?

 

Gregor

 

V V ned., 14. okt. 2018 ob 23:42 je oseba Michael Jerris <mike at jerris.com <mailto:mike at jerris.com> > napisala:

I’ll expand on Brian’s comment.  Proxy media should NEVER be used.  There is one exception to this rule, and that is when you are trying to pass codecs that freeswitch does not know about, and even that case has a better workaround than to use proxy media.  If you are using proxy media you should stop, if you are not sure how to do so you should discuss on the mailing list.  Proxy media mode is NOT the way to avoid transcoding, that can be done properly without this setting and this setting should never be used.

 

Mike





On Oct 12, 2018, at 5:07 PM, Brian West <brian at freeswitch.com <mailto:brian at freeswitch.com> > wrote:

 

Could you elaborate on why you're using Proxy Media for?

/b

 

On Fri, Oct 12, 2018 at 5:37 PM Marc Quore <marquore at gmail.com <mailto:marquore at gmail.com> > wrote:

Hi, All

 

I've got FreeSwitch 1.8.2 running in media proxy mode:

 

<param name="inbound-proxy-media" value="true"/>
<param name="inbound-late-negotiation" value="true"/>

 

Dialplan is configured to bridge incoming calls to remote gateway and most of calls completing successfully, however some of them fail with INCOMPATIBLE_DESTINATION cause. For failed calls everything goes normally until bridged connection gets early media from callee:

 

[NOTICE] sofia.c:7304 Pre-Answer sofia/external/2222 at 2.2.2.2 <mailto:2222 at 2.2.2.2> !
[INFO] switch_ivr_originate.c:3747 Sending early media
[DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 <mailto:1111 at 1.1.1.1>  PCMU/0 0 ms 960 samples 384000 bits 1 channels
[DEBUG] switch_core_codec.c:111 sofia/internal/1111 at 1.1.1.1 <mailto:1111 at 1.1.1.1>  Original read codec set to PCMU:0
[DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/1111 at 1.1.1.1 <mailto:1111 at 1.1.1.1> ] 1.1.1.1:18792->1.1.1.1:18792 <http://1.1.1.1:18792/>  codec: 0 ms: 20
[ERR] switch_core_media.c:9510 AUDIO RTP REPORTS ERROR: [Missing local host]
[NOTICE] switch_core_media.c:9511 Hangup sofia/internal/1111 at 1.1.1.1 <mailto:1111 at 1.1.1.1>  [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]

 

Here's a dump for this call:

 

<image.png>

 

Healthy call on the same machine:

 

[DEBUG] switch_core_media.c:3781 Set Codec sofia/vpn/2222 at 2.2.2.2 <mailto:2222 at 2.2.2.2>  PROXY/0 0 ms 160 samples 0 bits 1 channels
[DEBUG] switch_core_codec.c:111 sofia/internal/2222 at 2.2.2.2 <mailto:2222 at 2.2.2.2>  Original read codec set to PROXY:0
[DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/2222 at 2.2.2.2 <mailto:2222 at 2.2.2.2> ] 2.2.2.2:60818->2.2.2.2:60818 <http://2.2.2.2:60818/>  codec: 0 ms: 20

[DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 <mailto:1111 at 1.1.1.1>  PROXY/0 0 ms 160 samples 0 bits 1 channels
[DEBUG] switch_core_codec.c:111 sofia/external/1111 at 1.1.1.1 <mailto:1111 at 1.1.1.1>  Original read codec set to PROXY:0
[DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/external/1111 at 1.1.1.1 <mailto:1111 at 1.1.1.1> ] 1.1.1.1:31412->1.1.1.1:31412 <http://1.1.1.1:31412/>  codec: 0 ms: 20

So it looks like proxified media connection cannot be established for some calls. It shouldn't be a codec problem as call parties are configured with common codecs, FS is bond to local interface which is running with no errors. But I have no idea what "local host is missing" in my case so I'll be appreciated for any hint.

 

FreeSWITCH Version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git a98a958 2018-09-26 17:55:25Z 64bit)

CentOS Linux release 7.5.1804 (Core)

 

Best,

Marc

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

Official FreeSWITCH Sites
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-- 

  <https://hipchat.freeswitch.org/files/1/9111/w0eGOzyOVyZQdMg/email_logo.png> 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com <mailto:sales at freeswitch.com>  

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

Email: brian at freeswitch.com <mailto:brian at freeswitch.com> 

Mobile: 918-424-9378

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

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

Official FreeSWITCH Sites
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https://freeswitch.org/confluence
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FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com

 

_________________________________________________________________________
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> 
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com




 

-- 


Gregor Nanger

 

CTO
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia 
• www.infomedia.si <http://www.infomedia.si> 

_________________________________________________________________________
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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sales at freeswitch.com <mailto:sales at freeswitch.com> 
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
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FreeSWITCH-users mailing list
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https://freeswitch.com




 

-- 


Gregor Nanger

 

CTO
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia 
• www.infomedia.si <http://www.infomedia.si> 

 

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From gregor at infomedia.si  Tue Oct 16 14:47:54 2018
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 16 Oct 2018 16:47:54 +0200
Subject: [Freeswitch-users] [Missing local host] in proxy media mode
In-Reply-To: <206001d46555$9f155d30$dd401790$@freeswitch.org>
References: <CAMBkyc1ETujbQLgYEyv+-URcV2OwCuWW0Dh2k+b6U2XjUkAMig@mail.gmail.com>
 <CAEJMVkB9sggz6y4RfPvWTYFuxOEpkvvM2Htmsnav8nfxRZzRog@mail.gmail.com>
 <187AA53B-CE2E-416D-996E-AD6D8F12043B@jerris.com>
 <CAC7ttFGWWWicu3-uYT7pNaqkBotfd5GwdiB9eSLTe-VJd3bAWg@mail.gmail.com>
 <CAFiqYu=G2GWb4yLV1Nib6v3RBeVirteHc_8wiM9kGhTQr8fcUA@mail.gmail.com>
 <CAC7ttFFVcCWRKkQJrvbnbAx-ifgh7KsYPGJXqsGRjUgq=T_dDw@mail.gmail.com>
 <206001d46555$9f155d30$dd401790$@freeswitch.org>
Message-ID: <CAC7ttFHvMBK3FqWL6fR371OFgaxsGMiTuVt-ojCx5uoFJC9m3g@mail.gmail.com>

Thank you, Ken, for explanation. Just please tell me if I am using mod_g729
on Windows, would g729 passthrough works without proxy media mode? As I
remembered, it didn't work, that's why I use proxy media mode.

V V tor., 16. okt. 2018 ob 16:29 je oseba Ken Rice <krice at freeswitch.org>
napisala:

> Proxy media mode is a special mode that may or may not work the way you
> think it does, like bypass media mode, depending on your use cases it may
> cause more problems than its worth as a large number of things are bypassed
> disabled… there is no advantage for using proxy media mode unless you are
> just trying to use a new completely unsupported codec.  Proxy media mode is
> no faster than the default standard media mode when you are not transcoding.
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Gregor Nanger
> *Sent:* Monday, October 15, 2018 5:32 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] [Missing local host] in proxy media mode
>
>
>
> Yes, I am using mod_g729 and proxy media mode and it works. Just wanted to
> know, why should I avoid proxy media mode at all? In scenario where I use
> proxy media, I don't want to use any FS processing on channel, just to flow
> from one ip to another.
>
>
>
> V V tor., 16. okt. 2018 ob 00:17 je oseba Steven Ayre <steveayre at gmail.com>
> napisala:
>
> mod_g729 will allow passthrough g729 on windows, which is all proxy media
> allows. only mod_com_g729 is limited to linux.
>
>
>
> On Mon, 15 Oct 2018 at 13:22, Gregor Nanger <gregor at infomedia.si> wrote:
>
> Michael, can you please explain your comment a little bit more:
>
>
>
> "There is one exception to this rule, and that is when you are trying to
> pass codecs that freeswitch does not know about, and even that case has a
> better workaround than to use proxy media"
>
>
>
> We are using proxy, because FS on windows can't support g729 and
> transcoding. What is better workaround than use proxy media?
>
>
>
> Gregor
>
>
>
> V V ned., 14. okt. 2018 ob 23:42 je oseba Michael Jerris <mike at jerris.com>
> napisala:
>
> I’ll expand on Brian’s comment.  Proxy media should NEVER be used.  There
> is one exception to this rule, and that is when you are trying to pass
> codecs that freeswitch does not know about, and even that case has a better
> workaround than to use proxy media.  If you are using proxy media you
> should stop, if you are not sure how to do so you should discuss on the
> mailing list.  Proxy media mode is NOT the way to avoid transcoding, that
> can be done properly without this setting and this setting should never be
> used.
>
>
>
> Mike
>
>
>
> On Oct 12, 2018, at 5:07 PM, Brian West <brian at freeswitch.com> wrote:
>
>
>
> Could you elaborate on why you're using Proxy Media for?
>
> /b
>
>
>
> On Fri, Oct 12, 2018 at 5:37 PM Marc Quore <marquore at gmail.com> wrote:
>
> Hi, All
>
>
>
> I've got FreeSwitch 1.8.2 running in media proxy mode:
>
>
>
> <param name="inbound-proxy-media" value="true"/>
> <param name="inbound-late-negotiation" value="true"/>
>
>
>
> Dialplan is configured to bridge incoming calls to remote gateway and most
> of calls completing successfully, however some of them fail with
> INCOMPATIBLE_DESTINATION cause. For failed calls everything goes normally
> until bridged connection gets early media from callee:
>
>
>
> [NOTICE] sofia.c:7304 Pre-Answer sofia/external/2222 at 2.2.2.2!
> [INFO] switch_ivr_originate.c:3747 Sending early media
> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PCMU/0
> 0 ms 960 samples 384000 bits 1 channels
> [DEBUG] switch_core_codec.c:111 sofia/internal/1111 at 1.1.1.1 Original read
> codec set to PCMU:0
> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
> 1111 at 1.1.1.1] 1.1.1.1:18792->1.1.1.1:18792 codec: 0 ms: 20
> [ERR] switch_core_media.c:9510 AUDIO RTP REPORTS ERROR: *[Missing local
> host]*
> [NOTICE] switch_core_media.c:9511 Hangup sofia/internal/1111 at 1.1.1.1
> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
>
>
>
> Here's a dump for this call:
>
>
>
> <image.png>
>
>
>
> Healthy call on the same machine:
>
>
>
> [DEBUG] switch_core_media.c:3781 Set Codec sofia/vpn/2222 at 2.2.2.2 PROXY/0
> 0 ms 160 samples 0 bits 1 channels
> [DEBUG] switch_core_codec.c:111 sofia/internal/2222 at 2.2.2.2 Original read
> codec set to PROXY:0
> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/
> 2222 at 2.2.2.2] 2.2.2.2:60818->2.2.2.2:60818 codec: 0 ms: 20
>
> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PROXY/0
> 0 ms 160 samples 0 bits 1 channels
> [DEBUG] switch_core_codec.c:111 sofia/external/1111 at 1.1.1.1 Original read
> codec set to PROXY:0
> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/external/
> 1111 at 1.1.1.1] 1.1.1.1:31412->1.1.1.1:31412 codec: 0 ms: 20
>
> So it looks like proxified media connection cannot be established for some
> calls. It shouldn't be a codec problem as call parties are configured with
> common codecs, FS is bond to local interface which is running with no
> errors. But I have no idea what "local host is missing" in my case so I'll
> be appreciated for any hint.
>
>
>
> FreeSWITCH Version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git
> a98a958 2018-09-26 17:55:25Z 64bit)
>
> CentOS Linux release 7.5.1804 (Core)
>
>
>
> Best,
>
> Marc
>
> _________________________________________________________________________
> 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>
>
> _________________________________________________________________________
> 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
>
>
>
> _________________________________________________________________________
> 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
>
>
>
>
> --
>
> *Gregor Nanger*
>
>
>
> *CTO*
> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
> • www.infomedia.si
>
> _________________________________________________________________________
> 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
>
> _________________________________________________________________________
> 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
>
>
>
>
> --
>
> *Gregor Nanger*
>
>
>
> *CTO*
> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
> • www.infomedia.si
>
>
> _________________________________________________________________________
> 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



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
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From krice at tollfreegateway.com  Tue Oct 16 17:30:36 2018
From: krice at tollfreegateway.com (Ken Rice)
Date: Tue, 16 Oct 2018 12:30:36 -0500
Subject: [Freeswitch-users] [Missing local host] in proxy media mode
In-Reply-To: <CAC7ttFHvMBK3FqWL6fR371OFgaxsGMiTuVt-ojCx5uoFJC9m3g@mail.gmail.com>
References: <CAMBkyc1ETujbQLgYEyv+-URcV2OwCuWW0Dh2k+b6U2XjUkAMig@mail.gmail.com>
 <CAEJMVkB9sggz6y4RfPvWTYFuxOEpkvvM2Htmsnav8nfxRZzRog@mail.gmail.com>
 <187AA53B-CE2E-416D-996E-AD6D8F12043B@jerris.com>
 <CAC7ttFGWWWicu3-uYT7pNaqkBotfd5GwdiB9eSLTe-VJd3bAWg@mail.gmail.com>
 <CAFiqYu=G2GWb4yLV1Nib6v3RBeVirteHc_8wiM9kGhTQr8fcUA@mail.gmail.com>
 <CAC7ttFFVcCWRKkQJrvbnbAx-ifgh7KsYPGJXqsGRjUgq=T_dDw@mail.gmail.com>
 <206001d46555$9f155d30$dd401790$@freeswitch.org>
 <CAC7ttFHvMBK3FqWL6fR371OFgaxsGMiTuVt-ojCx5uoFJC9m3g@mail.gmail.com>
Message-ID: <F28A1176-1D51-490E-BD15-1A0A4C46847C@tollfreegateway.com>

mod_g729 is the passthru (non-transcoding) module for supporting g729. it works on all platforms. mod_com_g729 which is the g729 transcoding module only works on linux

Sent from my iPhone

> On Oct 16, 2018, at 09:47, Gregor Nanger <gregor at infomedia.si> wrote:
> 
> Thank you, Ken, for explanation. Just please tell me if I am using mod_g729 on Windows, would g729 passthrough works without proxy media mode? As I remembered, it didn't work, that's why I use proxy media mode.
> 
> V V tor., 16. okt. 2018 ob 16:29 je oseba Ken Rice <krice at freeswitch.org> napisala:
>> Proxy media mode is a special mode that may or may not work the way you think it does, like bypass media mode, depending on your use cases it may cause more problems than its worth as a large number of things are bypassed disabled… there is no advantage for using proxy media mode unless you are just trying to use a new completely unsupported codec.  Proxy media mode is no faster than the default standard media mode when you are not transcoding.
>> 
>>  
>> 
>> From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Gregor Nanger
>> Sent: Monday, October 15, 2018 5:32 PM
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-users] [Missing local host] in proxy media mode
>> 
>>  
>> 
>> Yes, I am using mod_g729 and proxy media mode and it works. Just wanted to know, why should I avoid proxy media mode at all? In scenario where I use proxy media, I don't want to use any FS processing on channel, just to flow from one ip to another.
>> 
>>  
>> 
>> V V tor., 16. okt. 2018 ob 00:17 je oseba Steven Ayre <steveayre at gmail.com> napisala:
>> 
>> mod_g729 will allow passthrough g729 on windows, which is all proxy media allows. only mod_com_g729 is limited to linux.
>> 
>>  
>> 
>> On Mon, 15 Oct 2018 at 13:22, Gregor Nanger <gregor at infomedia.si> wrote:
>> 
>> Michael, can you please explain your comment a little bit more:
>> 
>>  
>> 
>> "There is one exception to this rule, and that is when you are trying to pass codecs that freeswitch does not know about, and even that case has a better workaround than to use proxy media"
>> 
>>  
>> 
>> We are using proxy, because FS on windows can't support g729 and transcoding. What is better workaround than use proxy media?
>> 
>>  
>> 
>> Gregor
>> 
>>  
>> 
>> V V ned., 14. okt. 2018 ob 23:42 je oseba Michael Jerris <mike at jerris.com> napisala:
>> 
>> I’ll expand on Brian’s comment.  Proxy media should NEVER be used.  There is one exception to this rule, and that is when you are trying to pass codecs that freeswitch does not know about, and even that case has a better workaround than to use proxy media.  If you are using proxy media you should stop, if you are not sure how to do so you should discuss on the mailing list.  Proxy media mode is NOT the way to avoid transcoding, that can be done properly without this setting and this setting should never be used.
>> 
>>  
>> 
>> Mike
>> 
>> 
>> 
>> 
>> On Oct 12, 2018, at 5:07 PM, Brian West <brian at freeswitch.com> wrote:
>> 
>>  
>> 
>> Could you elaborate on why you're using Proxy Media for?
>> 
>> /b
>> 
>>  
>> 
>> On Fri, Oct 12, 2018 at 5:37 PM Marc Quore <marquore at gmail.com> wrote:
>> 
>> Hi, All
>> 
>>  
>> 
>> I've got FreeSwitch 1.8.2 running in media proxy mode:
>> 
>>  
>> 
>> <param name="inbound-proxy-media" value="true"/>
>> <param name="inbound-late-negotiation" value="true"/>
>> 
>>  
>> 
>> Dialplan is configured to bridge incoming calls to remote gateway and most of calls completing successfully, however some of them fail with INCOMPATIBLE_DESTINATION cause. For failed calls everything goes normally until bridged connection gets early media from callee:
>> 
>>  
>> 
>> [NOTICE] sofia.c:7304 Pre-Answer sofia/external/2222 at 2.2.2.2!
>> [INFO] switch_ivr_originate.c:3747 Sending early media
>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PCMU/0 0 ms 960 samples 384000 bits 1 channels
>> [DEBUG] switch_core_codec.c:111 sofia/internal/1111 at 1.1.1.1 Original read codec set to PCMU:0
>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/1111 at 1.1.1.1] 1.1.1.1:18792->1.1.1.1:18792 codec: 0 ms: 20
>> [ERR] switch_core_media.c:9510 AUDIO RTP REPORTS ERROR: [Missing local host]
>> [NOTICE] switch_core_media.c:9511 Hangup sofia/internal/1111 at 1.1.1.1 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
>> 
>>  
>> 
>> Here's a dump for this call:
>> 
>>  
>> 
>> <image.png>
>> 
>>  
>> 
>> Healthy call on the same machine:
>> 
>>  
>> 
>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/vpn/2222 at 2.2.2.2 PROXY/0 0 ms 160 samples 0 bits 1 channels
>> [DEBUG] switch_core_codec.c:111 sofia/internal/2222 at 2.2.2.2 Original read codec set to PROXY:0
>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/internal/2222 at 2.2.2.2] 2.2.2.2:60818->2.2.2.2:60818 codec: 0 ms: 20
>> 
>> [DEBUG] switch_core_media.c:3781 Set Codec sofia/internal/1111 at 1.1.1.1 PROXY/0 0 ms 160 samples 0 bits 1 channels
>> [DEBUG] switch_core_codec.c:111 sofia/external/1111 at 1.1.1.1 Original read codec set to PROXY:0
>> [DEBUG] switch_core_media.c:8577 PROXY AUDIO RTP [sofia/external/1111 at 1.1.1.1] 1.1.1.1:31412->1.1.1.1:31412 codec: 0 ms: 20
>> 
>> So it looks like proxified media connection cannot be established for some calls. It shouldn't be a codec problem as call parties are configured with common codecs, FS is bond to local interface which is running with no errors. But I have no idea what "local host is missing" in my case so I'll be appreciated for any hint.
>> 
>>  
>> 
>> FreeSWITCH Version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git a98a958 2018-09-26 17:55:25Z 64bit)
>> 
>> CentOS Linux release 7.5.1804 (Core)
>> 
>>  
>> 
>> Best,
>> 
>> Marc
>> 
>> _________________________________________________________________________
>> 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
>> 
>> Email: brian at freeswitch.com
>> 
>> Mobile: 918-424-9378
>> 
>> Website: https://www.FreeSWITCH.com
>> 
>> 
>> 
>> _________________________________________________________________________
>> 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
>> 
>>  
>> 
>> _________________________________________________________________________
>> 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
>> 
>> 
>> 
>>  
>> 
>> --
>> 
>> Gregor Nanger
>>  
>> CTO
>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 
>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia 
>> • www.infomedia.si
>> _________________________________________________________________________
>> 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
>> 
>> _________________________________________________________________________
>> 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
>> 
>> 
>> 
>>  
>> 
>> --
>> 
>> Gregor Nanger
>>  
>> CTO
>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 
>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia 
>> • www.infomedia.si
>>  
>> 
>> _________________________________________________________________________
>> 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
> 
> 
> -- 
> Gregor Nanger
>  
> CTO
> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 
> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia 
> • www.infomedia.si
> _________________________________________________________________________
> 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 wsimon at stratusvideo.com  Tue Oct 16 20:10:54 2018
From: wsimon at stratusvideo.com (William Simon)
Date: Tue, 16 Oct 2018 20:10:54 +0000
Subject: [Freeswitch-users] Accepting an "optional" SRTP offer (crypto
 in RTP/AVP) and establishing SRTP
In-Reply-To: <C85C798A-BE0B-4EE2-8300-6051CE2103FC@jerris.com>
References: <2AE1EF96-BD07-4E78-A638-D9D53FF8FF1E@stratusvideo.com>
 <C85C798A-BE0B-4EE2-8300-6051CE2103FC@jerris.com>
Message-ID: <A37C45B3-A694-404B-A230-552E9BE98906@stratusvideo.com>

Michael, are you aware of a bug that would cause this behavior in 1.6.20? Just trying to narrow down whether we are up against a bug or a configuration error on our part.


> On Oct 12, 2018, at 6:00 PM, Michael Jerris <mike at jerris.com> wrote:
>
> Are you trying this on 1.8.2 version?
>
>> On Oct 5, 2018, at 10:27 AM, William Simon <wsimon at stratusvideo.com> wrote:
>>
>> We are working with an endpoint that offers optional SRTP in the non-RFC-compliant way of an a=crypto attribute within RTP/AVP.
>>
>> I have told FreeSWITCH to allow this with rtp_allow_crypto_in_avp=true at the right place in the dialplan.
>>
>> Immediately after that in the dialplan I have to reject SRTP by using rtp_secure_media=forbidden, otherwise the call setup still fails. By setting the value to forbidden, the call does proceed unencrypted.
>>
>> We want FreeSWITCH to proceed with media encryption. Setting rtp_secure_media to any other value results in FreeSWITCH rejecting the offer like this:
>>
>> v=0
>> o=FreeSWITCH 1538660754 1538660755 IN IP4 192.168.100.104
>> s=FreeSWITCH
>> c=IN IP4 192.168.100.104
>> t=0 0
>> m=audio 0 RTP/AVP 19
>> m=video 0 RTP/AVP 19
>>
>> Is there anything else I can do to force SRTP in the answer?
>>
>>
>
>
> _________________________________________________________________________
> 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 gregor at infomedia.si  Tue Oct 16 20:56:02 2018
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 16 Oct 2018 22:56:02 +0200
Subject: [Freeswitch-users] Multi gateways in profile
In-Reply-To: <CA+fRVSUQFB9a4urdqAvoeiw1L+iotJq8oHQ1kcTZ+KfGgy8kUw@mail.gmail.com>
References: <CA+fRVSUkFuJRgkUAQCxr=3p6fj-joQR_yQD=c=YD4WRu8r=P-g@mail.gmail.com>
 <f7699f6b-56aa-4f82-8841-1c1258cef39b@gmail.com>
 <CA+fRVSXA5-bzKqNgmrRaq=FFBqe06iPYYgv0rc48chwY9_PZ1A@mail.gmail.com>
 <6416dd6b-6cee-4feb-bfe2-f55a80337ae6@gmail.com>
 <CA+fRVSUQFB9a4urdqAvoeiw1L+iotJq8oHQ1kcTZ+KfGgy8kUw@mail.gmail.com>
Message-ID: <CAC7ttFGf0TkcPB8JjsJHqXJ8yM30gwyzrszEBk-X1z0yNOMF7g@mail.gmail.com>

What if you create gateway with ip and without registration? Maybe FS will
pair it and you can put variable. Just a thought.

On Tue, 16 Oct 2018, 16:29 Mickael Hubert, <mickael at winlux.fr> wrote:

> Hi Srigo,
> for my case, I have gateway without REGISTER (it's IP peering)
>
> In this case, when a incoming call arrives into FS, it does not find the
> good gateway and all variables into gateway conf are not populate
>
> ++
>
> Le lun. 15 oct. 2018 à 18:55, Srigo Kanapathipillai <ksrigo at gmail.com> a
> écrit :
>
>> Why you can't set the same variable in both gateways profile:
>>
>> In gw1:
>>
>> <variable name="incoming_gw" value="gw1"/>
>>
>>
>> In gw2:
>>
>> <variable name="incoming_gw" value="gw2"/>
>>
>>
>> And check in your dialplan if incoming_gw is set to gw1 OR gw2?
>>
>>
>> Srigo Kana
>> On 14 Oct 2018, at 18:33, Mickael Hubert <mickael at winlux.fr> wrote:
>>>
>>> Hi
>>> Thanks a lot !
>>>
>>> I think the best way (maybe only way) it's to match with from IP.
>>> It's sure when inbound call arrives from GW2, any variables in gateway
>>> configuration aren't populate.
>>>
>>> But how Can I do math with IP please ?
>>>
>>> ++
>>>
>>>
>>> Le sam. 13 oct. 2018 18:26, Srigo Kanapathipillai <ksrigo at gmail.com> a
>>> écrit :
>>>
>>>> Hi Mickael,
>>>>
>>>> <gateway>
>>>> ..params...
>>>> <variables>
>>>> <variable name="inbound_var_name" value="this is inbound"
>>>> direction="inbound"/>
>>>> <variable name="outbound_var_name" value="this is outbound"
>>>> direction="outbound"/>
>>>> <variable name="both_var_name" value="this on any direction"/>
>>>> </variables>
>>>> </gateway>
>>>>
>>>> Did you try to set variables on your gateway and use it in your
>>>> dialplan?
>>>>
>>>> Srigo Kana
>>>> On 13 Oct 2018, at 00:31, Mickael Hubert <mickael at winlux.fr> wrote:
>>>>>
>>>>> Hi guys,
>>>>> I want to identify source of inbounds calls.
>>>>> In my external profile I have two gateways:
>>>>>
>>>>> - One with REGISTER method
>>>>> - One without REGISTER method (IP to IP), only iptables filtering
>>>>>
>>>>> <gateways>
>>>>>       <gateway name="GW1">
>>>>>         <param name="from-domain" value="**********"/>
>>>>>         <param name="caller-id-in-from" value="true"/>
>>>>>         <param name="proxy" value="1.1.1.1"/>
>>>>>         <param name="register" value="true"/>
>>>>>         <param name="username" value="**************"/>
>>>>>         <param name="password" value="****************"/>
>>>>>         <param name="expire-seconds" value="3600"/>
>>>>>         <param name="retry-seconds" value="30"/>
>>>>>         <param name="ping" value="25"/>
>>>>>       </gateway>
>>>>>       <gateway name="GW2">
>>>>>         <param name="from-domain" value="*********"/>
>>>>>         <param name="caller-id-in-from" value="true"/>
>>>>>         <param name="proxy" value="2.2.2.2"/>
>>>>>         <param name="register" value="false"/>
>>>>>         <param name="retry-seconds" value="30"/>
>>>>>         <param name="ping" value="25"/>
>>>>>       </gateway>
>>>>>     </gateways>
>>>>>
>>>>> In my dialplan I have this conf:
>>>>>
>>>>> <condition field="${sip_gateway}"  expression="GW1" break="never">
>>>>>    <action application="set" data="number_dest=${destination_number}"
>>>>> />
>>>>>    <action application="execute_extension" data="to-private XML
>>>>> from-gw1"/>
>>>>> </condition>
>>>>> <condition field="${sip_gateway}"  expression="GW2" break="never">
>>>>>    <action application="set" data="number_dest=${destination_number}"
>>>>> />
>>>>>    <action application="execute_extension" data="to-private XML
>>>>> from-gw2"/>
>>>>> </condition>
>>>>>
>>>>> When a call arrives throught GW1 (REGISTERED), ${sip_gateway} is
>>>>> populate with "GW1" perfect !. But when another call arrives throught GW2
>>>>> (not REGISTERED), ${sip_gateway} is empty.
>>>>>
>>>>> Please, how can I identify the source gateway where calls arrive ?
>>>>> maybe it's not a good way to do that ?
>>>>>
>>>>> thanks in advance
>>>>>
>>>>> ------------------------------
>>>>>
>>>>> 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
>>>>>
>>>>>
>>>> _________________________________________________________________________
>>>> 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
>>>
>>> ------------------------------
>>>
>>> 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
>>>
>>> _________________________________________________________________________
>> 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
>
> _________________________________________________________________________
> 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 brian at freeswitch.com  Tue Oct 16 23:45:08 2018
From: brian at freeswitch.com (Brian West)
Date: Tue, 16 Oct 2018 18:45:08 -0500
Subject: [Freeswitch-users] Multi gateways in profile
In-Reply-To: <CAC7ttFGf0TkcPB8JjsJHqXJ8yM30gwyzrszEBk-X1z0yNOMF7g@mail.gmail.com>
References: <CA+fRVSUkFuJRgkUAQCxr=3p6fj-joQR_yQD=c=YD4WRu8r=P-g@mail.gmail.com>
 <f7699f6b-56aa-4f82-8841-1c1258cef39b@gmail.com>
 <CA+fRVSXA5-bzKqNgmrRaq=FFBqe06iPYYgv0rc48chwY9_PZ1A@mail.gmail.com>
 <6416dd6b-6cee-4feb-bfe2-f55a80337ae6@gmail.com>
 <CA+fRVSUQFB9a4urdqAvoeiw1L+iotJq8oHQ1kcTZ+KfGgy8kUw@mail.gmail.com>
 <CAC7ttFGf0TkcPB8JjsJHqXJ8yM30gwyzrszEBk-X1z0yNOMF7g@mail.gmail.com>
Message-ID: <CAEJMVkD1=6MbfVW1o9Q8SNjK+xw9VMx3tV67waShJfMKHB_uRg@mail.gmail.com>

It won’t ;)

On Tue, Oct 16, 2018 at 2:03 PM Gregor Nanger <gregor at infomedia.si> wrote:

> What if you create gateway with ip and without registration? Maybe FS will
> pair it and you can put variable. Just a thought.
>
> On Tue, 16 Oct 2018, 16:29 Mickael Hubert, <mickael at winlux.fr> wrote:
>
>> Hi Srigo,
>> for my case, I have gateway without REGISTER (it's IP peering)
>>
>> In this case, when a incoming call arrives into FS, it does not find the
>> good gateway and all variables into gateway conf are not populate
>>
>> ++
>>
>> Le lun. 15 oct. 2018 à 18:55, Srigo Kanapathipillai <ksrigo at gmail.com> a
>> écrit :
>>
>>> Why you can't set the same variable in both gateways profile:
>>>
>>> In gw1:
>>>
>>> <variable name="incoming_gw" value="gw1"/>
>>>
>>>
>>> In gw2:
>>>
>>> <variable name="incoming_gw" value="gw2"/>
>>>
>>>
>>> And check in your dialplan if incoming_gw is set to gw1 OR gw2?
>>>
>>>
>>> Srigo Kana
>>> On 14 Oct 2018, at 18:33, Mickael Hubert <mickael at winlux.fr> wrote:
>>>>
>>>> Hi
>>>> Thanks a lot !
>>>>
>>>> I think the best way (maybe only way) it's to match with from IP.
>>>> It's sure when inbound call arrives from GW2, any variables in gateway
>>>> configuration aren't populate.
>>>>
>>>> But how Can I do math with IP please ?
>>>>
>>>> ++
>>>>
>>>>
>>>> Le sam. 13 oct. 2018 18:26, Srigo Kanapathipillai <ksrigo at gmail.com> a
>>>> écrit :
>>>>
>>>>> Hi Mickael,
>>>>>
>>>>> <gateway>
>>>>> ..params...
>>>>> <variables>
>>>>> <variable name="inbound_var_name" value="this is inbound"
>>>>> direction="inbound"/>
>>>>> <variable name="outbound_var_name" value="this is outbound"
>>>>> direction="outbound"/>
>>>>> <variable name="both_var_name" value="this on any direction"/>
>>>>> </variables>
>>>>> </gateway>
>>>>>
>>>>> Did you try to set variables on your gateway and use it in your
>>>>> dialplan?
>>>>>
>>>>> Srigo Kana
>>>>> On 13 Oct 2018, at 00:31, Mickael Hubert <mickael at winlux.fr> wrote:
>>>>>>
>>>>>> Hi guys,
>>>>>> I want to identify source of inbounds calls.
>>>>>> In my external profile I have two gateways:
>>>>>>
>>>>>> - One with REGISTER method
>>>>>> - One without REGISTER method (IP to IP), only iptables filtering
>>>>>>
>>>>>> <gateways>
>>>>>>       <gateway name="GW1">
>>>>>>         <param name="from-domain" value="**********"/>
>>>>>>         <param name="caller-id-in-from" value="true"/>
>>>>>>         <param name="proxy" value="1.1.1.1"/>
>>>>>>         <param name="register" value="true"/>
>>>>>>         <param name="username" value="**************"/>
>>>>>>         <param name="password" value="****************"/>
>>>>>>         <param name="expire-seconds" value="3600"/>
>>>>>>         <param name="retry-seconds" value="30"/>
>>>>>>         <param name="ping" value="25"/>
>>>>>>       </gateway>
>>>>>>       <gateway name="GW2">
>>>>>>         <param name="from-domain" value="*********"/>
>>>>>>         <param name="caller-id-in-from" value="true"/>
>>>>>>         <param name="proxy" value="2.2.2.2"/>
>>>>>>         <param name="register" value="false"/>
>>>>>>         <param name="retry-seconds" value="30"/>
>>>>>>         <param name="ping" value="25"/>
>>>>>>       </gateway>
>>>>>>     </gateways>
>>>>>>
>>>>>> In my dialplan I have this conf:
>>>>>>
>>>>>> <condition field="${sip_gateway}"  expression="GW1" break="never">
>>>>>>    <action application="set" data="number_dest=${destination_number}"
>>>>>> />
>>>>>>    <action application="execute_extension" data="to-private XML
>>>>>> from-gw1"/>
>>>>>> </condition>
>>>>>> <condition field="${sip_gateway}"  expression="GW2" break="never">
>>>>>>    <action application="set" data="number_dest=${destination_number}"
>>>>>> />
>>>>>>    <action application="execute_extension" data="to-private XML
>>>>>> from-gw2"/>
>>>>>> </condition>
>>>>>>
>>>>>> When a call arrives throught GW1 (REGISTERED), ${sip_gateway} is
>>>>>> populate with "GW1" perfect !. But when another call arrives throught GW2
>>>>>> (not REGISTERED), ${sip_gateway} is empty.
>>>>>>
>>>>>> Please, how can I identify the source gateway where calls arrive ?
>>>>>> maybe it's not a good way to do that ?
>>>>>>
>>>>>> thanks in advance
>>>>>>
>>>>>> ------------------------------
>>>>>>
>>>>>> 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
>>>>>>
>>>>>>
>>>>> _________________________________________________________________________
>>>>> 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
>>>>
>>>> ------------------------------
>>>>
>>>> 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
>>>>
>>>>
>>> _________________________________________________________________________
>>> 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
>>
>> _________________________________________________________________________
>> 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
>
> _________________________________________________________________________
> 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 brian at freeswitch.com  Tue Oct 16 23:46:54 2018
From: brian at freeswitch.com (Brian West)
Date: Tue, 16 Oct 2018 18:46:54 -0500
Subject: [Freeswitch-users] Accepting an "optional" SRTP offer (crypto
 in RTP/AVP) and establishing SRTP
In-Reply-To: <A37C45B3-A694-404B-A230-552E9BE98906@stratusvideo.com>
References: <2AE1EF96-BD07-4E78-A638-D9D53FF8FF1E@stratusvideo.com>
 <C85C798A-BE0B-4EE2-8300-6051CE2103FC@jerris.com>
 <A37C45B3-A694-404B-A230-552E9BE98906@stratusvideo.com>
Message-ID: <CAEJMVkAryKuO7H9SWpqJ7_h_qEOWJ=ottn83X-cbUjXp6AbvpA@mail.gmail.com>

1.6.20 is unsupported now that 1.8 is out

On Tue, Oct 16, 2018 at 1:55 PM William Simon <wsimon at stratusvideo.com>
wrote:

> Michael, are you aware of a bug that would cause this behavior in 1.6.20?
> Just trying to narrow down whether we are up against a bug or a
> configuration error on our part.
>
>
> > On Oct 12, 2018, at 6:00 PM, Michael Jerris <mike at jerris.com> wrote:
> >
> > Are you trying this on 1.8.2 version?
> >
> >> On Oct 5, 2018, at 10:27 AM, William Simon <wsimon at stratusvideo.com>
> wrote:
> >>
> >> We are working with an endpoint that offers optional SRTP in the
> non-RFC-compliant way of an a=crypto attribute within RTP/AVP.
> >>
> >> I have told FreeSWITCH to allow this with rtp_allow_crypto_in_avp=true
> at the right place in the dialplan.
> >>
> >> Immediately after that in the dialplan I have to reject SRTP by using
> rtp_secure_media=forbidden, otherwise the call setup still fails. By
> setting the value to forbidden, the call does proceed unencrypted.
> >>
> >> We want FreeSWITCH to proceed with media encryption. Setting
> rtp_secure_media to any other value results in FreeSWITCH rejecting the
> offer like this:
> >>
> >> v=0
> >> o=FreeSWITCH 1538660754 1538660755 IN IP4 192.168.100.104
> >> s=FreeSWITCH
> >> c=IN IP4 192.168.100.104
> >> t=0 0
> >> m=audio 0 RTP/AVP 19
> >> m=video 0 RTP/AVP 19
> >>
> >> Is there anything else I can do to force SRTP in the answer?
> >>
> >>
> >
> >
> > _________________________________________________________________________
> > 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 information transmitted is intended only for the person or entity to
> which it is addressed and may contain proprietary, business-confidential
> and/or privileged material. If you are not the intended recipient of this
> message you are hereby notified that any use, review, retransmission,
> dissemination, distribution, reproduction or any action taken in reliance
> upon this message is prohibited. If you received this in error, please
> contact the sender and delete the material from any computer.”
> _________________________________________________________________________
> 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 enp at itx.ru  Wed Oct 17 11:36:16 2018
From: enp at itx.ru (Eugene Prokopiev)
Date: Wed, 17 Oct 2018 14:36:16 +0300
Subject: [Freeswitch-users] How to play some file into parked call via
	inbound ESL connection?
Message-ID: <CANM4RujtOKRx1FQQCqbjcBzj7NepQFwxuJTeFinDJjPP60r5Ag@mail.gmail.com>

Hi,

Is it possible to play some file into parked call via inbound ESL
connection? I have dialplan like this:

    <extension name="in">
      <condition>
        <action application="park"/>
      </condition>
    </extension>

I tried:

$ telnet localhost 8021

auth ClueCon

Content-Type: command/reply
Reply-Text: +OK accepted

event plain CHANNEL_PARK

Content-Type: command/reply
Reply-Text: +OK event listener enabled plain

Content-Length: 5397
Content-Type: text/event-plain

Event-Name: CHANNEL_PARK
...
Unique-ID: 783e9456-d1f2-11e8-a77f-95321f80797f
...

api uuid_answer 783e9456-d1f2-11e8-a77f-95321f80797f

Content-Type: api/response
Content-Length: 4

+OK

api uuid_displace 783e9456-d1f2-11e8-a77f-95321f80797f start
/usr/share/freeswitch/sounds/ru/RU/elena/ivr/8000/ivr-welcome.wav

Content-Type: api/response
Content-Length: 12

+OK Success

api uuid_broadcast 783e9456-d1f2-11e8-a77f-95321f80797f
playback::/usr/share/freeswitch/sounds/ru/RU/elena/ivr/8000/ivr-welcome.wav

Content-Type: api/response
Content-Length: 17

+OK Message sent

api uuid_kill 783e9456-d1f2-11e8-a77f-95321f80797f

Content-Type: api/response
Content-Length: 4

So, both api uuid_displace and api uuid_broadcast returns immediately
with +OK. Is it possible to play and wait until the end of file? Or
this is possible via sendmsg instead of api and suscribing on
PLAYBACK_STOP event?

-- 
WBR,
Eugene Prokopiev


From brian at freeswitch.com  Thu Oct 18 01:03:02 2018
From: brian at freeswitch.com (Brian West)
Date: Wed, 17 Oct 2018 18:03:02 -0700
Subject: [Freeswitch-users] How to play some file into parked call via
 inbound ESL connection?
In-Reply-To: <CANM4RujtOKRx1FQQCqbjcBzj7NepQFwxuJTeFinDJjPP60r5Ag@mail.gmail.com>
References: <CANM4RujtOKRx1FQQCqbjcBzj7NepQFwxuJTeFinDJjPP60r5Ag@mail.gmail.com>
Message-ID: <CAEJMVkCUYa6YM3usY66j8KudXUuBYgQb-3h50b0Pfs13rL3=8g@mail.gmail.com>

You don’t have to use that method,

sendmsg: <uuid>
call-command: execute
exec-app-name: playback
exec-app-args: file.wav
event-lock:true




On Wed, Oct 17, 2018 at 7:02 AM Eugene Prokopiev <enp at itx.ru> wrote:

> Hi,
>
> Is it possible to play some file into parked call via inbound ESL
> connection? I have dialplan like this:
>
>     <extension name="in">
>       <condition>
>         <action application="park"/>
>       </condition>
>     </extension>
>
> I tried:
>
> $ telnet localhost 8021
>
> auth ClueCon
>
> Content-Type: command/reply
> Reply-Text: +OK accepted
>
> event plain CHANNEL_PARK
>
> Content-Type: command/reply
> Reply-Text: +OK event listener enabled plain
>
> Content-Length: 5397
> Content-Type: text/event-plain
>
> Event-Name: CHANNEL_PARK
> ...
> Unique-ID: 783e9456-d1f2-11e8-a77f-95321f80797f
> ...
>
> api uuid_answer 783e9456-d1f2-11e8-a77f-95321f80797f
>
> Content-Type: api/response
> Content-Length: 4
>
> +OK
>
> api uuid_displace 783e9456-d1f2-11e8-a77f-95321f80797f start
> /usr/share/freeswitch/sounds/ru/RU/elena/ivr/8000/ivr-welcome.wav
>
> Content-Type: api/response
> Content-Length: 12
>
> +OK Success
>
> api uuid_broadcast 783e9456-d1f2-11e8-a77f-95321f80797f
> playback::/usr/share/freeswitch/sounds/ru/RU/elena/ivr/8000/ivr-welcome.wav
>
> Content-Type: api/response
> Content-Length: 17
>
> +OK Message sent
>
> api uuid_kill 783e9456-d1f2-11e8-a77f-95321f80797f
>
> Content-Type: api/response
> Content-Length: 4
>
> So, both api uuid_displace and api uuid_broadcast returns immediately
> with +OK. Is it possible to play and wait until the end of file? Or
> this is possible via sendmsg instead of api and suscribing on
> PLAYBACK_STOP event?
>
> --
> WBR,
> Eugene Prokopiev
>
> _________________________________________________________________________
> 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 sagarmalam at gmail.com  Thu Oct 18 10:24:06 2018
From: sagarmalam at gmail.com (sagar malam)
Date: Thu, 18 Oct 2018 15:54:06 +0530
Subject: [Freeswitch-users] Shared presence not working with callcenter
 application of Freeswitch
In-Reply-To: <6ABFDA54-2016-495F-BB0D-DCFB55874429@jerris.com>
References: <CAEZuF3vgN4OdaRha=edmCkhd_Ac-ZLThs6_uXx2LT3FGdVgfdQ@mail.gmail.com>
 <LNXP265MB1180E1DFD30C173B4DFC6759EBE70@LNXP265MB1180.GBRP265.PROD.OUTLOOK.COM>
 <CAEZuF3tbp1tPH9Bt7jDqiVeUAzBToBsD3Ah9mNS2RF+o7B+C9A@mail.gmail.com>
 <6ABFDA54-2016-495F-BB0D-DCFB55874429@jerris.com>
Message-ID: <CAEZuF3tZz101nOiVLugcszQUa+TxrixnL79375jdW=VOPWf8Rg@mail.gmail.com>

Yes.
Below parameters are enabled :
manage-presence
manage-shared-appearance
dbname = share_presence

On Sat, Oct 13, 2018 at 4:07 AM Michael Jerris <mike at jerris.com> wrote:

> Is manage presense enabled?
>
> On Oct 10, 2018, at 5:29 AM, sagar malam <sagarmalam at gmail.com> wrote:
>
> I have already tried that setting along with sip_invite_domain.But without
> any success.Further when i enable SLA and Presence debug , i dont see any
> SLA / presence logs which appears in case of one to one calls.
>
> On Wed, Oct 10, 2018 at 4:54 AM Shaun Stokes <
> shaun.stokes at itec-support.co.uk> wrote:
>
>> You need to set the presence_id variable of your agent when you bridge to
>> the agent in the agent contact field.
>>
>> Shaun
>>
>> Get Outlook for iOS <https://aka.ms/o0ukef>
>> ------------------------------
>> *From:* 20110170300n behalf of
>> *Sent:* Tuesday, October 9, 2018 19:48
>> *To:* FreeSWITCH Users Help
>> *Subject:* [Freeswitch-users] Shared presence not working with
>> callcenter application of Freeswitch
>>
>> I am facing issue related to presence/ shared presence for agents in
>> callcenter(mod_callcenter).
>> Suppose we are dialling a user 1001 at example.com and it is a shared user
>> in two phones. Shared presence works perfectly if we directly dial
>> extension 1001.
>>
>> But if same extension is configured as agent of callcenter and call is
>> bridged to agent(extension) through callcenter, shared presence does not
>> work. Freeswitch does not generate notify for agent state change.
>>
>> Is this expected behaviour ? If not then what can fix this ?
>>
>> Thanks in advance
>>
>
> _________________________________________________________________________
> 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 josedavid at zennio.com  Fri Oct 19 12:56:11 2018
From: josedavid at zennio.com (Jose David Jurado Alonso)
Date: Fri, 19 Oct 2018 14:56:11 +0200
Subject: [Freeswitch-users] VideoCall hangup always after 32 seconds to
	answer
Message-ID: <CANeCd7qQ7vNkGBGDopLPKnLcyR3UMRUhYOhd3nWUXsCNovzRmA@mail.gmail.com>

Hi,

I'ḿ get the next dialplan configuration in order to set the call max limit
timeout on 120 seconds after answer:

    <extension name="Local_Extension">
      <condition field="destination_number" expression="^(.*)$">
        <action application="export" data="dialed_extension=$1"/>

        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="ignore_early_media=false"/>
        <action application="set" data="execute_on_answer=sched_hangup
+120" />

        <action application="bridge" data="user/${dialed_extension}@
${domain_name}"/>
      </condition>
    </extension>


But the call always disconnect after 32 seconds... (calling linphone to
linphone android clients). I tried always using "call_timeout",
"originate_timeout" but the result is the same.

What I really want to do is:

+ Set timeout of 30 seconds to hangup if NO answer (nobody responds).
+ Set a maximum total time of 120 seconds since the call is answered.

Can someone help me with this configuration?

FS version: FreeSWITCH Version 1.9.0-654-ed4920e~64bit (-654-ed4920e 64bit)

Thanks!

José David Jurado Alonso.
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From tculjaga at gmail.com  Fri Oct 19 13:01:14 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 19 Oct 2018 15:01:14 +0200
Subject: [Freeswitch-users] How to play some file into parked call via
 inbound ESL connection?
In-Reply-To: <CAEJMVkCUYa6YM3usY66j8KudXUuBYgQb-3h50b0Pfs13rL3=8g@mail.gmail.com>
References: <CANM4RujtOKRx1FQQCqbjcBzj7NepQFwxuJTeFinDJjPP60r5Ag@mail.gmail.com>
 <CAEJMVkCUYa6YM3usY66j8KudXUuBYgQb-3h50b0Pfs13rL3=8g@mail.gmail.com>
Message-ID: <CAJoKKdV9bnJ9Qe4WP9rrNFo99FBJLiNuNbMPSk9SX1Nf0zt1zQ@mail.gmail.com>

its strange you want turn async into sync... but okay :=)

>From my opinion, would be better to subscribe PLAYBACK_START and
PLAYBACK_STOP events (and other events to track call state) and in this
case react on  PLAYBACK_STOP to process next command. This way you can
handle simultaneous calls :=)
As for commands, its better to use bgapi instead of api so you avoid being
blocking during command execution ( e.g. originate ).



regards,
Tihomir.




On Thu, 18 Oct 2018 at 12:28, Brian West <brian at freeswitch.com> wrote:

> You don’t have to use that method,
>
> sendmsg: <uuid>
> call-command: execute
> exec-app-name: playback
> exec-app-args: file.wav
> event-lock:true
>
>
>
>
> On Wed, Oct 17, 2018 at 7:02 AM Eugene Prokopiev <enp at itx.ru> wrote:
>
>> Hi,
>>
>> Is it possible to play some file into parked call via inbound ESL
>> connection? I have dialplan like this:
>>
>>     <extension name="in">
>>       <condition>
>>         <action application="park"/>
>>       </condition>
>>     </extension>
>>
>> I tried:
>>
>> $ telnet localhost 8021
>>
>> auth ClueCon
>>
>> Content-Type: command/reply
>> Reply-Text: +OK accepted
>>
>> event plain CHANNEL_PARK
>>
>> Content-Type: command/reply
>> Reply-Text: +OK event listener enabled plain
>>
>> Content-Length: 5397
>> Content-Type: text/event-plain
>>
>> Event-Name: CHANNEL_PARK
>> ...
>> Unique-ID: 783e9456-d1f2-11e8-a77f-95321f80797f
>> ...
>>
>> api uuid_answer 783e9456-d1f2-11e8-a77f-95321f80797f
>>
>> Content-Type: api/response
>> Content-Length: 4
>>
>> +OK
>>
>> api uuid_displace 783e9456-d1f2-11e8-a77f-95321f80797f start
>> /usr/share/freeswitch/sounds/ru/RU/elena/ivr/8000/ivr-welcome.wav
>>
>> Content-Type: api/response
>> Content-Length: 12
>>
>> +OK Success
>>
>> api uuid_broadcast 783e9456-d1f2-11e8-a77f-95321f80797f
>>
>> playback::/usr/share/freeswitch/sounds/ru/RU/elena/ivr/8000/ivr-welcome.wav
>>
>> Content-Type: api/response
>> Content-Length: 17
>>
>> +OK Message sent
>>
>> api uuid_kill 783e9456-d1f2-11e8-a77f-95321f80797f
>>
>> Content-Type: api/response
>> Content-Length: 4
>>
>> So, both api uuid_displace and api uuid_broadcast returns immediately
>> with +OK. Is it possible to play and wait until the end of file? Or
>> this is possible via sendmsg instead of api and suscribing on
>> PLAYBACK_STOP event?
>>
>> --
>> WBR,
>> Eugene Prokopiev
>>
>> _________________________________________________________________________
>> 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>
> _________________________________________________________________________
> 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  Fri Oct 19 22:30:43 2018
From: social at bohboh.info (Social Boh)
Date: Fri, 19 Oct 2018 17:30:43 -0500
Subject: [Freeswitch-users] CentOS 7, FreeSWITCH 1.8.2, mod_flite error
Message-ID: <fcc23d99-f3bf-a4bc-49cd-aa63eb598b9d@bohboh.info>

Hello list,

I compiled FreeSWITCH 1.8.2+git~20180926T175525Z~a98a958ac3~64bit en 
CentOS 7.5.1804 without problem with flite-devel version 2.0.0-1.el7.

Now, when I try to load the module:

/[CRIT] switch_loadable_module.c:1522 Error Loading module 
/usr/lib/freeswitch/mod/mod_flite.so//
//**/lib64/libflite.so.1: undefined symbol: snd_pcm_hw_params_any**/

Any Hint?

Thank you

Regards

-- 
---
I'm SoCIaL, MayBe

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From mouli123 at gmail.com  Sat Oct 20 13:16:32 2018
From: mouli123 at gmail.com (Chandramouli P)
Date: Sat, 20 Oct 2018 18:46:32 +0530
Subject: [Freeswitch-users] DTMF Issue - Unexpected digits are receiving
Message-ID: <CALR0W682r3CHbf4S1ZqVTZ+f3-+LnySXXrz9Z_ukb7L5s_mkqg@mail.gmail.com>

Hello,

We developed an application (FreeSwitch) that receives pressed digit by the
end user on mobile phone dialer through DTMF and some action will be
performed. Everything is working fine till now. But, we noticed one issue
recently. From the FreeSwitch log, we came to know that FreeSwitch is
received the letter i.e. "A" from DTMF, instead of digit. We tried to
regenerate this issue. But, when we are trying to regenerate, FreeSwitch is
receiving the digit from DTMF as usual. I would like to know How FreeSwitch
is received the letter i.e. "A", instead of digit. Which key was pressed by
the end user on mobile phone dialer?

Thank you.

Best Regards,
Chandramouli.
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From s.safarov at gmail.com  Sat Oct 20 18:07:57 2018
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 20 Oct 2018 21:07:57 +0300
Subject: [Freeswitch-users] CentOS 7, FreeSWITCH 1.8.2, mod_flite error
In-Reply-To: <fcc23d99-f3bf-a4bc-49cd-aa63eb598b9d@bohboh.info>
References: <fcc23d99-f3bf-a4bc-49cd-aa63eb598b9d@bohboh.info>
Message-ID: <CAHtxdDcbCjxAX0nf0C3W+06BGgP_QzNTrUWqY0wgoMWd_JRAyA@mail.gmail.com>

https://freeswitch.org/jira/browse/FS-8084

сб, 20 окт. 2018 г. в 20:07, Social Boh <social at bohboh.info>:

> Hello list,
>
> I compiled FreeSWITCH 1.8.2+git~20180926T175525Z~a98a958ac3~64bit en
> CentOS 7.5.1804 without problem with flite-devel version 2.0.0-1.el7.
>
> Now, when I try to load the module:
>
> *[CRIT] switch_loadable_module.c:1522 Error Loading module
> /usr/lib/freeswitch/mod/mod_flite.so*
> ***/lib64/libflite.so.1: undefined symbol: snd_pcm_hw_params_any***
>
> Any Hint?
>
> Thank you
>
> Regards
>
> --
> ---
> I'm SoCIaL, MayBe
>
> _________________________________________________________________________
> 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 tollfreegateway.com  Sat Oct 20 20:56:00 2018
From: krice at tollfreegateway.com (Ken Rice)
Date: Sat, 20 Oct 2018 15:56:00 -0500
Subject: [Freeswitch-users] VideoCall hangup always after 32 seconds to
	answer
In-Reply-To: <CANeCd7qQ7vNkGBGDopLPKnLcyR3UMRUhYOhd3nWUXsCNovzRmA@mail.gmail.com>
References: <CANeCd7qQ7vNkGBGDopLPKnLcyR3UMRUhYOhd3nWUXsCNovzRmA@mail.gmail.com>
Message-ID: <948455BA-EE8C-4A5C-ACD5-813597948723@tollfreegateway.com>

that sounds more like a nat issue. enable sip tracing and watch to see if the 200 ok is getting ack’d properly

Sent from my iPhone

> On Oct 19, 2018, at 07:56, Jose David Jurado Alonso <josedavid at zennio.com> wrote:
> 
> Hi,
> 
> I'ḿ get the next dialplan configuration in order to set the call max limit timeout on 120 seconds after answer:
> 
>     <extension name="Local_Extension">
>       <condition field="destination_number" expression="^(.*)$">
>         <action application="export" data="dialed_extension=$1"/>
> 
>         <action application="set" data="hangup_after_bridge=true"/>
>         <action application="set" data="continue_on_fail=true"/>
>         <action application="set" data="ignore_early_media=false"/>
>         <action application="set" data="execute_on_answer=sched_hangup +120" />
> 
>         <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
>       </condition>
>     </extension>
> 
> 
> But the call always disconnect after 32 seconds... (calling linphone to linphone android clients). I tried always using "call_timeout", "originate_timeout" but the result is the same.
> 
> What I really want to do is:
> 
> + Set timeout of 30 seconds to hangup if NO answer (nobody responds).
> + Set a maximum total time of 120 seconds since the call is answered.
> 
> Can someone help me with this configuration?
> 
> FS version: FreeSWITCH Version 1.9.0-654-ed4920e~64bit (-654-ed4920e 64bit)
> 
> Thanks!
> José David Jurado Alonso.
> 
> _________________________________________________________________________
> 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  Sat Oct 20 23:12:34 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sun, 21 Oct 2018 00:12:34 +0100
Subject: [Freeswitch-users] VideoCall hangup always after 32 seconds to
	answer
In-Reply-To: <948455BA-EE8C-4A5C-ACD5-813597948723@tollfreegateway.com>
References: <CANeCd7qQ7vNkGBGDopLPKnLcyR3UMRUhYOhd3nWUXsCNovzRmA@mail.gmail.com>
 <948455BA-EE8C-4A5C-ACD5-813597948723@tollfreegateway.com>
Message-ID: <CAFGRPVq2pZUt86jYFdcsSAGuc2Wn70r1ak8v-xx64Mg1UuDe2Q@mail.gmail.com>

32 seconds is ALWAYS an unacknowledged 200 OK, or an rtp timeout. Checkout
a trace, look at what rtp ip both sides are offering, make sure no side
offers a private ip address. Also make sure fs gets the ack to the 200ok.

On Sat, Oct 20, 2018, 23:00 Ken Rice <krice at tollfreegateway.com> wrote:

> that sounds more like a nat issue. enable sip tracing and watch to see if
> the 200 ok is getting ack’d properly
>
> Sent from my iPhone
>
> On Oct 19, 2018, at 07:56, Jose David Jurado Alonso <josedavid at zennio.com>
> wrote:
>
> Hi,
>
> I'ḿ get the next dialplan configuration in order to set the call max limit
> timeout on 120 seconds after answer:
>
>     <extension name="Local_Extension">
>       <condition field="destination_number" expression="^(.*)$">
>         <action application="export" data="dialed_extension=$1"/>
>
>         <action application="set" data="hangup_after_bridge=true"/>
>         <action application="set" data="continue_on_fail=true"/>
>         <action application="set" data="ignore_early_media=false"/>
>         <action application="set" data="execute_on_answer=sched_hangup
> +120" />
>
>         <action application="bridge" data="user/${dialed_extension}@
> ${domain_name}"/>
>       </condition>
>     </extension>
>
>
> But the call always disconnect after 32 seconds... (calling linphone to
> linphone android clients). I tried always using "call_timeout",
> "originate_timeout" but the result is the same.
>
> What I really want to do is:
>
> + Set timeout of 30 seconds to hangup if NO answer (nobody responds).
> + Set a maximum total time of 120 seconds since the call is answered.
>
> Can someone help me with this configuration?
>
> FS version: FreeSWITCH Version 1.9.0-654-ed4920e~64bit (-654-ed4920e 64bit)
>
> Thanks!
>
> José David Jurado Alonso.
>
> _________________________________________________________________________
> 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
>
> _________________________________________________________________________
> 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  Sat Oct 20 23:19:55 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sun, 21 Oct 2018 00:19:55 +0100
Subject: [Freeswitch-users] DTMF Issue - Unexpected digits are receiving
In-Reply-To: <CALR0W682r3CHbf4S1ZqVTZ+f3-+LnySXXrz9Z_ukb7L5s_mkqg@mail.gmail.com>
References: <CALR0W682r3CHbf4S1ZqVTZ+f3-+LnySXXrz9Z_ukb7L5s_mkqg@mail.gmail.com>
Message-ID: <CAFGRPVoB-6VXTVvO9oJthH-38ky0n5PQHZdrih1WTkyyoHML_Q@mail.gmail.com>

>From what i know, "A" is a multifrequency tone, 697hz+1633hz. Maybe the
client is actually sending that.

On Sat, Oct 20, 2018, 17:54 Chandramouli P <mouli123 at gmail.com> wrote:

> Hello,
>
> We developed an application (FreeSwitch) that receives pressed digit by
> the end user on mobile phone dialer through DTMF and some action will be
> performed. Everything is working fine till now. But, we noticed one issue
> recently. From the FreeSwitch log, we came to know that FreeSwitch is
> received the letter i.e. "A" from DTMF, instead of digit. We tried to
> regenerate this issue. But, when we are trying to regenerate, FreeSwitch is
> receiving the digit from DTMF as usual. I would like to know How FreeSwitch
> is received the letter i.e. "A", instead of digit. Which key was pressed by
> the end user on mobile phone dialer?
>
> Thank you.
>
> Best Regards,
> Chandramouli.
> _________________________________________________________________________
> 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 ch.chhatra at gmail.com  Sun Oct 21 03:50:08 2018
From: ch.chhatra at gmail.com (Chhorm Chhatra)
Date: Sun, 21 Oct 2018 10:50:08 +0700
Subject: [Freeswitch-users] Freeswitch failed to initiate outbound call
 using SIPs + SRTP (SRTP unprotect )
In-Reply-To: <CAHtxdDfRHo+TYa9WmxByqW-me1wSzCbHMVKoi-_QcVETyrDi+w@mail.gmail.com>
References: <CAJssr-gTObUuiwFLYiDAdhuYTGyh9L57sL885i-uEA38rHksJg@mail.gmail.com>
 <CAEJMVkAe0Jpbr5NR7BP-K2W5rmKT4E=Z5oEg09G_-jaNnN=AFQ@mail.gmail.com>
 <CAJssr-g2C=tUhV_iXpRJCEWMEZL1MiH=D14eKVSE=jfD-DuM8Q@mail.gmail.com>
 <CAHtxdDfv7WTyt9jsquiGD78cYHyQit9rM=haYLbW1X+9P1pk1w@mail.gmail.com>
 <CAHtxdDfRHo+TYa9WmxByqW-me1wSzCbHMVKoi-_QcVETyrDi+w@mail.gmail.com>
Message-ID: <CAJssr-jiq2YqN52gSy0sTAXWxUNc22agoZ_rBpvFu01L_OuOFg@mail.gmail.com>

Thank you Safarov for your solution.
I am not quite sure about reverting the dial-string part.
Could you please electorate on how can I revert the dial-string?
Do I need to remove the dial-string from the dialplan or from the user
directory configuration?
Thank you in advance.

On Sat, Sep 29, 2018 at 10:52 PM Sergey Safarov <s.safarov at gmail.com> wrote:

> Need to revert back "dial-string" in directory config
> Also important "sips" and "sip" uri different. Please make sure you not
> use sips uri in client side.
>
> Sergey
>
> сб, 29 сент. 2018 г. в 13:35, Sergey Safarov <s.safarov at gmail.com>:
>
>> As i understand you try overwrite transport to user B registration.
>> In many case users is located behind NAT and FS cannot establish TLS
>> connections to B-user.
>>
>> Think in your case need to disable all non TLS sockets and then simple
>> try bridge "user/{user}@{domain}"
>>
>> сб, 29 сент. 2018 г. в 13:20, Chhorm Chhatra <ch.chhatra at gmail.com>:
>>
>>> Dear Brain West,
>>> thank you for your response.
>>> I would like to confirm that either using export or set on a leg of
>>> "rtp_secure_media=true" with the following dial-string is not working for
>>> me. One leg call is fine but it does not work for 2-leg call (I could not
>>> hear the sound and the call terminates after
>>>   {rtp_secure_media=${regex(${sofia_contact(${dialed_user}@
>>> ${dialed_domain})}|transport=tls)},presence_id=${dialed_user}@
>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"
>>>
>>> On Wed, 1 Aug 2018 at 23:20, Brian West <brian at freeswitch.com> wrote:
>>>
>>>> don't us export, set it inside {}, or on use set on a-leg.
>>>>
>>>> /b
>>>>
>>>>
>>>> On Tue, Jul 31, 2018 at 9:23 AM, Chhorm Chhatra <ch.chhatra at gmail.com>
>>>> wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> Currently, I faced a problem regarding SRTP outbound call to user (Leg
>>>>> B).
>>>>>
>>>>> The scenario is like this,
>>>>>
>>>>>    - We set up our own root CA to an IP address (e.g 192.168.0.13)
>>>>>    - We create a server certificate for freeswitch at 192.168.0.13
>>>>>    - Linphone is used as SIP client and is configured to trust our
>>>>>    root CA by default.
>>>>>    - Linphone A is configured to register to Freeswitch vis TLS +
>>>>>    SRTP. (One leg call to server has both SIPs and SRTP – completely secure)
>>>>>    - Linphone B is registered to Freeswitch via TLS + SRTP, and
>>>>>    waiting for Linphone A to call to.
>>>>>
>>>>> (One leg call to server, e.g. 9196 (echo test), is completely secure
>>>>> with SRTP + SIPs)
>>>>>
>>>>>    - Unfortunately, if A call to B, only A leg has SIPs + SRTP, but
>>>>>    Leg B is not encrypted with SRTP and SIPs at all. This causes *SRTP
>>>>>    unprotect failed with code 7 (auth check failed)**.*
>>>>>
>>>>> + Dialplan Configuration
>>>>>
>>>>> <action application="set" data="rtp_secure_media=true"/>
>>>>>
>>>>> <action application="export" data="rtp_secure_media=true"/>
>>>>>
>>>>> The dial-string is <action application="bridge"
>>>>> data="user/${dialed_extension}@${domain_name}"/>
>>>>>
>>>>> + Directory Configruation:
>>>>>
>>>>> <param name="dial-string"
>>>>> value="{rtp_secure_media=${regex(${sofia_contact(${dialed_user}@
>>>>> ${dialed_domain})}|transport=tls)},presence_id=${dialed_user}@
>>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" />
>>>>>
>>>>> My question is that, is there any configuration left that I have to
>>>>> set up in order to let freeswitch initiate an outbound call to Leg B
>>>>> correctly with SRTP and SIPs (tls)?
>>>>>
>>>>> Any help would be really appreciated.
>>>>> Thank you so much.
>>>>> Best Regard,
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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 <(918)%20424-9378>
>>>>
>>>> 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>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>>>
>>> _________________________________________________________________________
>>> 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
>>
>> _________________________________________________________________________
> 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 brian at freeswitch.com  Sun Oct 21 04:41:46 2018
From: brian at freeswitch.com (Brian West)
Date: Sat, 20 Oct 2018 23:41:46 -0500
Subject: [Freeswitch-users] How to play some file into parked call via
 inbound ESL connection?
In-Reply-To: <CAJoKKdV9bnJ9Qe4WP9rrNFo99FBJLiNuNbMPSk9SX1Nf0zt1zQ@mail.gmail.com>
References: <CANM4RujtOKRx1FQQCqbjcBzj7NepQFwxuJTeFinDJjPP60r5Ag@mail.gmail.com>
 <CAEJMVkCUYa6YM3usY66j8KudXUuBYgQb-3h50b0Pfs13rL3=8g@mail.gmail.com>
 <CAJoKKdV9bnJ9Qe4WP9rrNFo99FBJLiNuNbMPSk9SX1Nf0zt1zQ@mail.gmail.com>
Message-ID: <CAEJMVkBLP3nk=HX19NdB1_YFBu-0+WfObYtTTj2qLz1gJCf_9w@mail.gmail.com>

No, it can stack them for you, no need to do it manually.

/b

On Sat, Oct 20, 2018 at 10:50 AM Tihomir Culjaga <tculjaga at gmail.com> wrote:

> its strange you want turn async into sync... but okay :=)
>
> From my opinion, would be better to subscribe PLAYBACK_START and
> PLAYBACK_STOP events (and other events to track call state) and in this
> case react on  PLAYBACK_STOP to process next command. This way you can
> handle simultaneous calls :=)
> As for commands, its better to use bgapi instead of api so you avoid being
> blocking during command execution ( e.g. originate ).
>
>
>
> regards,
> Tihomir.
>
>
>
>
> On Thu, 18 Oct 2018 at 12:28, Brian West <brian at freeswitch.com> wrote:
>
>> You don’t have to use that method,
>>
>> sendmsg: <uuid>
>> call-command: execute
>> exec-app-name: playback
>> exec-app-args: file.wav
>> event-lock:true
>>
>>
>>
>>
>> On Wed, Oct 17, 2018 at 7:02 AM Eugene Prokopiev <enp at itx.ru> wrote:
>>
>>> Hi,
>>>
>>> Is it possible to play some file into parked call via inbound ESL
>>> connection? I have dialplan like this:
>>>
>>>     <extension name="in">
>>>       <condition>
>>>         <action application="park"/>
>>>       </condition>
>>>     </extension>
>>>
>>> I tried:
>>>
>>> $ telnet localhost 8021
>>>
>>> auth ClueCon
>>>
>>> Content-Type: command/reply
>>> Reply-Text: +OK accepted
>>>
>>> event plain CHANNEL_PARK
>>>
>>> Content-Type: command/reply
>>> Reply-Text: +OK event listener enabled plain
>>>
>>> Content-Length: 5397
>>> Content-Type: text/event-plain
>>>
>>> Event-Name: CHANNEL_PARK
>>> ...
>>> Unique-ID: 783e9456-d1f2-11e8-a77f-95321f80797f
>>> ...
>>>
>>> api uuid_answer 783e9456-d1f2-11e8-a77f-95321f80797f
>>>
>>> Content-Type: api/response
>>> Content-Length: 4
>>>
>>> +OK
>>>
>>> api uuid_displace 783e9456-d1f2-11e8-a77f-95321f80797f start
>>> /usr/share/freeswitch/sounds/ru/RU/elena/ivr/8000/ivr-welcome.wav
>>>
>>> Content-Type: api/response
>>> Content-Length: 12
>>>
>>> +OK Success
>>>
>>> api uuid_broadcast 783e9456-d1f2-11e8-a77f-95321f80797f
>>>
>>> playback::/usr/share/freeswitch/sounds/ru/RU/elena/ivr/8000/ivr-welcome.wav
>>>
>>> Content-Type: api/response
>>> Content-Length: 17
>>>
>>> +OK Message sent
>>>
>>> api uuid_kill 783e9456-d1f2-11e8-a77f-95321f80797f
>>>
>>> Content-Type: api/response
>>> Content-Length: 4
>>>
>>> So, both api uuid_displace and api uuid_broadcast returns immediately
>>> with +OK. Is it possible to play and wait until the end of file? Or
>>> this is possible via sendmsg instead of api and suscribing on
>>> PLAYBACK_STOP event?
>>>
>>> --
>>> WBR,
>>> Eugene Prokopiev
>>>
>>> _________________________________________________________________________
>>> 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>
>> _________________________________________________________________________
>> 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
>
> _________________________________________________________________________
> 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  Mon Oct 22 06:21:07 2018
From: s.safarov at gmail.com (Sergey Safarov)
Date: Mon, 22 Oct 2018 09:21:07 +0300
Subject: [Freeswitch-users] Freeswitch failed to initiate outbound call
 using SIPs + SRTP (SRTP unprotect )
In-Reply-To: <CAJssr-jiq2YqN52gSy0sTAXWxUNc22agoZ_rBpvFu01L_OuOFg@mail.gmail.com>
References: <CAJssr-gTObUuiwFLYiDAdhuYTGyh9L57sL885i-uEA38rHksJg@mail.gmail.com>
 <CAEJMVkAe0Jpbr5NR7BP-K2W5rmKT4E=Z5oEg09G_-jaNnN=AFQ@mail.gmail.com>
 <CAJssr-g2C=tUhV_iXpRJCEWMEZL1MiH=D14eKVSE=jfD-DuM8Q@mail.gmail.com>
 <CAHtxdDfv7WTyt9jsquiGD78cYHyQit9rM=haYLbW1X+9P1pk1w@mail.gmail.com>
 <CAHtxdDfRHo+TYa9WmxByqW-me1wSzCbHMVKoi-_QcVETyrDi+w@mail.gmail.com>
 <CAJssr-jiq2YqN52gSy0sTAXWxUNc22agoZ_rBpvFu01L_OuOFg@mail.gmail.com>
Message-ID: <CAHtxdDfz2t-W5upCBnRDkD5gvrzicM2bBO1+xTBt-9=1dAmrgg@mail.gmail.com>

example of dialstring
https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/conf/vanilla/directory/default.xml#24

вс, 21 окт. 2018 г. в 21:12, Chhorm Chhatra <ch.chhatra at gmail.com>:

> Thank you Safarov for your solution.
> I am not quite sure about reverting the dial-string part.
> Could you please electorate on how can I revert the dial-string?
> Do I need to remove the dial-string from the dialplan or from the user
> directory configuration?
> Thank you in advance.
>
> On Sat, Sep 29, 2018 at 10:52 PM Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> Need to revert back "dial-string" in directory config
>> Also important "sips" and "sip" uri different. Please make sure you not
>> use sips uri in client side.
>>
>> Sergey
>>
>> сб, 29 сент. 2018 г. в 13:35, Sergey Safarov <s.safarov at gmail.com>:
>>
>>> As i understand you try overwrite transport to user B registration.
>>> In many case users is located behind NAT and FS cannot establish TLS
>>> connections to B-user.
>>>
>>> Think in your case need to disable all non TLS sockets and then simple
>>> try bridge "user/{user}@{domain}"
>>>
>>> сб, 29 сент. 2018 г. в 13:20, Chhorm Chhatra <ch.chhatra at gmail.com>:
>>>
>>>> Dear Brain West,
>>>> thank you for your response.
>>>> I would like to confirm that either using export or set on a leg of
>>>> "rtp_secure_media=true" with the following dial-string is not working for
>>>> me. One leg call is fine but it does not work for 2-leg call (I could not
>>>> hear the sound and the call terminates after
>>>>   {rtp_secure_media=${regex(${sofia_contact(${dialed_user}@
>>>> ${dialed_domain})}|transport=tls)},presence_id=${dialed_user}@
>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"
>>>>
>>>> On Wed, 1 Aug 2018 at 23:20, Brian West <brian at freeswitch.com> wrote:
>>>>
>>>>> don't us export, set it inside {}, or on use set on a-leg.
>>>>>
>>>>> /b
>>>>>
>>>>>
>>>>> On Tue, Jul 31, 2018 at 9:23 AM, Chhorm Chhatra <ch.chhatra at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Hello,
>>>>>>
>>>>>> Currently, I faced a problem regarding SRTP outbound call to user
>>>>>> (Leg B).
>>>>>>
>>>>>> The scenario is like this,
>>>>>>
>>>>>>    - We set up our own root CA to an IP address (e.g 192.168.0.13)
>>>>>>    - We create a server certificate for freeswitch at 192.168.0.13
>>>>>>    - Linphone is used as SIP client and is configured to trust our
>>>>>>    root CA by default.
>>>>>>    - Linphone A is configured to register to Freeswitch vis TLS +
>>>>>>    SRTP. (One leg call to server has both SIPs and SRTP – completely secure)
>>>>>>    - Linphone B is registered to Freeswitch via TLS + SRTP, and
>>>>>>    waiting for Linphone A to call to.
>>>>>>
>>>>>> (One leg call to server, e.g. 9196 (echo test), is completely secure
>>>>>> with SRTP + SIPs)
>>>>>>
>>>>>>    - Unfortunately, if A call to B, only A leg has SIPs + SRTP, but
>>>>>>    Leg B is not encrypted with SRTP and SIPs at all. This causes *SRTP
>>>>>>    unprotect failed with code 7 (auth check failed)**.*
>>>>>>
>>>>>> + Dialplan Configuration
>>>>>>
>>>>>> <action application="set" data="rtp_secure_media=true"/>
>>>>>>
>>>>>> <action application="export" data="rtp_secure_media=true"/>
>>>>>>
>>>>>> The dial-string is <action application="bridge"
>>>>>> data="user/${dialed_extension}@${domain_name}"/>
>>>>>>
>>>>>> + Directory Configruation:
>>>>>>
>>>>>> <param name="dial-string"
>>>>>> value="{rtp_secure_media=${regex(${sofia_contact(${dialed_user}@
>>>>>> ${dialed_domain})}|transport=tls)},presence_id=${dialed_user}@
>>>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"
>>>>>> />
>>>>>>
>>>>>> My question is that, is there any configuration left that I have to
>>>>>> set up in order to let freeswitch initiate an outbound call to Leg B
>>>>>> correctly with SRTP and SIPs (tls)?
>>>>>>
>>>>>> Any help would be really appreciated.
>>>>>> Thank you so much.
>>>>>> Best Regard,
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> 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 <(918)%20424-9378>
>>>>>
>>>>> 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>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>>>
>>> _________________________________________________________________________
>> 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
>
> _________________________________________________________________________
> 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 mouli123 at gmail.com  Mon Oct 22 06:24:43 2018
From: mouli123 at gmail.com (Chandramouli P)
Date: Mon, 22 Oct 2018 11:54:43 +0530
Subject: [Freeswitch-users] DTMF Issue - Unexpected digits are receiving
In-Reply-To: <CAFGRPVoB-6VXTVvO9oJthH-38ky0n5PQHZdrih1WTkyyoHML_Q@mail.gmail.com>
References: <CALR0W682r3CHbf4S1ZqVTZ+f3-+LnySXXrz9Z_ukb7L5s_mkqg@mail.gmail.com>
 <CAFGRPVoB-6VXTVvO9oJthH-38ky0n5PQHZdrih1WTkyyoHML_Q@mail.gmail.com>
Message-ID: <CALR0W69X_oLB8HT-874vQDZadR8sQwPYjUEuWc5AvZRjE996yA@mail.gmail.com>

Hello David,

Thanks for your reply and I understand. But, As I explained, we tried to
regenerate the issue by pressing different keys on dial pad. But,
FreeSwitch is receiving the digits as usual every time. So, which key was
pressed by the user on dial pad to send "A"? Any assumptions?

Thank you.

Best regards,
Chandramouli.

On Sun, Oct 21, 2018 at 10:53 PM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> From what i know, "A" is a multifrequency tone, 697hz+1633hz. Maybe the
> client is actually sending that.
>
> On Sat, Oct 20, 2018, 17:54 Chandramouli P <mouli123 at gmail.com> wrote:
>
>> Hello,
>>
>> We developed an application (FreeSwitch) that receives pressed digit by
>> the end user on mobile phone dialer through DTMF and some action will be
>> performed. Everything is working fine till now. But, we noticed one issue
>> recently. From the FreeSwitch log, we came to know that FreeSwitch is
>> received the letter i.e. "A" from DTMF, instead of digit. We tried to
>> regenerate this issue. But, when we are trying to regenerate, FreeSwitch is
>> receiving the digit from DTMF as usual. I would like to know How FreeSwitch
>> is received the letter i.e. "A", instead of digit. Which key was pressed by
>> the end user on mobile phone dialer?
>>
>> Thank you.
>>
>> Best Regards,
>> Chandramouli.
>> _________________________________________________________________________
>> 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
>
> _________________________________________________________________________
> 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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Virus-free.
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From david.villasmil.work at gmail.com  Mon Oct 22 11:03:49 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 22 Oct 2018 12:03:49 +0100
Subject: [Freeswitch-users] DTMF Issue - Unexpected digits are receiving
In-Reply-To: <CALR0W69X_oLB8HT-874vQDZadR8sQwPYjUEuWc5AvZRjE996yA@mail.gmail.com>
References: <CALR0W682r3CHbf4S1ZqVTZ+f3-+LnySXXrz9Z_ukb7L5s_mkqg@mail.gmail.com>
 <CAFGRPVoB-6VXTVvO9oJthH-38ky0n5PQHZdrih1WTkyyoHML_Q@mail.gmail.com>
 <CALR0W69X_oLB8HT-874vQDZadR8sQwPYjUEuWc5AvZRjE996yA@mail.gmail.com>
Message-ID: <CAFGRPVrFLSwJJVq56B3SLPYWG=4+pSCVoZj-_zeEBwE3mRbndA@mail.gmail.com>

I guess that depends where the dtmf is coming from, is it your client? Is
the client actually capable of sending that? You need traces.

On Mon, Oct 22, 2018, 09:43 Chandramouli P <mouli123 at gmail.com> wrote:

> Hello David,
>
> Thanks for your reply and I understand. But, As I explained, we tried to
> regenerate the issue by pressing different keys on dial pad. But,
> FreeSwitch is receiving the digits as usual every time. So, which key was
> pressed by the user on dial pad to send "A"? Any assumptions?
>
> Thank you.
>
> Best regards,
> Chandramouli.
>
>
> On Sun, Oct 21, 2018 at 10:53 PM David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> From what i know, "A" is a multifrequency tone, 697hz+1633hz. Maybe the
>> client is actually sending that.
>>
>> On Sat, Oct 20, 2018, 17:54 Chandramouli P <mouli123 at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> We developed an application (FreeSwitch) that receives pressed digit by
>>> the end user on mobile phone dialer through DTMF and some action will be
>>> performed. Everything is working fine till now. But, we noticed one issue
>>> recently. From the FreeSwitch log, we came to know that FreeSwitch is
>>> received the letter i.e. "A" from DTMF, instead of digit. We tried to
>>> regenerate this issue. But, when we are trying to regenerate, FreeSwitch is
>>> receiving the digit from DTMF as usual. I would like to know How FreeSwitch
>>> is received the letter i.e. "A", instead of digit. Which key was pressed by
>>> the end user on mobile phone dialer?
>>>
>>> Thank you.
>>>
>>> Best Regards,
>>> Chandramouli.
>>> _________________________________________________________________________
>>> 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
>>
>> _________________________________________________________________________
>> 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
>
>
>
> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> Virus-free.
> www.avg.com
> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
> <#m_-6596582178606799753_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
> _________________________________________________________________________
> 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
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From mouli123 at gmail.com  Mon Oct 22 11:58:09 2018
From: mouli123 at gmail.com (Chandramouli P)
Date: Mon, 22 Oct 2018 17:28:09 +0530
Subject: [Freeswitch-users] DTMF Issue - Unexpected digits are receiving
In-Reply-To: <CAFGRPVrFLSwJJVq56B3SLPYWG=4+pSCVoZj-_zeEBwE3mRbndA@mail.gmail.com>
References: <CALR0W682r3CHbf4S1ZqVTZ+f3-+LnySXXrz9Z_ukb7L5s_mkqg@mail.gmail.com>
 <CAFGRPVoB-6VXTVvO9oJthH-38ky0n5PQHZdrih1WTkyyoHML_Q@mail.gmail.com>
 <CALR0W69X_oLB8HT-874vQDZadR8sQwPYjUEuWc5AvZRjE996yA@mail.gmail.com>
 <CAFGRPVrFLSwJJVq56B3SLPYWG=4+pSCVoZj-_zeEBwE3mRbndA@mail.gmail.com>
Message-ID: <CALR0W6_Bfy=6=K76yzgGB3Wb=xPoJLJ-RbSd=R_R5a_F0NPspw@mail.gmail.com>

Hello David,

We developed a simple audio conferencing application with the help of
FreeSwitch module. Simply, an end user (Host) can call a PSTN number from
their mobile phone and group of people can join in the conference call as
usual. With respect to this scenario, once the call is bridged, the host
can press "0" to mute all other participants in the conference. Here users
or host use regular mobile phone dialer only. In this scenario, host
pressed something on mobile phone dialer instead of "0", and FreeSwitch
received the pressed input by the host as "A". I hope you understand. We
have gone through the log and didn't find any useful hint to know which key
(s) is pressed by the host.

*Please find the here: *https://pastebin.com/3NBU1xyu

Thank you.

Best regards,
Chandramouli.


On Mon, Oct 22, 2018 at 5:02 PM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> I guess that depends where the dtmf is coming from, is it your client? Is
> the client actually capable of sending that? You need traces.
>
> On Mon, Oct 22, 2018, 09:43 Chandramouli P <mouli123 at gmail.com> wrote:
>
>> Hello David,
>>
>> Thanks for your reply and I understand. But, As I explained, we tried to
>> regenerate the issue by pressing different keys on dial pad. But,
>> FreeSwitch is receiving the digits as usual every time. So, which key was
>> pressed by the user on dial pad to send "A"? Any assumptions?
>>
>> Thank you.
>>
>> Best regards,
>> Chandramouli.
>>
>>
>> On Sun, Oct 21, 2018 at 10:53 PM David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> From what i know, "A" is a multifrequency tone, 697hz+1633hz. Maybe the
>>> client is actually sending that.
>>>
>>> On Sat, Oct 20, 2018, 17:54 Chandramouli P <mouli123 at gmail.com> wrote:
>>>
>>>> Hello,
>>>>
>>>> We developed an application (FreeSwitch) that receives pressed digit by
>>>> the end user on mobile phone dialer through DTMF and some action will be
>>>> performed. Everything is working fine till now. But, we noticed one issue
>>>> recently. From the FreeSwitch log, we came to know that FreeSwitch is
>>>> received the letter i.e. "A" from DTMF, instead of digit. We tried to
>>>> regenerate this issue. But, when we are trying to regenerate, FreeSwitch is
>>>> receiving the digit from DTMF as usual. I would like to know How FreeSwitch
>>>> is received the letter i.e. "A", instead of digit. Which key was pressed by
>>>> the end user on mobile phone dialer?
>>>>
>>>> Thank you.
>>>>
>>>> Best Regards,
>>>> Chandramouli.
>>>>
>>>> _________________________________________________________________________
>>>> 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
>>>
>>> _________________________________________________________________________
>>> 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
>>
>>
>>
>> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> Virus-free.
>> www.avg.com
>> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
>> <#m_-956368409844694306_m_-6596582178606799753_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>> _________________________________________________________________________
>> 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
>
> _________________________________________________________________________
> 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 enp at itx.ru  Mon Oct 22 12:00:35 2018
From: enp at itx.ru (Eugene Prokopiev)
Date: Mon, 22 Oct 2018 15:00:35 +0300
Subject: [Freeswitch-users] How to play some file into parked call via
 inbound ESL connection?
In-Reply-To: <CAEJMVkCUYa6YM3usY66j8KudXUuBYgQb-3h50b0Pfs13rL3=8g@mail.gmail.com>
References: <CANM4RujtOKRx1FQQCqbjcBzj7NepQFwxuJTeFinDJjPP60r5Ag@mail.gmail.com>
 <CAEJMVkCUYa6YM3usY66j8KudXUuBYgQb-3h50b0Pfs13rL3=8g@mail.gmail.com>
Message-ID: <CANM4RujRzuUn91=rtUSnvCOSGhMLjB8oUUGBZtZrVqDt4srUBQ@mail.gmail.com>

Can you explain what is the difference between api/bgapi uuid_kill <uuid>
and sendmsg <uuid> \n call-command: hangup? I guess one is low level abd
another is some abstraction layer with additional featutes, right?

чт, 18 окт. 2018 г. в 13:15, Brian West <brian at freeswitch.com>:

> You don’t have to use that method,
>
> sendmsg: <uuid>
> call-command: execute
> exec-app-name: playback
> exec-app-args: file.wav
> event-lock:true
>
>
>
>
> On Wed, Oct 17, 2018 at 7:02 AM Eugene Prokopiev <enp at itx.ru> wrote:
>
>> Hi,
>>
>> Is it possible to play some file into parked call via inbound ESL
>> connection? I have dialplan like this:
>>
>>     <extension name="in">
>>       <condition>
>>         <action application="park"/>
>>       </condition>
>>     </extension>
>>
>> I tried:
>>
>> $ telnet localhost 8021
>>
>> auth ClueCon
>>
>> Content-Type: command/reply
>> Reply-Text: +OK accepted
>>
>> event plain CHANNEL_PARK
>>
>> Content-Type: command/reply
>> Reply-Text: +OK event listener enabled plain
>>
>> Content-Length: 5397
>> Content-Type: text/event-plain
>>
>> Event-Name: CHANNEL_PARK
>> ...
>> Unique-ID: 783e9456-d1f2-11e8-a77f-95321f80797f
>> ...
>>
>> api uuid_answer 783e9456-d1f2-11e8-a77f-95321f80797f
>>
>> Content-Type: api/response
>> Content-Length: 4
>>
>> +OK
>>
>> api uuid_displace 783e9456-d1f2-11e8-a77f-95321f80797f start
>> /usr/share/freeswitch/sounds/ru/RU/elena/ivr/8000/ivr-welcome.wav
>>
>> Content-Type: api/response
>> Content-Length: 12
>>
>> +OK Success
>>
>> api uuid_broadcast 783e9456-d1f2-11e8-a77f-95321f80797f
>>
>> playback::/usr/share/freeswitch/sounds/ru/RU/elena/ivr/8000/ivr-welcome.wav
>>
>> Content-Type: api/response
>> Content-Length: 17
>>
>> +OK Message sent
>>
>> api uuid_kill 783e9456-d1f2-11e8-a77f-95321f80797f
>>
>> Content-Type: api/response
>> Content-Length: 4
>>
>> So, both api uuid_displace and api uuid_broadcast returns immediately
>> with +OK. Is it possible to play and wait until the end of file? Or
>> this is possible via sendmsg instead of api and suscribing on
>> PLAYBACK_STOP event?
>>
>> --
>> WBR,
>> Eugene Prokopiev
>>
>> _________________________________________________________________________
>> 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>
> _________________________________________________________________________
> 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



-- 
WBR,
Eugene Prokopiev
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From vma at vallimamod.org  Mon Oct 22 12:24:53 2018
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Mon, 22 Oct 2018 14:24:53 +0200
Subject: [Freeswitch-users] DTMF Issue - Unexpected digits are receiving
In-Reply-To: <CALR0W69X_oLB8HT-874vQDZadR8sQwPYjUEuWc5AvZRjE996yA@mail.gmail.com>
References: <CALR0W682r3CHbf4S1ZqVTZ+f3-+LnySXXrz9Z_ukb7L5s_mkqg@mail.gmail.com>
 <CAFGRPVoB-6VXTVvO9oJthH-38ky0n5PQHZdrih1WTkyyoHML_Q@mail.gmail.com>
 <CALR0W69X_oLB8HT-874vQDZadR8sQwPYjUEuWc5AvZRjE996yA@mail.gmail.com>
Message-ID: <7F36BBCD-D236-45EE-B0A0-F7DE031AC1E6@vallimamod.org>

Hi,

I encountered the same issue in the past and have noticed that it was the mobile voicemail beep tone that was sometimes converted to 'A' dtmf on the isdn gateway.

Maybe your case is similar?


Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sip.solutions
linkedin.com/in/vallimamod
.


> On 22 Oct 2018, at 08:24, Chandramouli P <mouli123 at gmail.com> wrote:
> 
> Hello David,
> 
> Thanks for your reply and I understand. But, As I explained, we tried to regenerate the issue by pressing different keys on dial pad. But, FreeSwitch is receiving the digits as usual every time. So, which key was pressed by the user on dial pad to send "A"? Any assumptions?
> 
> Thank you.
> 
> Best regards,
> Chandramouli.
> 
> On Sun, Oct 21, 2018 at 10:53 PM David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
> From what i know, "A" is a multifrequency tone, 697hz+1633hz. Maybe the client is actually sending that.
> 
> 
> On Sat, Oct 20, 2018, 17:54 Chandramouli P <mouli123 at gmail.com <mailto:mouli123 at gmail.com>> wrote:
> Hello,
> 
> We developed an application (FreeSwitch) that receives pressed digit by the end user on mobile phone dialer through DTMF and some action will be performed. Everything is working fine till now. But, we noticed one issue recently. From the FreeSwitch log, we came to know that FreeSwitch is received the letter i.e. "A" from DTMF, instead of digit. We tried to regenerate this issue. But, when we are trying to regenerate, FreeSwitch is receiving the digit from DTMF as usual. I would like to know How FreeSwitch is received the letter i.e. "A", instead of digit. Which key was pressed by the end user on mobile phone dialer?
> 
> Thank you.
> 
> Best Regards,
> Chandramouli.

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From social at bohboh.info  Mon Oct 22 13:39:42 2018
From: social at bohboh.info (Social Boh)
Date: Mon, 22 Oct 2018 08:39:42 -0500
Subject: [Freeswitch-users] CentOS 7, FreeSWITCH 1.8.2, mod_flite error
In-Reply-To: <CAHtxdDcbCjxAX0nf0C3W+06BGgP_QzNTrUWqY0wgoMWd_JRAyA@mail.gmail.com>
References: <fcc23d99-f3bf-a4bc-49cd-aa63eb598b9d@bohboh.info>
 <CAHtxdDcbCjxAX0nf0C3W+06BGgP_QzNTrUWqY0wgoMWd_JRAyA@mail.gmail.com>
Message-ID: <06fd3b74-be95-9d4f-9f7d-628117c7d475@bohboh.info>

Hello,

I have read that page but I still do not understand what the solution is.

Thank you

Regards

---
I'm SoCIaL, MayBe

El 20/10/2018 a las 13:07, Sergey Safarov escribió:
> https://freeswitch.org/jira/browse/FS-8084
>
> сб, 20 окт. 2018 г. в 20:07, Social Boh <social at bohboh.info 
> <mailto:social at bohboh.info>>:
>
>     Hello list,
>
>     I compiled FreeSWITCH 1.8.2+git~20180926T175525Z~a98a958ac3~64bit
>     en CentOS 7.5.1804 without problem with flite-devel version
>     2.0.0-1.el7.
>
>     Now, when I try to load the module:
>
>     /[CRIT] switch_loadable_module.c:1522 Error Loading module
>     /usr/lib/freeswitch/mod/mod_flite.so//
>     //**/lib64/libflite.so.1: undefined symbol: snd_pcm_hw_params_any**/
>
>     Any Hint?
>
>     Thank you
>
>     Regards
>
>     -- 
>     ---
>     I'm SoCIaL, MayBe
>
>     _________________________________________________________________________
>     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
>
>
>
> _________________________________________________________________________
> 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 brian at freeswitch.com  Mon Oct 22 19:01:22 2018
From: brian at freeswitch.com (Brian West)
Date: Mon, 22 Oct 2018 14:01:22 -0500
Subject: [Freeswitch-users] Sometimes "hanging" channels
In-Reply-To: <C27A69BA-D5E4-4B30-8053-E53C34E585FA@vallimamod.org>
References: <87D934A1-4A37-4709-93B5-96FA0659ABBE@gmx.net>
 <CAFiqYun3i3NDvqR4pkvfDCquQNMALXXAA-fp2EPm6bXx0VuVmg@mail.gmail.com>
 <1E3CBAFE-0B7B-445C-AAF2-EF4C074392B7@gmx.net>
 <C27A69BA-D5E4-4B30-8053-E53C34E585FA@vallimamod.org>
Message-ID: <CAEJMVkCD1Rd-sTOwovGpq15RQjt280wJOrrK+yDK9cnmTukioA@mail.gmail.com>

It;s not that it was never received, it was probably deadlocked and never
sent.

/b


On Mon, Oct 15, 2018 at 7:46 AM Vallimamod Abdullah <vma at vallimamod.org>
wrote:

> Hi,
>
> The message "Locked, Waiting on external entities" is not alarming by
> itself as it is always logged on hangup. But it should be followed by:
>
> [NOTICE] switch_core_session.c:1683 Session XXX (sofia/internal/YYY) Ended
> [NOTICE] switch_core_session.c:1687 Close Channel sofia/internal/YYY
> [CS_DESTROY]
>
> If it is not the case then, you have an app or a module that is holding a
> lock on the session and preventing its release.
>
> And this happens after the reporting state i.e. after the hangup_complete
> event and CDR posting. So there may be other causes for your missing events.
>
>
>
> Best Regards,
> --
> Vallimamod Abdullah
> SIP Solutions
> vma at sip.solutions
> linkedin.com/in/vallimamod
> .
>
>
> > On 10 Oct 2018, at 15:54, Markus Bönke <mbodbg at gmx.net> wrote:
> >
> > So far I found out that the channel was kept open, because our ESL
> application never received a CHANNEL_HANGUP and CHANNEL_HANGUP_COMPLETE
> event. I’m running an ESL trace with tshark now. If it happens again I can
> see if those events are really not send by FS or they get lost in our app.
> If it turns out that FS sometimes is not sending the events we will upgrade
> to 1.8.2.
> >
> > Thanks and regards
> >
> > Markus
> >
> >
> >> Am 10.10.2018 um 14:03 schrieb Steven Ayre <steveayre at gmail.com>:
> >>
> >> If you take a gcore it'll generate a core dump. You could then dig into
> it with gdb to find the thread for that channel and see what lock it's
> waiting on.
> >> However the gcore will pause the process for a while, so it will impact
> on any other calls on that box.
> >>
> >> Have you tried reproducing it on 1.8?
> >>
> >> On Tue, 9 Oct 2018 at 20:36, Markus Bönke <mbodbg at gmx.net> wrote:
> >> Hello All,
> >>
> >> we are running freeswitch 1.6.20, calls are controlled via ESL, CDRs
> are written with mod_xml_cdrl. Sometimes we see „hanging“ channels. In such
> a case the CDR via mod_xml_cdr is written and the last log entry for such a
> call is "Locked, Waiting on external entities“.
> >> ...
> >>
> freeswitch.log.2018-10-09-10-19-07.1:db2908e3-c06b-4293-88b1-e18cffb66263
> 2018-10-09 10:18:15.197345 [DEBUG] switch_core_state_machine.c:610
> (sofia/internal/anonymous at anonymous.invalid) State Change CS_REPORTING ->
> CS_DESTROY
> >>
> freeswitch.log.2018-10-09-10-19-07.1:db2908e3-c06b-4293-88b1-e18cffb66263
> 2018-10-09 10:18:15.197345 [DEBUG] switch_core_session.c:1665 Session
> 166111 (sofia/internal/anonymous at anonymous.invalid) Locked, Waiting on
> external entities
> >> ...
> >> How can I proceed to further analyze the problem ? In the last log line
> I also see the session number (Session 166111) - is there a way to find out
> on which external entity it is waiting?
> >>
> >> Thanks and regards
> >>
> >> Markus
> >>
> _________________________________________________________________________
> >> 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
> >>
> _________________________________________________________________________
> >> 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
> >
> > _________________________________________________________________________
> > 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
>
>
> _________________________________________________________________________
> 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  Mon Oct 22 19:44:48 2018
From: s.safarov at gmail.com (Sergey Safarov)
Date: Mon, 22 Oct 2018 22:44:48 +0300
Subject: [Freeswitch-users] CentOS 7, FreeSWITCH 1.8.2, mod_flite error
In-Reply-To: <06fd3b74-be95-9d4f-9f7d-628117c7d475@bohboh.info>
References: <fcc23d99-f3bf-a4bc-49cd-aa63eb598b9d@bohboh.info>
 <CAHtxdDcbCjxAX0nf0C3W+06BGgP_QzNTrUWqY0wgoMWd_JRAyA@mail.gmail.com>
 <06fd3b74-be95-9d4f-9f7d-628117c7d475@bohboh.info>
Message-ID: <CAHtxdDf=c3zUUh2ESsD3LZQxixTaYQd+uL4959Js+1JH2+ZN8g@mail.gmail.com>

you have two option
1) wait when issue in fixed on FreeSwitch repo;
2) create own flite rpm file and then compile FreeSwitch RPM by self.

Sergey

пн, 22 окт. 2018 г. в 22:28, Social Boh <social at bohboh.info>:

> Hello,
>
> I have read that page but I still do not understand what the solution is.
>
> Thank you
>
> Regards
>
> ---
> I'm SoCIaL, MayBe
>
> El 20/10/2018 a las 13:07, Sergey Safarov escribió:
>
> https://freeswitch.org/jira/browse/FS-8084
>
> сб, 20 окт. 2018 г. в 20:07, Social Boh <social at bohboh.info>:
>
>> Hello list,
>>
>> I compiled FreeSWITCH 1.8.2+git~20180926T175525Z~a98a958ac3~64bit en
>> CentOS 7.5.1804 without problem with flite-devel version 2.0.0-1.el7.
>>
>> Now, when I try to load the module:
>>
>> *[CRIT] switch_loadable_module.c:1522 Error Loading module
>> /usr/lib/freeswitch/mod/mod_flite.so*
>> ***/lib64/libflite.so.1: undefined symbol: snd_pcm_hw_params_any***
>>
>> Any Hint?
>>
>> Thank you
>>
>> Regards
>>
>> --
>> ---
>> I'm SoCIaL, MayBe
>>
>> _________________________________________________________________________
>> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com
>
> Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com
>
>
> _________________________________________________________________________
> 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 chad at apartmentlines.com  Tue Oct 23 02:16:16 2018
From: chad at apartmentlines.com (Chad Phillips)
Date: Mon, 22 Oct 2018 19:16:16 -0700
Subject: [Freeswitch-users] Playing file into videoconference via mod_av
	freezes after server reboot
Message-ID: <CAJ0sghRWPnRHu5xxx5JftFymh7R+7QOMfKKPwxjC2KrmUFW0Cw@mail.gmail.com>

FreeSWITCH 1.8.2 on latest Debian Stretch, compiled stock from source, deps
installed via FS repo.

I have FreeSWITCH managed by systemd to start at boot, which works well.
However, I'm hitting a weird issue where, after initial system boot, if I
try to play a video file into a conference via mod_av, the video is frozen
on the first frame (audio is fine). If I restart FreeSWITCH, then the video
file plays fine.

Any thoughts on what might be causing this? My only guess: on boot systemd
starts FreeSWITCH before some other service or setup that is needed to play
video files via mod_av, and restarting FreeSWITCH fixes the issue because
said service is now running.
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From mouli123 at gmail.com  Tue Oct 23 06:23:24 2018
From: mouli123 at gmail.com (Chandramouli P)
Date: Tue, 23 Oct 2018 11:53:24 +0530
Subject: [Freeswitch-users] DTMF Issue - Unexpected digits are receiving
In-Reply-To: <7F36BBCD-D236-45EE-B0A0-F7DE031AC1E6@vallimamod.org>
References: <CALR0W682r3CHbf4S1ZqVTZ+f3-+LnySXXrz9Z_ukb7L5s_mkqg@mail.gmail.com>
 <CAFGRPVoB-6VXTVvO9oJthH-38ky0n5PQHZdrih1WTkyyoHML_Q@mail.gmail.com>
 <CALR0W69X_oLB8HT-874vQDZadR8sQwPYjUEuWc5AvZRjE996yA@mail.gmail.com>
 <7F36BBCD-D236-45EE-B0A0-F7DE031AC1E6@vallimamod.org>
Message-ID: <CALR0W6-yvbqFncMx6aXG8QZmXHF_PCXP-cZJZeuk1hzfAVL0=A@mail.gmail.com>

Hello Abdullah,

Thank you for your reply. By following your hint, I tried to regenerate the
issue. I tried by getting voice mail beep, other beeps like receiving SMS,
WhatsApp messages etc. But, unable to simulate.

Thank you.

Best regards,
Chandramouli.

On Mon, Oct 22, 2018 at 8:00 PM Vallimamod Abdullah <vma at vallimamod.org>
wrote:

> Hi,
>
> I encountered the same issue in the past and have noticed that it was the
> mobile voicemail beep tone that was sometimes converted to 'A' dtmf on the
> isdn gateway.
>
> Maybe your case is similar?
>
>
> Best Regards,
> --
> Vallimamod Abdullah
> SIP Solutions
> vma at sip.solutions
> linkedin.com/in/vallimamod
> .
>
>
> On 22 Oct 2018, at 08:24, Chandramouli P <mouli123 at gmail.com> wrote:
>
> Hello David,
>
> Thanks for your reply and I understand. But, As I explained, we tried to
> regenerate the issue by pressing different keys on dial pad. But,
> FreeSwitch is receiving the digits as usual every time. So, which key was
> pressed by the user on dial pad to send "A"? Any assumptions?
>
> Thank you.
>
> Best regards,
> Chandramouli.
>
> On Sun, Oct 21, 2018 at 10:53 PM David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> From what i know, "A" is a multifrequency tone, 697hz+1633hz. Maybe the
>> client is actually sending that.
>>
>> On Sat, Oct 20, 2018, 17:54 Chandramouli P <mouli123 at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> We developed an application (FreeSwitch) that receives pressed digit by
>>> the end user on mobile phone dialer through DTMF and some action will be
>>> performed. Everything is working fine till now. But, we noticed one issue
>>> recently. From the FreeSwitch log, we came to know that FreeSwitch is
>>> received the letter i.e. "A" from DTMF, instead of digit. We tried to
>>> regenerate this issue. But, when we are trying to regenerate, FreeSwitch is
>>> receiving the digit from DTMF as usual. I would like to know How FreeSwitch
>>> is received the letter i.e. "A", instead of digit. Which key was pressed by
>>> the end user on mobile phone dialer?
>>>
>>> Thank you.
>>>
>>> Best Regards,
>>> Chandramouli.
>>>
>>
> _________________________________________________________________________
> 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 asilva at wirelessmundi.com  Tue Oct 23 11:41:13 2018
From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=)
Date: Tue, 23 Oct 2018 13:41:13 +0200
Subject: [Freeswitch-users] h264 video call between endpoints with one end
 at bad resolution of 352x288
In-Reply-To: <C1EF7F9E-1917-42BC-9F90-FEAC5A673738@jerris.com>
References: <mailman.1680.1530191219.1211.freeswitch-users@lists.freeswitch.org>
 <C1EF7F9E-1917-42BC-9F90-FEAC5A673738@jerris.com>
Message-ID: <747be107-2baf-96d4-d433-d5cd04faa826@wirelessmundi.com>

Hi,

i found the issue for this problem, it was because the fmtp parameters 
weren't pass to the bleg when using late-negotiation, so the other end 
start sending the video at lower resolution.

I solve by exporting the first fmtp from the aleg to the bleg channel 
using the variable rtp_video_fmtp .


This issue it's supposed to be fixed in 
https://freeswitch.org/jira/browse/FS-5958 but is present in current 
master and 1.8 branch.

More info can be found in https://freeswitch.org/jira/browse/FS-11140.


On 11/07/2018 22:06, Michael Jerris wrote:
> sounds like you might be doing the option to mirror resolution
>
>
>> On Jun 28, 2018, at 9:07 AM, antonio via FreeSWITCH-users 
>> <freeswitch-users at lists.freeswitch.org 
>> <mailto:freeswitch-users at lists.freeswitch.org>> wrote:
>>
>>
>> *From: *antonio <asilva at wirelessmundi.com 
>> <mailto:asilva at wirelessmundi.com>>
>> *Subject: **h264 video call between endpoints with one end at bad 
>> resolution of 352x288*
>> *Date: *June 26, 2018 at 6:05:20 AM EDT
>> *To: *freeswitch-users at lists.freeswitch.org 
>> <mailto:freeswitch-users at lists.freeswitch.org>
>>
>>
>> Hi,
>>
>> Calling between two Grandstream GXV3275 i expected to have a video 
>> call at 1280x720, but the video that is present at destination as the 
>> resolution of 352x288 and the image as poor quality, originator is 
>> sending video at 1280×720.  If i swap the devices the result is the 
>> same, the destination always get a bad resolution of 352x288 but the 
>> originator receives the video at expected resolution of 1280x720.
>>
>> If i set rtp_direct between the two endpoints it work as expected.
>>
>> I also tested a conference room and force the it resolution to 
>> 1280*x720 it work well,  if i don't force the canvas-size i see:
>>
>> 2018-06-26 13:55:02.077824 [WARNING] mod_conference.c:3157 
>> Unspecified video-canvas-size, falling back to 1280x720
>>
>> And the conference is done at 1280x720.
>>
>>
>> I've open a jira where you can find a full log with this behavior:
>>
>> https://freeswitch.org/jira/browse/FS-11140
>>
>>
>> I don't understand why there is transcoding and mod_av is used, the 
>> codec negotiated is the same.. i understand that the resolution is at 
>> 352x288 for the first frames sent by the phone but then it send them 
>> at 1280x720,  mod_av should detected and adjust the new resolution, no?
>>
>>
>> Anyone experience this behavior?
>>
>
-- 
Saludos / Regards / Cumprimentos
António Silva

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From BJordan at E-Teleco.com  Tue Oct 23 19:19:09 2018
From: BJordan at E-Teleco.com (Branden Jordan)
Date: Tue, 23 Oct 2018 19:19:09 +0000
Subject: [Freeswitch-users] DTMF Issue - Unexpected digits are receiving
In-Reply-To: <CALR0W6-yvbqFncMx6aXG8QZmXHF_PCXP-cZJZeuk1hzfAVL0=A@mail.gmail.com>
References: <CALR0W682r3CHbf4S1ZqVTZ+f3-+LnySXXrz9Z_ukb7L5s_mkqg@mail.gmail.com>
 <CAFGRPVoB-6VXTVvO9oJthH-38ky0n5PQHZdrih1WTkyyoHML_Q@mail.gmail.com>
 <CALR0W69X_oLB8HT-874vQDZadR8sQwPYjUEuWc5AvZRjE996yA@mail.gmail.com>
 <7F36BBCD-D236-45EE-B0A0-F7DE031AC1E6@vallimamod.org>
 <CALR0W6-yvbqFncMx6aXG8QZmXHF_PCXP-cZJZeuk1hzfAVL0=A@mail.gmail.com>
Message-ID: <51dc1d4b2d63413285607f7c6ed801e4@MDF-EXCH1.MDF-Holdings.local>

A is a valid DTMF digit so I wouldn’t worry about it too much and just handle A-D in your valid TT fields. I see around a million calls a day and see a few hundred A-D DTMF. It’s rare, but some really old phones still have the A-D DTMF on them, and softphones have the ability to send them as well, maybe your user is using a softphone on their end?

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Chandramouli P
Sent: Monday, October 22, 2018 11:23 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] DTMF Issue - Unexpected digits are receiving

Hello Abdullah,

Thank you for your reply. By following your hint, I tried to regenerate the issue. I tried by getting voice mail beep, other beeps like receiving SMS, WhatsApp messages etc. But, unable to simulate.

Thank you.

Best regards,
Chandramouli.
On Mon, Oct 22, 2018 at 8:00 PM Vallimamod Abdullah <vma at vallimamod.org<mailto:vma at vallimamod.org>> wrote:
Hi,

I encountered the same issue in the past and have noticed that it was the mobile voicemail beep tone that was sometimes converted to 'A' dtmf on the isdn gateway.

Maybe your case is similar?


Best Regards,
--
Vallimamod Abdullah
SIP Solutions
vma at sip.solutions<mailto:vma at sip.solutions>
linkedin.com/in/vallimamod<http://linkedin.com/in/vallimamod>
.


On 22 Oct 2018, at 08:24, Chandramouli P <mouli123 at gmail.com<mailto:mouli123 at gmail.com>> wrote:

Hello David,

Thanks for your reply and I understand. But, As I explained, we tried to regenerate the issue by pressing different keys on dial pad. But, FreeSwitch is receiving the digits as usual every time. So, which key was pressed by the user on dial pad to send "A"? Any assumptions?

Thank you.

Best regards,
Chandramouli.
On Sun, Oct 21, 2018 at 10:53 PM David Villasmil <david.villasmil.work at gmail.com<mailto:david.villasmil.work at gmail.com>> wrote:

From what i know, "A" is a multifrequency tone, 697hz+1633hz. Maybe the client is actually sending that.

On Sat, Oct 20, 2018, 17:54 Chandramouli P <mouli123 at gmail.com<mailto:mouli123 at gmail.com>> wrote:
Hello,

We developed an application (FreeSwitch) that receives pressed digit by the end user on mobile phone dialer through DTMF and some action will be performed. Everything is working fine till now. But, we noticed one issue recently. From the FreeSwitch log, we came to know that FreeSwitch is received the letter i.e. "A" from DTMF, instead of digit. We tried to regenerate this issue. But, when we are trying to regenerate, FreeSwitch is receiving the digit from DTMF as usual. I would like to know How FreeSwitch is received the letter i.e. "A", instead of digit. Which key was pressed by the end user on mobile phone dialer?

Thank you.

Best Regards,
Chandramouli.

_________________________________________________________________________
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 mouli123 at gmail.com  Wed Oct 24 05:52:13 2018
From: mouli123 at gmail.com (Chandramouli P)
Date: Wed, 24 Oct 2018 11:22:13 +0530
Subject: [Freeswitch-users] DTMF Issue - Unexpected digits are receiving
In-Reply-To: <51dc1d4b2d63413285607f7c6ed801e4@MDF-EXCH1.MDF-Holdings.local>
References: <CALR0W682r3CHbf4S1ZqVTZ+f3-+LnySXXrz9Z_ukb7L5s_mkqg@mail.gmail.com>
 <CAFGRPVoB-6VXTVvO9oJthH-38ky0n5PQHZdrih1WTkyyoHML_Q@mail.gmail.com>
 <CALR0W69X_oLB8HT-874vQDZadR8sQwPYjUEuWc5AvZRjE996yA@mail.gmail.com>
 <7F36BBCD-D236-45EE-B0A0-F7DE031AC1E6@vallimamod.org>
 <CALR0W6-yvbqFncMx6aXG8QZmXHF_PCXP-cZJZeuk1hzfAVL0=A@mail.gmail.com>
 <51dc1d4b2d63413285607f7c6ed801e4@MDF-EXCH1.MDF-Holdings.local>
Message-ID: <CALR0W68_=W2NMiB+Fe9VsA54ixkwBTjFT=SycZ_uZP=NFWJ49A@mail.gmail.com>

Hello Branden,

Thank you for your reply. As you said, I do understand about A-D DTMF
digits. But, my problem is to regenerate the issue i.e. ("A"). Which key
(s), I have to press to send "A" as DTMF digit? I am sure that the end user
is used regular mobile phone dialer only, and not softphone.

Thank you.

Best Regards,
Chandramouli.

On Wed, Oct 24, 2018 at 2:58 AM Branden Jordan <BJordan at e-teleco.com> wrote:

> A is a valid DTMF digit so I wouldn’t worry about it too much and just
> handle A-D in your valid TT fields. I see around a million calls a day and
> see a few hundred A-D DTMF. It’s rare, but some really old phones still
> have the A-D DTMF on them, and softphones have the ability to send them as
> well, maybe your user is using a softphone on their end?
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Chandramouli P
> *Sent:* Monday, October 22, 2018 11:23 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] DTMF Issue - Unexpected digits are
> receiving
>
>
>
> Hello Abdullah,
>
>
>
> Thank you for your reply. By following your hint, I tried to regenerate
> the issue. I tried by getting voice mail beep, other beeps like receiving
> SMS, WhatsApp messages etc. But, unable to simulate.
>
>
>
> Thank you.
>
>
>
> Best regards,
>
> Chandramouli.
>
> On Mon, Oct 22, 2018 at 8:00 PM Vallimamod Abdullah <vma at vallimamod.org>
> wrote:
>
> Hi,
>
>
>
> I encountered the same issue in the past and have noticed that it was the
> mobile voicemail beep tone that was sometimes converted to 'A' dtmf on the
> isdn gateway.
>
>
>
> Maybe your case is similar?
>
>
>
>
>
> Best Regards,
> --
> Vallimamod Abdullah
> SIP Solutions
> vma at sip.solutions
> linkedin.com/in/vallimamod
> .
>
>
>
> On 22 Oct 2018, at 08:24, Chandramouli P <mouli123 at gmail.com> wrote:
>
>
>
> Hello David,
>
>
>
> Thanks for your reply and I understand. But, As I explained, we tried to
> regenerate the issue by pressing different keys on dial pad. But,
> FreeSwitch is receiving the digits as usual every time. So, which key was
> pressed by the user on dial pad to send "A"? Any assumptions?
>
>
>
> Thank you.
>
>
>
> Best regards,
>
> Chandramouli.
>
> On Sun, Oct 21, 2018 at 10:53 PM David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
> From what i know, "A" is a multifrequency tone, 697hz+1633hz. Maybe the
> client is actually sending that.
>
>
>
> On Sat, Oct 20, 2018, 17:54 Chandramouli P <mouli123 at gmail.com> wrote:
>
> Hello,
>
>
>
> We developed an application (FreeSwitch) that receives pressed digit by
> the end user on mobile phone dialer through DTMF and some action will be
> performed. Everything is working fine till now. But, we noticed one issue
> recently. From the FreeSwitch log, we came to know that FreeSwitch is
> received the letter i.e. "A" from DTMF, instead of digit. We tried to
> regenerate this issue. But, when we are trying to regenerate, FreeSwitch is
> receiving the digit from DTMF as usual. I would like to know How FreeSwitch
> is received the letter i.e. "A", instead of digit. Which key was pressed by
> the end user on mobile phone dialer?
>
>
>
> Thank you.
>
>
>
> Best Regards,
>
> Chandramouli.
>
>
>
> _________________________________________________________________________
> 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
>
> _________________________________________________________________________
> 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 bilaln018 at gmail.com  Wed Oct 24 07:27:23 2018
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Wed, 24 Oct 2018 12:27:23 +0500
Subject: [Freeswitch-users] [username password with @][http cache]
Message-ID: <CADSVerCEY9GioU5_PCp2ioKS5DpZNDWcaXxgFMjAwuQfHW0vSQ@mail.gmail.com>

Hey users,
I have to download a sound file from FTP server having password with @ in
it.
The mod_http_cache has the way username:password at url, i was looking to
escape the @ from it.
This might be very basic question to ask, may be some one has done that
already.

Regards
Abbasi
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From asilva at wirelessmundi.com  Wed Oct 24 09:21:23 2018
From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=)
Date: Wed, 24 Oct 2018 11:21:23 +0200
Subject: [Freeswitch-users] [username password with @][http cache]
In-Reply-To: <CADSVerCEY9GioU5_PCp2ioKS5DpZNDWcaXxgFMjAwuQfHW0vSQ@mail.gmail.com>
References: <CADSVerCEY9GioU5_PCp2ioKS5DpZNDWcaXxgFMjAwuQfHW0vSQ@mail.gmail.com>
Message-ID: <6275c9fd-984e-dcd9-3ed9-818470b36a77@wirelessmundi.com>

you could try url encode de @, to %40

Example:

username:password%40 at url


On 24/10/2018 09:27, Bilal Abbasi wrote:
> Hey users,
> I have to download a sound file from FTP server having password with @ 
> in it.
> The mod_http_cache has the way username:password at url, i was looking to 
> escape the @ from it.
> This might be very basic question to ask, may be some one has done 
> that already.
>
> Regards
> Abbasi
>
> _________________________________________________________________________
> 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

-- 
Saludos / Regards / Cumprimentos
António Silva

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From douglas.davenport at gmail.com  Wed Oct 24 14:54:25 2018
From: douglas.davenport at gmail.com (Douglas Davenport)
Date: Wed, 24 Oct 2018 10:54:25 -0400
Subject: [Freeswitch-users] SpanDSP V34 Support
Message-ID: <CAMDsFF2r-6oo-crwkoFSn8fXoM4uPkMEOfenByow-oAbqVqrcQ@mail.gmail.com>

There appears to be a build option to add V34 faxing support to freeswitch
SpanDSP. Does anyone know if this code is functional and how to enable it?

Thanks
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From nreis at wavecom.pt  Wed Oct 24 16:37:17 2018
From: nreis at wavecom.pt (Nuno Reis)
Date: Wed, 24 Oct 2018 17:37:17 +0100
Subject: [Freeswitch-users] CentOS 7, FreeSWITCH 1.8.2, mod_flite error
In-Reply-To: <CAHtxdDf=c3zUUh2ESsD3LZQxixTaYQd+uL4959Js+1JH2+ZN8g@mail.gmail.com>
References: <fcc23d99-f3bf-a4bc-49cd-aa63eb598b9d@bohboh.info>
 <CAHtxdDcbCjxAX0nf0C3W+06BGgP_QzNTrUWqY0wgoMWd_JRAyA@mail.gmail.com>
 <06fd3b74-be95-9d4f-9f7d-628117c7d475@bohboh.info>
 <CAHtxdDf=c3zUUh2ESsD3LZQxixTaYQd+uL4959Js+1JH2+ZN8g@mail.gmail.com>
Message-ID: <CAE2+g++JzFiVe5jXV-hvsiyJArVj9jQw67dMtaRYVokNaVzZUg@mail.gmail.com>

Hi there.

The solution doesn't require you to have different freeswitch rpms.
The problem is in flite build itself. The flite rpm package in freeswitch
repo is compiled with 'configure --enable-shared --with-audio=alsa' and it
should have been 'configure --enable-shared --with-audio=none'.
If you want just recompile flite from source with 'configure
--enable-shared --with-audio=none' and replace the version that came from
freeswitch repo and your problem will be solved.

--

*Nuno Miguel Reis* | *Unified Communication** Systems*
M. +351 913907481 | nreis at wavecom.pt
WAVECOM-Soluções Rádio, S.A.
Cacia Park | Rua do Progresso, Lote 15
3800-639 AVEIRO | Portugal
T. +351 309 700 225 | F. +351 234 919 191
*GPS
<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>
| www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>*

<http://www.wavecom.pt/>
<http://www.wavecom.pt/pt/>




On Mon, Oct 22, 2018 at 9:30 PM Sergey Safarov <s.safarov at gmail.com> wrote:

> you have two option
> 1) wait when issue in fixed on FreeSwitch repo;
> 2) create own flite rpm file and then compile FreeSwitch RPM by self.
>
> Sergey
>
> пн, 22 окт. 2018 г. в 22:28, Social Boh <social at bohboh.info>:
>
>> Hello,
>>
>> I have read that page but I still do not understand what the solution is.
>>
>> Thank you
>>
>> Regards
>>
>> ---
>> I'm SoCIaL, MayBe
>>
>> El 20/10/2018 a las 13:07, Sergey Safarov escribió:
>>
>> https://freeswitch.org/jira/browse/FS-8084
>>
>> сб, 20 окт. 2018 г. в 20:07, Social Boh <social at bohboh.info>:
>>
>>> Hello list,
>>>
>>> I compiled FreeSWITCH 1.8.2+git~20180926T175525Z~a98a958ac3~64bit en
>>> CentOS 7.5.1804 without problem with flite-devel version 2.0.0-1.el7.
>>>
>>> Now, when I try to load the module:
>>>
>>> *[CRIT] switch_loadable_module.c:1522 Error Loading module
>>> /usr/lib/freeswitch/mod/mod_flite.so*
>>> ***/lib64/libflite.so.1: undefined symbol: snd_pcm_hw_params_any***
>>>
>>> Any Hint?
>>>
>>> Thank you
>>>
>>> Regards
>>>
>>> --
>>> ---
>>> I'm SoCIaL, MayBe
>>>
>>> _________________________________________________________________________
>>> 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
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com
>>
>> Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com
>>
>>
>> _________________________________________________________________________
>> 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
>
> _________________________________________________________________________
> 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 italo at freeswitch.org  Wed Oct 24 19:53:39 2018
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Wed, 24 Oct 2018 16:53:39 -0300
Subject: [Freeswitch-users] Playing file into videoconference via mod_av
 freezes after server reboot
In-Reply-To: <CAJ0sghRWPnRHu5xxx5JftFymh7R+7QOMfKKPwxjC2KrmUFW0Cw@mail.gmail.com>
References: <CAJ0sghRWPnRHu5xxx5JftFymh7R+7QOMfKKPwxjC2KrmUFW0Cw@mail.gmail.com>
Message-ID: <CABMX_w9VD+5z=0hekQx8zoafyC6FwEhZDBzyxF6VJqyqfZqkPg@mail.gmail.com>

How do you restart freeswitch? Can you compare the system user and
priorities it's running before and after the restart?
Em ter, 23 de out de 2018 às 06:08, Chad Phillips <chad at apartmentlines.com>
escreveu:

> FreeSWITCH 1.8.2 on latest Debian Stretch, compiled stock from source,
> deps installed via FS repo.
>
> I have FreeSWITCH managed by systemd to start at boot, which works well.
> However, I'm hitting a weird issue where, after initial system boot, if I
> try to play a video file into a conference via mod_av, the video is frozen
> on the first frame (audio is fine). If I restart FreeSWITCH, then the video
> file plays fine.
>
> Any thoughts on what might be causing this? My only guess: on boot systemd
> starts FreeSWITCH before some other service or setup that is needed to play
> video files via mod_av, and restarting FreeSWITCH fixes the issue because
> said service is now running.
> _________________________________________________________________________
> 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 coppice12 at gmail.com  Wed Oct 24 20:51:37 2018
From: coppice12 at gmail.com (Steve Underwood)
Date: Wed, 24 Oct 2018 21:51:37 +0100
Subject: [Freeswitch-users] SpanDSP V34 Support
In-Reply-To: <CAMDsFF2r-6oo-crwkoFSn8fXoM4uPkMEOfenByow-oAbqVqrcQ@mail.gmail.com>
References: <CAMDsFF2r-6oo-crwkoFSn8fXoM4uPkMEOfenByow-oAbqVqrcQ@mail.gmail.com>
Message-ID: <bfafd458-4342-9338-f6dd-f7b1fede288d@gmail.com>

On 10/24/2018 03:54 PM, Douglas Davenport wrote:

> There appears to be a build option to add V34 faxing support to 
> freeswitch SpanDSP. Does anyone know if this code is functional and 
> how to enable it?
There is no V.34 or V.34 FAX support in spandsp. There are some 
fragments of V.34 code, but its far from complete. There is no V.34 FAX 
code at all.

Steve


From brian at freeswitch.com  Wed Oct 24 21:11:23 2018
From: brian at freeswitch.com (Brian West)
Date: Wed, 24 Oct 2018 16:11:23 -0500
Subject: [Freeswitch-users] CentOS 7, FreeSWITCH 1.8.2, mod_flite error
In-Reply-To: <CAE2+g++JzFiVe5jXV-hvsiyJArVj9jQw67dMtaRYVokNaVzZUg@mail.gmail.com>
References: <fcc23d99-f3bf-a4bc-49cd-aa63eb598b9d@bohboh.info>
 <CAHtxdDcbCjxAX0nf0C3W+06BGgP_QzNTrUWqY0wgoMWd_JRAyA@mail.gmail.com>
 <06fd3b74-be95-9d4f-9f7d-628117c7d475@bohboh.info>
 <CAHtxdDf=c3zUUh2ESsD3LZQxixTaYQd+uL4959Js+1JH2+ZN8g@mail.gmail.com>
 <CAE2+g++JzFiVe5jXV-hvsiyJArVj9jQw67dMtaRYVokNaVzZUg@mail.gmail.com>
Message-ID: <CAEJMVkD0PJ-vRRfqBHZnAVuMVE69jxt3j6n6_5khHwrA0kTzqw@mail.gmail.com>

That sounds like a bug report, Maybe you can see if that's not already
filed in JIRA and file it?

On Wed, Oct 24, 2018 at 3:47 PM Nuno Reis <nreis at wavecom.pt> wrote:

> Hi there.
>
> The solution doesn't require you to have different freeswitch rpms.
> The problem is in flite build itself. The flite rpm package in freeswitch
> repo is compiled with 'configure --enable-shared --with-audio=alsa' and it
> should have been 'configure --enable-shared --with-audio=none'.
> If you want just recompile flite from source with 'configure
> --enable-shared --with-audio=none' and replace the version that came from
> freeswitch repo and your problem will be solved.
>
> --
>
> *Nuno Miguel Reis* | *Unified Communication** Systems*
> M. +351 913907481 | nreis at wavecom.pt
> WAVECOM-Soluções Rádio, S.A.
> Cacia Park | Rua do Progresso, Lote 15
> 3800-639 AVEIRO | Portugal
> T. +351 309 700 225 | F. +351 234 919 191
> *GPS
> <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>
> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>*
>
> <http://www.wavecom.pt/>
> <http://www.wavecom.pt/pt/>
>
>
>
>
> On Mon, Oct 22, 2018 at 9:30 PM Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> you have two option
>> 1) wait when issue in fixed on FreeSwitch repo;
>> 2) create own flite rpm file and then compile FreeSwitch RPM by self.
>>
>> Sergey
>>
>> пн, 22 окт. 2018 г. в 22:28, Social Boh <social at bohboh.info>:
>>
>>> Hello,
>>>
>>> I have read that page but I still do not understand what the solution is.
>>>
>>> Thank you
>>>
>>> Regards
>>>
>>> ---
>>> I'm SoCIaL, MayBe
>>>
>>> El 20/10/2018 a las 13:07, Sergey Safarov escribió:
>>>
>>> https://freeswitch.org/jira/browse/FS-8084
>>>
>>> сб, 20 окт. 2018 г. в 20:07, Social Boh <social at bohboh.info>:
>>>
>>>> Hello list,
>>>>
>>>> I compiled FreeSWITCH 1.8.2+git~20180926T175525Z~a98a958ac3~64bit en
>>>> CentOS 7.5.1804 without problem with flite-devel version 2.0.0-1.el7.
>>>>
>>>> Now, when I try to load the module:
>>>>
>>>> *[CRIT] switch_loadable_module.c:1522 Error Loading module
>>>> /usr/lib/freeswitch/mod/mod_flite.so*
>>>> ***/lib64/libflite.so.1: undefined symbol: snd_pcm_hw_params_any***
>>>>
>>>> Any Hint?
>>>>
>>>> Thank you
>>>>
>>>> Regards
>>>>
>>>> --
>>>> ---
>>>> I'm SoCIaL, MayBe
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com
>>>
>>> Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com
>>>
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com
>>>
>>>
>>> _________________________________________________________________________
>>> 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
>>
>> _________________________________________________________________________
>> 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
>
> _________________________________________________________________________
> 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:
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From social at bohboh.info  Wed Oct 24 21:52:29 2018
From: social at bohboh.info (Social Boh)
Date: Wed, 24 Oct 2018 16:52:29 -0500
Subject: [Freeswitch-users] CentOS 7, FreeSWITCH 1.8.2, mod_flite error
In-Reply-To: <CAE2+g++JzFiVe5jXV-hvsiyJArVj9jQw67dMtaRYVokNaVzZUg@mail.gmail.com>
References: <fcc23d99-f3bf-a4bc-49cd-aa63eb598b9d@bohboh.info>
 <CAHtxdDcbCjxAX0nf0C3W+06BGgP_QzNTrUWqY0wgoMWd_JRAyA@mail.gmail.com>
 <06fd3b74-be95-9d4f-9f7d-628117c7d475@bohboh.info>
 <CAHtxdDf=c3zUUh2ESsD3LZQxixTaYQd+uL4959Js+1JH2+ZN8g@mail.gmail.com>
 <CAE2+g++JzFiVe5jXV-hvsiyJArVj9jQw67dMtaRYVokNaVzZUg@mail.gmail.com>
Message-ID: <fb90f75e-e789-3923-ee18-105af6fc7b44@bohboh.info>

works...

Thank you

---
I'm SoCIaL, MayBe

El 24/10/2018 a las 11:37, Nuno Reis escribió:
> Hi there.
>
> The solution doesn't require you to have different freeswitch rpms.
> The problem is in flite build itself. The flite rpm package in 
> freeswitch repo is compiled with 'configure --enable-shared 
> --with-audio=alsa' and it should have been 'configure --enable-shared 
> --with-audio=none'.
> If you want just recompile flite from source with 'configure 
> --enable-shared --with-audio=none' and replace the version that came 
> from freeswitch repo and your problem will be solved.
>
> --
>
> *Nuno Miguel Reis*| *Unified Communication**Systems*
> M. +351 913907481 | nreis at wavecom.pt <mailto:nreis at wavecom.pt>
>
> WAVECOM-Soluções Rádio, S.A.
> Cacia Park | Rua do Progresso, Lote 15
> 3800-639 AVEIRO | Portugal
> T. +351 309 700 225 | F. +351 234 919 191
> *GPS 
> <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88> 
> | www.wavecom.pt <http://www.wavecom.pt/>**<http://www.wavecom.pt/>*
>
> <http://www.wavecom.pt/>
> <http://www.wavecom.pt/pt/>
>
>
>
>
> On Mon, Oct 22, 2018 at 9:30 PM Sergey Safarov <s.safarov at gmail.com 
> <mailto:s.safarov at gmail.com>> wrote:
>
>     you have two option
>     1) wait when issue in fixed on FreeSwitch repo;
>     2) create own flite rpm file and then compile FreeSwitch RPM by self.
>
>     Sergey
>
>     пн, 22 окт. 2018 г. в 22:28, Social Boh <social at bohboh.info
>     <mailto:social at bohboh.info>>:
>
>         Hello,
>
>         I have read that page but I still do not understand what the
>         solution is.
>
>         Thank you
>
>         Regards
>
>         ---
>         I'm SoCIaL, MayBe
>
>         El 20/10/2018 a las 13:07, Sergey Safarov escribió:
>>         https://freeswitch.org/jira/browse/FS-8084
>>
>>         сб, 20 окт. 2018 г. в 20:07, Social Boh <social at bohboh.info
>>         <mailto:social at bohboh.info>>:
>>
>>             Hello list,
>>
>>             I compiled FreeSWITCH
>>             1.8.2+git~20180926T175525Z~a98a958ac3~64bit en CentOS
>>             7.5.1804 without problem with flite-devel version
>>             2.0.0-1.el7.
>>
>>             Now, when I try to load the module:
>>
>>             /[CRIT] switch_loadable_module.c:1522 Error Loading
>>             module /usr/lib/freeswitch/mod/mod_flite.so//
>>             //**/lib64/libflite.so.1: undefined symbol:
>>             snd_pcm_hw_params_any**/
>>
>>             Any Hint?
>>
>>             Thank you
>>
>>             Regards
>>
>>             -- 
>>             ---
>>             I'm SoCIaL, MayBe
>>
>>             _________________________________________________________________________
>>             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
>>
>>
>>
>>         _________________________________________________________________________
>>         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
>
>         _________________________________________________________________________
>         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
>
>     _________________________________________________________________________
>     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
>
>
>
> _________________________________________________________________________
> 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 mbodbg at gmx.net  Thu Oct 25 08:12:53 2018
From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=)
Date: Thu, 25 Oct 2018 10:12:53 +0200
Subject: [Freeswitch-users] Sometimes "hanging" channels
In-Reply-To: <CAEJMVkCD1Rd-sTOwovGpq15RQjt280wJOrrK+yDK9cnmTukioA@mail.gmail.com>
References: <87D934A1-4A37-4709-93B5-96FA0659ABBE@gmx.net>
 <CAFiqYun3i3NDvqR4pkvfDCquQNMALXXAA-fp2EPm6bXx0VuVmg@mail.gmail.com>
 <1E3CBAFE-0B7B-445C-AAF2-EF4C074392B7@gmx.net>
 <C27A69BA-D5E4-4B30-8053-E53C34E585FA@vallimamod.org>
 <CAEJMVkCD1Rd-sTOwovGpq15RQjt280wJOrrK+yDK9cnmTukioA@mail.gmail.com>
Message-ID: <58AF98E7-5BEA-4687-A393-3091A4904C23@gmx.net>

As there is no support for 1.6 anymore, we prepare to upgrade to 1.8.2. However in the 1.6 freeswitch logs there was nothing special to see, I only saw in the ESL trace that the CHANNEL_HANGLUP_COMPLETE event was missing. If we reproduce this on 1.8.2 is there anything special we should configure to better analyze if there is a deadlock?

Best regards

Markus 



> Am 22.10.2018 um 21:01 schrieb Brian West <brian at freeswitch.com>:
> 
> It;s not that it was never received, it was probably deadlocked and never sent.
> 
> /b
> 
> 
> On Mon, Oct 15, 2018 at 7:46 AM Vallimamod Abdullah <vma at vallimamod.org <mailto:vma at vallimamod.org>> wrote:
> Hi,
> 
> The message "Locked, Waiting on external entities" is not alarming by itself as it is always logged on hangup. But it should be followed by:
> 
> [NOTICE] switch_core_session.c:1683 Session XXX (sofia/internal/YYY) Ended
> [NOTICE] switch_core_session.c:1687 Close Channel sofia/internal/YYY [CS_DESTROY]
> 
> If it is not the case then, you have an app or a module that is holding a lock on the session and preventing its release.
> 
> And this happens after the reporting state i.e. after the hangup_complete event and CDR posting. So there may be other causes for your missing events.
> 
> 
> 
> Best Regards,
> -- 
> Vallimamod Abdullah
> SIP Solutions
> vma at sip.solutions
> linkedin.com/in/vallimamod <http://linkedin.com/in/vallimamod>
> .
> 
> 
> > On 10 Oct 2018, at 15:54, Markus Bönke <mbodbg at gmx.net <mailto:mbodbg at gmx.net>> wrote:
> > 
> > So far I found out that the channel was kept open, because our ESL application never received a CHANNEL_HANGUP and CHANNEL_HANGUP_COMPLETE event. I’m running an ESL trace with tshark now. If it happens again I can see if those events are really not send by FS or they get lost in our app. If it turns out that FS sometimes is not sending the events we will upgrade to 1.8.2.
> > 
> > Thanks and regards
> > 
> > Markus
> > 
> > 
> >> Am 10.10.2018 um 14:03 schrieb Steven Ayre <steveayre at gmail.com <mailto:steveayre at gmail.com>>:
> >> 
> >> If you take a gcore it'll generate a core dump. You could then dig into it with gdb to find the thread for that channel and see what lock it's waiting on.
> >> However the gcore will pause the process for a while, so it will impact on any other calls on that box.
> >> 
> >> Have you tried reproducing it on 1.8?
> >> 
> >> On Tue, 9 Oct 2018 at 20:36, Markus Bönke <mbodbg at gmx.net <mailto:mbodbg at gmx.net>> wrote:
> >> Hello All,
> >> 
> >> we are running freeswitch 1.6.20, calls are controlled via ESL, CDRs are written with mod_xml_cdrl. Sometimes we see „hanging“ channels. In such a case the CDR via mod_xml_cdr is written and the last log entry for such a call is "Locked, Waiting on external entities“.
> >> ...
> >> freeswitch.log.2018-10-09-10-19-07.1:db2908e3-c06b-4293-88b1-e18cffb66263 2018-10-09 10:18:15.197345 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/anonymous at anonymous.invalid) State Change CS_REPORTING -> CS_DESTROY
> >> freeswitch.log.2018-10-09-10-19-07.1:db2908e3-c06b-4293-88b1-e18cffb66263 2018-10-09 10:18:15.197345 [DEBUG] switch_core_session.c:1665 Session 166111 (sofia/internal/anonymous at anonymous.invalid) Locked, Waiting on external entities
> >> ...
> >> How can I proceed to further analyze the problem ? In the last log line I also see the session number (Session 166111) - is there a way to find out on which external entity it is waiting?
> >> 
> >> Thanks and regards
> >> 
> >> Markus 
> >> _________________________________________________________________________
> >> 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/>
> >> _________________________________________________________________________
> >> 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/>
> > 
> > _________________________________________________________________________
> > 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/>
> 
> 
> _________________________________________________________________________
> 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/>
> 
> -- 
> 
> Brian West | Co-founder and Developer
> Need Commercial support? email sales at freeswitch.com <mailto: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 <mailto:brian at freeswitch.com>
> Mobile: 918-424-9378
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>  <https://www.facebook.com/freeswitch>  <https://twitter.com/freeswitch>
> _________________________________________________________________________
> 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 brian at freeswitch.com  Thu Oct 25 15:08:34 2018
From: brian at freeswitch.com (Brian West)
Date: Thu, 25 Oct 2018 10:08:34 -0500
Subject: [Freeswitch-users] Upcoming maintenance!
Message-ID: <CAEJMVkCeeTSBqa44OoW1AFFr+PCyQCPBSe1LDGMoQMpAkjzp-g@mail.gmail.com>

FreeSWITCHers,

Over the weekend you'll see various public facing services for the
FreeSWITCH go down for a while.  Please be patient while we make changes.

Thanks,


-- 

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 bilaln018 at gmail.com  Thu Oct 25 19:29:33 2018
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Fri, 26 Oct 2018 00:29:33 +0500
Subject: [Freeswitch-users] [username password with @][http cache]
In-Reply-To: <mailman.6314.1540377699.1213.freeswitch-users@lists.freeswitch.org>
References: <CADSVerCEY9GioU5_PCp2ioKS5DpZNDWcaXxgFMjAwuQfHW0vSQ@mail.gmail.com>
 <mailman.6314.1540377699.1213.freeswitch-users@lists.freeswitch.org>
Message-ID: <CADSVerDq72OKh6xWqZz6_MUc2z23aHkcp1EL5F74G50mhsw-Ow@mail.gmail.com>

Yeah that did helped, i was just wondering can i use ftp URL in mod http
cache?
FTP URL seems to be not working in my case, i was looking for any best
solution to download ftp sound files.

Regards
Abbasi

On Wed, 24 Oct 2018 at 4:08 PM, António Silva via FreeSWITCH-users <
freeswitch-users at lists.freeswitch.org> wrote:

>
>
>
> ---------- Forwarded message ----------
> From: "António Silva" <asilva at wirelessmundi.com>
> To: freeswitch-users at lists.freeswitch.org
> Cc:
> Bcc:
> Date: Wed, 24 Oct 2018 11:21:23 +0200
> Subject: Re: [Freeswitch-users] [username password with @][http cache]
>
> you could try url encode de @, to %40
>
> Example:
>
> username:password%40 at url
>
>
> On 24/10/2018 09:27, Bilal Abbasi wrote:
>
> Hey users,
> I have to download a sound file from FTP server having password with @ in
> it.
> The mod_http_cache has the way username:password at url, i was looking to
> escape the @ from it.
> This might be very basic question to ask, may be some one has done that
> already.
>
> Regards
> Abbasi
>
> _________________________________________________________________________
> Professional FreeSWITCH Servicessales at freeswitch.comhttps://freeswitch.com
>
> Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com
>
> --
> Saludos / Regards / Cumprimentos
> António Silva
>
>
>
>
> ---------- Forwarded message ----------
> From: "António Silva via FreeSWITCH-users" <
> freeswitch-users at lists.freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Cc:
> Bcc:
> Date: Wed, 24 Oct 2018 04:08:06 -0700 (PDT)
> Subject: Re: [Freeswitch-users] [username password with @][http cache]
> _________________________________________________________________________
> 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 chad at apartmentlines.com  Thu Oct 25 19:31:30 2018
From: chad at apartmentlines.com (Chad Phillips)
Date: Thu, 25 Oct 2018 12:31:30 -0700
Subject: [Freeswitch-users] Playing file into videoconference via mod_av
 freezes after server reboot
In-Reply-To: <CABMX_w9VD+5z=0hekQx8zoafyC6FwEhZDBzyxF6VJqyqfZqkPg@mail.gmail.com>
References: <CAJ0sghRWPnRHu5xxx5JftFymh7R+7QOMfKKPwxjC2KrmUFW0Cw@mail.gmail.com>
 <CABMX_w9VD+5z=0hekQx8zoafyC6FwEhZDBzyxF6VJqyqfZqkPg@mail.gmail.com>
Message-ID: <CAJ0sghTnKSBr_s=zKXm6cnNwc6ZYoUQkdWsEGYMVOjgVZzmZWQ@mail.gmail.com>

Happy to report I figure this out!

The default service file provided in the repository is out of date, in at
least two ways:

1) syslog.target is no longer available
2) The [Unit] section needs a Requires= to match the After= declaration to
ensure the network target is started as a dependency.

I've fixed this in my own installation, will file a Jira to get it
corrected upstream.

On Wed, Oct 24, 2018 at 1:59 PM Ítalo Rossi <italo at freeswitch.org> wrote:

> How do you restart freeswitch? Can you compare the system user and
> priorities it's running before and after the restart?
> Em ter, 23 de out de 2018 às 06:08, Chad Phillips <chad at apartmentlines.com>
> escreveu:
>
>> FreeSWITCH 1.8.2 on latest Debian Stretch, compiled stock from source,
>> deps installed via FS repo.
>>
>> I have FreeSWITCH managed by systemd to start at boot, which works well.
>> However, I'm hitting a weird issue where, after initial system boot, if I
>> try to play a video file into a conference via mod_av, the video is frozen
>> on the first frame (audio is fine). If I restart FreeSWITCH, then the video
>> file plays fine.
>>
>> Any thoughts on what might be causing this? My only guess: on boot
>> systemd starts FreeSWITCH before some other service or setup that is needed
>> to play video files via mod_av, and restarting FreeSWITCH fixes the issue
>> because said service is now running.
>> _________________________________________________________________________
>> 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
>
> _________________________________________________________________________
> 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 koki.roul at gmail.com  Fri Oct 26 01:31:04 2018
From: koki.roul at gmail.com (Lyubo Popov)
Date: Thu, 25 Oct 2018 22:31:04 -0300
Subject: [Freeswitch-users] Prevent FS from changing # to %23 in the SIP URI
Message-ID: <CAMSNS4gLjBLk5iJRjS8UsuWw3Uz2X=hJ_to5vYmwszYBBRjXgw@mail.gmail.com>

Hello all,

I am facing a problem where FS will convert a "#" character to "% 23" (no
spaces) in the SIP URI. The problem is that the call will be recorded like
that in CDRs or send in RADIUS packets ( I am using mod_xml_radius) and my
billing will have problems accounting the call as it will expect # and not
"% 23". Is there a way to tel the switch to not encode ( changes)  such SIP
URIs? Thank you for your help!

Re.
Koki
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From social at bohboh.info  Fri Oct 26 20:31:22 2018
From: social at bohboh.info (Social Boh)
Date: Fri, 26 Oct 2018 15:31:22 -0500
Subject: [Freeswitch-users] Firefox WebRTC ICE error
Message-ID: <1f6f89d6-5a60-fc06-e837-9cf2ae1499e0@bohboh.info>

Hello list,

I'm trying to use WebRTC on FreeSWITCH 1.8.2 with Firefox. While with 
FreeSWITCH 1.6.2 all works correctly with FreeSWITCH 1.8.2 I can't 
connect the call from browser to phone and viceverda. I think the most 
relevant lines are:

[DEBUG] switch_core_media.c:4309 sofia/internal/1010 at mydomain.org no 
suitable candidates found

[ERR] mod_sofia.c:2493 CODEC NEGOTIATION ERROR.  SDP:

This problem is on 62.0.3 and 63 firefox versions.

No problem with Chrome.

Thank you

Regards

-- 
---
I'm SoCIaL, MayBe



From sagarmalam at gmail.com  Sat Oct 27 09:56:01 2018
From: sagarmalam at gmail.com (sagar malam)
Date: Sat, 27 Oct 2018 15:26:01 +0530
Subject: [Freeswitch-users] Audio files created using Voicemail Module and
 uuid_record API are having different size.
Message-ID: <CAEZuF3vB3-BHP2p7fo=TBrR+ihygKNTQ41eqMzeAv+KQvN6N7w@mail.gmail.com>

Hello,

Audio files created using Voicemail Module and uuid_record API are having
different size.
I made two calls, one to drop VM and other to record call using uuid_record
API.
Size for VM file is 13.75291 MB/hour
Size of file created using uuid_record API is 6.866692 MB/hour
File format is mp3 for both files.
Both the calls were using G711a codec.
Can anyone help me understand why there is a difference ?

Thanks in advance

-- 
Thanks,

Sagar
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From s.safarov at gmail.com  Sat Oct 27 13:46:15 2018
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 27 Oct 2018 16:46:15 +0300
Subject: [Freeswitch-users] XML unescaping
Message-ID: <CAHtxdDcqr0L71PgLniowy6jFf3ejziGVAgZ0C8j6fW+MUb9vhA@mail.gmail.com>

Hello
When need to assign variable value with XML symbols requires escaping is
suggested to use dialplan contractions like example
<action application="set"><![CDATA[
sip_h_Call-Info=<urn:nena:uid:callid:72270801514820004831:
lng.001.experient.co.us>;purpose=nena-CallId]]></action>

But more elegant will be
<action application="set"
data="sip_h_Call-Info=&lt;urn:nena:uid:callid:72270801514820004831:
lng.001.experient.co.us&gt;;purpose=nena-CallId"/>

I want add this functionality to FS. Could you give me reference to FS code
where is used escaping/unescaping.

Sergey
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From sukithaj at gmail.com  Sun Oct 28 15:00:57 2018
From: sukithaj at gmail.com (sukitha jayasinghe)
Date: Sun, 28 Oct 2018 20:30:57 +0530
Subject: [Freeswitch-users] Identify voice quality issues using CDR records
Message-ID: <CAAwzf1=vN=N6ZusvvMJuKDuwUeCm68Lvat9r8hjQeB=c_Pi4XA@mail.gmail.com>

Dear all,

I have serious voice quality issues in my freeswitch 1.6.20 installation.
In CDRs I found that the voice quality percentage is 100 and mos is 4.5,
but the quality is very bad for one side. I have pasted the audio part of
the CDR below. What mean by the skip packet count. Is it possible to
identify the issue by looking at these statistics?

"audio" : {
            "inbound" : {
                "raw_bytes" : 90298,
                "media_bytes" : 90298,
                "packet_count" : 1235,
                "media_packet_count" : 1235,
                "skip_packet_count" : 1184,
                "jitter_packet_count" : 0,
                "dtmf_packet_count" : 0,
                "cng_packet_count" : 0,
                "flush_packet_count" : 0,
                "largest_jb_size" : 0,
                "jitter_min_variance" : 137.097046,
                "jitter_max_variance" : 315.2,
                "jitter_loss_rate" : 0,
                "jitter_burst_rate" : 0,
                "mean_interval" : 27.274788,
                "flaw_total" : 0,
                "quality_percentage" : 100,
                "mos" : 4.5
            },
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From paul.muaddib83 at gmail.com  Sun Oct 28 21:58:35 2018
From: paul.muaddib83 at gmail.com (Paul Muaddib)
Date: Sun, 28 Oct 2018 22:58:35 +0100
Subject: [Freeswitch-users] enterprise originate + originate_delay_start
Message-ID: <CAEjNe_GcpgTtLNSYgme22T1S2BRExZEGcWWXmjBR5Srkid9uPg@mail.gmail.com>

Hi,

I am trying to accomplish this bridge command but I am facing two problems.

<action application="bridge"
data="<originate_timeout=90,ignore_early_media=true>${group_call(werkstatt+A)},pickup/global,pickup/werkstatt:_:{originate_delay_start=25000}${group_call(buero+A)}“/>

1. When I set originate_delay_start to 25000 the group gets not called at
all. If I set it to 10000, it takes about 18 seconds and not 10 seconds.
2. Even though I set ignore_early_media=true I don't hear any ringing.

Best regards,
Paul

PS: Why is the value for originate_delay_start millisconds and for
leg_delay_start it is seconds?
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From paul.muaddib83 at gmail.com  Sun Oct 28 22:20:36 2018
From: paul.muaddib83 at gmail.com (Paul Muaddib)
Date: Sun, 28 Oct 2018 23:20:36 +0100
Subject: [Freeswitch-users] transfer_fallback_extension + transfer_on_fail
Message-ID: <CAEjNe_Fx_ZjZ5NYR64NBZhjpVH5S69=8T34NaJKubEBVVbXboA@mail.gmail.com>

Hi,

I setup the following:
When someone calls an extensions the phone rings for 30 seconds. After that
the phone keeps ringing but also the phones at front desk, here 10 and 11
When the user is busy (USER_BUSY) or when the extensions does not exist
(NO_PICKUP) the caller will be transferred to the front desk. This is
working with transfer_on_fail

Now I would like to accomplish the following:
When someone does a blind transfer and the callee does not answer after 30
seconds, how can I transfer the callee back to the extension that initiated
the transfer?
In the documentation I found the channel variable
transfer_fallback_extension. But there is not explanation to it and how
does this work together with transfer_on_fail?

Best regards,
Paul

<extension name="Extension">
      <condition field="destination_number" expression="^([1-9][0-9])$">
        <action application="set" data="dialed_extension=$1"/>

        <action application="set"
data="transfer_on_fail=NO_PICKUP,USER_BUSY,NO_ANSWER,SUBSCRIBER_ABSENT
BRIDGE_FAILED_EXTENSION xml"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>

        <action application="bind_digit_action"
data="get_digits,~^([1-9][0-9])$,exec:lua,get_digits.lua,peer,self"/>
        <action application="digit_action_set_realm" data="get_digits"/>

        <action application="bridge"
data="<ignore_early_media=true>[leg_timeout=30]user/${dialed_extension},
pickup/global, pickup/${dialed_extension}"/>

        <action application="bridge"
data="<originate_timeout=90,ignore_early_media=true>
                                                user/${dialed_extension},
                                                user/10,
                                                user/11,
                                                pickup/global,

pickup/${dialed_extension}"/>
        <action application="hangup"/>

      </condition>
</extension>

<extension name="BRIDGE_FAILED_EXTENSION">
        <condition>
                <action application="log" data="INFO ######## Extension
failed - ${originate_disposition} ########" />

                <condition field="${originate_failed_cause}"
expression="(NO_PICKUP)|(USER_BUSY)|(NO_ANSWER)" break="on-true">
                        <action application="transfer" data="operator"/>
                </condition>

                <condition field="${originate_failed_cause}"
expression=".*" break="on-true">
                        <action application="transfer" data="operator"/>
                </condition>

        </condition>
</extension>
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From paul.muaddib83 at gmail.com  Sun Oct 28 22:44:30 2018
From: paul.muaddib83 at gmail.com (Paul Muaddib)
Date: Sun, 28 Oct 2018 23:44:30 +0100
Subject: [Freeswitch-users] bridge and originate - difference between
	opening a channel and ring ready
Message-ID: <CAEjNe_EMvu=pL1S4pqonjR6jde3e+SQsMXxM+h8Sbv3CXrhTZw@mail.gmail.com>

Hi,

I am trying to understand the difference between opening a channel and ring
ready

Following example:
<action application="bridge"
data="<originate_timeout=90,ignore_early_media=true>user/A, user/B,
[leg_delay_start]user/C"/>

When I call multiple endpoints with bridge in parallel and delay one of
them with leg_delay_start, then I can see in the log files that freeswitch
opens up three channels and delays the ring ready for C. But when at the
same time user/C gets called from someone else while leg_delay_start is not
over yet, can user/C still answer the second call? And which call reaches
the user first if the user lets pass by leg_delay_start from the first call?

When I do the whole thing with enterprise originate, it looks much more
efficient since freeswitch is creating the channels only after the time for
originate_delay_start has passed by. What should be used when?

Best regards,
Paul
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From mouli123 at gmail.com  Mon Oct 29 06:53:17 2018
From: mouli123 at gmail.com (Chandramouli P)
Date: Mon, 29 Oct 2018 12:23:17 +0530
Subject: [Freeswitch-users] DTMF Issue - Unexpected digits are receiving
In-Reply-To: <CALR0W68_=W2NMiB+Fe9VsA54ixkwBTjFT=SycZ_uZP=NFWJ49A@mail.gmail.com>
References: <CALR0W682r3CHbf4S1ZqVTZ+f3-+LnySXXrz9Z_ukb7L5s_mkqg@mail.gmail.com>
 <CAFGRPVoB-6VXTVvO9oJthH-38ky0n5PQHZdrih1WTkyyoHML_Q@mail.gmail.com>
 <CALR0W69X_oLB8HT-874vQDZadR8sQwPYjUEuWc5AvZRjE996yA@mail.gmail.com>
 <7F36BBCD-D236-45EE-B0A0-F7DE031AC1E6@vallimamod.org>
 <CALR0W6-yvbqFncMx6aXG8QZmXHF_PCXP-cZJZeuk1hzfAVL0=A@mail.gmail.com>
 <51dc1d4b2d63413285607f7c6ed801e4@MDF-EXCH1.MDF-Holdings.local>
 <CALR0W68_=W2NMiB+Fe9VsA54ixkwBTjFT=SycZ_uZP=NFWJ49A@mail.gmail.com>
Message-ID: <CALR0W6-mLbJ2nvpfxvXwVX6+pf4wMtV-3T_T14y52FM2btkb7g@mail.gmail.com>

Hello all,

Any update would be appreciated. Thank you.

Best Regards,
Chandramouli.

On Wed, Oct 24, 2018 at 11:22 AM Chandramouli P <mouli123 at gmail.com> wrote:

> Hello Branden,
>
> Thank you for your reply. As you said, I do understand about A-D DTMF
> digits. But, my problem is to regenerate the issue i.e. ("A"). Which key
> (s), I have to press to send "A" as DTMF digit? I am sure that the end user
> is used regular mobile phone dialer only, and not softphone.
>
> Thank you.
>
> Best Regards,
> Chandramouli.
>
> On Wed, Oct 24, 2018 at 2:58 AM Branden Jordan <BJordan at e-teleco.com>
> wrote:
>
>> A is a valid DTMF digit so I wouldn’t worry about it too much and just
>> handle A-D in your valid TT fields. I see around a million calls a day and
>> see a few hundred A-D DTMF. It’s rare, but some really old phones still
>> have the A-D DTMF on them, and softphones have the ability to send them as
>> well, maybe your user is using a softphone on their end?
>>
>>
>>
>> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
>> Behalf Of *Chandramouli P
>> *Sent:* Monday, October 22, 2018 11:23 PM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* Re: [Freeswitch-users] DTMF Issue - Unexpected digits are
>> receiving
>>
>>
>>
>> Hello Abdullah,
>>
>>
>>
>> Thank you for your reply. By following your hint, I tried to regenerate
>> the issue. I tried by getting voice mail beep, other beeps like receiving
>> SMS, WhatsApp messages etc. But, unable to simulate.
>>
>>
>>
>> Thank you.
>>
>>
>>
>> Best regards,
>>
>> Chandramouli.
>>
>> On Mon, Oct 22, 2018 at 8:00 PM Vallimamod Abdullah <vma at vallimamod.org>
>> wrote:
>>
>> Hi,
>>
>>
>>
>> I encountered the same issue in the past and have noticed that it was the
>> mobile voicemail beep tone that was sometimes converted to 'A' dtmf on the
>> isdn gateway.
>>
>>
>>
>> Maybe your case is similar?
>>
>>
>>
>>
>>
>> Best Regards,
>> --
>> Vallimamod Abdullah
>> SIP Solutions
>> vma at sip.solutions
>> linkedin.com/in/vallimamod
>> .
>>
>>
>>
>> On 22 Oct 2018, at 08:24, Chandramouli P <mouli123 at gmail.com> wrote:
>>
>>
>>
>> Hello David,
>>
>>
>>
>> Thanks for your reply and I understand. But, As I explained, we tried to
>> regenerate the issue by pressing different keys on dial pad. But,
>> FreeSwitch is receiving the digits as usual every time. So, which key was
>> pressed by the user on dial pad to send "A"? Any assumptions?
>>
>>
>>
>> Thank you.
>>
>>
>>
>> Best regards,
>>
>> Chandramouli.
>>
>> On Sun, Oct 21, 2018 at 10:53 PM David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>> From what i know, "A" is a multifrequency tone, 697hz+1633hz. Maybe the
>> client is actually sending that.
>>
>>
>>
>> On Sat, Oct 20, 2018, 17:54 Chandramouli P <mouli123 at gmail.com> wrote:
>>
>> Hello,
>>
>>
>>
>> We developed an application (FreeSwitch) that receives pressed digit by
>> the end user on mobile phone dialer through DTMF and some action will be
>> performed. Everything is working fine till now. But, we noticed one issue
>> recently. From the FreeSwitch log, we came to know that FreeSwitch is
>> received the letter i.e. "A" from DTMF, instead of digit. We tried to
>> regenerate this issue. But, when we are trying to regenerate, FreeSwitch is
>> receiving the digit from DTMF as usual. I would like to know How FreeSwitch
>> is received the letter i.e. "A", instead of digit. Which key was pressed by
>> the end user on mobile phone dialer?
>>
>>
>>
>> Thank you.
>>
>>
>>
>> Best Regards,
>>
>> Chandramouli.
>>
>>
>>
>> _________________________________________________________________________
>> 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
>>
>> _________________________________________________________________________
>> 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 joes.mailing.lists at gmail.com  Mon Oct 29 11:36:33 2018
From: joes.mailing.lists at gmail.com (.)
Date: Mon, 29 Oct 2018 20:36:33 +0900
Subject: [Freeswitch-users] Incorrect example for 'originate' with
 socket and async full
In-Reply-To: <CAEJMVkALhsdbXWscfAfp2GFBg-f1xRV1n0baY00jxP+P==zB_A@mail.gmail.com>
References: <CAHUGhbKq2FEPJ23mfjvfDBSkM+16yz5F6OCFanN_5tsKRoHj-g@mail.gmail.com>
 <7337B160-A5A0-4B70-8F69-926E133242AE@mgtech.com>
 <CAEJMVkDchaK2AutTfzxVKZ=b-XG8SX_711ODP7ZU=0fKKmij+g@mail.gmail.com>
 <1539008509.884661.1534620552.4EF2A2AC@webmail.messagingengine.com>
 <CAEJMVkALhsdbXWscfAfp2GFBg-f1xRV1n0baY00jxP+P==zB_A@mail.gmail.com>
Message-ID: <CAHUGhb+nNd8L5viEVbpJgvz0ErRjabS4mDtCt-cFm1t4ohwY=A@mail.gmail.com>

Thanks for the responses.  Is it possible to use this syntax then to call
the socket application like in my initial example?  It is not clear to me
how that works with the app:arg pair syntax, given that they socket app
takes multiple args, and in the socket case the colon is also used in the
IP:port parameter value.  Best, Joe


On Wed, Oct 10, 2018 at 5:07 AM Brian West <brian at freeswitch.com> wrote:

> Because the &app method has been deprecated for a while now and the way
> you should accomplish this is via inline dialplan
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+Inline+Dialplan
>
> Its actualy more powerful.
>
> On Mon, Oct 8, 2018 at 11:05 AM Michael Avers <michael at mailworks.org>
> wrote:
>
>> Hi Brian can you please clarify what you mean by online dialplan? Why
>> shouldn't &app be used anymore?
>>
>> Thank you
>> Mike
>>
>>
>> On Mon, Oct 8, 2018, at 3:13 AM, Brian West wrote:
>>
>> The &app notation shouldn’t be used anymore the inline dialplan is the
>> appropriate way to do this moving forward.
>>
>> /b
>>
>> On Fri, Oct 5, 2018 at 6:41 PM Mario <mario_fs at mgtech.com> wrote:
>>
>> Fixed.
>> Mario G
>>
>>
>> On Oct 4, 2018, at 5:30 AM, . <joes.mailing.lists at gmail.com> wrote:
>>
>> Hi,
>>
>> I wanted to report an error/typo in the confluence wiki related to
>> 'originate', and its use together with the socket application.
>>
>> Apologies, I don't have access to the wiki, but thought this was worth
>> passing on, so here it is:
>>
>>    -
>>    https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-originate
>>
>> The wiki example for ring_ready also includes the socket application and
>> is written as:
>>
>>    -
>>
>>    originate {return_ring_ready=true}sofia/gateway/someprovider/919246461929 &socket(127.0.0.1:8082 async full)
>>
>>
>> however this hangs forever if used as written. Eventually, after several
>> hours of hair tearing and googling, I stumbled on this thread:
>>
>>    -
>>    https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-3591/FS-3591.html
>>
>> Indeed this is the problem.  The correct syntax for the socket component
>> is:
>>
>>    -
>>
>>    &socket('127.0.0.1:8082 async full')
>>
>>    - Specifically single quotes around the command so that it is treated
>>       as a single string.
>>
>> After making this change there is no more hanging on the outbound calls
>> and all events come through in a timely fashion.
>>
>> Cheers,
>> Joe
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> 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
>>
>>
>> _________________________________________________________________________
>> 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>
>>
>> *_________________________________________________________________________*
>> 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
>>
>>
>> _________________________________________________________________________
>> 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>
> _________________________________________________________________________
> 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 agoulis at opensips.org  Sat Oct 27 19:39:46 2018
From: agoulis at opensips.org (Alex Goulis)
Date: Sat, 27 Oct 2018 14:39:46 -0500
Subject: [Freeswitch-users] mod_spy,
 does it only work for registered users?
In-Reply-To: <D2A08A40-D9D9-489E-BA5E-63DBAF123CF8@jerris.com>
Message-ID: <E1gGUR5-0002lB-Ia@mail.opensips.org>

Ok... sorry for the delay in getting back to this... So I got it working by actually setting the variable_domain_name  and variable_user_name on both a and b legs. Those were not set automatically it seems, but once I set them it worked!!Thank you!!!!Sent via the Samsung Galaxy Note8, an AT&T 4G LTE smartphone
-------- Original message --------From: Michael Jerris <mike at jerris.com> Date: 10/14/18  4:36 PM  (GMT-06:00) To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> Subject: Re: [Freeswitch-users] mod_spy, does it only work for registered users? Can you show me exactly how you are overriding them in dialplan?  This should work.On Oct 12, 2018, at 6:40 PM, Alex Goulis <agoulis at opensips.org> wrote:Man thanks for digging deeper... unfortunately I have an identical match to Pairs 0-3 in my tests.   Ie, they all have the same user/domain in them. And they all match the user/domain I set when invoking userspy.The events after the invite show them populated as well. I'm pretty sure the eavesdrop trigger happens before I can manually override them with dialplan on the incoming call to be targeted.Happy to test any scenario you can further think of though.AlexSent via the Samsung Galaxy Note8, an AT&T 4G LTE smartphone-------- Original message --------From: Michael Jerris <mike at jerris.com> Date: 10/12/18  3:53 PM  (GMT-07:00) To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> Subject: Re: [Freeswitch-users] mod_spy, does it only work for registered users? After talking to you about this, I just took a look. No its not based on registered users, but authed users will probably more reliably set the right variables to figure out the right user to match to.  Trick is to have the vars set to be able to look them up.    username[0] = switch_event_get_header(event, "Caller-Username");                                                                                                domain[0] = switch_event_get_header(event, "variable_domain_name");                                                                                                                                                                                                                                                             username[1] = switch_event_get_header(event, "variable_dialed_user");                                                                                           domain[1] = switch_event_get_header(event, "variable_dialed_domain");                                                                                                                                                                                                                                                           username[2] = switch_event_get_header(event, "variable_user_name");                                                                                             domain[2] = switch_event_get_header(event, "variable_domain_name");                                                                                                                                                                                                                                                             username[3] = switch_event_get_header(event, "variable_sip_to_user");                                                                                           domain[3] = switch_event_get_header(event, "variable_domain_name");                                                                                                                                                                                                                                                             username[4] = switch_event_get_header(event, "variable_verto_user");                                                                                            domain[4] = switch_event_get_header(event, "variable_verto_host");                                                                                          It has to match one of those pairs.On Oct 5, 2018, at 6:24 PM, Alex Goulis <agoulis at opensips.org> wrote:Hi all...Does the userspy application only target registered users or can you use it to spy calls to let's say calls to a DID?
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From agoulis at opensips.org  Sat Oct 27 19:54:56 2018
From: agoulis at opensips.org (Alex Goulis)
Date: Sat, 27 Oct 2018 14:54:56 -0500
Subject: [Freeswitch-users] Mod_spy part2
Message-ID: <E1gGUfm-0002nw-FB@mail.opensips.org>

Hi all...Now that I've managed to get mod_spy to work, I've noticed it only allows a perpetual listen and gives none of the dynamic dtmf or the ability to whisper on the eavesdrop app when it initiates it. I've set all the correct eavesdrop parameters before calling userspy, but they dont have any effect.Is there a way to get the eavesdrop to whisper or enable the dtmf functions to work using userspy?Thanks!
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From bipin at xbipin.com  Mon Oct 29 14:20:36 2018
From: bipin at xbipin.com (Bipin Patel)
Date: Mon, 29 Oct 2018 18:20:36 +0400
Subject: [Freeswitch-users] filter * DTMF when using bind_meta_app and
 gateway that supports inband DTMF
Message-ID: <6dab49e4-f418-c93c-1345-641a12fcb26b@xbipin.com>

hi,

i have a SIP gateway which doesn't support info or rfc DTMF so im forced 
with using inband so in my dialplan i use start_dtmf_generate to 
translate from info/RFC used by IP phones to inband so gateway can send 
that ahead, the problem i face is im using bind_meta_app to start a on 
demand recording using *2 so when the local extension presses *2, the * 
DTMF still goes to the gateway inband and 2 is never heard by callee. 
What i want is to filter out the complete *2 so the callee doesn't hear 
those at all but i dont want to drop all dtmf from going to gateway.

Is there a way to filter out selective DTMF like the * or any other star 
codes used in bind_meta_app etc?


-- 
Regards,
Bipin


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From social at bohboh.info  Mon Oct 29 14:38:08 2018
From: social at bohboh.info (Social Boh)
Date: Mon, 29 Oct 2018 09:38:08 -0500
Subject: [Freeswitch-users] Firefox WebRTC ICE error
In-Reply-To: <1f6f89d6-5a60-fc06-e837-9cf2ae1499e0@bohboh.info>
References: <1f6f89d6-5a60-fc06-e837-9cf2ae1499e0@bohboh.info>
Message-ID: <94ca8782-1c63-f4ab-c5c7-d63872f1ae1c@bohboh.info>

The exact error is:


/[DEBUG] switch_core_media.c:4256 Searching for rtp candidate.
[DEBUG] switch_core_media.c:4256 Searching for rtcp candidate.
[DEBUG] switch_core_media.c:4303 Look for Relay Candidates as last resort
[DEBUG] switch_core_media.c:4256 Searching for rtp candidate.
[DEBUG] switch_core_media.c:4256 Searching for rtcp candidate.
[DEBUG] switch_core_media.c:4309 sofia/internal/1019 at mydomain.org no 
suitable candidates found.

/Thank you/
/

---
I'm SoCIaL, MayBe

El 26/10/2018 a las 15:31, Social Boh escribió:
> Hello list,
>
> I'm trying to use WebRTC on FreeSWITCH 1.8.2 with Firefox. While with 
> FreeSWITCH 1.6.2 all works correctly with FreeSWITCH 1.8.2 I can't 
> connect the call from browser to phone and viceverda. I think the most 
> relevant lines are:
>
> [DEBUG] switch_core_media.c:4309 sofia/internal/1010 at mydomain.org no 
> suitable candidates found
>
> [ERR] mod_sofia.c:2493 CODEC NEGOTIATION ERROR.  SDP:
>
> This problem is on 62.0.3 and 63 firefox versions.
>
> No problem with Chrome.
>
> Thank you
>
> Regards
>

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From grcamauer at gmail.com  Mon Oct 29 14:43:15 2018
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Mon, 29 Oct 2018 11:43:15 -0300
Subject: [Freeswitch-users] Audio files created using Voicemail Module
 and uuid_record API are having different size.
In-Reply-To: <CAEZuF3vB3-BHP2p7fo=TBrR+ihygKNTQ41eqMzeAv+KQvN6N7w@mail.gmail.com>
References: <CAEZuF3vB3-BHP2p7fo=TBrR+ihygKNTQ41eqMzeAv+KQvN6N7w@mail.gmail.com>
Message-ID: <CAPNTnnoCuKsbJSOnM5hg1VxVmY_Bihqbj0n4AuUEVFUZr_sT0A@mail.gmail.com>

Just a wild guess from looking at the file size, it looks like one is twice
the size of the other.  Could one be stereo and the othe mono?  Or one be
recording 2 legs and the other just one?

Guillermo

On Mon, Oct 29, 2018 at 10:49 AM sagar malam <sagarmalam at gmail.com> wrote:

> Hello,
>
> Audio files created using Voicemail Module and uuid_record API are having
> different size.
> I made two calls, one to drop VM and other to record call using
> uuid_record API.
> Size for VM file is 13.75291 MB/hour
> Size of file created using uuid_record API is 6.866692 MB/hour
> File format is mp3 for both files.
> Both the calls were using G711a codec.
> Can anyone help me understand why there is a difference ?
>
> Thanks in advance
>
> --
> Thanks,
>
> Sagar
> _________________________________________________________________________
> 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



-- 
Guillermo Ruiz Camauer
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From steveayre at gmail.com  Tue Oct 30 12:16:51 2018
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 30 Oct 2018 12:16:51 +0000
Subject: [Freeswitch-users] bridge and originate - difference between
 opening a channel and ring ready
In-Reply-To: <CAEjNe_EMvu=pL1S4pqonjR6jde3e+SQsMXxM+h8Sbv3CXrhTZw@mail.gmail.com>
References: <CAEjNe_EMvu=pL1S4pqonjR6jde3e+SQsMXxM+h8Sbv3CXrhTZw@mail.gmail.com>
Message-ID: <CAFiqYu=_EudhumCe=rbyvUJ06cNTJHeO6eY5t3eXJ4rwseuYbg@mail.gmail.com>

> But when at the same time user/C gets called from someone else while
leg_delay_start is not over yet, can user/C still answer the second call?

Maybe. They can get sent calls concurrently. How they handle that would
depend on their client.

On Mon, 29 Oct 2018 at 14:37, Paul Muaddib <paul.muaddib83 at gmail.com> wrote:

> Hi,
>
> I am trying to understand the difference between opening a channel and
> ring ready
>
> Following example:
> <action application="bridge"
> data="<originate_timeout=90,ignore_early_media=true>user/A, user/B,
> [leg_delay_start]user/C"/>
>
> When I call multiple endpoints with bridge in parallel and delay one of
> them with leg_delay_start, then I can see in the log files that freeswitch
> opens up three channels and delays the ring ready for C. But when at the
> same time user/C gets called from someone else while leg_delay_start is not
> over yet, can user/C still answer the second call? And which call reaches
> the user first if the user lets pass by leg_delay_start from the first call?
>
> When I do the whole thing with enterprise originate, it looks much more
> efficient since freeswitch is creating the channels only after the time for
> originate_delay_start has passed by. What should be used when?
>
> Best regards,
> Paul
>
> _________________________________________________________________________
> 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 sagarmalam at gmail.com  Tue Oct 30 12:32:17 2018
From: sagarmalam at gmail.com (sagar malam)
Date: Tue, 30 Oct 2018 18:02:17 +0530
Subject: [Freeswitch-users] Audio files created using Voicemail Module
 and uuid_record API are having different size.
In-Reply-To: <CAPNTnnoCuKsbJSOnM5hg1VxVmY_Bihqbj0n4AuUEVFUZr_sT0A@mail.gmail.com>
References: <CAEZuF3vB3-BHP2p7fo=TBrR+ihygKNTQ41eqMzeAv+KQvN6N7w@mail.gmail.com>
 <CAPNTnnoCuKsbJSOnM5hg1VxVmY_Bihqbj0n4AuUEVFUZr_sT0A@mail.gmail.com>
Message-ID: <CAEZuF3sPUVvohH80oaENPe0x+5fJLpLMMRtqUfniKzd7jsepxw@mail.gmail.com>

Guillermo,
Both files are mono.

About the legs, VM is recording one leg and uuid_record is recording 2
legs.But does that matter ?


On Mon, Oct 29, 2018 at 8:37 PM Guillermo Ruiz Camauer <grcamauer at gmail.com>
wrote:

> Just a wild guess from looking at the file size, it looks like one is
> twice the size of the other.  Could one be stereo and the othe mono?  Or
> one be recording 2 legs and the other just one?
>
> Guillermo
>
> On Mon, Oct 29, 2018 at 10:49 AM sagar malam <sagarmalam at gmail.com> wrote:
>
>> Hello,
>>
>> Audio files created using Voicemail Module and uuid_record API are having
>> different size.
>> I made two calls, one to drop VM and other to record call using
>> uuid_record API.
>> Size for VM file is 13.75291 MB/hour
>> Size of file created using uuid_record API is 6.866692 MB/hour
>> File format is mp3 for both files.
>> Both the calls were using G711a codec.
>> Can anyone help me understand why there is a difference ?
>>
>> Thanks in advance
>>
>> --
>> Thanks,
>>
>> Sagar
>> _________________________________________________________________________
>> 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
>
>
>
> --
> Guillermo Ruiz Camauer
> _________________________________________________________________________
> 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



-- 
Thanks,

Sagar
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From sagarmalam at gmail.com  Tue Oct 30 12:34:18 2018
From: sagarmalam at gmail.com (sagar malam)
Date: Tue, 30 Oct 2018 18:04:18 +0530
Subject: [Freeswitch-users] Shared presence not working with callcenter
 application of Freeswitch
In-Reply-To: <CAEZuF3tZz101nOiVLugcszQUa+TxrixnL79375jdW=VOPWf8Rg@mail.gmail.com>
References: <CAEZuF3vgN4OdaRha=edmCkhd_Ac-ZLThs6_uXx2LT3FGdVgfdQ@mail.gmail.com>
 <LNXP265MB1180E1DFD30C173B4DFC6759EBE70@LNXP265MB1180.GBRP265.PROD.OUTLOOK.COM>
 <CAEZuF3tbp1tPH9Bt7jDqiVeUAzBToBsD3Ah9mNS2RF+o7B+C9A@mail.gmail.com>
 <6ABFDA54-2016-495F-BB0D-DCFB55874429@jerris.com>
 <CAEZuF3tZz101nOiVLugcszQUa+TxrixnL79375jdW=VOPWf8Rg@mail.gmail.com>
Message-ID: <CAEZuF3vsdivA5KbMs-JspTYDrOgEUTo5AZiac4LMc8-yU+5c4Q@mail.gmail.com>

Any update on this ?
Has anyone got this working? I have tested this with FS 1.6.20
without success.

On Thu, Oct 18, 2018 at 3:54 PM sagar malam <sagarmalam at gmail.com> wrote:

> Yes.
> Below parameters are enabled :
> manage-presence
> manage-shared-appearance
> dbname = share_presence
>
> On Sat, Oct 13, 2018 at 4:07 AM Michael Jerris <mike at jerris.com> wrote:
>
>> Is manage presense enabled?
>>
>> On Oct 10, 2018, at 5:29 AM, sagar malam <sagarmalam at gmail.com> wrote:
>>
>> I have already tried that setting along with sip_invite_domain.But
>> without any success.Further when i enable SLA and Presence debug , i dont
>> see any SLA / presence logs which appears in case of one to one calls.
>>
>> On Wed, Oct 10, 2018 at 4:54 AM Shaun Stokes <
>> shaun.stokes at itec-support.co.uk> wrote:
>>
>>> You need to set the presence_id variable of your agent when you bridge
>>> to the agent in the agent contact field.
>>>
>>> Shaun
>>>
>>> Get Outlook for iOS <https://aka.ms/o0ukef>
>>> ------------------------------
>>> *From:* 20110170300n behalf of
>>> *Sent:* Tuesday, October 9, 2018 19:48
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* [Freeswitch-users] Shared presence not working with
>>> callcenter application of Freeswitch
>>>
>>> I am facing issue related to presence/ shared presence for agents in
>>> callcenter(mod_callcenter).
>>> Suppose we are dialling a user 1001 at example.com and it is a shared user
>>> in two phones. Shared presence works perfectly if we directly dial
>>> extension 1001.
>>>
>>> But if same extension is configured as agent of callcenter and call is
>>> bridged to agent(extension) through callcenter, shared presence does not
>>> work. Freeswitch does not generate notify for agent state change.
>>>
>>> Is this expected behaviour ? If not then what can fix this ?
>>>
>>> Thanks in advance
>>>
>>
>> _________________________________________________________________________
>> 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
>
>

-- 
Thanks,

Sagar
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From sagarmalam at gmail.com  Tue Oct 30 12:37:05 2018
From: sagarmalam at gmail.com (sagar malam)
Date: Tue, 30 Oct 2018 18:07:05 +0530
Subject: [Freeswitch-users] FREESWITCH NOT PARSING SDP OF HOLD REINVITE
	PROPERLY
In-Reply-To: <CAEZuF3tD1W-PK2=Ak8UX6RuKL3UWwDbms_xSVdpPi7h0+t9Www@mail.gmail.com>
References: <CAEZuF3tD1W-PK2=Ak8UX6RuKL3UWwDbms_xSVdpPi7h0+t9Www@mail.gmail.com>
Message-ID: <CAEZuF3s3ZZJb8e7WFpVxhsxo3R+sHZFwyNg9biSOYJ0kK_6z1A@mail.gmail.com>

Can anyone help me on this ?
Please advice if this is not right place to ask for fix and need to create
a Jira.

Thanks.

On Sun, Sep 23, 2018 at 6:36 PM sagar malam <sagarmalam at gmail.com> wrote:

> Hello ,
>
> I am working on developing PBX application using FS 1.6.17.I am facing
> issue which i think is a bug in FS.
> 1) User A calls User B
> 2) User B blind transfers to  User C
> 3) Now User A and User C is connected.
> 4) User A puts call on hold.At this step ,When FS executes code(in
> switch_core_media.c file )  to check if HOLD RE-INVITE has any crypto
> attributes , it gets positive even though RE-INIVITE does not have any
> crypto attributes.Below are the logs for fs_cli ( look for logs in BOLD )
>
> =======================LOGS TAKEN FROM FS_CLI================
>
> 2018-09-23 12:41:24.408118 [DEBUG] sofia.c:7052 Channel sofia/register/
> 1025 at 202.131.119.122:53814 entering state [received][100]
> 2018-09-23 12:41:24.408118 [DEBUG] sofia.c:7062 Remote SDP:
> v=0
> o=- 1537706456 1537706459 IN IP4 10.50.7.253
> s=Polycom IP Phone
> c=IN IP4 10.50.7.253
> t=0 0
> a=sendonly
> m=audio 14852 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=sendonly
> m=video 0 RTP/AVP 103 31 34
>
> 2018-09-23 12:41:24.658118 [DEBUG] sofia.c:7052 Channel sofia/register/
> 1025 at 202.131.119.122:53814 entering state [completed][200]
> 2018-09-23 12:41:24.758055 [DEBUG] sofia.c:7052 Channel sofia/register/
> 1025 at 202.131.119.122:53814 entering state [ready][200]
> 2018-09-23 12:41:24.898112 [DEBUG] sofia.c:7052 Channel sofia/register/
> 1013 at ecosmob.sip.teledge.com entering state [calling][0]
> 2018-09-23 12:41:25.218098 [DEBUG] sofia.c:7052 Channel sofia/register/
> 1013 at ecosmob.sip.teledge.com entering state [completing][200]
> 2018-09-23 12:41:25.218098 [DEBUG] sofia.c:7062 Remote SDP:
> v=0
> o=- 1537706452 1537706455 IN IP4 10.50.7.253
> s=Polycom IP Phone
> c=IN IP4 10.50.7.253
> t=0 0
> a=sendrecv
> m=audio 34858 RTP/AVP 0 8 18 9 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=direction:active
> m=video 0 RTP/AVP 103 31 34
> a=direction:active
> a=oldmediaip:172.16.16.241
>
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[G729:18:8000:20:8000:1]
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4502 Audio Codec
> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1]
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[G729:18:8000:20:8000:1]
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4502 Audio Codec
> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1]
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [G729:18:8000:20:8000:1]/[G729:18:8000:20:8000:1]
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4502 Audio Codec
> Compare [G729:18:8000:20:8000:1] ++++ is saved as a match
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1]
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1]
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [G729:18:8000:20:8000:1]/[G722:9:8000:20:64000:1]
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [G722:9:8000:20:64000:1]/[G729:18:8000:20:8000:1]
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1]
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4502 Audio Codec
> Compare [G722:9:8000:20:64000:1] ++++ is saved as a match
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4363 Set
> telephone-event payload to 101 at 8000
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4706 Set
> telephone-event payload to 101 at 8000
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4765 sofia/register/
> 1013 at ecosmob.sip.teledge.com Set 2833 dtmf send payload to 101 recv
> payload to 101
> 2018-09-23 12:41:25.218098 [DEBUG] sofia.c:7344 RESTABLISH MEDIA SDP:
> v=0
> o=FreeSWITCH 1537693452 1537693458 IN IP4 10.50.7.251
> s=FreeSWITCH
> c=IN IP4 10.50.7.251
> t=0 0
> m=audio 13002 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
> m=video 17310 RTP/AVP 97 31 34
> b=AS:1024
> a=rtpmap:97 H264/90000
> a=rtpmap:31 H261/90000
> a=rtpmap:34 H263/90000
>
> 2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:6860 Audio params
> are unchanged for sofia/register/1013 at ecosmob.sip.teledge.com.
> 2018-09-23 12:41:25.218098 [DEBUG] sofia.c:7052 Channel sofia/register/
> 1025 at 202.131.119.122:53814 entering state [calling][0]
> 2018-09-23 12:41:25.238128 [DEBUG] sofia.c:7052 Channel sofia/register/
> 1013 at ecosmob.sip.teledge.com entering state [ready][200]
> 2018-09-23 12:41:25.498123 [DEBUG] sofia.c:7052 Channel sofia/register/
> 1025 at 202.131.119.122:53814 entering state [completing][200]
> 2018-09-23 12:41:25.498123 [DEBUG] sofia.c:7062 Remote SDP:
> *v=0*
> *o=- 1537706456 1537706460 IN IP4 10.50.7.253*
> *s=Polycom IP Phone*
> *c=IN IP4 10.50.7.253*
> *t=0 0*
> *a=sendonly*
> *m=audio 14852 RTP/AVP 18 0 8 9 101*
> *a=rtpmap:18 G729/8000*
> *a=fmtp:18 annexb=no*
> *a=rtpmap:0 PCMU/8000*
> *a=rtpmap:8 PCMA/8000*
> *a=rtpmap:9 G722/8000*
> *a=rtpmap:101 telephone-event/8000*
> *a=direction:active*
> *m=video 0 RTP/AVP 103 31 34*
> *a=direction:active*
> *a=oldmediaip:172.16.16.249*
>
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [G729:18:8000:20:8000:1]/[G729:18:8000:20:8000:1]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4502 Audio Codec
> Compare [G729:18:8000:20:8000:1] ++++ is saved as a match*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [G729:18:8000:20:8000:1]/[G722:9:8000:20:64000:1]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[G729:18:8000:20:8000:1]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4502 Audio Codec
> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[G729:18:8000:20:8000:1]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4502 Audio Codec
> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [G722:9:8000:20:64000:1]/[G729:18:8000:20:8000:1]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio Codec
> Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4502 Audio Codec
> Compare [G722:9:8000:20:64000:1] ++++ is saved as a match*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4363 Set
> telephone-event payload to 101 at 8000*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4706 Set
> telephone-event payload to 101 at 8000*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4765
> sofia/register/1025 at 202.131.119.122:53814
> <http://1025 at 202.131.119.122:53814> Set 2833 dtmf send payload to 101 recv
> payload to 101*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set Local
> audio crypto Key [1 AEAD_AES_256_GCM_8
> inline:R/pN8McEpZCkZFA8lsTNl19uoua1R8bSPnthVWaqMpse/3n/Nqk8Mb7FXcQ]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set Local
> video crypto Key [1 AEAD_AES_256_GCM_8
> inline:JVeqouE9G4frRG7FF5pLNBJfufVOKkrObsLVUTLImMpzRo2fKYVa2ZNU4ko]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set Local
> audio crypto Key [2 AEAD_AES_128_GCM_8
> inline:IJcN/B4PZbaX/WiLGLRMcGukkvHnQp/7SXLOdg]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set Local
> video crypto Key [2 AEAD_AES_128_GCM_8
> inline:dAlbL/sVbX2VkpVzWpWFt8/9wixwugT9pnwHeA]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set Local
> audio crypto Key [3 AES_CM_256_HMAC_SHA1_80
> inline:bCEgc1rOR76vYxwpTnXoblfXgSQIDLJ0jQrMEghJh9akfkn6jcSoyzgsF4JHtw]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set Local
> video crypto Key [3 AES_CM_256_HMAC_SHA1_80
> inline:Ae2/2r0cISukm84lGWzNL0O+r54lJgeLsjxKzxrBWAdIOlVt6fuPqXbMj+fgmA]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set Local
> audio crypto Key [4 AES_CM_192_HMAC_SHA1_80
> inline:Ux4vMz/xZsXLDsc1YD7cPDUSsaC/Xs8KJv+QBm5nj8v9O70NR8E]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set Local
> video crypto Key [4 AES_CM_192_HMAC_SHA1_80
> inline:XLdnwildLOkebarqHXItf3RnJoXM/2+6tqwL925rjW4WJtIkF74]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set Local
> audio crypto Key [5 AES_CM_128_HMAC_SHA1_80
> inline:soWN8HkHRNDRz2itaR8JcJ3C6PRzWYjoyLZGl2d5]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set Local
> video crypto Key [5 AES_CM_128_HMAC_SHA1_80
> inline:f3fCmj6oXqrCGj/Wh1H38lY9liE6Awiev4Mo4syD]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set Local
> audio crypto Key [6 AES_CM_256_HMAC_SHA1_32
> inline:ikMbUpS+1V5t4baA+GNGP+T3iBnN83VzOJqdI0gL3LLThnhCg/M3IxEO0Crxww]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set Local
> video crypto Key [6 AES_CM_256_HMAC_SHA1_32
> inline:d5RXBHswLr0PC+9In+CxOQVtrQ9puAdOXsM/dfyTLuTVzp1nxVDotupDNgClgw]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set Local
> audio crypto Key [7 AES_CM_192_HMAC_SHA1_32
> inline:jZ9p+h023hhYcRw6KesYK/cLOn8FYkHfWksh8vEGSktPCG6wiz0]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set Local
> video crypto Key [7 AES_CM_192_HMAC_SHA1_32
> inline:oI21eom6/FW+T8fXWYdYZ76w8RYJGJ1a3n/VKFWip/brFXEF/w8]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set Local
> audio crypto Key [8 AES_CM_128_HMAC_SHA1_32
> inline:vWCsqgSoEf8lNmgmHqTz8GUYSvm21JBbNou/fext]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set Local
> video crypto Key [8 AES_CM_128_HMAC_SHA1_32
> inline:XUEC/U88QVT3YhCBunaxMftY8jY2GW4GOq2ivrhN]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set Local
> audio crypto Key [9 AES_CM_128_NULL_AUTH
> inline:En7TIXDB7g3YBNp8SSaSyn8W/b3BjDZapNo19sbY]*
> *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set Local
> video crypto Key [9 AES_CM_128_NULL_AUTH
> inline:0IunsVM9Agglm6leczPQI11+bpkaZuM4x21lVK1D]*
> 2018-09-23 12:41:25.498123 [DEBUG] sofia.c:7344 RESTABLISH MEDIA SDP:
> v=0
> o=FreeSWITCH 1537686256 1537686263 IN IP4 10.50.7.251
> s=FreeSWITCH
> c=IN IP4 10.50.7.251
> t=0 0
> m=audio 20198 RTP/AVP 18 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=recvonly
> m=video 29982 RTP/SAVP 97 31 34
> b=AS:1024
> a=rtpmap:97 H264/90000
> a=rtpmap:31 H261/90000
> a=rtpmap:34 H263/90000
> a=crypto:1 AEAD_AES_256_GCM_8
> inline:JVeqouE9G4frRG7FF5pLNBJfufVOKkrObsLVUTLImMpzRo2fKYVa2ZNU4ko
> a=crypto:2 AEAD_AES_128_GCM_8 inline:dAlbL/sVbX2VkpVzWpWFt8/9wixwugT9pnwHeA
> a=crypto:3 AES_CM_256_HMAC_SHA1_80
> inline:Ae2/2r0cISukm84lGWzNL0O+r54lJgeLsjxKzxrBWAdIOlVt6fuPqXbMj+fgmA
> a=crypto:4 AES_CM_192_HMAC_SHA1_80
> inline:XLdnwildLOkebarqHXItf3RnJoXM/2+6tqwL925rjW4WJtIkF74
> a=crypto:5 AES_CM_128_HMAC_SHA1_80
> inline:f3fCmj6oXqrCGj/Wh1H38lY9liE6Awiev4Mo4syD
> a=crypto:6 AES_CM_256_HMAC_SHA1_32
> inline:d5RXBHswLr0PC+9In+CxOQVtrQ9puAdOXsM/dfyTLuTVzp1nxVDotupDNgClgw
> a=crypto:7 AES_CM_192_HMAC_SHA1_32
> inline:oI21eom6/FW+T8fXWYdYZ76w8RYJGJ1a3n/VKFWip/brFXEF/w8
> a=crypto:8 AES_CM_128_HMAC_SHA1_32
> inline:XUEC/U88QVT3YhCBunaxMftY8jY2GW4GOq2ivrhN
> a=crypto:9 AES_CM_128_NULL_AUTH
> inline:0IunsVM9Agglm6leczPQI11+bpkaZuM4x21lVK1D
> m=video 29982 RTP/AVP 97 31 34
> b=AS:1024
> a=rtpmap:97 H264/90000
> a=rtpmap:31 H261/90000
> a=rtpmap:34 H263/90000
>
>
> ========================================================================
> Please note :
> 1)  After each call is answered i am bypassing media from FS using
> 'uuid_media' API.So bypass media API was executed on step 1 and Step 3.
> 2) Sofia parameter : "resume-media-on-hold" is enabled.
> 3) All the SRTP and ZRTP related parameters are disabled.
> 4) Above issue happens only when media bypassed from FS for transferred
> calls.
> 5) NO SRTP/ZRTP channel variables were set in dialplan.
>
>
> Thanks in advance.I can provide more information if needed.
>
>
>
>
>
>

-- 
Thanks,

Sagar
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From asilva at wirelessmundi.com  Wed Oct 31 09:25:00 2018
From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=)
Date: Wed, 31 Oct 2018 10:25:00 +0100
Subject: [Freeswitch-users] FREESWITCH NOT PARSING SDP OF HOLD REINVITE
 PROPERLY
In-Reply-To: <CAEZuF3s3ZZJb8e7WFpVxhsxo3R+sHZFwyNg9biSOYJ0kK_6z1A@mail.gmail.com>
References: <CAEZuF3tD1W-PK2=Ak8UX6RuKL3UWwDbms_xSVdpPi7h0+t9Www@mail.gmail.com>
 <CAEZuF3s3ZZJb8e7WFpVxhsxo3R+sHZFwyNg9biSOYJ0kK_6z1A@mail.gmail.com>
Message-ID: <bdcd46af-6381-63fd-c333-9e4629700ba3@wirelessmundi.com>

Hi,

did you try with master version or tag 1.8.  If your problem is still 
presence on this version (at least master) fill up a jira:

https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA


Version 1.6 is deprecated.

On 30/10/2018 13:37, sagar malam wrote:
> Can anyone help me on this ?
> Please advice if this is not right place to ask for fix and need to 
> create a Jira.
>
> Thanks.
>
> On Sun, Sep 23, 2018 at 6:36 PM sagar malam <sagarmalam at gmail.com 
> <mailto:sagarmalam at gmail.com>> wrote:
>
>     Hello ,
>
>     I am working on developing PBX application using FS 1.6.17.I am
>     facing issue which i think is a bug in FS.
>     1) User A calls User B
>     2) User B blind transfers to  User C
>     3) Now User A and User C is connected.
>     4) User A puts call on hold.At this step ,When FS executes code(in
>     switch_core_media.c file )  to check if HOLD RE-INVITE has any
>     crypto attributes , it gets positive even though RE-INIVITE does
>     not have any crypto attributes.Below are the logs for fs_cli (
>     look for logs in BOLD )
>
>     =======================LOGS TAKEN FROM FS_CLI================
>
>     2018-09-23 12:41:24.408118 [DEBUG] sofia.c:7052 Channel
>     sofia/register/1025 at 202.131.119.122:53814
>     <http://1025 at 202.131.119.122:53814> entering state [received][100]
>     2018-09-23 12:41:24.408118 [DEBUG] sofia.c:7062 Remote SDP:
>     v=0
>     o=- 1537706456 1537706459 IN IP4 10.50.7.253
>     s=Polycom IP Phone
>     c=IN IP4 10.50.7.253
>     t=0 0
>     a=sendonly
>     m=audio 14852 RTP/AVP 18 101
>     a=rtpmap:18 G729/8000
>     a=fmtp:18 annexb=no
>     a=rtpmap:101 telephone-event/8000
>     a=sendonly
>     m=video 0 RTP/AVP 103 31 34
>
>     2018-09-23 12:41:24.658118 [DEBUG] sofia.c:7052 Channel
>     sofia/register/1025 at 202.131.119.122:53814
>     <http://1025 at 202.131.119.122:53814> entering state [completed][200]
>     2018-09-23 12:41:24.758055 [DEBUG] sofia.c:7052 Channel
>     sofia/register/1025 at 202.131.119.122:53814
>     <http://1025 at 202.131.119.122:53814> entering state [ready][200]
>     2018-09-23 12:41:24.898112 [DEBUG] sofia.c:7052 Channel
>     sofia/register/1013 at ecosmob.sip.teledge.com
>     <mailto:1013 at ecosmob.sip.teledge.com> entering state [calling][0]
>     2018-09-23 12:41:25.218098 [DEBUG] sofia.c:7052 Channel
>     sofia/register/1013 at ecosmob.sip.teledge.com
>     <mailto:1013 at ecosmob.sip.teledge.com> entering state [completing][200]
>     2018-09-23 12:41:25.218098 [DEBUG] sofia.c:7062 Remote SDP:
>     v=0
>     o=- 1537706452 1537706455 IN IP4 10.50.7.253
>     s=Polycom IP Phone
>     c=IN IP4 10.50.7.253
>     t=0 0
>     a=sendrecv
>     m=audio 34858 RTP/AVP 0 8 18 9 101
>     a=rtpmap:0 PCMU/8000
>     a=rtpmap:8 PCMA/8000
>     a=rtpmap:18 G729/8000
>     a=fmtp:18 annexb=no
>     a=rtpmap:9 G722/8000
>     a=rtpmap:101 telephone-event/8000
>     a=direction:active
>     m=video 0 RTP/AVP 103 31 34
>     a=direction:active
>     a=oldmediaip:172.16.16.241
>
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [PCMU:0:8000:20:64000:1]/[G729:18:8000:20:8000:1]
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4502 Audio
>     Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1]
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [PCMA:8:8000:20:64000:1]/[G729:18:8000:20:8000:1]
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4502 Audio
>     Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1]
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [G729:18:8000:20:8000:1]/[G729:18:8000:20:8000:1]
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4502 Audio
>     Codec Compare [G729:18:8000:20:8000:1] ++++ is saved as a match
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1]
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1]
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [G729:18:8000:20:8000:1]/[G722:9:8000:20:64000:1]
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [G722:9:8000:20:64000:1]/[G729:18:8000:20:8000:1]
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1]
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4502 Audio
>     Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4363 Set
>     telephone-event payload to 101 at 8000
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4706 Set
>     telephone-event payload to 101 at 8000
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:4765
>     sofia/register/1013 at ecosmob.sip.teledge.com
>     <mailto:1013 at ecosmob.sip.teledge.com> Set 2833 dtmf send payload
>     to 101 recv payload to 101
>     2018-09-23 12:41:25.218098 [DEBUG] sofia.c:7344 RESTABLISH MEDIA SDP:
>     v=0
>     o=FreeSWITCH 1537693452 1537693458 IN IP4 10.50.7.251
>     s=FreeSWITCH
>     c=IN IP4 10.50.7.251
>     t=0 0
>     m=audio 13002 RTP/AVP 0 101
>     a=rtpmap:0 PCMU/8000
>     a=rtpmap:101 telephone-event/8000
>     a=fmtp:101 0-16
>     a=ptime:20
>     a=sendrecv
>     m=video 17310 RTP/AVP 97 31 34
>     b=AS:1024
>     a=rtpmap:97 H264/90000
>     a=rtpmap:31 H261/90000
>     a=rtpmap:34 H263/90000
>
>     2018-09-23 12:41:25.218098 [DEBUG] switch_core_media.c:6860 Audio
>     params are unchanged for
>     sofia/register/1013 at ecosmob.sip.teledge.com
>     <mailto:1013 at ecosmob.sip.teledge.com>.
>     2018-09-23 12:41:25.218098 [DEBUG] sofia.c:7052 Channel
>     sofia/register/1025 at 202.131.119.122:53814
>     <http://1025 at 202.131.119.122:53814> entering state [calling][0]
>     2018-09-23 12:41:25.238128 [DEBUG] sofia.c:7052 Channel
>     sofia/register/1013 at ecosmob.sip.teledge.com
>     <mailto:1013 at ecosmob.sip.teledge.com> entering state [ready][200]
>     2018-09-23 12:41:25.498123 [DEBUG] sofia.c:7052 Channel
>     sofia/register/1025 at 202.131.119.122:53814
>     <http://1025 at 202.131.119.122:53814> entering state [completing][200]
>     2018-09-23 12:41:25.498123 [DEBUG] sofia.c:7062 Remote SDP:
>     *v=0*
>     *o=- 1537706456 1537706460 IN IP4 10.50.7.253*
>     *s=Polycom IP Phone*
>     *c=IN IP4 10.50.7.253*
>     *t=0 0*
>     *a=sendonly*
>     *m=audio 14852 RTP/AVP 18 0 8 9 101*
>     *a=rtpmap:18 G729/8000*
>     *a=fmtp:18 annexb=no*
>     *a=rtpmap:0 PCMU/8000*
>     *a=rtpmap:8 PCMA/8000*
>     *a=rtpmap:9 G722/8000*
>     *a=rtpmap:101 telephone-event/8000*
>     *a=direction:active*
>     *m=video 0 RTP/AVP 103 31 34*
>     *a=direction:active*
>     *a=oldmediaip:172.16.16.249*
>     *
>     *
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [G729:18:8000:20:8000:1]/[G729:18:8000:20:8000:1]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4502 Audio
>     Codec Compare [G729:18:8000:20:8000:1] ++++ is saved as a match*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [G729:18:8000:20:8000:1]/[G722:9:8000:20:64000:1]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [PCMU:0:8000:20:64000:1]/[G729:18:8000:20:8000:1]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4502 Audio
>     Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [PCMA:8:8000:20:64000:1]/[G729:18:8000:20:8000:1]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4502 Audio
>     Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [G722:9:8000:20:64000:1]/[G729:18:8000:20:8000:1]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4447 Audio
>     Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4502 Audio
>     Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4363 Set
>     telephone-event payload to 101 at 8000*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4706 Set
>     telephone-event payload to 101 at 8000*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:4765
>     sofia/register/1025 at 202.131.119.122:53814
>     <http://1025 at 202.131.119.122:53814> Set 2833 dtmf send payload to
>     101 recv payload to 101*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set
>     Local audio crypto Key [1 AEAD_AES_256_GCM_8
>     inline:R/pN8McEpZCkZFA8lsTNl19uoua1R8bSPnthVWaqMpse/3n/Nqk8Mb7FXcQ]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set
>     Local video crypto Key [1 AEAD_AES_256_GCM_8
>     inline:JVeqouE9G4frRG7FF5pLNBJfufVOKkrObsLVUTLImMpzRo2fKYVa2ZNU4ko]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set
>     Local audio crypto Key [2 AEAD_AES_128_GCM_8
>     inline:IJcN/B4PZbaX/WiLGLRMcGukkvHnQp/7SXLOdg]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set
>     Local video crypto Key [2 AEAD_AES_128_GCM_8
>     inline:dAlbL/sVbX2VkpVzWpWFt8/9wixwugT9pnwHeA]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set
>     Local audio crypto Key [3 AES_CM_256_HMAC_SHA1_80
>     inline:bCEgc1rOR76vYxwpTnXoblfXgSQIDLJ0jQrMEghJh9akfkn6jcSoyzgsF4JHtw]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set
>     Local video crypto Key [3 AES_CM_256_HMAC_SHA1_80
>     inline:Ae2/2r0cISukm84lGWzNL0O+r54lJgeLsjxKzxrBWAdIOlVt6fuPqXbMj+fgmA]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set
>     Local audio crypto Key [4 AES_CM_192_HMAC_SHA1_80
>     inline:Ux4vMz/xZsXLDsc1YD7cPDUSsaC/Xs8KJv+QBm5nj8v9O70NR8E]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set
>     Local video crypto Key [4 AES_CM_192_HMAC_SHA1_80
>     inline:XLdnwildLOkebarqHXItf3RnJoXM/2+6tqwL925rjW4WJtIkF74]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set
>     Local audio crypto Key [5 AES_CM_128_HMAC_SHA1_80
>     inline:soWN8HkHRNDRz2itaR8JcJ3C6PRzWYjoyLZGl2d5]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set
>     Local video crypto Key [5 AES_CM_128_HMAC_SHA1_80
>     inline:f3fCmj6oXqrCGj/Wh1H38lY9liE6Awiev4Mo4syD]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set
>     Local audio crypto Key [6 AES_CM_256_HMAC_SHA1_32
>     inline:ikMbUpS+1V5t4baA+GNGP+T3iBnN83VzOJqdI0gL3LLThnhCg/M3IxEO0Crxww]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set
>     Local video crypto Key [6 AES_CM_256_HMAC_SHA1_32
>     inline:d5RXBHswLr0PC+9In+CxOQVtrQ9puAdOXsM/dfyTLuTVzp1nxVDotupDNgClgw]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set
>     Local audio crypto Key [7 AES_CM_192_HMAC_SHA1_32
>     inline:jZ9p+h023hhYcRw6KesYK/cLOn8FYkHfWksh8vEGSktPCG6wiz0]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set
>     Local video crypto Key [7 AES_CM_192_HMAC_SHA1_32
>     inline:oI21eom6/FW+T8fXWYdYZ76w8RYJGJ1a3n/VKFWip/brFXEF/w8]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set
>     Local audio crypto Key [8 AES_CM_128_HMAC_SHA1_32
>     inline:vWCsqgSoEf8lNmgmHqTz8GUYSvm21JBbNou/fext]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set
>     Local video crypto Key [8 AES_CM_128_HMAC_SHA1_32
>     inline:XUEC/U88QVT3YhCBunaxMftY8jY2GW4GOq2ivrhN]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set
>     Local audio crypto Key [9 AES_CM_128_NULL_AUTH
>     inline:En7TIXDB7g3YBNp8SSaSyn8W/b3BjDZapNo19sbY]*
>     *2018-09-23 12:41:25.498123 [DEBUG] switch_core_media.c:1124 Set
>     Local video crypto Key [9 AES_CM_128_NULL_AUTH
>     inline:0IunsVM9Agglm6leczPQI11+bpkaZuM4x21lVK1D]*
>     2018-09-23 12:41:25.498123 [DEBUG] sofia.c:7344 RESTABLISH MEDIA SDP:
>     v=0
>     o=FreeSWITCH 1537686256 1537686263 IN IP4 10.50.7.251
>     s=FreeSWITCH
>     c=IN IP4 10.50.7.251
>     t=0 0
>     m=audio 20198 RTP/AVP 18 101
>     a=rtpmap:18 G729/8000
>     a=fmtp:18 annexb=no
>     a=rtpmap:101 telephone-event/8000
>     a=fmtp:101 0-16
>     a=ptime:20
>     a=recvonly
>     m=video 29982 RTP/SAVP 97 31 34
>     b=AS:1024
>     a=rtpmap:97 H264/90000
>     a=rtpmap:31 H261/90000
>     a=rtpmap:34 H263/90000
>     a=crypto:1 AEAD_AES_256_GCM_8
>     inline:JVeqouE9G4frRG7FF5pLNBJfufVOKkrObsLVUTLImMpzRo2fKYVa2ZNU4ko
>     a=crypto:2 AEAD_AES_128_GCM_8
>     inline:dAlbL/sVbX2VkpVzWpWFt8/9wixwugT9pnwHeA
>     a=crypto:3 AES_CM_256_HMAC_SHA1_80
>     inline:Ae2/2r0cISukm84lGWzNL0O+r54lJgeLsjxKzxrBWAdIOlVt6fuPqXbMj+fgmA
>     a=crypto:4 AES_CM_192_HMAC_SHA1_80
>     inline:XLdnwildLOkebarqHXItf3RnJoXM/2+6tqwL925rjW4WJtIkF74
>     a=crypto:5 AES_CM_128_HMAC_SHA1_80
>     inline:f3fCmj6oXqrCGj/Wh1H38lY9liE6Awiev4Mo4syD
>     a=crypto:6 AES_CM_256_HMAC_SHA1_32
>     inline:d5RXBHswLr0PC+9In+CxOQVtrQ9puAdOXsM/dfyTLuTVzp1nxVDotupDNgClgw
>     a=crypto:7 AES_CM_192_HMAC_SHA1_32
>     inline:oI21eom6/FW+T8fXWYdYZ76w8RYJGJ1a3n/VKFWip/brFXEF/w8
>     a=crypto:8 AES_CM_128_HMAC_SHA1_32
>     inline:XUEC/U88QVT3YhCBunaxMftY8jY2GW4GOq2ivrhN
>     a=crypto:9 AES_CM_128_NULL_AUTH
>     inline:0IunsVM9Agglm6leczPQI11+bpkaZuM4x21lVK1D
>     m=video 29982 RTP/AVP 97 31 34
>     b=AS:1024
>     a=rtpmap:97 H264/90000
>     a=rtpmap:31 H261/90000
>     a=rtpmap:34 H263/90000
>
>     ========================================================================
>     Please note :
>     1)  After each call is answered i am bypassing media from FS using
>     'uuid_media' API.So bypass media API was executed on step 1 and
>     Step 3.
>     2) Sofia parameter : "resume-media-on-hold" is enabled.
>     3) All the SRTP and ZRTP related parameters are disabled.
>     4) Above issue happens only when media bypassed from FS for
>     transferred calls.
>     5) NO SRTP/ZRTP channel variables were set in dialplan.
>
>
>     Thanks in advance.I can provide more information if needed.
>
>
>
>
>
>
>
> -- 
> Thanks,
>
> Sagar
>
> _________________________________________________________________________
> 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

-- 
Saludos / Regards / Cumprimentos
António Silva

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From julien.terrasson at gmail.com  Wed Oct 31 11:05:32 2018
From: julien.terrasson at gmail.com (Julien Terrasson)
Date: Wed, 31 Oct 2018 12:05:32 +0100
Subject: [Freeswitch-users] debian 9 stable release install : how to put the
	debug symbols on ?
Message-ID: <CAFi=dz819EHcktLvZ+fbw_i3FfhRcCr4+Qxa33huk8JU0vUZuA@mail.gmail.com>

I'm having coredumps in my call handling scenarios since i moved to debian
9 + FS Version 1.8.2 -3-a98a958ac3 64bit.
I need to put the debug symbols on so that the coredumps can be interpreted.

However :
Installing the stable release with debian package procedure doesn't put the
debug symbols on.
Installing the stable release with sources + compilation procedure (witch i
beleive from the Debugger
<https://freeswitch.org/confluence/display/FREESWITCH/Debugger> guidelines
should put the debug symbols on) ends up in FS Version 1.8.2 git a98a958
being installed.

How can i have the latest stable release (1.8.2 -3-a98a958ac3 64bit)
installed with the debug symbols on ?
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From mouli123 at gmail.com  Wed Oct 31 12:36:07 2018
From: mouli123 at gmail.com (Chandramouli P)
Date: Wed, 31 Oct 2018 18:06:07 +0530
Subject: [Freeswitch-users] Forward issue - FS is inviting itself
Message-ID: <CALR0W69B6XJ-_N5FZWQNAZckg2ZEHnG8z7pxKmtYXH=QNLN3Ug@mail.gmail.com>

Hello,

We have a very simple IVR application using FreeSwitch and two Yealink IP
phones and all are in same LAN. For assumption, below are the IP addresses
along with the extension numbers:

FreeSwitch server: 192.168.1.1
IP phone1 (Ext. 1456): 192.168.1.2
IP phone2 (Ext. 1459): 192.168.1.3

Based on the input received from IVR, FS is working fine and respective
extension phone is ringing. Till now, everything is working fine. Now, I
entered second phone extension number i.e. 1459 in first phone's "Forward"
configuration settings using web GUI of phone. Simply, I am doing the call
forwarding. If I enter, 1459 at 192.168.1.3 in "Forward" menu option, call is
forwarding to second phone and it is ringing. *But, If I simply enter the
extension number i.e. 1459 with out IP address of the second phone, FS is
adding it's IP address itself like 1459 at 192.168.1.1 <1459 at 192.168.1.1> and
sending invite itself and call is disconnecting.* The objective is simply I
just want to enter the extension number only with out IP address. Can
anybody share the thoughts to overcome this issue?

Thank you.

Best Regards,
Chandra.
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From sagarmalam at gmail.com  Wed Oct 31 12:49:15 2018
From: sagarmalam at gmail.com (sagar malam)
Date: Wed, 31 Oct 2018 18:19:15 +0530
Subject: [Freeswitch-users] Freeswitch crashing : Program terminated with
	signal 11, Segmentation fault.
Message-ID: <CAEZuF3ts2ivaQ2AeMw71oweeJJe9QisZCu71gARNXKVhkNZX9w@mail.gmail.com>

Hello,

I am using FS 1.6.17 for PBX system.It was working fine since last one
month.Since last two days it has started crashing randomly.All the core
dump show same error as mentioned below :

#0  hash (h=0x2e362e312f706973, k=k at entry=0x7f3ea0001100) at
./src/include/private/switch_hashtable_private.h:53
        i = <optimized out>
#1  switch_hashtable_search (h=0x2e362e312f706973, k=k at entry=0x7f3ea0001100)
at src/switch_hashtable.c:231
        e = <optimized out>
        hashvalue = 2433273728
#2  0x00007f3edfb59885 in switch_core_hash_find (hash=<optimized out>,
key=key at entry=0x7f3ea0001100 "register") at src/switch_core_hash.c:178
No locals.
#3  0x00007f3e90d28690 in sofia_glue_find_profile__
(file=file at entry=0x7f3e90e2c27a
"sofia_presence.c",
    func=func at entry=0x7f3e90e32500 <__func__.29963>
"actual_sofia_presence_mwi_event_handler", line=line at entry=521,
key=0x7f3ea0001100 "register") at sofia_glue.c:1630
        profile = <optimized out>

Please find whole core dump :
https://pastebin.com/xP2ziRMX

System is in production.Please help.
-- 
Thanks,

Sagar
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From backes at solthor.de  Wed Oct 31 15:43:36 2018
From: backes at solthor.de (Ulrich Backes)
Date: Wed, 31 Oct 2018 16:43:36 +0100 (CET)
Subject: [Freeswitch-users] Cannot activate late negotiation
Message-ID: <184568100.424524.1541000616837@webmail.strato.de>

Hi all,

how can I switch to late negotiation? FS 1.8, new installed, default settings!

internal.xml:
<param name=”inbound-late-negotiation” value=”true”/>

default.xml (added 'inherit_codec'):
<action application="set" data="inherit_codec=true"/>

Expected result: late-negotiation. Actual result: early-negotiation (see image below).

Thanks and kind regards.
Uli


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From brian at freeswitch.com  Wed Oct 31 22:17:31 2018
From: brian at freeswitch.com (Brian West)
Date: Wed, 31 Oct 2018 17:17:31 -0500
Subject: [Freeswitch-users] Freeswitch crashing : Program terminated
 with signal 11, Segmentation fault.
In-Reply-To: <CAEZuF3ts2ivaQ2AeMw71oweeJJe9QisZCu71gARNXKVhkNZX9w@mail.gmail.com>
References: <CAEZuF3ts2ivaQ2AeMw71oweeJJe9QisZCu71gARNXKVhkNZX9w@mail.gmail.com>
Message-ID: <CAEJMVkBRgm+yeL+fSBygGcEjZoiH-q9nd2art9KBtF1twODm1w@mail.gmail.com>

1.6.x is EOL and no longer supported, 1.6.20 was the last release of 1.6,
you should try that to see if the issue is fixed.  If the problem persists
please try 1.8.2 and then file a JIRA.

Thanks,
/b


On Wed, Oct 31, 2018 at 8:13 AM sagar malam <sagarmalam at gmail.com> wrote:

> Hello,
>
> I am using FS 1.6.17 for PBX system.It was working fine since last one
> month.Since last two days it has started crashing randomly.All the core
> dump show same error as mentioned below :
>
> #0  hash (h=0x2e362e312f706973, k=k at entry=0x7f3ea0001100) at
> ./src/include/private/switch_hashtable_private.h:53
>         i = <optimized out>
> #1  switch_hashtable_search (h=0x2e362e312f706973, k=k at entry=0x7f3ea0001100)
> at src/switch_hashtable.c:231
>         e = <optimized out>
>         hashvalue = 2433273728
> #2  0x00007f3edfb59885 in switch_core_hash_find (hash=<optimized out>,
> key=key at entry=0x7f3ea0001100 "register") at src/switch_core_hash.c:178
> No locals.
> #3  0x00007f3e90d28690 in sofia_glue_find_profile__ (file=file at entry=0x7f3e90e2c27a
> "sofia_presence.c",
>     func=func at entry=0x7f3e90e32500 <__func__.29963>
> "actual_sofia_presence_mwi_event_handler", line=line at entry=521,
> key=0x7f3ea0001100 "register") at sofia_glue.c:1630
>         profile = <optimized out>
>
> Please find whole core dump :
> https://pastebin.com/xP2ziRMX
>
> System is in production.Please help.
> --
> Thanks,
>
> Sagar
> _________________________________________________________________________
> 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 brian at freeswitch.com  Wed Oct 31 22:20:32 2018
From: brian at freeswitch.com (Brian West)
Date: Wed, 31 Oct 2018 17:20:32 -0500
Subject: [Freeswitch-users] Freeswitch crashing : Program terminated
 with signal 11, Segmentation fault.
In-Reply-To: <CAEJMVkBRgm+yeL+fSBygGcEjZoiH-q9nd2art9KBtF1twODm1w@mail.gmail.com>
References: <CAEZuF3ts2ivaQ2AeMw71oweeJJe9QisZCu71gARNXKVhkNZX9w@mail.gmail.com>
 <CAEJMVkBRgm+yeL+fSBygGcEjZoiH-q9nd2art9KBtF1twODm1w@mail.gmail.com>
Message-ID: <CAEJMVkCKZPE9bHAzEPXYR99Jab-so1DWwNvV7Lcwi4PWfQ7TqA@mail.gmail.com>

Guessing based on the flow of data, this is your issue too
https://freeswitch.org/jira/browse/FS-11493?

/b

On Wed, Oct 31, 2018 at 5:17 PM Brian West <brian at freeswitch.com> wrote:

> 1.6.x is EOL and no longer supported, 1.6.20 was the last release of 1.6,
> you should try that to see if the issue is fixed.  If the problem persists
> please try 1.8.2 and then file a JIRA.
>
> Thanks,
> /b
>
>
> On Wed, Oct 31, 2018 at 8:13 AM sagar malam <sagarmalam at gmail.com> wrote:
>
>> Hello,
>>
>> I am using FS 1.6.17 for PBX system.It was working fine since last one
>> month.Since last two days it has started crashing randomly.All the core
>> dump show same error as mentioned below :
>>
>> #0  hash (h=0x2e362e312f706973, k=k at entry=0x7f3ea0001100) at
>> ./src/include/private/switch_hashtable_private.h:53
>>         i = <optimized out>
>> #1  switch_hashtable_search (h=0x2e362e312f706973, k=k at entry=0x7f3ea0001100)
>> at src/switch_hashtable.c:231
>>         e = <optimized out>
>>         hashvalue = 2433273728
>> #2  0x00007f3edfb59885 in switch_core_hash_find (hash=<optimized out>,
>> key=key at entry=0x7f3ea0001100 "register") at src/switch_core_hash.c:178
>> No locals.
>> #3  0x00007f3e90d28690 in sofia_glue_find_profile__ (file=file at entry=0x7f3e90e2c27a
>> "sofia_presence.c",
>>     func=func at entry=0x7f3e90e32500 <__func__.29963>
>> "actual_sofia_presence_mwi_event_handler", line=line at entry=521,
>> key=0x7f3ea0001100 "register") at sofia_glue.c:1630
>>         profile = <optimized out>
>>
>> Please find whole core dump :
>> https://pastebin.com/xP2ziRMX
>>
>> System is in production.Please help.
>> --
>> Thanks,
>>
>> Sagar
>> _________________________________________________________________________
>> 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
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>
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> --
>
> 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
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> Mobile: 918-424-9378
>
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>
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> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
>


-- 

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 piotrek.gregor at gmail.com  Wed Oct 31 22:49:45 2018
From: piotrek.gregor at gmail.com (Piotr Gregor)
Date: Wed, 31 Oct 2018 22:49:45 +0000
Subject: [Freeswitch-users] DTMF Issue - Unexpected digits are receiving
In-Reply-To: <CALR0W6-mLbJ2nvpfxvXwVX6+pf4wMtV-3T_T14y52FM2btkb7g@mail.gmail.com>
References: <CALR0W682r3CHbf4S1ZqVTZ+f3-+LnySXXrz9Z_ukb7L5s_mkqg@mail.gmail.com>
 <CAFGRPVoB-6VXTVvO9oJthH-38ky0n5PQHZdrih1WTkyyoHML_Q@mail.gmail.com>
 <CALR0W69X_oLB8HT-874vQDZadR8sQwPYjUEuWc5AvZRjE996yA@mail.gmail.com>
 <7F36BBCD-D236-45EE-B0A0-F7DE031AC1E6@vallimamod.org>
 <CALR0W6-yvbqFncMx6aXG8QZmXHF_PCXP-cZJZeuk1hzfAVL0=A@mail.gmail.com>
 <51dc1d4b2d63413285607f7c6ed801e4@MDF-EXCH1.MDF-Holdings.local>
 <CALR0W68_=W2NMiB+Fe9VsA54ixkwBTjFT=SycZ_uZP=NFWJ49A@mail.gmail.com>
 <CALR0W6-mLbJ2nvpfxvXwVX6+pf4wMtV-3T_T14y52FM2btkb7g@mail.gmail.com>
Message-ID: <CAO3obxZ5S4Kv_NT3t1yGcbFkhKPGKt17uWPFJB2PJR7-hyatYw@mail.gmail.com>

Hi Chandramouli,

This is DTMF association to symbols:

 569      *
 570      * Tones:
 571      *          1209 Hz |   1336 Hz     | 1477 Hz   |   1633 Hz
 572      *  697 Hz  '1'         '2'             '3'         'A'
 573      *  770 Hz  '4'         '5'             '6'         'B'
 574      *  852 Hz  '7'         '8'             '9'         'C'
 575      *  941 Hz  '*'         '0'             '#'         'D'

If the device has different keypad from above then you need to find out
which key it has mapped to DTMF comprised from 697 Hz/1633 Hz frequencies.

cheers,
Piotr
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