From tanim at surroundapps.com  Wed Jan  1 07:36:53 2020
From: tanim at surroundapps.com (Md,Mehedi Hasan Kabir(Tanim))
Date: Wed, 1 Jan 2020 13:36:53 +0600
Subject: [Freeswitch-users] No audio for conference call
In-Reply-To: <CAE23trtVg8tggsmyqZrZ_fOg_q7wDo=Cs-PzbjMDJ6f9hvPmNg@mail.gmail.com>
References: <mailman.788.1577802862.1226.freeswitch-users@lists.freeswitch.org>
 <CAE23trtVg8tggsmyqZrZ_fOg_q7wDo=Cs-PzbjMDJ6f9hvPmNg@mail.gmail.com>
Message-ID: <CAF2MuSNix9HEevSDLi3UX9AkZj7m-qk_Xbi_Q10tPckzrS-y2g@mail.gmail.com>

Hi David

Search your log for 'Choose rtp'. Did you mean fs_cli log? at fs_cli log, i
didn't find anything 'Choose rtp'.

However, siptrace log is as follows

send 883 bytes to udp/[103.50.168.50]:46849 at 07:23:03.166391:

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

   SIP/2.0 407 Proxy Authentication Required

   Via: SIP/2.0/UDP 192.168.103.30:46849
;branch=z9hG4bK-524287-1---0e342a14418bcca4;rport=46849;received=103.50.168.50

   From: <sip:2112 at 52.221.122.187;transport=UDP>;tag=ee839502

   To: <sip:3000 at 52.221.122.187>;tag=B4gvey7Ha1e1p

   Call-ID: mdzZMAwx9AxiY9ax3MxA0A..

   CSeq: 1 INVITE

   User-Agent:
FreeSWITCH-mod_sofia/1.8.6+git~20190606T190249Z~93b4c92e75~64bit

   Accept: application/sdp

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

   Supported: timer, path, replaces

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

   Proxy-Authenticate: Digest realm="52.221.122.187",
nonce="4ff129a0-1845-4036-bb10-8f88d20bbdef", algorithm=MD5, qop="auth"

   Content-Length: 0


recv 813 bytes from udp/[103.50.168.50]:46849 at 07:23:03.165432:

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

   INVITE sip:3000 at 52.221.122.187;transport=UDP SIP/2.0

   Via: SIP/2.0/UDP 192.168.103.30:46849
;branch=z9hG4bK-524287-1---0e342a14418bcca4;rport

   Max-Forwards: 70

   Contact: <sip:2112 at 103.50.168.50:46849;transport=UDP>

   To: <sip:3000 at 52.221.122.187>

   From: <sip:2112 at 52.221.122.187;transport=UDP>;tag=ee839502

   Call-ID: mdzZMAwx9AxiY9ax3MxA0A..

   CSeq: 1 INVITE

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

   Content-Type: application/sdp

   User-Agent: Zoiper rv2.9.27

   Allow-Events: presence, kpml, talk

   Content-Length: 254



   v=0

   o=Z 1577863381110 1 IN IP4 103.50.168.50

   s=Z

   c=IN IP4 103.50.168.50

   t=0 0

   m=audio 55560 RTP/AVP 3 101 110 97 8 0

   a=rtpmap:101 telephone-event/8000

   a=fmtp:101 0-16

   a=rtpmap:110 speex/8000

   a=rtpmap:97 iLBC/8000

   a=fmtp:97 mode=20

   a=sendrecv

Here i see private ip at Via: SIP/2.0/UDP
192.168.103.30:46849;branch=z9hG4bK-524287-1---0e342a14418bcca4;rport=46849;received=103.50.168.50.
Is it any problem?

Regards
Tanim

On Tue, Dec 31, 2019 at 9:56 PM David P <davidswalkabout at gmail.com> wrote:

> Search your log for 'Choose rtp'; the IP number following this shows what
> FS will use for a media stream. There is a line like this for each stream.
> For your scenario, you want these not to be private IPs.
>
> If they are private, you can try to auto-reject them by changing
> acl.conf.xml. But you'll have limited success due to this bug
> https://github.com/signalwire/freeswitch/issues/157
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From tanim at surroundapps.com  Wed Jan  1 09:55:07 2020
From: tanim at surroundapps.com (Md,Mehedi Hasan Kabir(Tanim))
Date: Wed, 1 Jan 2020 15:55:07 +0600
Subject: [Freeswitch-users] No audio for conference call
In-Reply-To: <CAFGRPVqdUoMAZZzb1skuWv-zkOEj6VwJ28dE5+En-sRfFExU1w@mail.gmail.com>
References: <CAF2MuSPfgCQer1_F2=B7Thg2yU0GvH-m4F0So75n1rBX5ZeaHw@mail.gmail.com>
 <CAHtxdDc5iVU0Pb5unZWX4+3WGmnG9tVHtMBy9-TgjN1__sRoTQ@mail.gmail.com>
 <CAF2MuSPYN4QAhVOFhOfNDGz+w0qLfe2CYaVsadw6Ca_0hDK1vA@mail.gmail.com>
 <CAFGRPVpLky8z2ufsd=XtoEHLqX-XLDdBNsKyH8Ke0K6=H_xzzw@mail.gmail.com>
 <CAF2MuSPKr=vNxfzqKPU8=6o8z2j7FzXig_Q9L8cSJfy0_Qso=g@mail.gmail.com>
 <CAFGRPVqdUoMAZZzb1skuWv-zkOEj6VwJ28dE5+En-sRfFExU1w@mail.gmail.com>
Message-ID: <CAF2MuSOTfRM619ETWYwR-ZhWc7gunev_xe9Tmou5O5fqfDujJA@mail.gmail.com>

Hi David

Please find the attached log.

in cli, conference 3000-dialengine.sensor.buzz json_list command output is
as follows

[{

"conference_name": "3000-dialengine.sensor.buzz",

"member_count": 2,

"ghost_count": 0,

"rate": 8000,

"run_time": 235,

"conference_uuid": "af76568b-80e1-4def-8c0a-f2e2b3076925",

"canvas_count": 0,

"max_bw_in": 0,

"force_bw_in": 0,

"video_floor_packets": 0,

"locked": false,

"destruct": false,

"wait_mod": false,

"audio_always": false,

"running": true,

"answered": true,

"enforce_min": true,

"bridge_to": false,

"dynamic": true,

"exit_sound": true,

"enter_sound": true,

"recording": false,

"video_bridge": false,

"video_floor_only": false,

"video_rfc4579": false,

"variables": {

},

"members": [{

"type": "caller",

"id": 20,

"flags": {

"can_hear": true,

"can_see": true,

"can_speak": true,

"hold": false,

"mute_detect": false,

"talking": false,

"has_video": false,

"video_bridge": false,

"has_floor": false,

"is_moderator": false,

"end_conference": false

},

"uuid": "584d5363-6243-4051-bb3e-6b2365ea4ed6",

"caller_id_name": "2112",

"caller_id_number": "2112",

"join_time": 21,

"last_talking": 0,

"energy": 100,

"volume_in": 0,

"volume_out": 0,

"output-volume": 0,

"input-volume": 0

}, {

"type": "caller",

"id": 19,

"flags": {

"can_hear": true,

"can_see": true,

"can_speak": true,

"hold": false,

"mute_detect": false,

"talking": false,

"has_video": false,

"video_bridge": false,

"has_floor": true,

"is_moderator": false,

"end_conference": false

},

"uuid": "3eb72d21-f767-4c49-a83f-f4d6ddc78eb4",

"caller_id_name": "2111",

"caller_id_number": "2111",

"join_time": 235,

"last_talking": 0,

"energy": 100,

"volume_in": 0,

"volume_out": 0,

"output-volume": 0,

"input-volume": 0

}]

}]

Regards
Tanim

On Tue, Dec 31, 2019 at 8:33 PM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> can you please share a SIP trace and all FS logs?
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
>
> On Tue, Dec 31, 2019 at 2:59 PM Md,Mehedi Hasan Kabir(Tanim) <
> tanim at surroundapps.com> wrote:
>
>> Hi David
>>
>> Yes,ext-sip-ip and ext-rtp-ip is set.Audio is ok for call between two
>> zoiper client. problem occurs only for conference call.
>>
>> Regards
>> Tanim
>>
>>
>> On Tue, Dec 31, 2019, 4:36 PM David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Check the IP addresses offered from Zoiper and freeswitch. Make sure
>>> none is private.
>>>
>>> Is fs behind NAT? If so, did you set the ext-sip-ip and ext-rtp-ip  ?
>>>
>>> On Tue, 31 Dec 2019 at 05:24, Md,Mehedi Hasan Kabir(Tanim) <
>>> tanim at surroundapps.com> wrote:
>>>
>>>> Hi Sergey
>>>>
>>>> No, call is not disconnected. Only some noise can be heard after
>>>> enabling speaker.
>>>>
>>>> Regards
>>>> Tanim
>>>>
>>>> On Mon, Dec 30, 2019 at 7:51 PM Sergey Safarov <s.safarov at gmail.com>
>>>> wrote:
>>>>
>>>>> Is call disconnected after 34 second?
>>>>>
>>>>> On Mon, Dec 30, 2019 at 4:37 PM Md,Mehedi Hasan Kabir(Tanim) <
>>>>> tanim at surroundapps.com> wrote:
>>>>>
>>>>>> Hi
>>>>>>
>>>>>> I have tried to test FreeSWITCH conference by dialing 3000 from two
>>>>>> zoiper clients. When I call the conference number, I can hear hold
>>>>>> music. when the second call dial the conference number hold music
>>>>>> stops, but no voice between legs. what would be the reason for this?
>>>>>>
>>>>>> Audio is fine when I call other users directly.
>>>>>>
>>>>>> Regards
>>>>>> Tanim
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>
>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>> https://signalwire.com
>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>> services.
>>>>>> Build your next product on our scalable cloud platform.
>>>>>>
>>>>>> Join our online community to chat in real time
>>>>>> https://signalwire.community
>>>>>>
>>>>>> Professional FreeSWITCH Services
>>>>>> sales at freeswitch.com
>>>>>> https://freeswitch.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> https://freeswitch.com/oss
>>>>>> https://freeswitch.org/confluence
>>>>>> https://cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> https://freeswitch.com
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> --
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.work at gmail.com
>>> phone: +34669448337
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From sagarmalam at gmail.com  Wed Jan  1 12:29:09 2020
From: sagarmalam at gmail.com (sagar malam)
Date: Wed, 1 Jan 2020 17:59:09 +0530
Subject: [Freeswitch-users] uuid_simplify after intercept/bridge
In-Reply-To: <CADmXuNQBbHW90EiMAtevuurC39Kn2F3-BdYYRG8_OW6vje-RzQ@mail.gmail.com>
References: <CADmXuNT=GoQX5dMFYKJPaqzEzb-ViXyhQQry8KpAO_yvv7A75A@mail.gmail.com>
 <CE02F8BE-6D45-4F64-941A-5098993A0115@freeswitch.org>
 <CADmXuNQBbHW90EiMAtevuurC39Kn2F3-BdYYRG8_OW6vje-RzQ@mail.gmail.com>
Message-ID: <CAEZuF3v=pzXou=WpP-gR589aT0aBbbVLC9Oase+LZMEKFotSyw@mail.gmail.com>

Try this :
 <action application="set" data="api_result=${bgapi originate
{ignore_early_media=true, *api_on_media=uuid_simplify ${uuid*}
} sofia/gateway/fritzbox/**620 &intercept(${uuid})"/>




On Tue, Dec 31, 2019 at 4:27 PM Mitchell Langs <napole at gmail.com> wrote:

> Thank you for the idea, Mike!
> However bypass_media_after_bridge only handles the media. The sip messages
> are still being exchanged over Freeswitch after this. That means the calls
> between freeswitch and the gateway are still active. The gateway has a
> limit for concurrent calls. Therefore I want to call uuid_simplify when the
> bridge starts. For the gateway this would reduce the number of concurrent
> calls by two.
>
> Am Mo., 30. Dez. 2019 um 20:57 Uhr schrieb Mike Jerris <
> mike at freeswitch.org>:
>
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/bypass_media_after_bridge
>>
>> On Dec 29, 2019, at 9:57 AM, Mitchell Langs <napole at gmail.com> wrote:
>>
>> Hi,
>> I need to originate 2 calls and later bridge them. Then Freeswitch needs
>> to be removed from the call path.
>>
>> uuid_simplify accomplishes this. But where do I put this command so that
>> it is executed once the two calls are bridged? As far as I understand the
>> dialplan or scripts pause when I issue the "intercept" or "uuid_bridge"
>> command until the bridge is terminated.
>>
>> I have also tried to set sip_auto_simplify=true but it had no effect.
>>
>>
>> Here's what I am executing:
>>
>> originate
>> {codec_string=PCMA,ignore_early_media=true}sofia/gateway/fritzbox/**620
>> testing XML public
>>
>> <extension name="testing">
>>   <condition field="destination_number" expression="^testing$">
>>     <action application="set" data="api_result=${bgapi originate
>> {ignore_early_media=true}sofia/gateway/fritzbox/**620 &intercept(${uuid})"/>
>>    ...
>>     <action application="playback" data="$${hold_music}"/>
>>   </condition>
>> </extension>
>>
>> Kind Regards
>>
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Thanks,

Sagar
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From bilaln018 at gmail.com  Wed Jan  1 14:16:22 2020
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Wed, 1 Jan 2020 19:16:22 +0500
Subject: [Freeswitch-users] [Issue with setting variable while
	bridge/transfer]
Message-ID: <CADSVerC+nKtOSmWv0txasH_ftqT4eMJnB4zovv=Foactp_bCYA@mail.gmail.com>

Hi Users,
I have a switch where people are calling and they press 1 to get
transferred to client's DID.
I need to have stats for live total calls on my switch along with calls
that are transferred(realtime stats).
I am doing following commands to get the live total calls.(basically
increasing variable ${CampaignId}_live on getting call and on hangup i
decrement that variable).
<action application="set" data="ignore=${memcache(increment
${CampaignId}_live 1)}"/>
<action application="set" data="api_hangup_hook=memcache decrement
${CampaignId}_live 1"/>

I am doing this while i transfer the calls to client's number(basically
increasing variable ${CampaignId}_transfer on when somebody presses
transfer button and on hangup i decrement that variable).
<action application="bridge" data="{api_hangup_hook='memcache decrement
${CampaignId}_transfer 1 both',ignore='${memcache(increment
${CampaignId}_transfer
1)}'transfer_trigger=1}sofia/external/${TransferTo}"/>

Now i have live calls stats coming correctly.
But in case of transfer calls, when i have low number of calls(like 3-4
calls) it works good, but when i have 50-60 calls the stats are not
presented correctly(it does not increase/decrease memcache variable).

Can someone guide me how to do this in a better way?, or what mistake i am
making in doing it.

Regards
Abbasi
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From gmaruzz at gmail.com  Wed Jan  1 14:54:52 2020
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 1 Jan 2020 15:54:52 +0100
Subject: [Freeswitch-users] [Issue with setting variable while
	bridge/transfer]
In-Reply-To: <CADSVerC+nKtOSmWv0txasH_ftqT4eMJnB4zovv=Foactp_bCYA@mail.gmail.com>
References: <CADSVerC+nKtOSmWv0txasH_ftqT4eMJnB4zovv=Foactp_bCYA@mail.gmail.com>
Message-ID: <CALXCt0pPBXJhd8KzngbAyVWsOKHxyhXMwAVaaG_42YVvk4Qs0Q@mail.gmail.com>

you have probably race conditions

eg, more than one call modify same counter

maybe you want to keep a uniqe counter for each callid, and then sum them
up ?

just first thoughts, maybe completely wrong




On Wed, Jan 1, 2020 at 3:17 PM Bilal Abbasi <bilaln018 at gmail.com> wrote:

> Hi Users,
> I have a switch where people are calling and they press 1 to get
> transferred to client's DID.
> I need to have stats for live total calls on my switch along with calls
> that are transferred(realtime stats).
> I am doing following commands to get the live total calls.(basically
> increasing variable ${CampaignId}_live on getting call and on hangup i
> decrement that variable).
> <action application="set" data="ignore=${memcache(increment
> ${CampaignId}_live 1)}"/>
> <action application="set" data="api_hangup_hook=memcache decrement
> ${CampaignId}_live 1"/>
>
> I am doing this while i transfer the calls to client's number(basically
> increasing variable ${CampaignId}_transfer on when somebody presses
> transfer button and on hangup i decrement that variable).
> <action application="bridge" data="{api_hangup_hook='memcache decrement
> ${CampaignId}_transfer 1 both',ignore='${memcache(increment
> ${CampaignId}_transfer
> 1)}'transfer_trigger=1}sofia/external/${TransferTo}"/>
>
> Now i have live calls stats coming correctly.
> But in case of transfer calls, when i have low number of calls(like 3-4
> calls) it works good, but when i have 50-60 calls the stats are not
> presented correctly(it does not increase/decrease memcache variable).
>
> Can someone guide me how to do this in a better way?, or what mistake i am
> making in doing it.
>
> Regards
> Abbasi
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From bilaln018 at gmail.com  Wed Jan  1 15:45:42 2020
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Wed, 1 Jan 2020 20:45:42 +0500
Subject: [Freeswitch-users] [Issue with setting variable while
	bridge/transfer]
In-Reply-To: <CALXCt0pPBXJhd8KzngbAyVWsOKHxyhXMwAVaaG_42YVvk4Qs0Q@mail.gmail.com>
References: <CADSVerC+nKtOSmWv0txasH_ftqT4eMJnB4zovv=Foactp_bCYA@mail.gmail.com>
 <CALXCt0pPBXJhd8KzngbAyVWsOKHxyhXMwAVaaG_42YVvk4Qs0Q@mail.gmail.com>
Message-ID: <CADSVerB=uzzjdGLVk0xn6n7GsSFjkWnE1gSxaTadVbw=ziDFDw@mail.gmail.com>

Hi Marruzelli,
I am doing exactly same for live calls stats, and those are working good.
Its only issue when i do that on transfer calls.
Its not even 10CPS calls.


Regards
Abbasi

On Wed, 1 Jan 2020 at 8:24 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> you have probably race conditions
>
> eg, more than one call modify same counter
>
> maybe you want to keep a uniqe counter for each callid, and then sum them
> up ?
>
> just first thoughts, maybe completely wrong
>
>
>
>
> On Wed, Jan 1, 2020 at 3:17 PM Bilal Abbasi <bilaln018 at gmail.com> wrote:
>
>> Hi Users,
>> I have a switch where people are calling and they press 1 to get
>> transferred to client's DID.
>> I need to have stats for live total calls on my switch along with calls
>> that are transferred(realtime stats).
>> I am doing following commands to get the live total calls.(basically
>> increasing variable ${CampaignId}_live on getting call and on hangup i
>> decrement that variable).
>> <action application="set" data="ignore=${memcache(increment
>> ${CampaignId}_live 1)}"/>
>> <action application="set" data="api_hangup_hook=memcache decrement
>> ${CampaignId}_live 1"/>
>>
>> I am doing this while i transfer the calls to client's number(basically
>> increasing variable ${CampaignId}_transfer on when somebody presses
>> transfer button and on hangup i decrement that variable).
>> <action application="bridge" data="{api_hangup_hook='memcache decrement
>> ${CampaignId}_transfer 1 both',ignore='${memcache(increment
>> ${CampaignId}_transfer
>> 1)}'transfer_trigger=1}sofia/external/${TransferTo}"/>
>>
>> Now i have live calls stats coming correctly.
>> But in case of transfer calls, when i have low number of calls(like 3-4
>> calls) it works good, but when i have 50-60 calls the stats are not
>> presented correctly(it does not increase/decrease memcache variable).
>>
>> Can someone guide me how to do this in a better way?, or what mistake i
>> am making in doing it.
>>
>> Regards
>> Abbasi
>>
> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From napole at gmail.com  Wed Jan  1 16:10:03 2020
From: napole at gmail.com (Mitchell Langs)
Date: Wed, 1 Jan 2020 17:10:03 +0100
Subject: [Freeswitch-users] uuid_simplify after intercept/bridge
In-Reply-To: <CAEZuF3v=pzXou=WpP-gR589aT0aBbbVLC9Oase+LZMEKFotSyw@mail.gmail.com>
References: <CADmXuNT=GoQX5dMFYKJPaqzEzb-ViXyhQQry8KpAO_yvv7A75A@mail.gmail.com>
 <CE02F8BE-6D45-4F64-941A-5098993A0115@freeswitch.org>
 <CADmXuNQBbHW90EiMAtevuurC39Kn2F3-BdYYRG8_OW6vje-RzQ@mail.gmail.com>
 <CAEZuF3v=pzXou=WpP-gR589aT0aBbbVLC9Oase+LZMEKFotSyw@mail.gmail.com>
Message-ID: <CADmXuNSKa3uvKNiP407MApaYfZHL4KMVN-ixWE_hG0A-DW2e2Q@mail.gmail.com>

This doesn't work because the bridge has not been established when
uuid_simplify
is executed.

Am Mi., 1. Jan. 2020 um 15:14 Uhr schrieb sagar malam <sagarmalam at gmail.com
>:

> Try this :
>  <action application="set" data="api_result=${bgapi originate
> {ignore_early_media=true, *api_on_media=uuid_simplify ${uuid*}
> } sofia/gateway/fritzbox/**620 &intercept(${uuid})"/>
>
>
>
>
> On Tue, Dec 31, 2019 at 4:27 PM Mitchell Langs <napole at gmail.com> wrote:
>
>> Thank you for the idea, Mike!
>> However bypass_media_after_bridge only handles the media. The sip
>> messages are still being exchanged over Freeswitch after this. That means
>> the calls between freeswitch and the gateway are still active. The gateway
>> has a limit for concurrent calls. Therefore I want to call uuid_simplify
>> when the bridge starts. For the gateway this would reduce the number of
>> concurrent calls by two.
>>
>> Am Mo., 30. Dez. 2019 um 20:57 Uhr schrieb Mike Jerris <
>> mike at freeswitch.org>:
>>
>>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/bypass_media_after_bridge
>>>
>>> On Dec 29, 2019, at 9:57 AM, Mitchell Langs <napole at gmail.com> wrote:
>>>
>>> Hi,
>>> I need to originate 2 calls and later bridge them. Then Freeswitch needs
>>> to be removed from the call path.
>>>
>>> uuid_simplify accomplishes this. But where do I put this command so that
>>> it is executed once the two calls are bridged? As far as I understand the
>>> dialplan or scripts pause when I issue the "intercept" or "uuid_bridge"
>>> command until the bridge is terminated.
>>>
>>> I have also tried to set sip_auto_simplify=true but it had no effect.
>>>
>>>
>>> Here's what I am executing:
>>>
>>> originate
>>> {codec_string=PCMA,ignore_early_media=true}sofia/gateway/fritzbox/**620
>>> testing XML public
>>>
>>> <extension name="testing">
>>>   <condition field="destination_number" expression="^testing$">
>>>     <action application="set" data="api_result=${bgapi originate
>>> {ignore_early_media=true}sofia/gateway/fritzbox/**620 &intercept(${uuid})"/>
>>>    ...
>>>     <action application="playback" data="$${hold_music}"/>
>>>   </condition>
>>> </extension>
>>>
>>> Kind Regards
>>>
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
> Thanks,
>
> Sagar
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From obelousov at gmail.com  Wed Jan  1 17:36:55 2020
From: obelousov at gmail.com (Oleg Belousov)
Date: Wed, 1 Jan 2020 20:36:55 +0300
Subject: [Freeswitch-users] bypass_media_after_bridge
In-Reply-To: <961B66B1-260E-4B87-912F-A9895BA6B641@freeswitch.org>
References: <CAH-zrmLuhA9WDMZdrazyLrcMUABJvanf=1TFU8S5DZ-PMpOTRA@mail.gmail.com>
 <961B66B1-260E-4B87-912F-A9895BA6B641@freeswitch.org>
Message-ID: <CAH-zrmKGLtrthhD+R+sSBEnvEqtS_Nx_dYijewS7wcddBrbemQ@mail.gmail.com>

Thank you for the advice, Mike.
Have tried  'uuid_bridge', however call is dropped immediately (as per log
becouse of script exit: has executed the last dialplan instruction, hanging
up).
Tried  another bridge option - 'intercept', together with 'uuid_media off'
- it allowed to achieved desired behavior - FS send re-invite, excluding
itself from RTP path.
--
obelousov.tel


On Tue, Dec 31, 2019 at 9:51 PM Mike Jerris <mike at freeswitch.org> wrote:

> May not work with freeswitch.bridge.  try doing a uuid_bridge and see if
> it makes any difference.
>
> On Dec 31, 2019, at 7:40 AM, Oleg Belousov <obelousov at gmail.com> wrote:
>
> Happy festive season.
>
> Would you please guide me with usage of 'bypass_media_after_bridge'
> feature?
>
> Call flow is as following (running in lua script):
> 1. freeswitch preanswer call from UAC, then play some prompts
> 2, create second_session
> 3. when second session is ready - assign bypass_media_after_bridge=true
> and, see var is assigned.
> EXECUTE [depth=0] sofia/external/XXXXXXXXXX
> set(bypass_media_after_bridge=true)
> 2019-12-31 14:21:19.253279 [DEBUG] mod_dptools.c:1672 SET
> sofia/external/XXXXXXXXXX [bypass_media_after_bridge]=[true]
> 4. bridge both calls. At that time expect freeswtch fire re-invite to get
> out of rtp path, however do not see any re-invites.
>
> extract from script:
> ----
>
> ConnectStr = "sofia/gateway/aed8dd2f-202f-4093-9851-436e21bafcf4/"..DN
> second_session = freeswitch.Session(ConnectStr)
>
> if (second_session:ready()) then
>     second_session:execute("set","bypass_media_after_bridge=true")
>     freeswitch.bridge(session, second_session)
>
> ---
>
> What could be wrong in my script?
> version: FreeSWITCH Version 1.10.1-release-12
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From obelousov at gmail.com  Thu Jan  2 08:20:08 2020
From: obelousov at gmail.com (Oleg Belousov)
Date: Thu, 2 Jan 2020 11:20:08 +0300
Subject: [Freeswitch-users] uuid_simplify after intercept/bridge
In-Reply-To: <CADmXuNSKa3uvKNiP407MApaYfZHL4KMVN-ixWE_hG0A-DW2e2Q@mail.gmail.com>
References: <CADmXuNT=GoQX5dMFYKJPaqzEzb-ViXyhQQry8KpAO_yvv7A75A@mail.gmail.com>
 <CE02F8BE-6D45-4F64-941A-5098993A0115@freeswitch.org>
 <CADmXuNQBbHW90EiMAtevuurC39Kn2F3-BdYYRG8_OW6vje-RzQ@mail.gmail.com>
 <CAEZuF3v=pzXou=WpP-gR589aT0aBbbVLC9Oase+LZMEKFotSyw@mail.gmail.com>
 <CADmXuNSKa3uvKNiP407MApaYfZHL4KMVN-ixWE_hG0A-DW2e2Q@mail.gmail.com>
Message-ID: <CAH-zrm+VckmxmPVoGqxuk-w3tzSSbMDd+ZaAApLj+nwQxXWAjw@mail.gmail.com>

Bridge requires some time to establish. I believe you should use sched_api
when call uuid_simplify.
--
Regards,
Oleg.


On Wed, Jan 1, 2020 at 8:58 PM Mitchell Langs <napole at gmail.com> wrote:

> This doesn't work because the bridge has not been established when uuid_simplify
> is executed.
>
> Am Mi., 1. Jan. 2020 um 15:14 Uhr schrieb sagar malam <
> sagarmalam at gmail.com>:
>
>> Try this :
>>  <action application="set" data="api_result=${bgapi originate
>> {ignore_early_media=true, *api_on_media=uuid_simplify ${uuid*}
>> } sofia/gateway/fritzbox/**620 &intercept(${uuid})"/>
>>
>>
>>
>>
>> On Tue, Dec 31, 2019 at 4:27 PM Mitchell Langs <napole at gmail.com> wrote:
>>
>>> Thank you for the idea, Mike!
>>> However bypass_media_after_bridge only handles the media. The sip
>>> messages are still being exchanged over Freeswitch after this. That means
>>> the calls between freeswitch and the gateway are still active. The gateway
>>> has a limit for concurrent calls. Therefore I want to call uuid_simplify
>>> when the bridge starts. For the gateway this would reduce the number of
>>> concurrent calls by two.
>>>
>>> Am Mo., 30. Dez. 2019 um 20:57 Uhr schrieb Mike Jerris <
>>> mike at freeswitch.org>:
>>>
>>>>
>>>> https://freeswitch.org/confluence/display/FREESWITCH/bypass_media_after_bridge
>>>>
>>>> On Dec 29, 2019, at 9:57 AM, Mitchell Langs <napole at gmail.com> wrote:
>>>>
>>>> Hi,
>>>> I need to originate 2 calls and later bridge them. Then Freeswitch
>>>> needs to be removed from the call path.
>>>>
>>>> uuid_simplify accomplishes this. But where do I put this command so
>>>> that it is executed once the two calls are bridged? As far as I understand
>>>> the dialplan or scripts pause when I issue the "intercept" or "uuid_bridge"
>>>> command until the bridge is terminated.
>>>>
>>>> I have also tried to set sip_auto_simplify=true but it had no effect.
>>>>
>>>>
>>>> Here's what I am executing:
>>>>
>>>> originate
>>>> {codec_string=PCMA,ignore_early_media=true}sofia/gateway/fritzbox/**620
>>>> testing XML public
>>>>
>>>> <extension name="testing">
>>>>   <condition field="destination_number" expression="^testing$">
>>>>     <action application="set" data="api_result=${bgapi originate
>>>> {ignore_early_media=true}sofia/gateway/fritzbox/**620 &intercept(${uuid})"/>
>>>>    ...
>>>>     <action application="playback" data="$${hold_music}"/>
>>>>   </condition>
>>>> </extension>
>>>>
>>>> Kind Regards
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>>
>> --
>> Thanks,
>>
>> Sagar
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From davidswalkabout at gmail.com  Thu Jan  2 17:09:36 2020
From: davidswalkabout at gmail.com (David P)
Date: Thu, 2 Jan 2020 12:09:36 -0500
Subject: [Freeswitch-users] No audio for conference call
In-Reply-To: <mailman.804.1577864256.1226.freeswitch-users@lists.freeswitch.org>
References: <mailman.804.1577864256.1226.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAE23tru6jKqo7uOSdt5Xkyv9abMD2BWy7EKLbJAYyNsdMagbrg@mail.gmail.com>

You need a more detailed log, which on linux is usually at
/var/log/freeswitch/freeswitch.log IIRC. Search your file system for this
filename.

You might also need to increase your logging level; do a web search for:
freeswitch log level
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From bilaln018 at gmail.com  Thu Jan  2 17:21:28 2020
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Thu, 2 Jan 2020 22:21:28 +0500
Subject: [Freeswitch-users] [Issue with setting variable while
	bridge/transfer]
In-Reply-To: <CADSVerB=uzzjdGLVk0xn6n7GsSFjkWnE1gSxaTadVbw=ziDFDw@mail.gmail.com>
References: <CADSVerC+nKtOSmWv0txasH_ftqT4eMJnB4zovv=Foactp_bCYA@mail.gmail.com>
 <CALXCt0pPBXJhd8KzngbAyVWsOKHxyhXMwAVaaG_42YVvk4Qs0Q@mail.gmail.com>
 <CADSVerB=uzzjdGLVk0xn6n7GsSFjkWnE1gSxaTadVbw=ziDFDw@mail.gmail.com>
Message-ID: <CADSVerAGDpLG+VMm8VR=kBfHZ6ziK05d2VW9-SrdMUSj=ryjew@mail.gmail.com>

Can anyone help me out with this please.

Regards
Abbasi

On Wed, 1 Jan 2020 at 8:45 PM, Bilal Abbasi <bilaln018 at gmail.com> wrote:

> Hi Marruzelli,
> I am doing exactly same for live calls stats, and those are working good.
> Its only issue when i do that on transfer calls.
> Its not even 10CPS calls.
>
>
> Regards
> Abbasi
>
> On Wed, 1 Jan 2020 at 8:24 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> you have probably race conditions
>>
>> eg, more than one call modify same counter
>>
>> maybe you want to keep a uniqe counter for each callid, and then sum them
>> up ?
>>
>> just first thoughts, maybe completely wrong
>>
>>
>>
>>
>> On Wed, Jan 1, 2020 at 3:17 PM Bilal Abbasi <bilaln018 at gmail.com> wrote:
>>
>>> Hi Users,
>>> I have a switch where people are calling and they press 1 to get
>>> transferred to client's DID.
>>> I need to have stats for live total calls on my switch along with calls
>>> that are transferred(realtime stats).
>>> I am doing following commands to get the live total calls.(basically
>>> increasing variable ${CampaignId}_live on getting call and on hangup i
>>> decrement that variable).
>>> <action application="set" data="ignore=${memcache(increment
>>> ${CampaignId}_live 1)}"/>
>>> <action application="set" data="api_hangup_hook=memcache decrement
>>> ${CampaignId}_live 1"/>
>>>
>>> I am doing this while i transfer the calls to client's number(basically
>>> increasing variable ${CampaignId}_transfer on when somebody presses
>>> transfer button and on hangup i decrement that variable).
>>> <action application="bridge" data="{api_hangup_hook='memcache decrement
>>> ${CampaignId}_transfer 1 both',ignore='${memcache(increment
>>> ${CampaignId}_transfer
>>> 1)}'transfer_trigger=1}sofia/external/${TransferTo}"/>
>>>
>>> Now i have live calls stats coming correctly.
>>> But in case of transfer calls, when i have low number of calls(like 3-4
>>> calls) it works good, but when i have 50-60 calls the stats are not
>>> presented correctly(it does not increase/decrease memcache variable).
>>>
>>> Can someone guide me how to do this in a better way?, or what mistake i
>>> am making in doing it.
>>>
>>> Regards
>>> Abbasi
>>>
>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
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From david.villasmil.work at gmail.com  Thu Jan  2 18:31:08 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 2 Jan 2020 19:31:08 +0100
Subject: [Freeswitch-users] [Issue with setting variable while
	bridge/transfer]
In-Reply-To: <CADSVerAGDpLG+VMm8VR=kBfHZ6ziK05d2VW9-SrdMUSj=ryjew@mail.gmail.com>
References: <CADSVerC+nKtOSmWv0txasH_ftqT4eMJnB4zovv=Foactp_bCYA@mail.gmail.com>
 <CALXCt0pPBXJhd8KzngbAyVWsOKHxyhXMwAVaaG_42YVvk4Qs0Q@mail.gmail.com>
 <CADSVerB=uzzjdGLVk0xn6n7GsSFjkWnE1gSxaTadVbw=ziDFDw@mail.gmail.com>
 <CADSVerAGDpLG+VMm8VR=kBfHZ6ziK05d2VW9-SrdMUSj=ryjew@mail.gmail.com>
Message-ID: <CAFGRPVrdVXVPWnNb1RTX6M0TouQcz-_1TAUZJOaWT4MvfhwEuQ@mail.gmail.com>

Have you tried setting separate variables? I.e: increase  “conn” by 1 when
connected and increase “disc” by 1 when disconnected.

This would help you see if race conditions are messing it up. To get the
actual connected you’d simply subtract.

On Thu, 2 Jan 2020 at 18:45, Bilal Abbasi <bilaln018 at gmail.com> wrote:

> Can anyone help me out with this please.
>
> Regards
> Abbasi
>
> On Wed, 1 Jan 2020 at 8:45 PM, Bilal Abbasi <bilaln018 at gmail.com> wrote:
>
>> Hi Marruzelli,
>> I am doing exactly same for live calls stats, and those are working good.
>> Its only issue when i do that on transfer calls.
>> Its not even 10CPS calls.
>>
>>
>> Regards
>> Abbasi
>>
>> On Wed, 1 Jan 2020 at 8:24 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>
>>> you have probably race conditions
>>>
>>> eg, more than one call modify same counter
>>>
>>> maybe you want to keep a uniqe counter for each callid, and then sum
>>> them up ?
>>>
>>> just first thoughts, maybe completely wrong
>>>
>>>
>>>
>>>
>>> On Wed, Jan 1, 2020 at 3:17 PM Bilal Abbasi <bilaln018 at gmail.com> wrote:
>>>
>>>> Hi Users,
>>>> I have a switch where people are calling and they press 1 to get
>>>> transferred to client's DID.
>>>> I need to have stats for live total calls on my switch along with calls
>>>> that are transferred(realtime stats).
>>>> I am doing following commands to get the live total calls.(basically
>>>> increasing variable ${CampaignId}_live on getting call and on hangup i
>>>> decrement that variable).
>>>> <action application="set" data="ignore=${memcache(increment
>>>> ${CampaignId}_live 1)}"/>
>>>> <action application="set" data="api_hangup_hook=memcache decrement
>>>> ${CampaignId}_live 1"/>
>>>>
>>>> I am doing this while i transfer the calls to client's number(basically
>>>> increasing variable ${CampaignId}_transfer on when somebody presses
>>>> transfer button and on hangup i decrement that variable).
>>>> <action application="bridge" data="{api_hangup_hook='memcache decrement
>>>> ${CampaignId}_transfer 1 both',ignore='${memcache(increment
>>>> ${CampaignId}_transfer
>>>> 1)}'transfer_trigger=1}sofia/external/${TransferTo}"/>
>>>>
>>>> Now i have live calls stats coming correctly.
>>>> But in case of transfer calls, when i have low number of calls(like 3-4
>>>> calls) it works good, but when i have 50-60 calls the stats are not
>>>> presented correctly(it does not increase/decrease memcache variable).
>>>>
>>>> Can someone guide me how to do this in a better way?, or what mistake i
>>>> am making in doing it.
>>>>
>>>> Regards
>>>> Abbasi
>>>>
>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> OpenTelecom.IT
>>> cell: +39 347 266 56 18
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From babak.freeswitch at gmail.com  Sat Jan  4 06:44:40 2020
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Sat, 4 Jan 2020 10:14:40 +0330
Subject: [Freeswitch-users] calling sleep after answer results in no
	audio
In-Reply-To: <20378B1F-DE92-4B5A-9FC2-89E3A5B270F8@freeswitch.org>
References: <CAEVfcaZ47-tounRyEtVvkSGQcbeQpZHqN1A_tm8zerwes__jyA@mail.gmail.com>
 <20378B1F-DE92-4B5A-9FC2-89E3A5B270F8@freeswitch.org>
Message-ID: <CAEVfcaZMagvVw34EQ8EPP40AgwmW5T8v-qnSb6A=jm1fc1y+gA@mail.gmail.com>

I tested on 1.8.7 release
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From bilaln018 at gmail.com  Sat Jan  4 15:51:35 2020
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Sat, 4 Jan 2020 20:51:35 +0500
Subject: [Freeswitch-users] [Issue with setting variable while
	bridge/transfer]
In-Reply-To: <CAFGRPVrdVXVPWnNb1RTX6M0TouQcz-_1TAUZJOaWT4MvfhwEuQ@mail.gmail.com>
References: <CADSVerC+nKtOSmWv0txasH_ftqT4eMJnB4zovv=Foactp_bCYA@mail.gmail.com>
 <CALXCt0pPBXJhd8KzngbAyVWsOKHxyhXMwAVaaG_42YVvk4Qs0Q@mail.gmail.com>
 <CADSVerB=uzzjdGLVk0xn6n7GsSFjkWnE1gSxaTadVbw=ziDFDw@mail.gmail.com>
 <CADSVerAGDpLG+VMm8VR=kBfHZ6ziK05d2VW9-SrdMUSj=ryjew@mail.gmail.com>
 <CAFGRPVrdVXVPWnNb1RTX6M0TouQcz-_1TAUZJOaWT4MvfhwEuQ@mail.gmail.com>
Message-ID: <CADSVerDwETypFrhL3Ew1j8_deLTKb-yKC=CUWeds+pTmW9ttHA@mail.gmail.com>

I tried this, still dec_count and inc_count are not same.
I got a difference of 14 between them. And i got exact 14error responses
while transferring the call(looks like api hangup is not executing on
bridge resulting in error(congestions etc)).
PFA as dec is less than inc(difference is 14).

Regards
Abbasi

On Fri, 3 Jan 2020 at 12:14 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Have you tried setting separate variables? I.e: increase  “conn” by 1 when
> connected and increase “disc” by 1 when disconnected.
>
> This would help you see if race conditions are messing it up. To get the
> actual connected you’d simply subtract.
>
> On Thu, 2 Jan 2020 at 18:45, Bilal Abbasi <bilaln018 at gmail.com> wrote:
>
>> Can anyone help me out with this please.
>>
>> Regards
>> Abbasi
>>
>> On Wed, 1 Jan 2020 at 8:45 PM, Bilal Abbasi <bilaln018 at gmail.com> wrote:
>>
>>> Hi Marruzelli,
>>> I am doing exactly same for live calls stats, and those are working
>>> good. Its only issue when i do that on transfer calls.
>>> Its not even 10CPS calls.
>>>
>>>
>>> Regards
>>> Abbasi
>>>
>>> On Wed, 1 Jan 2020 at 8:24 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>
>>>> you have probably race conditions
>>>>
>>>> eg, more than one call modify same counter
>>>>
>>>> maybe you want to keep a uniqe counter for each callid, and then sum
>>>> them up ?
>>>>
>>>> just first thoughts, maybe completely wrong
>>>>
>>>>
>>>>
>>>>
>>>> On Wed, Jan 1, 2020 at 3:17 PM Bilal Abbasi <bilaln018 at gmail.com>
>>>> wrote:
>>>>
>>>>> Hi Users,
>>>>> I have a switch where people are calling and they press 1 to get
>>>>> transferred to client's DID.
>>>>> I need to have stats for live total calls on my switch along with
>>>>> calls that are transferred(realtime stats).
>>>>> I am doing following commands to get the live total calls.(basically
>>>>> increasing variable ${CampaignId}_live on getting call and on hangup i
>>>>> decrement that variable).
>>>>> <action application="set" data="ignore=${memcache(increment
>>>>> ${CampaignId}_live 1)}"/>
>>>>> <action application="set" data="api_hangup_hook=memcache decrement
>>>>> ${CampaignId}_live 1"/>
>>>>>
>>>>> I am doing this while i transfer the calls to client's
>>>>> number(basically increasing variable ${CampaignId}_transfer on when
>>>>> somebody presses transfer button and on hangup i decrement that variable).
>>>>> <action application="bridge" data="{api_hangup_hook='memcache
>>>>> decrement ${CampaignId}_transfer 1 both',ignore='${memcache(increment
>>>>> ${CampaignId}_transfer
>>>>> 1)}'transfer_trigger=1}sofia/external/${TransferTo}"/>
>>>>>
>>>>> Now i have live calls stats coming correctly.
>>>>> But in case of transfer calls, when i have low number of calls(like
>>>>> 3-4 calls) it works good, but when i have 50-60 calls the stats are not
>>>>> presented correctly(it does not increase/decrease memcache variable).
>>>>>
>>>>> Can someone guide me how to do this in a better way?, or what mistake
>>>>> i am making in doing it.
>>>>>
>>>>> Regards
>>>>> Abbasi
>>>>>
>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>>>
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> OpenTelecom.IT
>>>> cell: +39 347 266 56 18
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From borik.internet at gmail.com  Sun Jan  5 13:45:31 2020
From: borik.internet at gmail.com (Dmitriy Borisov)
Date: Sun, 5 Jan 2020 16:45:31 +0300
Subject: [Freeswitch-users] detect_speech unimrcp ... ends with 'Aborted
 (core dumped)'
In-Reply-To: <CAHtxdDdx3XHs-565iog4dXW9gTthZe5j0cL-eTWHfeLduuamDA@mail.gmail.com>
References: <CAFOPyrhy-o7Rm1CYqHUtL71m7xfhOA6S3vVsrTGbiehyPbjdrA@mail.gmail.com>
 <CAHtxdDdx3XHs-565iog4dXW9gTthZe5j0cL-eTWHfeLduuamDA@mail.gmail.com>
Message-ID: <CAFOPyriPZE8BhWnLTyYE=zg4C-540XkM=tm-hu4E4RBCNkChMg@mail.gmail.com>

Thank you! Problem is solved in 1.10 and situated only in 1.8 and earlier
versions

ср, 25 дек. 2019 г. в 04:07, Sergey Safarov <s.safarov at gmail.com>:

> Looks as buffer not initialized.
> Need investigate backtrace.
> Please create ticket here
> https://github.com/signalwire/freeswitch/issues
>
>
> On Wed, Dec 25, 2019 at 12:39 AM Dmitriy Borisov <borik.internet at gmail.com>
> wrote:
>
>> Hi, All!
>>
>> I have a strange problem... I have a big Lua script to initiate a call.
>> It starts detect_speech with
>>
>>> execute_on_media=detect_speech unimrcp ...
>>
>> Script initiate Leg B on leg A answer. And when leg B answers FreeSWITCH
>> halted with this message
>>
>>> 2019-12-25 00:00:55.903734 [INFO] switch_ivr_async.c:219 Digit parser
>>> DPTOOLS: Setting realm to 'temp'
>>> 2019-12-25 00:00:55.903734 [DEBUG] switch_ivr_async.c:344 Digit parser
>>> DPTOOLS: binding *#/temp/0 callback: 0x7fb375cf9700 data: 0x7fb2fc119818
>>> 2019-12-25 00:00:55.903734 [DEBUG] switch_ivr_bridge.c:1672
>>> (sofia/internal/1-FROM_GEN at 10.17.19.111:5060) State Change
>>> CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA
>>> 2019-12-25 00:00:55.903734 [DEBUG] switch_core_state_machine.c:584
>>> (sofia/internal/1-FROM_GEN at 10.17.19.111:5060) Running State Change
>>> CS_CONSUME_MEDIA (Cur 2 Tot 2)
>>> 2019-12-25 00:00:55.903734 [DEBUG] switch_core_state_machine.c:662
>>> (sofia/internal/1-FROM_GEN at 10.17.19.111:5060) State CONSUME_MEDIA
>>> 2019-12-25 00:00:55.903734 [DEBUG] switch_ivr_bridge.c:1063
>>> sofia/internal/1-FROM_GEN at 10.17.19.111:5060 CUSTOM HOLD
>>> 2019-12-25 00:00:55.903734 [DEBUG] switch_core_state_machine.c:662
>>> (sofia/internal/1-FROM_GEN at 10.17.19.111:5060) State CONSUME_MEDIA going
>>> to sleep
>>> freeswitch: src/switch_buffer.c:348: switch_buffer_zero: Assertion
>>> `buffer->data != ((void *)0)' failed.
>>> Aborted (core dumped)
>>>
>> I need to fix it ASAP, but I don't know where to see at all! What is it
>> can be?
>>
>> --
>> With best regards
>>    Dmitry Borisov
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
With best regards
   Dmitry Borisov
-------------- next part --------------
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From tanim at surroundapps.com  Mon Jan  6 05:48:40 2020
From: tanim at surroundapps.com (Md,Mehedi Hasan Kabir(Tanim))
Date: Mon, 6 Jan 2020 11:48:40 +0600
Subject: [Freeswitch-users] No audio for conference call
In-Reply-To: <CAF2MuSOTfRM619ETWYwR-ZhWc7gunev_xe9Tmou5O5fqfDujJA@mail.gmail.com>
References: <CAF2MuSPfgCQer1_F2=B7Thg2yU0GvH-m4F0So75n1rBX5ZeaHw@mail.gmail.com>
 <CAHtxdDc5iVU0Pb5unZWX4+3WGmnG9tVHtMBy9-TgjN1__sRoTQ@mail.gmail.com>
 <CAF2MuSPYN4QAhVOFhOfNDGz+w0qLfe2CYaVsadw6Ca_0hDK1vA@mail.gmail.com>
 <CAFGRPVpLky8z2ufsd=XtoEHLqX-XLDdBNsKyH8Ke0K6=H_xzzw@mail.gmail.com>
 <CAF2MuSPKr=vNxfzqKPU8=6o8z2j7FzXig_Q9L8cSJfy0_Qso=g@mail.gmail.com>
 <CAFGRPVqdUoMAZZzb1skuWv-zkOEj6VwJ28dE5+En-sRfFExU1w@mail.gmail.com>
 <CAF2MuSOTfRM619ETWYwR-ZhWc7gunev_xe9Tmou5O5fqfDujJA@mail.gmail.com>
Message-ID: <CAF2MuSNAp9Yk59JjVj8CqcG_AoYqSx_Bw1g6B1L85Jh_0RSeUA@mail.gmail.com>

Hi David Villasmil

Can you guess anything from the attached log?

Regards
Tanim

On Wed, Jan 1, 2020 at 3:55 PM Md,Mehedi Hasan Kabir(Tanim) <
tanim at surroundapps.com> wrote:

> Hi David
>
> Please find the attached log.
>
> in cli, conference 3000-dialengine.sensor.buzz json_list command output is
> as follows
>
> [{
>
> "conference_name": "3000-dialengine.sensor.buzz",
>
> "member_count": 2,
>
> "ghost_count": 0,
>
> "rate": 8000,
>
> "run_time": 235,
>
> "conference_uuid": "af76568b-80e1-4def-8c0a-f2e2b3076925",
>
> "canvas_count": 0,
>
> "max_bw_in": 0,
>
> "force_bw_in": 0,
>
> "video_floor_packets": 0,
>
> "locked": false,
>
> "destruct": false,
>
> "wait_mod": false,
>
> "audio_always": false,
>
> "running": true,
>
> "answered": true,
>
> "enforce_min": true,
>
> "bridge_to": false,
>
> "dynamic": true,
>
> "exit_sound": true,
>
> "enter_sound": true,
>
> "recording": false,
>
> "video_bridge": false,
>
> "video_floor_only": false,
>
> "video_rfc4579": false,
>
> "variables": {
>
> },
>
> "members": [{
>
> "type": "caller",
>
> "id": 20,
>
> "flags": {
>
> "can_hear": true,
>
> "can_see": true,
>
> "can_speak": true,
>
> "hold": false,
>
> "mute_detect": false,
>
> "talking": false,
>
> "has_video": false,
>
> "video_bridge": false,
>
> "has_floor": false,
>
> "is_moderator": false,
>
> "end_conference": false
>
> },
>
> "uuid": "584d5363-6243-4051-bb3e-6b2365ea4ed6",
>
> "caller_id_name": "2112",
>
> "caller_id_number": "2112",
>
> "join_time": 21,
>
> "last_talking": 0,
>
> "energy": 100,
>
> "volume_in": 0,
>
> "volume_out": 0,
>
> "output-volume": 0,
>
> "input-volume": 0
>
> }, {
>
> "type": "caller",
>
> "id": 19,
>
> "flags": {
>
> "can_hear": true,
>
> "can_see": true,
>
> "can_speak": true,
>
> "hold": false,
>
> "mute_detect": false,
>
> "talking": false,
>
> "has_video": false,
>
> "video_bridge": false,
>
> "has_floor": true,
>
> "is_moderator": false,
>
> "end_conference": false
>
> },
>
> "uuid": "3eb72d21-f767-4c49-a83f-f4d6ddc78eb4",
>
> "caller_id_name": "2111",
>
> "caller_id_number": "2111",
>
> "join_time": 235,
>
> "last_talking": 0,
>
> "energy": 100,
>
> "volume_in": 0,
>
> "volume_out": 0,
>
> "output-volume": 0,
>
> "input-volume": 0
>
> }]
>
> }]
>
> Regards
> Tanim
>
> On Tue, Dec 31, 2019 at 8:33 PM David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> can you please share a SIP trace and all FS logs?
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>>
>>
>> On Tue, Dec 31, 2019 at 2:59 PM Md,Mehedi Hasan Kabir(Tanim) <
>> tanim at surroundapps.com> wrote:
>>
>>> Hi David
>>>
>>> Yes,ext-sip-ip and ext-rtp-ip is set.Audio is ok for call between two
>>> zoiper client. problem occurs only for conference call.
>>>
>>> Regards
>>> Tanim
>>>
>>>
>>> On Tue, Dec 31, 2019, 4:36 PM David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>>> Check the IP addresses offered from Zoiper and freeswitch. Make sure
>>>> none is private.
>>>>
>>>> Is fs behind NAT? If so, did you set the ext-sip-ip and ext-rtp-ip  ?
>>>>
>>>> On Tue, 31 Dec 2019 at 05:24, Md,Mehedi Hasan Kabir(Tanim) <
>>>> tanim at surroundapps.com> wrote:
>>>>
>>>>> Hi Sergey
>>>>>
>>>>> No, call is not disconnected. Only some noise can be heard after
>>>>> enabling speaker.
>>>>>
>>>>> Regards
>>>>> Tanim
>>>>>
>>>>> On Mon, Dec 30, 2019 at 7:51 PM Sergey Safarov <s.safarov at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Is call disconnected after 34 second?
>>>>>>
>>>>>> On Mon, Dec 30, 2019 at 4:37 PM Md,Mehedi Hasan Kabir(Tanim) <
>>>>>> tanim at surroundapps.com> wrote:
>>>>>>
>>>>>>> Hi
>>>>>>>
>>>>>>> I have tried to test FreeSWITCH conference by dialing 3000 from two
>>>>>>> zoiper clients. When I call the conference number, I can hear hold
>>>>>>> music. when the second call dial the conference number hold music
>>>>>>> stops, but no voice between legs. what would be the reason for this?
>>>>>>>
>>>>>>> Audio is fine when I call other users directly.
>>>>>>>
>>>>>>> Regards
>>>>>>> Tanim
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>>
>>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>>> https://signalwire.com
>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>>> services.
>>>>>>> Build your next product on our scalable cloud platform.
>>>>>>>
>>>>>>> Join our online community to chat in real time
>>>>>>> https://signalwire.community
>>>>>>>
>>>>>>> Professional FreeSWITCH Services
>>>>>>> sales at freeswitch.com
>>>>>>> https://freeswitch.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> https://freeswitch.com/oss
>>>>>>> https://freeswitch.org/confluence
>>>>>>> https://cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> https://freeswitch.com
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>
>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>> https://signalwire.com
>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>> services.
>>>>>> Build your next product on our scalable cloud platform.
>>>>>>
>>>>>> Join our online community to chat in real time
>>>>>> https://signalwire.community
>>>>>>
>>>>>> Professional FreeSWITCH Services
>>>>>> sales at freeswitch.com
>>>>>> https://freeswitch.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> https://freeswitch.com/oss
>>>>>> https://freeswitch.org/confluence
>>>>>> https://cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> https://freeswitch.com
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>> --
>>>> Regards,
>>>>
>>>> David Villasmil
>>>> email: david.villasmil.work at gmail.com
>>>> phone: +34669448337
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
-------------- next part --------------
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From david.villasmil.work at gmail.com  Mon Jan  6 10:21:57 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 6 Jan 2020 11:21:57 +0100
Subject: [Freeswitch-users] No audio for conference call
In-Reply-To: <CAF2MuSNAp9Yk59JjVj8CqcG_AoYqSx_Bw1g6B1L85Jh_0RSeUA@mail.gmail.com>
References: <CAF2MuSPfgCQer1_F2=B7Thg2yU0GvH-m4F0So75n1rBX5ZeaHw@mail.gmail.com>
 <CAHtxdDc5iVU0Pb5unZWX4+3WGmnG9tVHtMBy9-TgjN1__sRoTQ@mail.gmail.com>
 <CAF2MuSPYN4QAhVOFhOfNDGz+w0qLfe2CYaVsadw6Ca_0hDK1vA@mail.gmail.com>
 <CAFGRPVpLky8z2ufsd=XtoEHLqX-XLDdBNsKyH8Ke0K6=H_xzzw@mail.gmail.com>
 <CAF2MuSPKr=vNxfzqKPU8=6o8z2j7FzXig_Q9L8cSJfy0_Qso=g@mail.gmail.com>
 <CAFGRPVqdUoMAZZzb1skuWv-zkOEj6VwJ28dE5+En-sRfFExU1w@mail.gmail.com>
 <CAF2MuSOTfRM619ETWYwR-ZhWc7gunev_xe9Tmou5O5fqfDujJA@mail.gmail.com>
 <CAF2MuSNAp9Yk59JjVj8CqcG_AoYqSx_Bw1g6B1L85Jh_0RSeUA@mail.gmail.com>
Message-ID: <CAFGRPVo2mpy6cC1wvc3iEdW1_boq=5sUeg8sGCvbEE-kJO+r7g@mail.gmail.com>

Hello,

This is a webrtc conference?

On Mon, 6 Jan 2020 at 07:07, Md,Mehedi Hasan Kabir(Tanim) <
tanim at surroundapps.com> wrote:

> Hi David Villasmil
>
> Can you guess anything from the attached log?
>
> Regards
> Tanim
>
> On Wed, Jan 1, 2020 at 3:55 PM Md,Mehedi Hasan Kabir(Tanim) <
> tanim at surroundapps.com> wrote:
>
>> Hi David
>>
>> Please find the attached log.
>>
>> in cli, conference 3000-dialengine.sensor.buzz json_list command output
>> is as follows
>>
>> [{
>>
>> "conference_name": "3000-dialengine.sensor.buzz",
>>
>> "member_count": 2,
>>
>> "ghost_count": 0,
>>
>> "rate": 8000,
>>
>> "run_time": 235,
>>
>> "conference_uuid": "af76568b-80e1-4def-8c0a-f2e2b3076925",
>>
>> "canvas_count": 0,
>>
>> "max_bw_in": 0,
>>
>> "force_bw_in": 0,
>>
>> "video_floor_packets": 0,
>>
>> "locked": false,
>>
>> "destruct": false,
>>
>> "wait_mod": false,
>>
>> "audio_always": false,
>>
>> "running": true,
>>
>> "answered": true,
>>
>> "enforce_min": true,
>>
>> "bridge_to": false,
>>
>> "dynamic": true,
>>
>> "exit_sound": true,
>>
>> "enter_sound": true,
>>
>> "recording": false,
>>
>> "video_bridge": false,
>>
>> "video_floor_only": false,
>>
>> "video_rfc4579": false,
>>
>> "variables": {
>>
>> },
>>
>> "members": [{
>>
>> "type": "caller",
>>
>> "id": 20,
>>
>> "flags": {
>>
>> "can_hear": true,
>>
>> "can_see": true,
>>
>> "can_speak": true,
>>
>> "hold": false,
>>
>> "mute_detect": false,
>>
>> "talking": false,
>>
>> "has_video": false,
>>
>> "video_bridge": false,
>>
>> "has_floor": false,
>>
>> "is_moderator": false,
>>
>> "end_conference": false
>>
>> },
>>
>> "uuid": "584d5363-6243-4051-bb3e-6b2365ea4ed6",
>>
>> "caller_id_name": "2112",
>>
>> "caller_id_number": "2112",
>>
>> "join_time": 21,
>>
>> "last_talking": 0,
>>
>> "energy": 100,
>>
>> "volume_in": 0,
>>
>> "volume_out": 0,
>>
>> "output-volume": 0,
>>
>> "input-volume": 0
>>
>> }, {
>>
>> "type": "caller",
>>
>> "id": 19,
>>
>> "flags": {
>>
>> "can_hear": true,
>>
>> "can_see": true,
>>
>> "can_speak": true,
>>
>> "hold": false,
>>
>> "mute_detect": false,
>>
>> "talking": false,
>>
>> "has_video": false,
>>
>> "video_bridge": false,
>>
>> "has_floor": true,
>>
>> "is_moderator": false,
>>
>> "end_conference": false
>>
>> },
>>
>> "uuid": "3eb72d21-f767-4c49-a83f-f4d6ddc78eb4",
>>
>> "caller_id_name": "2111",
>>
>> "caller_id_number": "2111",
>>
>> "join_time": 235,
>>
>> "last_talking": 0,
>>
>> "energy": 100,
>>
>> "volume_in": 0,
>>
>> "volume_out": 0,
>>
>> "output-volume": 0,
>>
>> "input-volume": 0
>>
>> }]
>>
>> }]
>>
>> Regards
>> Tanim
>>
>> On Tue, Dec 31, 2019 at 8:33 PM David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> can you please share a SIP trace and all FS logs?
>>>
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.work at gmail.com
>>> phone: +34669448337
>>>
>>>
>>> On Tue, Dec 31, 2019 at 2:59 PM Md,Mehedi Hasan Kabir(Tanim) <
>>> tanim at surroundapps.com> wrote:
>>>
>>>> Hi David
>>>>
>>>> Yes,ext-sip-ip and ext-rtp-ip is set.Audio is ok for call between two
>>>> zoiper client. problem occurs only for conference call.
>>>>
>>>> Regards
>>>> Tanim
>>>>
>>>>
>>>> On Tue, Dec 31, 2019, 4:36 PM David Villasmil <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>>> Check the IP addresses offered from Zoiper and freeswitch. Make sure
>>>>> none is private.
>>>>>
>>>>> Is fs behind NAT? If so, did you set the ext-sip-ip and ext-rtp-ip  ?
>>>>>
>>>>> On Tue, 31 Dec 2019 at 05:24, Md,Mehedi Hasan Kabir(Tanim) <
>>>>> tanim at surroundapps.com> wrote:
>>>>>
>>>>>> Hi Sergey
>>>>>>
>>>>>> No, call is not disconnected. Only some noise can be heard after
>>>>>> enabling speaker.
>>>>>>
>>>>>> Regards
>>>>>> Tanim
>>>>>>
>>>>>> On Mon, Dec 30, 2019 at 7:51 PM Sergey Safarov <s.safarov at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Is call disconnected after 34 second?
>>>>>>>
>>>>>>> On Mon, Dec 30, 2019 at 4:37 PM Md,Mehedi Hasan Kabir(Tanim) <
>>>>>>> tanim at surroundapps.com> wrote:
>>>>>>>
>>>>>>>> Hi
>>>>>>>>
>>>>>>>> I have tried to test FreeSWITCH conference by dialing 3000 from two
>>>>>>>> zoiper clients. When I call the conference number, I can hear hold
>>>>>>>> music. when the second call dial the conference number hold music
>>>>>>>> stops, but no voice between legs. what would be the reason for
>>>>>>>> this?
>>>>>>>>
>>>>>>>> Audio is fine when I call other users directly.
>>>>>>>>
>>>>>>>> Regards
>>>>>>>> Tanim
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>>
>>>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>>>> https://signalwire.com
>>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>>>> services.
>>>>>>>> Build your next product on our scalable cloud platform.
>>>>>>>>
>>>>>>>> Join our online community to chat in real time
>>>>>>>> https://signalwire.community
>>>>>>>>
>>>>>>>> Professional FreeSWITCH Services
>>>>>>>> sales at freeswitch.com
>>>>>>>> https://freeswitch.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> https://freeswitch.com/oss
>>>>>>>> https://freeswitch.org/confluence
>>>>>>>> https://cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> https://freeswitch.com
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>>
>>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>>> https://signalwire.com
>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>>> services.
>>>>>>> Build your next product on our scalable cloud platform.
>>>>>>>
>>>>>>> Join our online community to chat in real time
>>>>>>> https://signalwire.community
>>>>>>>
>>>>>>> Professional FreeSWITCH Services
>>>>>>> sales at freeswitch.com
>>>>>>> https://freeswitch.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> https://freeswitch.com/oss
>>>>>>> https://freeswitch.org/confluence
>>>>>>> https://cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> https://freeswitch.com
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>
>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>> https://signalwire.com
>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>> services.
>>>>>> Build your next product on our scalable cloud platform.
>>>>>>
>>>>>> Join our online community to chat in real time
>>>>>> https://signalwire.community
>>>>>>
>>>>>> Professional FreeSWITCH Services
>>>>>> sales at freeswitch.com
>>>>>> https://freeswitch.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> https://freeswitch.com/oss
>>>>>> https://freeswitch.org/confluence
>>>>>> https://cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> https://freeswitch.com
>>>>>
>>>>> --
>>>>> Regards,
>>>>>
>>>>> David Villasmil
>>>>> email: david.villasmil.work at gmail.com
>>>>> phone: +34669448337
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From tanim at surroundapps.com  Mon Jan  6 13:37:04 2020
From: tanim at surroundapps.com (Md,Mehedi Hasan Kabir(Tanim))
Date: Mon, 6 Jan 2020 19:37:04 +0600
Subject: [Freeswitch-users] No audio for conference call
In-Reply-To: <CAFGRPVo2mpy6cC1wvc3iEdW1_boq=5sUeg8sGCvbEE-kJO+r7g@mail.gmail.com>
References: <CAF2MuSPfgCQer1_F2=B7Thg2yU0GvH-m4F0So75n1rBX5ZeaHw@mail.gmail.com>
 <CAHtxdDc5iVU0Pb5unZWX4+3WGmnG9tVHtMBy9-TgjN1__sRoTQ@mail.gmail.com>
 <CAF2MuSPYN4QAhVOFhOfNDGz+w0qLfe2CYaVsadw6Ca_0hDK1vA@mail.gmail.com>
 <CAFGRPVpLky8z2ufsd=XtoEHLqX-XLDdBNsKyH8Ke0K6=H_xzzw@mail.gmail.com>
 <CAF2MuSPKr=vNxfzqKPU8=6o8z2j7FzXig_Q9L8cSJfy0_Qso=g@mail.gmail.com>
 <CAFGRPVqdUoMAZZzb1skuWv-zkOEj6VwJ28dE5+En-sRfFExU1w@mail.gmail.com>
 <CAF2MuSOTfRM619ETWYwR-ZhWc7gunev_xe9Tmou5O5fqfDujJA@mail.gmail.com>
 <CAF2MuSNAp9Yk59JjVj8CqcG_AoYqSx_Bw1g6B1L85Jh_0RSeUA@mail.gmail.com>
 <CAFGRPVo2mpy6cC1wvc3iEdW1_boq=5sUeg8sGCvbEE-kJO+r7g@mail.gmail.com>
Message-ID: <CAF2MuSOVooh_L+uWQ5W-vMWm-GiVBdgOVB=f0EPO7qHyUpHjgw@mail.gmail.com>

Hi David Villasmil

No. I just dialled 3000 from two zoiper android client to test freeswitch
default conference dialplan at conf/dialplan/default.xml. My freeswitch
version is 1.8.6.Direct audio call between two extension is fine.

Regards
Tanim

On Mon, Jan 6, 2020 at 4:21 PM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello,
>
> This is a webrtc conference?
>
> On Mon, 6 Jan 2020 at 07:07, Md,Mehedi Hasan Kabir(Tanim) <
> tanim at surroundapps.com> wrote:
>
>> Hi David Villasmil
>>
>> Can you guess anything from the attached log?
>>
>> Regards
>> Tanim
>>
>> On Wed, Jan 1, 2020 at 3:55 PM Md,Mehedi Hasan Kabir(Tanim) <
>> tanim at surroundapps.com> wrote:
>>
>>> Hi David
>>>
>>> Please find the attached log.
>>>
>>> in cli, conference 3000-dialengine.sensor.buzz json_list command output
>>> is as follows
>>>
>>> [{
>>>
>>> "conference_name": "3000-dialengine.sensor.buzz",
>>>
>>> "member_count": 2,
>>>
>>> "ghost_count": 0,
>>>
>>> "rate": 8000,
>>>
>>> "run_time": 235,
>>>
>>> "conference_uuid": "af76568b-80e1-4def-8c0a-f2e2b3076925",
>>>
>>> "canvas_count": 0,
>>>
>>> "max_bw_in": 0,
>>>
>>> "force_bw_in": 0,
>>>
>>> "video_floor_packets": 0,
>>>
>>> "locked": false,
>>>
>>> "destruct": false,
>>>
>>> "wait_mod": false,
>>>
>>> "audio_always": false,
>>>
>>> "running": true,
>>>
>>> "answered": true,
>>>
>>> "enforce_min": true,
>>>
>>> "bridge_to": false,
>>>
>>> "dynamic": true,
>>>
>>> "exit_sound": true,
>>>
>>> "enter_sound": true,
>>>
>>> "recording": false,
>>>
>>> "video_bridge": false,
>>>
>>> "video_floor_only": false,
>>>
>>> "video_rfc4579": false,
>>>
>>> "variables": {
>>>
>>> },
>>>
>>> "members": [{
>>>
>>> "type": "caller",
>>>
>>> "id": 20,
>>>
>>> "flags": {
>>>
>>> "can_hear": true,
>>>
>>> "can_see": true,
>>>
>>> "can_speak": true,
>>>
>>> "hold": false,
>>>
>>> "mute_detect": false,
>>>
>>> "talking": false,
>>>
>>> "has_video": false,
>>>
>>> "video_bridge": false,
>>>
>>> "has_floor": false,
>>>
>>> "is_moderator": false,
>>>
>>> "end_conference": false
>>>
>>> },
>>>
>>> "uuid": "584d5363-6243-4051-bb3e-6b2365ea4ed6",
>>>
>>> "caller_id_name": "2112",
>>>
>>> "caller_id_number": "2112",
>>>
>>> "join_time": 21,
>>>
>>> "last_talking": 0,
>>>
>>> "energy": 100,
>>>
>>> "volume_in": 0,
>>>
>>> "volume_out": 0,
>>>
>>> "output-volume": 0,
>>>
>>> "input-volume": 0
>>>
>>> }, {
>>>
>>> "type": "caller",
>>>
>>> "id": 19,
>>>
>>> "flags": {
>>>
>>> "can_hear": true,
>>>
>>> "can_see": true,
>>>
>>> "can_speak": true,
>>>
>>> "hold": false,
>>>
>>> "mute_detect": false,
>>>
>>> "talking": false,
>>>
>>> "has_video": false,
>>>
>>> "video_bridge": false,
>>>
>>> "has_floor": true,
>>>
>>> "is_moderator": false,
>>>
>>> "end_conference": false
>>>
>>> },
>>>
>>> "uuid": "3eb72d21-f767-4c49-a83f-f4d6ddc78eb4",
>>>
>>> "caller_id_name": "2111",
>>>
>>> "caller_id_number": "2111",
>>>
>>> "join_time": 235,
>>>
>>> "last_talking": 0,
>>>
>>> "energy": 100,
>>>
>>> "volume_in": 0,
>>>
>>> "volume_out": 0,
>>>
>>> "output-volume": 0,
>>>
>>> "input-volume": 0
>>>
>>> }]
>>>
>>> }]
>>>
>>> Regards
>>> Tanim
>>>
>>> On Tue, Dec 31, 2019 at 8:33 PM David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>>> can you please share a SIP trace and all FS logs?
>>>>
>>>> Regards,
>>>>
>>>> David Villasmil
>>>> email: david.villasmil.work at gmail.com
>>>> phone: +34669448337
>>>>
>>>>
>>>> On Tue, Dec 31, 2019 at 2:59 PM Md,Mehedi Hasan Kabir(Tanim) <
>>>> tanim at surroundapps.com> wrote:
>>>>
>>>>> Hi David
>>>>>
>>>>> Yes,ext-sip-ip and ext-rtp-ip is set.Audio is ok for call between two
>>>>> zoiper client. problem occurs only for conference call.
>>>>>
>>>>> Regards
>>>>> Tanim
>>>>>
>>>>>
>>>>> On Tue, Dec 31, 2019, 4:36 PM David Villasmil <
>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>
>>>>>> Check the IP addresses offered from Zoiper and freeswitch. Make sure
>>>>>> none is private.
>>>>>>
>>>>>> Is fs behind NAT? If so, did you set the ext-sip-ip and ext-rtp-ip  ?
>>>>>>
>>>>>> On Tue, 31 Dec 2019 at 05:24, Md,Mehedi Hasan Kabir(Tanim) <
>>>>>> tanim at surroundapps.com> wrote:
>>>>>>
>>>>>>> Hi Sergey
>>>>>>>
>>>>>>> No, call is not disconnected. Only some noise can be heard after
>>>>>>> enabling speaker.
>>>>>>>
>>>>>>> Regards
>>>>>>> Tanim
>>>>>>>
>>>>>>> On Mon, Dec 30, 2019 at 7:51 PM Sergey Safarov <s.safarov at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Is call disconnected after 34 second?
>>>>>>>>
>>>>>>>> On Mon, Dec 30, 2019 at 4:37 PM Md,Mehedi Hasan Kabir(Tanim) <
>>>>>>>> tanim at surroundapps.com> wrote:
>>>>>>>>
>>>>>>>>> Hi
>>>>>>>>>
>>>>>>>>> I have tried to test FreeSWITCH conference by dialing 3000 from
>>>>>>>>> two zoiper clients. When I call the conference number, I can hear
>>>>>>>>> hold music. when the second call dial the conference number hold
>>>>>>>>> music stops, but no voice between legs. what would be the reason
>>>>>>>>> for this?
>>>>>>>>>
>>>>>>>>> Audio is fine when I call other users directly.
>>>>>>>>>
>>>>>>>>> Regards
>>>>>>>>> Tanim
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>>
>>>>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>>>>> https://signalwire.com
>>>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and
>>>>>>>>> PSTN services.
>>>>>>>>> Build your next product on our scalable cloud platform.
>>>>>>>>>
>>>>>>>>> Join our online community to chat in real time
>>>>>>>>> https://signalwire.community
>>>>>>>>>
>>>>>>>>> Professional FreeSWITCH Services
>>>>>>>>> sales at freeswitch.com
>>>>>>>>> https://freeswitch.com
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> https://freeswitch.com/oss
>>>>>>>>> https://freeswitch.org/confluence
>>>>>>>>> https://cluecon.com
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> https://freeswitch.com
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>>
>>>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>>>> https://signalwire.com
>>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>>>> services.
>>>>>>>> Build your next product on our scalable cloud platform.
>>>>>>>>
>>>>>>>> Join our online community to chat in real time
>>>>>>>> https://signalwire.community
>>>>>>>>
>>>>>>>> Professional FreeSWITCH Services
>>>>>>>> sales at freeswitch.com
>>>>>>>> https://freeswitch.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> https://freeswitch.com/oss
>>>>>>>> https://freeswitch.org/confluence
>>>>>>>> https://cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> https://freeswitch.com
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>>
>>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>>> https://signalwire.com
>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>>> services.
>>>>>>> Build your next product on our scalable cloud platform.
>>>>>>>
>>>>>>> Join our online community to chat in real time
>>>>>>> https://signalwire.community
>>>>>>>
>>>>>>> Professional FreeSWITCH Services
>>>>>>> sales at freeswitch.com
>>>>>>> https://freeswitch.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> https://freeswitch.com/oss
>>>>>>> https://freeswitch.org/confluence
>>>>>>> https://cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> https://freeswitch.com
>>>>>>
>>>>>> --
>>>>>> Regards,
>>>>>>
>>>>>> David Villasmil
>>>>>> email: david.villasmil.work at gmail.com
>>>>>> phone: +34669448337
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>
>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>> https://signalwire.com
>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>> services.
>>>>>> Build your next product on our scalable cloud platform.
>>>>>>
>>>>>> Join our online community to chat in real time
>>>>>> https://signalwire.community
>>>>>>
>>>>>> Professional FreeSWITCH Services
>>>>>> sales at freeswitch.com
>>>>>> https://freeswitch.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> https://freeswitch.com/oss
>>>>>> https://freeswitch.org/confluence
>>>>>> https://cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> https://freeswitch.com
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From govoiper at gmail.com  Mon Jan  6 15:11:56 2020
From: govoiper at gmail.com (SamyGo)
Date: Mon, 6 Jan 2020 10:11:56 -0500
Subject: [Freeswitch-users] No audio for conference call
In-Reply-To: <CAF2MuSOVooh_L+uWQ5W-vMWm-GiVBdgOVB=f0EPO7qHyUpHjgw@mail.gmail.com>
References: <CAF2MuSPfgCQer1_F2=B7Thg2yU0GvH-m4F0So75n1rBX5ZeaHw@mail.gmail.com>
 <CAHtxdDc5iVU0Pb5unZWX4+3WGmnG9tVHtMBy9-TgjN1__sRoTQ@mail.gmail.com>
 <CAF2MuSPYN4QAhVOFhOfNDGz+w0qLfe2CYaVsadw6Ca_0hDK1vA@mail.gmail.com>
 <CAFGRPVpLky8z2ufsd=XtoEHLqX-XLDdBNsKyH8Ke0K6=H_xzzw@mail.gmail.com>
 <CAF2MuSPKr=vNxfzqKPU8=6o8z2j7FzXig_Q9L8cSJfy0_Qso=g@mail.gmail.com>
 <CAFGRPVqdUoMAZZzb1skuWv-zkOEj6VwJ28dE5+En-sRfFExU1w@mail.gmail.com>
 <CAF2MuSOTfRM619ETWYwR-ZhWc7gunev_xe9Tmou5O5fqfDujJA@mail.gmail.com>
 <CAF2MuSNAp9Yk59JjVj8CqcG_AoYqSx_Bw1g6B1L85Jh_0RSeUA@mail.gmail.com>
 <CAFGRPVo2mpy6cC1wvc3iEdW1_boq=5sUeg8sGCvbEE-kJO+r7g@mail.gmail.com>
 <CAF2MuSOVooh_L+uWQ5W-vMWm-GiVBdgOVB=f0EPO7qHyUpHjgw@mail.gmail.com>
Message-ID: <CAJUJwtjesTiBr3aO1e-6CErM943SFhukb2Ujp6LxkxBDKR39bg@mail.gmail.com>

Hi,
We had a FS 1.8.6 dev setup and got exactly the same problem, no audio from
participants in a conference. We upgraded to latest version 1.8.7 and the
conference started working.
All connected parties were able to record their joining name, and could
hear conference announcing the joining participants recorded msg but there
was no audio being mixed. So I tried undead, unmute ALL but in vain.  Only
version upgrading helped.

PS. We have a mix of WebRTC + regular devices on the conference and now
everything works fine. WebRTC users are fronted by Kamailio+RTPEngine.

Regards,
Sammy




On Mon, Jan 6, 2020 at 8:52 AM Md,Mehedi Hasan Kabir(Tanim) <
tanim at surroundapps.com> wrote:

> Hi David Villasmil
>
> No. I just dialled 3000 from two zoiper android client to test freeswitch
> default conference dialplan at conf/dialplan/default.xml. My freeswitch
> version is 1.8.6.Direct audio call between two extension is fine.
>
> Regards
> Tanim
>
> On Mon, Jan 6, 2020 at 4:21 PM David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello,
>>
>> This is a webrtc conference?
>>
>> On Mon, 6 Jan 2020 at 07:07, Md,Mehedi Hasan Kabir(Tanim) <
>> tanim at surroundapps.com> wrote:
>>
>>> Hi David Villasmil
>>>
>>> Can you guess anything from the attached log?
>>>
>>> Regards
>>> Tanim
>>>
>>> On Wed, Jan 1, 2020 at 3:55 PM Md,Mehedi Hasan Kabir(Tanim) <
>>> tanim at surroundapps.com> wrote:
>>>
>>>> Hi David
>>>>
>>>> Please find the attached log.
>>>>
>>>> in cli, conference 3000-dialengine.sensor.buzz json_list command output
>>>> is as follows
>>>>
>>>> [{
>>>>
>>>> "conference_name": "3000-dialengine.sensor.buzz",
>>>>
>>>> "member_count": 2,
>>>>
>>>> "ghost_count": 0,
>>>>
>>>> "rate": 8000,
>>>>
>>>> "run_time": 235,
>>>>
>>>> "conference_uuid": "af76568b-80e1-4def-8c0a-f2e2b3076925",
>>>>
>>>> "canvas_count": 0,
>>>>
>>>> "max_bw_in": 0,
>>>>
>>>> "force_bw_in": 0,
>>>>
>>>> "video_floor_packets": 0,
>>>>
>>>> "locked": false,
>>>>
>>>> "destruct": false,
>>>>
>>>> "wait_mod": false,
>>>>
>>>> "audio_always": false,
>>>>
>>>> "running": true,
>>>>
>>>> "answered": true,
>>>>
>>>> "enforce_min": true,
>>>>
>>>> "bridge_to": false,
>>>>
>>>> "dynamic": true,
>>>>
>>>> "exit_sound": true,
>>>>
>>>> "enter_sound": true,
>>>>
>>>> "recording": false,
>>>>
>>>> "video_bridge": false,
>>>>
>>>> "video_floor_only": false,
>>>>
>>>> "video_rfc4579": false,
>>>>
>>>> "variables": {
>>>>
>>>> },
>>>>
>>>> "members": [{
>>>>
>>>> "type": "caller",
>>>>
>>>> "id": 20,
>>>>
>>>> "flags": {
>>>>
>>>> "can_hear": true,
>>>>
>>>> "can_see": true,
>>>>
>>>> "can_speak": true,
>>>>
>>>> "hold": false,
>>>>
>>>> "mute_detect": false,
>>>>
>>>> "talking": false,
>>>>
>>>> "has_video": false,
>>>>
>>>> "video_bridge": false,
>>>>
>>>> "has_floor": false,
>>>>
>>>> "is_moderator": false,
>>>>
>>>> "end_conference": false
>>>>
>>>> },
>>>>
>>>> "uuid": "584d5363-6243-4051-bb3e-6b2365ea4ed6",
>>>>
>>>> "caller_id_name": "2112",
>>>>
>>>> "caller_id_number": "2112",
>>>>
>>>> "join_time": 21,
>>>>
>>>> "last_talking": 0,
>>>>
>>>> "energy": 100,
>>>>
>>>> "volume_in": 0,
>>>>
>>>> "volume_out": 0,
>>>>
>>>> "output-volume": 0,
>>>>
>>>> "input-volume": 0
>>>>
>>>> }, {
>>>>
>>>> "type": "caller",
>>>>
>>>> "id": 19,
>>>>
>>>> "flags": {
>>>>
>>>> "can_hear": true,
>>>>
>>>> "can_see": true,
>>>>
>>>> "can_speak": true,
>>>>
>>>> "hold": false,
>>>>
>>>> "mute_detect": false,
>>>>
>>>> "talking": false,
>>>>
>>>> "has_video": false,
>>>>
>>>> "video_bridge": false,
>>>>
>>>> "has_floor": true,
>>>>
>>>> "is_moderator": false,
>>>>
>>>> "end_conference": false
>>>>
>>>> },
>>>>
>>>> "uuid": "3eb72d21-f767-4c49-a83f-f4d6ddc78eb4",
>>>>
>>>> "caller_id_name": "2111",
>>>>
>>>> "caller_id_number": "2111",
>>>>
>>>> "join_time": 235,
>>>>
>>>> "last_talking": 0,
>>>>
>>>> "energy": 100,
>>>>
>>>> "volume_in": 0,
>>>>
>>>> "volume_out": 0,
>>>>
>>>> "output-volume": 0,
>>>>
>>>> "input-volume": 0
>>>>
>>>> }]
>>>>
>>>> }]
>>>>
>>>> Regards
>>>> Tanim
>>>>
>>>> On Tue, Dec 31, 2019 at 8:33 PM David Villasmil <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>>> can you please share a SIP trace and all FS logs?
>>>>>
>>>>> Regards,
>>>>>
>>>>> David Villasmil
>>>>> email: david.villasmil.work at gmail.com
>>>>> phone: +34669448337
>>>>>
>>>>>
>>>>> On Tue, Dec 31, 2019 at 2:59 PM Md,Mehedi Hasan Kabir(Tanim) <
>>>>> tanim at surroundapps.com> wrote:
>>>>>
>>>>>> Hi David
>>>>>>
>>>>>> Yes,ext-sip-ip and ext-rtp-ip is set.Audio is ok for call between two
>>>>>> zoiper client. problem occurs only for conference call.
>>>>>>
>>>>>> Regards
>>>>>> Tanim
>>>>>>
>>>>>>
>>>>>> On Tue, Dec 31, 2019, 4:36 PM David Villasmil <
>>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>>
>>>>>>> Check the IP addresses offered from Zoiper and freeswitch. Make sure
>>>>>>> none is private.
>>>>>>>
>>>>>>> Is fs behind NAT? If so, did you set the ext-sip-ip and ext-rtp-ip  ?
>>>>>>>
>>>>>>> On Tue, 31 Dec 2019 at 05:24, Md,Mehedi Hasan Kabir(Tanim) <
>>>>>>> tanim at surroundapps.com> wrote:
>>>>>>>
>>>>>>>> Hi Sergey
>>>>>>>>
>>>>>>>> No, call is not disconnected. Only some noise can be heard after
>>>>>>>> enabling speaker.
>>>>>>>>
>>>>>>>> Regards
>>>>>>>> Tanim
>>>>>>>>
>>>>>>>> On Mon, Dec 30, 2019 at 7:51 PM Sergey Safarov <s.safarov at gmail.com>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>>> Is call disconnected after 34 second?
>>>>>>>>>
>>>>>>>>> On Mon, Dec 30, 2019 at 4:37 PM Md,Mehedi Hasan Kabir(Tanim) <
>>>>>>>>> tanim at surroundapps.com> wrote:
>>>>>>>>>
>>>>>>>>>> Hi
>>>>>>>>>>
>>>>>>>>>> I have tried to test FreeSWITCH conference by dialing 3000 from
>>>>>>>>>> two zoiper clients. When I call the conference number, I can
>>>>>>>>>> hear hold music. when the second call dial the conference number
>>>>>>>>>> hold music stops, but no voice between legs. what would be the
>>>>>>>>>> reason for this?
>>>>>>>>>>
>>>>>>>>>> Audio is fine when I call other users directly.
>>>>>>>>>>
>>>>>>>>>> Regards
>>>>>>>>>> Tanim
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>
>>>>>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>>>>>> https://signalwire.com
>>>>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and
>>>>>>>>>> PSTN services.
>>>>>>>>>> Build your next product on our scalable cloud platform.
>>>>>>>>>>
>>>>>>>>>> Join our online community to chat in real time
>>>>>>>>>> https://signalwire.community
>>>>>>>>>>
>>>>>>>>>> Professional FreeSWITCH Services
>>>>>>>>>> sales at freeswitch.com
>>>>>>>>>> https://freeswitch.com
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> https://freeswitch.com/oss
>>>>>>>>>> https://freeswitch.org/confluence
>>>>>>>>>> https://cluecon.com
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> https://freeswitch.com
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>>
>>>>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>>>>> https://signalwire.com
>>>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and
>>>>>>>>> PSTN services.
>>>>>>>>> Build your next product on our scalable cloud platform.
>>>>>>>>>
>>>>>>>>> Join our online community to chat in real time
>>>>>>>>> https://signalwire.community
>>>>>>>>>
>>>>>>>>> Professional FreeSWITCH Services
>>>>>>>>> sales at freeswitch.com
>>>>>>>>> https://freeswitch.com
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> https://freeswitch.com/oss
>>>>>>>>> https://freeswitch.org/confluence
>>>>>>>>> https://cluecon.com
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> https://freeswitch.com
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>>
>>>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>>>> https://signalwire.com
>>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>>>> services.
>>>>>>>> Build your next product on our scalable cloud platform.
>>>>>>>>
>>>>>>>> Join our online community to chat in real time
>>>>>>>> https://signalwire.community
>>>>>>>>
>>>>>>>> Professional FreeSWITCH Services
>>>>>>>> sales at freeswitch.com
>>>>>>>> https://freeswitch.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> https://freeswitch.com/oss
>>>>>>>> https://freeswitch.org/confluence
>>>>>>>> https://cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> https://freeswitch.com
>>>>>>>
>>>>>>> --
>>>>>>> Regards,
>>>>>>>
>>>>>>> David Villasmil
>>>>>>> email: david.villasmil.work at gmail.com
>>>>>>> phone: +34669448337
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>>
>>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>>> https://signalwire.com
>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>>> services.
>>>>>>> Build your next product on our scalable cloud platform.
>>>>>>>
>>>>>>> Join our online community to chat in real time
>>>>>>> https://signalwire.community
>>>>>>>
>>>>>>> Professional FreeSWITCH Services
>>>>>>> sales at freeswitch.com
>>>>>>> https://freeswitch.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> https://freeswitch.com/oss
>>>>>>> https://freeswitch.org/confluence
>>>>>>> https://cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> https://freeswitch.com
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>
>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>> https://signalwire.com
>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>> services.
>>>>>> Build your next product on our scalable cloud platform.
>>>>>>
>>>>>> Join our online community to chat in real time
>>>>>> https://signalwire.community
>>>>>>
>>>>>> Professional FreeSWITCH Services
>>>>>> sales at freeswitch.com
>>>>>> https://freeswitch.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> https://freeswitch.com/oss
>>>>>> https://freeswitch.org/confluence
>>>>>> https://cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> https://freeswitch.com
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> --
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From napole at gmail.com  Mon Jan  6 21:43:15 2020
From: napole at gmail.com (Mitchell Langs)
Date: Mon, 6 Jan 2020 22:43:15 +0100
Subject: [Freeswitch-users] uuid_simplify after intercept/bridge
In-Reply-To: <CAH-zrm+VckmxmPVoGqxuk-w3tzSSbMDd+ZaAApLj+nwQxXWAjw@mail.gmail.com>
References: <CADmXuNT=GoQX5dMFYKJPaqzEzb-ViXyhQQry8KpAO_yvv7A75A@mail.gmail.com>
 <CE02F8BE-6D45-4F64-941A-5098993A0115@freeswitch.org>
 <CADmXuNQBbHW90EiMAtevuurC39Kn2F3-BdYYRG8_OW6vje-RzQ@mail.gmail.com>
 <CAEZuF3v=pzXou=WpP-gR589aT0aBbbVLC9Oase+LZMEKFotSyw@mail.gmail.com>
 <CADmXuNSKa3uvKNiP407MApaYfZHL4KMVN-ixWE_hG0A-DW2e2Q@mail.gmail.com>
 <CAH-zrm+VckmxmPVoGqxuk-w3tzSSbMDd+ZaAApLj+nwQxXWAjw@mail.gmail.com>
Message-ID: <CADmXuNR4H=RtGnx5dBQLv+WoU+XNWdud1cTUpfT7d1Dqsq1Umg@mail.gmail.com>

Thank you Oleg.
I guess I will have to use that.

Am Do., 2. Jan. 2020 um 09:53 Uhr schrieb Oleg Belousov <obelousov at gmail.com
>:

> Bridge requires some time to establish. I believe you should use sched_api
> when call uuid_simplify.
> --
> Regards,
> Oleg.
>
>
> On Wed, Jan 1, 2020 at 8:58 PM Mitchell Langs <napole at gmail.com> wrote:
>
>> This doesn't work because the bridge has not been established when uuid_simplify
>> is executed.
>>
>> Am Mi., 1. Jan. 2020 um 15:14 Uhr schrieb sagar malam <
>> sagarmalam at gmail.com>:
>>
>>> Try this :
>>>  <action application="set" data="api_result=${bgapi originate
>>> {ignore_early_media=true, *api_on_media=uuid_simplify ${uuid*}
>>> } sofia/gateway/fritzbox/**620 &intercept(${uuid})"/>
>>>
>>>
>>>
>>>
>>> On Tue, Dec 31, 2019 at 4:27 PM Mitchell Langs <napole at gmail.com> wrote:
>>>
>>>> Thank you for the idea, Mike!
>>>> However bypass_media_after_bridge only handles the media. The sip
>>>> messages are still being exchanged over Freeswitch after this. That means
>>>> the calls between freeswitch and the gateway are still active. The gateway
>>>> has a limit for concurrent calls. Therefore I want to call uuid_simplify
>>>> when the bridge starts. For the gateway this would reduce the number of
>>>> concurrent calls by two.
>>>>
>>>> Am Mo., 30. Dez. 2019 um 20:57 Uhr schrieb Mike Jerris <
>>>> mike at freeswitch.org>:
>>>>
>>>>>
>>>>> https://freeswitch.org/confluence/display/FREESWITCH/bypass_media_after_bridge
>>>>>
>>>>> On Dec 29, 2019, at 9:57 AM, Mitchell Langs <napole at gmail.com> wrote:
>>>>>
>>>>> Hi,
>>>>> I need to originate 2 calls and later bridge them. Then Freeswitch
>>>>> needs to be removed from the call path.
>>>>>
>>>>> uuid_simplify accomplishes this. But where do I put this command so
>>>>> that it is executed once the two calls are bridged? As far as I understand
>>>>> the dialplan or scripts pause when I issue the "intercept" or "uuid_bridge"
>>>>> command until the bridge is terminated.
>>>>>
>>>>> I have also tried to set sip_auto_simplify=true but it had no effect.
>>>>>
>>>>>
>>>>> Here's what I am executing:
>>>>>
>>>>> originate
>>>>> {codec_string=PCMA,ignore_early_media=true}sofia/gateway/fritzbox/**620
>>>>> testing XML public
>>>>>
>>>>> <extension name="testing">
>>>>>   <condition field="destination_number" expression="^testing$">
>>>>>     <action application="set" data="api_result=${bgapi originate
>>>>> {ignore_early_media=true}sofia/gateway/fritzbox/**620 &intercept(${uuid})"/>
>>>>>    ...
>>>>>     <action application="playback" data="$${hold_music}"/>
>>>>>   </condition>
>>>>> </extension>
>>>>>
>>>>> Kind Regards
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>>
>>>
>>> --
>>> Thanks,
>>>
>>> Sagar
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From tanim at surroundapps.com  Tue Jan  7 04:47:11 2020
From: tanim at surroundapps.com (Md,Mehedi Hasan Kabir(Tanim))
Date: Tue, 7 Jan 2020 10:47:11 +0600
Subject: [Freeswitch-users] No audio for conference call
In-Reply-To: <CAJUJwtjesTiBr3aO1e-6CErM943SFhukb2Ujp6LxkxBDKR39bg@mail.gmail.com>
References: <CAF2MuSPfgCQer1_F2=B7Thg2yU0GvH-m4F0So75n1rBX5ZeaHw@mail.gmail.com>
 <CAHtxdDc5iVU0Pb5unZWX4+3WGmnG9tVHtMBy9-TgjN1__sRoTQ@mail.gmail.com>
 <CAF2MuSPYN4QAhVOFhOfNDGz+w0qLfe2CYaVsadw6Ca_0hDK1vA@mail.gmail.com>
 <CAFGRPVpLky8z2ufsd=XtoEHLqX-XLDdBNsKyH8Ke0K6=H_xzzw@mail.gmail.com>
 <CAF2MuSPKr=vNxfzqKPU8=6o8z2j7FzXig_Q9L8cSJfy0_Qso=g@mail.gmail.com>
 <CAFGRPVqdUoMAZZzb1skuWv-zkOEj6VwJ28dE5+En-sRfFExU1w@mail.gmail.com>
 <CAF2MuSOTfRM619ETWYwR-ZhWc7gunev_xe9Tmou5O5fqfDujJA@mail.gmail.com>
 <CAF2MuSNAp9Yk59JjVj8CqcG_AoYqSx_Bw1g6B1L85Jh_0RSeUA@mail.gmail.com>
 <CAFGRPVo2mpy6cC1wvc3iEdW1_boq=5sUeg8sGCvbEE-kJO+r7g@mail.gmail.com>
 <CAF2MuSOVooh_L+uWQ5W-vMWm-GiVBdgOVB=f0EPO7qHyUpHjgw@mail.gmail.com>
 <CAJUJwtjesTiBr3aO1e-6CErM943SFhukb2Ujp6LxkxBDKR39bg@mail.gmail.com>
Message-ID: <CAF2MuSN1=hSx-R=nkTrOSnNOPZOcMBD+xfSAtff=qEoLmYj=vw@mail.gmail.com>

Hi Sammy

Thanks for the point

Regards
Tanim

On Mon, Jan 6, 2020 at 9:11 PM SamyGo <govoiper at gmail.com> wrote:

> Hi,
> We had a FS 1.8.6 dev setup and got exactly the same problem, no audio
> from participants in a conference. We upgraded to latest version 1.8.7 and
> the conference started working.
> All connected parties were able to record their joining name, and could
> hear conference announcing the joining participants recorded msg but there
> was no audio being mixed. So I tried undead, unmute ALL but in vain.  Only
> version upgrading helped.
>
> PS. We have a mix of WebRTC + regular devices on the conference and now
> everything works fine. WebRTC users are fronted by Kamailio+RTPEngine.
>
> Regards,
> Sammy
>
>
>
>
> On Mon, Jan 6, 2020 at 8:52 AM Md,Mehedi Hasan Kabir(Tanim) <
> tanim at surroundapps.com> wrote:
>
>> Hi David Villasmil
>>
>> No. I just dialled 3000 from two zoiper android client to test freeswitch
>> default conference dialplan at conf/dialplan/default.xml. My freeswitch
>> version is 1.8.6.Direct audio call between two extension is fine.
>>
>> Regards
>> Tanim
>>
>> On Mon, Jan 6, 2020 at 4:21 PM David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Hello,
>>>
>>> This is a webrtc conference?
>>>
>>> On Mon, 6 Jan 2020 at 07:07, Md,Mehedi Hasan Kabir(Tanim) <
>>> tanim at surroundapps.com> wrote:
>>>
>>>> Hi David Villasmil
>>>>
>>>> Can you guess anything from the attached log?
>>>>
>>>> Regards
>>>> Tanim
>>>>
>>>> On Wed, Jan 1, 2020 at 3:55 PM Md,Mehedi Hasan Kabir(Tanim) <
>>>> tanim at surroundapps.com> wrote:
>>>>
>>>>> Hi David
>>>>>
>>>>> Please find the attached log.
>>>>>
>>>>> in cli, conference 3000-dialengine.sensor.buzz json_list command
>>>>> output is as follows
>>>>>
>>>>> [{
>>>>>
>>>>> "conference_name": "3000-dialengine.sensor.buzz",
>>>>>
>>>>> "member_count": 2,
>>>>>
>>>>> "ghost_count": 0,
>>>>>
>>>>> "rate": 8000,
>>>>>
>>>>> "run_time": 235,
>>>>>
>>>>> "conference_uuid": "af76568b-80e1-4def-8c0a-f2e2b3076925",
>>>>>
>>>>> "canvas_count": 0,
>>>>>
>>>>> "max_bw_in": 0,
>>>>>
>>>>> "force_bw_in": 0,
>>>>>
>>>>> "video_floor_packets": 0,
>>>>>
>>>>> "locked": false,
>>>>>
>>>>> "destruct": false,
>>>>>
>>>>> "wait_mod": false,
>>>>>
>>>>> "audio_always": false,
>>>>>
>>>>> "running": true,
>>>>>
>>>>> "answered": true,
>>>>>
>>>>> "enforce_min": true,
>>>>>
>>>>> "bridge_to": false,
>>>>>
>>>>> "dynamic": true,
>>>>>
>>>>> "exit_sound": true,
>>>>>
>>>>> "enter_sound": true,
>>>>>
>>>>> "recording": false,
>>>>>
>>>>> "video_bridge": false,
>>>>>
>>>>> "video_floor_only": false,
>>>>>
>>>>> "video_rfc4579": false,
>>>>>
>>>>> "variables": {
>>>>>
>>>>> },
>>>>>
>>>>> "members": [{
>>>>>
>>>>> "type": "caller",
>>>>>
>>>>> "id": 20,
>>>>>
>>>>> "flags": {
>>>>>
>>>>> "can_hear": true,
>>>>>
>>>>> "can_see": true,
>>>>>
>>>>> "can_speak": true,
>>>>>
>>>>> "hold": false,
>>>>>
>>>>> "mute_detect": false,
>>>>>
>>>>> "talking": false,
>>>>>
>>>>> "has_video": false,
>>>>>
>>>>> "video_bridge": false,
>>>>>
>>>>> "has_floor": false,
>>>>>
>>>>> "is_moderator": false,
>>>>>
>>>>> "end_conference": false
>>>>>
>>>>> },
>>>>>
>>>>> "uuid": "584d5363-6243-4051-bb3e-6b2365ea4ed6",
>>>>>
>>>>> "caller_id_name": "2112",
>>>>>
>>>>> "caller_id_number": "2112",
>>>>>
>>>>> "join_time": 21,
>>>>>
>>>>> "last_talking": 0,
>>>>>
>>>>> "energy": 100,
>>>>>
>>>>> "volume_in": 0,
>>>>>
>>>>> "volume_out": 0,
>>>>>
>>>>> "output-volume": 0,
>>>>>
>>>>> "input-volume": 0
>>>>>
>>>>> }, {
>>>>>
>>>>> "type": "caller",
>>>>>
>>>>> "id": 19,
>>>>>
>>>>> "flags": {
>>>>>
>>>>> "can_hear": true,
>>>>>
>>>>> "can_see": true,
>>>>>
>>>>> "can_speak": true,
>>>>>
>>>>> "hold": false,
>>>>>
>>>>> "mute_detect": false,
>>>>>
>>>>> "talking": false,
>>>>>
>>>>> "has_video": false,
>>>>>
>>>>> "video_bridge": false,
>>>>>
>>>>> "has_floor": true,
>>>>>
>>>>> "is_moderator": false,
>>>>>
>>>>> "end_conference": false
>>>>>
>>>>> },
>>>>>
>>>>> "uuid": "3eb72d21-f767-4c49-a83f-f4d6ddc78eb4",
>>>>>
>>>>> "caller_id_name": "2111",
>>>>>
>>>>> "caller_id_number": "2111",
>>>>>
>>>>> "join_time": 235,
>>>>>
>>>>> "last_talking": 0,
>>>>>
>>>>> "energy": 100,
>>>>>
>>>>> "volume_in": 0,
>>>>>
>>>>> "volume_out": 0,
>>>>>
>>>>> "output-volume": 0,
>>>>>
>>>>> "input-volume": 0
>>>>>
>>>>> }]
>>>>>
>>>>> }]
>>>>>
>>>>> Regards
>>>>> Tanim
>>>>>
>>>>> On Tue, Dec 31, 2019 at 8:33 PM David Villasmil <
>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>
>>>>>> can you please share a SIP trace and all FS logs?
>>>>>>
>>>>>> Regards,
>>>>>>
>>>>>> David Villasmil
>>>>>> email: david.villasmil.work at gmail.com
>>>>>> phone: +34669448337
>>>>>>
>>>>>>
>>>>>> On Tue, Dec 31, 2019 at 2:59 PM Md,Mehedi Hasan Kabir(Tanim) <
>>>>>> tanim at surroundapps.com> wrote:
>>>>>>
>>>>>>> Hi David
>>>>>>>
>>>>>>> Yes,ext-sip-ip and ext-rtp-ip is set.Audio is ok for call between
>>>>>>> two zoiper client. problem occurs only for conference call.
>>>>>>>
>>>>>>> Regards
>>>>>>> Tanim
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Dec 31, 2019, 4:36 PM David Villasmil <
>>>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>>>
>>>>>>>> Check the IP addresses offered from Zoiper and freeswitch. Make
>>>>>>>> sure none is private.
>>>>>>>>
>>>>>>>> Is fs behind NAT? If so, did you set the ext-sip-ip and ext-rtp-ip
>>>>>>>> ?
>>>>>>>>
>>>>>>>> On Tue, 31 Dec 2019 at 05:24, Md,Mehedi Hasan Kabir(Tanim) <
>>>>>>>> tanim at surroundapps.com> wrote:
>>>>>>>>
>>>>>>>>> Hi Sergey
>>>>>>>>>
>>>>>>>>> No, call is not disconnected. Only some noise can be heard after
>>>>>>>>> enabling speaker.
>>>>>>>>>
>>>>>>>>> Regards
>>>>>>>>> Tanim
>>>>>>>>>
>>>>>>>>> On Mon, Dec 30, 2019 at 7:51 PM Sergey Safarov <
>>>>>>>>> s.safarov at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Is call disconnected after 34 second?
>>>>>>>>>>
>>>>>>>>>> On Mon, Dec 30, 2019 at 4:37 PM Md,Mehedi Hasan Kabir(Tanim) <
>>>>>>>>>> tanim at surroundapps.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> Hi
>>>>>>>>>>>
>>>>>>>>>>> I have tried to test FreeSWITCH conference by dialing 3000 from
>>>>>>>>>>> two zoiper clients. When I call the conference number, I can
>>>>>>>>>>> hear hold music. when the second call dial the conference
>>>>>>>>>>> number hold music stops, but no voice between legs. what would be
>>>>>>>>>>> the reason for this?
>>>>>>>>>>>
>>>>>>>>>>> Audio is fine when I call other users directly.
>>>>>>>>>>>
>>>>>>>>>>> Regards
>>>>>>>>>>> Tanim
>>>>>>>>>>>
>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>>
>>>>>>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>>>>>>> https://signalwire.com
>>>>>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and
>>>>>>>>>>> PSTN services.
>>>>>>>>>>> Build your next product on our scalable cloud platform.
>>>>>>>>>>>
>>>>>>>>>>> Join our online community to chat in real time
>>>>>>>>>>> https://signalwire.community
>>>>>>>>>>>
>>>>>>>>>>> Professional FreeSWITCH Services
>>>>>>>>>>> sales at freeswitch.com
>>>>>>>>>>> https://freeswitch.com
>>>>>>>>>>>
>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>> https://freeswitch.com/oss
>>>>>>>>>>> https://freeswitch.org/confluence
>>>>>>>>>>> https://cluecon.com
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> https://freeswitch.com
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>
>>>>>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>>>>>> https://signalwire.com
>>>>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and
>>>>>>>>>> PSTN services.
>>>>>>>>>> Build your next product on our scalable cloud platform.
>>>>>>>>>>
>>>>>>>>>> Join our online community to chat in real time
>>>>>>>>>> https://signalwire.community
>>>>>>>>>>
>>>>>>>>>> Professional FreeSWITCH Services
>>>>>>>>>> sales at freeswitch.com
>>>>>>>>>> https://freeswitch.com
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> https://freeswitch.com/oss
>>>>>>>>>> https://freeswitch.org/confluence
>>>>>>>>>> https://cluecon.com
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> https://freeswitch.com
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>>
>>>>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>>>>> https://signalwire.com
>>>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and
>>>>>>>>> PSTN services.
>>>>>>>>> Build your next product on our scalable cloud platform.
>>>>>>>>>
>>>>>>>>> Join our online community to chat in real time
>>>>>>>>> https://signalwire.community
>>>>>>>>>
>>>>>>>>> Professional FreeSWITCH Services
>>>>>>>>> sales at freeswitch.com
>>>>>>>>> https://freeswitch.com
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> https://freeswitch.com/oss
>>>>>>>>> https://freeswitch.org/confluence
>>>>>>>>> https://cluecon.com
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> https://freeswitch.com
>>>>>>>>
>>>>>>>> --
>>>>>>>> Regards,
>>>>>>>>
>>>>>>>> David Villasmil
>>>>>>>> email: david.villasmil.work at gmail.com
>>>>>>>> phone: +34669448337
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>>
>>>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>>>> https://signalwire.com
>>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>>>> services.
>>>>>>>> Build your next product on our scalable cloud platform.
>>>>>>>>
>>>>>>>> Join our online community to chat in real time
>>>>>>>> https://signalwire.community
>>>>>>>>
>>>>>>>> Professional FreeSWITCH Services
>>>>>>>> sales at freeswitch.com
>>>>>>>> https://freeswitch.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> https://freeswitch.com/oss
>>>>>>>> https://freeswitch.org/confluence
>>>>>>>> https://cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> https://freeswitch.com
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>>
>>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>>> https://signalwire.com
>>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>>> services.
>>>>>>> Build your next product on our scalable cloud platform.
>>>>>>>
>>>>>>> Join our online community to chat in real time
>>>>>>> https://signalwire.community
>>>>>>>
>>>>>>> Professional FreeSWITCH Services
>>>>>>> sales at freeswitch.com
>>>>>>> https://freeswitch.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> https://freeswitch.com/oss
>>>>>>> https://freeswitch.org/confluence
>>>>>>> https://cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> https://freeswitch.com
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>
>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>> https://signalwire.com
>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>> services.
>>>>>> Build your next product on our scalable cloud platform.
>>>>>>
>>>>>> Join our online community to chat in real time
>>>>>> https://signalwire.community
>>>>>>
>>>>>> Professional FreeSWITCH Services
>>>>>> sales at freeswitch.com
>>>>>> https://freeswitch.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> https://freeswitch.com/oss
>>>>>> https://freeswitch.org/confluence
>>>>>> https://cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> https://freeswitch.com
>>>>>
>>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> --
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.work at gmail.com
>>> phone: +34669448337
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From borik.internet at gmail.com  Tue Jan  7 14:55:22 2020
From: borik.internet at gmail.com (Dmitriy Borisov)
Date: Tue, 7 Jan 2020 17:55:22 +0300
Subject: [Freeswitch-users] uuid_simplify after intercept/bridge
In-Reply-To: <CADmXuNSKa3uvKNiP407MApaYfZHL4KMVN-ixWE_hG0A-DW2e2Q@mail.gmail.com>
References: <CADmXuNT=GoQX5dMFYKJPaqzEzb-ViXyhQQry8KpAO_yvv7A75A@mail.gmail.com>
 <CE02F8BE-6D45-4F64-941A-5098993A0115@freeswitch.org>
 <CADmXuNQBbHW90EiMAtevuurC39Kn2F3-BdYYRG8_OW6vje-RzQ@mail.gmail.com>
 <CAEZuF3v=pzXou=WpP-gR589aT0aBbbVLC9Oase+LZMEKFotSyw@mail.gmail.com>
 <CADmXuNSKa3uvKNiP407MApaYfZHL4KMVN-ixWE_hG0A-DW2e2Q@mail.gmail.com>
Message-ID: <CAFOPyrgaMxk-uay8ER1preBFZxNH1fRzKKT=bV9Ojb_+LPApRw@mail.gmail.com>

You can use api_on_originate or api_on_post_originate to run some API
command (such uuid_simplify) on a successful bridge

ср, 1 янв. 2020 г. в 20:16, Mitchell Langs <napole at gmail.com>:

> This doesn't work because the bridge has not been established when uuid_simplify
> is executed.
>
> Am Mi., 1. Jan. 2020 um 15:14 Uhr schrieb sagar malam <
> sagarmalam at gmail.com>:
>
>> Try this :
>>  <action application="set" data="api_result=${bgapi originate
>> {ignore_early_media=true, *api_on_media=uuid_simplify ${uuid*}
>> } sofia/gateway/fritzbox/**620 &intercept(${uuid})"/>
>>
>>
>>
>>
>> On Tue, Dec 31, 2019 at 4:27 PM Mitchell Langs <napole at gmail.com> wrote:
>>
>>> Thank you for the idea, Mike!
>>> However bypass_media_after_bridge only handles the media. The sip
>>> messages are still being exchanged over Freeswitch after this. That means
>>> the calls between freeswitch and the gateway are still active. The gateway
>>> has a limit for concurrent calls. Therefore I want to call uuid_simplify
>>> when the bridge starts. For the gateway this would reduce the number of
>>> concurrent calls by two.
>>>
>>> Am Mo., 30. Dez. 2019 um 20:57 Uhr schrieb Mike Jerris <
>>> mike at freeswitch.org>:
>>>
>>>>
>>>> https://freeswitch.org/confluence/display/FREESWITCH/bypass_media_after_bridge
>>>>
>>>> On Dec 29, 2019, at 9:57 AM, Mitchell Langs <napole at gmail.com> wrote:
>>>>
>>>> Hi,
>>>> I need to originate 2 calls and later bridge them. Then Freeswitch
>>>> needs to be removed from the call path.
>>>>
>>>> uuid_simplify accomplishes this. But where do I put this command so
>>>> that it is executed once the two calls are bridged? As far as I understand
>>>> the dialplan or scripts pause when I issue the "intercept" or "uuid_bridge"
>>>> command until the bridge is terminated.
>>>>
>>>> I have also tried to set sip_auto_simplify=true but it had no effect.
>>>>
>>>>
>>>> Here's what I am executing:
>>>>
>>>> originate
>>>> {codec_string=PCMA,ignore_early_media=true}sofia/gateway/fritzbox/**620
>>>> testing XML public
>>>>
>>>> <extension name="testing">
>>>>   <condition field="destination_number" expression="^testing$">
>>>>     <action application="set" data="api_result=${bgapi originate
>>>> {ignore_early_media=true}sofia/gateway/fritzbox/**620 &intercept(${uuid})"/>
>>>>    ...
>>>>     <action application="playback" data="$${hold_music}"/>
>>>>   </condition>
>>>> </extension>
>>>>
>>>> Kind Regards
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>>
>> --
>> Thanks,
>>
>> Sagar
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
With best regards
   Dmitry Borisov
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From napole at gmail.com  Tue Jan  7 21:00:23 2020
From: napole at gmail.com (Mitchell Langs)
Date: Tue, 7 Jan 2020 22:00:23 +0100
Subject: [Freeswitch-users] uuid_simplify after intercept/bridge
In-Reply-To: <CAFOPyrgaMxk-uay8ER1preBFZxNH1fRzKKT=bV9Ojb_+LPApRw@mail.gmail.com>
References: <CADmXuNT=GoQX5dMFYKJPaqzEzb-ViXyhQQry8KpAO_yvv7A75A@mail.gmail.com>
 <CE02F8BE-6D45-4F64-941A-5098993A0115@freeswitch.org>
 <CADmXuNQBbHW90EiMAtevuurC39Kn2F3-BdYYRG8_OW6vje-RzQ@mail.gmail.com>
 <CAEZuF3v=pzXou=WpP-gR589aT0aBbbVLC9Oase+LZMEKFotSyw@mail.gmail.com>
 <CADmXuNSKa3uvKNiP407MApaYfZHL4KMVN-ixWE_hG0A-DW2e2Q@mail.gmail.com>
 <CAFOPyrgaMxk-uay8ER1preBFZxNH1fRzKKT=bV9Ojb_+LPApRw@mail.gmail.com>
Message-ID: <CADmXuNTtm1UoNGFiMN8G7fo5PQa8h1Y4+OCxK8GLoVsHpD+k9g@mail.gmail.com>

No, in my setup I originate 2 calls and only after they are both answered
bridge/intercept them. Therefore api_on_post_originate will run before they
are bridged.

Am Di., 7. Jan. 2020 um 16:37 Uhr schrieb Dmitriy Borisov <
borik.internet at gmail.com>:

> You can use api_on_originate or api_on_post_originate to run some API
> command (such uuid_simplify) on a successful bridge
>
> ср, 1 янв. 2020 г. в 20:16, Mitchell Langs <napole at gmail.com>:
>
>> This doesn't work because the bridge has not been established when uuid_simplify
>> is executed.
>>
>> Am Mi., 1. Jan. 2020 um 15:14 Uhr schrieb sagar malam <
>> sagarmalam at gmail.com>:
>>
>>> Try this :
>>>  <action application="set" data="api_result=${bgapi originate
>>> {ignore_early_media=true, *api_on_media=uuid_simplify ${uuid*}
>>> } sofia/gateway/fritzbox/**620 &intercept(${uuid})"/>
>>>
>>>
>>>
>>>
>>> On Tue, Dec 31, 2019 at 4:27 PM Mitchell Langs <napole at gmail.com> wrote:
>>>
>>>> Thank you for the idea, Mike!
>>>> However bypass_media_after_bridge only handles the media. The sip
>>>> messages are still being exchanged over Freeswitch after this. That means
>>>> the calls between freeswitch and the gateway are still active. The gateway
>>>> has a limit for concurrent calls. Therefore I want to call uuid_simplify
>>>> when the bridge starts. For the gateway this would reduce the number of
>>>> concurrent calls by two.
>>>>
>>>> Am Mo., 30. Dez. 2019 um 20:57 Uhr schrieb Mike Jerris <
>>>> mike at freeswitch.org>:
>>>>
>>>>>
>>>>> https://freeswitch.org/confluence/display/FREESWITCH/bypass_media_after_bridge
>>>>>
>>>>> On Dec 29, 2019, at 9:57 AM, Mitchell Langs <napole at gmail.com> wrote:
>>>>>
>>>>> Hi,
>>>>> I need to originate 2 calls and later bridge them. Then Freeswitch
>>>>> needs to be removed from the call path.
>>>>>
>>>>> uuid_simplify accomplishes this. But where do I put this command so
>>>>> that it is executed once the two calls are bridged? As far as I understand
>>>>> the dialplan or scripts pause when I issue the "intercept" or "uuid_bridge"
>>>>> command until the bridge is terminated.
>>>>>
>>>>> I have also tried to set sip_auto_simplify=true but it had no effect.
>>>>>
>>>>>
>>>>> Here's what I am executing:
>>>>>
>>>>> originate
>>>>> {codec_string=PCMA,ignore_early_media=true}sofia/gateway/fritzbox/**620
>>>>> testing XML public
>>>>>
>>>>> <extension name="testing">
>>>>>   <condition field="destination_number" expression="^testing$">
>>>>>     <action application="set" data="api_result=${bgapi originate
>>>>> {ignore_early_media=true}sofia/gateway/fritzbox/**620 &intercept(${uuid})"/>
>>>>>    ...
>>>>>     <action application="playback" data="$${hold_music}"/>
>>>>>   </condition>
>>>>> </extension>
>>>>>
>>>>> Kind Regards
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> The FreeSWITCH project is sponsored by SignalWire
>>>> https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>> services.
>>>> Build your next product on our scalable cloud platform.
>>>>
>>>> Join our online community to chat in real time
>>>> https://signalwire.community
>>>>
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>>
>>>
>>> --
>>> Thanks,
>>>
>>> Sagar
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
>
> --
> With best regards
>    Dmitry Borisov
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From Alexander.Haugg at c4b.de  Thu Jan  9 07:17:04 2020
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Thu, 9 Jan 2020 07:17:04 +0000
Subject: [Freeswitch-users] WebRTC STUN request for a SIP profile is not
	working
Message-ID: <5bf5b03fd2ee48b3afbb1cd6f0b7ed24@c4b.de>

Hi,

I’d read the manuals and configure my sofia profile to work with the external IP in the ext-rtp-ip setting.
<param name="stun-enabled" value="true"/>
<param name="stun-auto-disable" value="false"/>
<param name="ext-rtp-ip" value="stun:stun.freeswitch.org" />

The Freeswitch is started with the parameter –nonat
In the pcap trace is no STUN request visible.
I try to restart the Freeswitch, restart the profile, do an outbound call, but there is no STUN request.
If I try to get my external IP via the CLI cmd “stun stun.freeswitch.org”, then works it successfully.

The status output of the profile is:
=================================================================================================
Name                                SBC2_B2Bua
Domain Name                 N/A
Auto-NAT                         false
DBName                           sofia_reg_SBC2_B2Bua
Pres Hosts
Dialplan                            XML
Context                            SBC2_SP_outbound
Challenge Realm             auto_to
RTP-IP                               MY_LOCAL_IP
Ext-RTP-IP                        stun:stun.freeswitch.org
SIP-IP                                MY_LOCAL_IP
Ext-SIP-IP                         MY_LOCAL_IP
URL                                   sip:mod_sofia at MY_LOCAL_IP:4901
BIND-URL                  sip:mod_sofia at MY_LOCAL_IP:4901;maddr=MY_LOCAL_IP;transport=udp,tcp
HOLD-MUSIC                   local_stream://moh
OUTBOUND-PROXY       N/A
CODECS IN                       OPUS,PCMU,PCMA,VP8
CODECS OUT                   OPUS,PCMU,PCMA,VP8
TEL-EVENT                       98
DTMF-MODE                   rfc2833
CNG                                  13
SESSION-TO                     0
MAX-DIALOG                   0
NOMEDIA                        false
LATE-NEG                         true
PROXY-MEDIA                 false
ZRTP-PASSTHRU             false
AGGRESSIVENAT           false
CALLS-IN                           0
FAILED-CALLS-IN             0
CALLS-OUT                       0
FAILED-CALLS-OUT        0
REGISTRATIONS              0

Thanks a lot!!!
Alex


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From Alexander.Haugg at c4b.de  Fri Jan 10 12:20:49 2020
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Fri, 10 Jan 2020 12:20:49 +0000
Subject: [Freeswitch-users] WebRTC STUN request for a SIP profile is not
 working
Message-ID: <de53c6fc0191427f86ee99e25404d3b7@c4b.de>

Hi,

if I set the parameter „<X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=stun:stun.freeswitch.org"/> “ in the vars.xml and use this variable $${external_rtp_ip}  in the SIP Profile, then the Ext-RTP-IP show the real external IP.
I think, that’s OK.

But I miss the srflx candidate in the sdp.

I tried to experimental with the apply-candidate-acl (rfc1918.auto, any_v4.auto, wan_v4.auto), but nothing works.
What’s wrong?

Here is my sofia profile config:
<profile xmlns:xsd="http://www.w3.org/2001/XMLSchema" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" name="SBC2_B2Bua">
  <aliases />
  <domains>
    <domain name="h3k.sip.c4b" alias="false" parse="true" />
  </domains>
  <settings>
    <param name="user-agent-string" value="XPhone Call Controller" />
    <param name="debug" value="0" />
    <param name="stun-enabled" value="true"/>
    <param name="stun-auto-disable" value="false"/>
    <param name="sip-trace" value="no" />
    <param name="sip-capture" value="no" />
    <param name="dialplan" value="XML" />
    <param name="context" value="SBC2_SP_outbound" />
    <param name="dtmf-type" value="rfc2833" />
    <param name="dtmf-duration" value="100" />
    <param name="rfc2833-pt" value="98" />
    <param name="inbound-codec-prefs" value="OPUS,PCMU,PCMA,VP8" />
    <param name="outbound-codec-prefs" value="OPUS,PCMU,PCMA,VP8" />
    <param name="hold-music" value="local_stream://moh" />
    <param name="rtp-timer-name" value="none" />
    <param name="rtp-rewrite-timestamps" value="true" />
    <param name="manage-presence" value="false" />
    <param name="inbound-codec-negotiation" value="generous" />
    <param name="inbound-late-negotiation" value="true" />
    <param name="nonce-ttl" value="60" />
    <param name="auth-calls" value="false" />
    <param name="sip-port" value="4901" />
    <param name="rtp-ip" value="MyLocalIP" />
    <param name="sip-ip" value=" MyLocalIP " />
    <param name="ext-rtp-ip" value="$${external_rtp_ip}" />
    <param name="ext-sip-ip" value=" MyLocalIP " />
    <param name="local-network-acl" value="SBC2_localnet.auto" />
    <param name="apply-candidate-acl" value="rfc1918.auto" />
    <param name="apply-candidate-acl" value="SBC2_localnet.auto" />
    <param name="apply-candidate-acl" value="any_v4.auto" />
    <param name="apply-candidate-acl" value="wan_v4.auto" />
    <param name="apply-inbound-acl" value="SBC2_localnet.auto" />
    <param name="rtp-timeout-sec" value="300" />
    <param name="rtp-hold-timeout-sec" value="1800" />
    <param name="enable-3pcc" value="true" />
    <param name="inbound-use-callid-as-uuid" value="true" />
    <param name="tls" value="false" />
    <param name="tls-only" value="false" />
    <param name="tls-bind-params" value="transport=tls" />
    <param name="tls-sip-port" value="" />
    <param name="tls-passphrase" value="" />
    <param name="tls-verify-date" value="true" />
    <param name="tls-verify-depth" value="2" />
    <param name="tls-verify-in-subjects" value="" />
    <param name="tls-version" value="tlsv1,tlsv1.1,tlsv1.2" />
    <param name="tls-ciphers" value="ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH" />
    <param name="zrtp-passthru" value="true" />
    <param name="inbound-reg-force-matching-username" value="true" />
    <param name="auth-all-packets" value="false" />
    <param name="wss-binding" value=":4447" />
    <param name="caller-id-type" value="rpid" />
    <param name="manual-redirect" value="true" />
    <param name="rtcp-audio-interval-msec" value="5000" />
  </settings>
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/GW_SBC2_B2Bua.conf.xml" />
  </gateways>
</profile>

Von: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> Im Auftrag von Alexander Haugg
Gesendet: Donnerstag, 9. Januar 2020 08:17
An: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Betreff: [Freeswitch-users] WebRTC STUN request for a SIP profile is not working

Hi,

I’d read the manuals and configure my sofia profile to work with the external IP in the ext-rtp-ip setting.
<param name="stun-enabled" value="true"/>
<param name="stun-auto-disable" value="false"/>
<param name="ext-rtp-ip" value="stun:stun.freeswitch.org" />

The Freeswitch is started with the parameter –nonat
In the pcap trace is no STUN request visible.
I try to restart the Freeswitch, restart the profile, do an outbound call, but there is no STUN request.

If I try to get my external IP via the CLI cmd “stun stun.freeswitch.org”, then works it successfully.

The status output of the profile is:
=================================================================================================
Name                                SBC2_B2Bua
Domain Name                 N/A
Auto-NAT                         false
DBName                           sofia_reg_SBC2_B2Bua
Pres Hosts
Dialplan                            XML
Context                            SBC2_SP_outbound
Challenge Realm             auto_to
RTP-IP                               MY_LOCAL_IP
Ext-RTP-IP                        stun:stun.freeswitch.org
SIP-IP                                MY_LOCAL_IP
Ext-SIP-IP                         MY_LOCAL_IP
URL                                   sip:mod_sofia at MY_LOCAL_IP:4901
BIND-URL                  sip:mod_sofia at MY_LOCAL_IP:4901;maddr=MY_LOCAL_IP;transport=udp,tcp
HOLD-MUSIC                   local_stream://moh
OUTBOUND-PROXY       N/A
CODECS IN                       OPUS,PCMU,PCMA,VP8
CODECS OUT                   OPUS,PCMU,PCMA,VP8
TEL-EVENT                       98
DTMF-MODE                   rfc2833
CNG                                  13
SESSION-TO                     0
MAX-DIALOG                   0
NOMEDIA                        false
LATE-NEG                         true
PROXY-MEDIA                 false
ZRTP-PASSTHRU             false
AGGRESSIVENAT           false
CALLS-IN                           0
FAILED-CALLS-IN             0
CALLS-OUT                       0
FAILED-CALLS-OUT        0
REGISTRATIONS              0

Thanks a lot!!!
Alex


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From david.villasmil.work at gmail.com  Fri Jan 10 12:59:10 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 10 Jan 2020 12:59:10 +0000
Subject: [Freeswitch-users] WebRTC STUN request for a SIP profile is not
	working
In-Reply-To: <de53c6fc0191427f86ee99e25404d3b7@c4b.de>
References: <de53c6fc0191427f86ee99e25404d3b7@c4b.de>
Message-ID: <CAFGRPVrpJJkUsf53bXodr6ER_ehhOONoYsebPRbKkZVfdTK-4g@mail.gmail.com>

Does it work with a manually entered IP?

On Fri, 10 Jan 2020 at 12:42, Alexander Haugg <Alexander.Haugg at c4b.de>
wrote:

> Hi,
>
>
>
> if I set the parameter „<X-PRE-PROCESS cmd="stun-set"
> data="external_rtp_ip=stun:stun.freeswitch.org"/> “ in the vars.xml and
> use this variable $${external_rtp_ip}  in the SIP Profile, then the
> Ext-RTP-IP show the real external IP.
>
> I think, that’s OK.
>
>
>
> But I miss the srflx candidate in the sdp.
>
>
>
> I tried to experimental with the apply-candidate-acl (rfc1918.auto,
> any_v4.auto, wan_v4.auto), but nothing works.
>
> What’s wrong?
>
>
>
> Here is my sofia profile config:
>
> <profile xmlns:xsd="http://www.w3.org/2001/XMLSchema" xmlns:xsi="
> http://www.w3.org/2001/XMLSchema-instance" name="SBC2_B2Bua">
>
>   <aliases />
>
>   <domains>
>
>     <domain name="h3k.sip.c4b" alias="false" parse="true" />
>
>   </domains>
>
>   <settings>
>
>     <param name="user-agent-string" value="XPhone Call Controller" />
>
>     <param name="debug" value="0" />
>
> *    <param name="stun-enabled" value="true"/>*
>
> *    <param name="stun-auto-disable" value="false"/>*
>
>     <param name="sip-trace" value="no" />
>
>     <param name="sip-capture" value="no" />
>
>     <param name="dialplan" value="XML" />
>
>     <param name="context" value="SBC2_SP_outbound" />
>
>     <param name="dtmf-type" value="rfc2833" />
>
>     <param name="dtmf-duration" value="100" />
>
>     <param name="rfc2833-pt" value="98" />
>
>     <param name="inbound-codec-prefs" value="OPUS,PCMU,PCMA,VP8" />
>
>     <param name="outbound-codec-prefs" value="OPUS,PCMU,PCMA,VP8" />
>
>     <param name="hold-music" value="local_stream://moh" />
>
>     <param name="rtp-timer-name" value="none" />
>
>     <param name="rtp-rewrite-timestamps" value="true" />
>
>     <param name="manage-presence" value="false" />
>
>     <param name="inbound-codec-negotiation" value="generous" />
>
>     <param name="inbound-late-negotiation" value="true" />
>
>     <param name="nonce-ttl" value="60" />
>
>     <param name="auth-calls" value="false" />
>
>     <param name="sip-port" value="4901" />
>
>     <param name="rtp-ip" value="MyLocalIP" />
>
>     <param name="sip-ip" value=" MyLocalIP " />
>
>     *<param name="ext-rtp-ip" value="$${external_rtp_ip}" />*
>
>     <param name="ext-sip-ip" value=" MyLocalIP " />
>
>     <param name="local-network-acl" value="SBC2_localnet.auto" />
>
> *    <param name="apply-candidate-acl" value="rfc1918.auto" />*
>
> *    <param name="apply-candidate-acl" value="SBC2_localnet.auto" />*
>
> *    <param name="apply-candidate-acl" value="any_v4.auto" />*
>
> *    <param name="apply-candidate-acl" value="wan_v4.auto" />*
>
>     <param name="apply-inbound-acl" value="SBC2_localnet.auto" />
>
>     <param name="rtp-timeout-sec" value="300" />
>
>     <param name="rtp-hold-timeout-sec" value="1800" />
>
>     <param name="enable-3pcc" value="true" />
>
>     <param name="inbound-use-callid-as-uuid" value="true" />
>
>     <param name="tls" value="false" />
>
>     <param name="tls-only" value="false" />
>
>     <param name="tls-bind-params" value="transport=tls" />
>
>     <param name="tls-sip-port" value="" />
>
>     <param name="tls-passphrase" value="" />
>
>     <param name="tls-verify-date" value="true" />
>
>     <param name="tls-verify-depth" value="2" />
>
>     <param name="tls-verify-in-subjects" value="" />
>
>     <param name="tls-version" value="tlsv1,tlsv1.1,tlsv1.2" />
>
>     <param name="tls-ciphers" value="ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH" />
>
>     <param name="zrtp-passthru" value="true" />
>
>     <param name="inbound-reg-force-matching-username" value="true" />
>
>     <param name="auth-all-packets" value="false" />
>
>     <param name="wss-binding" value=":4447" />
>
>     <param name="caller-id-type" value="rpid" />
>
>     <param name="manual-redirect" value="true" />
>
>     <param name="rtcp-audio-interval-msec" value="5000" />
>
>   </settings>
>
>   <gateways>
>
>     <X-PRE-PROCESS cmd="include" data="external/GW_SBC2_B2Bua.conf.xml" />
>
>   </gateways>
>
> </profile>
>
>
>
> *Von:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *Im
> Auftrag von *Alexander Haugg
> *Gesendet:* Donnerstag, 9. Januar 2020 08:17
> *An:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Betreff:* [Freeswitch-users] WebRTC STUN request for a SIP profile is
> not working
>
>
>
> Hi,
>
>
>
> I’d read the manuals and configure my sofia profile to work with the
> external IP in the ext-rtp-ip setting.
>
> <param name="stun-enabled" value="true"/>
>
> <param name="stun-auto-disable" value="false"/>
>
> <param name="ext-rtp-ip" value="stun:stun.freeswitch.org" />
>
>
>
> The Freeswitch is started with the parameter –nonat
>
> In the pcap trace is no STUN request visible.
>
> I try to restart the Freeswitch, restart the profile, do an outbound call,
> but there is no STUN request.
>
>
>
> If I try to get my external IP via the CLI cmd “stun stun.freeswitch.org”,
> then works it successfully.
>
>
>
> The status output of the profile is:
>
>
> =================================================================================================
>
> Name                                SBC2_B2Bua
>
> Domain Name                 N/A
>
> Auto-NAT                         false
>
> DBName                           sofia_reg_SBC2_B2Bua
>
> Pres Hosts
>
> Dialplan                            XML
>
> Context                            SBC2_SP_outbound
>
> Challenge Realm             auto_to
>
> RTP-IP                               MY_LOCAL_IP
>
> Ext-RTP-IP                        stun:stun.freeswitch.org
>
> SIP-IP                                MY_LOCAL_IP
>
> Ext-SIP-IP                         MY_LOCAL_IP
>
> URL                                   sip:mod_sofia at MY_LOCAL_IP:4901
>
> BIND-URL
> sip:mod_sofia at MY_LOCAL_IP:4901;maddr=MY_LOCAL_IP;transport=udp,tcp
>
> HOLD-MUSIC                   local_stream://moh
>
> OUTBOUND-PROXY       N/A
>
> CODECS IN                       OPUS,PCMU,PCMA,VP8
>
> CODECS OUT                   OPUS,PCMU,PCMA,VP8
>
> TEL-EVENT                       98
>
> DTMF-MODE                   rfc2833
>
> CNG                                  13
>
> SESSION-TO                     0
>
> MAX-DIALOG                   0
>
> NOMEDIA                        false
>
> LATE-NEG                         true
>
> PROXY-MEDIA                 false
>
> ZRTP-PASSTHRU             false
>
> AGGRESSIVENAT           false
>
> CALLS-IN                           0
>
> FAILED-CALLS-IN             0
>
> CALLS-OUT                       0
>
> FAILED-CALLS-OUT        0
>
> REGISTRATIONS              0
>
>
>
> Thanks a lot!!!
>
> Alex
>
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From Alexander.Haugg at c4b.de  Fri Jan 10 13:45:21 2020
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Fri, 10 Jan 2020 13:45:21 +0000
Subject: [Freeswitch-users] WebRTC STUN request for a SIP profile is
	not	working
In-Reply-To: <CAFGRPVrpJJkUsf53bXodr6ER_ehhOONoYsebPRbKkZVfdTK-4g@mail.gmail.com>
References: <de53c6fc0191427f86ee99e25404d3b7@c4b.de>
 <CAFGRPVrpJJkUsf53bXodr6ER_ehhOONoYsebPRbKkZVfdTK-4g@mail.gmail.com>
Message-ID: <489523a3fcb34915abd8b6e87406bf09@c4b.de>

OK,
I have the solution.

1.       My think was to see the external candidate in the SDP. Freeswitch do that automatically, if the Client Registration from outside the local NW.

2.       To force adding the external candidate, there is the channel var “include_external_ip=true”. For example -> originate {include_external_ip=true,media_webrtc=true}sofia/gateway/GW_SBC2_B2Bua/22100 &park

3.       The srflx candidate will be set, if the internal port different from the external port (stun request). See the switch_core_media implementation for the candidate generation.

My questions are solved (at the moment ;-). My biggest problem was to understand the server stack vs. client stack behaviour for WebRTC.

Thanks a lot!
Alex

Von: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> Im Auftrag von David Villasmil
Gesendet: Freitag, 10. Januar 2020 13:59
An: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Betreff: Re: [Freeswitch-users] WebRTC STUN request for a SIP profile is not working

Does it work with a manually entered IP?

On Fri, 10 Jan 2020 at 12:42, Alexander Haugg <Alexander.Haugg at c4b.de<mailto:Alexander.Haugg at c4b.de>> wrote:
Hi,

if I set the parameter „<X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=stun:stun.freeswitch.org<http://stun.freeswitch.org>"/> “ in the vars.xml and use this variable $${external_rtp_ip}  in the SIP Profile, then the Ext-RTP-IP show the real external IP.
I think, that’s OK.

But I miss the srflx candidate in the sdp.

I tried to experimental with the apply-candidate-acl (rfc1918.auto, any_v4.auto, wan_v4.auto), but nothing works.
What’s wrong?

Here is my sofia profile config:
<profile xmlns:xsd="http://www.w3.org/2001/XMLSchema" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" name="SBC2_B2Bua">
  <aliases />
  <domains>
    <domain name="h3k.sip.c4b" alias="false" parse="true" />
  </domains>
  <settings>
    <param name="user-agent-string" value="XPhone Call Controller" />
    <param name="debug" value="0" />
    <param name="stun-enabled" value="true"/>
    <param name="stun-auto-disable" value="false"/>
    <param name="sip-trace" value="no" />
    <param name="sip-capture" value="no" />
    <param name="dialplan" value="XML" />
    <param name="context" value="SBC2_SP_outbound" />
    <param name="dtmf-type" value="rfc2833" />
    <param name="dtmf-duration" value="100" />
    <param name="rfc2833-pt" value="98" />
    <param name="inbound-codec-prefs" value="OPUS,PCMU,PCMA,VP8" />
    <param name="outbound-codec-prefs" value="OPUS,PCMU,PCMA,VP8" />
    <param name="hold-music" value="local_stream://moh" />
    <param name="rtp-timer-name" value="none" />
    <param name="rtp-rewrite-timestamps" value="true" />
    <param name="manage-presence" value="false" />
    <param name="inbound-codec-negotiation" value="generous" />
    <param name="inbound-late-negotiation" value="true" />
    <param name="nonce-ttl" value="60" />
    <param name="auth-calls" value="false" />
    <param name="sip-port" value="4901" />
    <param name="rtp-ip" value="MyLocalIP" />
    <param name="sip-ip" value=" MyLocalIP " />
    <param name="ext-rtp-ip" value="$${external_rtp_ip}" />
    <param name="ext-sip-ip" value=" MyLocalIP " />
    <param name="local-network-acl" value="SBC2_localnet.auto" />
    <param name="apply-candidate-acl" value="rfc1918.auto" />
    <param name="apply-candidate-acl" value="SBC2_localnet.auto" />
    <param name="apply-candidate-acl" value="any_v4.auto" />
    <param name="apply-candidate-acl" value="wan_v4.auto" />
    <param name="apply-inbound-acl" value="SBC2_localnet.auto" />
    <param name="rtp-timeout-sec" value="300" />
    <param name="rtp-hold-timeout-sec" value="1800" />
    <param name="enable-3pcc" value="true" />
    <param name="inbound-use-callid-as-uuid" value="true" />
    <param name="tls" value="false" />
    <param name="tls-only" value="false" />
    <param name="tls-bind-params" value="transport=tls" />
    <param name="tls-sip-port" value="" />
    <param name="tls-passphrase" value="" />
    <param name="tls-verify-date" value="true" />
    <param name="tls-verify-depth" value="2" />
    <param name="tls-verify-in-subjects" value="" />
    <param name="tls-version" value="tlsv1,tlsv1.1,tlsv1.2" />
    <param name="tls-ciphers" value="ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH" />
    <param name="zrtp-passthru" value="true" />
    <param name="inbound-reg-force-matching-username" value="true" />
    <param name="auth-all-packets" value="false" />
    <param name="wss-binding" value=":4447" />
    <param name="caller-id-type" value="rpid" />
    <param name="manual-redirect" value="true" />
    <param name="rtcp-audio-interval-msec" value="5000" />
  </settings>
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/GW_SBC2_B2Bua.conf.xml" />
  </gateways>
</profile>

Von: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> Im Auftrag von Alexander Haugg
Gesendet: Donnerstag, 9. Januar 2020 08:17
An: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Betreff: [Freeswitch-users] WebRTC STUN request for a SIP profile is not working

Hi,

I’d read the manuals and configure my sofia profile to work with the external IP in the ext-rtp-ip setting.
<param name="stun-enabled" value="true"/>
<param name="stun-auto-disable" value="false"/>
<param name="ext-rtp-ip" value="stun:stun.freeswitch.org<http://stun.freeswitch.org>" />

The Freeswitch is started with the parameter –nonat
In the pcap trace is no STUN request visible.
I try to restart the Freeswitch, restart the profile, do an outbound call, but there is no STUN request.

If I try to get my external IP via the CLI cmd “stun stun.freeswitch.org<http://stun.freeswitch.org>”, then works it successfully.

The status output of the profile is:
=================================================================================================
Name                                SBC2_B2Bua
Domain Name                 N/A
Auto-NAT                         false
DBName                           sofia_reg_SBC2_B2Bua
Pres Hosts
Dialplan                            XML
Context                            SBC2_SP_outbound
Challenge Realm             auto_to
RTP-IP                               MY_LOCAL_IP
Ext-RTP-IP                        stun:stun.freeswitch.org<http://stun.freeswitch.org>
SIP-IP                                MY_LOCAL_IP
Ext-SIP-IP                         MY_LOCAL_IP
URL                                   sip:mod_sofia at MY_LOCAL_IP:4901
BIND-URL                  sip:mod_sofia at MY_LOCAL_IP:4901;maddr=MY_LOCAL_IP;transport=udp,tcp
HOLD-MUSIC                   local_stream://moh
OUTBOUND-PROXY       N/A
CODECS IN                       OPUS,PCMU,PCMA,VP8
CODECS OUT                   OPUS,PCMU,PCMA,VP8
TEL-EVENT                       98
DTMF-MODE                   rfc2833
CNG                                  13
SESSION-TO                     0
MAX-DIALOG                   0
NOMEDIA                        false
LATE-NEG                         true
PROXY-MEDIA                 false
ZRTP-PASSTHRU             false
AGGRESSIVENAT           false
CALLS-IN                           0
FAILED-CALLS-IN             0
CALLS-OUT                       0
FAILED-CALLS-OUT        0
REGISTRATIONS              0

Thanks a lot!!!
Alex


_________________________________________________________________________

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

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

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

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

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

David Villasmil
email: david.villasmil.work at gmail.com<mailto:david.villasmil.work at gmail.com>
phone: +34669448337
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From th982a at googlemail.com  Sat Jan 11 21:21:56 2020
From: th982a at googlemail.com (Tamer Higazi)
Date: Sat, 11 Jan 2020 22:21:56 +0100
Subject: portaudio and pulsaudio works on FS, but INCOPATIBLE DESTINATION
Message-ID: <36fb0991-947d-2f93-bf2a-f856d2aab302@googlemail.com>

Hi people,

After long try and there I got "finally" portaudio running on my cpu.

I got the dumb error message "connection refused" rid of me, and FS 
seems to access the soundcard.
I can choose the devices of my choice, and place a call.

But when I place a call, I still get this:

2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:935 
(sofia/external/123456) State REPORTING
2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:174 
sofia/external/123456 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION
2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:935 
(sofia/external/123456) State REPORTING going to sleep
2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:610 
(sofia/external/123456) State Change CS_REPORTING -> CS_DESTROY
2020-01-11 22:18:33.847800 [DEBUG] switch_core_session.c:1715 Session 17 
(sofia/external/123456) Locked, Waiting on external entities
2020-01-11 22:18:33.847800 [DEBUG] switch_ivr_originate.c:3941 Originate 
Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
2020-01-11 22:18:33.847800 [NOTICE] switch_core_session.c:1733 Session 
17 (sofia/external/123456) Ended
2020-01-11 22:18:33.847800 [NOTICE] switch_core_session.c:1737 Close 
Channel sofia/external/123456 [CS_DESTROY]
2020-01-11 22:18:33.847800 [DEBUG] switch_channel.c:2047 
(portaudio/1123456) Callstate Change RING_WAIT -> ACTIVE
2020-01-11 22:18:33.847800 [INFO] mod_dptools.c:3518 Originate Failed.  
Cause: INCOMPATIBLE_DESTINATION
2020-01-11 22:18:33.847800 [NOTICE] switch_channel.c:4857 Hangup 
portaudio/1123456 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:738 
(sofia/external/123456) Running State Change CS_DESTROY (Cur 1 Tot 17)

Any ideas why this is the case ?


best, Tamer



From david.villasmil.work at gmail.com  Sun Jan 12 00:12:41 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sun, 12 Jan 2020 00:12:41 +0000
Subject: [Freeswitch-users] portaudio and pulsaudio works on FS,
 but INCOPATIBLE DESTINATION
In-Reply-To: <mailman.912.1578777754.1226.freeswitch-users@lists.freeswitch.org>
References: <mailman.912.1578777754.1226.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAFGRPVqVZqT+pktvCFJAUQp056bB9Fzr7W4x+gHvAioQThcuUw@mail.gmail.com>

That usually comes up when the far side and fs can’t agree on a codec. Make
sure both sides have the same set of codecs enabled.

On Sat, 11 Jan 2020 at 21:37, Tamer Higazi via FreeSWITCH-users <
freeswitch-users at lists.freeswitch.org> wrote:

>
>
>
> ---------- Forwarded message ----------
> From: Tamer Higazi <th982a at googlemail.com>
> To: freeswitch-users at lists.freeswitch.org
> Cc:
> Bcc:
> Date: Sat, 11 Jan 2020 22:21:56 +0100
> Subject: portaudio and pulsaudio works on FS, but INCOPATIBLE DESTINATION
> Hi people,
>
> After long try and there I got "finally" portaudio running on my cpu.
>
> I got the dumb error message "connection refused" rid of me, and FS
> seems to access the soundcard.
> I can choose the devices of my choice, and place a call.
>
> But when I place a call, I still get this:
>
> 2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:935
> (sofia/external/123456) State REPORTING
> 2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:174
> sofia/external/123456 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION
> 2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:935
> (sofia/external/123456) State REPORTING going to sleep
> 2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:610
> (sofia/external/123456) State Change CS_REPORTING -> CS_DESTROY
> 2020-01-11 22:18:33.847800 [DEBUG] switch_core_session.c:1715 Session 17
> (sofia/external/123456) Locked, Waiting on external entities
> 2020-01-11 22:18:33.847800 [DEBUG] switch_ivr_originate.c:3941 Originate
> Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
> 2020-01-11 22:18:33.847800 [NOTICE] switch_core_session.c:1733 Session
> 17 (sofia/external/123456) Ended
> 2020-01-11 22:18:33.847800 [NOTICE] switch_core_session.c:1737 Close
> Channel sofia/external/123456 [CS_DESTROY]
> 2020-01-11 22:18:33.847800 [DEBUG] switch_channel.c:2047
> (portaudio/1123456) Callstate Change RING_WAIT -> ACTIVE
> 2020-01-11 22:18:33.847800 [INFO] mod_dptools.c:3518 Originate Failed.
> Cause: INCOMPATIBLE_DESTINATION
> 2020-01-11 22:18:33.847800 [NOTICE] switch_channel.c:4857 Hangup
> portaudio/1123456 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
> 2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:738
> (sofia/external/123456) Running State Change CS_DESTROY (Cur 1 Tot 17)
>
> Any ideas why this is the case ?
>
>
> best, Tamer
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Tamer Higazi via FreeSWITCH-users <
> freeswitch-users at lists.freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Cc:
> Bcc:
> Date: Sat, 11 Jan 2020 13:37:37 -0800 (PST)
> Subject: [Freeswitch-users] portaudio and pulsaudio works on FS, but
> INCOPATIBLE DESTINATION
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From th982a at googlemail.com  Sun Jan 12 00:16:57 2020
From: th982a at googlemail.com (Tamer Higazi)
Date: Sun, 12 Jan 2020 01:16:57 +0100
Subject: [Freeswitch-users] portaudio and pulsaudio works on FS, but
 INCOPATIBLE DESTINATION
In-Reply-To: <CAFGRPVqVZqT+pktvCFJAUQp056bB9Fzr7W4x+gHvAioQThcuUw@mail.gmail.com>
References: <mailman.912.1578777754.1226.freeswitch-users@lists.freeswitch.org>
 <CAFGRPVqVZqT+pktvCFJAUQp056bB9Fzr7W4x+gHvAioQThcuUw@mail.gmail.com>
Message-ID: <8fa48b57-8a55-17e3-282a-ff84cc321026@googlemail.com>

Dear David,

This is what I thought as well (just shortly).

with the SIP phone dialing it bridges through the gateway, it works 
without any problems.

But with mod_portaudio or mod_alsa I have this problem.

Any ideas how to fix it or to get more informations or how to fix it ?

Because in the FS pages it's written nowhere what codecs should be used 
or how to configure it ....


best, Tamer

On 2020-01-12 01:12, David Villasmil wrote:
>
> That usually comes up when the far side and fs can’t agree on a codec. 
> Make sure both sides have the same set of codecs enabled.
>
> On Sat, 11 Jan 2020 at 21:37, Tamer Higazi via FreeSWITCH-users 
> <freeswitch-users at lists.freeswitch.org 
> <mailto:freeswitch-users at lists.freeswitch.org>> wrote:
>
>
>
>
>     ---------- Forwarded message ----------
>     From: Tamer Higazi <th982a at googlemail.com
>     <mailto:th982a at googlemail.com>>
>     To: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     Cc:
>     Bcc:
>     Date: Sat, 11 Jan 2020 22:21:56 +0100
>     Subject: portaudio and pulsaudio works on FS, but INCOPATIBLE
>     DESTINATION
>     Hi people,
>
>     After long try and there I got "finally" portaudio running on my cpu.
>
>     I got the dumb error message "connection refused" rid of me, and FS
>     seems to access the soundcard.
>     I can choose the devices of my choice, and place a call.
>
>     But when I place a call, I still get this:
>
>     2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:935
>     (sofia/external/123456) State REPORTING
>     2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:174
>     sofia/external/123456 Standard REPORTING, cause:
>     INCOMPATIBLE_DESTINATION
>     2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:935
>     (sofia/external/123456) State REPORTING going to sleep
>     2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:610
>     (sofia/external/123456) State Change CS_REPORTING -> CS_DESTROY
>     2020-01-11 22:18:33.847800 [DEBUG] switch_core_session.c:1715
>     Session 17
>     (sofia/external/123456) Locked, Waiting on external entities
>     2020-01-11 22:18:33.847800 [DEBUG] switch_ivr_originate.c:3941
>     Originate
>     Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
>     2020-01-11 22:18:33.847800 [NOTICE] switch_core_session.c:1733
>     Session
>     17 (sofia/external/123456) Ended
>     2020-01-11 22:18:33.847800 [NOTICE] switch_core_session.c:1737 Close
>     Channel sofia/external/123456 [CS_DESTROY]
>     2020-01-11 22:18:33.847800 [DEBUG] switch_channel.c:2047
>     (portaudio/1123456) Callstate Change RING_WAIT -> ACTIVE
>     2020-01-11 22:18:33.847800 [INFO] mod_dptools.c:3518 Originate
>     Failed.
>     Cause: INCOMPATIBLE_DESTINATION
>     2020-01-11 22:18:33.847800 [NOTICE] switch_channel.c:4857 Hangup
>     portaudio/1123456 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
>     2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:738
>     (sofia/external/123456) Running State Change CS_DESTROY (Cur 1 Tot 17)
>
>     Any ideas why this is the case ?
>
>
>     best, Tamer
>
>
>
>
>
>     ---------- Forwarded message ----------
>     From: Tamer Higazi via FreeSWITCH-users
>     <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     To: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     Cc:
>     Bcc:
>     Date: Sat, 11 Jan 2020 13:37:37 -0800 (PST)
>     Subject: [Freeswitch-users] portaudio and pulsaudio works on FS,
>     but INCOPATIBLE DESTINATION
>     _________________________________________________________________________
>
>     The FreeSWITCH project is sponsored by SignalWire
>     https://signalwire.com
>     Enhance your FreeSWITCH install with disruptive priced SMS and
>     PSTN services.
>     Build your next product on our scalable cloud platform.
>
>     Join our online community to chat in real time
>     https://signalwire.community
>
>     Professional FreeSWITCH Services
>     sales at freeswitch.com <mailto:sales at freeswitch.com>
>     https://freeswitch.com
>
>     Official FreeSWITCH Sites
>     https://freeswitch.com/oss
>     https://freeswitch.org/confluence
>     https://cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     https://freeswitch.com
>
> -- 
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com 
> <mailto:david.villasmil.work at gmail.com>
> phone: +34669448337


From covici at ccs.covici.com  Sun Jan 12 07:06:14 2020
From: covici at ccs.covici.com (John Covici)
Date: Sun, 12 Jan 2020 02:06:14 -0500
Subject: [Freeswitch-users] portaudio and pulsaudio works on FS,
	but INCOPATIBLE DESTINATION
In-Reply-To: <mailman.913.1578788251.1226.freeswitch-users@lists.freeswitch.org>
References: <mailman.912.1578777754.1226.freeswitch-users@lists.freeswitch.org>
 <CAFGRPVqVZqT+pktvCFJAUQp056bB9Fzr7W4x+gHvAioQThcuUw@mail.gmail.com>
 <mailman.913.1578788251.1226.freeswitch-users@lists.freeswitch.org>
Message-ID: <m34kx1taqh.wl-covici@ccs.covici.com>

How about <action application="export"
data="media_mix_inbound_outbound_codecs=true"/>  See if that works.

On Sat, 11 Jan 2020 19:52:54 -0500,
Tamer Higazi via FreeSWITCH-users wrote:
> 
> [1  <message/rfc822 (7bit)>]
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on FS, but INCOPATIBLE DESTINATION
> From: Tamer Higazi <th982a at googlemail.com>
> Date: Sun, 12 Jan 2020 01:16:57 +0100
> 
> Dear David,
> 
> This is what I thought as well (just shortly).
> 
> with the SIP phone dialing it bridges through the gateway, it
> works without any problems.
> 
> But with mod_portaudio or mod_alsa I have this problem.
> 
> Any ideas how to fix it or to get more informations or how to fix it ?
> 
> Because in the FS pages it's written nowhere what codecs should
> be used or how to configure it ....
> 
> 
> best, Tamer
> 
> On 2020-01-12 01:12, David Villasmil wrote:
> > 
> > That usually comes up when the far side and fs can’t agree on a
> > codec. Make sure both sides have the same set of codecs
> > enabled.
> > 
> > On Sat, 11 Jan 2020 at 21:37, Tamer Higazi via FreeSWITCH-users
> > <freeswitch-users at lists.freeswitch.org
> > <mailto:freeswitch-users at lists.freeswitch.org>> wrote:
> > 
> > 
> > 
> > 
> >     ---------- Forwarded message ----------
> >     From: Tamer Higazi <th982a at googlemail.com
> >     <mailto:th982a at googlemail.com>>
> >     To: freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>
> >     Cc:
> >     Bcc:
> >     Date: Sat, 11 Jan 2020 22:21:56 +0100
> >     Subject: portaudio and pulsaudio works on FS, but INCOPATIBLE
> >     DESTINATION
> >     Hi people,
> > 
> >     After long try and there I got "finally" portaudio running on my cpu.
> > 
> >     I got the dumb error message "connection refused" rid of me, and FS
> >     seems to access the soundcard.
> >     I can choose the devices of my choice, and place a call.
> > 
> >     But when I place a call, I still get this:
> > 
> >     2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:935
> >     (sofia/external/123456) State REPORTING
> >     2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:174
> >     sofia/external/123456 Standard REPORTING, cause:
> >     INCOMPATIBLE_DESTINATION
> >     2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:935
> >     (sofia/external/123456) State REPORTING going to sleep
> >     2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:610
> >     (sofia/external/123456) State Change CS_REPORTING -> CS_DESTROY
> >     2020-01-11 22:18:33.847800 [DEBUG] switch_core_session.c:1715
> >     Session 17
> >     (sofia/external/123456) Locked, Waiting on external entities
> >     2020-01-11 22:18:33.847800 [DEBUG] switch_ivr_originate.c:3941
> >     Originate
> >     Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
> >     2020-01-11 22:18:33.847800 [NOTICE] switch_core_session.c:1733
> >     Session
> >     17 (sofia/external/123456) Ended
> >     2020-01-11 22:18:33.847800 [NOTICE] switch_core_session.c:1737 Close
> >     Channel sofia/external/123456 [CS_DESTROY]
> >     2020-01-11 22:18:33.847800 [DEBUG] switch_channel.c:2047
> >     (portaudio/1123456) Callstate Change RING_WAIT -> ACTIVE
> >     2020-01-11 22:18:33.847800 [INFO] mod_dptools.c:3518 Originate
> >     Failed.
> >     Cause: INCOMPATIBLE_DESTINATION
> >     2020-01-11 22:18:33.847800 [NOTICE] switch_channel.c:4857 Hangup
> >     portaudio/1123456 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
> >     2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:738
> >     (sofia/external/123456) Running State Change CS_DESTROY (Cur 1 Tot 17)
> > 
> >     Any ideas why this is the case ?
> > 
> > 
> >     best, Tamer
> > 
> > 
> > 
> > 
> > 
> >     ---------- Forwarded message ----------
> >     From: Tamer Higazi via FreeSWITCH-users
> >     <freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>>
> >     To: freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>
> >     Cc:
> >     Bcc:
> >     Date: Sat, 11 Jan 2020 13:37:37 -0800 (PST)
> >     Subject: [Freeswitch-users] portaudio and pulsaudio works on FS,
> >     but INCOPATIBLE DESTINATION
> >     _________________________________________________________________________
> > 
> >     The FreeSWITCH project is sponsored by SignalWire
> >     https://signalwire.com
> >     Enhance your FreeSWITCH install with disruptive priced SMS and
> >     PSTN services.
> >     Build your next product on our scalable cloud platform.
> > 
> >     Join our online community to chat in real time
> >     https://signalwire.community
> > 
> >     Professional FreeSWITCH Services
> >     sales at freeswitch.com <mailto:sales at freeswitch.com>
> >     https://freeswitch.com
> > 
> >     Official FreeSWITCH Sites
> >     https://freeswitch.com/oss
> >     https://freeswitch.org/confluence
> >     https://cluecon.com
> > 
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     https://freeswitch.com
> > 
> > -- 
> > Regards,
> > 
> > David Villasmil
> > email: david.villasmil.work at gmail.com
> > <mailto:david.villasmil.work at gmail.com>
> > phone: +34669448337
> 
> [2  <text/plain; utf-8 (base64)>]
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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

         John Covici wb2una
         covici at ccs.covici.com


From avi at avimarcus.net  Sun Jan 12 13:06:28 2020
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 12 Jan 2020 13:06:28 +0000
Subject: [Freeswitch-users] Calls repeat at 80 seconds?
Message-ID: <0100016f99dce3c1-51f0a57f-e4f7-4c52-85e7-fc08c20a713f-000000@email.amazonses.com>

FreeSWITCH Version 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit)

Is anyone having an issue with calls suddenly "restarting" at about 80
seconds? I thought it was 1 carrier on 1 route, but I'm getting reports now
across various destinations using different carriers. It's not on every
call, just some.

I've upgraded to FreeSWITCH Version 1.10.2-release-13-f7bdd3845a~64bit
(-release-13-f7bdd3845a 64bit) and we'll see if I still get the issue.



Thanks,
-Avi Marcus
-------------- next part --------------
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From th982a at googlemail.com  Sun Jan 12 13:49:46 2020
From: th982a at googlemail.com (Tamer Higazi)
Date: Sun, 12 Jan 2020 14:49:46 +0100
Subject: [Freeswitch-users] portaudio and pulsaudio works on FS, but
 INCOPATIBLE DESTINATION (SOLVED)
In-Reply-To: <m34kx1taqh.wl-covici@ccs.covici.com>
References: <mailman.912.1578777754.1226.freeswitch-users@lists.freeswitch.org>
 <CAFGRPVqVZqT+pktvCFJAUQp056bB9Fzr7W4x+gHvAioQThcuUw@mail.gmail.com>
 <mailman.913.1578788251.1226.freeswitch-users@lists.freeswitch.org>
 <m34kx1taqh.wl-covici@ccs.covici.com>
Message-ID: <6a472945-8b42-c351-99d7-05efd504463f@googlemail.com>

Dear John,

Thank you very much for your advise.
It works very well!

It works only good and without any problems, if I choose the device 
directly.
If I choose "pulse" as ringdev, outdev and indev I have this "cracking" 
perioudicly in the line.


Thanks, Tamer

On 2020-01-12 08:06, John Covici wrote:
> How about <action application="export"
> data="media_mix_inbound_outbound_codecs=true"/>  See if that works.
>
> On Sat, 11 Jan 2020 19:52:54 -0500,
> Tamer Higazi via FreeSWITCH-users wrote:
>> [1  <message/rfc822 (7bit)>]
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on FS, but INCOPATIBLE DESTINATION
>> From: Tamer Higazi <th982a at googlemail.com>
>> Date: Sun, 12 Jan 2020 01:16:57 +0100
>>
>> Dear David,
>>
>> This is what I thought as well (just shortly).
>>
>> with the SIP phone dialing it bridges through the gateway, it
>> works without any problems.
>>
>> But with mod_portaudio or mod_alsa I have this problem.
>>
>> Any ideas how to fix it or to get more informations or how to fix it ?
>>
>> Because in the FS pages it's written nowhere what codecs should
>> be used or how to configure it ....
>>
>>
>> best, Tamer
>>
>> On 2020-01-12 01:12, David Villasmil wrote:
>>> That usually comes up when the far side and fs can’t agree on a
>>> codec. Make sure both sides have the same set of codecs
>>> enabled.
>>>
>>> On Sat, 11 Jan 2020 at 21:37, Tamer Higazi via FreeSWITCH-users
>>> <freeswitch-users at lists.freeswitch.org
>>> <mailto:freeswitch-users at lists.freeswitch.org>> wrote:
>>>
>>>
>>>
>>>
>>>      ---------- Forwarded message ----------
>>>      From: Tamer Higazi <th982a at googlemail.com
>>>      <mailto:th982a at googlemail.com>>
>>>      To: freeswitch-users at lists.freeswitch.org
>>>      <mailto:freeswitch-users at lists.freeswitch.org>
>>>      Cc:
>>>      Bcc:
>>>      Date: Sat, 11 Jan 2020 22:21:56 +0100
>>>      Subject: portaudio and pulsaudio works on FS, but INCOPATIBLE
>>>      DESTINATION
>>>      Hi people,
>>>
>>>      After long try and there I got "finally" portaudio running on my cpu.
>>>
>>>      I got the dumb error message "connection refused" rid of me, and FS
>>>      seems to access the soundcard.
>>>      I can choose the devices of my choice, and place a call.
>>>
>>>      But when I place a call, I still get this:
>>>
>>>      2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:935
>>>      (sofia/external/123456) State REPORTING
>>>      2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:174
>>>      sofia/external/123456 Standard REPORTING, cause:
>>>      INCOMPATIBLE_DESTINATION
>>>      2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:935
>>>      (sofia/external/123456) State REPORTING going to sleep
>>>      2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:610
>>>      (sofia/external/123456) State Change CS_REPORTING -> CS_DESTROY
>>>      2020-01-11 22:18:33.847800 [DEBUG] switch_core_session.c:1715
>>>      Session 17
>>>      (sofia/external/123456) Locked, Waiting on external entities
>>>      2020-01-11 22:18:33.847800 [DEBUG] switch_ivr_originate.c:3941
>>>      Originate
>>>      Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
>>>      2020-01-11 22:18:33.847800 [NOTICE] switch_core_session.c:1733
>>>      Session
>>>      17 (sofia/external/123456) Ended
>>>      2020-01-11 22:18:33.847800 [NOTICE] switch_core_session.c:1737 Close
>>>      Channel sofia/external/123456 [CS_DESTROY]
>>>      2020-01-11 22:18:33.847800 [DEBUG] switch_channel.c:2047
>>>      (portaudio/1123456) Callstate Change RING_WAIT -> ACTIVE
>>>      2020-01-11 22:18:33.847800 [INFO] mod_dptools.c:3518 Originate
>>>      Failed.
>>>      Cause: INCOMPATIBLE_DESTINATION
>>>      2020-01-11 22:18:33.847800 [NOTICE] switch_channel.c:4857 Hangup
>>>      portaudio/1123456 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
>>>      2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:738
>>>      (sofia/external/123456) Running State Change CS_DESTROY (Cur 1 Tot 17)
>>>
>>>      Any ideas why this is the case ?
>>>
>>>
>>>      best, Tamer
>>>
>>>
>>>
>>>
>>>
>>>      ---------- Forwarded message ----------
>>>      From: Tamer Higazi via FreeSWITCH-users
>>>      <freeswitch-users at lists.freeswitch.org
>>>      <mailto:freeswitch-users at lists.freeswitch.org>>
>>>      To: freeswitch-users at lists.freeswitch.org
>>>      <mailto:freeswitch-users at lists.freeswitch.org>
>>>      Cc:
>>>      Bcc:
>>>      Date: Sat, 11 Jan 2020 13:37:37 -0800 (PST)
>>>      Subject: [Freeswitch-users] portaudio and pulsaudio works on FS,
>>>      but INCOPATIBLE DESTINATION
>>>      _________________________________________________________________________
>>>
>>>      The FreeSWITCH project is sponsored by SignalWire
>>>      https://signalwire.com
>>>      Enhance your FreeSWITCH install with disruptive priced SMS and
>>>      PSTN services.
>>>      Build your next product on our scalable cloud platform.
>>>
>>>      Join our online community to chat in real time
>>>      https://signalwire.community
>>>
>>>      Professional FreeSWITCH Services
>>>      sales at freeswitch.com <mailto:sales at freeswitch.com>
>>>      https://freeswitch.com
>>>
>>>      Official FreeSWITCH Sites
>>>      https://freeswitch.com/oss
>>>      https://freeswitch.org/confluence
>>>      https://cluecon.com
>>>
>>>      FreeSWITCH-users mailing list
>>>      FreeSWITCH-users at lists.freeswitch.org
>>>      <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>      https://freeswitch.com
>>>
>>> -- 
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.work at gmail.com
>>> <mailto:david.villasmil.work at gmail.com>
>>> phone: +34669448337
>> [2  <text/plain; utf-8 (base64)>]
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com


From david.villasmil.work at gmail.com  Sun Jan 12 14:16:38 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sun, 12 Jan 2020 14:16:38 +0000
Subject: [Freeswitch-users] portaudio and pulsaudio works on FS,
 but INCOPATIBLE DESTINATION (SOLVED)
In-Reply-To: <mailman.916.1578837024.1226.freeswitch-users@lists.freeswitch.org>
References: <mailman.912.1578777754.1226.freeswitch-users@lists.freeswitch.org>
 <CAFGRPVqVZqT+pktvCFJAUQp056bB9Fzr7W4x+gHvAioQThcuUw@mail.gmail.com>
 <mailman.913.1578788251.1226.freeswitch-users@lists.freeswitch.org>
 <m34kx1taqh.wl-covici@ccs.covici.com>
 <mailman.916.1578837024.1226.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAFGRPVq583AqTJKHbfXtzxR_q2KaBpp_7_CgXq=L08LOKhSuCg@mail.gmail.com>

Can you provide more logging? And also a trace?

On Sun, 12 Jan 2020 at 14:00, Tamer Higazi via FreeSWITCH-users <
freeswitch-users at lists.freeswitch.org> wrote:

>
>
>
> ---------- Forwarded message ----------
> From: Tamer Higazi <th982a at googlemail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Sun, 12 Jan 2020 14:49:46 +0100
> Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on FS, but
> INCOPATIBLE DESTINATION (SOLVED)
> Dear John,
>
> Thank you very much for your advise.
> It works very well!
>
> It works only good and without any problems, if I choose the device
> directly.
> If I choose "pulse" as ringdev, outdev and indev I have this "cracking"
> perioudicly in the line.
>
>
> Thanks, Tamer
>
> On 2020-01-12 08:06, John Covici wrote:
> > How about <action application="export"
> > data="media_mix_inbound_outbound_codecs=true"/>  See if that works.
> >
> > On Sat, 11 Jan 2020 19:52:54 -0500,
> > Tamer Higazi via FreeSWITCH-users wrote:
> >> [1  <message/rfc822 (7bit)>]
> >> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> >> Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on FS,
> but INCOPATIBLE DESTINATION
> >> From: Tamer Higazi <th982a at googlemail.com>
> >> Date: Sun, 12 Jan 2020 01:16:57 +0100
> >>
> >> Dear David,
> >>
> >> This is what I thought as well (just shortly).
> >>
> >> with the SIP phone dialing it bridges through the gateway, it
> >> works without any problems.
> >>
> >> But with mod_portaudio or mod_alsa I have this problem.
> >>
> >> Any ideas how to fix it or to get more informations or how to fix it ?
> >>
> >> Because in the FS pages it's written nowhere what codecs should
> >> be used or how to configure it ....
> >>
> >>
> >> best, Tamer
> >>
> >> On 2020-01-12 01:12, David Villasmil wrote:
> >>> That usually comes up when the far side and fs can’t agree on a
> >>> codec. Make sure both sides have the same set of codecs
> >>> enabled.
> >>>
> >>> On Sat, 11 Jan 2020 at 21:37, Tamer Higazi via FreeSWITCH-users
> >>> <freeswitch-users at lists.freeswitch.org
> >>> <mailto:freeswitch-users at lists.freeswitch.org>> wrote:
> >>>
> >>>
> >>>
> >>>
> >>>      ---------- Forwarded message ----------
> >>>      From: Tamer Higazi <th982a at googlemail.com
> >>>      <mailto:th982a at googlemail.com>>
> >>>      To: freeswitch-users at lists.freeswitch.org
> >>>      <mailto:freeswitch-users at lists.freeswitch.org>
> >>>      Cc:
> >>>      Bcc:
> >>>      Date: Sat, 11 Jan 2020 22:21:56 +0100
> >>>      Subject: portaudio and pulsaudio works on FS, but INCOPATIBLE
> >>>      DESTINATION
> >>>      Hi people,
> >>>
> >>>      After long try and there I got "finally" portaudio running on my
> cpu.
> >>>
> >>>      I got the dumb error message "connection refused" rid of me, and
> FS
> >>>      seems to access the soundcard.
> >>>      I can choose the devices of my choice, and place a call.
> >>>
> >>>      But when I place a call, I still get this:
> >>>
> >>>      2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:935
> >>>      (sofia/external/123456) State REPORTING
> >>>      2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:174
> >>>      sofia/external/123456 Standard REPORTING, cause:
> >>>      INCOMPATIBLE_DESTINATION
> >>>      2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:935
> >>>      (sofia/external/123456) State REPORTING going to sleep
> >>>      2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:610
> >>>      (sofia/external/123456) State Change CS_REPORTING -> CS_DESTROY
> >>>      2020-01-11 22:18:33.847800 [DEBUG] switch_core_session.c:1715
> >>>      Session 17
> >>>      (sofia/external/123456) Locked, Waiting on external entities
> >>>      2020-01-11 22:18:33.847800 [DEBUG] switch_ivr_originate.c:3941
> >>>      Originate
> >>>      Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
> >>>      2020-01-11 22:18:33.847800 [NOTICE] switch_core_session.c:1733
> >>>      Session
> >>>      17 (sofia/external/123456) Ended
> >>>      2020-01-11 22:18:33.847800 [NOTICE] switch_core_session.c:1737
> Close
> >>>      Channel sofia/external/123456 [CS_DESTROY]
> >>>      2020-01-11 22:18:33.847800 [DEBUG] switch_channel.c:2047
> >>>      (portaudio/1123456) Callstate Change RING_WAIT -> ACTIVE
> >>>      2020-01-11 22:18:33.847800 [INFO] mod_dptools.c:3518 Originate
> >>>      Failed.
> >>>      Cause: INCOMPATIBLE_DESTINATION
> >>>      2020-01-11 22:18:33.847800 [NOTICE] switch_channel.c:4857 Hangup
> >>>      portaudio/1123456 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
> >>>      2020-01-11 22:18:33.847800 [DEBUG] switch_core_state_machine.c:738
> >>>      (sofia/external/123456) Running State Change CS_DESTROY (Cur 1
> Tot 17)
> >>>
> >>>      Any ideas why this is the case ?
> >>>
> >>>
> >>>      best, Tamer
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>      ---------- Forwarded message ----------
> >>>      From: Tamer Higazi via FreeSWITCH-users
> >>>      <freeswitch-users at lists.freeswitch.org
> >>>      <mailto:freeswitch-users at lists.freeswitch.org>>
> >>>      To: freeswitch-users at lists.freeswitch.org
> >>>      <mailto:freeswitch-users at lists.freeswitch.org>
> >>>      Cc:
> >>>      Bcc:
> >>>      Date: Sat, 11 Jan 2020 13:37:37 -0800 (PST)
> >>>      Subject: [Freeswitch-users] portaudio and pulsaudio works on FS,
> >>>      but INCOPATIBLE DESTINATION
> >>>
> _________________________________________________________________________
> >>>
> >>>      The FreeSWITCH project is sponsored by SignalWire
> >>>      https://signalwire.com
> >>>      Enhance your FreeSWITCH install with disruptive priced SMS and
> >>>      PSTN services.
> >>>      Build your next product on our scalable cloud platform.
> >>>
> >>>      Join our online community to chat in real time
> >>>      https://signalwire.community
> >>>
> >>>      Professional FreeSWITCH Services
> >>>      sales at freeswitch.com <mailto:sales at freeswitch.com>
> >>>      https://freeswitch.com
> >>>
> >>>      Official FreeSWITCH Sites
> >>>      https://freeswitch.com/oss
> >>>      https://freeswitch.org/confluence
> >>>      https://cluecon.com
> >>>
> >>>      FreeSWITCH-users mailing list
> >>>      FreeSWITCH-users at lists.freeswitch.org
> >>>      <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>>      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>      UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>      https://freeswitch.com
> >>>
> >>> --
> >>> Regards,
> >>>
> >>> David Villasmil
> >>> email: david.villasmil.work at gmail.com
> >>> <mailto:david.villasmil.work at gmail.com>
> >>> phone: +34669448337
> >> [2  <text/plain; utf-8 (base64)>]
> >>
> _________________________________________________________________________
> >>
> >> The FreeSWITCH project is sponsored by SignalWire
> https://signalwire.com
> >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> >> Build your next product on our scalable cloud platform.
> >>
> >> Join our online community to chat in real time
> https://signalwire.community
> >>
> >> Professional FreeSWITCH Services
> >> sales at freeswitch.com
> >> https://freeswitch.com
> >>
> >> Official FreeSWITCH Sites
> >> https://freeswitch.com/oss
> >> https://freeswitch.org/confluence
> >> https://cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> https://freeswitch.com
>
>
>
>
> ---------- Forwarded message ----------
> From: Tamer Higazi via FreeSWITCH-users <
> freeswitch-users at lists.freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Sun, 12 Jan 2020 06:00:54 -0800 (PST)
> Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on FS, but
> INCOPATIBLE DESTINATION (SOLVED)
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From th982a at googlemail.com  Sun Jan 12 14:28:01 2020
From: th982a at googlemail.com (Tamer Higazi)
Date: Sun, 12 Jan 2020 15:28:01 +0100
Subject: [Freeswitch-users] portaudio and pulsaudio works on FS, but
 INCOPATIBLE DESTINATION (SOLVED)
In-Reply-To: <CAFGRPVq583AqTJKHbfXtzxR_q2KaBpp_7_CgXq=L08LOKhSuCg@mail.gmail.com>
References: <mailman.912.1578777754.1226.freeswitch-users@lists.freeswitch.org>
 <CAFGRPVqVZqT+pktvCFJAUQp056bB9Fzr7W4x+gHvAioQThcuUw@mail.gmail.com>
 <mailman.913.1578788251.1226.freeswitch-users@lists.freeswitch.org>
 <m34kx1taqh.wl-covici@ccs.covici.com>
 <mailman.916.1578837024.1226.freeswitch-users@lists.freeswitch.org>
 <CAFGRPVq583AqTJKHbfXtzxR_q2KaBpp_7_CgXq=L08LOKhSuCg@mail.gmail.com>
Message-ID: <8c1241f8-1981-83d8-dfdb-4e528e3a974f@googlemail.com>

Yes....of course!

I try to simulate it once again with PulseAudio in combination with 
mod_portaudio.

In the meantime, could you please tell me how to raise the logginglevel 
and debugging for you and where it is saved?

I think logging on my machine is through mod_cdrcsv log.


Thanks.

On 2020-01-12 15:16, David Villasmil wrote:
> Can you provide more logging? And also a trace?
>
> On Sun, 12 Jan 2020 at 14:00, Tamer Higazi via FreeSWITCH-users 
> <freeswitch-users at lists.freeswitch.org 
> <mailto:freeswitch-users at lists.freeswitch.org>> wrote:
>
>
>
>
>     ---------- Forwarded message ----------
>     From: Tamer Higazi <th982a at googlemail.com
>     <mailto:th982a at googlemail.com>>
>     To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     Cc:
>     Bcc:
>     Date: Sun, 12 Jan 2020 14:49:46 +0100
>     Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on
>     FS, but INCOPATIBLE DESTINATION (SOLVED)
>     Dear John,
>
>     Thank you very much for your advise.
>     It works very well!
>
>     It works only good and without any problems, if I choose the device
>     directly.
>     If I choose "pulse" as ringdev, outdev and indev I have this
>     "cracking"
>     perioudicly in the line.
>
>
>     Thanks, Tamer
>
>     On 2020-01-12 08:06, John Covici wrote:
>     > How about <action application="export"
>     > data="media_mix_inbound_outbound_codecs=true"/>  See if that works.
>     >
>     > On Sat, 11 Jan 2020 19:52:54 -0500,
>     > Tamer Higazi via FreeSWITCH-users wrote:
>     >> [1  <message/rfc822 (7bit)>]
>     >> To: FreeSWITCH Users Help
>     <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     >> Subject: Re: [Freeswitch-users] portaudio and pulsaudio works
>     on FS, but INCOPATIBLE DESTINATION
>     >> From: Tamer Higazi <th982a at googlemail.com
>     <mailto:th982a at googlemail.com>>
>     >> Date: Sun, 12 Jan 2020 01:16:57 +0100
>     >>
>     >> Dear David,
>     >>
>     >> This is what I thought as well (just shortly).
>     >>
>     >> with the SIP phone dialing it bridges through the gateway, it
>     >> works without any problems.
>     >>
>     >> But with mod_portaudio or mod_alsa I have this problem.
>     >>
>     >> Any ideas how to fix it or to get more informations or how to
>     fix it ?
>     >>
>     >> Because in the FS pages it's written nowhere what codecs should
>     >> be used or how to configure it ....
>     >>
>     >>
>     >> best, Tamer
>     >>
>     >> On 2020-01-12 01:12, David Villasmil wrote:
>     >>> That usually comes up when the far side and fs can’t agree on a
>     >>> codec. Make sure both sides have the same set of codecs
>     >>> enabled.
>     >>>
>     >>> On Sat, 11 Jan 2020 at 21:37, Tamer Higazi via FreeSWITCH-users
>     >>> <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     >>> <mailto:freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>> wrote:
>     >>>
>     >>>
>     >>>
>     >>>
>     >>>      ---------- Forwarded message ----------
>     >>>      From: Tamer Higazi <th982a at googlemail.com
>     <mailto:th982a at googlemail.com>
>     >>>      <mailto:th982a at googlemail.com
>     <mailto:th982a at googlemail.com>>>
>     >>>      To: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     >>>      <mailto:freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     >>>      Cc:
>     >>>      Bcc:
>     >>>      Date: Sat, 11 Jan 2020 22:21:56 +0100
>     >>>      Subject: portaudio and pulsaudio works on FS, but INCOPATIBLE
>     >>>      DESTINATION
>     >>>      Hi people,
>     >>>
>     >>>      After long try and there I got "finally" portaudio
>     running on my cpu.
>     >>>
>     >>>      I got the dumb error message "connection refused" rid of
>     me, and FS
>     >>>      seems to access the soundcard.
>     >>>      I can choose the devices of my choice, and place a call.
>     >>>
>     >>>      But when I place a call, I still get this:
>     >>>
>     >>>      2020-01-11 22:18:33.847800 [DEBUG]
>     switch_core_state_machine.c:935
>     >>>      (sofia/external/123456) State REPORTING
>     >>>      2020-01-11 22:18:33.847800 [DEBUG]
>     switch_core_state_machine.c:174
>     >>>      sofia/external/123456 Standard REPORTING, cause:
>     >>>      INCOMPATIBLE_DESTINATION
>     >>>      2020-01-11 22:18:33.847800 [DEBUG]
>     switch_core_state_machine.c:935
>     >>>      (sofia/external/123456) State REPORTING going to sleep
>     >>>      2020-01-11 22:18:33.847800 [DEBUG]
>     switch_core_state_machine.c:610
>     >>>      (sofia/external/123456) State Change CS_REPORTING ->
>     CS_DESTROY
>     >>>      2020-01-11 22:18:33.847800 [DEBUG] switch_core_session.c:1715
>     >>>      Session 17
>     >>>      (sofia/external/123456) Locked, Waiting on external entities
>     >>>      2020-01-11 22:18:33.847800 [DEBUG]
>     switch_ivr_originate.c:3941
>     >>>      Originate
>     >>>      Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
>     >>>      2020-01-11 22:18:33.847800 [NOTICE]
>     switch_core_session.c:1733
>     >>>      Session
>     >>>      17 (sofia/external/123456) Ended
>     >>>      2020-01-11 22:18:33.847800 [NOTICE]
>     switch_core_session.c:1737 Close
>     >>>      Channel sofia/external/123456 [CS_DESTROY]
>     >>>      2020-01-11 22:18:33.847800 [DEBUG] switch_channel.c:2047
>     >>>      (portaudio/1123456) Callstate Change RING_WAIT -> ACTIVE
>     >>>      2020-01-11 22:18:33.847800 [INFO] mod_dptools.c:3518
>     Originate
>     >>>      Failed.
>     >>>      Cause: INCOMPATIBLE_DESTINATION
>     >>>      2020-01-11 22:18:33.847800 [NOTICE] switch_channel.c:4857
>     Hangup
>     >>>      portaudio/1123456 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
>     >>>      2020-01-11 22:18:33.847800 [DEBUG]
>     switch_core_state_machine.c:738
>     >>>      (sofia/external/123456) Running State Change CS_DESTROY
>     (Cur 1 Tot 17)
>     >>>
>     >>>      Any ideas why this is the case ?
>     >>>
>     >>>
>     >>>      best, Tamer
>     >>>
>     >>>
>     >>>
>     >>>
>     >>>
>     >>>      ---------- Forwarded message ----------
>     >>>      From: Tamer Higazi via FreeSWITCH-users
>     >>>      <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     >>>      <mailto:freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>>
>     >>>      To: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     >>>      <mailto:freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     >>>      Cc:
>     >>>      Bcc:
>     >>>      Date: Sat, 11 Jan 2020 13:37:37 -0800 (PST)
>     >>>      Subject: [Freeswitch-users] portaudio and pulsaudio works
>     on FS,
>     >>>      but INCOPATIBLE DESTINATION
>     >>>
>     _________________________________________________________________________
>     >>>
>     >>>      The FreeSWITCH project is sponsored by SignalWire
>     >>> https://signalwire.com
>     >>>      Enhance your FreeSWITCH install with disruptive priced
>     SMS and
>     >>>      PSTN services.
>     >>>      Build your next product on our scalable cloud platform.
>     >>>
>     >>>      Join our online community to chat in real time
>     >>> https://signalwire.community
>     >>>
>     >>>      Professional FreeSWITCH Services
>     >>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>     <mailto: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>
>     >>>      <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>>     
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >>> https://freeswitch.com
>     >>>
>     >>> --
>     >>> Regards,
>     >>>
>     >>> David Villasmil
>     >>> email: david.villasmil.work at gmail.com
>     <mailto:david.villasmil.work at gmail.com>
>     >>> <mailto:david.villasmil.work at gmail.com
>     <mailto:david.villasmil.work at gmail.com>>
>     >>> phone: +34669448337
>     >> [2  <text/plain; utf-8 (base64)>]
>     >>
>     _________________________________________________________________________
>     >>
>     >> The FreeSWITCH project is sponsored by SignalWire
>     https://signalwire.com
>     >> Enhance your FreeSWITCH install with disruptive priced SMS and
>     PSTN services.
>     >> Build your next product on our scalable cloud platform.
>     >>
>     >> Join our online community to chat in real time
>     https://signalwire.community
>     >>
>     >> Professional FreeSWITCH Services
>     >> sales at freeswitch.com <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
>
>
>
>
>     ---------- Forwarded message ----------
>     From: Tamer Higazi via FreeSWITCH-users
>     <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     Cc:
>     Bcc:
>     Date: Sun, 12 Jan 2020 06:00:54 -0800 (PST)
>     Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on
>     FS, but INCOPATIBLE DESTINATION (SOLVED)
>     _________________________________________________________________________
>
>     The FreeSWITCH project is sponsored by SignalWire
>     https://signalwire.com
>     Enhance your FreeSWITCH install with disruptive priced SMS and
>     PSTN services.
>     Build your next product on our scalable cloud platform.
>
>     Join our online community to chat in real time
>     https://signalwire.community
>
>     Professional FreeSWITCH Services
>     sales at freeswitch.com <mailto:sales at freeswitch.com>
>     https://freeswitch.com
>
>     Official FreeSWITCH Sites
>     https://freeswitch.com/oss
>     https://freeswitch.org/confluence
>     https://cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     https://freeswitch.com
>
> -- 
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com 
> <mailto:david.villasmil.work at gmail.com>
> phone: +34669448337


From david.villasmil.work at gmail.com  Sun Jan 12 15:06:06 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sun, 12 Jan 2020 15:06:06 +0000
Subject: [Freeswitch-users] portaudio and pulsaudio works on FS,
 but INCOPATIBLE DESTINATION (SOLVED)
In-Reply-To: <mailman.919.1578839315.1226.freeswitch-users@lists.freeswitch.org>
References: <mailman.912.1578777754.1226.freeswitch-users@lists.freeswitch.org>
 <CAFGRPVqVZqT+pktvCFJAUQp056bB9Fzr7W4x+gHvAioQThcuUw@mail.gmail.com>
 <mailman.913.1578788251.1226.freeswitch-users@lists.freeswitch.org>
 <m34kx1taqh.wl-covici@ccs.covici.com>
 <mailman.916.1578837024.1226.freeswitch-users@lists.freeswitch.org>
 <CAFGRPVq583AqTJKHbfXtzxR_q2KaBpp_7_CgXq=L08LOKhSuCg@mail.gmail.com>
 <mailman.919.1578839315.1226.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAFGRPVq3876QjYNwrcOdtN9JBe2DR-ruL=XPsm8ENWiScPHiDQ@mail.gmail.com>

On the cli:

sofia profile [internal/external] siptrace on

Where internal/external is the profile you’re using.

Should get the trace on the cli itself.

On Sun, 12 Jan 2020 at 15:02, Tamer Higazi via FreeSWITCH-users <
freeswitch-users at lists.freeswitch.org> wrote:

>
>
>
> ---------- Forwarded message ----------
> From: Tamer Higazi <th982a at googlemail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Sun, 12 Jan 2020 15:28:01 +0100
> Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on FS, but
> INCOPATIBLE DESTINATION (SOLVED)
> Yes....of course!
>
> I try to simulate it once again with PulseAudio in combination with
> mod_portaudio.
>
> In the meantime, could you please tell me how to raise the logginglevel
> and debugging for you and where it is saved?
>
> I think logging on my machine is through mod_cdrcsv log.
>
>
> Thanks.
>
> On 2020-01-12 15:16, David Villasmil wrote:
> > Can you provide more logging? And also a trace?
> >
> > On Sun, 12 Jan 2020 at 14:00, Tamer Higazi via FreeSWITCH-users
> > <freeswitch-users at lists.freeswitch.org
> > <mailto:freeswitch-users at lists.freeswitch.org>> wrote:
> >
> >
> >
> >
> >     ---------- Forwarded message ----------
> >     From: Tamer Higazi <th982a at googlemail.com
> >     <mailto:th982a at googlemail.com>>
> >     To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>>
> >     Cc:
> >     Bcc:
> >     Date: Sun, 12 Jan 2020 14:49:46 +0100
> >     Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on
> >     FS, but INCOPATIBLE DESTINATION (SOLVED)
> >     Dear John,
> >
> >     Thank you very much for your advise.
> >     It works very well!
> >
> >     It works only good and without any problems, if I choose the device
> >     directly.
> >     If I choose "pulse" as ringdev, outdev and indev I have this
> >     "cracking"
> >     perioudicly in the line.
> >
> >
> >     Thanks, Tamer
> >
> >     On 2020-01-12 08:06, John Covici wrote:
> >     > How about <action application="export"
> >     > data="media_mix_inbound_outbound_codecs=true"/>  See if that works.
> >     >
> >     > On Sat, 11 Jan 2020 19:52:54 -0500,
> >     > Tamer Higazi via FreeSWITCH-users wrote:
> >     >> [1  <message/rfc822 (7bit)>]
> >     >> To: FreeSWITCH Users Help
> >     <freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>>
> >     >> Subject: Re: [Freeswitch-users] portaudio and pulsaudio works
> >     on FS, but INCOPATIBLE DESTINATION
> >     >> From: Tamer Higazi <th982a at googlemail.com
> >     <mailto:th982a at googlemail.com>>
> >     >> Date: Sun, 12 Jan 2020 01:16:57 +0100
> >     >>
> >     >> Dear David,
> >     >>
> >     >> This is what I thought as well (just shortly).
> >     >>
> >     >> with the SIP phone dialing it bridges through the gateway, it
> >     >> works without any problems.
> >     >>
> >     >> But with mod_portaudio or mod_alsa I have this problem.
> >     >>
> >     >> Any ideas how to fix it or to get more informations or how to
> >     fix it ?
> >     >>
> >     >> Because in the FS pages it's written nowhere what codecs should
> >     >> be used or how to configure it ....
> >     >>
> >     >>
> >     >> best, Tamer
> >     >>
> >     >> On 2020-01-12 01:12, David Villasmil wrote:
> >     >>> That usually comes up when the far side and fs can’t agree on a
> >     >>> codec. Make sure both sides have the same set of codecs
> >     >>> enabled.
> >     >>>
> >     >>> On Sat, 11 Jan 2020 at 21:37, Tamer Higazi via FreeSWITCH-users
> >     >>> <freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>
> >     >>> <mailto:freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>>> wrote:
> >     >>>
> >     >>>
> >     >>>
> >     >>>
> >     >>>      ---------- Forwarded message ----------
> >     >>>      From: Tamer Higazi <th982a at googlemail.com
> >     <mailto:th982a at googlemail.com>
> >     >>>      <mailto:th982a at googlemail.com
> >     <mailto:th982a at googlemail.com>>>
> >     >>>      To: freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>
> >     >>>      <mailto:freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>>
> >     >>>      Cc:
> >     >>>      Bcc:
> >     >>>      Date: Sat, 11 Jan 2020 22:21:56 +0100
> >     >>>      Subject: portaudio and pulsaudio works on FS, but
> INCOPATIBLE
> >     >>>      DESTINATION
> >     >>>      Hi people,
> >     >>>
> >     >>>      After long try and there I got "finally" portaudio
> >     running on my cpu.
> >     >>>
> >     >>>      I got the dumb error message "connection refused" rid of
> >     me, and FS
> >     >>>      seems to access the soundcard.
> >     >>>      I can choose the devices of my choice, and place a call.
> >     >>>
> >     >>>      But when I place a call, I still get this:
> >     >>>
> >     >>>      2020-01-11 22:18:33.847800 [DEBUG]
> >     switch_core_state_machine.c:935
> >     >>>      (sofia/external/123456) State REPORTING
> >     >>>      2020-01-11 22:18:33.847800 [DEBUG]
> >     switch_core_state_machine.c:174
> >     >>>      sofia/external/123456 Standard REPORTING, cause:
> >     >>>      INCOMPATIBLE_DESTINATION
> >     >>>      2020-01-11 22:18:33.847800 [DEBUG]
> >     switch_core_state_machine.c:935
> >     >>>      (sofia/external/123456) State REPORTING going to sleep
> >     >>>      2020-01-11 22:18:33.847800 [DEBUG]
> >     switch_core_state_machine.c:610
> >     >>>      (sofia/external/123456) State Change CS_REPORTING ->
> >     CS_DESTROY
> >     >>>      2020-01-11 22:18:33.847800 [DEBUG]
> switch_core_session.c:1715
> >     >>>      Session 17
> >     >>>      (sofia/external/123456) Locked, Waiting on external entities
> >     >>>      2020-01-11 22:18:33.847800 [DEBUG]
> >     switch_ivr_originate.c:3941
> >     >>>      Originate
> >     >>>      Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
> >     >>>      2020-01-11 22:18:33.847800 [NOTICE]
> >     switch_core_session.c:1733
> >     >>>      Session
> >     >>>      17 (sofia/external/123456) Ended
> >     >>>      2020-01-11 22:18:33.847800 [NOTICE]
> >     switch_core_session.c:1737 Close
> >     >>>      Channel sofia/external/123456 [CS_DESTROY]
> >     >>>      2020-01-11 22:18:33.847800 [DEBUG] switch_channel.c:2047
> >     >>>      (portaudio/1123456) Callstate Change RING_WAIT -> ACTIVE
> >     >>>      2020-01-11 22:18:33.847800 [INFO] mod_dptools.c:3518
> >     Originate
> >     >>>      Failed.
> >     >>>      Cause: INCOMPATIBLE_DESTINATION
> >     >>>      2020-01-11 22:18:33.847800 [NOTICE] switch_channel.c:4857
> >     Hangup
> >     >>>      portaudio/1123456 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
> >     >>>      2020-01-11 22:18:33.847800 [DEBUG]
> >     switch_core_state_machine.c:738
> >     >>>      (sofia/external/123456) Running State Change CS_DESTROY
> >     (Cur 1 Tot 17)
> >     >>>
> >     >>>      Any ideas why this is the case ?
> >     >>>
> >     >>>
> >     >>>      best, Tamer
> >     >>>
> >     >>>
> >     >>>
> >     >>>
> >     >>>
> >     >>>      ---------- Forwarded message ----------
> >     >>>      From: Tamer Higazi via FreeSWITCH-users
> >     >>>      <freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>
> >     >>>      <mailto:freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>>>
> >     >>>      To: freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>
> >     >>>      <mailto:freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>>
> >     >>>      Cc:
> >     >>>      Bcc:
> >     >>>      Date: Sat, 11 Jan 2020 13:37:37 -0800 (PST)
> >     >>>      Subject: [Freeswitch-users] portaudio and pulsaudio works
> >     on FS,
> >     >>>      but INCOPATIBLE DESTINATION
> >     >>>
> >
>  _________________________________________________________________________
> >     >>>
> >     >>>      The FreeSWITCH project is sponsored by SignalWire
> >     >>> https://signalwire.com
> >     >>>      Enhance your FreeSWITCH install with disruptive priced
> >     SMS and
> >     >>>      PSTN services.
> >     >>>      Build your next product on our scalable cloud platform.
> >     >>>
> >     >>>      Join our online community to chat in real time
> >     >>> https://signalwire.community
> >     >>>
> >     >>>      Professional FreeSWITCH Services
> >     >>> sales at freeswitch.com <mailto:sales at freeswitch.com>
> >     <mailto: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>
> >     >>>      <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
> >     >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >>>
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     >>> https://freeswitch.com
> >     >>>
> >     >>> --
> >     >>> Regards,
> >     >>>
> >     >>> David Villasmil
> >     >>> email: david.villasmil.work at gmail.com
> >     <mailto:david.villasmil.work at gmail.com>
> >     >>> <mailto:david.villasmil.work at gmail.com
> >     <mailto:david.villasmil.work at gmail.com>>
> >     >>> phone: +34669448337
> >     >> [2  <text/plain; utf-8 (base64)>]
> >     >>
> >
>  _________________________________________________________________________
> >     >>
> >     >> The FreeSWITCH project is sponsored by SignalWire
> >     https://signalwire.com
> >     >> Enhance your FreeSWITCH install with disruptive priced SMS and
> >     PSTN services.
> >     >> Build your next product on our scalable cloud platform.
> >     >>
> >     >> Join our online community to chat in real time
> >     https://signalwire.community
> >     >>
> >     >> Professional FreeSWITCH Services
> >     >> sales at freeswitch.com <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
> >
> >
> >
> >
> >     ---------- Forwarded message ----------
> >     From: Tamer Higazi via FreeSWITCH-users
> >     <freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>>
> >     To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>>
> >     Cc:
> >     Bcc:
> >     Date: Sun, 12 Jan 2020 06:00:54 -0800 (PST)
> >     Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on
> >     FS, but INCOPATIBLE DESTINATION (SOLVED)
> >
>  _________________________________________________________________________
> >
> >     The FreeSWITCH project is sponsored by SignalWire
> >     https://signalwire.com
> >     Enhance your FreeSWITCH install with disruptive priced SMS and
> >     PSTN services.
> >     Build your next product on our scalable cloud platform.
> >
> >     Join our online community to chat in real time
> >     https://signalwire.community
> >
> >     Professional FreeSWITCH Services
> >     sales at freeswitch.com <mailto:sales at freeswitch.com>
> >     https://freeswitch.com
> >
> >     Official FreeSWITCH Sites
> >     https://freeswitch.com/oss
> >     https://freeswitch.org/confluence
> >     https://cluecon.com
> >
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     https://freeswitch.com
> >
> > --
> > Regards,
> >
> > David Villasmil
> > email: david.villasmil.work at gmail.com
> > <mailto:david.villasmil.work at gmail.com>
> > phone: +34669448337
>
>
>
>
> ---------- Forwarded message ----------
> From: Tamer Higazi via FreeSWITCH-users <
> freeswitch-users at lists.freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Sun, 12 Jan 2020 07:02:41 -0800 (PST)
> Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on FS, but
> INCOPATIBLE DESTINATION (SOLVED)
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From brians at iptel.co  Sun Jan 12 16:32:01 2020
From: brians at iptel.co (Brian :)
Date: Sun, 12 Jan 2020 16:32:01 +0000
Subject: [Freeswitch-users] Calls repeat at 80 seconds?
In-Reply-To: <0100016f99dce3c1-51f0a57f-e4f7-4c52-85e7-fc08c20a713f-000000@email.amazonses.com>
References: <0100016f99dce3c1-51f0a57f-e4f7-4c52-85e7-fc08c20a713f-000000@email.amazonses.com>
Message-ID: <CAGPQfi8qpqGk-mQ-VfF66Jw1L3VVZSVcBG=qndnCyibM4tMR+g@mail.gmail.com>

There were some carries that would do this with high value middle Eastern
destinations. Record first x seconds of call then hangup the actual call to
the high value destination and play the recording back. The caller would
stay connected and be billed. The unscrupulous carrier thus getting high
value per minute rates and not paying it on as the call was hung up.


On Sunday, January 12, 2020, Avi Marcus <avi at avimarcus.net> wrote:
> FreeSWITCH Version 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit)
> Is anyone having an issue with calls suddenly "restarting" at about 80
seconds? I thought it was 1 carrier on 1 route, but I'm getting reports now
across various destinations using different carriers. It's not on every
call, just some.
> I've upgraded to FreeSWITCH Version 1.10.2-release-13-f7bdd3845a~64bit
(-release-13-f7bdd3845a 64bit) and we'll see if I still get the issue.
>
>
> Thanks,
> -Avi Marcus
>
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From david.villasmil.work at gmail.com  Sun Jan 12 18:03:24 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sun, 12 Jan 2020 18:03:24 +0000
Subject: [Freeswitch-users] Calls repeat at 80 seconds?
In-Reply-To: <CAGPQfi8qpqGk-mQ-VfF66Jw1L3VVZSVcBG=qndnCyibM4tMR+g@mail.gmail.com>
References: <0100016f99dce3c1-51f0a57f-e4f7-4c52-85e7-fc08c20a713f-000000@email.amazonses.com>
 <CAGPQfi8qpqGk-mQ-VfF66Jw1L3VVZSVcBG=qndnCyibM4tMR+g@mail.gmail.com>
Message-ID: <CAFGRPVoJo0xmp6hyCpUSiun84inPTDK2Oxt7sb2-3RkhPg8GXQ@mail.gmail.com>

+1
Regards,

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


On Sun, Jan 12, 2020 at 5:01 PM Brian : <brians at iptel.co> wrote:

> There were some carries that would do this with high value middle Eastern
> destinations. Record first x seconds of call then hangup the actual call to
> the high value destination and play the recording back. The caller would
> stay connected and be billed. The unscrupulous carrier thus getting high
> value per minute rates and not paying it on as the call was hung up.
>
>
> On Sunday, January 12, 2020, Avi Marcus <avi at avimarcus.net> wrote:
> > FreeSWITCH Version 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit)
> > Is anyone having an issue with calls suddenly "restarting" at about 80
> seconds? I thought it was 1 carrier on 1 route, but I'm getting reports now
> across various destinations using different carriers. It's not on every
> call, just some.
> > I've upgraded to FreeSWITCH Version 1.10.2-release-13-f7bdd3845a~64bit
> (-release-13-f7bdd3845a 64bit) and we'll see if I still get the issue.
> >
> >
> > Thanks,
> > -Avi Marcus
> > _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From gregor at infomedia.si  Sun Jan 12 22:21:15 2020
From: gregor at infomedia.si (Gregor Nanger)
Date: Sun, 12 Jan 2020 23:21:15 +0100
Subject: [Freeswitch-users] Calls repeat at 80 seconds?
In-Reply-To: <CAFGRPVoJo0xmp6hyCpUSiun84inPTDK2Oxt7sb2-3RkhPg8GXQ@mail.gmail.com>
References: <0100016f99dce3c1-51f0a57f-e4f7-4c52-85e7-fc08c20a713f-000000@email.amazonses.com>
 <CAGPQfi8qpqGk-mQ-VfF66Jw1L3VVZSVcBG=qndnCyibM4tMR+g@mail.gmail.com>
 <CAFGRPVoJo0xmp6hyCpUSiun84inPTDK2Oxt7sb2-3RkhPg8GXQ@mail.gmail.com>
Message-ID: <CAC7ttFGNwq1qkP3CbTQWpXftDCXHFgrk6y2-J7jGw+jxLSmwFw@mail.gmail.com>

It's called FAS 😊

On Sun, 12 Jan 2020, 19:04 David Villasmil, <david.villasmil.work at gmail.com>
wrote:

> +1
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
>
> On Sun, Jan 12, 2020 at 5:01 PM Brian : <brians at iptel.co> wrote:
>
>> There were some carries that would do this with high value middle Eastern
>> destinations. Record first x seconds of call then hangup the actual call to
>> the high value destination and play the recording back. The caller would
>> stay connected and be billed. The unscrupulous carrier thus getting high
>> value per minute rates and not paying it on as the call was hung up.
>>
>>
>> On Sunday, January 12, 2020, Avi Marcus <avi at avimarcus.net> wrote:
>> > FreeSWITCH Version 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit)
>> > Is anyone having an issue with calls suddenly "restarting" at about 80
>> seconds? I thought it was 1 carrier on 1 route, but I'm getting reports now
>> across various destinations using different carriers. It's not on every
>> call, just some.
>> > I've upgraded to FreeSWITCH Version 1.10.2-release-13-f7bdd3845a~64bit
>> (-release-13-f7bdd3845a 64bit) and we'll see if I still get the issue.
>> >
>> >
>> > Thanks,
>> > -Avi Marcus
>> >
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From sanjeevdb4 at gmail.com  Tue Jan 14 09:41:53 2020
From: sanjeevdb4 at gmail.com (sanjeev dubey)
Date: Tue, 14 Jan 2020 15:11:53 +0530
Subject: [Freeswitch-users] need help of setting the proxy with kamailio
	server with freeswitch
In-Reply-To: <489523a3fcb34915abd8b6e87406bf09@c4b.de>
References: <de53c6fc0191427f86ee99e25404d3b7@c4b.de>
 <CAFGRPVrpJJkUsf53bXodr6ER_ehhOONoYsebPRbKkZVfdTK-4g@mail.gmail.com>
 <489523a3fcb34915abd8b6e87406bf09@c4b.de>
Message-ID: <CAE_Vhmoq7GeGkcvn=HEPE4tVcwqJcY_KC0pJOO1hXOgsnzJ4vQ@mail.gmail.com>

Hi Team,

Please help on the setting as proxy with kamailio sip server.
Thanks

On Fri, Jan 10, 2020, 7:33 PM Alexander Haugg <Alexander.Haugg at c4b.de>
wrote:

> OK,
>
> I have the solution.
>
> 1.       My think was to see the external candidate in the SDP.
> Freeswitch do that automatically, if the Client Registration from outside
> the local NW.
>
> 2.       To force adding the external candidate, there is the channel var
> “include_external_ip=true”. For example -> originate
> {include_external_ip=true,media_webrtc=true}sofia/gateway/GW_SBC2_B2Bua/22100
> &park
>
> 3.       The srflx candidate will be set, if the internal port different
> from the external port (stun request). See the switch_core_media
> implementation for the candidate generation.
>
>
>
> My questions are solved (at the moment ;-). My biggest problem was to
> understand the server stack vs. client stack behaviour for WebRTC.
>
>
>
> Thanks a lot!
>
> Alex
>
>
>
> *Von:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *Im
> Auftrag von *David Villasmil
> *Gesendet:* Freitag, 10. Januar 2020 13:59
> *An:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Betreff:* Re: [Freeswitch-users] WebRTC STUN request for a SIP profile
> is not working
>
>
>
> Does it work with a manually entered IP?
>
>
>
> On Fri, 10 Jan 2020 at 12:42, Alexander Haugg <Alexander.Haugg at c4b.de>
> wrote:
>
> Hi,
>
>
>
> if I set the parameter „<X-PRE-PROCESS cmd="stun-set"
> data="external_rtp_ip=stun:stun.freeswitch.org"/> “ in the vars.xml and
> use this variable $${external_rtp_ip}  in the SIP Profile, then the
> Ext-RTP-IP show the real external IP.
>
> I think, that’s OK.
>
>
>
> But I miss the srflx candidate in the sdp.
>
>
>
> I tried to experimental with the apply-candidate-acl (rfc1918.auto,
> any_v4.auto, wan_v4.auto), but nothing works.
>
> What’s wrong?
>
>
>
> Here is my sofia profile config:
>
> <profile xmlns:xsd="http://www.w3.org/2001/XMLSchema" xmlns:xsi="
> http://www.w3.org/2001/XMLSchema-instance" name="SBC2_B2Bua">
>
>   <aliases />
>
>   <domains>
>
>     <domain name="h3k.sip.c4b" alias="false" parse="true" />
>
>   </domains>
>
>   <settings>
>
>     <param name="user-agent-string" value="XPhone Call Controller" />
>
>     <param name="debug" value="0" />
>
> *    <param name="stun-enabled" value="true"/>*
>
> *    <param name="stun-auto-disable" value="false"/>*
>
>     <param name="sip-trace" value="no" />
>
>     <param name="sip-capture" value="no" />
>
>     <param name="dialplan" value="XML" />
>
>     <param name="context" value="SBC2_SP_outbound" />
>
>     <param name="dtmf-type" value="rfc2833" />
>
>     <param name="dtmf-duration" value="100" />
>
>     <param name="rfc2833-pt" value="98" />
>
>     <param name="inbound-codec-prefs" value="OPUS,PCMU,PCMA,VP8" />
>
>     <param name="outbound-codec-prefs" value="OPUS,PCMU,PCMA,VP8" />
>
>     <param name="hold-music" value="local_stream://moh" />
>
>     <param name="rtp-timer-name" value="none" />
>
>     <param name="rtp-rewrite-timestamps" value="true" />
>
>     <param name="manage-presence" value="false" />
>
>     <param name="inbound-codec-negotiation" value="generous" />
>
>     <param name="inbound-late-negotiation" value="true" />
>
>     <param name="nonce-ttl" value="60" />
>
>     <param name="auth-calls" value="false" />
>
>     <param name="sip-port" value="4901" />
>
>     <param name="rtp-ip" value="MyLocalIP" />
>
>     <param name="sip-ip" value=" MyLocalIP " />
>
>     *<param name="ext-rtp-ip" value="$${external_rtp_ip}" />*
>
>     <param name="ext-sip-ip" value=" MyLocalIP " />
>
>     <param name="local-network-acl" value="SBC2_localnet.auto" />
>
> *    <param name="apply-candidate-acl" value="rfc1918.auto" />*
>
> *    <param name="apply-candidate-acl" value="SBC2_localnet.auto" />*
>
> *    <param name="apply-candidate-acl" value="any_v4.auto" />*
>
> *    <param name="apply-candidate-acl" value="wan_v4.auto" />*
>
>     <param name="apply-inbound-acl" value="SBC2_localnet.auto" />
>
>     <param name="rtp-timeout-sec" value="300" />
>
>     <param name="rtp-hold-timeout-sec" value="1800" />
>
>     <param name="enable-3pcc" value="true" />
>
>     <param name="inbound-use-callid-as-uuid" value="true" />
>
>     <param name="tls" value="false" />
>
>     <param name="tls-only" value="false" />
>
>     <param name="tls-bind-params" value="transport=tls" />
>
>     <param name="tls-sip-port" value="" />
>
>     <param name="tls-passphrase" value="" />
>
>     <param name="tls-verify-date" value="true" />
>
>     <param name="tls-verify-depth" value="2" />
>
>     <param name="tls-verify-in-subjects" value="" />
>
>     <param name="tls-version" value="tlsv1,tlsv1.1,tlsv1.2" />
>
>     <param name="tls-ciphers" value="ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH" />
>
>     <param name="zrtp-passthru" value="true" />
>
>     <param name="inbound-reg-force-matching-username" value="true" />
>
>     <param name="auth-all-packets" value="false" />
>
>     <param name="wss-binding" value=":4447" />
>
>     <param name="caller-id-type" value="rpid" />
>
>     <param name="manual-redirect" value="true" />
>
>     <param name="rtcp-audio-interval-msec" value="5000" />
>
>   </settings>
>
>   <gateways>
>
>     <X-PRE-PROCESS cmd="include" data="external/GW_SBC2_B2Bua.conf.xml" />
>
>   </gateways>
>
> </profile>
>
>
>
> *Von:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *Im
> Auftrag von *Alexander Haugg
> *Gesendet:* Donnerstag, 9. Januar 2020 08:17
> *An:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Betreff:* [Freeswitch-users] WebRTC STUN request for a SIP profile is
> not working
>
>
>
> Hi,
>
>
>
> I’d read the manuals and configure my sofia profile to work with the
> external IP in the ext-rtp-ip setting.
>
> <param name="stun-enabled" value="true"/>
>
> <param name="stun-auto-disable" value="false"/>
>
> <param name="ext-rtp-ip" value="stun:stun.freeswitch.org" />
>
>
>
> The Freeswitch is started with the parameter –nonat
>
> In the pcap trace is no STUN request visible.
>
> I try to restart the Freeswitch, restart the profile, do an outbound call,
> but there is no STUN request.
>
>
>
> If I try to get my external IP via the CLI cmd “stun stun.freeswitch.org”,
> then works it successfully.
>
>
>
> The status output of the profile is:
>
>
> =================================================================================================
>
> Name                                SBC2_B2Bua
>
> Domain Name                 N/A
>
> Auto-NAT                         false
>
> DBName                           sofia_reg_SBC2_B2Bua
>
> Pres Hosts
>
> Dialplan                            XML
>
> Context                            SBC2_SP_outbound
>
> Challenge Realm             auto_to
>
> RTP-IP                               MY_LOCAL_IP
>
> Ext-RTP-IP                        stun:stun.freeswitch.org
>
> SIP-IP                                MY_LOCAL_IP
>
> Ext-SIP-IP                         MY_LOCAL_IP
>
> URL                                   sip:mod_sofia at MY_LOCAL_IP:4901
>
> BIND-URL
> sip:mod_sofia at MY_LOCAL_IP:4901;maddr=MY_LOCAL_IP;transport=udp,tcp
>
> HOLD-MUSIC                   local_stream://moh
>
> OUTBOUND-PROXY       N/A
>
> CODECS IN                       OPUS,PCMU,PCMA,VP8
>
> CODECS OUT                   OPUS,PCMU,PCMA,VP8
>
> TEL-EVENT                       98
>
> DTMF-MODE                   rfc2833
>
> CNG                                  13
>
> SESSION-TO                     0
>
> MAX-DIALOG                   0
>
> NOMEDIA                        false
>
> LATE-NEG                         true
>
> PROXY-MEDIA                 false
>
> ZRTP-PASSTHRU             false
>
> AGGRESSIVENAT           false
>
> CALLS-IN                           0
>
> FAILED-CALLS-IN             0
>
> CALLS-OUT                       0
>
> FAILED-CALLS-OUT        0
>
> REGISTRATIONS              0
>
>
>
> Thanks a lot!!!
>
> Alex
>
>
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
> --
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.work at gmail.com
>
> phone: +34669448337
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From spencer.angerbauer at gmail.com  Sun Jan 19 22:54:39 2020
From: spencer.angerbauer at gmail.com (Spencer Angerbauer)
Date: Sun, 19 Jan 2020 15:54:39 -0700
Subject: [Freeswitch-users] Requiring Pin of "1" for all conferences +
	announcement
References: <45214DC2-EBE2-4D6F-8C0F-BD7EF556D1E6@ventureslopes.com>
Message-ID: <D3FC2D27-F683-41C3-8B98-9D42D2EF0BF7@gmail.com>

I am trying to get all conferences to say a phrase before entering the pin #1 prior to entering (as we are connecting via outbound dialer from api). I have installed TTS to do a “say” function, and have updated the conf/autoload_configs/conference.conf.xml by making the “pin” required and to require pin as “1”, but every time a caller connects, it just transfers them directly into the conference without announcing that they need to press “1” to enter, nor does it require a “pin” to connect to the conference.

Is there somewhere else I need to update the dialing plan to require all my conference calls require a “1” when using the web api outbound dialer to connect everyone?  (I am using http://<<servername>>:8080/webapi/conference?<<friendlyname>> to connect multiple calls together).

Also, is there another location to add speak to text for the beginning of every conference?

Thank you for your help on both these related issues.

-Spence

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From ch.chhatra at gmail.com  Tue Jan 21 06:13:01 2020
From: ch.chhatra at gmail.com (Chhorm Chhatra)
Date: Tue, 21 Jan 2020 13:13:01 +0700
Subject: [Freeswitch-users] How to generate only one CDR record regardless
	of transfer?
Message-ID: <CAJssr-i2e+xWO=bS_WSf7xCDTVb+weYjGxzY8mCo1wEgRqo09w@mail.gmail.com>

Hi FreeSWITCH mailing list,
Currently, I encountered some problem when the call get transferred,
freeswitch generated multiple CDR for it.

How could I configure freeswitch to only create one CDR for both leg A and
B even if it get transferred?

Any help would be really appreciated.
Thank you in advance.

best regards,
Chhorm Chhatra
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From alex at digitalmail.com  Wed Jan 22 14:55:19 2020
From: alex at digitalmail.com (Alex Lake)
Date: Wed, 22 Jan 2020 14:55:19 +0000
Subject: [Freeswitch-users] Zoiper and push notification
Message-ID: <5cb3756c-b534-b4e3-6e44-0d9f1c5968a7@digitalmail.com>

Hi,

Is anyone here using Zoiper mobile app and their push notification system?

I had a go at following their instructions, but I don't think it worked.

Cheers

Alex


From victor.chukalovskiy at gmail.com  Wed Jan 22 15:46:46 2020
From: victor.chukalovskiy at gmail.com (Victor C)
Date: Wed, 22 Jan 2020 10:46:46 -0500
Subject: [Freeswitch-users] Zoiper and push notification
In-Reply-To: <5cb3756c-b534-b4e3-6e44-0d9f1c5968a7@digitalmail.com>
References: <5cb3756c-b534-b4e3-6e44-0d9f1c5968a7@digitalmail.com>
Message-ID: <8333B04F-CBE8-4F73-A747-E0CEC4B3D048@gmail.com>

Don't know about Zoiper, but in case it helps Session Talk with push works great for my FS needs.


> On Jan 22, 2020, at 09:55, Alex Lake <alex at digitalmail.com> wrote:
> 
> Hi,
> 
> Is anyone here using Zoiper mobile app and their push notification system?
> 
> I had a go at following their instructions, but I don't think it worked.
> 
> Cheers
> 
> Alex
> 
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com


From arjun.nainwal at startelelogic.co.in  Tue Jan 21 11:29:25 2020
From: arjun.nainwal at startelelogic.co.in (Arjunstl)
Date: Tue, 21 Jan 2020 04:29:25 -0700 (MST)
Subject: [Freeswitch-users] FreeSWITCH uniMRCP dynamic speech context
Message-ID: <1579606165435-0.post@n2.nabble.com>

Hi,
I'm working on ASR application with uniMRCP in lua. but when doing with
dynamic speech context mentioned in unimrcp usage manual. i'm unable to get
exact results.
my aim is to recognize only those words which i want to define in phrases(my
defined speech context). most of the times, recognizer recognizes random
words, that are not defined in speech context.

any help or suggestion would be highly appreciated.

Best Regards



--
Sent from: http://freeswitch-users.2379917.n2.nabble.com/


From s.safarov at gmail.com  Wed Jan 22 22:01:40 2020
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 23 Jan 2020 01:01:40 +0300
Subject: [Freeswitch-users] FreeSWITCH uniMRCP dynamic speech context
In-Reply-To: <1579606165435-0.post@n2.nabble.com>
References: <1579606165435-0.post@n2.nabble.com>
Message-ID: <CAHtxdDcvpbrdMxJ0kQ=4qTj5Ds4eCnWHfB4FQ4+oycaSeu+CNg@mail.gmail.com>

Think need to make manual test by excluding UniMRCP.
As example, if you use Google Seech plugging then need recognize wav record
file using REST API and check result.
If ASR returns correct results, then need open ticket on UniMRCP   portal
to resolve this issue.

On Wed, Jan 22, 2020 at 10:55 PM Arjunstl <arjun.nainwal at startelelogic.co.in>
wrote:

> Hi,
> I'm working on ASR application with uniMRCP in lua. but when doing with
> dynamic speech context mentioned in unimrcp usage manual. i'm unable to get
> exact results.
> my aim is to recognize only those words which i want to define in
> phrases(my
> defined speech context). most of the times, recognizer recognizes random
> words, that are not defined in speech context.
>
> any help or suggestion would be highly appreciated.
>
> Best Regards
>
>
>
> --
> Sent from: http://freeswitch-users.2379917.n2.nabble.com/
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From arjun.nainwal at startelelogic.co.in  Thu Jan 23 14:08:29 2020
From: arjun.nainwal at startelelogic.co.in (Arjun Nainwal)
Date: Thu, 23 Jan 2020 19:38:29 +0530
Subject: [Freeswitch-users] FreeSWITCH uniMRCP dynamic speech context
In-Reply-To: <CAHtxdDcvpbrdMxJ0kQ=4qTj5Ds4eCnWHfB4FQ4+oycaSeu+CNg@mail.gmail.com>
References: <1579606165435-0.post@n2.nabble.com>
 <CAHtxdDcvpbrdMxJ0kQ=4qTj5Ds4eCnWHfB4FQ4+oycaSeu+CNg@mail.gmail.com>
Message-ID: <CAFJjoZy0o9nOarw1ewGkg9akF1Ry3xB99iByZVAprtSBg7z0Ug@mail.gmail.com>

Thanks Sergey,
I'll do the same for testing.

On Thu, Jan 23, 2020, 4:08 AM Sergey Safarov <s.safarov at gmail.com> wrote:

> Think need to make manual test by excluding UniMRCP.
> As example, if you use Google Seech plugging then need recognize wav
> record file using REST API and check result.
> If ASR returns correct results, then need open ticket on UniMRCP   portal
> to resolve this issue.
>
> On Wed, Jan 22, 2020 at 10:55 PM Arjunstl <
> arjun.nainwal at startelelogic.co.in> wrote:
>
>> Hi,
>> I'm working on ASR application with uniMRCP in lua. but when doing with
>> dynamic speech context mentioned in unimrcp usage manual. i'm unable to
>> get
>> exact results.
>> my aim is to recognize only those words which i want to define in
>> phrases(my
>> defined speech context). most of the times, recognizer recognizes random
>> words, that are not defined in speech context.
>>
>> any help or suggestion would be highly appreciated.
>>
>> Best Regards
>>
>>
>>
>> --
>> Sent from: http://freeswitch-users.2379917.n2.nabble.com/
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From kliment.toshkov at netfinity.bg  Thu Jan 23 15:14:29 2020
From: kliment.toshkov at netfinity.bg (Kliment Toshkov, Netfinity JSC)
Date: Thu, 23 Jan 2020 17:14:29 +0200
Subject: [Freeswitch-users] Cannot force transcoding between PCMU and PCMA
Message-ID: <54E6EE36-B241-444B-8BB7-B8080809D49F@netfinity.bg>

Hello,

I have to force transcoding between PCMU and PCMA in the following scenario:

Customer [PCMA, PCMU, G729] > Kamailio > Freeswitch 1..N > termination gateway [PCMA]

I need transcoding from PCMU and G729 to take place on Freeswitch boxes. Currently it works for G729 but for PCMU calls they are originated with PCMU.

Inbound profile:
<param name="inbound-codec-prefs" value="$${global_codec_prefs}”/>
<param name="disable-transcoding" value="false"/>

Outbound profile:
<param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="disable-transcoding" value="false"/>

vars.xml:
  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,PCMU,G729"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA"/>


In xml dial plan:
              <action application="set" data="inherit_codec=false"/>
              <action application="set" data="media_mix_inbound_outbound_codecs=true"/>

But the calls are still not transcoded between PCMU and PCMA.

Only one thing worked - forcing the codec in bridge app:
                <action application="bridge" data="{absolute_codec_string='PCMA'} ..."/>

But I really need to know what I am missing and how to fix it using config files only. Brute force is not preferred.

Thank you!

Kliment Toshkov
Key Accounts

24/7: 0700 30000 w: www.netfinity.bg <http://www.netfinity.bg/>
t: +359 2 4918888 f: +359 2 4815555

 <http://www.netfinity.bg/>

This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. 




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From mike at freeswitch.org  Thu Jan 23 18:24:20 2020
From: mike at freeswitch.org (Mike Jerris)
Date: Thu, 23 Jan 2020 11:24:20 -0700
Subject: [Freeswitch-users] No audio for conference call
In-Reply-To: <CAJUJwtjesTiBr3aO1e-6CErM943SFhukb2Ujp6LxkxBDKR39bg@mail.gmail.com>
References: <CAF2MuSPfgCQer1_F2=B7Thg2yU0GvH-m4F0So75n1rBX5ZeaHw@mail.gmail.com>
 <CAHtxdDc5iVU0Pb5unZWX4+3WGmnG9tVHtMBy9-TgjN1__sRoTQ@mail.gmail.com>
 <CAF2MuSPYN4QAhVOFhOfNDGz+w0qLfe2CYaVsadw6Ca_0hDK1vA@mail.gmail.com>
 <CAFGRPVpLky8z2ufsd=XtoEHLqX-XLDdBNsKyH8Ke0K6=H_xzzw@mail.gmail.com>
 <CAF2MuSPKr=vNxfzqKPU8=6o8z2j7FzXig_Q9L8cSJfy0_Qso=g@mail.gmail.com>
 <CAFGRPVqdUoMAZZzb1skuWv-zkOEj6VwJ28dE5+En-sRfFExU1w@mail.gmail.com>
 <CAF2MuSOTfRM619ETWYwR-ZhWc7gunev_xe9Tmou5O5fqfDujJA@mail.gmail.com>
 <CAF2MuSNAp9Yk59JjVj8CqcG_AoYqSx_Bw1g6B1L85Jh_0RSeUA@mail.gmail.com>
 <CAFGRPVo2mpy6cC1wvc3iEdW1_boq=5sUeg8sGCvbEE-kJO+r7g@mail.gmail.com>
 <CAF2MuSOVooh_L+uWQ5W-vMWm-GiVBdgOVB=f0EPO7qHyUpHjgw@mail.gmail.com>
 <CAJUJwtjesTiBr3aO1e-6CErM943SFhukb2Ujp6LxkxBDKR39bg@mail.gmail.com>
Message-ID: <3762B8AB-2C3B-4370-AF96-3AB2412C4EEA@freeswitch.org>

Of note: 1.8.x is not the latest version.  You should be looking at the latest 1.10.x releases.

> On Jan 6, 2020, at 8:11 AM, SamyGo <govoiper at gmail.com> wrote:
> 
> Hi,
> We had a FS 1.8.6 dev setup and got exactly the same problem, no audio from participants in a conference. We upgraded to latest version 1.8.7 and the conference started working. 
> All connected parties were able to record their joining name, and could hear conference announcing the joining participants recorded msg but there was no audio being mixed. So I tried undead, unmute ALL but in vain.  Only  version upgrading helped. 
> 
> PS. We have a mix of WebRTC + regular devices on the conference and now everything works fine. WebRTC users are fronted by Kamailio+RTPEngine. 
> 
> Regards,
> Sammy
> 
> 
> 
> 
> On Mon, Jan 6, 2020 at 8:52 AM Md,Mehedi Hasan Kabir(Tanim) <tanim at surroundapps.com <mailto:tanim at surroundapps.com>> wrote:
> Hi David Villasmil
> 
> No. I just dialled 3000 from two zoiper android client to test freeswitch default conference dialplan at conf/dialplan/default.xml. My freeswitch version is 1.8.6.Direct audio call between two extension is fine. 
> 
> Regards
> Tanim
> 
> On Mon, Jan 6, 2020 at 4:21 PM David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
> Hello,
> 
> This is a webrtc conference?
> 
> On Mon, 6 Jan 2020 at 07:07, Md,Mehedi Hasan Kabir(Tanim) <tanim at surroundapps.com <mailto:tanim at surroundapps.com>> wrote:
> Hi David Villasmil
> 
> Can you guess anything from the attached log?
> 
> Regards
> Tanim
> 
> On Wed, Jan 1, 2020 at 3:55 PM Md,Mehedi Hasan Kabir(Tanim) <tanim at surroundapps.com <mailto:tanim at surroundapps.com>> wrote:
> Hi David
> 
> Please find the attached log. 
> 
> in cli, conference 3000-dialengine.sensor.buzz json_list command output is as follows
> [{
> 		"conference_name":	"3000-dialengine.sensor.buzz",
> 		"member_count":	2,
> 		"ghost_count":	0,
> 		"rate":	8000,
> 		"run_time":	235,
> 		"conference_uuid":	"af76568b-80e1-4def-8c0a-f2e2b3076925",
> 		"canvas_count":	0,
> 		"max_bw_in":	0,
> 		"force_bw_in":	0,
> 		"video_floor_packets":	0,
> 		"locked":	false,
> 		"destruct":	false,
> 		"wait_mod":	false,
> 		"audio_always":	false,
> 		"running":	true,
> 		"answered":	true,
> 		"enforce_min":	true,
> 		"bridge_to":	false,
> 		"dynamic":	true,
> 		"exit_sound":	true,
> 		"enter_sound":	true,
> 		"recording":	false,
> 		"video_bridge":	false,
> 		"video_floor_only":	false,
> 		"video_rfc4579":	false,
> 		"variables":	{
> 		},
> 		"members":	[{
> 				"type":	"caller",
> 				"id":	20,
> 				"flags":	{
> 					"can_hear":	true,
> 					"can_see":	true,
> 					"can_speak":	true,
> 					"hold":	false,
> 					"mute_detect":	false,
> 					"talking":	false,
> 					"has_video":	false,
> 					"video_bridge":	false,
> 					"has_floor":	false,
> 					"is_moderator":	false,
> 					"end_conference":	false
> 				},
> 				"uuid":	"584d5363-6243-4051-bb3e-6b2365ea4ed6",
> 				"caller_id_name":	"2112",
> 				"caller_id_number":	"2112",
> 				"join_time":	21,
> 				"last_talking":	0,
> 				"energy":	100,
> 				"volume_in":	0,
> 				"volume_out":	0,
> 				"output-volume":	0,
> 				"input-volume":	0
> 			}, {
> 				"type":	"caller",
> 				"id":	19,
> 				"flags":	{
> 					"can_hear":	true,
> 					"can_see":	true,
> 					"can_speak":	true,
> 					"hold":	false,
> 					"mute_detect":	false,
> 					"talking":	false,
> 					"has_video":	false,
> 					"video_bridge":	false,
> 					"has_floor":	true,
> 					"is_moderator":	false,
> 					"end_conference":	false
> 				},
> 				"uuid":	"3eb72d21-f767-4c49-a83f-f4d6ddc78eb4",
> 				"caller_id_name":	"2111",
> 				"caller_id_number":	"2111",
> 				"join_time":	235,
> 				"last_talking":	0,
> 				"energy":	100,
> 				"volume_in":	0,
> 				"volume_out":	0,
> 				"output-volume":	0,
> 				"input-volume":	0
> 			}]
> 	}]
> 
> Regards
> Tanim
> 
> On Tue, Dec 31, 2019 at 8:33 PM David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
> can you please share a SIP trace and all FS logs?
> 
> Regards,
> 
> David Villasmil
> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
> phone: +34669448337
> 
> 
> On Tue, Dec 31, 2019 at 2:59 PM Md,Mehedi Hasan Kabir(Tanim) <tanim at surroundapps.com <mailto:tanim at surroundapps.com>> wrote:
> Hi David
> 
> Yes,ext-sip-ip and ext-rtp-ip is set.Audio is ok for call between two zoiper client. problem occurs only for conference call. 
> 
> Regards
> Tanim
> 
> 
> On Tue, Dec 31, 2019, 4:36 PM David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
> Check the IP addresses offered from Zoiper and freeswitch. Make sure none is private.
> 
> Is fs behind NAT? If so, did you set the ext-sip-ip and ext-rtp-ip  ?
> 
> On Tue, 31 Dec 2019 at 05:24, Md,Mehedi Hasan Kabir(Tanim) <tanim at surroundapps.com <mailto:tanim at surroundapps.com>> wrote:
> Hi Sergey
> 
> No, call is not disconnected. Only some noise can be heard after enabling speaker.
> 
> Regards
> Tanim
> 
> On Mon, Dec 30, 2019 at 7:51 PM Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
> Is call disconnected after 34 second?
> 
> On Mon, Dec 30, 2019 at 4:37 PM Md,Mehedi Hasan Kabir(Tanim) <tanim at surroundapps.com <mailto:tanim at surroundapps.com>> wrote:
> Hi
> 
> I have tried to test FreeSWITCH conference by dialing 3000 from two zoiper clients. When I call the conference number, I can hear hold music. when the second call dial the conference number hold music stops, but no voice between legs. what would be the reason for this?
> 
> Audio is fine when I call other users directly.
> 
> Regards
> Tanim
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From mike at freeswitch.org  Thu Jan 23 18:29:42 2020
From: mike at freeswitch.org (Mike Jerris)
Date: Thu, 23 Jan 2020 11:29:42 -0700
Subject: [Freeswitch-users] Cannot force transcoding between PCMU and
	PCMA
In-Reply-To: <54E6EE36-B241-444B-8BB7-B8080809D49F@netfinity.bg>
References: <54E6EE36-B241-444B-8BB7-B8080809D49F@netfinity.bg>
Message-ID: <8750D75F-EF71-4116-B2CF-91E570E11B3F@freeswitch.org>

Absolute_codec_string is how you would force only an individual codec out the b leg as you’ve noted.  The other ways are about passing through the a leg codecs with some preferences and tweaks, mix adds our other codecs at the end.  These codecs are offered to the b leg then it chooses which to use.  If you ONLY want to offer PCMA out the b leg, absolute_codec_string is the right way.

> On Jan 23, 2020, at 8:14 AM, Kliment Toshkov, Netfinity JSC <kliment.toshkov at netfinity.bg> wrote:
> 
> Hello,
> 
> I have to force transcoding between PCMU and PCMA in the following scenario:
> 
> Customer [PCMA, PCMU, G729] > Kamailio > Freeswitch 1..N > termination gateway [PCMA]
> 
> I need transcoding from PCMU and G729 to take place on Freeswitch boxes. Currently it works for G729 but for PCMU calls they are originated with PCMU.
> 
> Inbound profile:
> <param name="inbound-codec-prefs" value="$${global_codec_prefs}”/>
> <param name="disable-transcoding" value="false"/>
> 
> Outbound profile:
> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
> <param name="disable-transcoding" value="false"/>
> 
> vars.xml:
>   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA,PCMU,G729"/>
>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA"/>
> 
> 
> In xml dial plan:
>               <action application="set" data="inherit_codec=false"/>
>               <action application="set" data="media_mix_inbound_outbound_codecs=true"/>
> 
> But the calls are still not transcoded between PCMU and PCMA.
> 
> Only one thing worked - forcing the codec in bridge app:
>                 <action application="bridge" data="{absolute_codec_string='PCMA'} ..."/>
> 
> But I really need to know what I am missing and how to fix it using config files only. Brute force is not preferred.
> 
> Thank you!
> 
> Kliment Toshkov
> Key Accounts
> 
> 24/7: 0700 30000 w: www.netfinity.bg <http://www.netfinity.bg/>
> t: +359 2 4918888 f: +359 2 4815555
> 
>  <http://www.netfinity.bg/>
> 
> This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. 
> 
> 

I’m sorry but we do not allow confidential emails on this mailing list.  Any emails to the list will not be confidential regardless of your disclaimer and will not later be removed or deleted (yes we’ve been asked to do this multiple times).  Please do not send disclaimers of confidentiality to the mailing list, they will be ignored.



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From spencer.angerbauer at gmail.com  Thu Jan 23 19:18:09 2020
From: spencer.angerbauer at gmail.com (Spencer Angerbauer)
Date: Thu, 23 Jan 2020 12:18:09 -0700
Subject: [Freeswitch-users] Requiring Pin of "1" for all conferences +
	announcement
In-Reply-To: <D3FC2D27-F683-41C3-8B98-9D42D2EF0BF7@gmail.com>
References: <45214DC2-EBE2-4D6F-8C0F-BD7EF556D1E6@ventureslopes.com>
 <D3FC2D27-F683-41C3-8B98-9D42D2EF0BF7@gmail.com>
Message-ID: <B395AF32-101C-42EA-9FB1-C072D09ACB0D@gmail.com>

Hello Freeswitch Pros,

I just wanted to see if by chance anyone had any suggestions on the below info and issue.  Thank you so much for your feedback and direction as it is greatly appreciated!

Thank you,

-Spence

> On Jan 19, 2020, at 3:54 PM, Spencer Angerbauer <spencer.angerbauer at gmail.com> wrote:
> 
> I am trying to get all conferences to say a phrase before entering the pin #1 prior to entering (as we are connecting via outbound dialer from api). I have installed TTS to do a “say” function, and have updated the conf/autoload_configs/conference.conf.xml by making the “pin” required and to require pin as “1”, but every time a caller connects, it just transfers them directly into the conference without announcing that they need to press “1” to enter, nor does it require a “pin” to connect to the conference.
> 
> Is there somewhere else I need to update the dialing plan to require all my conference calls require a “1” when using the web api outbound dialer to connect everyone?  (I am using http://<<servername>>:8080/webapi/originate?sofia/gateway/signalwire/<<phone number>>%20&conference(<<conference name>>) <http://api.joonto.com:8080/webapi/originate?sofia/gateway/signalwire/18015737111%20&conference(1234)>to connect multiple calls together).
> 
> Also, is there another location to add speak to text for the beginning of every conference from webapi or does it all need to be handled by the dial plan?
> 
> Thank you for your help on both these related issues.
> 
> -Spence
> 

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From david.villasmil.work at gmail.com  Fri Jan 24 00:07:58 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 24 Jan 2020 00:07:58 +0000
Subject: [Freeswitch-users] Requiring Pin of "1" for all conferences +
	announcement
In-Reply-To: <B395AF32-101C-42EA-9FB1-C072D09ACB0D@gmail.com>
References: <45214DC2-EBE2-4D6F-8C0F-BD7EF556D1E6@ventureslopes.com>
 <D3FC2D27-F683-41C3-8B98-9D42D2EF0BF7@gmail.com>
 <B395AF32-101C-42EA-9FB1-C072D09ACB0D@gmail.com>
Message-ID: <CAFGRPVpoWGRRgmpjt9M46DfkYi2RQg7SvNYzOg__xNT=d6zCDQ@mail.gmail.com>

Why don’t you originate and send to XML XXXX where XXXX is an extension and
on the extension you do first the announcement and gather the digit and if
it is 1 then send to the conference?

On Thu, 23 Jan 2020 at 19:26, Spencer Angerbauer <
spencer.angerbauer at gmail.com> wrote:

> Hello Freeswitch Pros,
>
> I just wanted to see if by chance anyone had any suggestions on the below
> info and issue.  Thank you so much for your feedback and direction as it is
> greatly appreciated!
>
> Thank you,
>
> -Spence
>
> On Jan 19, 2020, at 3:54 PM, Spencer Angerbauer <
> spencer.angerbauer at gmail.com> wrote:
>
> I am trying to get all conferences to say a phrase before entering the pin
> #1 prior to entering (as we are connecting via outbound dialer from api). I
> have installed TTS to do a “say” function, and have updated the
> conf/autoload_configs/conference.conf.xml by making the “pin” required
> and to require pin as “1”, but every time a caller connects, it just
> transfers them directly into the conference without announcing that they
> need to press “1” to enter, nor does it require a “pin” to connect to the
> conference.
>
> Is there somewhere else I need to update the dialing plan to require all
> my conference calls require a “1” when using the web api outbound dialer to
> connect everyone?  (I am using http://<<servername>>:8080/webapi/originate?sofia/gateway/signalwire/<<phone
> number>>%20&conference(<<conference name>>)
> <http://api.joonto.com:8080/webapi/originate?sofia/gateway/signalwire/18015737111%20&conference(1234)>to connect
> multiple calls together).
>
>
> Also, is there another location to add speak to text for the beginning of
> every conference from webapi or does it all need to be handled by the dial
> plan?
>
>
>
> Thank you for your help on both these related issues.
>
> -Spence
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From david.villasmil.work at gmail.com  Fri Jan 24 00:19:36 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 24 Jan 2020 00:19:36 +0000
Subject: [Freeswitch-users] Requiring Pin of "1" for all conferences +
	announcement
In-Reply-To: <CAFGRPVpoWGRRgmpjt9M46DfkYi2RQg7SvNYzOg__xNT=d6zCDQ@mail.gmail.com>
References: <45214DC2-EBE2-4D6F-8C0F-BD7EF556D1E6@ventureslopes.com>
 <D3FC2D27-F683-41C3-8B98-9D42D2EF0BF7@gmail.com>
 <B395AF32-101C-42EA-9FB1-C072D09ACB0D@gmail.com>
 <CAFGRPVpoWGRRgmpjt9M46DfkYi2RQg7SvNYzOg__xNT=d6zCDQ@mail.gmail.com>
Message-ID: <CAFGRPVrFuD0-Cs8DOH1qrN+a+GgXsDH5kg=qK7PnNgdBDc5G_w@mail.gmail.com>

Have you tried “+1”?

Like

http://<<servername>>:8080/webapi/conference%2B1?<<friendlyname>>


On Fri, 24 Jan 2020 at 00:07, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Why don’t you originate and send to XML XXXX where XXXX is an extension
> and on the extension you do first the announcement and gather the digit and
> if it is 1 then send to the conference?
>
> On Thu, 23 Jan 2020 at 19:26, Spencer Angerbauer <
> spencer.angerbauer at gmail.com> wrote:
>
>> Hello Freeswitch Pros,
>>
>> I just wanted to see if by chance anyone had any suggestions on the below
>> info and issue.  Thank you so much for your feedback and direction as it is
>> greatly appreciated!
>>
>> Thank you,
>>
>> -Spence
>>
>> On Jan 19, 2020, at 3:54 PM, Spencer Angerbauer <
>> spencer.angerbauer at gmail.com> wrote:
>>
>> I am trying to get all conferences to say a phrase before entering the
>> pin #1 prior to entering (as we are connecting via outbound dialer from
>> api). I have installed TTS to do a “say” function, and have updated the
>> conf/autoload_configs/conference.conf.xml by making the “pin” required
>> and to require pin as “1”, but every time a caller connects, it just
>> transfers them directly into the conference without announcing that they
>> need to press “1” to enter, nor does it require a “pin” to connect to the
>> conference.
>>
>> Is there somewhere else I need to update the dialing plan to require all
>> my conference calls require a “1” when using the web api outbound dialer to
>> connect everyone?  (I am using http://<<servername>>:8080/webapi/originate?sofia/gateway/signalwire/<<phone
>> number>>%20&conference(<<conference name>>)
>> <http://api.joonto.com:8080/webapi/originate?sofia/gateway/signalwire/18015737111%20&conference(1234)>to connect
>> multiple calls together).
>>
>>
>> Also, is there another location to add speak to text for the beginning of
>> every conference from webapi or does it all need to be handled by the dial
>> plan?
>>
>>
>>
>> Thank you for your help on both these related issues.
>>
>> -Spence
>>
>>
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From spencer.angerbauer at gmail.com  Fri Jan 24 00:39:08 2020
From: spencer.angerbauer at gmail.com (Spencer Angerbauer)
Date: Thu, 23 Jan 2020 17:39:08 -0700
Subject: [Freeswitch-users] Requiring Pin of "1" for all conferences +
 announcement
In-Reply-To: <CAFGRPVrFuD0-Cs8DOH1qrN+a+GgXsDH5kg=qK7PnNgdBDc5G_w@mail.gmail.com>
References: <45214DC2-EBE2-4D6F-8C0F-BD7EF556D1E6@ventureslopes.com>
 <D3FC2D27-F683-41C3-8B98-9D42D2EF0BF7@gmail.com>
 <B395AF32-101C-42EA-9FB1-C072D09ACB0D@gmail.com>
 <CAFGRPVpoWGRRgmpjt9M46DfkYi2RQg7SvNYzOg__xNT=d6zCDQ@mail.gmail.com>
 <CAFGRPVrFuD0-Cs8DOH1qrN+a+GgXsDH5kg=qK7PnNgdBDc5G_w@mail.gmail.com>
Message-ID: <7E0EFC95-8F9C-4009-A0B7-10B224049FC2@gmail.com>

Yes, I have tried that and unfortunately the bridge just hangs up right after it does an outgoing connection...

> On Jan 23, 2020, at 5:19 PM, David Villasmil <david.villasmil.work at gmail.com> wrote:
> 
> Have you tried “+1”?
> 
> Like
> 
> http://<<servername>>:8080/webapi/conference%2B1?<<friendlyname>>
> 
> 
> On Fri, 24 Jan 2020 at 00:07, David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
> Why don’t you originate and send to XML XXXX where XXXX is an extension and on the extension you do first the announcement and gather the digit and if it is 1 then send to the conference?
> 
> On Thu, 23 Jan 2020 at 19:26, Spencer Angerbauer <spencer.angerbauer at gmail.com <mailto:spencer.angerbauer at gmail.com>> wrote:
> Hello Freeswitch Pros,
> 
> I just wanted to see if by chance anyone had any suggestions on the below info and issue.  Thank you so much for your feedback and direction as it is greatly appreciated!
> 
> Thank you,
> 
> -Spence
> 
>> On Jan 19, 2020, at 3:54 PM, Spencer Angerbauer <spencer.angerbauer at gmail.com <mailto:spencer.angerbauer at gmail.com>> wrote:
>> 
>> I am trying to get all conferences to say a phrase before entering the pin #1 prior to entering (as we are connecting via outbound dialer from api). I have installed TTS to do a “say” function, and have updated the conf/autoload_configs/conference.conf.xml by making the “pin” required and to require pin as “1”, but every time a caller connects, it just transfers them directly into the conference without announcing that they need to press “1” to enter, nor does it require a “pin” to connect to the conference.
>> 
>> Is there somewhere else I need to update the dialing plan to require all my conference calls require a “1” when using the web api outbound dialer to connect everyone?  (I am using http://<<servername>>:8080/webapi/originate?sofia/gateway/signalwire/<<phone number>>%20&conference(<<conference name>>) <http://api.joonto.com:8080/webapi/originate?sofia/gateway/signalwire/18015737111%20&conference(1234)>to connect multiple calls together).
>> 
>> Also, is there another location to add speak to text for the beginning of every conference from webapi or does it all need to be handled by the dial plan?
> 
>> 
>> 
>> Thank you for your help on both these related issues.
>> 
>> -Spence
>> 
> 
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>-- 
> Regards,
> 
> David Villasmil
> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
> phone: +34669448337
> -- 
> Regards,
> 
> David Villasmil
> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
> phone: +34669448337
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From spencer.angerbauer at gmail.com  Fri Jan 24 05:08:04 2020
From: spencer.angerbauer at gmail.com (Spencer Angerbauer)
Date: Thu, 23 Jan 2020 22:08:04 -0700
Subject: [Freeswitch-users] Requiring Pin of "1" for all conferences +
 announcement
In-Reply-To: <7E0EFC95-8F9C-4009-A0B7-10B224049FC2@gmail.com>
References: <45214DC2-EBE2-4D6F-8C0F-BD7EF556D1E6@ventureslopes.com>
 <D3FC2D27-F683-41C3-8B98-9D42D2EF0BF7@gmail.com>
 <B395AF32-101C-42EA-9FB1-C072D09ACB0D@gmail.com>
 <CAFGRPVpoWGRRgmpjt9M46DfkYi2RQg7SvNYzOg__xNT=d6zCDQ@mail.gmail.com>
 <CAFGRPVrFuD0-Cs8DOH1qrN+a+GgXsDH5kg=qK7PnNgdBDc5G_w@mail.gmail.com>
 <7E0EFC95-8F9C-4009-A0B7-10B224049FC2@gmail.com>
Message-ID: <87A2B57A-E57A-4EF0-9E63-BBD064C47781@gmail.com>

I’ve tried every variation in front of the conference call name as well as after it….  I’m still really hoping I can get the “1” pin code to work via webapi…. Any other suggestions?

> On Jan 23, 2020, at 5:39 PM, Spencer Angerbauer <spencer.angerbauer at gmail.com> wrote:
> 
> Yes, I have tried that and unfortunately the bridge just hangs up right after it does an outgoing connection...
> 
>> On Jan 23, 2020, at 5:19 PM, David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
>> 
>> Have you tried “+1”?
>> 
>> Like
>> 
>> http://<<servername>>:8080/webapi/conference%2B1?<<friendlyname>>
>> 
>> 
>> On Fri, 24 Jan 2020 at 00:07, David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
>> Why don’t you originate and send to XML XXXX where XXXX is an extension and on the extension you do first the announcement and gather the digit and if it is 1 then send to the conference?
>> 
>> On Thu, 23 Jan 2020 at 19:26, Spencer Angerbauer <spencer.angerbauer at gmail.com <mailto:spencer.angerbauer at gmail.com>> wrote:
>> Hello Freeswitch Pros,
>> 
>> I just wanted to see if by chance anyone had any suggestions on the below info and issue.  Thank you so much for your feedback and direction as it is greatly appreciated!
>> 
>> Thank you,
>> 
>> -Spence
>> 
>>> On Jan 19, 2020, at 3:54 PM, Spencer Angerbauer <spencer.angerbauer at gmail.com <mailto:spencer.angerbauer at gmail.com>> wrote:
>>> 
>>> I am trying to get all conferences to say a phrase before entering the pin #1 prior to entering (as we are connecting via outbound dialer from api). I have installed TTS to do a “say” function, and have updated the conf/autoload_configs/conference.conf.xml by making the “pin” required and to require pin as “1”, but every time a caller connects, it just transfers them directly into the conference without announcing that they need to press “1” to enter, nor does it require a “pin” to connect to the conference.
>>> 
>>> Is there somewhere else I need to update the dialing plan to require all my conference calls require a “1” when using the web api outbound dialer to connect everyone?  (I am using http://<<servername>>:8080/webapi/originate?sofia/gateway/signalwire/<<phone number>>%20&conference(<<conference name>>) <http://api.joonto.com:8080/webapi/originate?sofia/gateway/signalwire/18015737111%20&conference(1234)>to connect multiple calls together).
>>> 
>>> Also, is there another location to add speak to text for the beginning of every conference from webapi or does it all need to be handled by the dial plan?
>> 
>>> 
>>> 
>>> Thank you for your help on both these related issues.
>>> 
>>> -Spence
>>> 
>> 
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>> Build your next product on our scalable cloud platform.
>> 
>> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>> https://cluecon.com <https://cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> https://freeswitch.com <https://freeswitch.com/>-- 
>> Regards,
>> 
>> David Villasmil
>> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
>> phone: +34669448337
>> -- 
>> Regards,
>> 
>> David Villasmil
>> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
>> phone: +34669448337
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>> Build your next product on our scalable cloud platform.
>> 
>> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>> https://cluecon.com <https://cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> https://freeswitch.com <https://freeswitch.com/>
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From spencer.angerbauer at gmail.com  Fri Jan 24 05:28:18 2020
From: spencer.angerbauer at gmail.com (Spencer Angerbauer)
Date: Thu, 23 Jan 2020 22:28:18 -0700
Subject: [Freeswitch-users] Requiring Pin of "1" for all conferences +
 announcement
In-Reply-To: <7E0EFC95-8F9C-4009-A0B7-10B224049FC2@gmail.com>
References: <45214DC2-EBE2-4D6F-8C0F-BD7EF556D1E6@ventureslopes.com>
 <D3FC2D27-F683-41C3-8B98-9D42D2EF0BF7@gmail.com>
 <B395AF32-101C-42EA-9FB1-C072D09ACB0D@gmail.com>
 <CAFGRPVpoWGRRgmpjt9M46DfkYi2RQg7SvNYzOg__xNT=d6zCDQ@mail.gmail.com>
 <CAFGRPVrFuD0-Cs8DOH1qrN+a+GgXsDH5kg=qK7PnNgdBDc5G_w@mail.gmail.com>
 <7E0EFC95-8F9C-4009-A0B7-10B224049FC2@gmail.com>
Message-ID: <83DE024C-2765-4E87-84FD-CBA5F8479A3D@gmail.com>

Ive also update all conference.con.xml to include the following:

 <!-- Conference pin -->
      <param name="pin" value="1"/>
      <param name="moderator-pin" value="1”/>



But still no luck on pin being required upon entrance into a conference...



> On Jan 23, 2020, at 5:39 PM, Spencer Angerbauer <spencer.angerbauer at gmail.com> wrote:
> 
> Yes, I have tried that and unfortunately the bridge just hangs up right after it does an outgoing connection...
> 
>> On Jan 23, 2020, at 5:19 PM, David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
>> 
>> Have you tried “+1”?
>> 
>> Like
>> 
>> http://<<servername>>:8080/webapi/conference%2B1?<<friendlyname>>
>> 
>> 
>> On Fri, 24 Jan 2020 at 00:07, David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
>> Why don’t you originate and send to XML XXXX where XXXX is an extension and on the extension you do first the announcement and gather the digit and if it is 1 then send to the conference?
>> 
>> On Thu, 23 Jan 2020 at 19:26, Spencer Angerbauer <spencer.angerbauer at gmail.com <mailto:spencer.angerbauer at gmail.com>> wrote:
>> Hello Freeswitch Pros,
>> 
>> I just wanted to see if by chance anyone had any suggestions on the below info and issue.  Thank you so much for your feedback and direction as it is greatly appreciated!
>> 
>> Thank you,
>> 
>> -Spence
>> 
>>> On Jan 19, 2020, at 3:54 PM, Spencer Angerbauer <spencer.angerbauer at gmail.com <mailto:spencer.angerbauer at gmail.com>> wrote:
>>> 
>>> I am trying to get all conferences to say a phrase before entering the pin #1 prior to entering (as we are connecting via outbound dialer from api). I have installed TTS to do a “say” function, and have updated the conf/autoload_configs/conference.conf.xml by making the “pin” required and to require pin as “1”, but every time a caller connects, it just transfers them directly into the conference without announcing that they need to press “1” to enter, nor does it require a “pin” to connect to the conference.
>>> 
>>> Is there somewhere else I need to update the dialing plan to require all my conference calls require a “1” when using the web api outbound dialer to connect everyone?  (I am using http://<<servername>>:8080/webapi/originate?sofia/gateway/signalwire/<<phone number>>%20&conference(<<conference name>>) <http://api.joonto.com:8080/webapi/originate?sofia/gateway/signalwire/18015737111%20&conference(1234)>to connect multiple calls together).
>>> 
>>> Also, is there another location to add speak to text for the beginning of every conference from webapi or does it all need to be handled by the dial plan?
>> 
>>> 
>>> 
>>> Thank you for your help on both these related issues.
>>> 
>>> -Spence
>>> 
>> 
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>> Build your next product on our scalable cloud platform.
>> 
>> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>> https://cluecon.com <https://cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> https://freeswitch.com <https://freeswitch.com/>-- 
>> Regards,
>> 
>> David Villasmil
>> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
>> phone: +34669448337
>> -- 
>> Regards,
>> 
>> David Villasmil
>> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
>> phone: +34669448337
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>> Build your next product on our scalable cloud platform.
>> 
>> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>> https://cluecon.com <https://cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> https://freeswitch.com <https://freeswitch.com/>
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From sdisanti.eteleco at gmail.com  Fri Jan 24 19:50:30 2020
From: sdisanti.eteleco at gmail.com (Sean DiSanti)
Date: Fri, 24 Jan 2020 11:50:30 -0800
Subject: [Freeswitch-users] Requiring Pin of "1" for all conferences +
	announcement
In-Reply-To: <83DE024C-2765-4E87-84FD-CBA5F8479A3D@gmail.com>
References: <45214DC2-EBE2-4D6F-8C0F-BD7EF556D1E6@ventureslopes.com>
 <D3FC2D27-F683-41C3-8B98-9D42D2EF0BF7@gmail.com>
 <B395AF32-101C-42EA-9FB1-C072D09ACB0D@gmail.com>
 <CAFGRPVpoWGRRgmpjt9M46DfkYi2RQg7SvNYzOg__xNT=d6zCDQ@mail.gmail.com>
 <CAFGRPVrFuD0-Cs8DOH1qrN+a+GgXsDH5kg=qK7PnNgdBDc5G_w@mail.gmail.com>
 <7E0EFC95-8F9C-4009-A0B7-10B224049FC2@gmail.com>
 <83DE024C-2765-4E87-84FD-CBA5F8479A3D@gmail.com>
Message-ID: <CAANGHxt1nZ_5-7kUiyiGjY3t4Qs5zS5mHm7nnA_HZU0sTMf=eQ@mail.gmail.com>

I'm sure this is a silly question, but I don't want to assume. You are
reloading xml after your changes, right?

On Thu, Jan 23, 2020 at 9:28 PM Spencer Angerbauer <
spencer.angerbauer at gmail.com> wrote:

> Ive also update all conference.con.xml to include the following:
>
>  <!-- Conference pin -->
>       <param name="pin" value="1"/>
>       <param name="moderator-pin" value="1”/>
>
>
>
> But still no luck on pin being required upon entrance into a conference...
>
>
>
> On Jan 23, 2020, at 5:39 PM, Spencer Angerbauer <
> spencer.angerbauer at gmail.com> wrote:
>
> Yes, I have tried that and unfortunately the bridge just hangs up right
> after it does an outgoing connection...
>
> On Jan 23, 2020, at 5:19 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
> Have you tried “+1”?
>
> Like
>
> http://<<servername>>:8080/webapi/conference%2B1?<<friendlyname>>
>
>
> On Fri, 24 Jan 2020 at 00:07, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Why don’t you originate and send to XML XXXX where XXXX is an extension
>> and on the extension you do first the announcement and gather the digit and
>> if it is 1 then send to the conference?
>>
>> On Thu, 23 Jan 2020 at 19:26, Spencer Angerbauer <
>> spencer.angerbauer at gmail.com> wrote:
>>
>>> Hello Freeswitch Pros,
>>>
>>> I just wanted to see if by chance anyone had any suggestions on the
>>> below info and issue.  Thank you so much for your feedback and direction as
>>> it is greatly appreciated!
>>>
>>> Thank you,
>>>
>>> -Spence
>>>
>>> On Jan 19, 2020, at 3:54 PM, Spencer Angerbauer <
>>> spencer.angerbauer at gmail.com> wrote:
>>>
>>> I am trying to get all conferences to say a phrase before entering the
>>> pin #1 prior to entering (as we are connecting via outbound dialer from
>>> api). I have installed TTS to do a “say” function, and have updated the
>>> conf/autoload_configs/conference.conf.xml by making the “pin” required
>>> and to require pin as “1”, but every time a caller connects, it just
>>> transfers them directly into the conference without announcing that they
>>> need to press “1” to enter, nor does it require a “pin” to connect to the
>>> conference.
>>>
>>> Is there somewhere else I need to update the dialing plan to require all
>>> my conference calls require a “1” when using the web api outbound dialer to
>>> connect everyone?  (I am using http://<<servername>>:8080/webapi/originate?sofia/gateway/signalwire/<<phone
>>> number>>%20&conference(<<conference name>>)
>>> <http://api.joonto.com:8080/webapi/originate?sofia/gateway/signalwire/18015737111%20&conference(1234)>to connect
>>> multiple calls together).
>>>
>>>
>>> Also, is there another location to add speak to text for the beginning
>>> of every conference from webapi or does it all need to be handled by the
>>> dial plan?
>>>
>>>
>>>
>>> Thank you for your help on both these related issues.
>>>
>>> -Spence
>>>
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> --
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From spencer.angerbauer at gmail.com  Fri Jan 24 19:52:43 2020
From: spencer.angerbauer at gmail.com (Spencer Angerbauer)
Date: Fri, 24 Jan 2020 12:52:43 -0700
Subject: [Freeswitch-users] Requiring Pin of "1" for all conferences +
	announcement
In-Reply-To: <CAANGHxt1nZ_5-7kUiyiGjY3t4Qs5zS5mHm7nnA_HZU0sTMf=eQ@mail.gmail.com>
References: <CAANGHxt1nZ_5-7kUiyiGjY3t4Qs5zS5mHm7nnA_HZU0sTMf=eQ@mail.gmail.com>
Message-ID: <97D9E988-784E-4A2B-987E-580174F57011@gmail.com>

Yes. I am reloading each time on any changes but that would have been awesome if I hadn’t been. Lol. 

Sent from my iPhone

> On Jan 24, 2020, at 12:51 PM, Sean DiSanti <sdisanti.eteleco at gmail.com> wrote:
> 
> 
> I'm sure this is a silly question, but I don't want to assume. You are reloading xml after your changes, right?
> 
>> On Thu, Jan 23, 2020 at 9:28 PM Spencer Angerbauer <spencer.angerbauer at gmail.com> wrote:
>> Ive also update all conference.con.xml to include the following:
>> 
>>  <!-- Conference pin -->
>>       <param name="pin" value="1"/>
>>       <param name="moderator-pin" value="1”/>
>> 
>> 
>> 
>> But still no luck on pin being required upon entrance into a conference...
>> 
>> 
>> 
>>> On Jan 23, 2020, at 5:39 PM, Spencer Angerbauer <spencer.angerbauer at gmail.com> wrote:
>>> 
>>> Yes, I have tried that and unfortunately the bridge just hangs up right after it does an outgoing connection...
>>> 
>>>> On Jan 23, 2020, at 5:19 PM, David Villasmil <david.villasmil.work at gmail.com> wrote:
>>>> 
>>>> Have you tried “+1”?
>>>> 
>>>> Like
>>>> 
>>>> http://<<servername>>:8080/webapi/conference%2B1?<<friendlyname>>
>>>> 
>>>> 
>>>>> On Fri, 24 Jan 2020 at 00:07, David Villasmil <david.villasmil.work at gmail.com> wrote:
>>>>> Why don’t you originate and send to XML XXXX where XXXX is an extension and on the extension you do first the announcement and gather the digit and if it is 1 then send to the conference?
>>>>> 
>>>>>> On Thu, 23 Jan 2020 at 19:26, Spencer Angerbauer <spencer.angerbauer at gmail.com> wrote:
>>>>>> Hello Freeswitch Pros,
>>>>>> 
>>>>>> I just wanted to see if by chance anyone had any suggestions on the below info and issue.  Thank you so much for your feedback and direction as it is greatly appreciated!
>>>>>> 
>>>>>> Thank you,
>>>>>> 
>>>>>> -Spence
>>>>>> 
>>>>>>>> On Jan 19, 2020, at 3:54 PM, Spencer Angerbauer <spencer.angerbauer at gmail.com> wrote:
>>>>>>>> 
>>>>>>>> I am trying to get all conferences to say a phrase before entering the pin #1 prior to entering (as we are connecting via outbound dialer from api). I have installed TTS to do a “say” function, and have updated the conf/autoload_configs/conference.conf.xml by making the “pin” required and to require pin as “1”, but every time a caller connects, it just transfers them directly into the conference without announcing that they need to press “1” to enter, nor does it require a “pin” to connect to the conference.
>>>>>>>> 
>>>>>>>> Is there somewhere else I need to update the dialing plan to require all my conference calls require a “1” when using the web api outbound dialer to connect everyone?  (I am using http://<<servername>>:8080/webapi/originate?sofia/gateway/signalwire/<<phone number>>%20&conference(<<conference name>>)to connect multiple calls together).
>>>>>>>> 
>>>>>>>> Also, is there another location to add speak to text for the beginning of every conference from webapi or does it all need to be handled by the dial plan?
>>>>>>> 
>>>>>>> 
>>>>>>> Thank you for your help on both these related issues.
>>>>>>> 
>>>>>>> -Spence
>>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> 
>>>>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>>>>>> Build your next product on our scalable cloud platform.
>>>>>> 
>>>>>> Join our online community to chat in real time https://signalwire.community
>>>>>> 
>>>>>> Professional FreeSWITCH Services
>>>>>> sales at freeswitch.com
>>>>>> https://freeswitch.com
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> https://freeswitch.com/oss
>>>>>> https://freeswitch.org/confluence
>>>>>> https://cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> https://freeswitch.com
>>>>> -- 
>>>>> Regards,
>>>>> 
>>>>> David Villasmil
>>>>> email: david.villasmil.work at gmail.com
>>>>> phone: +34669448337
>>>> -- 
>>>> Regards,
>>>> 
>>>> David Villasmil
>>>> email: david.villasmil.work at gmail.com
>>>> phone: +34669448337
>>>> _________________________________________________________________________
>>>> 
>>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>>>> Build your next product on our scalable cloud platform.
>>>> 
>>>> Join our online community to chat in real time https://signalwire.community
>>>> 
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>> 
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>> 
>>> _________________________________________________________________________
>>> 
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>>> Build your next product on our scalable cloud platform.
>>> 
>>> Join our online community to chat in real time https://signalwire.community
>>> 
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>> 
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>> 
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>> Build your next product on our scalable cloud platform.
>> 
>> Join our online community to chat in real time https://signalwire.community
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From william at williamcollsassoc.ca  Sat Jan 25 21:35:43 2020
From: william at williamcollsassoc.ca (William Colls)
Date: Sat, 25 Jan 2020 16:35:43 -0500
Subject: [Freeswitch-users] Grandstream ATA702 - dropped calls
Message-ID: <8d017c74-8e0c-cc44-9509-f857d9c02157@williamcollsassoc.ca>

When I make an outgoing call through this device, the call is often 
disconnected after about 30-40 seconds. Incoming calls are fine. I have 
SIP transport set to UDP, NAT Transversal set to NO, Enable SIP Options 
Keel Alive set to YES, SIP Options Keep Alive Interval set at 30, and 
SIP Options Keep Alive Max Lost at 3. Freeswitch version 1.10.1  was 
started with -nonat. Obviously I have missed, or miss-understood, 
something. Any pointers or suggestions will be most welcome.

Thanks for your time.

William.



From david.villasmil.work at gmail.com  Sat Jan 25 22:17:11 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 25 Jan 2020 22:17:11 +0000
Subject: [Freeswitch-users] Grandstream ATA702 - dropped calls
In-Reply-To: <8d017c74-8e0c-cc44-9509-f857d9c02157@williamcollsassoc.ca>
References: <8d017c74-8e0c-cc44-9509-f857d9c02157@williamcollsassoc.ca>
Message-ID: <CAFGRPVow1_WPDu=9sLh0-JnGEZxbHh=TyPQEaH+8Vef0423gtA@mail.gmail.com>

Have you checked a trace? What's the hangup reason? Calls getting dropped
at 30 seconds usually means there's no 200 OK ACK. Get a trace and verify
that, also check the RTP IPs.

Regards,

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


On Sat, Jan 25, 2020 at 10:01 PM William Colls <william at williamcollsassoc.ca>
wrote:

> When I make an outgoing call through this device, the call is often
> disconnected after about 30-40 seconds. Incoming calls are fine. I have
> SIP transport set to UDP, NAT Transversal set to NO, Enable SIP Options
> Keel Alive set to YES, SIP Options Keep Alive Interval set at 30, and
> SIP Options Keep Alive Max Lost at 3. Freeswitch version 1.10.1  was
> started with -nonat. Obviously I have missed, or miss-understood,
> something. Any pointers or suggestions will be most welcome.
>
> Thanks for your time.
>
> William.
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From thilo at ginkel.com  Mon Jan 27 14:06:04 2020
From: thilo at ginkel.com (Thilo-Alexander Ginkel)
Date: Mon, 27 Jan 2020 15:06:04 +0100
Subject: [Freeswitch-users] Announcing conference participant after PIN entry
Message-ID: <CANvSZQ8oR__E6wGc00T6hiKyCvhLkUwtNgB55+sVpCMpv-fVXQ@mail.gmail.com>

Hello everyone,

let's say, I have a FreeSWITCH conference that is PIN-protected and where I
want to record each participant's name. How do I announce the name of a
participant *after* the caller's PIN has been validated?

So far I tried something like this:

        <extension name="Record Name and schedule conf announce">
          <condition field="destination_number" expression="^34(\d{2})$">
            <action application="answer"/>
            <action application="set" data="namefile=/tmp/${uuid}-name.wav"
inline="true"/>
            <action application="sleep" data="1000"/>
            <action application="playback"
data="voicemail/vm-record_name1.wav"/>
            <action application="playback"
data="tone_stream://%(500,0,500)"/>
            <action application="record" data="${namefile} 5"/>
            <action application="playback"
data="ivr/ivr-call_being_transferred.wav"/>
            <action application="set" data="res=${sched_api +1 none
conference 31$1-${domain_name} play
file_string://${namefile}!conference/conf-has_joined.wav}"/>
            <action application="transfer" data="31$1 XML default"/>
          </condition>
        </extension>

        <extension name="wb_conferences">
          <condition field="destination_number" expression="^(31\d{2})$">
            <action application="answer"/>
            <action application="conference"
data="$1-${domain_name}@wideband${conference_pin}"/>
          </condition>
        </extension>

Participant dials 34xx, name is recorded, playback scheduled, call
transferred to 31xx, which initiates the conference, but sched_api plays
the join message before the pin has been validated.

Is there a way to hook into the conference so that the join message is only
scheduled after the PIN has been validated successfully?

Thanks,
Thilo
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From vineet.verma at bics.com  Mon Jan 27 14:50:49 2020
From: vineet.verma at bics.com (vineet)
Date: Mon, 27 Jan 2020 07:50:49 -0700 (MST)
Subject: Freeswitch Performance issue while RPC request
Message-ID: <1580136649723-0.post@n2.nabble.com>

Dear All,

we are facing freeswitch performance issue
we are sending https RPC request to originate calls from freeswitch.
it seems like after 35 caps/ per second.
freeswitch is choking .
delay keeps on growing and we start receiving Http time out .
some time response is coming after 60 seconds.

Can you please help ?

Thanks,
vineet



--
Sent from: http://freeswitch-users.2379917.n2.nabble.com/


From tom at tomlynn.com  Tue Jan 28 00:56:13 2020
From: tom at tomlynn.com (Tom Lynn)
Date: Mon, 27 Jan 2020 16:56:13 -0800
Subject: [Freeswitch-users] mod_portaudio: pa devlist returns ERR unless FS
	running under root
Message-ID: <CAERdjiaD34Vu-PUNAcUZT1_wK9To6oO9VG+6NZjAAyArFaDWbQ@mail.gmail.com>

I'm having some trouble understanding why the CLI pa command "pa devlist"
returns ERR unless I'm running it under root.  I installed on an ARM single
board computer over the weekend from the ARM repository following the
instructions for raspberry pi.
I can use aplay from the command line to play the music on hold wav files
included with FS.  I've plugged in a USB sound card device and made it the
default sound device and rebooted.  Same result.
The OS image is dietpi and there's no other running processes running that
use sound.  I saw alsactl running and killed it, restarted FS service, but
still no devlist.
Looking back through emails here in the list, there seem to be similar
questions over the years, but with no solve that I found.  Does anybody
have first hand experience with this?  I'm away from home, but I can follow
this up with some console output it if helps.
I'm trying to set up a loudspeaker paging device.

Tom
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From dujinfang at gmail.com  Tue Jan 28 13:23:22 2020
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 28 Jan 2020 21:23:22 +0800
Subject: [Freeswitch-users] Freeswitch Performance issue while RPC
	request
In-Reply-To: <mailman.1041.1580136650.1226.freeswitch-users@lists.freeswitch.org>
References: <mailman.1041.1580136650.1226.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAJs+-5qsb28dTMuz6-U2-0KvwG3BYX9iCA8XKVyEN1xMVKQbnQ@mail.gmail.com>

I had this before and my solution was to fork mod_xml_rpc to mod_xml_rpc2
... 3 .. 4 .. and proxy_pass with nginx at front. It does help.

You could also try
https://freeswitch.org/confluence/display/FREESWITCH/RESTful+Verto see if
it performs better.


On Mon, Jan 27, 2020 at 11:01 PM vineet via FreeSWITCH-users <
freeswitch-users at lists.freeswitch.org> wrote:

>
>
>
> ---------- Forwarded message ----------
> From: vineet <vineet.verma at bics.com>
> To: freeswitch-users at lists.freeswitch.org
> Cc:
> Bcc:
> Date: Mon, 27 Jan 2020 07:50:49 -0700 (MST)
> Subject: Freeswitch Performance issue while RPC request
> Dear All,
>
> we are facing freeswitch performance issue
> we are sending https RPC request to originate calls from freeswitch.
> it seems like after 35 caps/ per second.
> freeswitch is choking .
> delay keeps on growing and we start receiving Http time out .
> some time response is coming after 60 seconds.
>
> Can you please help ?
>
> Thanks,
> vineet
>
>
>
> --
> Sent from: http://freeswitch-users.2379917.n2.nabble.com/
>
>
>
>
> ---------- Forwarded message ----------
> From: vineet via FreeSWITCH-users <freeswitch-users at lists.freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Cc:
> Bcc:
> Date: Mon, 27 Jan 2020 07:01:53 -0800 (PST)
> Subject: [Freeswitch-users] Freeswitch Performance issue while RPC request
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn
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From bilaln018 at gmail.com  Tue Jan 28 18:10:48 2020
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Tue, 28 Jan 2020 23:10:48 +0500
Subject: [Freeswitch-users] [Issue with setting variable while
	bridge/transfer]
In-Reply-To: <CADSVerDwETypFrhL3Ew1j8_deLTKb-yKC=CUWeds+pTmW9ttHA@mail.gmail.com>
References: <CADSVerC+nKtOSmWv0txasH_ftqT4eMJnB4zovv=Foactp_bCYA@mail.gmail.com>
 <CALXCt0pPBXJhd8KzngbAyVWsOKHxyhXMwAVaaG_42YVvk4Qs0Q@mail.gmail.com>
 <CADSVerB=uzzjdGLVk0xn6n7GsSFjkWnE1gSxaTadVbw=ziDFDw@mail.gmail.com>
 <CADSVerAGDpLG+VMm8VR=kBfHZ6ziK05d2VW9-SrdMUSj=ryjew@mail.gmail.com>
 <CAFGRPVrdVXVPWnNb1RTX6M0TouQcz-_1TAUZJOaWT4MvfhwEuQ@mail.gmail.com>
 <CADSVerDwETypFrhL3Ew1j8_deLTKb-yKC=CUWeds+pTmW9ttHA@mail.gmail.com>
Message-ID: <CADSVerBw+fW=_q3Ov5DkasRqAqevFJGeQudu7cs5safHPvqMvw@mail.gmail.com>

Any help with this issue, i tried hiredis as well. Same issue.
Its simple problem i could not figure out.

Regards
Abbasi

On Sat, 4 Jan 2020 at 8:51 PM, Bilal Abbasi <bilaln018 at gmail.com> wrote:

> I tried this, still dec_count and inc_count are not same.
> I got a difference of 14 between them. And i got exact 14error responses
> while transferring the call(looks like api hangup is not executing on
> bridge resulting in error(congestions etc)).
> PFA as dec is less than inc(difference is 14).
>
> Regards
> Abbasi
>
> On Fri, 3 Jan 2020 at 12:14 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Have you tried setting separate variables? I.e: increase  “conn” by 1
>> when connected and increase “disc” by 1 when disconnected.
>>
>> This would help you see if race conditions are messing it up. To get the
>> actual connected you’d simply subtract.
>>
>> On Thu, 2 Jan 2020 at 18:45, Bilal Abbasi <bilaln018 at gmail.com> wrote:
>>
>>> Can anyone help me out with this please.
>>>
>>> Regards
>>> Abbasi
>>>
>>> On Wed, 1 Jan 2020 at 8:45 PM, Bilal Abbasi <bilaln018 at gmail.com> wrote:
>>>
>>>> Hi Marruzelli,
>>>> I am doing exactly same for live calls stats, and those are working
>>>> good. Its only issue when i do that on transfer calls.
>>>> Its not even 10CPS calls.
>>>>
>>>>
>>>> Regards
>>>> Abbasi
>>>>
>>>> On Wed, 1 Jan 2020 at 8:24 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>>> wrote:
>>>>
>>>>> you have probably race conditions
>>>>>
>>>>> eg, more than one call modify same counter
>>>>>
>>>>> maybe you want to keep a uniqe counter for each callid, and then sum
>>>>> them up ?
>>>>>
>>>>> just first thoughts, maybe completely wrong
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Wed, Jan 1, 2020 at 3:17 PM Bilal Abbasi <bilaln018 at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Hi Users,
>>>>>> I have a switch where people are calling and they press 1 to get
>>>>>> transferred to client's DID.
>>>>>> I need to have stats for live total calls on my switch along with
>>>>>> calls that are transferred(realtime stats).
>>>>>> I am doing following commands to get the live total calls.(basically
>>>>>> increasing variable ${CampaignId}_live on getting call and on hangup i
>>>>>> decrement that variable).
>>>>>> <action application="set" data="ignore=${memcache(increment
>>>>>> ${CampaignId}_live 1)}"/>
>>>>>> <action application="set" data="api_hangup_hook=memcache decrement
>>>>>> ${CampaignId}_live 1"/>
>>>>>>
>>>>>> I am doing this while i transfer the calls to client's
>>>>>> number(basically increasing variable ${CampaignId}_transfer on when
>>>>>> somebody presses transfer button and on hangup i decrement that variable).
>>>>>> <action application="bridge" data="{api_hangup_hook='memcache
>>>>>> decrement ${CampaignId}_transfer 1 both',ignore='${memcache(increment
>>>>>> ${CampaignId}_transfer
>>>>>> 1)}'transfer_trigger=1}sofia/external/${TransferTo}"/>
>>>>>>
>>>>>> Now i have live calls stats coming correctly.
>>>>>> But in case of transfer calls, when i have low number of calls(like
>>>>>> 3-4 calls) it works good, but when i have 50-60 calls the stats are not
>>>>>> presented correctly(it does not increase/decrease memcache variable).
>>>>>>
>>>>>> Can someone guide me how to do this in a better way?, or what mistake
>>>>>> i am making in doing it.
>>>>>>
>>>>>> Regards
>>>>>> Abbasi
>>>>>>
>>>>>
>>>>>> _________________________________________________________________________
>>>>>>
>>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>>> https://signalwire.com
>>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>>> services.
>>>>>> Build your next product on our scalable cloud platform.
>>>>>>
>>>>>> Join our online community to chat in real time
>>>>>> https://signalwire.community
>>>>>>
>>>>>> Professional FreeSWITCH Services
>>>>>> sales at freeswitch.com
>>>>>> https://freeswitch.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> https://freeswitch.com/oss
>>>>>> https://freeswitch.org/confluence
>>>>>> https://cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> https://freeswitch.com
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> OpenTelecom.IT
>>>>> cell: +39 347 266 56 18
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>> The FreeSWITCH project is sponsored by SignalWire
>>>>> https://signalwire.com
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>>>> services.
>>>>> Build your next product on our scalable cloud platform.
>>>>>
>>>>> Join our online community to chat in real time
>>>>> https://signalwire.community
>>>>>
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com
>>>>> https://freeswitch.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss
>>>>> https://freeswitch.org/confluence
>>>>> https://cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> https://freeswitch.com
>>>>
>>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> --
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
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From bogdan at opensips.org  Wed Jan 29 08:36:14 2020
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 29 Jan 2020 10:36:14 +0200
Subject: [Freeswitch-users] Invitation for OpenSIPS Summit 2020 Call for
	Paper
Message-ID: <2af18091-fefa-8688-717d-4075f53a4ad4@opensips.org>

Hello fellows VOIPer,

If you want to share with the rest of the VoIP & RTC community some 
news, interesting or breaking through ideas, or even more, some 
experience you had in terms of designing, integrating or operating 
various solutions or platform based on Open Source Softwares, then you 
should consider submitting a paper for the OpenSIPS Summit 2020 in May, 
Amsterdam.

https://www.papercall.io/opensips-summit-2020

We welcome anyone ready to share knowledge!

Best regards,
Bogdan

-- 
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS Summit, Amsterdam, May 2020
   https://www.opensips.org/events/Summit-2020Amsterdam/
OpenSIPS Bootcamp, Miami, March 2020
   https://opensips.org/training/OpenSIPS_Bootcamp_2020/



From mickael at winlux.fr  Wed Jan 29 09:15:05 2020
From: mickael at winlux.fr (Mickael Hubert)
Date: Wed, 29 Jan 2020 10:15:05 +0100
Subject: [Freeswitch-users] Limit cps per source IP and port
In-Reply-To: <mailman.2048.1563829398.1387.freeswitch-users@lists.freeswitch.org>
References: <mailman.2048.1563829398.1387.freeswitch-users@lists.freeswitch.org>
Message-ID: <CA+fRVSWs2k3xWYyCQA7YK-Fc14gk+XbxkNuCWMGo=UPd+fSZ=Q@mail.gmail.com>

Maybe it's very late,
but I use this:
https://github.com/Mickaelh51/freeswitch/blob/master/pingcalls_limit.xml
it's not per IP, but it's easy to change this code

Le lun. 22 juil. 2019 à 23:03, António Silva via FreeSWITCH-users <
freeswitch-users at lists.freeswitch.org> a écrit :

>
>
>
> ---------- Forwarded message ----------
> From: "António Silva" <asilva at wirelessmundi.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Thu, 18 Jul 2019 10:52:14 +0200
> Subject: Limit cps per source IP and port
> Hi,
>
> is it possible to limit the number of calls per source ip and port
> before the dialplan?
>
> I know that this can be done using mod_limit in the dialplan, i was
> thinking of a limit like the sessions-per-second global limit, this way
> less resources are used if some ip/port is flooding our machine.
>
> Another possibility is using the firewall... but i like this option less
> because i wont know of the drop calls.
>
> Thanks,
>
> --
> Saludos / Regards / Cumprimentos
> António Silva
>
>
>
>
>
> ---------- Forwarded message ----------
> From: "António Silva via FreeSWITCH-users" <
> freeswitch-users at lists.freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Mon, 22 Jul 2019 14:03:23 -0700 (PDT)
> Subject: [Freeswitch-users] Limit cps per source IP and port
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
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From asilva at wirelessmundi.com  Wed Jan 29 10:09:37 2020
From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=)
Date: Wed, 29 Jan 2020 11:09:37 +0100
Subject: [Freeswitch-users] Limit cps per source IP and port
In-Reply-To: <CA+fRVSWs2k3xWYyCQA7YK-Fc14gk+XbxkNuCWMGo=UPd+fSZ=Q@mail.gmail.com>
References: <mailman.2048.1563829398.1387.freeswitch-users@lists.freeswitch.org>
 <CA+fRVSWs2k3xWYyCQA7YK-Fc14gk+XbxkNuCWMGo=UPd+fSZ=Q@mail.gmail.com>
Message-ID: <7e6569b2-6886-1c42-26af-63f0af354fbe@wirelessmundi.com>

Hi Michael,

Thanks a lot for the reply.  I was searching of something at sofia 
level, before it  enters the xml routing decision.

We already have the parameters sessions-per-second and max-sessions, but 
is for the entire switch, my idea has to have the same type of 
parameters per account/source ip, but i is not implemented. i've a draft 
on a lab machine where i'm trying to implement this but for now is not 
good...

Right now I'm using iptables to limit the maximum of new invites to FS, 
and it keeps me "safe" and i can limit resources per source IP.


On 29/01/2020 10:15, Mickael Hubert wrote:
> Maybe it's very late,
> but I use this: 
> https://github.com/Mickaelh51/freeswitch/blob/master/pingcalls_limit.xml
> it's not per IP, but it's easy to change this code
>
> Le lun. 22 juil. 2019 à 23:03, António Silva via FreeSWITCH-users 
> <freeswitch-users at lists.freeswitch.org 
> <mailto:freeswitch-users at lists.freeswitch.org>> a écrit :
>
>
>
>
>     ---------- Forwarded message ----------
>     From: "António Silva" <asilva at wirelessmundi.com
>     <mailto:asilva at wirelessmundi.com>>
>     To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     Cc:
>     Bcc:
>     Date: Thu, 18 Jul 2019 10:52:14 +0200
>     Subject: Limit cps per source IP and port
>     Hi,
>
>     is it possible to limit the number of calls per source ip and port
>     before the dialplan?
>
>     I know that this can be done using mod_limit in the dialplan, i was
>     thinking of a limit like the sessions-per-second global limit,
>     this way
>     less resources are used if some ip/port is flooding our machine.
>
>     Another possibility is using the firewall... but i like this
>     option less
>     because i wont know of the drop calls.
>
>     Thanks,
>
>     -- 
>     Saludos / Regards / Cumprimentos
>     António Silva
>
>
>
>
>
>     ---------- Forwarded message ----------
>     From: "António Silva via FreeSWITCH-users"
>     <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     Cc:
>     Bcc:
>     Date: Mon, 22 Jul 2019 14:03:23 -0700 (PDT)
>     Subject: [Freeswitch-users] Limit cps per source IP and port
>     _________________________________________________________________________
>
>     The FreeSWITCH project is sponsored by SignalWire
>     https://signalwire.com
>     Enhance your FreeSWITCH install with disruptive priced SMS and
>     PSTN services.
>     Build your next product on our scalable cloud platform.
>
>     Join our online community to chat in real time
>     https://signalwire.community
>
>     Professional FreeSWITCH Services
>     sales at freeswitch.com <mailto:sales at freeswitch.com>
>     https://freeswitch.com
>
>     Official FreeSWITCH Sites
>     https://freeswitch.com/oss
>     https://freeswitch.org/confluence
>     https://cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     https://freeswitch.com
>
-- 
Saludos / Regards / Cumprimentos
António Silva

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From david.villasmil.work at gmail.com  Wed Jan 29 11:08:30 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 29 Jan 2020 11:08:30 +0000
Subject: [Freeswitch-users] Limit cps per source IP and port
In-Reply-To: <mailman.1069.1580292611.1226.freeswitch-users@lists.freeswitch.org>
References: <mailman.2048.1563829398.1387.freeswitch-users@lists.freeswitch.org>
 <CA+fRVSWs2k3xWYyCQA7YK-Fc14gk+XbxkNuCWMGo=UPd+fSZ=Q@mail.gmail.com>
 <mailman.1069.1580292611.1226.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAFGRPVoPGb4HODA9Z3PO-izQKM5PgMqpnassi1ocoY-F9c8C+A@mail.gmail.com>

Can you share your potables? Might be interesting for the the community.

On Wed, 29 Jan 2020 at 10:32, António Silva via FreeSWITCH-users <
freeswitch-users at lists.freeswitch.org> wrote:

>
>
>
> ---------- Forwarded message ----------
> From: "António Silva" <asilva at wirelessmundi.com>
> To: Mickael Hubert <mickael at winlux.fr>, FreeSWITCH Users Help <
> freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Wed, 29 Jan 2020 11:09:37 +0100
> Subject: Re: [Freeswitch-users] Limit cps per source IP and port
>
> Hi Michael,
>
> Thanks a lot for the reply.  I was searching of something at sofia level,
> before it  enters the xml routing decision.
>
> We already have the parameters sessions-per-second and max-sessions, but
> is for the entire switch, my idea has to have the same type of parameters
> per account/source ip, but i is not implemented. i've a draft on a lab
> machine where i'm trying to implement this but for now is not good...
>
> Right now I'm using iptables to limit the maximum of new invites to FS,
> and it keeps me "safe" and i can limit resources per source IP.
>
>
> On 29/01/2020 10:15, Mickael Hubert wrote:
>
> Maybe it's very late,
> but I use this:
> https://github.com/Mickaelh51/freeswitch/blob/master/pingcalls_limit.xml
> it's not per IP, but it's easy to change this code
>
> Le lun. 22 juil. 2019 à 23:03, António Silva via FreeSWITCH-users <
> freeswitch-users at lists.freeswitch.org> a écrit :
>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: "António Silva" <asilva at wirelessmundi.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Bcc:
>> Date: Thu, 18 Jul 2019 10:52:14 +0200
>> Subject: Limit cps per source IP and port
>> Hi,
>>
>> is it possible to limit the number of calls per source ip and port
>> before the dialplan?
>>
>> I know that this can be done using mod_limit in the dialplan, i was
>> thinking of a limit like the sessions-per-second global limit, this way
>> less resources are used if some ip/port is flooding our machine.
>>
>> Another possibility is using the firewall... but i like this option less
>> because i wont know of the drop calls.
>>
>> Thanks,
>>
>> --
>> Saludos / Regards / Cumprimentos
>> António Silva
>>
>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: "António Silva via FreeSWITCH-users" <
>> freeswitch-users at lists.freeswitch.org>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Cc:
>> Bcc:
>> Date: Mon, 22 Jul 2019 14:03:23 -0700 (PDT)
>> Subject: [Freeswitch-users] Limit cps per source IP and port
>> _________________________________________________________________________
>>
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>> services.
>> Build your next product on our scalable cloud platform.
>>
>> Join our online community to chat in real time
>> https://signalwire.community
>>
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> --
> Saludos / Regards / Cumprimentos
> António Silva
>
>
>
>
> ---------- Forwarded message ----------
> From: "António Silva via FreeSWITCH-users" <
> freeswitch-users at lists.freeswitch.org>
> To: Mickael Hubert <mickael at winlux.fr>, FreeSWITCH Users Help <
> freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Wed, 29 Jan 2020 02:32:08 -0800 (PST)
> Subject: Re: [Freeswitch-users] Limit cps per source IP and port
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From davidswalkabout at gmail.com  Wed Jan 29 17:21:59 2020
From: davidswalkabout at gmail.com (David P)
Date: Wed, 29 Jan 2020 09:21:59 -0800
Subject: [Freeswitch-users] Detecting chosen devices in verto
Message-ID: <CAE23trvWk0wmWjNkxCxKWsZHbeQN5q0=R8c7Vd7OWN3+JfqssQ@mail.gmail.com>

How can a verto-based app report which devices verto has chosen to use, esp
the mic?

If it's possible that verto changes its choice during a call, perhaps due
to a device being just plugged in, how can the app detect this?
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From napole at gmail.com  Wed Jan 29 19:26:01 2020
From: napole at gmail.com (Mitchell Langs)
Date: Wed, 29 Jan 2020 20:26:01 +0100
Subject: [Freeswitch-users] spandsp detects tone only while sleeping?
Message-ID: <CADmXuNTMkpNED8E9V6oOzCk8Rwuabn-Ge8xULRq-W_7zMzSdZw@mail.gmail.com>

I want to originate a call and do something as soon as the busy tone
is detected. I use this lua script but nothing is detected:

session = freeswitch.Session("{codec_string=PCMA}sofia/external/ringbusyde at 192.168.2.4:5080");
consumer = freeswitch.EventConsumer("DETECTED_TONE");
session:execute("spandsp_start_tone_detect", "49");
event = consumer:pop(1,6000);

However when I sleep after spandsp_start_tone_detect the tone is
detected properly:

session = freeswitch.Session("{codec_string=PCMA}sofia/external/ringbusyde at 192.168.2.4:5080");
consumer = freeswitch.EventConsumer("DETECTED_TONE");
session:execute("spandsp_start_tone_detect", "49");
session:execute("sleep", "2000");
event = consumer:pop(1,6000);

I'd like to understand why it is not working in the first case.


log output if needed:
https://pastebin.com/CnFsKNsJ


From spencer.angerbauer at gmail.com  Fri Jan 31 00:33:26 2020
From: spencer.angerbauer at gmail.com (Spencer Angerbauer)
Date: Thu, 30 Jan 2020 17:33:26 -0700
Subject: [Freeswitch-users] Requiring Pin of "1" for all conferences +
 announcement
In-Reply-To: <83DE024C-2765-4E87-84FD-CBA5F8479A3D@gmail.com>
References: <45214DC2-EBE2-4D6F-8C0F-BD7EF556D1E6@ventureslopes.com>
 <D3FC2D27-F683-41C3-8B98-9D42D2EF0BF7@gmail.com>
 <B395AF32-101C-42EA-9FB1-C072D09ACB0D@gmail.com>
 <CAFGRPVpoWGRRgmpjt9M46DfkYi2RQg7SvNYzOg__xNT=d6zCDQ@mail.gmail.com>
 <CAFGRPVrFuD0-Cs8DOH1qrN+a+GgXsDH5kg=qK7PnNgdBDc5G_w@mail.gmail.com>
 <7E0EFC95-8F9C-4009-A0B7-10B224049FC2@gmail.com>
 <83DE024C-2765-4E87-84FD-CBA5F8479A3D@gmail.com>
Message-ID: <437549DE-925D-45ED-A370-B2CB681C56C9@gmail.com>

Just following up after troubleshooting a variety of items… is there an easier way to hard code the XML to require PIN “1” in the conference.conf.xml files for every and all conference calls on the freeswitch pbx?  It seems the issue is requiring a PIN on a web api call or an originate call from the console. I am simply trying to hard set Pins for all conferences to be “1” for testing purposes.  Any help would be greatly appreciated. Am I missing an additional required parameter?

Thanks,

-Spence

> On Jan 23, 2020, at 10:28 PM, Spencer Angerbauer <spencer.angerbauer at gmail.com> wrote:
> 
> Ive also update all conference.con.xml to include the following:
> 
>  <!-- Conference pin -->
>       <param name="pin" value="1"/>
>       <param name="moderator-pin" value="1”/>
> 
> 
> 
> But still no luck on pin being required upon entrance into a conference...
> 
> 
> 
>> On Jan 23, 2020, at 5:39 PM, Spencer Angerbauer <spencer.angerbauer at gmail.com <mailto:spencer.angerbauer at gmail.com>> wrote:
>> 
>> Yes, I have tried that and unfortunately the bridge just hangs up right after it does an outgoing connection...
>> 
>>> On Jan 23, 2020, at 5:19 PM, David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
>>> 
>>> Have you tried “+1”?
>>> 
>>> Like
>>> 
>>> http://<<servername>>:8080/webapi/conference%2B1?<<friendlyname>>
>>> 
>>> 
>>> On Fri, 24 Jan 2020 at 00:07, David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
>>> Why don’t you originate and send to XML XXXX where XXXX is an extension and on the extension you do first the announcement and gather the digit and if it is 1 then send to the conference?
>>> 
>>> On Thu, 23 Jan 2020 at 19:26, Spencer Angerbauer <spencer.angerbauer at gmail.com <mailto:spencer.angerbauer at gmail.com>> wrote:
>>> Hello Freeswitch Pros,
>>> 
>>> I just wanted to see if by chance anyone had any suggestions on the below info and issue.  Thank you so much for your feedback and direction as it is greatly appreciated!
>>> 
>>> Thank you,
>>> 
>>> -Spence
>>> 
>>>> On Jan 19, 2020, at 3:54 PM, Spencer Angerbauer <spencer.angerbauer at gmail.com <mailto:spencer.angerbauer at gmail.com>> wrote:
>>>> 
>>>> I am trying to get all conferences to say a phrase before entering the pin #1 prior to entering (as we are connecting via outbound dialer from api). I have installed TTS to do a “say” function, and have updated the conf/autoload_configs/conference.conf.xml by making the “pin” required and to require pin as “1”, but every time a caller connects, it just transfers them directly into the conference without announcing that they need to press “1” to enter, nor does it require a “pin” to connect to the conference.
>>>> 
>>>> Is there somewhere else I need to update the dialing plan to require all my conference calls require a “1” when using the web api outbound dialer to connect everyone?  (I am using http://<<servername>>:8080/webapi/originate?sofia/gateway/signalwire/<<phone number>>%20&conference(<<conference name>>) <http://api.joonto.com:8080/webapi/originate?sofia/gateway/signalwire/18015737111%20&conference(1234)>to connect multiple calls together).
>>>> 
>>>> Also, is there another location to add speak to text for the beginning of every conference from webapi or does it all need to be handled by the dial plan?
>>> 
>>>> 
>>>> 
>>>> Thank you for your help on both these related issues.
>>>> 
>>>> -Spence
>>>> 
>>> 
>>> _________________________________________________________________________
>>> 
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>>> Build your next product on our scalable cloud platform.
>>> 
>>> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
>>> 
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>>> https://freeswitch.com <https://freeswitch.com/>
>>> 
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>>> https://cluecon.com <https://cluecon.com/>
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> https://freeswitch.com <https://freeswitch.com/>-- 
>>> Regards,
>>> 
>>> David Villasmil
>>> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
>>> phone: +34669448337
>>> -- 
>>> Regards,
>>> 
>>> David Villasmil
>>> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
>>> phone: +34669448337
>>> _________________________________________________________________________
>>> 
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>>> Build your next product on our scalable cloud platform.
>>> 
>>> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
>>> 
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>>> https://freeswitch.com <https://freeswitch.com/>
>>> 
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>>> https://cluecon.com <https://cluecon.com/>
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> https://freeswitch.com <https://freeswitch.com/>
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>> Build your next product on our scalable cloud platform.
>> 
>> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>> https://freeswitch.com
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
> 
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

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From nneul at mst.edu  Fri Jan 31 02:02:57 2020
From: nneul at mst.edu (Nathan Neulinger)
Date: Thu, 30 Jan 2020 20:02:57 -0600
Subject: [Freeswitch-users] Requiring Pin of "1" for all conferences +
 announcement
In-Reply-To: <437549DE-925D-45ED-A370-B2CB681C56C9@gmail.com>
References: <45214DC2-EBE2-4D6F-8C0F-BD7EF556D1E6@ventureslopes.com>
 <D3FC2D27-F683-41C3-8B98-9D42D2EF0BF7@gmail.com>
 <B395AF32-101C-42EA-9FB1-C072D09ACB0D@gmail.com>
 <CAFGRPVpoWGRRgmpjt9M46DfkYi2RQg7SvNYzOg__xNT=d6zCDQ@mail.gmail.com>
 <CAFGRPVrFuD0-Cs8DOH1qrN+a+GgXsDH5kg=qK7PnNgdBDc5G_w@mail.gmail.com>
 <7E0EFC95-8F9C-4009-A0B7-10B224049FC2@gmail.com>
 <83DE024C-2765-4E87-84FD-CBA5F8479A3D@gmail.com>
 <437549DE-925D-45ED-A370-B2CB681C56C9@gmail.com>
Message-ID: <2183e072-51ea-e9fb-c638-e6572e7793c1@mst.edu>

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From spencer.angerbauer at gmail.com  Fri Jan 31 08:16:55 2020
From: spencer.angerbauer at gmail.com (Spencer Angerbauer)
Date: Fri, 31 Jan 2020 01:16:55 -0700
Subject: [Freeswitch-users] Requiring Pin of "1" for all conferences +
 announcement
In-Reply-To: <2183e072-51ea-e9fb-c638-e6572e7793c1@mst.edu>
References: <45214DC2-EBE2-4D6F-8C0F-BD7EF556D1E6@ventureslopes.com>
 <D3FC2D27-F683-41C3-8B98-9D42D2EF0BF7@gmail.com>
 <B395AF32-101C-42EA-9FB1-C072D09ACB0D@gmail.com>
 <CAFGRPVpoWGRRgmpjt9M46DfkYi2RQg7SvNYzOg__xNT=d6zCDQ@mail.gmail.com>
 <CAFGRPVrFuD0-Cs8DOH1qrN+a+GgXsDH5kg=qK7PnNgdBDc5G_w@mail.gmail.com>
 <7E0EFC95-8F9C-4009-A0B7-10B224049FC2@gmail.com>
 <83DE024C-2765-4E87-84FD-CBA5F8479A3D@gmail.com>
 <437549DE-925D-45ED-A370-B2CB681C56C9@gmail.com>
 <2183e072-51ea-e9fb-c638-e6572e7793c1@mst.edu>
Message-ID: <A07B3996-77CB-4087-A394-1F9BB4E2F1E2@gmail.com>

Thanks Nathan for the quick reply.  We are creating our conferences through the webapi and triggering a web request for each number joining a particular call with the same conference ID… so for example, if we were bridging 3 calls, we are sending a GET post to the API sequentially by the following example phone numbers:

http://12.34.56.78:8080/webapi/originate?sofia/gateway/signalwire/15555555555%20&conference(1234) <http://12.34.56.78:8080/webapi/originate?sofia/gateway/signalwire/15555555555%20&conference(1234)>
http://12.34.56.78:8080/webapi/originate?sofia/gateway/signalwire/17777777777%20&conference(1234) <http://12.34.56.78:8080/webapi/originate?sofia/gateway/signalwire/17777777777%20&conference(1234)>
http://12.34.56.78:8080/webapi/originate?sofia/gateway/signalwire/18888888888%20&conference(1234) <http://12.34.56.78:8080/webapi/originate?sofia/gateway/signalwire/18888888888%20&conference(1234)>


We’ve tried to pass the pin via the webapi post without luck, and as mentioned below have tried to even hard-set the xml in conference.conf.xml to hardcoded to “1” as the PIN with no luck.

Is there a way to require the pin from the Webapi example above or even from the originate example here?:

originate sofia/gateway/signalwire/15555555555 &conference(1234)

Thank you again Nathan for your quick response today!

-Spence


> On Jan 30, 2020, at 7:02 PM, Nathan Neulinger <nneul at mst.edu> wrote:
> 
> How are you creating your conference?
> 
> If you have that pin setting in the conference profile named "mysettings", you should be able to create your conference using the "conference" application specifying that profile.
> 
> i.e.
> 
> <action application="conference" data="myconfname at mysettings" />
> 
> or you can just do it on that create
> 
> <action application="conference" data="myconfname+1" />
> 
> 
> See docs here: https://freeswitch.org/confluence/display/FREESWITCH/mod_conference <https://freeswitch.org/confluence/display/FREESWITCH/mod_conference>
> 
> -- Nathan
> 
> From: Spencer Angerbauer [mailto:spencer.angerbauer at gmail.com <mailto:spencer.angerbauer at gmail.com>]
> Sent: Thursday, January 30, 2020, 6:33 PM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> <mailto:freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Requiring Pin of "1" for all conferences + announcement
> 
>> Just following up after troubleshooting a variety of items… is there an easier way to hard code the XML to require PIN “1” in the conference.conf.xml files for every and all conference calls on the freeswitch pbx?  It seems the issue is requiring a PIN on a web api call or an originate call from the console. I am simply trying to hard set Pins for all conferences to be “1” for testing purposes.  Any help would be greatly appreciated. Am I missing an additional required parameter?
>> 
>> Thanks,
>> 
>> -Spence
>> 
>>> On Jan 23, 2020, at 10:28 PM, Spencer Angerbauer <spencer.angerbauer at gmail.com <mailto:spencer.angerbauer at gmail.com>> wrote:
>>> 
>>> Ive also update all conference.con.xml to include the following:
>>> 
>>>  <!-- Conference pin -->
>>>       <param name="pin" value="1"/>
>>>       <param name="moderator-pin" value="1”/>
>>> 
>>> 
>>> 
>>> But still no luck on pin being required upon entrance into a conference...
>>> 
>>> 
>>> 
>>>> On Jan 23, 2020, at 5:39 PM, Spencer Angerbauer <spencer.angerbauer at gmail.com <mailto:spencer.angerbauer at gmail.com>> wrote:
>>>> 
>>>> Yes, I have tried that and unfortunately the bridge just hangs up right after it does an outgoing connection...
>>>> 
>>>>> On Jan 23, 2020, at 5:19 PM, David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
>>>>> 
>>>>> Have you tried “+1”?
>>>>> 
>>>>> Like
>>>>> 
>>>>> http://<<servername>>:8080/webapi/conference%2B1?<<friendlyname>>
>>>>> 
>>>>> 
>>>>> On Fri, 24 Jan 2020 at 00:07, David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
>>>>> Why don’t you originate and send to XML XXXX where XXXX is an extension and on the extension you do first the announcement and gather the digit and if it is 1 then send to the conference?
>>>>> 
>>>>> On Thu, 23 Jan 2020 at 19:26, Spencer Angerbauer <spencer.angerbauer at gmail.com <mailto:spencer.angerbauer at gmail.com>> wrote:
>>>>> Hello Freeswitch Pros,
>>>>> 
>>>>> I just wanted to see if by chance anyone had any suggestions on the below info and issue.  Thank you so much for your feedback and direction as it is greatly appreciated!
>>>>> 
>>>>> Thank you,
>>>>> 
>>>>> -Spence
>>>>> 
>>>>>> On Jan 19, 2020, at 3:54 PM, Spencer Angerbauer <spencer.angerbauer at gmail.com <mailto:spencer.angerbauer at gmail.com>> wrote:
>>>>>> 
>>>>>> I am trying to get all conferences to say a phrase before entering the pin #1 prior to entering (as we are connecting via outbound dialer from api). I have installed TTS to do a “say” function, and have updated the conf/autoload_configs/conference.conf.xml by making the “pin” required and to require pin as “1”, but every time a caller connects, it just transfers them directly into the conference without announcing that they need to press “1” to enter, nor does it require a “pin” to connect to the conference.
>>>>>> 
>>>>>> Is there somewhere else I need to update the dialing plan to require all my conference calls require a “1” when using the web api outbound dialer to connect everyone?  (I am using http://<<servername>>:8080/webapi/originate?sofia/gateway/signalwire/<<phone number>>%20&conference(<<conference name>>) <http://api.joonto.com:8080/webapi/originate?sofia/gateway/signalwire/18015737111%20&conference(1234)>to connect multiple calls together).
>>>>>> 
>>>>>> Also, is there another location to add speak to text for the beginning of every conference from webapi or does it all need to be handled by the dial plan?
>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Thank you for your help on both these related issues.
>>>>>> 
>>>>>> -Spence
>>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> 
>>>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>>>>> Build your next product on our scalable cloud platform.
>>>>> 
>>>>> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
>>>>> 
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>>>>> https://freeswitch.com <https://freeswitch.com/>
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>>>>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>>>>> https://cluecon.com <https://cluecon.com/>
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> https://freeswitch.com <https://freeswitch.com/>-- 
>>>>> Regards,
>>>>> 
>>>>> David Villasmil
>>>>> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
>>>>> phone: +34669448337
>>>>> -- 
>>>>> Regards,
>>>>> 
>>>>> David Villasmil
>>>>> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
>>>>> phone: +34669448337
>>>>> _________________________________________________________________________
>>>>> 
>>>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>>>>> Build your next product on our scalable cloud platform.
>>>>> 
>>>>> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
>>>>> 
>>>>> Professional FreeSWITCH Services
>>>>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>>>>> https://freeswitch.com <https://freeswitch.com/>
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>>>>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>>>>> https://cluecon.com <https://cluecon.com/>
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> https://freeswitch.com <https://freeswitch.com/>
>>>> _________________________________________________________________________
>>>> 
>>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>>>> Build your next product on our scalable cloud platform.
>>>> 
>>>> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
>>>> 
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>>>> https://freeswitch.com <https://freeswitch.com/>
>>>> 
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>>>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>>>> https://cluecon.com <https://cluecon.com/>
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> https://freeswitch.com <https://freeswitch.com/>
>>> _________________________________________________________________________
>>> 
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>>> Build your next product on our scalable cloud platform.
>>> 
>>> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
>>> 
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>>> https://freeswitch.com <https://freeswitch.com/>
>>> 
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>>> https://cluecon.com <https://cluecon.com/>
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> 
>> _________________________________________________________________________
>> 
>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/>
>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
>> Build your next product on our scalable cloud platform.
>> 
>> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/>
>> 
>> Professional FreeSWITCH Services
>> sales at freeswitch.com <mailto:sales at freeswitch.com>
>> https://freeswitch.com <https://freeswitch.com/>
>> 
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss <https://freeswitch.com/oss>
>> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>> https://cluecon.com <https://cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> https://freeswitch.com <https://freeswitch.com/>-- 
> 
> ------------------------------------------------------------
> Nathan Neulinger                            nneul at mst.edu <mailto:nneul at mst.edu>
> Missouri S&T Information Technology         (573) 341-6679
> System Administrator - Architect            (573) 612-1412
> System and Desktop Infrastructure Team Manager

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From david.villasmil.work at gmail.com  Fri Jan 31 10:02:14 2020
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 31 Jan 2020 10:02:14 +0000
Subject: [Freeswitch-users] Requiring Pin of "1" for all conferences +
	announcement
In-Reply-To: <A07B3996-77CB-4087-A394-1F9BB4E2F1E2@gmail.com>
References: <45214DC2-EBE2-4D6F-8C0F-BD7EF556D1E6@ventureslopes.com>
 <D3FC2D27-F683-41C3-8B98-9D42D2EF0BF7@gmail.com>
 <B395AF32-101C-42EA-9FB1-C072D09ACB0D@gmail.com>
 <CAFGRPVpoWGRRgmpjt9M46DfkYi2RQg7SvNYzOg__xNT=d6zCDQ@mail.gmail.com>
 <CAFGRPVrFuD0-Cs8DOH1qrN+a+GgXsDH5kg=qK7PnNgdBDc5G_w@mail.gmail.com>
 <7E0EFC95-8F9C-4009-A0B7-10B224049FC2@gmail.com>
 <83DE024C-2765-4E87-84FD-CBA5F8479A3D@gmail.com>
 <437549DE-925D-45ED-A370-B2CB681C56C9@gmail.com>
 <2183e072-51ea-e9fb-c638-e6572e7793c1@mst.edu>
 <A07B3996-77CB-4087-A394-1F9BB4E2F1E2@gmail.com>
Message-ID: <CAFGRPVpNc05C_b7HUy_Shhx-vcAkCW8GuCQuKLza0gt9POjg8g@mail.gmail.com>

Does that api request treats & as fs or as a variable separator? I
wonder... never used the  web api, though.

I think you’d be better off using ESL.

Regards,

David

On Fri, 31 Jan 2020 at 08:48, Spencer Angerbauer <
spencer.angerbauer at gmail.com> wrote:

> Thanks Nathan for the quick reply.  We are creating our conferences
> through the webapi and triggering a web request for each number joining a
> particular call with the same conference ID… so for example, if we were
> bridging 3 calls, we are sending a GET post to the API sequentially by the
> following example phone numbers:
>
>
> http://12.34.56.78:8080/webapi/originate?sofia/gateway/signalwire/15555555555%20&conference
> (1234)
>
> http://12.34.56.78:8080/webapi/originate?sofia/gateway/signalwire/17777777777%20&conference
> (1234)
>
> http://12.34.56.78:8080/webapi/originate?sofia/gateway/signalwire/18888888888%20&conference
> (1234)
>
>
> We’ve tried to pass the pin via the webapi post without luck, and as
> mentioned below have tried to even hard-set the xml in conference.conf.xml
> to hardcoded to “1” as the PIN with no luck.
>
> Is there a way to require the pin from the Webapi example above or even
> from the originate example here?:
>
> originate sofia/gateway/signalwire/15555555555 &conference(1234)
>
> Thank you again Nathan for your quick response today!
>
> -Spence
>
>
> On Jan 30, 2020, at 7:02 PM, Nathan Neulinger <nneul at mst.edu> wrote:
>
> How are you creating your conference?
>
> If you have that pin setting in the conference profile named "mysettings",
> you should be able to create your conference using the "conference"
> application specifying that profile.
>
> i.e.
>
> <action application="conference" data="myconfname at mysettings" />
>
> or you can just do it on that create
>
> <action application="conference" data="myconfname+1" />
>
>
> See docs here:
> https://freeswitch.org/confluence/display/FREESWITCH/mod_conference
>
> -- Nathan
>
> ------------------------------
> *From:* Spencer Angerbauer [mailto:spencer.angerbauer at gmail.com
> <spencer.angerbauer at gmail.com>]
> *Sent:* Thursday, January 30, 2020, 6:33 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Requiring Pin of "1" for all conferences +
> announcement
>
> Just following up after troubleshooting a variety of items… is there an
> easier way to hard code the XML to require PIN “1” in the
> conference.conf.xml files for every and all conference calls on the
> freeswitch pbx?  It seems the issue is requiring a PIN on a web api call or
> an originate call from the console. I am simply trying to hard set Pins for
> all conferences to be “1” for testing purposes.  Any help would be greatly
> appreciated. Am I missing an additional required parameter?
>
> Thanks,
>
> -Spence
>
> On Jan 23, 2020, at 10:28 PM, Spencer Angerbauer <
> spencer.angerbauer at gmail.com> wrote:
>
> Ive also update all conference.con.xml to include the following:
>
>  <!-- Conference pin -->
>       <param name="pin" value="1"/>
>       <param name="moderator-pin" value="1”/>
>
>
>
> But still no luck on pin being required upon entrance into a conference...
>
>
>
> On Jan 23, 2020, at 5:39 PM, Spencer Angerbauer <
> spencer.angerbauer at gmail.com> wrote:
>
> Yes, I have tried that and unfortunately the bridge just hangs up right
> after it does an outgoing connection...
>
> On Jan 23, 2020, at 5:19 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
> Have you tried “+1”?
>
> Like
>
> http://<<servername>>:8080/webapi/conference%2B1?<<friendlyname>>
>
>
> On Fri, 24 Jan 2020 at 00:07, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Why don’t you originate and send to XML XXXX where XXXX is an extension
>> and on the extension you do first the announcement and gather the digit and
>> if it is 1 then send to the conference?
>>
>> On Thu, 23 Jan 2020 at 19:26, Spencer Angerbauer <
>> spencer.angerbauer at gmail.com> wrote:
>>
>>> Hello Freeswitch Pros,
>>>
>>> I just wanted to see if by chance anyone had any suggestions on the
>>> below info and issue.  Thank you so much for your feedback and direction as
>>> it is greatly appreciated!
>>>
>>> Thank you,
>>>
>>> -Spence
>>>
>>> On Jan 19, 2020, at 3:54 PM, Spencer Angerbauer <
>>> spencer.angerbauer at gmail.com> wrote:
>>>
>>> I am trying to get all conferences to say a phrase before entering the
>>> pin #1 prior to entering (as we are connecting via outbound dialer from
>>> api). I have installed TTS to do a “say” function, and have updated the
>>> conf/autoload_configs/conference.conf.xml by making the “pin” required
>>> and to require pin as “1”, but every time a caller connects, it just
>>> transfers them directly into the conference without announcing that they
>>> need to press “1” to enter, nor does it require a “pin” to connect to the
>>> conference.
>>>
>>> Is there somewhere else I need to update the dialing plan to require all
>>> my conference calls require a “1” when using the web api outbound dialer to
>>> connect everyone?  (I am using http://<<servername>>:8080/webapi/originate?sofia/gateway/signalwire/<<phone
>>> number>>%20&conference(<<conference name>>)
>>> <http://api.joonto.com:8080/webapi/originate?sofia/gateway/signalwire/18015737111%20&conference(1234)>to connect
>>> multiple calls together).
>>>
>>>
>>> Also, is there another location to add speak to text for the beginning
>>> of every conference from webapi or does it all need to be handled by the
>>> dial plan?
>>>
>>>
>>>
>>> Thank you for your help on both these related issues.
>>>
>>> -Spence
>>>
>>>
>>> _________________________________________________________________________
>>>
>>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
>>> services.
>>> Build your next product on our scalable cloud platform.
>>>
>>> Join our online community to chat in real time
>>> https://signalwire.community
>>>
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> --
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time https://signalwire.community
>
> Professional FreeSWITCH 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
>
> --
>
> ------------------------------------------------------------
> Nathan Neulinger                            nneul at mst.edu
> Missouri S&T Information Technology         (573) 341-6679
> System Administrator - Architect            (573) 612-1412
> System and Desktop Infrastructure Team Manager
>
>
> _________________________________________________________________________
>
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN
> services.
> Build your next product on our scalable cloud platform.
>
> Join our online community to chat in real time
> https://signalwire.community
>
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From nneul at mst.edu  Fri Jan 31 13:06:46 2020
From: nneul at mst.edu (Nathan Neulinger)
Date: Fri, 31 Jan 2020 07:06:46 -0600
Subject: [Freeswitch-users] Requiring Pin of "1" for all conferences +
 announcement
In-Reply-To: <CAFGRPVpNc05C_b7HUy_Shhx-vcAkCW8GuCQuKLza0gt9POjg8g@mail.gmail.com>
References: <45214DC2-EBE2-4D6F-8C0F-BD7EF556D1E6@ventureslopes.com>
 <D3FC2D27-F683-41C3-8B98-9D42D2EF0BF7@gmail.com>
 <B395AF32-101C-42EA-9FB1-C072D09ACB0D@gmail.com>
 <CAFGRPVpoWGRRgmpjt9M46DfkYi2RQg7SvNYzOg__xNT=d6zCDQ@mail.gmail.com>
 <CAFGRPVrFuD0-Cs8DOH1qrN+a+GgXsDH5kg=qK7PnNgdBDc5G_w@mail.gmail.com>
 <7E0EFC95-8F9C-4009-A0B7-10B224049FC2@gmail.com>
 <83DE024C-2765-4E87-84FD-CBA5F8479A3D@gmail.com>
 <437549DE-925D-45ED-A370-B2CB681C56C9@gmail.com>
 <2183e072-51ea-e9fb-c638-e6572e7793c1@mst.edu>
 <A07B3996-77CB-4087-A394-1F9BB4E2F1E2@gmail.com>
 <CAFGRPVpNc05C_b7HUy_Shhx-vcAkCW8GuCQuKLza0gt9POjg8g@mail.gmail.com>
Message-ID: <38f2a340-1bef-693e-a339-09b645ff22f2@mst.edu>

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From kashif.abbasi at hotmail.com  Wed Jan 29 09:38:09 2020
From: kashif.abbasi at hotmail.com (Kashif Abbasi)
Date: Wed, 29 Jan 2020 09:38:09 +0000
Subject: [Freeswitch-users] freeswitch call-transfer - inbound call
Message-ID: <DM6PR22MB193060B2F35F273BFD467EC2E5050@DM6PR22MB1930.namprd22.prod.outlook.com>

Hi,

I am working on a call flow which includes call transfer (A calls B (B is registered extension at freeswitch), B answers and then transfers the call to C and finally A and C are connected to each other). During call transfer, I want outbound leg to be handled as inbound leg through localhost interface at freeswitch hence creating separate CDR for that (basically this is a requirement at my prepaid billing system to have another inbound leg coming from freeswitch).

Can someone please tell how can I configure public dialplan to work as above and route the call to my default dialplan for further routing? My current matching condition in public dialplan is as below where transferred leg with new prefix is simply going to default dialplan as outbound leg.

    <extension name="xfer_calls">
      <condition field="destination_number" expression="^(1122)(\d+)$">
        <action application="transfer" data="$1$2 XML default"/>
      </condition>
    </extension>

I tried to put "fs_path=sip:127.0.0.1" in above transfer application but that doesn't seem to be working. Please suggest. Thanks

Regards
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