Dialing into a BBB conference using freeswitch
asumm...@microtechnow.com
I have configured freeswitch following the instructions in
HostBBB_BigBlueButton_Tips.pdf found at
http://groups.google.com/group/bigbluebutton-dev/browse_thread/thread/46d6a9a9262867d2/a2d5df507fa59a2d?lnk=gst&q=summit#a2d5df507fa59a2d.
I sucessfully have it connect and ask for the conference pin number,
but whatever I put in at that point it tells me that it is an invalid
conference pin number and asks for the conference pin number again.
I'm new to freeswitch, so I'm hoping I'm just overlooking something. I
did read on the same pdf page with the config instructions that the
pin number is the first 3 of the conference ID and 11. Also, if you
create a meetingID with a name as oppossed to numbers, how would the
conferenceID be applicable?
HostBBB.com
you can set the pin... to whatever you want.
<action application="conference" data="${pin}@wideband+${pin:0:3}11"/>
this is what is in the example, which takes the first 3 digit of
confernence number and adds 11
so a confernence 75468 would have a pin of 75411
you could just use +1234 and have 1234 be the pin.
you also can change the voice prompts, this example just asks for
number twice, firs tis room, second pin.
As for non numeric conferences they are allowed but you need to make
changes to dialplan to allow user to select from keypad.
regards,
Stephen
http://hostbbb.com
On Dec 8, 4:35 pm, "asummer...@microtechnow.com"
<asummer...@microtechnow.com> wrote:
> I have a BBB server running version 0.8-beta 3.
> I have configured freeswitch following the instructions in
> HostBBB_BigBlueButton_Tips.pdf found athttp://groups.google.com/group/bigbluebutton-dev/browse_thread/thread....
asumm...@microtechnow.com
The only changes I made were this to public.xml:
<extension name="phone pin">
<condition field="destination_number" expression="8662900475">
<action application="answer"/>
<action application="sleep" data="500"/>
<action application="play_and_get_digits" data="2 5 3 7000 #
conferen$
<action application="transfer" data="SEND_TO_CONFERENCE XML
default"/>
</condition>
</extension>
I added this into default.xml:
<extension name="phone conference">
<condition field="destination_number"
expression="^(SEND_TO_CONFERENCE)$">
<action application="conference" data="${pin}@wideband+${pin:
0:3}11"/>
</condition>
</extension>
And I added this into /var/lib/tomcat6/webapps/bigbluebutton/WEB-INF/
classes/bigbluebutton.properties :
The Conference # is: %%CONFNUM%%.
I created a conference # 53454. When I join using BigBlueButton, it
tells me that the Conference # is 75443. When I call in it only asks
me to enter the conference pin number. In this case, I have tried
53411, 75411 and then 53454 and 75443 just for kicks. Each time it
tells me that it is an invalid entry.
What changes would need to be put in place for non numeric
conferences?
On Dec 9, 10:39 am, "HostBBB.com" <sd...@207me.com> wrote:
> asummerall post the actual code your using to join the confernece...
>
> you can set the pin... to whatever you want.
>
> <action application="conference" data="${pin}@wideband+${pin:0:3}11"/>
>
> this is what is in the example, which takes the first 3 digit of
> confernence number and adds 11
>
> so a confernence 75468 would have a pin of 75411
> you could just use +1234 and have 1234 be the pin.
>
> you also can change the voice prompts, this example just asks for
> number twice, firs tis room, second pin.
>
> As for non numeric conferences they are allowed but you need to make
> changes to dialplan to allow user to select from keypad.
>
> regards,
> Stephenhttp://hostbbb.com
>
> On Dec 8, 4:35 pm, "asummer...@microtechnow.com"
>
>
>
> <asummer...@microtechnow.com> wrote:
> > I have a BBB server running version 0.8-beta 3.
> > I have configured freeswitch following the instructions in
> > HostBBB_BigBlueButton_Tips.pdf found athttp://groups.google.com/group/bigbluebutton-dev/browse_thread/thread....
> > I sucessfully have it connect and ask for the conference pin number,
> > but whatever I put in at that point it tells me that it is an invalid
> > conference pin number and asks for the conference pin number again.
>
> > I'm new to freeswitch, so I'm hoping I'm just overlooking something. I
> > did read on the same pdf page with the config instructions that the
> > pin number is the first 3 of the conference ID and 11. Also, if you
> > create a meetingID with a name as oppossed to numbers, how would the
> > conferenceID be applicable?- Hide quoted text -
>
> - Show quoted text -
HostBBB.com
from BBB meeting. like 76453 then prompt again you enter the first
three digits 764 and 11 so 76411 and you should join the conference.
you need to setup dialplan so dialing any number like 9200 would match
in dialplan, then you just transfer to the conferencename@profile. Im
not sure about use of alpha conf in BBB... works fine in freeswich,
but i have not tried in BBB
regards,
Stephen
hostbbb.com
On Dec 9, 3:55 pm, "asummer...@microtechnow.com"
asumm...@microtechnow.com
number it starts out asking to enter the conference pin. I'm guessing
I have something off in my config for FreeSwitch.
Other than the configuration files at
http://code.google.com/p/bigbluebutton/wiki/ConfigurationFiles#FreeSWITCH
and the pdf with tips at
http://groups.google.com/group/bigbluebutton-dev/browse_thread/thread/46d6a9a9262867d2/a2d5df507fa59a2d?lnk=gst&q=summit#a2d5df507fa59a2d
is there any additional documentation on how to setup Freeswitch for
the voice bridge in BBB?
Also, do I need to have the something setup in the API when the
conference is created?
> > > - Show quoted text -- Hide quoted text -
asumm...@microtechnow.com
don't see any indication that FreeSwitch is even detecting the DTMF
tones.
On Dec 9, 9:10 pm, "HostBBB.com" <sd...@207me.com> wrote:
> > > - Show quoted text -- Hide quoted text -
Richard Alam
<asumm...@microtechnow.com> wrote:
> Additionally when watching through fs_cli with debug level set to 7, I
> don't see any indication that FreeSwitch is even detecting the DTMF
> tones.
>
Can you post your dialplan and the console logs to pastebin.com when
you dial in to connect to the conference?
Ricahrd
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BigBlueButton Developer
http://www.bigbluebutton.org
http://code.google.com/p/bigbluebutton
asumm...@microtechnow.com
public.xml? or is it broken up into public.xml default.xml and public/
00_inbound_did.xml ?
On Dec 12, 12:26 pm, Richard Alam <ritza...@gmail.com> wrote:
> On Mon, Dec 12, 2011 at 11:59 AM, asummer...@microtechnow.com
> BigBlueButton Developerhttp://www.bigbluebutton.orghttp://code.google.com/p/bigbluebutton- Hide quoted text -
asumm...@microtechnow.com
I posted to pastebin per your request.
http://pastebin.com/dSASDK4e
Let me know if there is anything else you need.
On Dec 12, 12:26 pm, Richard Alam <ritza...@gmail.com> wrote:
> On Mon, Dec 12, 2011 at 11:59 AM, asummer...@microtechnow.com
> > For more options, visit this group athttp://groups.google.com/group/bigbluebutton-setup?hl=en.
>
> --
> -----
> BigBlueButton Developerhttp://www.bigbluebutton.orghttp://code.google.com/p/bigbluebutton- Hide quoted text -
HostBBB.com
you here the mute/unmute message.... that will tell you whether DTMF
is being passed thru you call.
DTMF, can be pass inband, or thru sip signaling.... there might be a
setting in your sip provider.
regards,
Stephen
hostbbb.com
On Dec 13, 9:15 am, "asummer...@microtechnow.com"
<asummer...@microtechnow.com> wrote:
> Richard,
>
> I posted to pastebin per your request.http://pastebin.com/dSASDK4e
> > BigBlueButton Developerhttp://www.bigbluebutton.orghttp://code.google.com/p/bigbluebutton-Hide quoted text -
asumm...@microtechnow.com
working.
I cannot dial into the conference, that's why I'm posting here to try
to figure it out.
Any other ideas?
> > > BigBlueButton Developerhttp://www.bigbluebutton.orghttp://code.google.com/p/bigbluebutton-Hidequoted text -
Richard Alam
<asumm...@microtechnow.com> wrote:
> If I could dial into the conference, then I would know that DTMF is
> working.
> I cannot dial into the conference, that's why I'm posting here to try
> to figure it out.
> Any other ideas?
>
> On Dec 13, 10:48 am, "HostBBB.com" <sd...@207me.com> wrote:
>> dial in the conference and press 0, then press 0 again, and see if
>> you here the mute/unmute message.... that will tell you whether DTMF
>> is being passed thru you call.
>>
>> DTMF, can be pass inband, or thru sip signaling.... there might be a
>> setting in your sip provider.
>>
>> regards,
>> Stephen
>> hostbbb.com
>>
>> On Dec 13, 9:15 am, "asummer...@microtechnow.com"
>>
>>
>>
>> <asummer...@microtechnow.com> wrote:
>> > Richard,
>>
>> > I posted to pastebin per your request.http://pastebin.com/dSASDK4e
>>
>> > Let me know if there is anything else you need.
Did you fix the error in the log and tried again?
From the log, you don't have an 8Khz audio installed (see line 463 of
pastebin). FS play_and_get_digits might be failing there
and can't continue. I suggest you get the 8Khz audio from
http://files.freeswitch.org/ and install.
Try putting line 106-110 into public.xml as the dialplan might not be
finding it at default.xml.
Then try removing +${pin:0:3}11 and make sure you enter 5-digits when
prompted so it will match line 187.
HTH.
Richard
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits
asumm...@microtechnow.com
Thanks for the repsonse. I downloaded
http://files.freeswitch.org/freeswitch-sounds-en-us-callie-8000-1.0.16.tar.gz
and put the files in place. I am not longer getting the errors
regarding 8k audio.
I also found that in my cli log I was getting:
[ERR] mod_native_file.c:74 Error opening /opt/freeswitch/sounds/en/us/
callie/ivr/i$.PCMU
and that in my public.xml it said <action
application="play_and_get_digits" data="2 5 3 7000 # conference/conf-
pin.wav ivr/i$
I changed it to be <action application="play_and_get_digits" data="2 5
3 7000 # conference/conf-pin.wav ivr/ivr-that_was_an_invalid_entry.wav
pin \d+"/>
I took your advice and now additionally have this in my public.xml :
<extension name="phone conference">
<condition field="destination_number"
expression="^(SEND_TO_CONFERENCE)$">
<action application="conference" data="$1@wideband"/>
</condition>
</extension>
I had it work once and it showed in the cli that it was receiving the
DTMF tones and all, but then I tried again and now it's not showing
dtmf recognition. So each time I run 'bbb-conf --clean' I get the dtmf
tones to be recognized for one call. Any ideas?
On Dec 14, 9:30 am, Richard Alam <ritza...@gmail.com> wrote:
> On Wed, Dec 14, 2011 at 9:05 AM, asummer...@microtechnow.com
>
>
>
>
>
> <asummer...@microtechnow.com> wrote:
> > If I could dial into the conference, then I would know that DTMF is
> > working.
> > I cannot dial into the conference, that's why I'm posting here to try
> > to figure it out.
> > Any other ideas?
>
> > On Dec 13, 10:48 am, "HostBBB.com" <sd...@207me.com> wrote:
> >> dial in the conference and press 0, then press 0 again, and see if
> >> you here the mute/unmute message.... that will tell you whether DTMF
> >> is being passed thru you call.
>
> >> DTMF, can be pass inband, or thru sip signaling.... there might be a
> >> setting in your sip provider.
>
> >> regards,
> >> Stephen
> >> hostbbb.com
>
> >> On Dec 13, 9:15 am, "asummer...@microtechnow.com"
>
> >> <asummer...@microtechnow.com> wrote:
> >> > Richard,
>
> >> > I posted to pastebin per your request.http://pastebin.com/dSASDK4e
>
> >> > Let me know if there is anything else you need.
>
> Did you fix the error in the log and tried again?
>
> From the log, you don't have an 8Khz audio installed (see line 463 of
> pastebin). FS play_and_get_digits might be failing there
> and can't continue. I suggest you get the 8Khz audio fromhttp://files.freeswitch.org/and install.
>
> Try putting line 106-110 into public.xml as the dialplan might not be
> finding it at default.xml.
>
> Then try removing +${pin:0:3}11 and make sure you enter 5-digits when
> prompted so it will match line 187.
>
> HTH.
>
> Richard
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits
>
> --
> -----
> BigBlueButton Developerhttp://www.bigbluebutton.orghttp://code.google.com/p/bigbluebutton- Hide quoted text -
asumm...@microtechnow.com
somewhere with the NAT'ing. I tried to follow the instructions at
http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT
however when I run 'sofia status' from the fs_cli I get this:
freeswitch@internal> sofia status
Name
Type Data State
=================================================================================================
internal-ipv6 profile
sip:mod_sofia@[::1]:5090 RUNNING (0)
internal profile
sip:mod_...@1.2.3.4:5090 RUNNING (0)
external profile
sip:mod_...@1.2.3.4:5060 RUNNING (0)
192.168.1.3
alias internal ALIASED
=================================================================================================
3 profiles 1 alias
(I've replaced my external/public IP with 1.2.3.4) What I find weird
is that for internal it shows my external IP and port 5090. Do I need
to have port 5090 forwarded on the router?
Also, when I run show nat_map, it returns '0 total.' & nat_map status
returns:
Nat Type: UNKNOWN, ExtIP:
0 total.
And lastly sofia status profile internal returns:
freeswitch@internal> sofia status profile internal
=================================================================================================
Name internal
Domain Name N/A
Auto-NAT false
DBName sofia_reg_internal
Pres Hosts
Dialplan XML
Context public
Challenge Realm auto_from
RTP-IP 192.168.1.3
Ext-RTP-IP 1.2.3.4
SIP-IP 192.168.1.3
Ext-SIP-IP 1.2.3.4
URL sip:mod_...@1.2.3.4:5090
BIND-URL sip:mod_...@1.2.3.4:5090;maddr=192.168.1.3
HOLD-MUSIC local_stream://moh
OUTBOUND-PROXY N/A
CODECS IN
speex@16000h@20i,speex@8000h@20i,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM
CODECS OUT
speex@16000h@20i,speex@8000h@20i,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM
TEL-EVENT 101
DTMF-MODE rfc2833
CNG 13
SESSION-TO 0
MAX-DIALOG 0
NOMEDIA false
LATE-NEG false
PROXY-MEDIA false
AGGRESSIVENAT false
STUN-ENABLED true
STUN-AUTO-DISABLE false
CALLS-IN 0
FAILED-CALLS-IN 0
CALLS-OUT 0
FAILED-CALLS-OUT 0
Registrations:
=================================================================================================
Total items returned: 0
=================================================================================================
I'm thinking this is probably a setting somewhere that I'm
overlooking, but so far it has eluded me.
On Dec 14, 12:50 pm, "asummer...@microtechnow.com"
<asummer...@microtechnow.com> wrote:
> Richard,
>
> Thanks for the repsonse. I downloadedhttp://files.freeswitch.org/freeswitch-sounds-en-us-callie-8000-1.0.1...
> > BigBlueButton Developerhttp://www.bigbluebutton.orghttp://code.google.com/p/bigbluebutton-Hide quoted text -
>
> > - Show quoted text -- Hide quoted text -
Papa Charlie
Did you get the BBB conference with FreeSwitch working ? How did you do it ? I'm running BBB 2.0.0 and using the instructions here:
http://docs.bigbluebutton.org/install/install.html#add-a-phone-number-to-the-conference-bridge
Before I upgraded from 2.0.0-beta I could see the conference PIN being dialled from fs_cli, after the upgrade, I don't see the PIN being dialled and the prompt keeps saying the PIN is invalid. I also see the line below in fs_cli:
Regex (FAIL) [check_if_conference_active] ${conference ${pin} list}(No active conferences.
) =~ //sofia/g/ break=on-false
I'm in a session that displays the phone number to call and PIN to join the session.
Any pointers on where to check ?
Chad Pilkey
Papa Charlie
Like I said in my previous post, I have followed the instructions here:
http://docs.bigbluebutton.org/install/install.html#add-a-phone-number-to-the-conference-bridge
I keep getting invalid PIN when I key in the PIN displayed in the BigBlueButton session, also fs_cli does not show the PIN being entered.
Fred Dixon
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Papa Charlie
Will upgrade ASAP and report back.
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Papa Charlie
I've upgraded to 2.2-beta. The problem persists with calling in to conference from a phone.
I get the ivr prompt to key in the PIN displayed in the BigBlueButton session, each time it says the PIN is incorrect. Prior to upgrading to 2.0, I could see the PIN being entered from fs_cli, now the PIN is not echoed in the fs_cli console. Additionally, I see a line in the fs_cli which says:
Dialplan: sofia/external/14164...@192.159.66.3 parsing [public->check_if_conference_active] continue=false
Dialplan: sofia/external/14164...@192.159.66.3 Regex (FAIL) [check_if_conference_active] ${conference ${pin} list}(+OK No active conferences.
) =~ //sofia/g/ break=on-false
Please how can I get the system to read the PIN being keyed in and why does it say "no active conference" when I am inside a BigBlueButton session which displays the PIN and number to dial in ? Thanks.
On Sunday, 7 July 2019 18:02:17 UTC+1, Fred Dixon wrote:
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Chad Pilkey
<condition field="destination_number" expression="^EXTERNALDID$">
<action application="answer"/>
<action application="sleep" data="500"/>
<action application="play_and_get_digits" data="5 5 3 7000 # conference/conf-pin.wav ivr/ivr-that_was_an_invalid_entry.wav pin \d+"/>
<action application="transfer" data="${pin} XML default"/>
</condition>
</extension>
Papa Charlie
Chad Pilkey
2019-07-09 20:51:19.355136 [DEBUG] switch_channel.c:3820 (sofia/external/anon...@64.2.142.13) Callstate Change EARLY -> ACTIVE
2019-07-09 20:51:19.355136 [DEBUG] sofia.c:7323 Channel sofia/external/anon...@64.2.142.13 entering state [completed][200]
EXECUTE [depth=0] sofia/external/anon...@64.2.142.13 sleep(500)
2019-07-09 20:51:19.415117 [DEBUG] sofia.c:7323 Channel sofia/external/anon...@64.2.142.13 entering state [ready][200]
2019-07-09 20:51:19.735149 [DEBUG] switch_rtp.c:1895 rtcp_stats_init: audio ssrc[1281275913] base_seq[8940]
2019-07-09 20:51:19.755205 [DEBUG] switch_rtp.c:7555 Correct audio ip/port confirmed.
EXECUTE [depth=0] sofia/external/anon...@64.2.142.13 set(bbb_authorized=true)
2019-07-09 20:51:19.855146 [DEBUG] mod_dptools.c:1598 SET sofia/external/anon...@64.2.142.13 [bbb_authorized]=[true]
EXECUTE [depth=0] sofia/external/anon...@64.2.142.13 play_and_get_digits(5 5 3 7000 # conference/conf-pin.wav ivr/ivr-that_was_an_invalid_entry.wav pin \d+)
2019-07-09 20:51:19.855146 [DEBUG] switch_ivr_play_say.c:1497 Codec Activated L16@8000hz 1 channels 20ms
2019-07-09 20:51:21.595175 [DEBUG] switch_ivr_play_say.c:1941 done playing file /opt/freeswitch/share/freeswitch/sounds/en/us/callie/conference/conf-pin.wav
2019-07-09 20:51:28.615160 [DEBUG] switch_ivr_play_say.c:1497 Codec Activated L16@8000hz 1 channels 20ms
2019-07-09 20:51:30.275124 [DEBUG] switch_ivr_play_say.c:1941 done playing file /opt/freeswitch/share/freeswitch/sounds/en/us/callie/ivr/ivr-that_was_an_invalid_entry.wav
2019-07-09 20:51:30.275124 [DEBUG] switch_ivr_play_say.c:1497 Codec Activated L16@8000hz 1 channels 20ms
2019-07-09 20:51:32.015134 [DEBUG] switch_ivr_play_say.c:1941 done playing file /opt/freeswitch/share/freeswitch/sounds/en/us/callie/conference/conf-pin.wav
2019-07-09 20:51:39.015143 [DEBUG] switch_ivr_play_say.c:1497 Codec Activated L16@8000hz 1 channels 20ms
2019-07-09 20:51:39.195146 [DEBUG] switch_rtp.c:7794 RTP RECV DTMF 7:800
2019-07-09 20:51:39.195146 [INFO] switch_channel.c:517 RECV DTMF 7:800
2019-07-09 20:51:39.855137 [DEBUG] switch_rtp.c:7794 RTP RECV DTMF 7:800
2019-07-09 20:51:39.855137 [INFO] switch_channel.c:517 RECV DTMF 7:800
2019-07-09 20:51:40.675145 [DEBUG] switch_ivr_play_say.c:1941 done playing file /opt/freeswitch/share/freeswitch/sounds/en/us/callie/ivr/ivr-that_was_an_invalid_entry.wav
2019-07-09 20:51:40.675145 [DEBUG] switch_ivr_play_say.c:1497 Codec Activated L16@8000hz 1 channels 20ms
2019-07-09 20:51:40.675145 [DEBUG] switch_ivr_play_say.c:1941 done playing file /opt/freeswitch/share/freeswitch/sounds/en/us/callie/conference/conf-pin.wav
2019-07-09 20:51:41.555134 [DEBUG] switch_rtp.c:7794 RTP RECV DTMF 9:800
2019-07-09 20:51:41.555134 [INFO] switch_channel.c:517 RECV DTMF 9:800
2019-07-09 20:51:43.135133 [DEBUG] switch_rtp.c:7794 RTP RECV DTMF 3:800
2019-07-09 20:51:43.135133 [INFO] switch_channel.c:517 RECV DTMF 3:800
2019-07-09 20:51:44.095147 [DEBUG] switch_rtp.c:7794 RTP RECV DTMF 5:800
2019-07-09 20:51:44.095147 [INFO] switch_channel.c:517 RECV DTMF 5:800
EXECUTE [depth=0] sofia/external/anon...@64.2.142.13 transfer(77935 XML default)
2019-07-09 20:51:44.095147 [DEBUG] switch_ivr.c:2208 (sofia/external/anon...@64.2.142.13) State Change CS_EXECUTE -> CS_ROUTING
2019-07-09 20:51:44.095147 [NOTICE] switch_ivr.c:2215 Transfer sofia/external/anon...@64.2.142.13 to XML[77935@default]
2019-07-09 20:51:44.095147 [DEBUG] switch_core_state_machine.c:650 (sofia/external/anon...@64.2.142.13) State EXECUTE going to sleep
2019-07-09 20:51:44.095147 [DEBUG] switch_core_state_machine.c:584 (sofia/external/anon...@64.2.142.13) Running State Change CS_ROUTING (Cur 1 Tot 12)
2019-07-09 20:51:44.095147 [DEBUG] switch_channel.c:2288 (sofia/external/anon...@64.2.142.13) Callstate Change ACTIVE -> RINGING
2019-07-09 20:51:44.095147 [DEBUG] switch_core_state_machine.c:643 (sofia/external/anon...@64.2.142.13) State ROUTING
2019-07-09 20:51:44.095147 [DEBUG] mod_sofia.c:154 sofia/external/anon...@64.2.142.13 SOFIA ROUTING
2019-07-09 20:51:44.095147 [DEBUG] switch_core_state_machine.c:236 sofia/external/anon...@64.2.142.13 Standard ROUTING
2019-07-09 20:51:44.095147 [INFO] mod_dialplan_xml.c:637 Processing anonymous <anonymous>->77935 in context default
Dialplan: sofia/external/anon...@64.2.142.13 parsing [default->unloop] continue=false
Dialplan: sofia/external/anon...@64.2.142.13 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/anon...@64.2.142.13 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/anon...@64.2.142.13 parsing [default->bbb_conferences_ws] continue=false
Dialplan: sofia/external/anon...@64.2.142.13 Regex (PASS) [bbb_conferences_ws] ${bbb_authorized}(true) =~ /true/ break=on-false
Dialplan: sofia/external/anon...@64.2.142.13 Regex (FAIL) [bbb_conferences_ws] ${sip_via_protocol}(udp) =~ /^wss?$/ break=on-false
Dialplan: sofia/external/anon...@64.2.142.13 parsing [default->bbb_conferences] continue=false
Dialplan: sofia/external/anon...@64.2.142.13 Regex (PASS) [bbb_conferences] ${bbb_authorized}(true) =~ /true/ break=on-false
Dialplan: sofia/external/anon...@64.2.142.13 Regex (PASS) [bbb_conferences] destination_number(77935) =~ /^(\d{5,6})$/ break=on-false
Dialplan: sofia/external/anon...@64.2.142.13 Action set(jitterbuffer_msec=20:400)
Dialplan: sofia/external/anon...@64.2.142.13 Action answer()
Dialplan: sofia/external/anon...@64.2.142.13 Action conference(77935@cdquality)
Papa Charlie
I see the "Set 2833 dtmf" (see below) but no DTMF output: How can I fix ?
2019-07-09 22:36:50.242582 [DEBUG] switch_core_media.c:5407 Set telephone-event payload to 101@8000
2019-07-09 22:36:50.242582 [DEBUG] switch_core_media.c:3781 Set Codec sofia/external/14164...@192.159.66.3 PCMU/8000 20 ms 160 samples 64000 bits 1
channels
2019-07-09 22:36:50.242582 [DEBUG] switch_core_codec.c:111 sofia/external/14164...@192.159.66.3 Original read codec set to PCMU:0
2019-07-09 22:36:50.242582 [DEBUG] switch_core_media.c:5750 Set telephone-event payload to 101@8000
2019-07-09 22:36:50.242582 [DEBUG] switch_core_media.c:5808 sofia/external/14164...@192.159.66.3 Set 2833 dtmf send payload to 101 recv payload to
101
2019-07-09 22:36:50.242582 [DEBUG] switch_core_media.c:8524 AUDIO RTP [sofia/external/14164...@192.159.66.3] 10.1.75.27 port 17856 -> 137.192.78.8
port 32342 codec: 0 ms: 20
2019-07-09 22:36:50.242582 [DEBUG] switch_rtp.c:4305 Starting timer [soft] 160 bytes per 20ms
2019-07-09 22:36:50.262503 [DEBUG] switch_core_media.c:8743 Activating RTCP PORT 0
2019-07-09 22:36:50.262503 [DEBUG] switch_rtp.c:4701 RTCP send rate is: 5000 and packet rate is: 20000 Remote Port: 32343
2019-07-09 22:36:50.262503 [DEBUG] switch_rtp.c:2577 Setting RTCP remote addr to 137.192.78.8:32343 2
2019-07-09 22:36:50.262503 [DEBUG] switch_core_media.c:8826 sofia/external/14164...@192.159.66.3 Set 2833 dtmf send payload to 101
2019-07-09 22:36:50.262503 [DEBUG] switch_core_media.c:8833 sofia/external/14164...@192.159.66.3 Set 2833 dtmf receive payload to 101
2019-07-09 22:36:50.262503 [DEBUG] switch_core_media.c:8856 sofia/external/14164...@192.159.66.3 Set rtp dtmf delay to 40
2019-07-09 22:36:50.262503 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/14164...@192.159.66.3!
2019-07-09 22:36:50.262503 [DEBUG] switch_channel.c:3521 (sofia/external/14164...@192.159.66.3) Callstate Change RINGING -> EARLY
2019-07-09 22:36:50.262503 [DEBUG] switch_core_media.c:8507 Audio params are unchanged for sofia/external/14164...@192.159.66.3.
2019-07-09 22:36:50.262503 [DEBUG] mod_sofia.c:882 Local SDP sofia/external/14164...@192.159.66.3:
v=0
o=FreeSWITCH 1562693954 1562693955 IN IP4 10.1.75.27
s=FreeSWITCH
c=IN IP4 10.1.75.27
t=0 0
m=audio 17856 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=rtcp:17857 IN IP4 10.1.75.27
2019-07-09 22:36:50.262503 [NOTICE] mod_dptools.c:1360 Channel [sofia/external/14164...@192.159.66.3] has been answered
2019-07-09 22:36:50.262503 [DEBUG] switch_channel.c:3820 (sofia/external/14164...@192.159.66.3) Callstate Change EARLY -> ACTIVE
2019-07-09 22:36:50.262503 [DEBUG] sofia.c:7323 Channel sofia/external/14164...@192.159.66.3 entering state [completed][200]
EXECUTE [depth=0] sofia/external/14164...@192.159.66.3 sleep(500)
EXECUTE [depth=0] sofia/external/14164...@192.159.66.3 set(bbb_authorized=true)
2019-07-09 22:36:50.762503 [DEBUG] mod_dptools.c:1598 SET sofia/external/14164...@192.159.66.3 [bbb_authorized]=[true]
EXECUTE [depth=0] sofia/external/14164...@192.159.66.3 play_and_get_digits(5 5 3 7000 # conference/conf-pin.wav ivr/ivr-that_was_an_invalid_entry.w
av pin \d+)
2019-07-09 22:36:50.762503 [DEBUG] switch_ivr_play_say.c:1497 Codec Activated L16@8000hz 1 channels 20ms
2019-07-09 22:36:52.482502 [DEBUG] switch_ivr_play_say.c:1941 done playing file /opt/freeswitch/share/freeswitch/sounds/en/us/callie/conference/conf-p
in.wav
2019-07-09 22:36:59.502500 [DEBUG] switch_ivr_play_say.c:1497 Codec Activated L16@8000hz 1 channels 20ms
2019-07-09 22:37:01.162501 [DEBUG] switch_ivr_play_say.c:1941 done playing file /opt/freeswitch/share/freeswitch/sounds/en/us/callie/ivr/ivr-that_was_
an_invalid_entry.wav
2019-07-09 22:37:01.182512 [DEBUG] switch_ivr_play_say.c:1497 Codec Activated L16@8000hz 1 channels 20ms
2019-07-09 22:37:02.902501 [DEBUG] switch_ivr_play_say.c:1941 done playing file /opt/freeswitch/share/freeswitch/sounds/en/us/callie/conference/conf-p
in.wav
Chad Pilkey
Papa Charlie
Chad Pilkey
sofia loglevel all 9
sofia global siptrace on
Papa Charlie
Please find the full log at the link below: (Let me know if I should paste it directly in my response)
https://pastebin.com/hpByYirj
> ><a href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits" rel="nofollow" target="_blank" onmousedown="this.href='http://www.google.com/url?q\x3dhttp%3A%2F%2Fwiki.freeswitch.org%2Fwiki%2FMisc._Dialplan_Tools_play_and_get_digits\x26sa\x3dD\x26sntz\x3d1\x26usg\x3dAFQjCNGFmBZFAkhbgva4azGIvALNVjlqYw';return true;" onclick="this.href='http://www.google.com/url?q\x3dhttp%3A%2F%2Fwiki.freeswitch.org%2Fw
Chad Pilkey
> ><a href="<a href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits" target="_blank" rel="nofollow" onmousedown="this.href='http://www.google.com/url?q\x3dhttp%3A%2F%2Fwiki.freeswitch.org%2Fwiki%2FMisc._Dialplan_Tools_play_and_get_digits\x26sa\x3dD\x26sntz
Papa Charlie
>
Chad Pilkey
> > >> <a href="http://hostbbb.com" rel="nofollow" target="_blank" onmousedown="
Papa Charlie
The problem was that I was getting the ICE 1007 error when trying to join the BBB conference with Microphone, so I went ahead and joined listen only. I wronly assumed this was due to the fact the microphone on my computer is faulty until someone else tried and got the same error. I then followed the steps in the troubleshooting section of the installation guide and the error changed to "1002 websocket . . . ." I then setup a "dummy NIC" as per the troubleshooting instructions and the problem was solved. The interesting thing is the test for if you require a dummy NIC or not came back suggesting I did not need a dummy NIC. Further, bbb-conf --status came back showing all services running even though there was a warning at startup about "SIP". So I guess the lesson is never ignore ANY warning/error message unless you are damn sure you know what you are doing.