SAIL Community Supported PBX

ok i can dial out and connect to my cell  but i have no audio
i notice this line in the call progress out
  -- Launched AGI Script /usr/share/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Set) Options: (__DYNAMIC_FEATURES=automon#clear#outpause#outresume)

but i get this later in the output log
ast_get_chan_applicationmap: Unknown DYNAMIC_FEATURES item 'automon' on channel SIP/401-00000002.

could this be the problem and how do i fix it

complete call log below

== Using SIP RTP CoS mark 5
       > 0x7fb9c8020340 -- Strict RTP learning after remote address set to: 192.168.10.197:11786
    -- Executing [15079953174@internal:1] AGI("SIP/401-00000002", "sarkhpe,OutCos,15079953174,,") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/sarkhpe
    -- <SIP/401-00000002>AGI Script sarkhpe completed, returning 0
    -- Executing [15079953174@401opencos:1] AGI("SIP/401-00000002", "sarkhpe,OutCluster,15079953174,,") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/sarkhpe
    -- <SIP/401-00000002>AGI Script sarkhpe completed, returning 0
    -- Executing [15079953174@qrxvtmny:1] AGI("SIP/401-00000002", "sarkhpe,OutRoute,flow_out_bnd,,") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/sarkhpe
    -- AGI Script Executing Application: (Set) Options: (__DYNAMIC_FEATURES=automon#clear#outpause#outresume)
    -- AGI Script Executing Application: (Set) Options: (CALLERID(number)=401)
    -- AGI Script Executing Application: (Set) Options: (TIMEOUT(absolute)=14400)
    -- Channel will hangup at 2020-07-12 03:30:23.003 CDT.
    -- AGI Script Executing Application: (Dial) Options: (SIP/15079953174@78769303,,T)
  == Using SIP RTP CoS mark 5
    -- Called SIP/15079953174@78769303
       > 0x7fb9a0009830 -- Strict RTP learning after remote address set to: 23.29.23.42:33502
    -- SIP/78769303-00000003 is ringing
    -- SIP/78769303-00000003 is making progress passing it to SIP/401-00000002
       > 0x7fb9c8020340 -- Strict RTP switching to RTP target address 192.168.10.197:11786 as source
       > 0x7fb9c8020340 -- Strict RTP learning complete - Locking on source address 192.168.10.197:11786
    -- SIP/78769303-00000003 answered SIP/401-00000002
[2020-07-11 23:30:31] WARNING[1289][C-00000002]: features_config.c:1366 ast_get_chan_applicationmap: Unknown DYNAMIC_FEATURES item 'automon' on channel SIP/401-00000002.
[2020-07-11 23:30:31] WARNING[1296][C-00000002]: features_config.c:1366 ast_get_chan_applicationmap: Unknown DYNAMIC_FEATURES item 'automon' on channel SIP/78769303-00000003.
    -- Channel SIP/78769303-00000003 joined 'simple_bridge' basic-bridge <bda3e27e-6cad-4f3a-a10f-5902e4332edf>
[2020-07-11 23:30:31] WARNING[1289][C-00000002]: features_config.c:1366 ast_get_chan_applicationmap: Unknown DYNAMIC_FEATURES item 'automon' on channel SIP/401-00000002.
    -- Channel SIP/401-00000002 joined 'simple_bridge' basic-bridge <bda3e27e-6cad-4f3a-a10f-5902e4332edf>
    -- Channel SIP/78769303-00000003 left 'simple_bridge' basic-bridge <bda3e27e-6cad-4f3a-a10f-5902e4332edf>
    -- Channel SIP/401-00000002 left 'simple_bridge' basic-bridge <bda3e27e-6cad-4f3a-a10f-5902e4332edf>
    -- <SIP/401-00000002>AGI Script sarkhpe completed, returning 0
    -- Auto fallthrough, channel 'SIP/401-00000002' status is 'ANSWER'

sorry -- note to self WEAR YOUR GLASSES

how do i add my sip provider from with in sark 6 dedian is there a howto for ver 6 debian
i know how to add it manually to sip.conf

By hard coded, I simply meant a hard value in the callerID field of the extension (Extension->Edit->Outbound Caller ID).  :-)

SARK won't change a callerID unless you ask it to.   There are several places where this can be done; in the extension, in the tenant or in the trunk definition.   If any of these have a CLID then it will be used.

You can verify this by running agi debug at the asterisk console.   If SARK makes a change then it will show up in the trace.   It will show up looking like this

<Local/07nnnnnnnn@internal-00000031;2>AGI Rx << EXEC Set CALLERID(number)=01924nnnnnn
-- AGI Script Executing Application: (Set) Options: (CALLERID(number)=01924nnnnnn)

It's also worth noting that your SIP carrier may not allow you to send a CLID which you don't own, in which case they will usually just override it with the CLID (DDI) associated with the trunk you are sending on.

I just ran a test here on a V6.0.1-51 image with no CLID explicitly set and it sent the originating CLID to my cellphone, which was twinned with my extension.

If it still isn't working for you, let me know the SARK release and send me a console log of the call arriving and leaving the PBX

Hi S
This does sound dynamic "have a hard coded value in the CLID field of your extension definition". Every incoming call sent back out on the twin function has a different CLID.
Unless you or I are thinking of something else.

Yep, I noticed the sounds pack a day or two after I posted this.   I've moved a copy into the repo so I think you should be all good now.   
   

Under normal circumstances it will indeed send the originating CLID unless you have a hard coded value in the CLID field of your extension definition, in which case it will send that instead (see the help paragraph under that field entry in the V6 extension->edit panel).    Usually, you should leave the extension CLID value blank unless you want to send a specific DiD to the callee. 

Hi S
Is there a way to forward the callerid (origin) on to the forwarded number when using the "twin" function on the extension?

The problem is you get a call from your own extension on "twin" but can not return the call as the source number is unknown.

Thanks
G

Quote from: sysadmin on April 18, 2020, 10:05:57 AMGlad you like the V6 layout.  Did you try running it on a tablet or smartphone yet?  It's pretty cool in the way it adapts to differing screen sizes and layouts.

Yes I tried it on my phone and a tablet and it works really well, very impressive how it adapts and prioritised data/menu/columns to fit.

Quote from: sysadmin on April 17, 2020, 09:06:34 PM
OK, let's see.   Which release of 6 are you running?  I wasn't aware of issues with internal calls not showing up.  I'm assuming you are talking about the recordings browser component, not the underlying data.  Yes?

It shows as running 6.0.1-52.

Yes the internal calls are all recorded fine and they are searched and found fine, it's just the recording class PHP file (sark/www/origrecs/lib/Recording.class.php ) didn't have anything in it that would match those calls to be shown in a table on the browser, so they don't actually show up in the list to playback. I had to add the line from my first post along with table data in the PHP file to get Internal calls to show up when searched.


For conference recording I ended up (since my post) adding the following to the "sark_customer_extensions_globals.conf" (perhaps incorrect way to do this now reading your post) so I could eliminate the need to rename the files when moving them.


[conferences]
exten => 101,1,NoOp(conference 101)
same => n,Answer(500)
same => n,Authenticate(1234)
same => n,Set(Confbridge(bridge,record_file)=/var/spool/asterisk/confbridge/${STRFTIME(${EPOCH},,%s)}default-${EXTEN}-${EXTEN}.wav)
same => n,Set(Confbridge(bridge,record_file_timestamp)=no);
same => n,ConfBridge(${EXTEN},,sark_hosted_user)
same => n,Hangup()

To give me the format I needed and I added the "record_file_timestamp=no" because it seemed Asterisk added the time to the end of each file which I didn't want in my formatting. My files end up like this:

xxxxxx6795-default-101-101.wav

Only reason I went for this is I wanted it to look exactly like the other recordings so I didn't need to add much the the PHP file to get this to be matched up and shown in a table and because I've forced the conference room number as both the in and outgoing ID I can use that to match it with the line from my first post. I manually added the tenant name so potentially the recording could just be shown to the Tenant searching the recordings from the browser (I haven't looked at this yet but it looks like the recordings can be password protected with tenant as users)

Quote from: sysadmin on April 17, 2020, 09:06:34 PMIn terms of conference rooms and call recording, we took a slightly different path to you. 

Bear in mind, customers don't always want recording to be turned on for a particular conference room so we wanted to give them the choice.   What we have is an app which uses a prefix (in this case 2) to invoke the conference with recording enabled

I never thought about this and I've never touched custom apps before. I like this approach better and going to have a play when I get chance to see if I can get this working, thanks for the tip/code.

Quote from: sysadmin on April 17, 2020, 09:06:34 PM
Finally, rsync won't move files that are busy so you can run this as often as you like.

This was the missing piece, thanks for that, I'll try rsync so I don't have to wait to run the cron job.

Were you able to reproduce the GB sounds not working (playing Alison US) on a new member entering/leaving?