Re: [gentoo-user] Asterisk - need some help

[Thelma <thelma@sys-concept.com>]

On 2/2/24 11:21, Jack wrote:
> On 2024.02.02 12:53, Thelma wrote:
>> On 2/2/24 10:09, John Covici wrote:
>>>
>>> On Fri, 02 Feb 2024 10:26:09 -0500,
>>> Thelma wrote:
>>>>
>>>> Anybody on the list using Asterisk?
>>>> I need some help.
>>>>
>>>> Have save version of asterisk is working correctly on one computer but the other.
>>>>
>>>>
>>>
>>> I use asterisk all the time, but I don't use the gentoo package, I
>>> compile from source myself because some of the computers I use it on
>>> have different requirements and this way I have more conttrol as to
>>> what goes on.
>>
>> I have been using asterisk for some time but have run into strange problem now.
>>
>> I have home-asterisk and remote-location-asterisk they are connected via openvpn and IAX
>> and I use iax to register home-asterisk to remote-asterisk
>>
>> At home I have two computers (main-asterisk and backup-asterisk), running save version of Asterisk, same dial-plan, same config files, all files in /etc/asterisk are identical
>> on home-comuters, I compare them with "meld"
>>
>> - backup-asterisk to remote-asterisk works OK, I can call remote asterisk internally over IAX and remote asterisk receive the call and voice is working.
>>
>> - main-asterisk to remote-asterisk doesn't work well and I don't know how to troubleshoot.
>>
>> When I place a call to remote-asterisk, internally over IAX the phone is ringing but when somebody answer the call we can not hear each other.
>> When somebody from remote-asterisk calls me (home-asterisk) internally over IAX  voice is working correctly; it only happen when I place a call from
>> home-asterisk to remote-asterisk (and only from my main-computer) it is not working.
>>
>> When I call remote-asterisk over POTS line it works OK.
> 
> I do not use asterisk, but I would look at configuration files to see if there is some filtering of allowed IP addresses for connections.  My first suspicion would be that main-asterisk and/or backup-asterisk has changed its address as seen by remote-asterisk and is now being handled differently.

When home-asterisk register IAX with the remote-asterisk, I can see it on the asterisk-CLI commend line: "iax show registry" the are registered with each other,
otherwise the call wouldn't go through at all.

Here is the output, from both asterisks; one that works and one that doesn't:

=======NOT WORKING=============

"main-asterisk" (NOT WORKING):

== Using SIP RTP CoS mark 5
        > 0x7fe0e00339a0 -- Strict RTP learning after remote address set to: 10.0.0.110:6000
     -- Executing [877@internal:1] Dial("SIP/55-00000005", "IAX2/home_server:5xxxxxx7@192.168.143.1/877,30,rw") in new stack
     -- Called IAX2/home_server:5xxxxxx7@192.168.143.1/877
     -- Call accepted by 192.168.143.1:4569 (format ulaw)
     -- Format for call is (ulaw)
     -- IAX2/192.168.143.1:4569-6413 is ringing
     -- IAX2/192.168.143.1:4569-6413 is ringing
     -- Nobody picked up in 30000 ms
     -- Hungup 'IAX2/192.168.143.1:4569-6413'
     -- Executing [877@internal:2] Hangup("SIP/55-00000005", "") in new stack
   == Spawn extension (internal, 877, 2) exited non-zero on 'SIP/55-00000005'


Remote-Asterisk:

    -- Accepting AUTHENTICATED call from 192.168.143.7:
     --        > requested format = ulaw,
     --        > requested prefs = (ulaw|gsm|ilbc|speex|g729|g723|alaw),
     --        > actual format = ulaw,
     --        > host prefs = (ulaw|alaw),
     --        > priority = mine
     -- Executing [877@extensions:1] Set("IAX2/home_server-3394", "recordfilename=55-877-2024_02_01_2048.wav") in new stack
     -- Executing [877@extensions:2] MixMonitor("IAX2/home_server-3394", "55-877-2024_02_01_2048.wav") in new stack
     -- Executing [877@extensions:3] Dial("IAX2/home_server-3394", "SIP/877,25,trw") in new stack
   == Begin MixMonitor Recording IAX2/home_server-3394
   == Using SIP RTP CoS mark 5
     -- Called SIP/877
     -- SIP/877-0000001e is ringing
     -- Nobody picked up in 25000 ms
     -- Executing [877@extensions:4] Playback("IAX2/home_server-3394", "beep") in new stack
     -- <IAX2/home_server-3394> Playing 'beep.gsm' (language 'en')
     -- Executing [877@extensions:5] VoiceMail("IAX2/home_server-3394", "877") in new stack
     -- <IAX2/home_server-3394> Playing 'vm-intro.gsm' (language 'en')
   == Spawn extension (extensions, 877, 5) exited non-zero on 'IAX2/home_server-3394'
     -- Hungup 'IAX2/home_server-3394'
   == MixMonitor close filestream (mixed)
   == End MixMonitor Recording IAX2/home_server-3394

=========WORKING OK=========

"backup-asterisk" (WORKING):

   == Using SIP RTP CoS mark 5
        > 0x7fef4c023480 -- Strict RTP learning after remote address set to: 10.0.0.110:6020
     -- Executing [877@internal:1] Dial("SIP/55-00000000", "IAX2/home_server:5xxxxxx7@192.168.143.1/877,30,rw") in new stack
     -- Called IAX2/home_server:5xxxxxx7@192.168.143.1/877
     -- Call accepted by 192.168.143.1:4569 (format ulaw)
     -- Format for call is (ulaw)
     -- IAX2/192.168.143.1:4569-1894 is ringing
     -- IAX2/192.168.143.1:4569-1894 is ringing
     -- IAX2/192.168.143.1:4569-1894 answered SIP/55-00000000
     -- Channel IAX2/192.168.143.1:4569-1894 joined 'simple_bridge' basic-bridge <2e8c7579-0455-4847-a57a-1955e00a83d8>
     -- Channel SIP/55-00000000 joined 'simple_bridge' basic-bridge <2e8c7579-0455-4847-a57a-1955e00a83d8>
        > 0x7fef4c023480 -- Strict RTP switching to RTP target address 10.0.0.110:6020 as source
        > 0x7fef4c023480 -- Strict RTP learning complete - Locking on source address 10.0.0.110:6020
     -- Channel SIP/55-00000000 left 'simple_bridge' basic-bridge <2e8c7579-0455-4847-a57a-1955e00a83d8>
     -- Channel IAX2/192.168.143.1:4569-1894 left 'simple_bridge' basic-bridge <2e8c7579-0455-4847-a57a-1955e00a83d8>
   == Spawn extension (internal, 877, 1) exited non-zero on 'SIP/55-00000000'
     -- Hungup 'IAX2/192.168.143.1:4569-1894'

Remote-Asterisk:

    -- Accepting AUTHENTICATED call from 192.168.143.7:
     --        > requested format = ulaw,
     --        > requested prefs = (ulaw|gsm|ilbc|speex|g729|g723|alaw),
     --        > actual format = ulaw,
     --        > host prefs = (ulaw|alaw),
     --        > priority = mine
     -- Executing [877@extensions:1] Set("IAX2/home_server-2819", "recordfilename=55-877-2024_02_01_2101.wav") in new stack
     -- Executing [877@extensions:2] MixMonitor("IAX2/home_server-2819", "55-877-2024_02_01_2101.wav") in new stack
     -- Executing [877@extensions:3] Dial("IAX2/home_server-2819", "SIP/877,25,trw") in new stack
   == Begin MixMonitor Recording IAX2/home_server-2819
   == Using SIP RTP CoS mark 5
     -- Called SIP/877
     -- SIP/877-0000001f is ringing
     -- Nobody picked up in 25000 ms
     -- Executing [877@extensions:4] Playback("IAX2/home_server-2819", "beep") in new stack
     -- <IAX2/home_server-2819> Playing 'beep.gsm' (language 'en')
     -- Executing [877@extensions:5] VoiceMail("IAX2/home_server-2819", "877") in new stack
     -- <IAX2/home_server-2819> Playing 'vm-intro.gsm' (language 'en')
   == Spawn extension (extensions, 877, 5) exited non-zero on 'IAX2/home_server-2819'
     -- Hungup 'IAX2/home_server-2819'
   == MixMonitor close filestream (mixed)
   == End MixMonitor Recording IAX2/home_server-2819