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
