[gentoo-user] Asterisk - need some help

[Thelma]

Anybody on the list using Asterisk?
I need some help.

Have save version of asterisk is working correctly on one computer but the other.

[John Covici]

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.

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

John Covici wb2una
cov...@ccs.covici.com

[Thelma]

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.

[Jack]

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.

[John Covici]

Sounds like a firewall setting on the home computers, make sure all
the ports are open between the start and end of the asterisk rpt
ports. Are you using a router, or is the asterisk on a public ip?
That is what I can think of at the moment.

[Thelma]

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:5xxx...@192.168.143.1/877,30,rw") in new stack
-- Called IAX2/home_server:5xxx...@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:5xxx...@192.168.143.1/877,30,rw") in new stack
-- Called IAX2/home_server:5xxx...@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

[Thelma]

On home computer there are no ports that are open, remote computer have port open to make openvpn connection
I'm using router, and asterisks are not on public IP, on private IP.

client
dev tun
proto udp
port 9072
topology subnet
remote xxx.xxx.xxx.xx 9072 # Telus fiber

resolv-retry infinite
tun-mtu 1500
tun-mtu-extra 32
mssfix 1200
persist-key
persist-tun
remote-cert-tls server
ca "/etc/openvpn/clinic_i5/ca.crt"
cert "/etc/openvpn/clinic_i5/syscon7.crt"
key "/etc/openvpn/clinic_i5/syscon7.key"
tls-auth "/etc/openvpn/clinic_i5/ta.key" 1
comp-lzo no
log /var/log/openvpn_i5.log
log-append /var/log/openvpn_i5.log
auth-nocache
verb 3

openvpn-log doesn't show any errors (will double check tonight)

The home main-asterisk and backup-asterisk are using same openvpn file; I just drop connection on one, and start connection on the one I want to use.

[John Covici]

On Fri, 02 Feb 2024 14:17:23 -0500,

What do your logs reveal -- you should be surre to set debug in your
logs before testing. Also, what happens if you use sip instead of
aix?

[Thelma]

Good suggestion trying SIP instead of IAX since the call doesn't go over firewall.
I'll try it over the weekend.

[William Kenworthy]

Yes, was caught out recently by the replacement of sip with pjsip -
currently on v21.0.2 and working (sip only, simple home setup) Also had
some weird problems with two versions installed (so asterisk started on
old working version even though new one was installed - once I ran
depclean it failed due to the sip/pjsip issue.

BillK

[Thelma]

When did they implement switch-over from sip to pjsip?
I'm using AudioCode boxes.

I emerged and tried to load asterisk ver.18 but the audiocode would not register. I suppose ver.16 is the end of the line for me.

[John Covici]

I would use at least asterisk 18 in all cases and if you can later
versions. pjsip has been the preferred version for a while, sip is
still OK, however.

[William Kenworthy]

In v18 sip is still present but deprecated - after this its removed. 
There is a conversion script (sip->pjsip) for migration.  It required a
few sacrificial  chickens and much swearing until I got the upstream
trunk to register (iinet in AU).  Its all working good now, the pjsip
config is more programmer friendly but also allows much more complex
(read hard to follow/fault find) configuration.

Note that the CLI commands are not equivalent to sip (including help,
its now pjsip) with a different format.  After install, but before
re-configuration everything sip related disappears on restart.

BillK

[Thelma]

I think I was able solve my problem, it was as simple as disabling "jitterbuffer" in iax.conf
I can hear phone voicemail request from the remote asterisk, will know 100% on Monday.

Regarding Asterisk I'm on 16.30.1 ; tried emerging ver.18 but my AudioCodes box wouldn't even register to it.
Conversion scrip (sip->pjsip) will not do any good if the hardware (AudioCodes boxes, Sipura and other units) are not compatible with pjsip.
Is IAX is gone as well in newer versions?

I have an impression this is the end of old Asterisk that Digium started; very, very sad :-/
It had good community support.

Newer versions 18+ are not compatible with older hardware and learning curve/conversion is not worth it.
Sangoma - community support is almost not existent, few folks just bark at you if one mention still running ver. 16

I'll hang on to 16.30.1 as long as I can.

[William Kenworthy]

I believe asterisk is evolving way from a "carrier" mindset more into an
IT one. Losing support for ancient hardware is part of that. I no longer
use IAX (used it to trunk multiple instances across vpn's - worked
well.)  Currently I only have a couple of now quite old Cisco phones and
Jami softphones on android using SIP to a single asterisk.

I incrementally upgrade asterisk mostly by a clean install carrying over
config files into an LXC instance (using a golden master setup) which
has been trouble free until pjsip - and thats more my fault in missing
that SIP was deprecated and having to unexpectedly fault find it.  The
conversion script worked fine for internal extensions, but the uplink
trunk required a few hours extra work until I stumbled over the correct
syntax - too many options confusing things.

My recommendation would be to spin up a VM and install a test instance
and start afresh - older asterisk versions are usually a security risk
as time goes on.

BillK