ASL3 multiple issues - SIP/Telephone portal/IAX access

I have a few issues with ASL3, got the broadcastify working. Thanks to N8EI

Now onto getting remote IAX zoiper, working properly, SIP phone (polycom) and Telephone portal.
I see someone already asked about SIP ASL 3 and SIP or PJSIP

I found on this page: Setting up a SIP Phone - AllStarLink Manual
I think the lines you add to modules.conf should say => not just = for the load/noload

I added the other lines in the other sip.conf file etc etc etc as listed, I am unable to connect via SIP
This is a local connection to 192.168.0.144, my firewall is not an issue on the router.

Running the debug, I don’t get any messages for SIP… I used the polycom example for my polycom phone, and tried using zoiper as well on SIP.

Using Zoiper on IAX, I get 2 rings, then it disconnects when dialing my node 2295.

If I change the context it works, using the iaxrpt instead of iax-client context.

[iax-client]
                            ; for IAX VoIP clients.
exten => ${NODE},1,Ringing()
        same => n,Wait(10)
        same => n,Answer()
        same => n,Set(CALLSIGN=${CALLERID(name)})
        same => n,NoOp(Caller ID name is ${CALLSIGN})
        same => n,NoOp(Caller ID number is ${CALLERID(number)})
        same => n,GotoIf(${ISNULL(${CALLSIGN})}?hangit)
        same => n,Playback(rpt/connected-to&rpt/node)
        same => n,SayDigits(${NODE})
        same => n,rpt(${NODE}|P|${CALLSIGN}-P)
        same => n(hangit),NoOp(No Caller ID Name)
        same => n,Playback(connection-failed)
        same => n,Wait(1)
        same => n,Hangup

When connecting via the telephone portal, I get a busy signal after pressing 1 “for voice operated access”

IAX2 Debugging Enabled
Rx-Frame Retry[ No] – OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 00005ms SCall: 10161 DCall: 00000 162.248.93.134:4569
VERSION : 2
CALLED NUMBER : 32295
CODEC_PREFS : (ulaw)
CALLING NUMBER : 914XXXXXXX
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME : KC7DMF
LANGUAGE : en
REFERRING DNIS : 5092557827
USERNAME : allstar-sys
FORMAT : 4
CAPABILITY : 65406
ADSICPE : 2
DATE TIME : 2024-07-04 20:37:44
CALLTOKEN : Present

Tx-Frame Retry[ No] – OSeqno: 000 ISeqno: 001 Type: IAX Subclass: CTOKEN
Timestamp: 00005ms SCall: 00001 DCall: 10161 162.248.93.134:4569
CALLTOKEN : 51 bytes

Rx-Frame Retry[ No] – OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 00053ms SCall: 10161 DCall: 00000 162.248.93.134:4569
VERSION : 2
CALLED NUMBER : 32295
CODEC_PREFS : (ulaw)
CALLING NUMBER : 914XXXXXXX
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME : KC7DMF
LANGUAGE : en
REFERRING DNIS : 5092557827
USERNAME : allstar-sys
FORMAT : 4
CAPABILITY : 65406
ADSICPE : 2
DATE TIME : 2024-07-04 20:37:44
CALLTOKEN : 51 bytes

Tx-Frame Retry[-01] – OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 00053ms SCall: 09161 DCall: 10161 162.248.93.134:4569
Tx-Frame Retry[000] – OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ
Timestamp: 00012ms SCall: 09161 DCall: 10161 162.248.93.134:4569
AUTHMETHODS : 4
CHALLENGE : \x38\x30\x36\x33\x38\x37\x38\x37\x30
USERNAME : allstar-sys

Rx-Frame Retry[ No] – OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP
Timestamp: 00103ms SCall: 10161 DCall: 09161 162.248.93.134:4569
RSA RESULT : VeSOZSEh1tQb5U9xqoa8kro7coacvq3JzLDqUTutczFzChwO7Jymfjdpv6Xx/EfdF/Lq0yruPbWOn5NeSuH/PKxc87HLN6wymWv6+Zo8qSwtKhuylzkaCj0jyOOKePX1ZUlNWPNKyX4iqoFMXQgvTc5WOhdAlQfbvtcN25mNaCo=

Tx-Frame Retry[-01] – OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK
Timestamp: 00103ms SCall: 09161 DCall: 10161 162.248.93.134:4569
[2024-07-04 13:37:44.142] WARNING[26305]: chan_iax2.c:8261 authenticate_verify: Requested inkey ‘allstar’ for RSA authentication does not exist
Tx-Frame Retry[000] – OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REJECT
Timestamp: 01062ms SCall: 09161 DCall: 10161 162.248.93.134:4569
CAUSE : No authority found
CAUSE CODE : 50

[allstar-sys]

exten => _1.,1,Rpt(${EXTEN:1}|Rrpt/node:NODE:rpt/in-call:digits/0:PARKED|120)
exten => _1.,n,Hangup
exten => _2.,1,Ringing
exten => _2.,n,Wait(3)
exten => _2.,n,Answer
exten => _2.,n,Playback(rpt/node)
exten => _2.,n,Saydigits(${EXTEN:1})
exten => _2.,n,Rpt(${EXTEN:1}|P|${CALLERID(name)}-P)
exten => _2.,n,Hangup
exten => _3.,1,Ringing
exten => _3.,n,Wait(3)
exten => _3.,n,Answer
exten => _3.,n,Playback(rpt/node)
exten => _3.,n,Saydigits(${EXTEN:1})
exten => _3.,n,Rpt(${EXTEN:1}|Pv|${CALLERID(name)}-P)
exten => _3.,n,Hangup
exten => _4.,1,Ringing
exten => _4.,n,Wait(3)
exten => _4.,n,Answer
exten => _4.,n,Playback(rpt/node)
exten => _4.,n,Saydigits(${EXTEN:1})
exten => _4.,n,Rpt(${EXTEN:1}|D|${CALLERID(name)}-P)
exten => _4.,n,Hangup
exten => _5.,1,Ringing
exten => _5.,n,Wait(3)
exten => _5.,n,Answer

I did NOT copy any old config files, this is all purely what came with ASL3, and modifications/additions per the website.

Any ideas how to fix these?
The Settings for my node etc are set to enable calling from the telephone portal.

Thanks,
73

Mark
KC7DMF

A couple of things to check.

In extensions.conf is
NODE = xxxx ; change this to your node number
set to your node?

The SIP Phone instructions were updated recently. Take another look.

There is a fix coming for coming that may apply.

I had the same issue and warning:

I opened #80 and included a workaround for this yesterday if you just want to follow those steps…otherwise, @N8EI kindly took care of this already in 3.2.0-1!

The new deb package worked for the phone portal, thanks for that.

I added the fix for phone portal, that is working, however… SIP is still not working, after re-doing the config files per the link there. Testing in Zoiper I get Request timed out (408)
I can use IAX in zoiper, but i’m trying to use it to test, so i can setup my desk phone, which is SIP only.

Running the PJSIP debug, I don’t get anything coming back as to any kind of error etc.

Not sure if there is more I need to enable in modules or disable etc…