Portsip continues ringing

Asterisk Asterisk Dialplan
1

Hello everyone! :wave:

I have 4 pjsip extensions with PortSIP softphones. The extension I use to test my setup works like a charm, I can accept incoming calls without problems and call external numbers. But the 3 others keep ringing when accepting the calls.
My question is, is it possibly a problem with my Asterisk setup? But why would my extension work and not the others?
Just to be clear: I put my extension on the top of the list when I do testing and remove it afterwards:
extensions.conf
[incoming]
;exten => 33xxxxxxxx,1,Dial(PJSIP/test, 20)
exten => 33xxxxxxxx,1,Dial(PJSIP/1001,20)
exten => 33xxxxxxxx,2,Dial(PJSIP/1002,20)
exten => 33xxxxxxxx,3,Dial(PJSIP/1003,20)

Thanks for some tips and hints, I really appreciate it!
Best regards :v:
Marc-André

2

Do

exten => 33xxxxxxxx,1,Dial(PJSIP/1001&PJSIP/1002&PJSIP/1003,20)

Le 16/07/2024 à 16:54, marcandreappel via Asterisk Community a écrit :

exten => 33xxxxxxxx,1,Dial(PJSIP/1001,20)

–
Daniel

3

Thanks for this piece of advice, I have implemented it and will have to wait 'til the 3 softphones are connected and clients start calling. But I’m curious, will that fix the problem with the softphones? Because when I had my softphone in the original configuration on the first position, it didn’t continue ringing after accepting the call.

4

Sorry for the lack of information, I’ll provide some configuration:
extension.conf

[call2trunk]
exten => _0X!,1,Set(newnum=${FILTER(0-9,${EXTEN})})
same => n,Set(newnum=500033${newnum:1})
same => n,Goto(call2trunk-outgoing,${newnum},1)

exten => _+33X!,1,Set(newnum=500033${FILTER(0-9,${EXTEN:3})})
same => n,Goto(call2trunk-outgoing,${newnum},1)

exten => _0033X!,1,Set(newnum=500033${FILTER(0-9,${EXTEN:4})})
same => n,Goto(call2trunk-outgoing,${newnum},1)

[call2trunk-outgoing]
exten => _50X!,1,Dial(PJSIP/${EXTEN}@maniterm)
same => n,Hangup()

[trunk2call]
exten => _X.,1, NoOp(Appel entrant : ${CALLERID(num)})
same => n,Goto(trunk2call-incoming,${EXTEN},1)

[trunk2call-incoming]
exten => 33xxxxxxx,1,Dial(PJSIP/1001&PJSIP/1002&PJSIP/1003,20)

[main-context]
include => call2trunk
include => trunk2call

pjsip.conf

[maniterm]
qualify = yes

[maniterm]
type = aor
contact = sip:185.101.180.190
qualify_frequency = 60

[maniterm]
type = identify
endpoint = maniterm
match = 185.101.180.190

[maniterm]
type = endpoint
context = call2trunk
disallow = all
allow = alaw,ulaw,g729
rtp_symmetric = yes
rewrite_contact = yes
rtp_timeout = 60
direct_media = no
aors = maniterm
allow_overlap = yes

[manivox]
qualify = yes

[manivox]
type = aor
contact = sip:185.101.180.252
qualify_frequency = 60

[manivox]
type = identify
endpoint = manivox
match = 185.101.180.252

[manivox]
type = endpoint
context = trunk2call
disallow = all
allow = alaw,ulaw,g729
rtp_symmetric = yes
rewrite_contact = yes
rtp_timeout = 60
direct_media = no
aors = manivox
allow_overlap = yes

[1001] ; identical for all extensions
type = endpoint
context = main-context
disallow = all
allow = alaw,ulaw,g729
auth = test
aors = test

[1001]
type = auth
auth_type = userpass
password = supersecret
username = 1001

[1001]
type = aor
max_contacts = 1

Could there be elsewhere an issue with my setup that could prevent a softphone to accept a call correctly on another continent?

Thanks again for advise and inside!
Best regards

5

Unfortunately your suggestion didn’t resolve my issue, but it made my configuration cleaner, thanks for that!
Best regards

6

Do you somewhere Anwer() the call ?

7

Please forgive my ignorance, but I thought it to be obvious when the softphone accepts the call? So the call needs to be answered?
Still, I wonder why my PortSIP does act correctly when I add it to the list of extensions (or is it just luck?).
Thanks again!

8

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