Configuration trunk asterisk/yeastart-ipbx with pjsip.conf

Asterisk Asterisk SIP
1

Hi!
I’m trying to get an exit point with my yeastart that I have linked to my asterisk server. I can call the extensions created in yeastar but yeastar can’t send calls to my asterisk. here is the exit I have:
ipbx*CLI> pjsip show endpoints

Endpoint: <Endpoint/CID…> <State…> <Channels.>
I/OAuth: <AuthId/UserName…>
Aor: <Aor…>
Contact: <Aor/ContactUri…> <Hash…> <RTT(ms)…>
Transport: <TransportId…> <BindAddress…>
Identify: <Identify/Endpoint…>
Match: <criteria…>
Channel: <ChannelId…> <State…> <Time…>
Exten: <DialedExten…> CLCID: <ConnectedLineCID…>

Endpoint: 600 Unavailable 0 of inf
OutAuth: 600/600
InAuth: 600/600
Aor: 600 2

Endpoint: 601 Not in use 0 of inf
OutAuth: 601/601
InAuth: 601/601
Aor: 601 2
Contact: 601/sip:601@192.168.2.151:5060 340454419e NonQual nan
Contact: 601/sip:601@192.168.2.2:63574;transport=UD 6f6e89dbd2 NonQual nan

Endpoint: 602 Unavailable 0 of inf
OutAuth: 602/602
InAuth: 602/602
Aor: 602 1

Endpoint: 603 Unavailable 0 of inf
OutAuth: 603/603
InAuth: 603/603
Aor: 603 1

Endpoint: voipms Not in use 0 of inf
OutAuth: voipms/1000
Aor: voipms 0
Contact: voipms/sip:192.168.2.151 8557c956e2 Avail 7.308

Objects found: 5

[Dec 4 21:45:02] NOTICE[2367]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request ‘INVITE’ from ‘“1000” sip:1000@192.168.2.152’ failed for ‘192.168.2.151:5060’ (callid: f8def36b-a09e-487c-9fbd-7c54d6af6713) - No matching endpoint found
[Dec 4 21:45:02] NOTICE[2367]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request ‘INVITE’ from ‘“1000” sip:1000@192.168.2.152’ failed for ‘192.168.2.151:5060’ (callid: f8def36b-a09e-487c-9fbd-7c54d6af6713) - No matching endpoint found
[Dec 4 21:45:02] NOTICE[2367]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request ‘INVITE’ from ‘“1000” sip:1000@192.168.2.152’ failed for ‘192.168.2.151:5060’ (callid: f8def36b-a09e-487c-9fbd-7c54d6af6713) - Failed to authenticate
[Dec 4 21:45:18] NOTICE[2516]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request ‘NOTIFY’ from ‘sip:1000@192.168.2.151’ failed for ‘192.168.2.151:5060’ (callid: b556d33b-be9f-46da-8bc3-6cbffabb801d) - No matching endpoint found
[Dec 4 21:45:25] NOTICE[2367]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request ‘OPTIONS’ from ‘sip:unknown@192.168.2.151’ failed for ‘192.168.2.151:5060’ (callid: 4612affc-6cfb-44f6-9b1d-d22df4b39ff6) - No matching endpoint found
[Dec 4 21:45:32] NOTICE[2516]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request ‘OPTIONS’ from ‘sip:unknown@192.168.2.151’ failed for ‘192.168.2.151:5060’ (callid: b0ec9dd6-f9b3-4e38-8242-dab380abbd1a) - No matching endpoint found
[Dec 4 21:46:11] NOTICE[2516]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request ‘NOTIFY’ from ‘sip:1000@192.168.2.151’ failed for ‘192.168.2.151:5060’ (callid: 58650f41-be58-4bca-9792-b1063651d3c7) - No matching endpoint found
[Dec 4 21:46:25] NOTICE[2367]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request ‘OPTIONS’ from ‘sip:unknown@192.168.2.151’ failed for ‘192.168.2.151:5060’ (callid: 78f92373-3420-4445-8e14-8549410725e4) - No matching endpoint found

my pjsip.conf

[general]
hasvoicemail = yes
register => sip:1000@192.168.2.151
[voipms]
type=registration
outbound_auth=voipms
server_uri=sip:192.168.2.151
client_uri=sip:1000@192.168.2.151
auth_rejection_permanent=no

[voipms]
type=auth
auth_type=userpass
username=1000
password=1_C5l4h_6_58~@85
realm=192.169.2.151
[voipms]
type=aor
contact=sip:192.168.2.151
qualify_frequency=60

[voipms]
type=endpoint
context=inbound-calls
disallow=all
allow=g729,alaw
;auth=voipms
outbound_auth=voipms
aors=voipms
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
from_user=1000
from_domain=192.168.2.151
direct_media= no
outbound_proxy=sip:192.168.2.151
dtmf_mode = rfc4733
[voipms]
type=identify
endpoint=voipms
match=192.168.2.151

my extensions.conf

[general]
static=yes
writeprotect=yes
clearglobalvars=yes
language=fr

[googletts]
; si ça ne répond pas, redirige sur ce contexte
exten => _6XX,1,Answer()
exten => _X.,n,Set(CHANNEL(language)=fr)
exten => _X.,n,agi(googletts.agi,“Le numéro composé n’est pas attribué, ou n’est pas disponible pour le moment. Bonne journée à vous.”,fr,1.25)
exten => _X.,n,NoOp(depuis poste ${CALLERID(num)} : appel de ${EXTEN})
exten => _X.,n,Wait(.5)
exten => _X.,n,HangUp()

; Le contexte “privilegies” permet de passer des appels des postes 600 à 603, donc ceux d’ici peuvent appeler tous les utilisateurs
[privilegies]
exten => _6XZ,1,NoOp(appelant : ${CALLERID(num)} → ${EXTEN})
same => n,GotoIfTime(08:00-03:00,mon-sat,,?ouvert) ;Fait référence aux heures d’ouveture à l’isep
same => n,agi(googletts.agi,“notre entreprise n’est ouverte que du lundi au vendredi de 08h a 19h. Merci de rappeler plus tard”,fr,1.25) ;texte en cas de fermeture
same => n,HangUp()
same => n(ouvert), Set(CHANNEL(language)=fr) ;si isep est ouvert alors tout le monde peut s’appeler entre eux
same => n,Set(CHANNEL(musicclass)=mp3)
same => n,DIAL(PJSIP/${EXTEN},30,trm)
same => n,Voicemail(${EXTEN}@privilegies)
same => n,HangUp()

; messagerie pour “privilegies”
exten => 8000,1,Answer()
same => n,Set(CHANNEL(language)=fr)
same => n,VoiceMailMain(${CALLERID(num)}@privilegies)

include => googletts
include => inbound-calls

; LE contexte “standard” permet d’émettre des appels des postes 602 à 603
[standard]
exten => _X.,1,NoOp(appelant : ${CALLERID(num)} → ${EXTEN})
same => n,GotoIfTime(08:00-03:00,mon-sat,,?ouvert) ;Fait référence aux heures d’ouveture à l’isep
same => n,agi(googletts.agi,“notre entreprise n’est ouverte que du lundi au vendredi de 08h a 19h. Merci de rappeler plus tard”,fr,1.25) ;texte en cas de fermeture
same => n,HangUp()
same => n(ouvert), Set(CHANNEL(language)=fr) ;si isep est ouvert alors tout le monde peut s’appeler entre eux
same => n,Set(CHANNEL(musicclass)=mp3)
same => n,DIAL(PJSIP/${EXTEN},30,trm)
same => n,Voicemail(${EXTEN}@standard)
same => n,HangUp()

exten => 600,1,DIAL(PJSIP/602,30,tT), quand un poste essayer d’appeler le directeur depuis le contexte standard, il sera redirigé vers le sécrétaire
exten => 602,2,HangUp()

; messagerie pour “standard”
exten => 8001,1,Answer()
same => n,Set(CHANNEL(language)=fr)
same => n,VoiceMailMain(${CALLERID(num)}@standard)

include => googletts
include => inbound-calls

[inbound-calls]

exten => _X.,1,Dial(PJSIP/601,30,tT)
same => n,Hangup()

include=>privilegies
include=>standard

2

I thought transport was mandatory, so I don’t understand how it is even reading the incoming requests.

Does pjsip have a general section? It certainly doesn’t have register => lines.

Using the name of a SIP provider for local hardware that they don’t supply is confusing.

Are you not able to configure a proper, static address, trunk on the yeastar?

Could you please explain why you think you need rtp_symmetric, force_rport, and rewrite_contact?

3

In fact my yeastar is connected to my modem router (which was given by a provider) which came with a telephone line (PSTN), now I put the yeastar in the network of my router. that’s why I have this address

I have set up a local asterisk server which is also in the network of my modem router and the yeastar. With my phone line, I would like to create a trunk with the yeastar and my asterisk to be able to make outgoing calls from my asterisk

5

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