Asterisk 18 pjsip "No matching endpoint found"

Hi,
I had a perfectly running asterisk-PBX on an old version (I think 1.6) for years. I use two snom 320 hard-phones, a Zoiper soft-phone and a Patton sn4112 as Gateway to a single POTS-line from our telephone-provider.
After a crash of the PC I installed Asterisk 18 and now the phones can talk to eachother and All can call out via the Patton-GW. But if a call comes in, I get the message

– Contact sn4112/sip:sn4112@192.168.129.10:5060 is now Reachable. RTT: 18.226 msec
[Nov 30 17:49:33] NOTICE[5719]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request ‘INVITE’ from ‘sip:06643806107@192.168.129.12’ failed for ‘192.168.129.10:5060’ (callid: 47fd8fa8eaa56335) - No matching endpoint found

I have searched the internet for hours, but I have no clue. Any ideas?
TIA, Karl

I forgot, my PJsip.conf is:

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0

;;;;;;;;;;;;;;;; SN4112 ;;;;;;;;;;;;;;;;;;;;;;;;
[sn4112]
type=endpoint
transport=transport-udp
context=amt-in
disallow=all
allow=alaw
allow=ulaw
auth=sn4112
aors=sn4112
callerid=Amt

[sn4112]
type=auth
auth_type=userpass
password=2114
username=sn4112

[sn4112]
type=aor
max_contacts=2
remove_existing=yes
maximum_expiration=7200
minimum_expiration=60
qualify_frequency=60

[11]
type=endpoint
transport=transport-udp
context=z
disallow=all
allow=alaw
auth=11
aors=11

[11]
type=auth
auth_type=md5
md5_cred=4f6d6890e67c2a2e6420cbab568b1ec8
username=11

[11]
type=aor
max_contacts=2

[16]
type=endpoint
transport=transport-udp
context=p
disallow=all
allow=alaw
auth=16
aors=16

[16]
type=auth
auth_type=userpass ; md5
password=raskrask
;md5_cred=cff5b3ac28d1172fc82ca85eaefa62f9
username=16

[16]
type=aor
max_contacts=2

[25]
type=endpoint
transport=transport-udp
context=z
disallow=all
allow=alaw
auth=25
aors=25

[25]
type=auth
auth_type=md5
md5_cred=1619f6ca9247fb5728adf4206512c437
username=25

[25]
type=aor
max_contacts=2

The part of my extensions.conf, where the Patton comes in is:

[amt-in]
;Patton sends DID only or “0” if no DID

; with 0
exten => 0,1,Set(CALLERID(num)=0${CALLERID(num)}) ; fuegt Amtsholungskennziffer fuer Rueckruf ein
exten => 0,n,SIPAddHeader(“AlertInfo: http://192.168.129.12;info=alert-external;x-line-id=0”)
, changed 2017-07-12
exten => 0,1,Ringing()
exten => 0,n,hint,PJSIP/11
exten => 0,n,Dial(PJSIP/11,15,tT)
exten => 0,n,Playback(weiter)
exten => 0,n,Dial(PJSIP/11&PJSIP/16&PJSIP/25,15,tT)
exten => 0,n,Playback(sorry)
exten => 0,n,Set(box=11)
exten => 0,n,Goto(s-${DIALSTATUS},1)
exten => 0,n,Hangup()

I can’t find any type=identify sections.

Hi David,
since I did not see a reference to “identify” in the wiki’s example “Hello World” and not found much in the Asterisk-book, I did not see the necessity. Now I have added a section:>

[sn4112]
type=identify
endpoint=sn4112
match=192.168.129.0/24

where the match is my class-C network. I also tried the IP-adress of the Patton sn4112. Nothing changed - everybody can call eachother and call out to the PSTN, but calls from the PSTN do not come in.
Is my identify-section OK? Do I miss something else? How can I see which endpoint is looked for, that is not matching? Again, it all worked with my old asterisk, so I did not change the configurations of the phones or the gateway.
TIA, Karl

This is part of your problem, you must use outbound_auth=" use for outbound authentication
; instead of “auth=”, which is for inbound authentication.

For it to work without identify, the section name would need to be 06643806107, given the log contents.

Also it appears that the identify section will be ignored if " you have res_pjsip_endpoint_ identifier_ip loading after res_pjsip_endpoint_ identifier_user "

Thank you guys, after changing auth= to outbound_auth= it works! :smiley:
Sorry we can’t have a glass of :champagne: (or beer or wine) together.
Have a nice day and stay healthy!

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