Asterisk 18 pjsip "No matching endpoint found"

Asterisk Asterisk SIP
1

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()

2

I can’t find any type=identify sections.

3

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

4

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

5

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 "

6

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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7

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