G’day!
FreePBX 12.0.70 (in the flavor “Incredipble PBX v 12.0.30” distributed by Ward Mundy, PBX-in-a-Flash, PIAF)
Asterisk 13.6.0
Rasbian 7`
Problem:
a) SIP connecting in principle works from home net:
Test: Zoiper on PC with direct registration at VoIP trunk Germen Telekom, works in both directions
b) networking in principle works
incoming calls from Telekom (over VoIP trunk) are terminated on the Asterisk PBX, the correct endpoints ring, call can be accepted, voice works both ways, call termination ok
c) outgoing calls do NOT work.
Asterisk sends REGISTER to Telekom (or any other configured trunk, it’s the same with all of them), get a REJECT and does NOT send a new REGISTER with auth information.
No joy at all …
-- Executing [[email protected]:1] Macro("PJSIP/6009-00000028", "user-callerid,LIMIT,EXTERNAL,") in new stack
-- Executing [[email protected]:1] Set("PJSIP/6009-00000028", "TOUCH_MONITOR=1495794155.241") in new stack
-- Executing [[email protected]:2] Set("PJSIP/6009-00000028", "AMPUSER=6009") in new stack
-- Executing [[email protected]:3] GotoIf("PJSIP/6009-00000028", "0?report") in new stack
...
-- Executing [[email protected]:15] ExecIf("PJSIP/6009-00000028", "1?Set(DIAL_TRUNK_OPTIONS=M(setmusic^1FM-60s-today))") in new stack
-- Executing [[email protected]:16] ExecIf("PJSIP/6009-00000028", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^1FM-60s-today)M(confirm))") in new stack
-- Executing [[email protected]:17] Macro("PJSIP/6009-00000028", "dialout-trunk-predial-hook,") in new stack
-- Executing [[email protected]:1] MacroExit("PJSIP/6009-00000028", "") in new stack
-- Executing [[email protected]:18] GotoIf("PJSIP/6009-00000028", "0?bypass,1") in new stack
-- Executing [[email protected]:19] ExecIf("PJSIP/6009-00000028", "1?Set(CONNECTEDLINE(num,i)=017x564yyyy)") in new stack
-- Executing [[email protected]:20] ExecIf("PJSIP/6009-00000028", "1?Set(CONNECTEDLINE(name,i)=CID:01738964133)") in new stack
-- Executing [[email protected]:21] GotoIf("PJSIP/6009-00000028", "0?customtrunk") in new stack
-- Executing [[email protected]:22] Dial("PJSIP/6009-00000028", "PJSIP/[email protected],300,M(setmusic^1FM-60s-today)") in new stack
-- Called PJSIP/[email protected]
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [[email protected]:23] NoOp("PJSIP/6009-00000028", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
Call flow for db99ec3a-31cf-4d8c-8f5a-49a58c364a73 (Color by Request/Response)
│SIP/2.0 200 OK
192.168.80.12:5060 217.0.23.36:5060 │Via: SIP/2.0/UDP 84.178.yyy.zzz:5060;received=84.178.yyy.zzz;rport=5060;branch=z9hG4bKPj23966b68-40b2-
──────────┬───────── ──────────┬─────────│b1-8cac-8681eccad821
│ REGISTER │ │To: <sip:[email protected]>;tag=h7g4Esbg_92fb3a4d0e5db6f8e0c950f8c6a3522d
13:09:05.481550 │ ──────────────────────────> │ │From: <sip:[email protected]>;tag=7ab1ef92-2309-48ab-bd17-398bc690b5d4
+0.014514 │ 200 OK │ │Call-ID: db99ec3a-31cf-4d8c-8f5a-49a58c364a73
13:09:05.496064 │ <────────────────────────── │ │CSeq: 19985 REGISTER
+650.010981 │ REGISTER │ │Contact: <sip:[email protected]:5060>;expires=660
13:19:55.507045 │ ──────────────────────────> │ │P-Associated-Uri: <sip:[email protected]>
+0.014459 │ 200 OK │ │P-Associated-Uri: <tel:+4951XX92XZZZ>
13:19:55.521504 │ <────────────────────────── │ │Service-Route: <sip:217.0.23.36:5060;transport=udp;lr>
+650.016296 │ REGISTER │ │Content-Length: 0
13:30:45.537800 │ ──────────────────────────> │ │Authentication-Info: qop=auth,rspauth="a89e2fa5e75d1a74fdf4cae83023966e",cnonce="bab4ea33-27b0-40b9-
+0.014319 │ 200 OK │ │aa-83898e4be771",nc=00000001
13:30:45.552119 │ <────────────────────────── │ │
+650.013206 │ REGISTER │ │
13:41:35.565325 │ ──────────────────────────> │ │
+0.112846 │ 401 Unauthorized 010330345 │ │
13:41:35.678171 │ <────────────────────────── │ │
+0.018954 │ REGISTER │ │
13:41:35.697125 │ ──────────────────────────> │ │
+0.077489 │ 200 OK │ │
13:41:35.774614 │ <────────────────────────── │ │
+650.017069 │ REGISTER │ │
13:52:25.791683 │ ──────────────────────────> │ │
+0.014575 │ 200 OK │ │
13:52:25.806258 │ <────────────────────────── │ │
+650.035921 │ REGISTER │ │
14:03:15.842179 │ ──────────────────────────> │ │
+0.014282 │ 200 OK │ │
14:03:15.856461 │ <────────────────────────── │ │
+650.005191 │ REGISTER │ │
14:14:05.861652 │ ──────────────────────────> │ │
+0.014066 │ 200 OK │ │
14:14:05.875718 │ <────────────────────────── │ │
There must be a principle error in the Asterisk config - but … where?
It looks like Asterisk does not register correctly
at least the FreePBX generated pjsip*.conf file do look different from this:
https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
But: HOW do I enter the correct values into the FreePBX WebGUI? Changing asterisk conf files does not work of course, as the next “save” from the GUI will overwirte everything
[Telekom-XY49XXX]
type=registration
transport=0.0.0.0-udp
outbound_auth=Telekom-XY49XXX
retry_interval=60
expiration=3600
auth_rejection_permanent=no
contact_user=051XXXY49XXX
server_uri=sip:tel.t-online.de:5060
client_uri=sip:[email protected]:5060
pjsip.aor.conf
[Telekom-XY49XXX]
type=aor
qualify_frequency=60
contact=sip:[email protected]:5060
pjsip.auth.conf
[Telekom-XY49XXX]
type=auth
auth_type=userpass
password=
username=051XXXY49XXX
(password empty is the advised setting… Telekom authenticates only against the DSL line used)
pjsip.endpoint.conf
[Telekom-XY49XXX]
type=endpoint
transport=0.0.0.0-udp
context=ext-did-0002
disallow=all
allow=ulaw,alaw,gsm,g722,g729,g723,g726,speex,speex16,speex32
outbound_auth=Telekom-XY49XXX
aors=Telekom-XY49XXX
pjsip.identify.conf
[Telekom-XY49XXX]
type=identify
endpoint=Telekom-XY49XXX
match=tel.t-online.de
Any help appriciated
KInd regards