SOLVED Asterisk 13.6.0 ignores from_user / from_domain settings

[bofh42]

IncrediblePBX 13-12-3 for RaspberryPi 2

FreePBX 12.0.70 / IncredibleGUI 12.0.30
Asterisk 13.6.0
Raspbian Wheezy
chan_pjsip

all updates installed

Problem: Asterisk always send REGISTER and OPTIONS packets to the trunk with the
"default_from_user" instead of using the values for "from_user" and "from_domain" given in the config files

The result is that it is impossible to perform outbound calls due to not being registered

OPTIONS sip:[email protected]:5060 SIP/2.0
            192.168.80.12:5060             217.0.23.36:5060 │Via: SIP/2.0/UDP 84.178.92.172:5060;rport;branch=z9hG4bKPjb545c814-7679-474c-9d31-48a31d
          ──────────┬─────────          ──────────┬─────────│c579
                    │           OPTIONS           │         │From: <sip:[email protected]>;tag=4b740498-65f5-4156-9ab5-5474fe865813
  20:44:21.463773   │ ──────────────────────────> │         │To: <sip:[email protected]>
        +0.014493   │        403 Forbidden        │         │Contact: <sip:[email protected]:5060>
  20:44:21.478266   │ <────────────────────────── │         │Call-ID: 2545a727-0740-47b5-9f0a-34844b6f3d13
                    │                             │         │CSeq: 19859 OPTIONS
                    │                             │         │Max-Forwards: 70
                    │                             │         │User-Agent: FPBX-12.0.70(13.6.0)
                    │                             │         │Content-Length:  0

pjsip.endpoint.conf:from_user=05173xxx



[Telekom-xxx]
type=endpoint
transport=0.0.0.0-udp
context=ext-did-0002
disallow=all
allow=g722,ulaw,alaw,gsm,g729,g723,g726,speex,speex16,speex32
outbound_auth=Telekom-xxx
aors=Telekom-xxx
direct_media=no
from_domain=tel.t-online.de
from_user=05173xxx

(it can be put in the pjsip.registration.conf as well, that does not help either...)

raspberrypi*CLI> pjsip show registration Telekom-xxx

.....
force_avp                     : false
 force_rport                   : true
 from_domain                   :
 from_user                     :
 g726_non_standard             : false
.....

well ....

is there any update path to anything newer and (hopefully) bug-free?

 

[Jake]

For starters try using just Chan SIP for your trunks. I've yet to have success with PJSIP and trunks.

 

[bofh42]

Whatever I try with chan_sip - and I did try for weeks, using any howto I could find - it always ends with

[2017-06-17 08:51:44] ERROR[1475]: chan_sip.c:4181 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
[2017-06-17 08:51:44] NOTICE[1475]: chan_sip.c:15347 sip_reg_timeout:    -- Registration for '[email protected]' timed out, trying again (Attempt #2204)

so that is not an option.

 

[bofh42]

looks like

from_user=
from_domain=

have to be present in

pjsip.endpoint.conf / type=endpoint

FreePBX 12.x as shipped in Incredibple Pi 2 does not write it at all, in none of the config files

additional

contact_user=

has to be present in

pjsip.registration.con / type=registration

FreePBX 12 as shipped in PIAF writes it to

type=endpoint / pjsip.endpoint.conf

, which is WRONG.

Any mistake in this stops things from working

If parameters for type=registration are put into type=endpoint, the whole configuration for that item is silently (!!) ignored. Therefore, the wrong place, FreePBX writes contact_user to - stops the endpoints being configured...

The underlying problem here is, that even in the official asterisk wiki for v13.x on asterisk.org, the parameter contact_user is listed under type=endpoint (and type=registration), where it may not be put - it disables the endpoint.

Having made these changes, pjsip no works flawlessly.

But: every change in the FreePBX GUI + save destroys these settings again - after saving from the GUI, you have to correct all type=endpoint settings for all trunks and reload asterisk again from the command line.

 

[wardmundy]

Solution: Don't use PJsip for trunks. That's been the recommendation for at least the last 2 years.

 

[bofh42]

and again: after several weeks of trying: I could NOT get chan_sip to work in any way with any of the trunks.

Nor could I get any help beyond: that's easy and works.

chan_pjsip now works perfectly