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[serov.d.p at gmail.com]

Hello.

I am using asterisk and chan_sip a lot of years. And newbie in chan_pjsip.
Now i am transfering all from chan_sip to chan_pjsip. And have a lot of
questions. First of...

system: Asterisk 13.2 on slackware 14.1
Errors on outgoing call:
[2015-03-03 00:18:58] ERROR[6794]: chan_pjsip.c:1778 request: Failed to
create outgoing session to endpoint 'srv_d228'
[2015-03-03 00:18:58] WARNING[24528][C-00001bcc]: app_dial.c:2431
dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No
route to destination)

What settings has mistake? What logic to choose outgoing transport?

[transport-udp]
type=transport
bind=0.0.0.0:5070
protocol=udp

[srv_d228]
type=endpoint
language=ru
rtp_symmetric=yes
force_rport=yes
disable_direct_media_on_nat=yes
rewrite_contact=yes
ice_support=yes
disallow=all
allow=alaw

context=ext-fromservers
from_domain=sipnet.ru
from_user=talk37.ru
aors=srv_d228
auth=srv_d228
set_var=fromDeviceId=228
set_var=fromUserId=2
outbound_auth=srv_d228
;outbound_proxy=sip:sipnet.ru:5060
transport=transport-udp

[srv_d228]
type=aor
qualify_frequency=30
contact=sip:sipnet.ru:5060
;outbound_proxy=sip:sipnet.ru:5060
max_contacts=10
remove_existing=yes

[srv_d228]
type=auth
auth_type=userpass
username=talk37.ru
password=secret

[srv_d228]
type=registration
transport=transport-udp
outbound_auth=srv_d228
server_uri=sip:sipnet.ru
client_uri=sip:talk37.ru@sipnet.ru
retry_interval=60
;auth_rejection_permanent=no
contact_user=srv_d228

pjsip show registrations
<Registration/ServerURI..............................>
<Auth..........> <Status.......>
=========================================================================================
srv_d228/sip:sipnet.ru srv_d228 Registered

pjsip show endpoints
Endpoint: srv_d228 Not in
use 0 of inf
OutAuth: srv_d228/talk37.ru
InAuth: srv_d228/talk37.ru
Aor: srv_d228 10
Contact: srv_d228/sip:sipnet.ru:5060 Avail 9.858

Thanks!
Dmitriy Serov


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[serov.d.p at gmail.com]

Sorry, i found the source of problem.
https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels
dialing via pjsip have to change dialplan syntax :(

May be it will be usefull sombody else.

04.03.2015 21:54, Dmitriy Serov пишет: