Asterisk not able to send OPTIONS

[ameynaik86]

Hi,

I am using asterisk 18.12.1 and trying to integrate it with ringcentral.

I am successfully able to register with ringcentral but my endpoint contact is not able to qualify.

I enabled debug logs on the console and below is the output while trying to send OPTIONS to qualify my contact:

[Jun 9 12:08:15] DEBUG[196967]: res_pjsip/pjsip_options.c:927 sip_options_qualify_aor: Qualifying all contacts on AOR ‘14699833569’
[Jun 9 12:08:15] DEBUG[196967]: res_pjsip/pjsip_options.c:856 sip_options_qualify_contact: Qualifying contact ‘14699833569@@146530725eb3f5297ee5daa50dd02a01’ on AOR ‘14699833569’
[Jun 9 12:08:15] DEBUG[196967]: res_pjsip.c:1566 endpt_send_request: 0x7f55882eb9c0: Wrapper created
[Jun 9 12:08:15] DEBUG[196967]: res_pjsip.c:1581 endpt_send_request: 0x7f55882eb9c0: Set timer to 3000 msec
[Jun 9 12:08:15] ERROR[196967]: res_pjsip.c:1619 endpt_send_request: Error 171005 ‘Missing route set (for tel: URI) (PJSIP_ENOROUTESET)’ sending OPTIONS request to endpoint 14699833569
[Jun 9 12:08:15] DEBUG[196967]: res_pjsip.c:1543 send_request_wrapper_destructor: 0x7f55882eb9c0: wrapper destroyed
[Jun 9 12:08:15] ERROR[196967]: res_pjsip/pjsip_options.c:911 sip_options_qualify_contact: Unable to send request to qualify contact sip:sip.ringcentral.com:5060 on AOR 14699833569

I am not able to understand what configuration i am missing here. Please help me with this.

[jcolp]

What is the actual complete configuration?

[ameynaik86]

Here is my actual configuration what i did,

[14699833569]
type = endpoint
context = trunkinbound
dtmf_mode = rfc4733
disallow = all
allow = all
rtp_symmetric = yes
rewrite_contact = yes
outbound_proxy = sip10.ringcentral.com:5090
from_user = 14699833569
from_domain = sip.ringcentral.com
auth = 14699833569
outbound_auth = 14699833569
aors = 14699833569
user_eq_phone=yes

[14699833569]
type = aor
max_contacts=10
contact = sip:sip.ringcentral.com:5060
outbound_proxy=sip:SIP10.ringcentral.com:5090
qualify_frequency=30
support_path=yes

[14699833569]
type = identify
endpoint = 14699833569
match = sip.ringcentral.com

[14699833569]
type = auth
username = 14699833569
password =

[reg_SIP10.ringcentral.com]
type = registration
retry_interval = 20
max_retries = 10
contact_user = 14699833569
expiration = 120
transport = transport-udp
outbound_auth = auth_reg_SIP10.ringcentral.com
client_uri = sip:14699833569@sip.ringcentral.com:5060
server_uri = sip:SIP10.ringcentral.com:5090

[auth_reg_SIP10.ringcentral.com]
type = auth
password =
username = 804224570011

[jcolp]

outbound_proxy on the endpoint does not start with “sip:”. For outbound_proxy you also want to do loose routing:

outbound_proxy = sip:SIP10.ringcentral.com:5090\;lr

Your endpoint should also have explicit codecs, and not use “allow=all”. I also see no reason why you should set max_contacts on the AOR.

[ameynaik86]

Thanks it worked didn’t noticed it was SIP URI.

The configuration i pasted was the output of sip_to_pjsip.py from asterisk source files. It has not appended sip: ahead of URI.