Getting auth errors when trying to route calls via asterisk

Asterisk Asterisk SIP
1

Hi guys, I’m having trouble getting these endpoints to be able to call eachother. When I try to make calls, asterisk is giving me auth errors. Any help would be greatly appreciated. If anyone knows of any definitive documentation that explains the possible fields, their possible values, and how they cause asterisk to behave I would love that. That said, here is the relevent config:

pjsip.conf:

[7007]
type = endpoint
context = internal
allow = all
aors = 7007
auth = auth7007
;dtmf_mode = rfc2833

[7007]
type = aor
max_contacts = 1

[7007]
type = identify
match = 192.168.200.111

[auth7007]
type=auth
auth_type=userpass
password=7007
username=7007

[4008]
type = endpoint
context = internal
allow = all
aors = 4008
;dtmf_mode = rfc2833

[4008]
type = aor
contact = sip:4008@192.168.200.2:5060

[4008]
type = identify
match = 192.168.200.2

[1]
type = endpoint
context = internal
allow = all
aors = 1
;dtmf_mode = rfc2833

[1]
type = aor
contact = sip:1@192.168.200.1:5060

[1]
type = identify
match = 192.168.200.1

[4010]
type = endpoint
context = internal
allow = all
aors = 4010
;dtmf_mode = rfc2833

[4010]
type = aor
contact = sip:4010@192.168.200.2:5060

[4010]
type = identify
match = 192.168.200.2

extensions.conf:

exten=>4008,1,Dial(SIP/OUT1/$(EXTEN),20,r)
exten=>1,1,Dial(SIP/OUT2/$(EXTEN),20,r)
exten=>4010,1,Dial(SIP/OUT3/$(EXTEN),20,r)
exten=>7007,1,Dial(SIP/7007,20,r)

2

You should always include console output of attempts including SIP traffic, otherwise we’re relying on your analysis and have to guess at times.

What exactly are you looking for information on? The pjsip.conf configuration? There is a sample file[1], and it is documented on the docs site[2]. Your extensions.conf is also written to use chan_sip, not chan_pjsip. Is that expected?

[1] asterisk/configs/samples/pjsip.conf.sample at master · asterisk/asterisk · GitHub
[2] res_pjsip - Asterisk Documentation

3

Thanks for replying. I intend to use pjsip… how should i set up extensions.conf for that?

4

https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/Dialing-PJSIP-Channels/

5

Thank you

6

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