Another Failed to authenticate on INVITE d

Asterisk Asterisk Dialplan
1

After several days of searching I haven’t found a solution that works.

Outgoing calls fail with CLI:

Using SIP RTP CoS mark 5
– Executing [14135551212@default:1] Dial(“SIP/8180-0000001b”, “SIP/14135551212)@Bulk”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/14135551212)@Bulk
== Using SIP RTP CoS mark 5
[2024-08-28 00:32:26] NOTICE[11568][C-00000018]: chan_sip.c:26826 handle_request_invite: Call from ‘’ (74.69.41.245:5060) to extension ‘14135551212)’ rejected because extension not found in context ‘default’.

extensions.conf:
exten => _1XXXXXXXXXX,1,Dial(SIP/${EXTEN})@Bulk)

I also tried:
exten => _1XXXXXXXXXX,1,Dial(SIP/${EXTEN})@sip.bulkvs.com)

sip.conf:
[Bulkvs]
type=peer
context=default
host=sip.bulkvs.com
username=me
secret=xxxxxx
qualify=yes
disallow=all
allow=ulaw
dtmfmode=rfc2833
rfc2833compensate=yes
insecure=port,invite
trustrpid=yes
realm=sip.bulkvs.com
fromdomain=sip.bulkvs.com

The provider says the invite request doesn’t respond with Digest info (username/password).

Any suggestions?

Thanks.

2

Why are you using an unsupported and obsolete channel driver?

There is a bogus “)” in your dial string.

You don’t have a context called “Bulk”.

Your outgoing call seems to have been routed to the originating Asterisk system.

username is a deprecated name for the parameter, is rarely needed, and is not used when an explicit host is specified.

Why do you need insecure=port?

This is not consistent with the log provided, which appears to show the call not going to the provider at all. However, using a literal domain name in the dial string will mean that only the general section of sip.conf is used, so no password will be available, and the provider’'s response would be correct for that case.

3

Thank you, thank you. I’ve been working on this for days and have become a little groggy.

I never would have spotted the mistakes in the dial string. Fixed that and all is well.

Setting up pjsip was just too complicated for me. There aren’t any basic primers.

Thank you, thank you.

4

Looked good in the examples. But I don’t know what I’m doing. Really appreciate your patience.

5

Not working yet but seems to be a different problem. Not sure if this an unrelated problem or not.

None of my outgoing calls go through except to one particular number. The number rings, answering machine picks up, but no sound.

Using SIP RTP CoS mark 5
– Executing [14133586745@default:1] Dial(“SIP/8180-0000000e”, “SIP/14133586745@Bulkvs”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/14133586745@Bulkvs
– SIP/Bulkvs-0000000f is ringing
– SIP/Bulkvs-0000000f is making progress passing it to SIP/8180-0000000e
– SIP/Bulkvs-0000000f is making progress passing it to SIP/8180-0000000e
– SIP/Bulkvs-0000000f answered SIP/8180-0000000e
– Channel SIP/Bulkvs-0000000f joined ‘simple_bridge’ basic-bridge <944c0d6f-3f26-4df5-a823-cf7d473c701b>
– Channel SIP/8180-0000000e joined ‘simple_bridge’ basic-bridge <944c0d6f-3f26-4df5-a823-cf7d473c701b>
– Channel SIP/Bulkvs-0000000f left ‘native_rtp’ basic-bridge <944c0d6f-3f26-4df5-a823-cf7d473c701b>
– Channel SIP/8180-0000000e left ‘native_rtp’ basic-bridge <944c0d6f-3f26-4df5-a823-cf7d473c701b>
== Spawn extension (default, 14133586745, 1) exited non-zero on ‘SIP/8180-0000000e’

If I call any other number, nothing happens. This is the CLI output:

Using SIP RTP CoS mark 5
– Executing [1413xxxxxxx@default:1] Dial(“SIP/8180-00000010”, “SIP/1413xxxxxxx@Bulkvs”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/1413xxxxxxx@Bulkvs
– SIP/Bulkvs-00000011 answered SIP/8180-00000010
– Channel SIP/Bulkvs-00000011 joined ‘simple_bridge’ basic-bridge <00daba14-1541-4dd8-bb2a-431b84707c1a>
– Channel SIP/8180-00000010 joined ‘simple_bridge’ basic-bridge <00daba14-1541-4dd8-bb2a-431b84707c1a>
– Channel SIP/8180-00000010 left ‘native_rtp’ basic-bridge <00daba14-1541-4dd8-bb2a-431b84707c1a>
– Channel SIP/Bulkvs-00000011 left ‘native_rtp’ basic-bridge <00daba14-1541-4dd8-bb2a-431b84707c1a>
== Spawn extension (default, 14134425424, 1) exited non-zero on ‘SIP/8180-00000010’

sip.conf:
[Bulkvs]
type=peer
context=default
host=sip.bulkvs.com
username=6091
secret=xxxxx
qualify=yes
disallow=all
allow=ulaw
dtmfmode=rfc2833
rfc2833compensate=yes
;insecure=port,invite
trustrpid=yes
realm=sip.bulkvs.com
fromdomain=sip.bulkvs.com

6

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.