Asterisk and CommPeak integration - Failed to authenticate

Asterisk Asterisk Support
1

Hi all,

I’m trying to connect my Asterisk PBX with CommPeak VoIP provider to be able to place calls directory to landline phones that do not support VoIP.

The scenario is as follows:
Mobile phone ( zoiper ) - my asterisk pbx - commbeap - landline phone number

When I try to dial a ‘real’ phone number provided to them by PSTN, I get the following error:

== Using SIP RTP CoS mark 5 -- Executing [1234567890@sip_test_call:1] Dial("SIP/mijo-00000025", "SIP/commpeak/1234567890") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/commpeak/1234567890 [Sep 9 11:06:51] NOTICE[13067]: chan_sip.c:20354 handle_response_invite: Failed to authenticate on INVITE to '"mijo" <sip:mijo@192.168.x.y>;tag=as77e02638' -- SIP/commpeak-00000026 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0)
I somewhere read that I need to put the insecure=invite option but that didn’t help either.

My sip.conf

[code][general]
allowguest=no
udpbindaddr=0.0.0.0:5060
srvlookup=yes
nat=yes
qualify=yes
videosupport=yes
directmedia=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723
allow=gsm
allow=ilbc

[mijo]
type=friend
context=sip_test_call
secret=****
host=dynamic
insecure=invite

[commpeak]
host=sip.commpeak.com
type=peer
qualify=yes
username=**********
secret=*********
context=commpeak[/code]

Thanks a lot,
Best regards

2

You’ve provided the wrong authentication data.

insecure=invite is for incoming calls and should no longer be necessary with remotesecret.

“username” is deprecated because it is misleading. Most systems that insist on a user name want it as fromuser, but they might actually want authuser.

3

Hi David,

I have tried your suggestions but with no luck.
As you sad, it seems like an authentication issue but I’m in no position to figure out what’s wrong.

I have double checked my SIP profile on CommPeak and I’m using username / password authentication instead of IP based authentication.

The message I get is following:

== Using SIP RTP CoS mark 5 -- Executing [098873455@test_calls:1] Dial("SIP/mijo-00000008", "SIP/commpeak/098873455") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/commpeak/098873455 [Sep 15 18:10:17] NOTICE[1754]: chan_sip.c:20354 handle_response_invite: Failed to authenticate on INVITE to '"mijo" <sip:mijo@192.168.1.101>;tag=as549be759' -- SIP/commpeak-00000009 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/mijo-00000008' status is 'CONGESTION'

My sip.conf

[general]
allowguest=no
udpbindaddr=0.0.0.0:5060
srvlookup=yes
nat=yes
qualify=yes
videosupport=yes
directmedia=no
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723
allow=gsm

[mijo]
type=friend
context=test_calls
secret=**********
host=dynamic

[commpeak]
host=sip.commpeak.com
type=peer
qualify=yes
disallow=all
allow=g729
allow=ulaw
username=xxx
secret=yyy
context=commpeak

My extensions.conf

[code][general]
autofallthrough=yes
writeprotect=no

[sip_pozivi]
exten => 101,1,Dial(SIP/mijo)

exten => 102,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()

exten => _X.,1,Dial(SIP/commpeak/${EXTEN})[/code]

Do you need any more debug messages or something else for further determination of what could be wrong ?

Thanks in advance,
Best regards.

4

sip debug output and a statement from the people responsible for the other end as to what should actually be in the INVITE.

Do they need you to register?