SIP extension without username and password?

Asterisk Asterisk Support
1

We have several servers that connect to our main Asterisk server, treating it as a SIP/IAX2 gateway. One of the servers needs to connect without a username and password. Also, the server needs to communicate on port 6058 instead of port 5060.

I thought I had everything setup correctly but it looks like the servers are not speaking to eachother. Running asterisk -vvvr and sip debug I see the call going from the gateway to the server but the server never sees it. Going from the server to Asterisk they see it going through on their end but Asterisk never sees anything.

Systems can ping eachother, IP address is correct on both systems, both systems are on the same subnet, no NAT involved, no firewall on either system, server can connect with a SIP server in china correctly. Other servers in our NOC can communicate with Asterisk using SIP or IAX without error.

Anyone have any hints on how to get this working correctly?

sip.conf--------------
[server100]
type=user
username=server100
trunk=yes
notransfer=yes
qualify=no
context=phones-outbound-nold
host=###.###.###.### (removed for this post)
bindport=6058
call-limit=2
insecure=very
[server100]
type=peer
username=server100
trunk=yes
notransfer=yes
qualify=no
context=from-pstn
host=###.###.###.### (removed for this post)
bindport=6058
call-limit=2
insecure=very

extensions.conf------------------------
;Server100
exten => _71355512XX,1,wait,2
exten => _71355512XX,2,Dial(SIP/${EXTEN}@ClientIPAddress|60|)

2

If you are not authenticating the server, why do you need a username?

3

I’ve tried it both with and without.

4

I always choose the path of least resistance :smile: If most other servers use auth, adding user name password in one shouldn’t be too painful.

5

Unfortunately we’ve got a customer with a non-standard install who can’t or won’t pass authentication. I know it can be done, but apparently the way I am doing it is either incorrect or their config won’t simply won’t work with Asterisk.

6

Guess it’s time to roll up the sleeves and CLI> sip debug :cry:

7

Already been through sip debug. On an inbound call from the server Asterisk never sees anything, as if it never happened. On outbound everything looks normal until the 60 second timeout we have ends and then it dies with cause 18.

Thanks anyway, we’re just going to tell the client they need to fix it on their end because all of our standard SIP clients are working normally.

8

Any chance a sniffer catches a SIP packet?