Inbound VOIP SIP calls is rejected

Asterisk Asterisk Support
1

Dear All,

Please help me on a problem that I am facing as below:

Basically, I am trying to achieve the below scenario:

softswitch1 —> asterisk 1.6 —> softswitch2

from the asterisk, it has already managed to register itself to softswitch2 and from an extensions or clients in asterisk 1.6 are able to terminate a call to softswitch2.

But when I tried to make a call from softswitch1 to asterisk, I am getting the below error for the CLI commands:

May 26 18:45:53] NOTICE[3262]: chan_sip.c:19990 handle_request_invite: Failed to authenticate device 1234 sip:192.168.2.14;tag=8690e9b46762453781e9da1598aa51c1

Please do help me why is this issue happening?

Thank You.

2

Looks like your username and/or password is incorrect for softswitch 1…

3

Actually it looks like the softwitch has the wrong user and password for accessing Asterisk!

4

Dear David55/ Mket,

Thanks for replying my post.

Sorry for not clearing up the call scenario.

Actually, the scenario is as below:

sip device1/client1 --> softswitch1 --> asterisk 1.6 --> softswitch2 --> sip device2/ client2

sip device1/client1 are already registered to softswitch1 and the same goes to sip device2 to softswitch2.

And also asterisk 1.6 has successfully registered to softswich2 as well.

A client device registered on asterisk 1.6 can terminate a successful call to sip device2/ client2 through softswitch2.

But when I tried passing a call from sip device1 through softswitch1 to the asterisk 1.6 I have the below mentioned error code:

May 26 18:45:53] NOTICE[3262]: chan_sip.c:19990 handle_request_invite: Failed to authenticate device 1234 sip:192.168.2.14;tag=8690e9b46762453781e9da1598aa51c1

If the error meant that the username and password is wrong, does that means that if I need to pass a call to asterisk, I need to register the softswitch1 to asterisk as well?

Appreciate all for helping me.

Thank You.

5

If there is a secret in the sip.conf entry for softwswitch1, softswitch1 needs to be configured to use that password.

Please get a SIP trace of the dialogue between softswitch1 and Asterisk. Also please provide the sip.conf entries for general and softswitch1.

6

Dear All,

Thanks for helping me out here. Sorry for the late reply as it seems like the forum is down for a few days until now.

I have tried the following method instead:

sip/ client device1 —> asterisk 1.6 ----> softswitch2 ----> sip/client device2

sip device1 registered to asterisk and sip device2 registered to softswitch2.

The call flow mentioned above is already working fine.

Then I tried the reversal:

sip/client device2 --> softswitch2 --> asterisk 1.6 —> sip/ client device1

and guess what… the same error messages still appears in the CLI:

[Jun 2 20:04:55] NOTICE[3612]: chan_sip.c:19990 handle_request_invite: Failed to authenticate device 1234 sip:172.28.0.133;tag=21aa2c94b18442a099e7059715ca9d87

I tried tracing the call using wireshark, and basicall I can see that the softswitch2 is sending an INVITE to the asterisk 1.6, and immediately after that the asterisk 1.6 replied with a “403 Forbidden” , do let me know if you guys would like to trace file to look at.

And below are the sip.conf entries for general and the softswitch1 (cpgw1.cpm.com.my)

/etc/asterisk/sip.conf

[general]
context=from_outside
realm=CPMDOMAIN
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
allow=gsm
language=en

register => 60998888:1234@172.28.0.133

[xlite1]
type = friend
host = dynamic
secret = xlite1
context = users

[xlite2]
type = friend
host = dynamic
secret = xlite2
context = users

[xlite3]
type = friend
host = dynamic
secret = xlite3
context = users

[cpgw1.cpm.com.my]
type = peer
username = 60998888
fromuser = 60998888
fromdomain = CPMDOMAIN
secret = 1234
canreinvite = no
insecure = invite,port
host = 172.28.0.133
deny = 0.0.0.0/0
permit = 172.28.0.0/255.255.255.255
disallow = all
allow = gsm
allow = ulaw
allow = alaw
qualify = yes
nat = no

and

/etc/asterisk/extensions.conf

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[global]

[users]

[from_outside]
include => inside

[inside]
exten => 6001,1,Dial(SIP/xlite1,10)
exten => 6002,1,Dial(SIP/xlite2,10)
exten => 6003,1,Dial(SIP/xlite3,10)

Thanks again. Appreciate for all your help.

7

permit = 172.28.0.0/255.255.255.255 is wrong.

8

Dear david55,

Thanks for looking into my case.

That was the problem all along. Once I changed it to:

permit = 172.28.0.0/255.255.255.0

Its working perfectly fine. Even on the 1st transit/passthrough scenario.

A simple silly mistake that I made.

Appreciate all the help that you guys have given to me.

Thank You.