Please help with sip_reg_timeout

[gibrahimi]

Hi guys !
Recently I have downloaded goautodial-ce-2.0-final.iso image and installed it in virtual machine.
I did the setup for a SIP carrier following the guidance and always I get the same error sip_reg_timeout.
The command:sip show peers gives the status UNREACHABLE

The strange thing is that if I use the same sip trunk parameters in AsteriskNOW I connect succesfully from the same network (my home network).

Here below is my carrier configuration:

Registration String: register => xxxx:yyyy@mysip.provider.com:5060

[test1]
username=xxxx
type=peer
secret=yyyy
insecure=port,invite
host=mysip.provider.com
dtmfmode=rfc2833
canreinvite=no
canredirect=no
qualify=yes
context=trunkinbound

Please help !

[robin]

can you ping your sip provider from the server?

[gibrahimi]

I can't ping it from anywhere because they closed the ICMP port,but as I wrote in my post if I use the same config file with AsteriskNOW I can get registered and make calls.

Thanks !

[robin]

I noticed that, I was just wondering if your new machine has access to the web at all...

I'm oldskool, I always start with the plug

[gibrahimi]

If I activate sip debug I see in From field a value starting with word "asterisk".
How can I change that ?
I think to do that because apart the config I posted and used in AsteriskNOW I have changed the parameter useragent from Asterisk to something else and that instruction was given by my sip provider.

Thanks !

[robin]

That would be the useragent setting in sip.conf

[gibrahimi]

That would be the useragent setting in sip.conf

Thanks robin,for the AsteriskNOW I used that trick and I use the same with goautodial in my sip.conf

I still cannot get registered !

[robin]

But do you have an internet connection on the box? I know, it seems like a stupid question, but as I said, I'm oldskool...

Can you reach other domains? Do you have a dns?

[gibrahimi]

But do you have an internet connection on the box? I know, it seems like a stupid question, but as I said, I'm oldskool...

Can you reach other domains? Do you have a dns?

Yes,I do have internet connection. I can ping google and other hosts.

my /etc/resolv.conf
nameserver 8.8.8.8
nameserver 8.8.4.4

I tried to use also the ip instead of hostname ,but I cant be registered again.

[williamconley]

Code: Select all

ping vicidial.com -c 2

post results

[gibrahimi]

Code: Select all

ping vicidial.com -c 2

post results

Here below the output:
PING vicidial.com (208.38.149.188) 56(84) bytes of data.
64 bytes from public.vicihost.com (208.38.149.188): icmp_seq=1 ttl=40 time=278 ms
64 bytes from public.vicihost.com (208.38.149.188): icmp_seq=2 ttl=40 time=310 ms

--- vicidial.com ping statistics ---
2 packets transmitted, 2 received, 0% packet loss, time 1280ms
rtt min/avg/max/mdev = 278.702/294.469/310.236/15.767 ms

[robin]

Did you turn the AsteriskNOW off? Maybe you cannot register twice?

Did you contact your sip provider and ask them what they see when you try to register?

[gibrahimi]

Did you turn the AsteriskNOW off? Maybe you cannot register twice?

Did you contact your sip provider and ask them what they see when you try to register?

For sure that I turned off AsteriskNOW before switching on the goautodial machine.
I'm still waiting an answer from provider,but in the same time I'm comparing the sip debug messages of AsteriskNOW with Goautodial.
At AsteriskNOW ,in the sip header "FROM" I see always "Unknown" in the begging of the line,while at goautodial it starts with Asterisk.
Maybe that is the cause.

[robin]

Maybe you should wait for the answer of you sip provider so you can dig deeper...

I know, waiting sucks.

Although I find it strange you have to change the useragent...

[gibrahimi]

This is going to make me crazy ! It looks that the problem relays on SIP provider side even they didn't answered yet.
I open the AsteriskNOW and changed sip trunk config and reloaded sip.Switched off asterisknow and switched on goautodial and now I see that it is registered ! It is really strange why,but in the same time the sip show peers displays the UNREACHABLE status and I cannot make calls.
I get this error:

app_dial.c:1296 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)

Maybe I have to wait the answer from my SIP provider.