Can't Connect to Sip Trunk

[Alex Sage]

[2016-02-09 14:18:25] WARNING[1667] chan_sip.c: 'tcp' is not a valid transport type when tcpenable=no. If no other is specified, the defaults from general will be used.
[2016-02-09 14:18:25] WARNING[1667] chan_sip.c: 'tls' is not a valid transport type when tlsenable=no. If no other is specified, the defaults from general will be used.
[2016-02-09 14:18:25] WARNING[1667] chan_sip.c: 'tcp' is not a valid transport type when tcpenable=no. If no other is specified, the defaults from general will be used.
[2016-02-09 14:18:25] WARNING[1667] chan_sip.c: 'tls' is not a valid transport type when tlsenable=no. If no other is specified, the defaults from general will be used.
[2016-02-09 14:18:26] WARNING[1667] chan_sip.c: Section 'vitelity' lacks type
[2016-02-09 14:18:26] VERBOSE[1667] config.c: Parsing '/etc/asterisk/sip_notify.conf': Found
[2016-02-09 14:18:26] VERBOSE[1667] config.c: Parsing '/etc/asterisk/sip_notify_custom.conf': Found
[2016-02-09 14:18:26] VERBOSE[1667] config.c: Parsing '/etc/asterisk/sip_notify_additional.conf': Found
[2016-02-09 14:18:46] NOTICE[1667] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #2)
[2016-02-09 14:19:06] NOTICE[1667] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #3)
[2016-02-09 14:19:26] NOTICE[1667] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #4)
[2016-02-09 14:19:46] NOTICE[1667] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #5)

[gjacobse]

Tagging
@Mike-Ralston and @art_of_shred

[Alex Sage]

What steps should I try to troubleshoot this?

There is a proxy, however I have added it to FreePBX.

The ports are open. I can't ping inbound35.vitelity.net, however with the proxy I am not sure if I should be able to or not....

[art_of_shred]

Is tcp correct? Shouldn't that be udp for that traffic?

[art_of_shred]

Instead of ping, try a traceroute to the server. See what it hits, where it stops, etc.

[Alex Sage]

@art_of_shred said:

Is tcp correct? Shouldn't that be udp for that traffic?

Um.... I think so? Sorry, I am not a SIP expert...

[coliver]

Vitelity has the settings required on their website although you need to specify both inbound and outbound. I think it should be UDP.

[art_of_shred]

@coliver Beware the accuracy of what they list on their site. Try it, but you may end up needing to open a support ticket to get the most current settings. I've encountered that before, more than once.

[coliver]

@art_of_shred said:

@coliver Beware the accuracy of what they list on their site. Try it, but you may end up needing to open a support ticket to get the most current settings. I've encountered that before, more than once.

That's right, I remember you guys saying something like that. It's unfortunate they can't keep those settings up-to-date.

[Alex Sage]

Well that's not good. Maybe the proxy is in the way? or is it more likely a firewall issue?

Sadly, I don't have access to the firewall, but I have to email HQ.

traceroute to inbound35.vitelity.net (66.241.99.208), 30 hops max, 60 byte packe                 ts
 1  * * *
 2  * * *
 3  * * *
 4  * * *
 5  * * *
 6  * * *
 7  * * *
 8  * * *
 9  * * *
10  * * *

[Alex Sage]

I opened a ticket for the most up to date settings to use with FreePBX.

[Alex Sage]

Well that was quick:

Hello,

This is a courtesy message from Vitelity Communications regarding an active Trouble Ticket you have in our system for the account 'XXXXXXXX'.

Your Trouble Ticket has been updated with a response/resolution to your problem.

Posted by vcdreece on 2016-02-09 08:31:26
Hello,

You will want your trunk settings to look like this:

[vitel-inbound]
type=friend
dtmfmode=auto
host=inbound35.vitelity.net
context=from-trunk
username=
secret=
allow=all
insecure=port,invite
canreinvite=no

[vitel-outbound]
type=friend
dtmfmode=auto
host=outbound.vitelity.net
username=
fromuser=
secret=
trustrpid=yes
sendrpid=yes
allow=all
canreinvite=no

register => rfna_XXXX:[email protected]:5060

Please let us know if we can be of any further assistance.

Sincerely,
David
Network Technician

Please visit https://portal.vitelity.net/login.php?show=tickets for more information regarding this ticket.

Sincerely,

Vitelity Communications
[email protected]

[dafyre]

@anonymous said:

Well that was quick:
<snip>

The question is: Were you able to get it going again?

[Alex Sage]

@dafyre No, still not working

[Alex Sage]

Sent a email to HQ to confirm the ports are open.....

[scottalanmiller]

@art_of_shred said:

Is tcp correct? Shouldn't that be udp for that traffic?

SIP is TCP. RTP is normally UDP.

[scottalanmiller]

@anonymous said:

Well that's not good. Maybe the proxy is in the way? or is it more likely a firewall issue?

Do you have non-web traffic going through the proxy? If so, does it pass it?

[Alex Sage]

@scottalanmiller said:

SIP is TCP. RTP is normally UDP.

It is? HQ said they opened udp only.....

[scottalanmiller]

@anonymous said:

@scottalanmiller said:

SIP is TCP. RTP is normally UDP.

It is? HQ said they opened udp only.....

You CAN do SIP on UDP. Not many places do.

[Alex Sage]

@scottalanmiller Well that could be a problem