Shoretel <-> Asterisk SIP Trunk

[tubby]

I am trying to create a SIP trunk between my shortel switch and an asterisk server.

However, the sip connection never gets established and keeps timing out.


Does anyone know how to do this?

Thanks


Shoretel Switch IP: 192.168.12.6
Asterisk IP" 192.168.10.18

Shoretel Configuration
================

Trunk Groups
-------------
Name: Asterisk_SIP
Teleworkers: unchecked
Enable SIP Info for G.711 DTMF Signaling: Unchecked
Enable Digest Authentication
User ID: siptrunk
Passwordassword

Individual Trunk
---------------
Name: Asterisk_SIP
Trunk Group: Asterisk_SIP
Switch: Switch 1
SIP Trunk Type: Use IP Address 192.168.10.18


Asterisk Configuration (sip.conf)
========================
register = siptrunk[email protected]/siptrunk

[siptrunk]
type=friend
secret=password
context=shoretel_incoming
host=192.168.12.6
disallow=all
allow=ulaw
dtmfmode=rfc2833
username=siptrunk

[MiJaMu]

check this thread
http://www.shoretelforums.com/forums...read.php?t=220

~ MiJa

[1stSentinel]

OKay Checked the thread.... and after a little more digging here's what I came up with...

Shoretel Switch IP: SHORETEL VIRTUAL SWITCH IP
Asterisk IP: FREEPBX IP

Shoretel Configuration
================
created individual sip trunks
created sip trunk group
Setup Off System Ext. on SIP trunk group



SHORETEL Trunk Groups CONFIGS:
-------------
Name: Asterisk_SIP
Teleworkers: unchecked
Enable SIP Info for G.711 DTMF Signaling: Unchecked
Enable Digest Authentication
User ID: siptrunk
Password: password (not real password)

Shoretel Individual Trunk
---------------
Name: Asterisk_SIP
Trunk Group: Asterisk_SIP
Switch: Switch 1
SIP Trunk Type: Use IP Address ****SHORETEL VIRTUAL SWITCH IP***


Asterisk Configuration (sip.conf)
========================
register = siptrunk[email protected] switch ip/siptrunk

[siptrunk]
type=friend
secret=password (not real password)
context=shoretel_incoming
host=SHORETEL VIRTUAL IP SWITCH
disallow=all
allow=ulaw
dtmfmode=rfc2833
username=siptrunk


What's really funny is.... I can dial outbound all day long from Asterisk, but I cannot access Shoretel extensions....


ANy Suggestions?

[Jason Learmouth]

Make sure that you have Extension ticked on the ShoreTel SIP Trunk Inbound section. You may also need to do some work on the Asterisk to ensure that it is actually using the SIP trunk to dial the ShoreTel extensions. Perhaps a packet trace would help, it would show you if the call is being attempted and also show you why it's being rejected if it is.

Also it may just be a typo in you post, but I think the IP address entered for the ShoreTel individual trunks should be the FREEPBX IP.

[1stSentinel]

Typo!!!.... your are correct in the typo ST individual trunks is FreePBX IP.

hey
Thanks for all your posts on the Shoretel to Asterisk configs. They worked like a charm! Expanding a little, FreePBX using trunks on ST system for outbound dialing! (I enable tandum trunking)

The only caveat is I receive a 603 decline error (ST is denying the freePBX ext ->ST ext call) I get a busy signal.

Other than extension to extension calls from FreePBX to ST, everything works.....CallerID (CID) outbound calls, even ST ext to freePBX ext calls. But not FreePBX to ST.

I forgot to mention I do have ST SIP Profile option set for FakeDeclineAsRedirect=1

Suggestions?

[praetorian46]

This is a great Thread!
1stSentinel,
Did you ever figure out the Asterisk Ext to ShorTel Ext calling? I would really like to bridge to remote offices so we don't have to tie up and outside lines to make calls between them.
Thank you,
Cody

[bunbun]

Hi guys,

I have VOIP service through Vitelity. I'm trying to setup an Astricks box to just route the incoming calls directly to the ShoreTel switch. All outbound calls dial directly to the WAN IP of the VOIP provider Vitelity.

Since the SIP trunks point to an outside WAN IP, outbound calls work correctly from the ShoreTel switch.

What I'm trying to do is just have the Astricks server route calls to the ShoreTel switch, and let the ST switch handle the routing of the extensions.

This is what Vitelity has provided me with in regards to the sip config:

[vitel-inbound]
type=friend
dtmfmode=auto
host=inbound24.vitelity.net
context=inbound
allow=all
insecure=very

How would this apply to the current settings provided in the prior threads? Or am I doing this all wrong for what I'm trying to accomplish?

ps Just trying to test astricks to see if it strips the sip headers for ring back after the AA. Currently when someone calls our office it rings to the AA, but when they enter in an extension, no ringback is sent to the caller, and they just hear dead air. So I was thinking the Astricks server would take care of this..

Thanks everyone!

My first Post!