Sip Trunks

[camaro67]

Hello all I have been asked to research our cs1000rls.5.0 for the possibilty of connecting it to asterisk we currently have sip trunks to 3 other sites via a stand alone NRS i have the route,cdp entery and the dn I would like to use for the asterisk box. i enetred it as a static endpoint in the nrs but i get nothing when trying to dial the asterisk. the forums I've read all seem to be slightly different on where to register the asterisk.I'm just looking for some guidelines and what i may be missing.thank you!

 

[RyanEOD]

Camaro,

Funny you ask, I just got done doing this here. I am on CS1000E 5.0 to Asterisk and it is working like a champ! So, can you print out what all programming you have right now? Also if you think your truck is good you can do a SIP trace to see if it is going out to the right IP address:


oam> SIPOutput 1
oam> SIPTraceLevel 1
oam> SIPCallTrace on

 

[camaro67]

cant seem to get my trunk to regisiter



[Oct 29 08:20:13] NOTICE[2486]: chan_sip.c:11539 sip_reregister: -- Re-registration for asterisk@nortel
> doing dnsmgr_lookup for 'nortel'
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 10.208.72.5:5060:
REGISTER sip:nortel SIP/2.0
Via: SIP/2.0/UDP 10.208.72.50:5060;branch=z9hG4bK5f92aa33;rport
Max-Forwards: 70
From: <sip:asterisk@nortel>;tag=as0fc4d9c6
To: <sip:asterisk@nortel>
Call-ID: 466291c0303f3a185c89e55204b85220@10.208.72.50
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:s@10.208.72.50>
Content-Length: 0


---

<--- SIP read from UDP:10.208.72.5:5060 --->
SIP/2.0 501 Not Implemented
From: <sip:asterisk@nortel>;tag=as0fc4d9c6
To: <sip:asterisk@nortel>;tag=1f10fd08-548d00a-13c4-40030-c0af1-5c0b662c-c0af1
Call-ID: 466291c0303f3a185c89e55204b85220@10.208.72.50
CSeq: 102 REGISTER
Via: SIP/2.0/UDP 10.208.72.50:5060;rport=5060;branch=z9hG4bK5f92aa33
Supported: 100rel,x-nortel-sipvc,replaces,timer
User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.00.31
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
[Oct 29 08:20:13] WARNING[2486]: chan_sip.c:18591 handle_response: Host '10.208.72.5' does not implement 'REGISTER'
[Oct 29 08:20:13] NOTICE[2486]: chan_sip.c:11539 sip_reregister: -- Re-registration for asterisk@nortel
> doing dnsmgr_lookup for 'nortel'
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 10.208.72.6:5060:
REGISTER sip:High SIP/2.0
Via: SIP/2.0/UDP 10.208.72.50:5060;branch=z9hG4bK57daf344;rport
Max-Forwards: 70
From: <sip:asterisk@nortel>;tag=as28dbc20e
To: <sip:asterisk@nortel>
Call-ID: 4c1e1a057e7a0db42e9933a044d17fc9@High
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:s@10.208.72.50>
Content-Length: 0


---

<--- SIP read from UDP:10.208.72.6:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.208.72.50:5060;branch=z9hG4bK57daf344;rport;received=10.208.72.50
From: <sip:asterisk@nortel>;tag=as28dbc20e
To: <sip:asterisk@nortel>;tag=62192
Call-ID: 4c1e1a057e7a0db42e9933a044d17fc9@High
CSeq: 102 REGISTER
Contact: <sip:s@10.208.72.50>
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
[Oct 29 08:20:13] WARNING[2486]: chan_sip.c:18165 handle_response_register: Forbidden - wrong password on authentication for REGISTER for 'asterisk' to 'nortel'
Really destroying SIP dialog '4c1e1a057e7a0db42e9933a044d17fc9@High' Method: REGISTER

 

[RyanEOD]

What is 10.208.72.5? Do you have your asterisk set up as a Gateway Endpoint in your NRS?

 

[camaro67]

yes i have asterisk configured in the NRS....72.5 is the node ip.

 

[RyanEOD]

Can you post your NRS Gateway Endpoint config for the Asterisk?

 

[camaro67]

ok i have a routing entry in of 3 and a cost of 1



View Gateway Endpoint Property (High / highudp / highcdp)

Gateway endpoint ID

Endpoint name
asterisk
Endpoint description

Tandem gateway endpoint name

Endpoint authentication enabled
Not configured
Authentication password

E.164 country code

E.164 area code

E.164 international dialing access code

E.164 national dialing access code

E.164 local (subscriber) dialing access code

Private L1 domain (UDP location) dialing access code

Private special number 1

Private special number 2

Static endpoint address type
IP version 4
Static endpoint address
ip of the asterisk
H.323 Support
H.323 not supported
323 transport

H.323 port

SIP support
Static SIP endpoint
SIP transport
TCP UDP TLS
SIP port

Network Connection Server enabled

 

[RyanEOD]

Only differences I see is I am doing UDP 5060 and don't have Network Connection Server enabled checked.

After that, how do you have it configured in your Asterisk? Using FreePBX?

 

[camaro67]

no I'm using aterisk with the GUI ...i matched your settings still no luck.

 

[RyanEOD]

Looks like your Asterisk side is trying to register it, in your PEER side have:


Outgoing Settings; Trunk name: CS1000

host=CS1000IPADDY
type=peer


In your USER side have:

USER Context: from-CS1000


host=CS1000IPADDY
useregphone=no
user=phone
type=peer

Register String BLANK

Then do another SIP trace