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SIP Trunking HELP!!! 4

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rhawkins74

Vendor
Nov 5, 2010
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Hey everyone...
Well here is my issue, I am in the process of setting up my first SIP trunks. This is a completely new area for me, and have to be honest I am lost. I definitely need some help on this, I don't even know where to start.
 
Ok we have a registration! Yay!

Let's look at your SIP URI's. I'll tell you what I always do, ofcourse I'm not entirely sure if that will work for Nextiva, but we'll give it a go.

First up you should know that with SIP you cannot modify your outgoing CLI the ISDN-way.

With SIP outgoing CLI can be set in three places:

1. at the provider. Basically your default number for the trunk.
2. in the SIP URI. You can just hardcode the number you wish to set.
3. in the user under the SIP tab. This requires on of your SIP URI's to be set to 'use internal data'.

Now, we'll start with 2 URI's. You can set up more as you go. I always split incoming and outgoing URI's. There's not really a limit on the amount of URI's you can make, so why not?

Incoming:

In the three drop downs, put a star (*). This enables you to use the incoming call routes as usual.
You don't have to tie it to a registration, so leave that at 'None'.
Put the incoming line id on 0.
Put the outgoing line id on sth you will not use, i.e. 15.

Outgoing (different URI!!):

In the three drop downs, select Use client authentication name. You're outgoing CLI will now be the CLI the provider set for you.
Tie it to the registration by selecting it.
Put the incoming line id on sth you will not use, i.e. 15.
Put the outgoing line id on sth else, i.e. 99

You can now make as many URI's as you like for outgoing calls, all with different CLI's. If you put it on Use internal data, it sends the number you specifiy in the user. I find this handy if I need to send personal DID's. Make sure you specify a different Outgoing Line ID.

You can also make a URI with a number hardcoded in those there drop downs. I.e. if a whole department needs to send that specific number. Or the main number. Also specify a different Outgoing Line ID again.

ARS: (I assume you use 50:Main)

Put the following in the ARS:

N;/N"@sbc.voipdnsservers.com"/99

give it a go!
 
Oh the forget my previous post for now (put file it under 'usefull' :p).

You need to figure out why the line doesn't register.

That trace you sent earlier only showed tx messages (outgoing), you should also see rx messages (incoming from provider).

Did you run STUN btw? (System, LAN1, Network Topology).
 
346895mS SIP Reg/Opt Tx: 17
OPTIONS sip:Unknown@sbc.voipdnsservers.com SIP/2.0
Via: SIP/2.0/UDP 10.0.1.190:5060;rport;branch=z9hG4bK6fc9c5e3db8aa06acd2844a6659a13b2
From: <sip:Unknown@sbc.voipdnsservers.com>;tag=9c6137c9f6083e2f
To: <sip:Unknown@sbc.voipdnsservers.com>
Call-ID: 774e0390e556e46406b150ae88f2cba0@10.0.1.190
CSeq: 1812900218 OPTIONS
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0

346966mS SIP Reg/Opt Rx: 17
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.1.190:5060;received=97.64.214.34;branch=z9hG4bK6fc9c5e3db8aa06acd2844a6659a13b2;rport=5060
From: <sip:Unknown@sbc.voipdnsservers.com>;tag=9c6137c9f6083e2f
To: <sip:Unknown@sbc.voipdnsservers.com>;tag=aprqngfrt-c1c567hde8al7
Call-ID: 774e0390e556e46406b150ae88f2cba0@10.0.1.190
CSeq: 1812900218 OPTIONS
 
This is not a REGISTER sequence but and OPTIONS sequence.

Can you find a REGISTER message (TX!), highlight the Call ID and hit F4 (it will filter all messages with that string).
 
Start with STUN but make sure all relevant ports are open on your firewall first:

5060-5090 UDP
3478 UDP
49152-53246 UDP
 
also, use the following IP as your STUN server:

132.177.123.13
 
1246895mS SIP Reg/Opt Tx: 17
OPTIONS sip:Unknown@sbc.voipdnsservers.com SIP/2.0
Via: SIP/2.0/UDP 10.0.1.190:5060;rport;branch=z9hG4bKbb5d9651b59e014b779f0b2eb5da6326
From: <sip:Unknown@sbc.voipdnsservers.com>;tag=91427dc0e5bf9002
To: <sip:Unknown@sbc.voipdnsservers.com>
Call-ID: f05e5d19d246c46e3e8e8900605f5bc3@10.0.1.190
CSeq: 1522102245 OPTIONS
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0

1246895mS SIP Tx: UDP 10.0.1.190:5060 -> 208.73.146.95:5060
OPTIONS sip:Unknown@sbc.voipdnsservers.com SIP/2.0
Via: SIP/2.0/UDP 10.0.1.190:5060;rport;branch=z9hG4bKbb5d9651b59e014b779f0b2eb5da6326
From: <sip:Unknown@sbc.voipdnsservers.com>;tag=91427dc0e5bf9002
To: <sip:Unknown@sbc.voipdnsservers.com>
Call-ID: f05e5d19d246c46e3e8e8900605f5bc3@10.0.1.190
CSeq: 1522102245 OPTIONS
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0

1246967mS SIP Rx: UDP 208.73.146.95:5060 -> 10.0.1.190:5060
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.1.190:5060;received=97.64.214.34;branch=z9hG4bKbb5d9651b59e014b779f0b2eb5da6326;rport=5060
From: <sip:Unknown@sbc.voipdnsservers.com>;tag=91427dc0e5bf9002
To: <sip:Unknown@sbc.voipdnsservers.com>;tag=aprqngfrt-e4js7f9bptrae
Call-ID: f05e5d19d246c46e3e8e8900605f5bc3@10.0.1.190
CSeq: 1522102245 OPTIONS

1246968mS SIP Reg/Opt Rx: 17
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.1.190:5060;received=97.64.214.34;branch=z9hG4bKbb5d9651b59e014b779f0b2eb5da6326;rport=5060
From: <sip:Unknown@sbc.voipdnsservers.com>;tag=91427dc0e5bf9002
To: <sip:Unknown@sbc.voipdnsservers.com>;tag=aprqngfrt-e4js7f9bptrae
Call-ID: f05e5d19d246c46e3e8e8900605f5bc3@10.0.1.190
CSeq: 1522102245 OPTIONS

 
Again, these are OPTIONS sequences, not REGISTER sequences!!
 
100385mS SIP Call Tx: 17
INVITE sip:N@sbc.voipdnsservers.com SIP/2.0
Via: SIP/2.0/UDP 97.64.214.34:5060;rport;branch=z9hG4bK237242247d51a5696978ed2356642473
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=c3a6790ded564e6d
To: <sip:N@sbc.voipdnsservers.com>
Call-ID: c61f8e3d1674d60fc1c78ff02698e12e@97.64.214.34
CSeq: 1735785606 INVITE
Contact: <sip:anonymous@97.64.214.34:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
Privacy: id
P-Asserted-Identity: "advtt78895789578" <sip:advtt78895789578@sbc.voipdnsservers.com:5060>
Content-Length: 203

v=0
o=UserA 423514166 1150706941 IN IP4 97.64.214.34
s=Session SDP
c=IN IP4 97.64.214.34
t=0 0
m=audio 49152 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
100385mS SIP Tx: UDP 10.0.1.190:5060 -> 208.73.146.95:5060
INVITE sip:N@sbc.voipdnsservers.com SIP/2.0
Via: SIP/2.0/UDP 97.64.214.34:5060;rport;branch=z9hG4bK237242247d51a5696978ed2356642473
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=c3a6790ded564e6d
To: <sip:N@sbc.voipdnsservers.com>
Call-ID: c61f8e3d1674d60fc1c78ff02698e12e@97.64.214.34
CSeq: 1735785606 INVITE
Contact: <sip:anonymous@97.64.214.34:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
Privacy: id
P-Asserted-Identity: "advtt78895789578" <sip:advtt78895789578@sbc.voipdnsservers.com:5060>
Content-Length: 203

v=0
o=UserA 423514166 1150706941 IN IP4 97.64.214.34
s=Session SDP
c=IN IP4 97.64.214.34
t=0 0
m=audio 49152 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
100453mS SIP Rx: UDP 208.73.146.95:5060 -> 10.0.1.190:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 97.64.214.34:5060;received=97.64.214.34;branch=z9hG4bK237242247d51a5696978ed2356642473;rport=5060
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=c3a6790ded564e6d
To: <sip:N@sbc.voipdnsservers.com>
Call-ID: c61f8e3d1674d60fc1c78ff02698e12e@97.64.214.34
CSeq: 1735785606 INVITE

100455mS SIP Call Rx: 17
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 97.64.214.34:5060;received=97.64.214.34;branch=z9hG4bK237242247d51a5696978ed2356642473;rport=5060
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=c3a6790ded564e6d
To: <sip:N@sbc.voipdnsservers.com>
Call-ID: c61f8e3d1674d60fc1c78ff02698e12e@97.64.214.34
CSeq: 1735785606 INVITE

100460mS SIP Rx: UDP 208.73.146.95:5060 -> 10.0.1.190:5060
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 97.64.214.34:5060;received=97.64.214.34;branch=z9hG4bK237242247d51a5696978ed2356642473;rport=5060
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=c3a6790ded564e6d
To: <sip:N@sbc.voipdnsservers.com>;tag=aprqngfrt-d8ns7ovcl3vd8
Call-ID: c61f8e3d1674d60fc1c78ff02698e12e@97.64.214.34
CSeq: 1735785606 INVITE

100462mS SIP Call Rx: 17
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 97.64.214.34:5060;received=97.64.214.34;branch=z9hG4bK237242247d51a5696978ed2356642473;rport=5060
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=c3a6790ded564e6d
To: <sip:N@sbc.voipdnsservers.com>;tag=aprqngfrt-d8ns7ovcl3vd8
Call-ID: c61f8e3d1674d60fc1c78ff02698e12e@97.64.214.34
CSeq: 1735785606 INVITE

100464mS SIP Call Tx: 17
ACK sip:N@sbc.voipdnsservers.com SIP/2.0
Via: SIP/2.0/UDP 97.64.214.34:5060;rport;branch=z9hG4bK237242247d51a5696978ed2356642473
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=c3a6790ded564e6d
To: <sip:N@sbc.voipdnsservers.com>;tag=aprqngfrt-d8ns7ovcl3vd8
Call-ID: c61f8e3d1674d60fc1c78ff02698e12e@97.64.214.34
CSeq: 1735785606 ACK
Privacy: id
P-Asserted-Identity: "advtt78895789578" <sip:advtt78895789578@sbc.voipdnsservers.com:5060>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Length: 0

100464mS SIP Tx: UDP 10.0.1.190:5060 -> 208.73.146.95:5060
ACK sip:N@sbc.voipdnsservers.com SIP/2.0
Via: SIP/2.0/UDP 97.64.214.34:5060;rport;branch=z9hG4bK237242247d51a5696978ed2356642473
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=c3a6790ded564e6d
To: <sip:N@sbc.voipdnsservers.com>;tag=aprqngfrt-d8ns7ovcl3vd8
Call-ID: c61f8e3d1674d60fc1c78ff02698e12e@97.64.214.34
CSeq: 1735785606 ACK
Privacy: id
P-Asserted-Identity: "advtt78895789578" <sip:advtt78895789578@sbc.voipdnsservers.com:5060>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Length: 0

 
What was the outcome from running STUN?

You made a typo in your ARS. What did you put in?

Should be sth like this:

N;/N"@sbc.voipdnsservers.com"/99

Don't forget the quotes!
 
Cool! Now grab my previous lengthy post and try to set that up! :)

You're getting there!
 
NLPeterK... Let me say thank you so far for your help on this.

As for setting up incoming and outgoing, we are wanting to keep it very simple. Like I said, this is one of our demo kits, so there is no one being assigned DID's or anything of that nature.

As of right now, we have it set up with outgoing id:0 are POTS lines, and a shortcode of 9 to grab these lines

We have outgoing id:1 are lines coming from cable company, and a shortcode of 7 to grab these lines

was going to set up outgoing id:2 for these SIP Trunks and a shortcode of 8 to grab them.
 
I was just giving you the full monty so you know where to go from here if you decide to set this up at a client.. They usually have some more 'elaborate' wishes towards CLI and that can really be a pain in the behind on SIP :)

Did you figure it out with the outgoing calls? Based on what you just said it should be:

in System Shortcodes:
8N/N/ARS-for-SIP

in ARS-for-SIP:
N;/N"@sbc.voipdnsservers.com"/2 Don't forget the ; and the " :)
 
Oh I understand fully about the full Monty, I figure once I get this up and running properly, then I can save the config and start playing around on it, and work on setting it up in a full environment.

I set up the ars and the shortcode and I get a call rejected when trying to use them. also, if I try and call the numbers I get just dead silence.
 
To simplify slightly, since R6 there is no need for the "@sipprovider.com" part anymore, just pointing N or whatever (like ISDN) at the SIP line group is enough as the system adds it for you :)

 
I am assuming the call rejection must have something to do with the SIP URI?
 
Post your config (edited if necessary), we can check and advise accordingly, it will be quicker than all this trial and error :)

 
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