MITEL SIP Trunking to SIP provider

[ferdiksa]

I have set up SIP trunking to a SIP service provider. The log files indicate that the connection was established. When I dial from an extension the message comes up saying Access Denied. I have checked the COR and it is not that. Any ideas anyone? How can I test if a SIP trunk works if I can not use direct trunk access?

Regards

Ferdi

 

[M1tel]

maintenance commands

SIP
Qualifiers:
BUSY PEER <PEER_NAME>

BUSY FORCE PEER <PEER_NAME>

RTS PEER <PEER_NAME>

LINK STATE PEER <PEER_NAME>

LINK STATE ALL

LINK STATUS

SHOW ACTIVE ALL

SHOW ACTIVE PEER <PEER_NAME>

SHOW ACTIVE PROFILE <PROFILE_NAME>

SNMP COMMON STATS SUMMARY

SNMP OTHER COMMON STATS

SNMP STATS METHOD <METHOD_NAME>

SNMP TIMERS CONFIGURATION

SNMP COMMON CONFIGURATION

SNMP PORT CONFIGURATION

SNMP SUPPORTED METHODS

STATS PROFILE <PROFILE_NAME>

STATS ALL

SIP REGISTRAR CONFIG

SIP REGISTRAR STATS

SIP REGISTRAR CONTACTS ALL

SIP REGISTRAR CONTACTS <USER_NAME>

SIP MWI STATS

SIP MWI STATS CLEAR

SIP MWI SUBSCRIBER INFO ALL

SIP MWI SUBSCRIBER INFO <CONTACT NAME>

SIP URI TRANS DEBUGON

SIP URI TRANS DEBUGOFF

SIP URI TRANS LIST

SIP URI TRANS SORTURI

SIP UIR TRANS SORTNUM

Purpose:
Provides maintenance commands for SIP trunks.

Details:
Command
Details

SIP BUSY PEER <PEER_NAME>
Busies the designated peer. This is a courtesy busy out. Existing calls are allowed to complete normally, but new calls are blocked. When a courtesy busy out is used, the number of active calls is displayed. When all calls are completed, the state of the link changes to MAN BUSY.

<PEER_NAME> The peer name is the "Name" from the Network Element form, and the name is case sensitive.

Example:

sip busy peer SoftSw2A

SIP BUSY ( COURTESY ) PEER SoftSw2A succeeded.

Link state changes from IDLE to MAN BUSY.

SIP BUSY FORCE PEER <PEER_NAME>
Forces busy on the designed peer name. The calls in progress are dropped.

SIP RTS PEER <PEER_NAME>
Returns the busied peer to service.

Example:

sip rts peer SoftSw2A

SIP RTS PEER SoftSw2A succeeded.

Link state changes from MAN BUSY to IDLE.

SIP LINK STATE PEER <PEER_NAME>
Displays the link state for the selected peer name.

The state can be one of:

"IDLE" - Initial state; no calls have been made on the link.

"IN SERVICE" - The link is in service.

"OUT OF SERVICE" - A temporary or permanent network outage/error was detected on the link.

"VALIDATING" - A new call is being attempted and is bringing the link from OUT OF SERVICE to IN SERVICE, if possible.

"MAN BUSY PENDING" - The link is being busied out, and no new calls can be made. One or more existing calls are still up.

"MAN BUSY" - The link is busied.

"UNKNOWN" - Unexpected error on the system.

SIP LINK STATE ALL
Displays the link state for all peers.

Example:

sip link state all

There are 2 SIP link(s) programmed.

Peer 41 state is IN SERVICE.

Peer SoftSw2A state is IDLE.

SIP LINK STATUS
Displays the number of established links and non-established links.

SIP SHOW ACTIVE ALL
Displays a summary of all the active calls.

SIP SHOW ACTIVE PEER <PEER_NAME>
Displays an active call for the selected peer name.

SIP SHOW ACTIVE PROFILE <PROFILE_NAME>
Displays active calls for a specific profile name.

SIP SNMP COMMON STATS SUMMARY
Displays a summary of common statistics.

SIP SNMP OTHER COMMON STATS
Displays other common statistics.

SIP SNMP STATS METHOD <METHOD_NAME>
Displays the total number of incoming and outgoing requests for a particular method.

Method names:

INVITE, ACK, CANCEL, BYE, INFO, UPDATE, PRACK, OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, REFER

SIP SNMP TIMERS CONFIGURATION
Displays the values for the configured timers.

SIP SNMP COMMON CONFIGURATION
Displays values for the common configuration.

SIP SNMP PORT CONFIGURATION
Displays the list of ports and transport types being used by the SIP module.

SIP SNMP SUPPORTED METHODS
Displays the list of all supported SIP methods.

SIP STATS PROFILE <PROFILE_NAME>
Displays statistics data for a specific profile name, including the number of active calls, rejected calls, incoming and outgoing non-authorized calls, incoming and outgoing authorized calls, incoming and outgoing authorized failures.

SIP STATS ALL
Displays statistics data for all SIP links in the system, including the number of active calls, rejected calls, incoming and outgoing non-authorized calls, incoming and outgoing authorized calls, incoming and outgoing authorized failures.

SIP REGISTRAR CONFIG
Displays configuration data for the Registrar:

Domain name- the user name and IP address.

Authentication - whether Authentication is enabled.

Third Party Registrations - whether third-party registrations are allowed.

Default Expiry - the default time interval for registrations to be active.

Minimum Configurable Expiry Duration - the minimum expiry duration that may be requested by a User Agent for a particular contact.

Maximum Configurable Expiry Duration - the maximum expiry duration.



Registrar Configuration

-----------------------

Domain: mycompany.com IP Address 10.37.105.20

Authentication Enabled: yes

Allow Third Party Registrations: yes

Default Expiry Duration: 3600 seconds

Minimum Configurable Expiry Duration: 300 seconds

Maximum Configurable Expiry Duration: 7200 seconds

SIP REGISTRAR STATS
Displays the following statistics data:


Current - the current number registered.

Accepted - the number of registrations that have been accepted.

Refreshed - the number of times registrations have been refreshed.

Total failures - the total number of registration failures.

Rejected registrations - number of rejected registrations caused by some system or messaging error, and not specifically by one of the other categories.

Failure responses - the failure response with the associated SIP failure message (e.g., 400 bad request is a SIP error message).


Registrar Statistics

--------------------

Current: 3

Accepted: 3

Refreshed: 2

Total Failed: 0

(Rejected 0)

(400 Bad Request 0)

(403 Forbidden 0)

(404 Not Found 0)

(423 Min Expires 0)

SIP REGISTRAR CONTACTS <USER_NAME>
Displays the same information as SIP REGISTRAR CONTACTS ALL, except displays only those entries that have the user_name in the Addr: For example, if 40500 is the contact name, the following is displayed:



Registrar Entry(s)

--------------------

3 (0xa3cda40) State:Registered Addr:sip:40500@10.37.105.20

Contact1: sip:40500@10.37.102.44

Time Left:201 Expires:400 Priority:0 Port:506 Scheme:1

SIP REGISTRAR CONTACTS ALL
Displays the following for all users:


First number - local identifier followed by an address in brackets.

State - either registered or pending.

Address - the address that was registered and each contact registered is printed.

(All these examples only list a single contact.)

Time left - the time before expiry.

Expires - the expiry time is the total time negotiated. Priority - the contact's priority.

Port - the registered port.



Registrar Entry(s)

--------------------

3 (0xa3cda40) State:Registered Addr:sip:40500@10.37.105.20

Contact1: sip:40500@10.37.102.44

Time Left:287 Expires:400 Priority:0 Port:5060

2 (0xa3be408) State:Registered Addr:sip:40501@10.37.105.20

Contact1: sip:40501@10.37.102.43

Time Left:280 Expires:400 Priority:0 Port:5060

1 (0xa3be3e0) State:Registered Addr:sip:40503@10.37.105.20

Contact1: sip:40503@10.37.102.42:5060

Time Left:3208 Expires:3600 Priority:0 Port:5060

SIP MWI STATS
Displays the following information:


Current - the number of current MWI subscriptions. In the following example, three phones are registered, and two are also subscribed for MWI.

Successful - the number of successful subscriptions.

Failed - the number of failures.


MWI Subscription Statistics

--------------

Current: 2 (out of 3 Registrations)

Successful: 6

Failed: 2

SIP MWI STATS CLEAR
Clears the number of successful and failed registrations. The following message is displayed:


"MWI Subscription Statistics Cleared"

SIP MWI SUBSCRIBER INFO ALL
Displays the following information about the current state of MWI for all endpoints:



State - MWI state of the subscribers.

SIP URIs - SIP URIs for the endpoints.

Time left - amount of time (in seconds) left in the subscriptions.

Expires - the currently negotiated expiry time for this phone.



In the example below, the first entry shows the MWI lamp is OFF, and in the second entry, it's ON. The phone's address is printed along with the time left before the subscription expires.


MWI Entry(s)

------------

OFF sip:40500@10.37.105.20

Time Left:96 Expires:400

ON sip:40501@10.37.105.20

Time Left:288 Expires:400

SIP MWI SUBSCRIBER INFO <USER_NAME>
Displays the same information as SIP MWI SUBSCRIBER INFO ALL for a particular user, except the command uses a string to search for entries that match. For example, using 40500 as the contact name, it displays only the first entry. Using 405 displays both entries below.


MWI Entry(s)

------------

OFF sip:40500@10.37.105.20

Time Left:96 Expires:400

ON sip:40501@10.37.105.20

Time Left:288 Expires:400

SIP URI TRANS DEBUGON
Turns on debug tracing to display URI/Number translations on a per call basis.

SIP URI TRANS DEBUGOFF
Turns off debug.

SIP URI TRANS LIST
Displays an unsorted list of all URI/Number entries in the translation table.

SIP URI TRANS SORTURI
Displays a sorted list (by URI) of all URI/Number entries in the translation table.

SIP URI TRANS SORTNUM
Displays a sorted list (by number) of all URI/Number entries in the translation table.

 

[isrm]

Check public network access via DPNSS is enabled in the COS of the phone making the call

 

[ferdiksa]

Wow, Thanx guys. I have learned something with your assistance. Following your instructions I could determine that there must be something wrong on my setup. It looks like I have missed something...The SIP REGISTRAR CONTACTS ALL produces a blank entry.

Registrar Configuration
-----------------------
Domain: IP Address 192.168.1.2
Allow Third Party Registrations: no
Default Expiry Duration: 3600 seconds
Minimum Configurable Expiry Duration: 300 seconds
Maximum Configurable Expiry Duration: 7200 seconds


Registrar Statistics
--------------------
Current: 0
Accepted: 0
Refreshed: 0
Total Failed: 0
(Rejected 0)
(400 Bad Request 0)
(403 Forbidden 0)
(404 Not Found 0)
(423 Min Expires 0)

Registrar Entry(s)
--------------------

 

[veenstrahiem]

I had a problem with building a sip-trunk between 2 mitel controllers, but got it working now, I can dial between the controllers.
What you say about SIP REGISTRAR CONTACTS ALL producing an empty reply, that gives an empty reply with our systems as well, but once again we can dail extensions across.

What you could try for a sip-trunk building manual is use the manual for Quick Conference, it has detailed instructions in it.

 

[ferdiksa]

Thanx sankop, I am going to try that now.

 

[M1tel]

That command will produce nothing unless you are using sip registrar

 

[ferdiksa]

Hi Guys

Thanx for this information. I now have the incoming side working...i.e. If I dial the 087...number it rings at reception. However if I try to dial out from an extension I get an Access Denied. I have checked the Class of Restriction and moved the extension to a COR that is unrestricted.

 

[M1tel]

public access via dpnss in cos of sets

 

[mitelritz]

ive had that error before and it was a setup message of the trunks (trying running a wireshark and see what code you get when making the call.

and ask what SBC your Provider has.

i tend to see mitel and the Nextone not to be friendly.


Thanks

 

[mitelritz]

sory and if your not registering with a provider (direct ip authentication) in your network assignment form.. take out the ip addess in SIP registrar FQDN or ip address entry

if seen that issue on my end come up.

(IF YOUR Doing direct ip Authenication)
(and not registering with a user/pass)


what i normally do also is *9 for testing sip calls as you know you dial the full 10 digits (in US that is)


whats your ICP version also.

welcome to the world of sip haha.

 

[ferdiksa]

Thanx mitelritz

I will try that today. I also made a stupid mistake in that I never checked with the provider if they allow calls to 087... numbers for testing. I.e. they have given me an 087 number but it might not be activated until I pay the subscription. I hope this is the problem as I have now exhausted most options.

Regards

Ferdi

 

[vlevalois]

I've got a similar problem whereas I get a successful registration yet OUT OF SERVICE on outbound and disconnect on inbound. The trunk is registered with an outside service and they can see the attempts at the calls, yet get a 503 Service Unavailable.

We're running MCD 4.0. 192.168.253.10 is our 3300, which lives on its own VLAN along with all the IP phones which are getting DHCP from the 3300 after rejection of our regular DHCP server on VLAN 1 (192.168.254.x).

The SIP gateway is on router 192.168.254.4, which translates all SIP requests to a public IP. There's a static route enabled for packets from 192.168.254.4 to find 192.168.253.10. Again, registration is successful.

Here's some stats/info:


Registrar Entry(s)
--------------------

-------- SipCommmonConfiguration ------------

SipProtocolVersion: SIP/2.0
SipServiceOperStatus: 2
SipServiceStartTime: 1256692556
SipServiceLastChance: 0
SipMaxTransaction: 0
SipServiceNotificationEnable: 0
SipEntityType: 2

-------- SipCommonStatsSummary ------------

TotalNumOfIncomingRequests: 42
TotalNumOfOutgoingRequests: 3229
TotalNumOfIncomingResponses: 2553
TotalNumOfOutgoingResponses: 63


-------- SipOtherCommonStats ------------

NumUnsupportedMethods: 0
OtherwiseDiscardedMsgs: 24

------------ SipPorts -------------------

SipPort: 5060
SipTransportRcvType: 1

SipPort: 5060
SipTransportRcvType: 2

SipPort: 5061
SipTransportRcvType: 3

SipPort: 5060
SipTransportRcvType: 1

SipPort: 5060
SipTransportRcvType: 2

SipPort: 5061
SipTransportRcvType: 3

SipPort: 5060
SipTransportRcvType: 1

SipPort: 5060
SipTransportRcvType: 2

SipPort: 5061
SipTransportRcvType: 3

SipPort: 5060
SipTransportRcvType: 1

SipPort: 5060
SipTransportRcvType: 2

SipPort: 5061
SipTransportRcvType: 3

SipPort: 5060
SipTransportRcvType: 1

SipPort: 5060
SipTransportRcvType: 2

SipPort: 5061
SipTransportRcvType: 3

--------- SipSupportedMethods ----------------

SipSupportedMethodIndex: 1
SipMethodName: INVITE

SipSupportedMethodIndex: 2
SipMethodName: ACK

SipSupportedMethodIndex: 3
SipMethodName: UPDATE

SipSupportedMethodIndex: 4
SipMethodName: INFO

SipSupportedMethodIndex: 5
SipMethodName: CANCEL

SipSupportedMethodIndex: 6
SipMethodName: BYE

SipSupportedMethodIndex: 7
SipMethodName: PRACK

SipSupportedMethodIndex: 8
SipMethodName: REGISTER

SipSupportedMethodIndex: 9
SipMethodName: OPTIONS

SipSupportedMethodIndex: 10
SipMethodName: SUBSCRIBE

SipSupportedMethodIndex: 11
SipMethodName: NOTIFY

SipSupportedMethodIndex: 12
SipMethodName: REFER

There are 1 SIP link(s) programmed.
Link Telcentri state is OUT OF SERVICE (link down count 1)

Number of established links: 0
Number of non-established links: 1

Registrar Configuration
-----------------------
Domain: 192.168.253.10 IP Address 192.168.253.10
Allow Third Party Registrations: no
Default Expiry Duration: 3600 seconds
Maximum Configurable Expiry Duration: 7200 seconds

Registrar Entry(s)
--------------------

Registrar Entry(s)
--------------------

SIP RTS PEER Telcentri succeeded.
Link state changes from OUT OF SERVICE to OUT OF SERVICE.

LDS Interface Registered components:
------------------------------------------------------
Component Id : Name:
# 1: SSP
# 2: SSM
# 3: SIPMH
# 4: DM
# 5: NTM
------------------------------------------------------
End of list.

See attached Network Element and Sip Peer programming.

Something tells me it's a simple setting issue.

Any help would be appreciated!

Vince

 

[vlevalois]

Actually got this working now inbound/outbound but getting bidirectional loss of voice after 2-3 minutes... Workin on it!

 

[mitelritz]

Hey there just a thought

In the sip world theres alot of different things

Try and lower your session time (seems it might be at 300) and see if you lower it it could lower ur disconnect time..

one of my issues were a cust session timer was at 600 and the call would always disconnect at 5 min i guess it would signal it at 300 sec hence 5 min. And it was due to the ports were changing right at 5 min from the mitel to the sip provider. And i lost audio

Hope that helps

 

[vlevalois]

Thanks for the tip, I actually ended up removing the Session Timer (making it 0) and it's all good. I now have inbound and outbound working fine.

One of the keys is that the SIP gateway has to see the call coming from the same number as the in the registration. So CPN range has to match that.

These are the SIP Peer Profile settings that ended up working:

SIP Peer Profile Label: SIP
Network Element: Telcentri

Local Account Information
Registration User Name: xxxxxxxxxxx
Address Type: IP Address: 192.168.253.10


Call Routing and Administration Options
Interconnect Restriction: 1
Maximum Simultaneous Calls: 1
Outbound Proxy Server:
SMDR Tag: 0
Trunk Service: 10
Zone: 1
Alternate Destination Domain Enabled: No
Alternate Destination Domain FQDN or IP Address:
Enable Special Re-invite Collision Handling: No
Private SIP Trunk: No
Route Call Using To Header: No


Calling Line ID Options
Default CPN: xxxxxxxxxxx
CPN Restriction: No
Public Calling Party Number Passthrough: No
Use Diverting Party Number as Calling Party Number: No


Authentication Options
User Name: xxxxxxxxxxx
Password: *******
Confirm Password: *******
Authentication Option for Incoming Calls: No Authentication


SDP Options
Allow Peer To Use Multiple Active M-Lines: No
Enable Mitel Proprietary SDP: No
Force sending SDP in initial Invite message: No
Force sending SDP in initial Invite - Early Answer: No
NAT Keepalive: No
Prevent the Use of IP Address 0.0.0.0 in SDP Messages: No
Renegotiate SDP To Enforce Symmetric Codec: No
Repeat SDP Answer If Duplicate Offer Is Received: No
RTP Packetization Rate Override: No
RTP Packetization Rate: 20ms
Special handling of Offers in 2XX responses (INVITE): No
Suppress Use of SDP Inactive Media Streams: No


Signaling and Header Manipulation Options
Session Timer: 0
Build Contact Using Request URI Address: No
Disable Reliable Provisional Responses: Yes
Enable sending '+' for E.164 numbers: No
Ignore Incoming Loose Routing Indication: No
Use P-Asserted Identity Header: No
Use P-Preferred Identity Header: No
Use Restricted Character Set For Authentication: No
Use To Address in From Header on Outgoing Calls: No

In Network Element all I've got is one entry (No Proxy) with the local IP of the SIP gateway specifically UDP 5060 on all entries.

Make sure your ARS is programmed correctly and you're good to go!

 

[mitelritz]

yea that was my final outcome on what i did i put it to 0 i was gonna say if that was it 0 is your answer cool ....

out of curiousity who is your provider (is it telecentri) you are dealing with

(we dealt with alot of providers trying to pick best of the best)

thing i learned mitel and the nextone SBCs dont like eachother..

 

[vlevalois]

It's Telcentris. They support SIP Trunks over DSL/Cable while others like Broadband, Paetec, etc. will only support over T1. They provide an Edgemark router that becomes the local gateway replacing your DSL modem for example or you can build a Proxy with a Cable modem or T1. It's essentially the bandwidth and QoS police a the gate and does its best at that stage. Beyond that to the ether it's anyone's game and you need to advise customers (we're resellers) that it's best effort. Then again, SIP trunking over DSL/Cable is cool for smaller businesses

 

[MitelJam]

Ferdiksa,

Did you sort out that access denied problem with your SIP trunk? I am having the same problem.

Regards

 

[vlevalois]

MitelJam, yes, make sure that your SIP Peer Profile by Incoming DID settings are correct. In this case, the DID must be all 11 digits. If it's 1 off it will give you Access Denied. Consult with your provider as to what they need to see on outbound to authorize the call to be made.

Also, the Default CPN in Sip Peer Profile Form must also conform to their standards, which in my case is all 11 digits of the SIP trunk #.

Now if I could only get HOLD, transfers and Personal Ring Groups to work I'd be happy, but that's easier said than done.

Vince