SIP on BCM50 R3

[BackofficeEurope]

Has anyone successfully configured SIP on a BCM 50R3 ? We have been trying to get it to work but with no success so far.

Thanks
Mike

 

[KjellH]

It all depends on what your SIP provider does.

If your looking at SIP end user accounts, where each account would register to a Providers SIP server - then the answer would be, the BCM50 can't do it.
The BCM50 cant do REGISTER to a SIP provider.

If you are looking at SIP trunks - then the answer might be yes, it all depends on the Provider and how flexible they may be.

If you use NAT, you could run into problems, it all depends on your Firewall and how well it supports SIP and the ability to open up all traffic on port 5060.

Basic set up.

Global settings
You must type in a - Business Name else all outgoing calls will be "Anonymous"

Public network dialing plan - Public Unknown

Dialing Plan , Routing , set up route to the VOIP trunks,
set up destination digit to Route

Active sets
OLI and Public # recieve, its very important to have them correct

Resources, IP trunks
If you use Route all calls using SIP Proxy you don't need to use the Routing table.

Tab:Sip Setting : local Domain - this could be you BCM50, the outside on the firewall, basicly it is the adress you tell the provider to answer back to.( use IP-adress )

Tab: SIP Proxy :
Domain = SIP providers domain adress
tick Route all calls using proxy,
set up an outbound proxy tabell, could be more than one line depends on the info you get from Provider.

Tab: SIP URI map
Unknown/Unknown - delete content , should be blank

Tab: SIP Authentication
Remote Accounts : if you get an account from your prider, add it here.

Thats it, hope this will help.

 

[BackofficeEurope]

I have been trying and trying and trying ... without any success (obviously). I am going crazy over this one :-(

My settings (now) are hereafter. Can someone spot the problem. UIP log excerpt is at the end.
When I dial a number I get No answer (I used to get Network not available but now it says 'no answer 18'. Why the 18 stands, I have no idea.

Telephony Resources - IP-Trunks - Routing Table
Description: Wizz
Destination Digits: 0
Domain: wizz-telecom.com
IP Address: 81.88.110.36
Port: 5060
GW Type: Other
MCDN Protocol: None
VoIP Protocol: SIP
QOS Monitor: unchecked
Tx Threshold: 0.0
Telephony Resources - IP-Trunks - SIP Settings
Telephony Settings - Fallback to circuit switched: Disabled
SIP Settings: Local Domain: Backoffice.be
Call Signalling Port: 5060
RFC2833: Dynamic Payload: 120
Status: Gateway is running
Telephony Resources - IP-Trunks - SIP Proxy
SIP Proxy - Domain: wizz-telecom.com
SIP Proxy - Route all calls using proxy: unchecked
MCDN Protocol: None
Outbound Proxy Table: empty
Optional IP address: 81.88.110.36
Port: 5060
Telephony Resources - IP-Trunks - SIP media Parameters
Selected list: G.729, G.711-aLaw, G.711-uLaw and G.723
Settings: Enabel Voice Activity Detection: Checked
Jitter Buffer: Auto
G.729 payload size (ms): 20
G.723 payload size (ms): 30 (greyed)
G.711 payload size (ms): 20
Fax Transport: T.38
Provide in-band ringback: Unchecked
Telephony Resources - IP-Trunks - SIP URI Map
e.164/National: national.e164
e.164/Subscriber: subscriber.e164
e.164/Unknown: unknown.e164
e.164/special.e164
Private/UDP: udp
Private/CDP: cdp
Private/Special: special.private
Private/Unknown: unknown.private
Private/Subscriber: suscriber.private
Unknown/Unknown: <empty>
Telephony Resources - IP-Trunks - SIP Authentication
Local Sip Authentication - Local Authentication: unchecked
Local Sip Authentication - Quality of Protection: authentication only
Local Sip Authentication - 401 reason: Unauthorized
Local Accounts: None
Remote Accounts:
realm: wizz-telecom.com
User ID: 9980400023301
Pwd: *****
Description: WIZZ TCOM


Telephony - Dialing Plan - Routing - Routes
Route 000
External number: <empty>
Use Pool: A
DN Type: N/A
Service Type: N/A
Service ID: N/A

Route 001
External number: <empty>
Use Pool: BlocA
DN Type: Public (Unknown)
Service Type: N/A
Service ID: N/A

When monitoring a call with the BCM UIP Monitor I get this information:
05/28/08
Trunk 3 15:15:06 < CC < CRef Origin: CRef 4A4 SETUP
Bearer capability - Speech - G.711 A-law
Information Transfer Capability: speech
Coding Standard: CCITT standardized coding
Information Transfer Rate: 64 Kbits/s
Transfer Mode: circuit mode
User Information L1 Protocol ID: Recommendation G.711 A-law
Layer and Protocol Identification: User Information L1 Protocol
13 00 81 90 00 00 A3 00 00 00 00 00 00 00
Channel identification - B-
Information Channel Selection: B1 channel
D-Channel Indicator: channel identified is not the D-channel
Preferred/Exclusive: exclusive - only the indicated channel is acceptable
Interface Type: primary rate interface
Interface Identifier Present: interface explicily identified
Interface Identifier: 0
Channel Type/Map Element Type: B-channel units
Number/Map: channel indicated by number
Coding Standard: CCITT standard as in Recommendation Q.931
Channel Number: :
0F 00 E9 00 83 00 01 00 81 00
Calling party number - Unknown/Unknown - 32800031
Numbering Plan Identification: Unknown
Type of Number: Unknown/Reserved Value
Screening indicator: User provided, not screened
Presentation indicator: Presentation allowed
Number Digits: - 32800031
07 00 00 80 08 00 03 02 08 0A 0A 0A 03 01
Called party number - Unknown/Unknown - 003232800030
Numbering Plan Identification: Unknown
Type of Number: Unknown/Reserved Value
Number Digits: - 003232800030
03 00 80 00 0C 00 0A 0A 03 02 03 02 08 0A 0A 0A 03 0A
Trunk 3 15:15:16 < CC < CRef Origin: CRef 4A4 DISCONNECT
Cause - Normal
Location: User
Coding Standard: CCITT standardized coding as described below
Cause Class: Normal Event
Cause: Normal clearing
05 00 80 00 10 00

 

[KjellH]

Using the BCM UIP Monitor will not help you.
What you need is a Wireshark trace, with a filter on port 5060, you would also need a hub else you will se nothing relevant in the trace.

You could at the same time use BCM Line Monitor to check if the BCM is using the VoIP line when dialing out on the VoiP lines.

Since you are using Remote account the wireshark trace would start something like this for an outgoing call:
1.
INVITE "number"@81.88.110.36

2.
The provider would answer back 401 or 407

3.
INVITE "number"@81.88.110.36 , but this time the INVITE would contain Authentication detalis.


SIP Settings: Local Domain: Backoffice.be
Use the IP adress of BCM50

realm: wizz-telecom.com
if you get a 401 answer I would use IP-adress
In wirechark check second INVITE, check Authentication details, check realm, if it states "anonymous" then use IP adress.