More flexibility with Rel 10
SIP provider and trunk
parameter
SIP provider today and in the past
Provider settings
Name: Provider name
Proxy Server/Port
Registrar/Port
• If no entry made, PBX will not send any register messages
Outbound Server/Port
• if needed
Domain used for registration and outgoing calls
• If no entry is made, the entry from field “proxy server” is used
Provider settings
Additional Domain only necessary with exclusive providers.
• On outgoing calls the domain will be replaced by value of the additional parameter field
Standard (MSN)/DDI
• Call distribution for this trunk made via
> Call distribution Incoming
> Call distribution Incoming DDI
Provider settings
Own IP address
• Individual /System = Local IP of PBX
> Public IP has to be resolved by Provider
> PBX always acts as RTP proxy.
• Via STUN: (STUN Server + Port)
> PBX determine public IP via STUN Server (not necessary in 99%)
> Current SIP Provider uses ALG
• Input control: Set fix value for public IP
> used as originator during registration and in the SDP for RTP (Port forwarding or ALG necessary!)
Own Port
• Local SIP port: Range starts with 10670 + 1 for each new provider
• Each provider used own Port (easy trunk Identification / easier to trace)
• also used to identify Prefix for phone no. (incoming)
Provider settings
Registration Refresh: Determines the SIP re-registration interval
• 0 =prevent re-registration Force new registration
if line goes down NAT keep alive time
• send empty 4 byte packet every x seconds to keep NAT session in Router open
• Value 0 = off
• Some SIP provider also send packets to keep session open
• Alternative Port forwarding
Message response time
• PBX waits x sec for message response (busy WAN/LAN could lead to
SIP trunk breakdown caused by delayed response messages)
Provider settings
Send useragent ID
send User-Agent during registration/invite (PBX type + SW version) Example: Aastra 800 (R 1.461.1.2 aastra-softpbxwin)
Should be activated some SIP provider will reject registration without user agent
support call deflection
• 302 moved temporary (SIP partial rerouting)
• Call deflection is made at SIP provider - No channels used in PBX
• Call forward external has to be set in each SIP trunk too
Provider settings
Late RTP:
Especially for Belgacom, RTP (dial tone) will be send delayed after 200 OK to avoid ALG problems
Supervise trunk
• Useful for SIP Provider without registration
• Send SIP options to verify provider is still available
• Interval = timer used from the assigned VoIP-profile
• If provider doesn’t respond with 200OK, line goes down
Parameter
• Possible for future implementations
Provider settings: outgoing
Settings will be used to create outgoing invite messages in the
provider required format
Provider settings: outgoing
Time to ready dial out
• Wait x sec, after trunk is occupied + last digit was entered, to dial out
CLIR
• off : Provider doesn’t support CLIR
• RFC3325: Privacy header ID is used: “Privacy: header;id”
• anonymous: “from” and “Display info” are set to "anonymous“
• Sipgate: “Display info”: is set to "anonymous"
• Belgacom:
> Especially for Belgacom
> To: will look like *31*destination-number#
Provider settings: outgoing
CLIP no screening
• Send CLIP which doesn‘t belong to the trunk
• not supported by all provider
• Sipgate: send not screened Number in the Display info
• QSC:
• Manipulation of “from” also possible
•
special entry in call distribution outgoing necessary
FROM: display name
• Set fix value for the “display name”
• Every outgoing call will show the same number
Provider settings: outgoing
FROM: username create from
• DID : Direct Inward Dialing
• Value from the call distribution + internal number is used to create the “from” username (DDI trunks)
• SipId: SIP ID is used as value for the “from” username
• standard SIP trunk/MSN
• Sipid + DID: SIP ID + the DID value from the call distribution is used to create the “from” username (DDI Trunks/ QSC)
• Input control: set fix value for “from” username
Provider settings: outgoing
P-Preferred-Id: username create from
• Value which is used to create the P-Preferred-Id
> DID
> SIP ID
> Ignore
INVITE: Replace 00 by + (TO:username)
• Dialed number 00493061044515
• Will be send as To: <sip:[email protected]>
INVITE: Replace 00 by + (FROM:username)
• Original CLIP 00493061044515
• Will be send as From: <sip:[email protected]>
Provider settings: incoming
Defines which header information are used
• To realize call distribution
• For display CLIP information
• DID = Direct in dial number
> Chosen value is relayed to the Call manger (CI) and used to realize call distribution
> Wildcards in call distribution possible with use of “*”
Provider settings: incoming
DID: username take from
• Defines which value of the invite is used as DID for the call distribution
• TO(all)
>
Use for standard SIP accounts
>
Complete value in “to:” will be used for call distribution
• TO(SipID)
> If “to:” contains SIP ID + DDI (prefix binding QSC)
> SIP ID will be cut off and only the DDI is used for call distribution
• Example To: 00493061044515 = SIP ID 0049306104 + DDI 4515
• DID = 4515
• Requested Uri
> Value is transmitted to call distribution transparent
> Used for DDI Accounts
> Or “MSN SIP Accounts” example SKYPE
• Example 10 different number without no connectivity belong to one SIP account
• Entry in the call distribution has to be made for each number
Provider settings: incoming
CLIP: username take from
• Defines which value of the invite is used as CLIP information
> P-Asserted-Id
> FROM
> FROM: display name
> Ignore (show no CLIP to user)
QSC-Redirect-Header
• If a call is forwarded to a QSC SIP trunk the To: header contains number of the diverter
• Value can’t be used for call distribution because number doesn’t belong to the trunk
• Requested URI has to be taken from the X-ORGINAL-DDI-URI:
Provider settings: incoming
Do not repeat 180 Ringing
• PBX will send 180 Ringing once for an incoming invite
• Could solve problems with chopped ringtone for external calling parties
• QSC for example is creating a new RTP session for every
180 Ringing.
SIP trunk today and in the past
SIP trunks
SIP trunks
Name
SIP provider
• Used Provider Configuration for this trunk
Phone No.
• No function (different in the past Rel. 9)
• only for better overview in the call distribution
SIP ID
User Name + Password + Validation
• If no entry made, the PBX will send no register
• Sometimes used if ISP = SIP Provider and PBX has fix public IP address
SIP trunks
Reference trunk
• Constriction in the past: routes can contain up to 3 bundle only
> SIP trunk = bundle
> So each SIP trunk needed a own route if you wanted to use a specific one
• Allows use of more than 3 SIP trunks in the same route
• Normally used when several standard trunks with own registration belong to one Provider
• One trunk is assigned to a route and all other trunks use this SIP trunk as reference to use the same route for outgoing calls
• Important: Call distribution outgoing has to be defined.
Otherwise the SIP ID/CLIP of the reference trunk is used
in the outgoing invite
SIP trunks
Voip Profile
• Use predefined Codec’s
Company
• Trunk is assigned to Company (access rights; Operator; SMDR etc)
Request Uri: username take from
• Value from the “request Uri” is used to identify the correct trunk + prefix during an incoming call.
• If each Provider uses own local SIP port value can be set to ignore
(see “local Port” in the provider settings)
Request Uri: username take from
• SipId
> Used if value in request uri = SIP ID
•
mostly used for SIP standard trunk
•
or Provider which always use the SIP ID in Invite and To contains the DID
•
prefix binding (QSC/MKS Netzdienste)
> Value is taken from the SIP ID field in trunk configuration below
• DID
> Used for DDI SIP trunks
> Number format has to be equal to chosen value form “DID: username take from” in the Provider settings (without DDI)
> Value checked left-aligned
> example if Request-URI: sip:[email protected] and DDI range is 0-9999
> Invite req Uri = DID 00493061044515 without DDI = 0049306104
Request Uri: username take from
• Input control
> Value checked left-aligned
> Has to fit 100%
• Ignore
> Values from the Request-URI will be ignored
> Identification is done via used local SIP port
> Can’t be use if several trunks working on the same local SIP port
> Otherwise prefix for incoming calls will be wrong
> Used for SKYPE
•
One registration
•
But you have several different numbers in the invite, without connectivity
•
Similar to PtmP
SIP trunks
Call forwarding
• External: Trunk supports 302 moved temporarily – “SIP partial rerouting”
(has to be activated in the provider if supported by the provider)
• can be activated for each trunk separately
• Internal: Call fwd will be done internal (2 B-channel will be used) Current connections
• Available B-channels for this trunk Prefix for phone no. (incoming)
• Prefix used in front of CgPN (Route number/ID/RübelKZ) Fax/Modem not possible
• a/b Ports set to FAX/Kombi or ISDN devices using call type “data” will
not be able to use this trunk
SIP trunks
CLIP no screening
• Allows sending numbers which don‘t belong to the SIP account
E.164 conversion
• Incoming call: put country code in front of displayed CLIP
> original CgPN 03061044515 in invite is display as 00493061044515 at the phone
• Outgoing call: replaces dialed country code when destination is located in the same country
> called party number 00493061044515 will be dialed as 03061044515
Dial out Cache
• PBX caches already dialed numbers
• If number is dialed again, PBX allocates the trunk without waiting for
dial timeout
SIP trunks
Create Charges
• PBX will create defined charges per minute
> „Create charges“ has to be activated in user group setting also
> See LCR zones „Charges per minute“
• Used for Hotels or payphones
View
• Trunk is available in CTI50 busy lamp field
• Shows trunk status and used B-channels
Parameter
• No function (different in the past)
• Used for latee implementations
Call Distribution SIP Standard
Same procedure as for MSN
• Shown phone number = defined phone number in sip trunk configuration (better overview)
• Incoming = Outgoing
If using reference trunk, outgoing configuration
has to be made!
Call Distribution SIP DDI
Without wildcards
• Phone number = value defined in “DID: username take from” in the SIP provider settings
> Call distribution has to be made for every DDI
Call Distribution SIP DDI easy
Use of wildcards with “*”
• Phone number = phone number + “*” instead of DDI
• Only one entry necessary
> Useful if all DDI are equal to the internal destination
Call Distribution SIP DDI Mapping
Outgoing call without Mapping will be canceled by the PBX!!!
Mapping to internal extension is possible up to 5 digits (max int. nr length)
Examples
• 0306104* - *
> 1 to 1
• 0306104* - 22*
> 1 to 1 + use of prefix
> Route over QSIG
• 0306104* - 4515
> Map all DDIs to 1 extension
Same procedure for outgoing calls if no special mapping was made:
incoming = outgoing
How to get SIP trunk running
What you need
• Internet Access
• SIP Account
> helpful details about supported features or examples
• Supported Codec's, Payload type, CLIR method, CLIP no Screening support, call fwd 302 moved temporarily, Re-register timer, NAT keep-alive by provider,
