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Cannot forward unconditionally a call on the same SIP trunk 603 Decline

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digiteltlc

Technical User
Apr 19, 2005
56
IT
I have a single SIP trunk from a local provider
I'm not able to forward unconditionally a call coming from this trunk to another external number via the same trunk
The original call is targeted to a single extension via DDI , this extension is forwarded unconditionally to an external no.
A call coming from an ISDN line with this extension as target is correctly forwarded via SIP trunk
Tracking detail in sistem status say :
Direction = To pbx, From = <incomingCLInumber>@<providerIPaddress>, To = <forward_destination_no>@<providerHostname>, Answer = 603 Decline

Trunk config (I'm translating from Italian sorry for any incongruence):

SIP LINE folder:
Send caller ID = Absent
Association method = By origin IP address
Use URI tel = NOT checked
Routing call method = URI requested
"From" header caller ID and send clear header "From" both checked

URI SIP folder
Local URI, Contact and Displayed name = Use internal data
PAI = Absent

Any hint please ??



 
Do a SIP trace, what response do you get from the SIP provider?

"Trying is the first step to failure..." - Homer
 
Well, your SIP provider doesn't allow the call.

I suspect that it has to do with you sending Contact as anonymous in your INVITE.
Contact: <sip:anonymous@IPOFFICE_LAN_ADDRESS:39814;transport=udp>

This should have the forwarding extensions CLI so the service provider knows who forwarded the number.

You need to enter some CLI information on the SIP tab on the forwarding user, you could just use the main number.

"Trying is the first step to failure..." - Homer
 
Well, in fact I had the "anonymous" flag checked on forwarded extension SIP tab .....my fault
Anyway, once unchecked, the behaviour is the same

I have forgotten to say that the message received by the caller is "busy"
The thing that surprise me is that "busy" message returned by IPO :

CALL:2015/03/3016:25,00:00:00,000,CALLER_CLI@PROVIDER_IP_ADDRESS,I,674,674,CALLER_CLI,,,0,,""n/a,0
16:24:59 2262648855mS CD: CALL: 17.150590.1 BState=Idle Cut=1 Music=0.0 Aend="Line 17" (0.0) Bend="Extn674(674)" [] (0.0) CalledNum=674 (Extn674) CallingNum=CALLER_CLI@PROVIDER_IP_ADDRESS (CALLER_CLI) Internal=0 Time=62 AState=Dialled
16:24:59 2262648855mS CD: CALL: 17.150590.1 Deleted
16:24:59 2262648856mS CMCallEvt: 17.150590.1 -1 SIPTrunk Endpoint: StateChange: END=X CMCSDialled->CMCSCompleted 16:24:59 2262648857mS SIP Tx: UDP IPOFFICE_LAN_ADDRESS:5060 -> PROVIDER_IP_ADDRESS:5060
SIP/2.0 486 Busy Here Via: SIP/2.0/UDP PROVIDER_IP_ADDRESS:5060;branch=z9hG4bK7a3ddebc;rport From: "CALLER_CLI" <sip:CALLER_CLI@PROVIDER_IP_ADDRESS>;tag=as235f4942 To: <sip:CALLED_DID@IPOFFICE_PUBLIC_ADDRESS:39814>;tag=fbe1571d26f1e601 Call-ID: 30d36f3d325b4e4e542c7e3212008c94@PROVIDER_IP_ADDRESS
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer,100rel
Server: IP Office 8.1 (43)
Content-Length: 0

It seems IPO try to call himself , am I wrong ??


On extension SIP tab the full public DID of extension is set into all fields
I've tried to leave the "Contact" field empty, IPO fills it automatically with the extension number
Same overall behaviour anyway...

Some other idea ??

Thank you again
 
The IPO will send busy to the incoming call since the outgoing leg gets DECLINED and the call has nowhere to go.

Your contactfield should be a valid DDI on your system.

"Trying is the first step to failure..." - Homer
 
Just a thought :

A regular call placed from an IPO local phone looks like this :

INVITE sip:CALLED_NUMBER@PROVIDER_HOSTNAME SIP/2.0
Via: SIP/2.0/UDP IPO_LAN_IP_ADDRESS:39814;rport;branch=z9hG4bK7e92ca5b6aa867c6b9631f0230207766
From: "EXTENSION_DID" <sip:EXTENSION_DID@PROVIDER_HOSTNAME>;tag=646f4557a8f47d9d
To: <sip:CALLED_NUMBER@PROVIDER_HOSTNAME>
Call-ID: 7b2e4b09a8e43d23dcb34b5c4ac6a0ec
CSeq: 28374157 INVITE
Contact: "EXTENSION_DID" <sip:EXTENSION_DID@IPO_LAN_IP_ADDRESS:39814;transport=udp>

where EXTENSION_DID is the value present in extension SIP tab


a forwarded call, instead, looks like :

INVITE sip:FORWARD_DESTINATION_NUMBER@PROVIDER_HOSTNAME SIP/2.0
Via: SIP/2.0/UDP IPOFFICE_LAN_ADDRESS:39814;rport;branch=z9hG4bK8a91c3ccca4fd990aab2fc13ed7c2344
From: "CALLER_CLI" <sip:CALLER_CLI@PROVIDER_IP_ADDRESS>;tag=19ecae0f1146e294
To: <sip:FORWARD_DESTINATION_NUMBER@PROVIDER_HOSTNAME>
Call-ID: f399488e9216f65a971faf1f7a706a93
CSeq: 1697547974 INVITE
Contact: "EXTENSION_DID" <sip:EXTENSION_DID@IPO_LAN_IP_ADDRESS:39814;transport=udp>


In the second (not working) case, is it possible that From: "CALLER_CLI" <sip:CALLER_CLI@PROVIDER_IP_ADDRESS> the provider refuse call as coming from a number not of his own ??


 
It might be, you need to ask your service provider what they require in this case.

Here in the Nordics they are usually happy if they get the DID in the Contact field, but I know many providers want it in PAI or might not allow it at all.

"Trying is the first step to failure..." - Homer
 
In that case, can i force a forwarded call coming from an external caller to be presented out with the extension DID in the "From" field rather than the original caller CLI ??
 
On the SIP trunk you can enter "Originator number for forwarded and twinning calls".

You could also try to change the "Send Caller ID" to Diversion Header or P Asserted ID depending on what you have now.
Your service provider should support one of them.

"Trying is the first step to failure..." - Homer
 
On the SIP trunk you can enter "Originator number for forwarded and twinning calls".

Thank you, just done, it works like a charm !! (i forgot about...)

At least the main company number is displayed as ID on the called party.

As PAI is set to absent on URI tab , I'll try to enable it as "use internal data" and change "Send Caller ID" to use PAI, as soon as customer allows me for a reboot, doing so the extension DID should be displayed to the called.


I asking myself (honestly , asking to all of you reading here) if there is a method to pass to Forwarded destination party the original CLI of incoming caller while passing the IPOffice CLI to SIP provider in order to accept calls.




 
It all depends on what you service provider supports and how they want the information.
We usually get a technical description from the provider where they say if they are using Contact, Diversion Header etc for these type of calls.

"Trying is the first step to failure..." - Homer
 
Meanwhile I've done some tests

The best result achieved is to have incoming call forwarded to destination using the extension DID as callerID

It was enough to uncheck "send in clear "from" header" on SIP line, (as "use internal data" was already set on URI tab values".
All other combination fail as a unmanaged ID is sent to provider.

 
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