SIP trunking questions

[acejavelin]

I am setting up a new SIP peer... two questions regarding incoming, and incoming to outgoing forwarded calls. Outgoing calls are working fine.

The first one I know I have seen before and it was easy, but I cannot for the life remember what it is, any incoming call the Mitel sends back a 503 Service Unavailable message (verified in Wireshark it's from the 3300). Outgoing calls with this number set as Outgoing CLID work, the number is in the DID routing table to the proper extension, and it is in the SIP Number Peering form. I know this one is something stupid, I just can't remember what it is.

The second question is that the carrier has said they want the main number sent as the P-Assert Identity number for any call that is not sending a number provisioned on the SIP service... So when forwarding to a cell phone we want the callers CID to pass through the cell phone, in order to send an outbound CLID that is not a number on the SIP trunk, we need to send the main billing number in the P-Assert Identity field. Can I do that? I know there are some P-Assert options on the SIP trunks, but the descriptions are less than clear.

[acejavelin]

Well, fixed my incoming issue... forget to set the number of digits to absorb in Trunk Attributes, it as blank. Put in a zero and it worked first try.

Still not understanding the P-Assert Identity thing though.

[petr.necas]

So in P-Asserver-Identity should be the original callers number? For me this is also a magic, I cannot test it now but try:

Public Calling Party Number Passthrough                        set to Yes
Use Diverting Party Number as Calling Party Number      set to No
Use Original Calling Party Number If Available                set to Yes
Use P-Asserted Identity Header                                     set to Yes
   
   

[acejavelin]

So in P-Asserver-Identity should be the original callers number? For me this is also a magic, I cannot test it now but try:

Public Calling Party Number Passthrough                        set to Yes
Use Diverting Party Number as Calling Party Number      set to No
Use Original Calling Party Number If Available                set to Yes
Use P-Asserted Identity Header                                     set to Yes
   

No, I don't think you understand ... let me set an example

Main billing number on SIP trunk is 612-555-1212. A call comes in from a cell phone (212-222-1111) and rings x29999 which is forwarded to a different cell number (651-444-7777)... the 212 and 651 numbers are not part of the SIP service from the carrier, so they want to see an invite like:

TO: <sip:6514447777@ipaddr:5060;transport=UDP>
FROM: <sip:2122221111@ipaddr:5060;transport=UDP>
P-Asserted-Identiy:<sip:6125551212@ipaddr:5060>

So the correct original caller's CID (212-222-1111) is carried through to 651-444-7777 and displayed on the cell phone, but any billing associated with it done against 612-555-1212

At least that is how I understand it.

According to my notes, the exact wording the SIP carrier's tech used was: You need to send us the main billing number in the P-Assert Identity field for any call that does not send an outbound CID we recognize as a DID on the SIP trunk. Actually, you can send any DID number as the P-Assert Identity so we know how to bill the call.

EDIT Nevermind, I just tested the above scenario and it works as fine. This must be the default behavior somehow. Guess I don't really have a problem.