1-800 403 error

[RLyon]

I am trying to program 6 incoming numbers. 4 local numbers work, where as our toll free numbers are getting a 403 error on the DB side. Is there an area or flag i need to enable to allow 1-800 numbers to come in. Ever since I ported the numbers over to flowroute they do not work. Flowroute tested it and they saw the 403 on my end.

[acejavelin]

Numbers coming in are just numbers, the phone system doesn't know or care if they are toll free or not.

That said, we would need to know what type of trunks and the DNIS digits being presented and how you are routing them, the contents of the CRT, etc.

I am guessing these are SIP or PRI trunks and you are using a CRT? If so, add a E and + entries at the bottom and route it to known destination.

If these are SIP trunks, you could do a SIP capture and debug the invites to see what is being sent and route it accordingly. If this is a PRI then you can use the terminal to look at the Q.931 messages and determine the digits being presented.

[NEPhoneGuy]

Numbers coming in are just numbers, the phone system doesn't know or care if they are toll free or not.

That said, we would need to know what type of trunks and the DNIS digits being presented and how you are routing them, the contents of the CRT, etc.

I am guessing these are SIP or PRI trunks and you are using a CRT? If so, add a E and + entries at the bottom and route it to known destination.

If these are SIP trunks, you could do a SIP capture and debug the invites to see what is being sent and route it accordingly. If this is a PRI then you can use the terminal to look at the Q.931 messages and determine the digits being presented.

What digits are being sent to the PBX from the carrier - toll free numbers usually come in over a local number. With a provider change they may be sending it over something different.

[RLyon]

Still having issues. This is hard to read. It seems that is not hitting the trunk and what could be the issue?

-01:595- 09:56 03-20 *** SIP RX [''-'P9000']: INVITE sip:1877XXXXXXX@192.XXX.XXX.XXX:5060 SIP/2.0
Record-Route: <sip:216.115.69.144;lr>
From: "WIRELESS CALLER" <sip:+1715XXXXXXXX@fl.gg;isup-oli=00>;tag=gK04304e60
Max-Forwards: 66
Record-Route: <sip:216.115.69.132;lr>
To: <sip:+1877XXXXXXX@fl.gg>
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bKde0b.6707393c12425d571ef96fd6e22be0f1.0
Via: SIP/2.0/UDP 54.71.6.127:5060;branch=z9hG4bKde0b.e443fd5a6072fc6aa59002da0161a3ce.0
Via: SIP/2.0/UDP 216.115.69.132;branch=z9hG4bKde0b.661fd8965652f7ddbcdb5e6aae590039.0
Via: SIP/2.0/UDP 23.29.25.120:5060;branch=z9hG4bK04B33c4078feb58e22b
Call-ID: 1847888491_62665600@23.29.25.120
CSeq: 45452 INVITE
Contact: <sip:wlssuser@23.29.25.120:5060;sipappsessionid=app-p95s925otgga;wlssfcid=sip-h2pqyq5gib91>
Session-Expires: 1800
Min-SE: 90
Content-Length:   224
Content-Type: application/sdp
P-Asserted-Identity: "WIRELESS CALLER" <sip:+1715XXXXXX@fl.gg>

v=0
o=- 499329 454196 IN IP4 23.29.25.124
s=-
c=IN IP4 23.29.25.124
t=0 0
m=audio 52984 RTP/AVP 0 18 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20

-01:596- 09:56 03-20 *** SIP TX [''-'']: SIP/2.0 403 Forbidden
Record-Route: <sip:216.115.69.144;lr>,<sip:216.115.69.132;lr>
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bKde0b.6707393c12425d571ef96fd6e22be0f1.0,SIP/2.0/UDP 54.71.6.127:5060;branch=z9hG4bKde0b.e443fd5a6072fc6aa59002da0161a3ce.0,SIP/2.0/UDP 216.115.69.132;branch=z9hG4bKde0b.661fd8965652f7ddbcdb5e6aae590039.0,SIP/2.0/UDP 23.29.25.120:5060;branch=z9hG4bK04B33c4078feb58e22b
From: "WIRELESS CALLER" <sip:+1715XXXXX@fl.gg;isup-oli=00>;tag=gK04304e60
To: <sip:+1877XXXXXX@fl.gg>;tag=Mitel-5000_3558397864-26664
Call-ID: 1847888491_62665600@23.29.25.120
CSeq: 45452 INVITE
Content-Length: 0


-01:597- 09:56 03-20 *** SIP RX [''-'P9000']: ACK sip:1877-XXX-XXXX@192..XXX.XXX.XXX:5060 SIP/2.0
From: "WIRELESS CALLER" <sip:+1715XXXXXXX@fl.gg;isup-oli=00>;tag=gK04304e60
Max-Forwards: 66
To: <sip:+1877XXXXXXX@fl.gg>;tag=Mitel-5000_3558397864-26664
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bKde0b.6707393c12425d571ef96fd6e22be0f1.0
Call-ID: 1847888491_62665600@23.29.25.120
CSeq: 45452 ACK
Content-Length: 0

[NEPhoneGuy]

your getting a 403 below

To: <sip:+1877XXXXXX@fl.gg>;tag=Mitel-5000_3558397864-26664

i'd expect to see a not found - do other numbers work okay? What's the full to: number and are you routing that in your call routing table?

[RLyon]

1-877-384-3287. I shouldnt even need a CRT if all numbers are going to the same CRA

[acejavelin]

1-877-384-3287. I shouldnt even need a CRT if all numbers are going to the same CRA

Always use a CRT on a trunk that is sending "digits"... if all go to the same number, just use the E and + entries and point them to the same place.

[RLyon]

There is an E and +  at the end and it's still not working

[acejavelin]

There is an E and +  at the end and it's still not working

And you have that trunk group pointed to that CRT? Is it the same result if you enter the last 4 digits of the number as well (above the + and E, it is a top down first match list)?

[RLyon]

Yes, the trunk group is pointed towards the CRT. I have the E and the + at the end. I do not have inbound authentication disabled. I tried enabling it to see if there was a miracle, but it did not work. I tried using the last 4 digits. The Numbers are at the top of the CRT. I contacted Flowroute and this is what they said..I am not sure what to tell, your system is rejecting it, given the 403. Most often this is because the SIP Server IP or FQDN is not listed as PEER, inbound authentication is turned on, etc. I am pasting the two invites below for your consideration, as I am not to familiar with the MITEL 5000 and maybe it is blocking the call based on the Contact Header.

1-877 contact Contact: <sip:wlssuser@23.29.25.120:5060;sipappsessionid=app-uv7ef9sraytm;wlssfcid=sip-1xhb1vw736smn>
locall number Contact Contact: <sip:+1715XXXXXXX@74.XXX.XXX.XXX:5060>

[NEPhoneGuy]

Any chance you could post the setup of working DID from this provider along with another one of the non working number.

edit- on another look here To: <sip:+1877XXXXXX@fl.gg>;tag=Mitel-5000_3558397864-26664 - I'd expect to see your public or lan address behind the @ on the to field on the invite also.

I just reread your post again and see the contact fields are different for the local vs 800 number. I'm not real familiar with flow route but it's odd they aren't just sending the toll free number over another local number. You might have to engage support with this one unfortunately.

[sarond]

Yes you may have to trust the IP address that the call is coming from. You can do this via
System > Devices and Feature Codes > SIP Peers > SIP Trunk Groups > {Trunk Group} > Configuration > Alternate IP/FQDN

Add the IP addresses that a call could appear from.

[Tech Electronics]

RLyon,

In your Alternate FQDN/IP Address table put the following FQDN entry.

FQDN: default

This will allow for any incoming IP or FQDN address into the SIP Peer Trunk Group.

Thanks,

TE

[RLyon]

Tech Electronics.... Thank you. that worked. Is it safe to keep it as default?

[Tech Electronics]

RLyon,

Safe is a relative term and my guess is that since the carrier is using different methods to support their features then you are stuck with this as your answer to their solution unless you can find the pattern used.

It might be better to just put in the FQDN ff.gg and remove default which allows it to answer anything.

I typically use the default FQDN to gather information, but some carriers like NexVortex use other carriers to send the calls and default is the only option without putting in an MBG; which we will at that point since the MiVO doesn’t support DNS SRV.

Thanks,

TE


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