Need to implement REFER SIP

[619Tech]

System - MXe3300 - 4.0 SP3 (10.0.3.14_1) w/SIP Trunking that has been in and running without issue for years.

Scenario - Customer recently reported issues transferring to cell phones and created ticket with their carrier.

Carrier - Ran tests with packet captures and sent this update:
"I have confirmed my suspicions and I see that your Mitel phone system appears to have a mis-configuration. I will include a description of my findings and I will attach a call capture that you can forward to your phone system vendor to have this corrected. I see that on the initial inbound call, it gets to the point of sending us a re-invite to transfer the call, and the final SDP that we receive from the Mitel phone system tells us to send audio to ourselves. It appears that the Mitel is telling us to do this in an attempt to have us bridge the audio between the inbound initial call and the outbound transferred call. Unfortunately, since these are 2 separate calls, Windstream has no way of linking them together, as our network is unaware that they are related. You will see in the Wireshark PCAP file that I have attached, the Mitel sends us an ACK on line 25 in response to us sending a 200 ok to the transfer invite. In this ACK’s SDP, the Mitel sets the connection IP to our SBC IP of 10.1.254.5 and this in incorrect. This tells us to send audio to ourselves, which causes the one way audio.
The 2 options here to correct this are:
1) Have the Mitel use the REFER SIP message in the initial inbound call, which will tell our switch to anchor the audio and allow us to connect the 2 audio streams.
2) Have the Mitel correctly set the connection IP on the ACK of the transfer re invite, causing the Mitel to anchor the audio and connect the 2 call."

Actions - I tried to implement option 1; the 4.0 SIP Peer Profile help shows a "Routing Prefix For Unknown URIs" field that mentions the "REFER" message; so I programmed an ARS pattern (8888) to use, but can't find where to input it in the SIP Peer Profile. In the help (4.0 SP2) docs, the field is located between the "Outbound Proxy Server" and 'SMDR Tag", but I do not see it in the database SIP Peer Profile Form. I see the help docs are SP2 and the database is SP3.

I am not sure if option 2 is doable either. What IP, and where would I input it??


Any assistance/input is greatly appreciated other than upgrade/buy SWA, both of which are out of my control.

[ralph]

I seen something similar in ver 6.0.
I had to set "Suppress Redirection Headers" to NO.   I don't know if this is an option in version 4.0 or not.

I'd set that to NO and then set "Avoid Signaling Hold to the Peer" to NO. 
Then retry.

Ralph

[619Tech]

Hi Ralph,

THX for helping. I'm not seeing those. Here is the current Peer Profile Config:

Call Routing and Admin Options
Interconnect Restriction: 1
Max Simultanious Calls: 23
Outbound Proxy Server:
SMDR Tag: 101
Trunk Service: 5
Zone: 1
Alternate Destination Domain Enabled: No
Alternate Destination Domain FQDN/IP:
Enable Special Re-invite Coliision Handling: No
Private Sip Trunk: No
Route Call Using To Header: No

Calling Line ID Options
Default CPN: 714xxxxxxx
CPN Restriction: No
Public Calling Party Number Passthrough: No
Use Diverting number as Callin Party Number: No

SDP Options
Allow Peer To Use Multiple Active M-Lines: No
Enable MItel Proprietary SDP: Yes
Force sending SDP in initial Invite message: Yes
Force sending SDP in initial Invite - Early Answer: No
NAT Keepalive: No
Prevent use of 0.0.0.0 in SDP messages: No
Renegotiate SDP to Enforce Symmetric Codec: No
Repeat SDP Answer iIf Duplicate Offer is received: No
RTP Packetization Rate Override: No
RTP Packetization Rate: 20ms
Special handling of Offers in 2XX responses (INVITE): No
Puppress Use of SDP Inactive Media Streams: No

Signaling and Header Manipulation Options
Session Timer: 90
Build Contact Using Request URI address: No (Maybe this should be yes?)
Disable Reliable Prvisional Responses: Yes
Enable sending + for E.164 numbers: No
Fordce Max-Forward: 70 on Outgoing Calls: No
Ignore Incoming Loose Routing Indications: No
Use P-Asserted Identity Header: No
Use P-Preferred Identity Header: No
Use Restricted Character Set for Authentication: No
Use To Address in From Header on Outgoing Calls: No

[ralph]

I was afraid of that.
I sorta knew that "Suppress Redirection Headers" wasn't in 4.0 but the "Avoid Signaling Hold to the Peer" I thought was.
They've done so much work on SIP that it's hard to keep up with them.

Hopefully someone else can jump in on this.

Ralph

[marcolive]

We ran into similar issues with all kind of SIP equipments/providers. Your MCD is 4 years old. This is not a good thing. I would definitly try to upgrade to the latest release since SIP support has greatly impoved over time.