Mitel Forums - The Unofficial Source
Mitel Forums => Mitel MiVoice Business/MCD/3300 => Topic started by: celswood on May 01, 2020, 09:28:52 AM
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Hi all,
Picked up a customer with some legacy kit. Moved their PRI over to SIP and their Linksys SPA8000 fail to make a call outbound. From the CCS trace it shows the outbound call over the SIP UA leg with a CIM/CCM of 0 but doesn't show the outbound leg of the call over the SIP Trunking Peer, can see the device making the call using the correct trunk access digit.
Set up a SIP softphone using the same UA and can make calls outbound ok so I don't think its the MiVB (v9.0) configuration, had a look at the SPA8000 and there is tons of options, not sure where to start. If anyone has a basic config that I can start on would be greatly appreciated.
Also note that the SPA8000 is on the same subnet as the MBG which the SIP trunks terminate.
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You will probably have to do a SIP trace to understand what's going on, but what does it say on the display when you try to make a call?
Ralph
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Hi Ralph,
Ran a SIP trace from the MiVB, currently the SPA8000 is getting a 403 Forbidden from the MiVB with the following reason "399 x.x.x.x "29H Access Barred"", checked COS for PSTN Access over DPNSS and COR access over the route, no problems there. When I attach a SIP softphone using the same credentials I can make calls fine.
Thanks,
Craig
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Hi Ralph,
Ran a SIP trace from the MiVB, currently the SPA8000 is getting a 403 Forbidden from the MiVB with the following reason "399 x.x.x.x "29H Access Barred"", checked COS for PSTN Access over DPNSS and COR access over the route, no problems there. When I attach a SIP softphone using the same credentials I can make calls fine.
Thanks,
Craig
What is the SPA8000 sending out in the invite? Can you attach that?
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Session Initiation Protocol (INVITE)
Request-Line: INVITE sip:xxxxxxxxxx@x.x.x.xSIP/2.0
Message Header
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-c60f402a
From: "1122" <sip:xxx@x.x.x.x>;tag=28b0ef2c58330edao1
To: <sip:xxxxxxxxxx@x.x.x.x>
Remote-Party-ID: "xxxx" <sip:xxxx@x.x.x.x>;screen=yes;party=calling
[Expert Info (Note/Undecoded): Unrecognised SIP header (remote-party-id)]
[Unrecognised SIP header (remote-party-id)]
[Severity level: Note]
[Group: Undecoded]
Call-ID: 4b9a4b14-3ce32412@x.x.x.x
[Generated Call-ID: 4b9a4b14-3ce32412@x.x.x.x]
CSeq: 101 INVITE
Sequence Number: 101
Method: INVITE
Max-Forwards: 70
Contact: "xxxx" <sip:xxxx@x.x.x.x:5060>
Expires: 240
User-Agent: Linksys/SPA8000-6.1.12SR1
Allow-Events: talk, hold, conference
Content-Length: 444
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 21225879 21225879 IN IP4 x.x.x.x
Session Name (s): -
Connection Information (c): IN IP4 x.x.x.x
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 20827 RTP/AVP 0 2 4 8 18 96 97 98 100 101
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:2 G726-32/8000
Media Attribute (a): rtpmap:4 G723/8000
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute (a): rtpmap:18 G729a/8000
Media Attribute (a): rtpmap:96 G726-40/8000
Media Attribute (a): rtpmap:97 G726-24/8000
Media Attribute (a): rtpmap:98 G726-16/8000
Media Attribute (a): rtpmap:100 NSE/8000
Media Attribute (a): fmtp:100 192-193
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Attribute (a): ptime:20
Media Attribute (a): sendrecv
[Generated Call-ID: d00bf67-6c3188fe@x.x.x.x]
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You get the same result with a 4 digit and external call?
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I have the deployment guide for this from Mitel. Its really old from Jan 2009.
send me a message with your email and I can send it on to you if you want it.