Mitel Forums - The Unofficial Source
Mitel Forums => Mitel MiVoice Business/MCD/3300 => Topic started by: foanthyme on May 15, 2017, 10:43:07 AM
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Hi,
We have an existing auto dialling phone that was set up by a previous colleague. When you lift the handset, it calls a set number.
Recently we bought another phone (different model), a JR101-CB to be precise. I've added the phone as a Generic SIP phone, I've set the class of service on the phone to allow calling, the SIP Device Capabilities matches the existing working phone. The password matches the one that I've set in User Configuration, but I can't get the JR101 to dial out. It has a beeping sound as soon as a lift the receiver, but does not call.
I can call the auto dialler from my desk phone, so dialling into it is working correctly.
We have a 3CX testing environment on a separate network, the phone worked instantly on this, so I don't think it's a phone issue.
Here is part of the Wireshark
The phone is set as #408, trying to call #485 for testing purposes. 192.168.1.20 is the Mitel server.
INVITE sip:485@192.168.1.20 SIP/2.0
From: "408"<sip:408@192.168.1.20>;tag=72bb88-bc01a8c0-13c4-55013-d-62a1e43b-d
To: <sip:485@192.168.1.20>
Call-ID: 739a08-bc01a8c0-13c4-55013-d-513ef6cd-d
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.1.188:5060;rport;branch=z9hG4bK-d-358f-73fd6671
Max-Forwards: 70
Supported: replaces,eventlist,timer
Allow: REGISTER, INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, INFO, OPTIONS, PRACK, SUBSCRIBE, PUBLISH
User-Agent: JPhone/1.3.1.18 (JPHONE-Rev; 4A2652B42710)
Contact: <sip:408@192.168.1.188>
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 301
v=0
o=408 1494339123 1494339123 IN IP4 192.168.1.188
s=-
c=IN IP4 192.168.1.188
t=0 0
m=audio 4002 RTP/AVP 8 0 18 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.188:5060;rport;branch=z9hG4bK-d-358f-73fd6671
From: "408" <sip:408@192.168.1.20>;tag=72bb88-bc01a8c0-13c4-55013-d-62a1e43b-d
To: <sip:485@192.168.1.20>;tag=0_2471797312-60209365
Call-ID: 739a08-bc01a8c0-13c4-55013-d-513ef6cd-d
CSeq: 1 INVITE
Content-Length: 0
I've tried disabling the current phone and giving the new phone the same number as the old one, to no avail.
Any tips appreciated.
Thanks
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What about your Class of Restriction?
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What is the response to the INVITE request? Could you please post here the PCAP file?
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Hi,
Here's a link to the pcapng file, I can make a longer one if required..
http://gofile.me/6lqWm/BqNkOjhfm
Class of Restriction is set up correctly as per other phones. I've been adding/moving Mitel phones from this system for around 3 years, so I'm fairly familiar with a normal Mitel phone set up.
Thanks for your time.
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From the PCAP file I can see the phone is dialing extension 599 and the PBX is replying with a message that this extension is not found, does this extension exists?
Frame 28: 393 bytes on wire (3144 bits), 393 bytes captured (3144 bits) on interface 0
Ethernet II, Src: MitelCor_16:d7:16 (08:00:0f:16:d7:16), Dst: 4a:26:52:b4:27:06 (4a:26:52:b4:27:06)
Internet Protocol Version 4, Src: 192.168.1.20, Dst: 192.168.1.223
User Datagram Protocol, Src Port: 5060, Dst Port: 5060
Session Initiation Protocol (404)
Status-Line: SIP/2.0 404 Not Found
Message Header
Via: SIP/2.0/UDP 192.168.1.223:5060;rport;branch=z9hG4bK-c5-301df-5028bcf9
From: "408" <sip:408@192.168.1.20>;tag=72c378-df01a8c0-13c4-55013-c5-71dfe3-c5
To: <sip:599@192.168.1.20>;tag=0_2412653680-60222541
Call-ID: 739e98-df01a8c0-13c4-55013-c5-61db07f5-c5
CSeq: 1 INVITE
Contact: <sip:192.168.1.20>
Content-Length: 0
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After sending Wiresharks to the phone maker, they changed the firmware to force it to send the port number to the Mitel system, the phone was assuming 5060, but wasn't sending it. They said this is a new issue for them as all other phone systems assume 5060.
All sorted :)