Jump to content

Mitel interoperability


Kristan
 Share

Recommended Posts

Hi All,

 

We have a customer with a Mitel 3300ICP who would to add a remote office with PBXnSIP. We've managed to get some of the mitel setup done (my god it makes you appreciate how simple PBXnSIP is to get running!) but are falling down with the audio. If I make a call from the mitel to an unregistered extension on the PBX (to get voicemail), after the initial invite/trying, the PBX sends the 200 OK and gets this back in response :

[7] 2007/12/19 10:31:34: SIP Rx udp:192.168.10.2:5060:

ACK sip:602@192.168.10.240:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK281759984-57239807

Max-Forwards: 70

From: "xxx" <sip:228@192.168.10.2>;tag=MitelICP_281599984-57239806

To: <sip:602@192.168.10.240>;tag=b0bae1689f

Call-ID: 281599984-57239804

CSeq: 1 ACK

Contact: "xxx" <sip:228@192.168.10.2:5060;transport=udp>

Content-Type: application/sdp

Content-Length: 185

 

v=0

o=- 14633181335402739584 14633181335402739584 IN IP4 192.168.10.3

s=-

c=IN IP4 192.168.10.3

t=0 0

m=audio 20199 RTP/AVP 8 96

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-15

 

[6] 2007/12/19 10:31:34: Sending RTP for 281599984-57239804#b0bae1689f to 192.168.10.3:20199

[8] 2007/12/19 10:31:34: No codec available for sending

 

Mitel is on 192.168.10.2, PBX is on 192.168.10.240 - I've no idea what 192.168.10.3 is, I presume it's the IP of the phone on the mitel side making a call. Now PBXnSIP offers all it's supported codecs, but the mitel isn't negotiating any. If I do the same with one of our asterisk boxes, it looks like it just sends PCMU regardless of what the mitel says (which seems to work).

 

Any ideas???

Link to comment
Share on other sites

Your wish is my command:

 

[7] 2007/12/19 10:31:34: SIP Rx udp:192.168.10.2:5060:

INVITE sip:602@192.168.10.240:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK281599984-57239805

Route: <sip:192.168.10.240:5060;transport=udp;lr>

Max-Forwards: 70

Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,REFER,REGISTER,UPDATE

Supported: timer,replaces

From: "xxx" <sip:228@192.168.10.2>;tag=MitelICP_281599984-57239806

To: <sip:602@192.168.10.240>

Call-ID: 281599984-57239804

CSeq: 1 INVITE

Min-SE: 90

Session-Expires: 90;Refresher=uas

Contact: "xxx" <sip:228@192.168.10.2:5060;transport=udp>

P-Asserted-Identity: "xxx" <sip:228@192.168.10.2:5060;transport=udp>

Content-Length: 0

 

 

[7] 2007/12/19 10:31:34: UDP: Opening socket on port 61252

[7] 2007/12/19 10:31:34: UDP: Opening socket on port 61253

[5] 2007/12/19 10:31:34: Identify trunk (IP address/port and domain match) 2

[9] 2007/12/19 10:31:34: Resolve destination 69745: aaaa udp 192.168.10.2 5060

[9] 2007/12/19 10:31:34: Resolve destination 69745: a udp 192.168.10.2 5060

[9] 2007/12/19 10:31:34: Resolve destination 69745: udp 192.168.10.2 5060

[7] 2007/12/19 10:31:34: SIP Tx udp:192.168.10.2:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK281599984-57239805

From: "xxx" <sip:228@192.168.10.2>;tag=MitelICP_281599984-57239806

To: <sip:602@192.168.10.240>;tag=b0bae1689f

Call-ID: 281599984-57239804

CSeq: 1 INVITE

Content-Length: 0

 

 

[5] 2007/12/19 10:31:34: Trunk Mitel Trunk sends call to 602

[7] 2007/12/19 10:31:34: Calling extension 602

[8] 2007/12/19 10:31:34: Play audio_en/mb_this_is_the_mailbox_of.wav audio_en/bi_6.wav audio_en/bi_0.wav audio_en/bi_2.wav audio_en/mb_leave_msg_after_tone.wav audio_moh/mb_beep.wav

[9] 2007/12/19 10:31:34: Resolve destination 69746: aaaa udp 192.168.10.2 5060

[9] 2007/12/19 10:31:34: Resolve destination 69746: a udp 192.168.10.2 5060

[9] 2007/12/19 10:31:34: Resolve destination 69746: udp 192.168.10.2 5060

[7] 2007/12/19 10:31:34: SIP Tx udp:192.168.10.2:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK281599984-57239805

From: "xxx" <sip:228@192.168.10.2>;tag=MitelICP_281599984-57239806

To: <sip:602@192.168.10.240>;tag=b0bae1689f

Call-ID: 281599984-57239804

CSeq: 1 INVITE

Contact: <sip:602@192.168.10.240:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.2.2292

Content-Type: application/sdp

Content-Length: 378

 

v=0

o=- 17142 17142 IN IP4 192.168.10.240

s=-

c=IN IP4 192.168.10.240

t=0 0

m=audio 61252 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:lGnVrHxY4kJK6730M7LF2LB4T9urAIiLWKyeeTzH

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[9] 2007/12/19 10:31:34: Resolve destination 69747: aaaa udp 192.168.10.2 5060

[9] 2007/12/19 10:31:34: Resolve destination 69747: a udp 192.168.10.2 5060

[9] 2007/12/19 10:31:34: Resolve destination 69747: udp 192.168.10.2 5060

[7] 2007/12/19 10:31:34: SIP Tx udp:192.168.10.2:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK281599984-57239805

From: "xxx" <sip:228@192.168.10.2>;tag=MitelICP_281599984-57239806

To: <sip:602@192.168.10.240>;tag=b0bae1689f

Call-ID: 281599984-57239804

CSeq: 1 INVITE

Contact: <sip:602@192.168.10.240:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/2.1.2.2292

Content-Type: application/sdp

Content-Length: 378

 

v=0

o=- 17142 17142 IN IP4 192.168.10.240

s=-

c=IN IP4 192.168.10.240

t=0 0

m=audio 61252 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:lGnVrHxY4kJK6730M7LF2LB4T9urAIiLWKyeeTzH

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2007/12/19 10:31:34: SIP Rx udp:192.168.10.2:5060:

ACK sip:602@192.168.10.240:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK281759984-57239807

Max-Forwards: 70

From: "xxx" <sip:228@192.168.10.2>;tag=MitelICP_281599984-57239806

To: <sip:602@192.168.10.240>;tag=b0bae1689f

Call-ID: 281599984-57239804

CSeq: 1 ACK

Contact: "xxx" <sip:228@192.168.10.2:5060;transport=udp>

Content-Type: application/sdp

Content-Length: 185

 

v=0

o=- 14633181335402739584 14633181335402739584 IN IP4 192.168.10.3

s=-

c=IN IP4 192.168.10.3

t=0 0

m=audio 20199 RTP/AVP 8 96

a=rtpmap:96 telephone-event/8000

a=fmtp:96 0-15

 

[6] 2007/12/19 10:31:34: Sending RTP for 281599984-57239804#b0bae1689f to 192.168.10.3:20199

[8] 2007/12/19 10:31:34: No codec available for sending

 

There's an option on the Mitel SIP trunk to force SDP in the initial invite, but if I do that the Mitel comes back and says "Error" :)

Link to comment
Share on other sites

What is very strange is that the PBX sends a SRTP key in the offer via an insecure connection. I guess we need to test the cases again where the initial INVITE does not contain SDP...

 

I wondered that too. The whole thing is very strange, those Mitels as monster systems, I'll take PBXnSIP anyday :)

Link to comment
Share on other sites

Join the conversation

You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.

Guest
Reply to this topic...

×   Pasted as rich text.   Paste as plain text instead

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.

 Share

×
×
  • Create New...