Hosting SIP trunks with MTE

Submitted by eeman on Thu, 03/05/2009 Permalink

did you write a pattern on your trunk group?

voice grouped-trunk ROUTING

no description

trunk T01

accept $ cost 1

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Submitted by eeman on Thu, 03/05/2009 Permalink

btw adtran is very helpful when it comes to getting these total access devices working.

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Submitted by ryan.tuttle on Fri, 03/06/2009 Permalink

Thank you for the help, actually the calls are getting to the Asterisk Server its just sending back a 404 not found and playing a recording that say's the extension cannot be found. I think the problem is on the MTE, not the Adtran.

And yes, adtran is always very helpful when it comes to integrating their products.

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Submitted by eeman on Fri, 03/06/2009 Permalink

can you paste a capture of the console when calling out from the adtran? Do you have the adtran setup as an extension in pbxmanager or a trunk? What context did you select for the trunk if thats what you selected?

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Submitted by ryan.tuttle on Fri, 03/06/2009 Permalink

I have it built as a trunk right now, I've tried an extension as well. The trunks context is from-inside-SIPTRUNK (SIPTRUNK being the name of the Tenant Group).

<--- SIP read from 66.178.152.118:5060 --->

INVITE sip:5414923028@66.178.167.79:5060 SIP/2.0

From: "Test Phone";tag=234e388-0-13c4-12773-659b5c95-12773

To:

Call-ID: 2360438-0-13c4-12773-4a9d3c9c-12773@66.178.167.79

CSeq: 1 INVITE

Via: SIP/2.0/UDP 66.178.152.118:5060;branch=z9hG4bK-12773-482198a-4b6037fa

Max-Forwards: 70

Supported: 100rel,replaces

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTION S, PRACK, REFER, REGISTER

User-Agent: ADTRAN_Total_Access_908_2nd_Gen/A1.05.00.E

Contact:

Content-Type: application/SDP

Content-Length: 268

v=0

o=- 1215259785 1215259785 IN IP4 66.178.152.118

s=-

c=IN IP4 66.178.152.118

t=0 0

m=audio 10000 RTP/AVP 0 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=silenceSupp:off - - - -

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

<------------->

--- (13 headers 12 lines) -- -

[Kappsrv1*CLI>

Sending to 66.178.152.118 : 5060 (no NAT)

[Kappsrv1*CLI>

Using INVITE request as basis request - 2360438-0-13c4-12773-4a9d3c9c-12773@66.178.167.79

[Kappsrv1*CLI>

Found peer 'Test_Adtran'

[Kappsrv1*CLI>

<--- Reliably Transmitting (no NAT) to 66.178.152.118:5060 --->

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 66.178.152.118:5060;branch=z9hG4bK-12773-482198a-4b6037fa;received=66.178.152.118

From: "Test Phone";tag=234e388-0-13c4-12773-659b5c95-12773

To: ;tag=as324bd88d

Call-ID: 2360438-0-13c4-12773-4a9d3c9c-12773@66.178.167.79

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, C ANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="53a86005"

Content-Length: 0

<------------>

Scheduling destruction of SIP dialog '2360438-0-13c4-12773-4a9d3c9c-12773@66.178.167.79' in 6400 ms (Method: INVITE)

[Kappsrv1*CLI>

<--- SIP read from 66.178.152.118:5060 --->

ACK sip:5414923028@66.178.167.79:5060;transport=UDP SIP/2.0

From: "Test Phone";tag=234e388-0-13c4-12773-659b5c95-12773

To: ;tag=as324bd88d

Call-ID: 2360438-0-13c4-12773-4a9d3c9c-12773@66.178.167.79

CSeq: 1 ACK

Via: SIP/2.0/UDP 66.178.152.118:5060;branch=z9hG4bK-12773-482198a-4b6037fa

Max-Forwards: 70

Supported: 100rel,replaces

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

User-Agent: ADTRAN_Total_Access_908_2nd_Gen/A1.05.00.E

Contact:

Content-Length: 0

<------------->

--- (12 headers 0 lines) ---

[Kappsrv1*CLI>

<--- SIP read from 66.178.152.118:5060 --->

INVITE sip:5414923028@66.178.167.79:5060 SIP/2.0

From: "Test Phone";tag=234e388-0-13c4-12773-659b5c95-12773

To:

Call-ID: 2360438-0-13c4-12773-4a9d3c9c-12773@66.178.167.79

CSeq: 2 INVITE

Via: SIP/2.0/UDP 66.178.152.118:5060;branch=z9hG4bK-12773-48219a5-5cf61e0b

Max-Forwards: 70

Supported: 100rel,replaces

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTION S, PRACK, REFER, REGISTER

User-Agent: ADTRAN_Total_Access_908_2nd_Gen/A1.05.00.E

Contact:

Proxy-Authorization: Digest username="Test_Adtran",realm="asterisk",nonce="53a86005",uri="sip:5414923028@66.178.167.79:5060",response="28e927ab78305131e0e985195bf19710",algorithm=MD5

Content-Type: application/SDP

Content-Length: 268

v=0

o=- 1215259785 1215259785 IN IP4 66.178.152.118

s=-

c=IN IP4 66.178.152.118

t=0 0

m=audio 10000 RTP/AVP 0 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=silenceSupp:off - - - -

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

<------------->

--- (14 headers 12 lines) ---

Sending to 66.178.152.118 : 5060 (no NAT)

Using INVITE request as basis request - 2360438-0-13c4-12773-4a9d3c9c-12773@66.178.167.79

Found peer 'Test_Adtran'

Found RTP audio format 0

Found RTP audio format 18

Found RTP audio format 101

Peer audio RTP is at port 66.178.152.118:10000

Found audio description format PCM U for ID 0

Found audio description format G729 for ID 18

Found audio description format telephone-event for ID 101

Capabilities: us - 0x4 (ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)

Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)

Peer audio RTP is at port 66.178.152.118:10000

Looking for 5414923028 in from-inside-SIPTRUNK (domain 66.178.167.79)

list_route: hop:

<--- Transmitting (no NAT) to 66.178.152.118:5060 --->

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 66.178.152.118:5060;branch=z9hG4bK-12773-48219a5-5cf61e0b;received=66.178.152.118

From: "Test Phone";tag=234e388-0-13c4-12773-659b5c95-12773

To:

Call-ID: 2360438-0-13c4-12773-4a9d3c9c-12773@66.178.167.79

CSeq: 2 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REF ER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact:

Content-Length: 0

<------------>

[Kappsrv1*CLI>

-- Executing [5414923028@from-inside-SIPTRUNK:1] Macro("SIP/5414926005-08221e50", "tl-set-variables2|from-inside-redir-SIPTRUNK|SIPTRUNK") in new stack

[Kappsrv1*CLI>

-- Executing [s@macro-tl-set-variables2:1] Set("SIP/5414926005-08221e50", "__tenant=SIPTRUNK") in new stack

[Kappsrv1*CLI>

-- Executing [s@macro-tl-set-variables2:2] Set("SIP/5414926005-08221e50", "CDR(userfield)=SIPTRUNK") in new stack

[Kappsrv1*CLI>

-- Executing [s@macro-tl-set-variables2:3] Set("SIP/5414926005-08221e50", "__MOH=default-SIPTRUNK") in new stack

[Kappsrv1*CLI>

-- Executing [s@macro-tl-set-variables2:4] GotoIf("SIP/5414926005-08221e50", "1 ?setmoh") in new stack

-- Goto (macro-tl-set-variables2,s,6)

-- Executing [s@macro-tl-set-variables2:6] SetMusicOnHold("SIP/5414926005-08221e50", "default-SIPTRUNK") in new stack

[Kappsrv1*CLI>

-- Executing [s@macro-tl-set-variables2:7] Goto("SIP/5414926005-08221e50", "from-inside-redir-SIPTRUNK|5414923028|1") in new stack

[Kappsrv1*CLI>

-- Goto (from-inside-redir-SIPTRUNK,5414923028,1)

== Channel 'SIP/5414926005-08221e50' jumping out of macro 'tl-set-variables2'

-- Sent into invalid extension '5414923028' in context 'from-inside-redir-SIPTRUNK' on SIP/5414926005-08221e50

[Kappsrv1*CLI>

-- Executing [i@from-inside-redir-SIPTRUNK:1] Playback("SIP/5414926005-08221e50", "invalid") in new stack

[Kappsrv1*CLI>

Audio is at 66.178.167.79 port 16276

[Kappsrv1*CLI>

Adding codec 0x4 (ulaw) to SDP

Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 66.178.152.118:5060 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 66.178.152.118:5060;branch=z9hG4bK-12773-48219a5-5cf61e0b;received=66.178.152.118

From: "Test Phone";tag=234e388-0-13c4-12773-659b5c95-12773

To: ;tag=as071377ee

Call-ID: 2360438-0-13c4-12773-4a9d3c9c-12773@66.178.167.79

CSeq: 2 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact:

Content-Type: application/sdp

Content-Length: 240

v=0

o=root 5662 5662 IN IP4 66.178.167.79

s=session

c=IN IP4 66.178.167.79

t=0 0

m=audio 16276 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

<------------>

[Kappsrv1*CLI>

-- Playing 'invalid' (language 'en')

[Kappsrv1*CLI>

<--- SIP read from 66.178.152.118:5060 --->

ACK sip:5414923028@66.178.167.79;transport=UDP SIP/2.0

From: "Test Phone";tag=234e388-0-13c4-12773-659b5c95-12773

To: ;tag=as071377ee

Call-ID: 2360438-0-13c4-12773-4a9d3c9c-12773@66.178.167.79

CSeq: 2 ACK

Via: SIP/2.0/UDP 66.178.152.118:5060;branch=z9hG4bK-12773-48219c9-5b3a8b38

Max-Forwards: 70

Supported: 100rel,replaces

Allow: ACK, BYE, CANCEL, INFO, INVI TE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

User-Agent: ADTRAN_Total_Access_908_2nd_Gen/A1.05.00.E

Contact:

Proxy-Authorization: Digest username="Test_Adtran",realm="asterisk",nonce="53a86005",uri="sip:5414923028@66.178.167.79:5060",response="28e927ab78305131e0e985195bf19710",algorithm=MD5

Content-Length: 0

<------------->

--- (13 headers 0 lines) ---

[Kappsrv1*CLI>

<--- SIP read from 66.178.161.82:5060 --->

REGISTER sip:66.178.167.79 SIP/2.0

Via: SIP/2.0/UDP 66.178.161.82:5060;branch=z9hG4bKb8f06a83c536416c402e03c97c1ce8cc

Via: SIP/2.0/UDP 10.60.3.39:5060;branch=z9hG4bK4104539c36c33cea4.991d760fdb2f977ea

From: ;tag=11dc51ca39

To:

Call-ID: 66e9f8ede08c82b8

CSeq: 32467 REGISTER

Contact: "6006-SIPTRUNK" ;+sip.instance=""

Authorization: Digest username="6006-SIPTRUNK", realm="asterisk", nonce="43ea7bf4", uri="sip:66.178.167.79", response="8d97c8bfeac784c27fee30d6f3a516e4", algorithm=MD5

Allow: INVITE

Allow: ACK

Allow: CANCEL

Allow: BYE

Allow: NOTIFY

Allow: REFER

Allow: OPTIONS

Allow: UPDATE

Allow: PRACK

Allow: SUBSCRIBE

Allow: INFO

Max-forwards: 69

allow-events: talk

allow-events: hold

allow-events: conference

allow-events: LocalModeStatus

Supported: gruu

User-agent: Aastra 55i/2.4.1.37

Expires: 600

Content-Length: 0

<------------->

[Kappsrv1*CLI>

--- (29 headers 0 lines) ---

Using latest REGISTER request as basis request

Sending to 66.178.161.82 : 5060 (no NAT)

<--- Transmitting (no NAT) to 66.178.161.82:5060 --->

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 66.178.161.82:5060;branch=z9hG4bKb8f06a83c536416c402e03c97c1ce8cc;received=66.178.161.82

Via: SIP/2.0/UDP 10.60.3.39:5060;branch=z9hG4bK4104539c36c33cea4.991d760fdb2f977ea

From: ;tag=11dc51ca39

To:

Call-ID: 66e9f8ede08c82b8

CSeq: 32467 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Suppor ted: replaces

Contact:

Content-Length: 0

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Submitted by eeman on Fri, 03/06/2009 Permalink

do you have a 10 pattern set up for outbound routes for this tenant? _NXXNXXXXXX

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Submitted by ryan.tuttle on Wed, 03/11/2009 Permalink

That worked, thank you for your help.

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