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Hosting SIP trunks with MTE

Posted by ryan.tuttle on Thu, 03/05/2009
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Thirdlane platform General Questions

I'm trying to setup a SIP trunk to an Adtran 908 for PRI delivery to a PBX. The trunk comes up and I can call extensions within the context it is provisioned but it will not use the default trunk out of the context if the number cannot be found. The trunk registers and I can route inbound calls to it with no problem. Any help would be greatly appreciated.

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

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.

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?

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