Asterisk simple outbound Route Issue

Hello all,

Im running Elastix 2.2 and am stuck on being able to dial out. I have inbound routing working, with an IVR, and all internal calls from pstn and from extension to extension are working great. But I’m having an issue with not being able to call outbound at all. I have a Rhino 4fx0 card that will have 4 analog pstn lines plugged in. I dont want anything fancy for now as far as blocking or advanced routes, I just want to be able to press 9- then dial out to a local #.

Currently, I can dial 9, and it appears to establish the connection with a local # (my cell phone) but I get no audio. Im not sure if there is a dahdi_Channel issue regarding sound or if something with my SIP settings or outbound route is off?

My outbound route is:

9|NXXXXXX

My Trunk: ( * I figured once I got a local outbound route working, I could then build off of that)

1340+NXXXXXXX

Here is the output of my SIP Set debug- while making a call:

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2011.11.28 14:02:50 =~=~=~=~=~=~=~=~=~=~=~=

e[0KReally destroying SIP dialog ‘d8fb5dd74cfcad88’ Method: REGISTER

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.23.103:5060 —>
INVITE sip:9@192.168.23.108:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bK5d9ca4010b17174f0.5f7c30222ebd0e57c

Route: sip:192.168.23.108:5060;lr

Max-Forwards: 70

From: “CFM Office” sip:740@192.168.23.108:5060;tag=32a25aff3d

To: sip:9@192.168.23.108:5060;user=phone

Call-ID: d9892f31c0b91661

CSeq: 14528 INVITE

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

Allow-Events: talk, hold, conference, LocalModeStatus

Contact: “CFM Office” sip:740@192.168.23.103:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-1000-8000-00085D2BC146

Supported: path, 100rel, replaces

User-Agent: Aastra 6730i/3.2.2.56

Content-Type: application/sdp

Content-Length: 633

v=0

o=MxSIP 0 1 IN IP4 192.168.23.103

s=SIP Call

c=IN IP4 192.168.23.103

t=0 0

m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=rtpmap:106 BV16/8000

a=rtpmap:107 BV32/16000

a=rtpmap:113 L16/16000

a=rtpmap:110 PCMU/16000

a=rtpmap:111 PCMA/16000

a=rtpmap:112 L16/8000

a=rtpmap:98 G726-16/8000

a=rtpmap:97 G726-24/8000

a=rtpmap:115 G726-32/8000

a=rtpmap:96 G726-40/8000

a=rtpmap:9 G722/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=silenceSupp:on - - - -

a=fmtp:101 0-15

a=ptime:30

a=rtcp:3001 IN IP4 192.168.23.103

a=sendrecv

<------------->
— (15 headers 26 lines) —

e[Klocalhost*CLI>
e[0K == Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5

e[Klocalhost*CLI>
e[0KSending to 192.168.23.103 : 5060 (no NAT)

e[Klocalhost*CLI>
e[0KUsing INVITE request as basis request - d9892f31c0b91661
Found peer ‘740’ for ‘740’ from 192.168.23.103:5060

e[Klocalhost*CLI>
e[0K
<— Reliably Transmitting (NAT) to 192.168.23.103:5060 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bK5d9ca4010b17174f0.5f7c30222ebd0e57c;received=192.168.23.103

From: “CFM Office” sip:740@192.168.23.108:5060;tag=32a25aff3d

To: sip:9@192.168.23.108:5060;user=phone;tag=as133bfd87

Call-ID: d9892f31c0b91661

CSeq: 14528 INVITE

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“0e2e40f1”

Content-Length: 0

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

e[Klocalhost*CLI>
e[0KScheduling destruction of SIP dialog ‘d9892f31c0b91661’ in 6400 ms (Method: INVITE)

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.23.103:5060 —>
ACK sip:9@192.168.23.108:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bK5d9ca4010b17174f0.5f7c30222ebd0e57c

Route: sip:192.168.23.108:5060;lr

Max-Forwards: 70

From: “CFM Office” sip:740@192.168.23.108:5060;tag=32a25aff3d

To: sip:9@192.168.23.108:5060;user=phone;tag=as133bfd87

Call-ID: d9892f31c0b91661

CSeq: 14528 ACK

User-Agent: Aastra 6730i/3.2.2.56

Content-Length: 0

<------------->
— (10 headers 0 lines) —

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.23.103:5060 —>
INVITE sip:9@192.168.23.108:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bKf0f5693f170b2a68c.b586cb62e460b63a3

Route: sip:192.168.23.108:5060;lr

Max-Forwards: 70

From: “CFM Office” sip:740@192.168.23.108:5060;tag=32a25aff3d

To: sip:9@192.168.23.108:5060;user=phone

Call-ID: d9892f31c0b91661

CSeq: 14529 INVITE

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

Allow-Events: talk, hold, conference, LocalModeStatus

Authorization: Digest username=“740”,realm=“asterisk”,nonce=“0e2e40f1”,uri=“sip:9@192.168.23.108:5060;user=phone”,response=“2485c8b4a33983679457c4735fa49d9c”,algorithm=MD5

Contact: “CFM Office” sip:740@192.168.23.103:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-1000-8000-00085D2BC146

Supported: path, 100rel, replaces

User-Agent: Aastra 6730i/3.2.2.56

Content-Type: application/sdp

Content-Length: 633

v=0

o=MxSIP 0 1 IN IP4 192.168.23.103

s=SIP Call

c=IN IP4 192.168.23.103

t=0 0

m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=rtpmap:106 BV16/8000

a=rtpmap:107 BV32/16000

a=rtpmap:113 L16/16000

a=rtpmap:110 PCMU/16000

a=rtpmap:111 PCMA/16000

a=rtpmap:112 L16/8000

a=rtpmap:98 G726-16/8000

a=rtpmap:97 G726-24/8000

a=rtpmap:115 G726-32/8000

a=rtpmap:96 G726-40/8000

a=rtpmap:9 G722/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=silenceSupp:on - - - -

a=fmtp:101 0-15

a=ptime:30

a=rtcp:3001 IN IP4 192.168.23.103

a=sendrecv

<------------->
— (16 headers 26 lines) —
Sending to 192.168.23.103 : 5060 (NAT)
Using INVITE request as basis request - d9892f31c0b91661
Found peer ‘740’ for ‘740’ from 192.168.23.103:5060
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 106
Found RTP audio format 107
Found RTP audio format 113
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 112
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 115
Found RTP audio format 96
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format BV16 for ID 106
Found audio description format BV32 for ID 107
Found audio description format L16 for ID 113
Found audio description format PCMU for ID 110
Found audio description format PCMA for ID 111
Found audio description format L16 for ID 112
Found audio description format G726-16 for ID 98
Found audio description format G726-24 for ID 97
Found audio description format G726-32 for ID 115
Found audio description format G726-40 for ID 96
Found audio description format G722 for ID 9
Found audio description format PCMA for ID 8

e[Klocalhost*CLI>
e[0KFound audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8101f4c (ulaw|alaw|g726|slin|g729|speex|ilbc|g722|h263p|t140)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.23.103:3000
Looking for 9 in from-internal (domain 192.168.23.108)

<— Reliably Transmitting (NAT) to 192.168.23.103:5060 —>
SIP/2.0 484 Address Incomplete

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bKf0f5693f170b2a68c.b586cb62e460b63a3;received=192.168.23.103

From: “CFM Office” sip:740@192.168.23.108:5060;tag=32a25aff3d

To: sip:9@192.168.23.108:5060;user=phone;tag=as133bfd87

Call-ID: d9892f31c0b91661

CSeq: 14529 INVITE

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘d9892f31c0b91661’ in 6400 ms (Method: INVITE)

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.23.103:5060 —>
ACK sip:9@192.168.23.108:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bKf0f5693f170b2a68c.b586cb62e460b63a3

Route: sip:192.168.23.108:5060;lr

Max-Forwards: 70

From: “CFM Office” sip:740@192.168.23.108:5060;tag=32a25aff3d

To: sip:9@192.168.23.108:5060;user=phone;tag=as133bfd87

Call-ID: d9892f31c0b91661

CSeq: 14529 ACK

User-Agent: Aastra 6730i/3.2.2.56

Content-Length: 0

<------------->
— (10 headers 0 lines) —

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.23.103:5060 —>
REGISTER sip:192.168.23.108:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bKb47dcbdbc933bf71c

Route: sip:192.168.23.108:5060;lr

Max-Forwards: 70

From: “CFM Office” sip:740@192.168.23.108:5060;tag=80d6a1e214

To: “CFM Office” sip:740@192.168.23.108:5060

Call-ID: d8fb5dd74cfcad88

CSeq: 41628 REGISTER

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

Allow-Events: talk, hold, conference, LocalModeStatus

Authorization: Digest username=“740”,realm=“asterisk”,nonce=“0334539c”,uri=“sip:192.168.23.108:5060”,response=“b0db909f3c379dfcfaf4a743665bd22a”,algorithm=MD5

Contact: “CFM Office” sip:740@192.168.23.103:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-1000-8000-00085D2BC146”;expires=30

Supported: path, gruu

User-Agent: Aastra 6730i/3.2.2.56

Content-Length: 0

<------------->
— (15 headers 0 lines) —
Sending to 192.168.23.103 : 5060 (no NAT)

e[Klocalhost*CLI>
e[0K
<— Transmitting (NAT) to 192.168.23.103:5060 —>
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bKb47dcbdbc933bf71c;received=192.168.23.103

From: “CFM Office” sip:740@192.168.23.108:5060;tag=80d6a1e214

To: “CFM Office” sip:740@192.168.23.108:5060

Call-ID: d8fb5dd74cfcad88

CSeq: 41628 REGISTER

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

Content-Length: 0

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

e[Klocalhost*CLI>
e[0K
<— Transmitting (NAT) to 192.168.23.103:5060 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bKb47dcbdbc933bf71c;received=192.168.23.103

From: “CFM Office” sip:740@192.168.23.108:5060;tag=80d6a1e214

To: “CFM Office” sip:740@192.168.23.108:5060;tag=as65423cc3

Call-ID: d8fb5dd74cfcad88

CSeq: 41628 REGISTER

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“6a65f9bb”

Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘d8fb5dd74cfcad88’ in 32000 ms (Method: REGISTER)

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.23.103:5060 —>
REGISTER sip:192.168.23.108:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bK9c84a2ae2dc0ae29e

Route: sip:192.168.23.108:5060;lr

Max-Forwards: 70

From: “CFM Office” sip:740@192.168.23.108:5060;tag=80d6a1e214

To: “CFM Office” sip:740@192.168.23.108:5060

Call-ID: d8fb5dd74cfcad88

CSeq: 41629 REGISTER

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

Allow-Events: talk, hold, conference, LocalModeStatus

Authorization: Digest username=“740”,realm=“asterisk”,nonce=“6a65f9bb”,uri=“sip:192.168.23.108:5060”,response=“25ecc558de10f3187fff042411bc6688”,algorithm=MD5

Contact: “CFM Office” sip:740@192.168.23.103:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-1000-8000-00085D2BC146”;expires=30

Supported: path, gruu

User-Agent: Aastra 6730i/3.2.2.56

Content-Length: 0

<------------->
— (15 headers 0 lines) —
Sending to 192.168.23.103 : 5060 (NAT)

<— Transmitting (NAT) to 192.168.23.103:5060 —>
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bK9c84a2ae2dc0ae29e;received=192.168.23.103

From: “CFM Office” sip:740@192.168.23.108:5060;tag=80d6a1e214

To: “CFM Office” sip:740@192.168.23.108:5060

Call-ID: d8fb5dd74cfcad88

CSeq: 41629 REGISTER

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

Content-Length: 0

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

e[Klocalhost*CLI>
e[0KReliably Transmitting (NAT) to 192.168.23.103:5060:
OPTIONS sip:740@192.168.23.103:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.23.108:5060;branch=z9hG4bK2298baea;rport

Max-Forwards: 70

From: “Unknown” sip:Unknown@192.168.23.108;tag=as09cdc2d0

To: sip:740@192.168.23.103:5060;transport=udp

Contact: sip:Unknown@192.168.23.108

Call-ID: 68beeda10f18b15f10a5e6471e1d834f@192.168.23.108

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.6.2.13

Date: Mon, 28 Nov 2011 18:03:07 GMT

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

Supported: replaces, timer

Content-Length: 0


<— Transmitting (NAT) to 192.168.23.103:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bK9c84a2ae2dc0ae29e;received=192.168.23.103

From: “CFM Office” sip:740@192.168.23.108:5060;tag=80d6a1e214

To: “CFM Office” sip:740@192.168.23.108:5060;tag=as65423cc3

Call-ID: d8fb5dd74cfcad88

CSeq: 41629 REGISTER

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

Expires: 60

Contact: sip:740@192.168.23.103:5060;transport=udp;expires=60

Date: Mon, 28 Nov 2011 18:03:07 GMT

Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘d8fb5dd74cfcad88’ in 32000 ms (Method: REGISTER)

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.23.103:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.23.108:5060;branch=z9hG4bK2298baea;rport=5060;received=192.168.23.108

From: “Unknown” sip:Unknown@192.168.23.108;tag=as09cdc2d0

To: sip:740@192.168.23.103:5060;transport=udp;tag=3520069866

Call-ID: 68beeda10f18b15f10a5e6471e1d834f@192.168.23.108

CSeq: 102 OPTIONS

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

Server: Aastra 6730i/3.2.2.56

Supported: path

Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘68beeda10f18b15f10a5e6471e1d834f@192.168.23.108’ Method: OPTIONS

e[Klocalhost*CLI>
e[0KReally destroying SIP dialog ‘d9892f31c0b91661’ Method: ACK

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.23.103:5060 —>
INVITE sip:93446015@192.168.23.108:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bKf762261b85fb56758.17f21d947e38a4fbf

Route: sip:192.168.23.108:5060;lr

Max-Forwards: 70

From: “CFM Office” sip:740@192.168.23.108:5060;tag=952590a318

To: sip:93446015@192.168.23.108:5060;user=phone

Call-ID: dd74d34e56a464e0

CSeq: 19410 INVITE

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

Allow-Events: talk, hold, conference, LocalModeStatus

Contact: “CFM Office” sip:740@192.168.23.103:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-1000-8000-00085D2BC146

Supported: path, 100rel, replaces

User-Agent: Aastra 6730i/3.2.2.56

Content-Type: application/sdp

Content-Length: 633

v=0

o=MxSIP 0 1 IN IP4 192.168.23.103

s=SIP Call

c=IN IP4 192.168.23.103

t=0 0

m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=rtpmap:106 BV16/8000

a=rtpmap:107 BV32/16000

a=rtpmap:113 L16/16000

a=rtpmap:110 PCMU/16000

a=rtpmap:111 PCMA/16000

a=rtpmap:112 L16/8000

a=rtpmap:98 G726-16/8000

a=rtpmap:97 G726-24/8000

a=rtpmap:115 G726-32/8000

a=rtpmap:96 G726-40/8000

a=rtpmap:9 G722/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=silenceSupp:on - - - -

a=fmtp:101 0-15

a=ptime:30

a=rtcp:3001 IN IP4 192.168.23.103

a=sendrecv

<------------->
— (15 headers 26 lines) —
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 192.168.23.103 : 5060 (no NAT)
Using INVITE request as basis request - dd74d34e56a464e0
Found peer ‘740’ for ‘740’ from 192.168.23.103:5060

<— Reliably Transmitting (NAT) to 192.168.23.103:5060 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bKf762261b85fb56758.17f21d947e38a4fbf;received=192.168.23.103

From: “CFM Office” sip:740@192.168.23.108:5060;tag=952590a318

To: sip:93446015@192.168.23.108:5060;user=phone;tag=as62f0606f

Call-ID: dd74d34e56a464e0

CSeq: 19410 INVITE

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“0b5d7f3a”

Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘dd74d34e56a464e0’ in 6400 ms (Method: INVITE)

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.23.103:5060 —>
ACK sip:93446015@192.168.23.108:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bKf762261b85fb56758.17f21d947e38a4fbf

Route: sip:192.168.23.108:5060;lr

Max-Forwards: 70

From: “CFM Office” sip:740@192.168.23.108:5060;tag=952590a318

To: sip:93446015@192.168.23.108:5060;user=phone;tag=as62f0606f

Call-ID: dd74d34e56a464e0

CSeq: 19410 ACK

User-Agent: Aastra 6730i/3.2.2.56

Content-Length: 0

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

e[Klocalhost*CLI>
e[0K— (10 headers 0 lines) —

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.23.103:5060 —>
INVITE sip:93446015@192.168.23.108:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bK2deff9c9d95a6ec4c.aa5ea71901a205a79

Route: sip:192.168.23.108:5060;lr

Max-Forwards: 70

From: “CFM Office” sip:740@192.168.23.108:5060;tag=952590a318

To: sip:93446015@192.168.23.108:5060;user=phone

Call-ID: dd74d34e56a464e0

CSeq: 19411 INVITE

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

Allow-Events: talk, hold, conference, LocalModeStatus

Authorization: Digest username=“740”,realm=“asterisk”,nonce=“0b5d7f3a”,uri=“sip:93446015@192.168.23.108:5060;user=phone”,response=“f33849de2df3d105fa3574ee62673bf4”,algorithm=MD5

Contact: “CFM Office” sip:740@192.168.23.103:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-1000-8000-00085D2BC146

Supported: path, 100rel, replaces

User-Agent: Aastra 6730i/3.2.2.56

Content-Type: application/sdp

Content-Length: 633

v=0

o=MxSIP 0 1 IN IP4 192.168.23.103

s=SIP Call

c=IN IP4 192.168.23.103

t=0 0

m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=rtpmap:106 BV16/8000

a=rtpmap:107 BV32/16000

a=rtpmap:113 L16/16000

a=rtpmap:110 PCMU/16000

a=rtpmap:111 PCMA/16000

a=rtpmap:112 L16/8000

a=rtpmap:98 G726-16/8000

a=rtpmap:97 G726-24/8000

a=rtpmap:115 G726-32/8000

a=rtpmap:96 G726-40/8000

a=rtpmap:9 G722/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=silenceSupp:on - - - -

a=fmtp:101 0-15

a=ptime:30

a=rtcp:3001 IN IP4 192.168.23.103

a=sendrecv

<------------->
— (16 headers 26 lines) —
Sending to 192.168.23.103 : 5060 (NAT)
Using INVITE request as basis request - dd74d34e56a464e0
Found peer ‘740’ for ‘740’ from 192.168.23.103:5060
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 106
Found RTP audio format 107
Found RTP audio format 113
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 112

e[Klocalhost*CLI>
e[0KFound RTP audio format 98
Found RTP audio format 97
Found RTP audio format 115
Found RTP audio format 96
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format BV16 for ID 106
Found audio description format BV32 for ID 107
Found audio description format L16 for ID 113
Found audio description format PCMU for ID 110
Found audio description format PCMA for ID 111
Found audio description format L16 for ID 112
Found audio description format G726-16 for ID 98
Found audio description format G726-24 for ID 97
Found audio description format G726-32 for ID 115
Found audio description format G726-40 for ID 96
Found audio description format G722 for ID 9
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8101f4c (ulaw|alaw|g726|slin|g729|speex|ilbc|g722|h263p|t140)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.23.103:3000
Looking for 93446015 in from-internal (domain 192.168.23.108)

e[Klocalhost*CLI>
e[0Klist_route: hop: sip:740@192.168.23.103:5060;transport=udp

e[Klocalhost*CLI>
e[0K
<— Transmitting (NAT) to 192.168.23.103:5060 —>
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bK2deff9c9d95a6ec4c.aa5ea71901a205a79;received=192.168.23.103

From: “CFM Office” sip:740@192.168.23.108:5060;tag=952590a318

To: sip:93446015@192.168.23.108:5060;user=phone

Call-ID: dd74d34e56a464e0

CSeq: 19411 INVITE

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

Contact: sip:93446015@192.168.23.108

Content-Length: 0

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

e[Klocalhost*CLI>
e[0K – Executing [93446015@from-internal:1] e[1;36mMacroe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35muser-callerid,SKIPTTL,e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:1] e[1;36mSete[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mAMPUSER=740e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:2] e[1;36mGotoIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m0?reporte[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:3] e[1;36mExecIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m1?Set(REALCALLERIDNUM=740)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:4] e[1;36mSete[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mAMPUSER=740e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:5] e[1;36mSete[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mAMPUSERCIDNAME=CFM Officee[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:6] e[1;36mGotoIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m0?reporte[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:7] e[1;36mSete[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mAMPUSERCID=740e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:8] e[1;36mSete[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mCALLERID(all)=“CFM Office” <740>e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:9] e[1;36mExecIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m0?Set(CHANNEL(language)=)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:10] e[1;36mGotoIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m1?continuee[0m”) in new stack
– Goto (macro-user-callerid,s,19)

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:19] e[1;36mNoOpe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mUsing CallerID “CFM Office” <740>e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [93446015@from-internal:2] e[1;36mExecIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m1?Set(TRUNKCIDOVERRIDE=3407761158)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [93446015@from-internal:3] e[1;36mSete[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m_NODEST=e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [93446015@from-internal:4] e[1;36mMacroe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mrecord-enable,740,OUT,e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-record-enable:1] e[1;36mGotoIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m1?checke[0m”) in new stack
– Goto (macro-record-enable,s,4)

e[Klocalhost*CLI>
e[0K – Executing [s@macro-record-enable:4] e[1;36mExecIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m0?MacroExit()e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-record-enable:5] e[1;36mGotoIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m0?Group:OUTe[0m”) in new stack
– Goto (macro-record-enable,s,15)

e[Klocalhost*CLI>
e[0K – Executing [s@macro-record-enable:15] e[1;36mGotoIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m0?INe[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-record-enable:16] e[1;36mExecIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m1?MacroExit()e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [93446015@from-internal:5] e[1;36mMacroe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mdialout-trunk,1,3446015,e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:1] e[1;36mSete[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mDIAL_TRUNK=1e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:2] e[1;36mGosubIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m0?sub-pincheck,s,1e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:3] e[1;36mGotoIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m0?disabletrunk,1e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:4] e[1;36mSete[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mDIAL_NUMBER=3446015e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:5] e[1;36mSete[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mDIAL_TRUNK_OPTIONS=tre[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:6] e[1;36mSete[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mOUTBOUND_GROUP=OUT_1e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:7] e[1;36mGotoIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m0?nomaxe[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:8] e[1;36mGotoIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m0?chanfulle[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:9] e[1;36mGotoIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m0?skipoutcide[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:10] e[1;36mSete[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mDIAL_TRUNK_OPTIONS=e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:11] e[1;36mMacroe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35moutbound-callerid,1e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:1] e[1;36mExecIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m0?Set(CALLERPRES()=)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:2] e[1;36mExecIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m0?Set(REALCALLERIDNUM=740)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:3] e[1;36mGotoIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m1?normcide[0m”) in new stack
– Goto (macro-outbound-callerid,s,6)

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:6] e[1;36mSete[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mUSEROUTCID=e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:7] e[1;36mSete[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mEMERGENCYCID=e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:8] e[1;36mSete[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mTRUNKOUTCID=Testing Trunke[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:9] e[1;36mGotoIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m1?trunkcide[0m”) in new stack
– Goto (macro-outbound-callerid,s,12)

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:12] e[1;36mExecIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m1?Set(CALLERID(all)=Testing Trunk)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:13] e[1;36mExecIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m0?Set(CALLERID(all)=)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:14] e[1;36mExecIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m1?Set(CALLERID(all)=3407761158)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:15] e[1;36mExecIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m0?Set(CALLERPRES()=prohib_passed_screen)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:12] e[1;36mExecIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m1?AGI(fixlocalprefix)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix

e[Klocalhost*CLI>
e[0K – <SIP/740-00000012>AGI Script fixlocalprefix completed, returning 0

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:13] e[1;36mSete[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mOUTNUM=3446015e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:14] e[1;36mSete[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mcustom=DAHDI/1e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:15] e[1;36mExecIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:16] e[1;36mMacroe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mdialout-trunk-predial-hook,e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk-predial-hook:1] e[1;36mMacroExite[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35me[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:17] e[1;36mGotoIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m0?bypass,1e[0m”) in new stack
– Executing [s@macro-dialout-trunk:18] e[1;36mGotoIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m0?customtrunke[0m”) in new stack
– Executing [s@macro-dialout-trunk:19] e[1;36mDiale[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mDAHDI/1/3446015,300,e[0m”) in new stack
– Called 1/3446015

e[Klocalhost*CLI>
e[0K – DAHDI/1-1 answered SIP/740-00000012
Audio is at 192.168.23.108 port 11698
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

e[Klocalhost*CLI>
e[0K
<— Reliably Transmitting (NAT) to 192.168.23.103:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bK2deff9c9d95a6ec4c.aa5ea71901a205a79;received=192.168.23.103

From: “CFM Office” sip:740@192.168.23.108:5060;tag=952590a318

To: sip:93446015@192.168.23.108:5060;user=phone;tag=as4d743c8d

Call-ID: dd74d34e56a464e0

CSeq: 19411 INVITE

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

Contact: sip:93446015@192.168.23.108

Content-Type: application/sdp

Content-Length: 263

v=0

o=root 427321702 427321702 IN IP4 192.168.23.108

s=Asterisk PBX 1.6.2.13

c=IN IP4 192.168.23.108

t=0 0

m=audio 11698 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

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

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.23.103:5060 —>
ACK sip:93446015@192.168.23.108 SIP/2.0

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bK66b9f40b6b7c4f3d2

Max-Forwards: 70

From: “CFM Office” sip:740@192.168.23.108:5060;tag=952590a318

To: sip:93446015@192.168.23.108:5060;user=phone;tag=as4d743c8d

Call-ID: dd74d34e56a464e0

CSeq: 19411 ACK

Authorization: Digest username=“740”,realm=“asterisk”,nonce=“0b5d7f3a”,uri=“sip:93446015@192.168.23.108:5060;user=phone”,response=“f33849de2df3d105fa3574ee62673bf4”,algorithm=MD5

User-Agent: Aastra 6730i/3.2.2.56

Content-Length: 0

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

e[Klocalhost*CLI>
e[0K— (10 headers 0 lines) —

e[Klocalhost*CLI>
e[0KReally destroying SIP dialog ‘d8fb5dd74cfcad88’ Method: REGISTER

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.23.103:5060 —>
REGISTER sip:192.168.23.108:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bKc112d037fe1d8223a

Route: sip:192.168.23.108:5060;lr

Max-Forwards: 70

From: “CFM Office” sip:740@192.168.23.108:5060;tag=80d6a1e214

To: “CFM Office” sip:740@192.168.23.108:5060

Call-ID: d8fb5dd74cfcad88

CSeq: 41630 REGISTER

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

Allow-Events: talk, hold, conference, LocalModeStatus

Authorization: Digest username=“740”,realm=“asterisk”,nonce=“6a65f9bb”,uri=“sip:192.168.23.108:5060”,response=“25ecc558de10f3187fff042411bc6688”,algorithm=MD5

Contact: “CFM Office” sip:740@192.168.23.103:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-1000-8000-00085D2BC146”;expires=30

Supported: path, gruu

User-Agent: Aastra 6730i/3.2.2.56

Content-Length: 0

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

e[Klocalhost*CLI>
e[0K— (15 headers 0 lines) —
Sending to 192.168.23.103 : 5060 (no NAT)

e[Klocalhost*CLI>
e[0K
<— Transmitting (NAT) to 192.168.23.103:5060 —>
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bKc112d037fe1d8223a;received=192.168.23.103

From: “CFM Office” sip:740@192.168.23.108:5060;tag=80d6a1e214

To: “CFM Office” sip:740@192.168.23.108:5060

Call-ID: d8fb5dd74cfcad88

CSeq: 41630 REGISTER

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

Content-Length: 0

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

e[Klocalhost*CLI>
e[0K
<— Transmitting (NAT) to 192.168.23.103:5060 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bKc112d037fe1d8223a;received=192.168.23.103

From: “CFM Office” sip:740@192.168.23.108:5060;tag=80d6a1e214

To: “CFM Office” sip:740@192.168.23.108:5060;tag=as7403bb13

Call-ID: d8fb5dd74cfcad88

CSeq: 41630 REGISTER

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“597243e5”

Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘d8fb5dd74cfcad88’ in 32000 ms (Method: REGISTER)

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.23.103:5060 —>
REGISTER sip:192.168.23.108:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bKa745348711c4ab9d1

Route: sip:192.168.23.108:5060;lr

Max-Forwards: 70

From: “CFM Office” sip:740@192.168.23.108:5060;tag=80d6a1e214

To: “CFM Office” sip:740@192.168.23.108:5060

Call-ID: d8fb5dd74cfcad88

CSeq: 41631 REGISTER

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

Allow-Events: talk, hold, conference, LocalModeStatus

Authorization: Digest username=“740”,realm=“asterisk”,nonce=“597243e5”,uri=“sip:192.168.23.108:5060”,response=“5df5e26171f4994d85124a2cc922ad3f”,algorithm=MD5

Contact: “CFM Office” sip:740@192.168.23.103:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-1000-8000-00085D2BC146”;expires=30

Supported: path, gruu

User-Agent: Aastra 6730i/3.2.2.56

Content-Length: 0

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

e[Klocalhost*CLI>
e[0K— (15 headers 0 lines) —
Sending to 192.168.23.103 : 5060 (NAT)

<— Transmitting (NAT) to 192.168.23.103:5060 —>
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bKa745348711c4ab9d1;received=192.168.23.103

From: “CFM Office” sip:740@192.168.23.108:5060;tag=80d6a1e214

To: “CFM Office” sip:740@192.168.23.108:5060

Call-ID: d8fb5dd74cfcad88

CSeq: 41631 REGISTER

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

Content-Length: 0

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

e[Klocalhost*CLI>
e[0KReliably Transmitting (NAT) to 192.168.23.103:5060:
OPTIONS sip:740@192.168.23.103:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.23.108:5060;branch=z9hG4bK42fdc954;rport

Max-Forwards: 70

From: “Unknown” sip:Unknown@192.168.23.108;tag=as49ab99b2

To: sip:740@192.168.23.103:5060;transport=udp

Contact: sip:Unknown@192.168.23.108

Call-ID: 26421c2a6943736a002e2ac04bedc927@192.168.23.108

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.6.2.13

Date: Mon, 28 Nov 2011 18:03:52 GMT

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

Supported: replaces, timer

Content-Length: 0


<— Transmitting (NAT) to 192.168.23.103:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bKa745348711c4ab9d1;received=192.168.23.103

From: “CFM Office” sip:740@192.168.23.108:5060;tag=80d6a1e214

To: “CFM Office” sip:740@192.168.23.108:5060;tag=as7403bb13

Call-ID: d8fb5dd74cfcad88

CSeq: 41631 REGISTER

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

Expires: 60

Contact: sip:740@192.168.23.103:5060;transport=udp;expires=60

Date: Mon, 28 Nov 2011 18:03:52 GMT

Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘d8fb5dd74cfcad88’ in 32000 ms (Method: REGISTER)

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.23.103:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.23.108:5060;branch=z9hG4bK42fdc954;rport=5060;received=192.168.23.108

From: “Unknown” sip:Unknown@192.168.23.108;tag=as49ab99b2

To: sip:740@192.168.23.103:5060;transport=udp;tag=1941888541

Call-ID: 26421c2a6943736a002e2ac04bedc927@192.168.23.108

CSeq: 102 OPTIONS

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

Server: Aastra 6730i/3.2.2.56

Supported: path

Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘26421c2a6943736a002e2ac04bedc927@192.168.23.108’ Method: OPTIONS

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.23.103:5060 —>
BYE sip:93446015@192.168.23.108 SIP/2.0

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bKd4c185aaa34267314

Max-Forwards: 70

From: “CFM Office” sip:740@192.168.23.108:5060;tag=952590a318

To: sip:93446015@192.168.23.108:5060;user=phone;tag=as4d743c8d

Call-ID: dd74d34e56a464e0

CSeq: 19412 BYE

Authorization: Digest username=“740”,realm=“asterisk”,nonce=“0b5d7f3a”,uri="sip:93446015@192.168.23.108",response=“d87953f5fb23f4528bfc6a562810467a”,algorithm=MD5

User-Agent: Aastra 6730i/3.2.2.56

Content-Length: 0

<------------->
— (10 headers 0 lines) —
Sending to 192.168.23.103 : 5060 (NAT)

e[Klocalhost*CLI>
e[0K
<— Transmitting (NAT) to 192.168.23.103:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.23.103;branch=z9hG4bKd4c185aaa34267314;received=192.168.23.103

From: “CFM Office” sip:740@192.168.23.108:5060;tag=952590a318

To: sip:93446015@192.168.23.108:5060;user=phone;tag=as4d743c8d

Call-ID: dd74d34e56a464e0

CSeq: 19412 BYE

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

Content-Length: 0

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

e[Klocalhost*CLI>
e[0K – Executing [h@macro-dialout-trunk:1] e[1;36mMacroe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mhangupcall,e[0m”) in new stack
– Executing [s@macro-hangupcall:1] e[1;36mGotoIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m1?noautomone[0m”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] e[1;36mNoOpe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35mTOUCH_MONITOR_OUTPUT=e[0m”) in new stack
– Executing [s@macro-hangupcall:4] e[1;36mGotoIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m1?skiprge[0m”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [s@macro-hangupcall:7] e[1;36mGotoIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m1?skipblkvme[0m”) in new stack
– Goto (macro-hangupcall,s,10)
– Executing [s@macro-hangupcall:10] e[1;36mGotoIfe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35m1?theende[0m”) in new stack
– Goto (macro-hangupcall,s,12)
– Executing [s@macro-hangupcall:12] e[1;36mHangupe[0m(“e[1;35mSIP/740-00000012e[0m”, “e[1;35me[0m”) in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on ‘SIP/740-00000012’ in macro ‘hangupcall’

e[Klocalhost*CLI>
e[0K – Hungup ‘DAHDI/1-1’
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on ‘SIP/740-00000012’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 93446015, 5) exited non-zero on ‘SIP/740-00000012’

e[Klocalhost*CLI>
e[0KReally destroying SIP dialog ‘dd74d34e56a464e0’ Method: BYE