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