Shoretel SIP Transfer

tknman0700

Joined
Oct 30, 2009
Messages
49
Likes
0
Points
0
#1
I have been working with Asterisk in the form of the Digium AA50 appliance for about a year now.. I am new to FreePBX so I am trying to learn what is different in this way of administration. Here is a scenario I have.

I have a sip trunk between Elastix and the Shoretel. I can call from the Elastix to the Shoretel no problem but when I reach a menu in the shoretel and the shoretel tries to do a blind transfer I think they do it differently than one might wish... when the calls comes to my shoretel phone the Name of the called reads "exit" on my Shoretel phone. This is sort of frustrating and I am trying to determine why this is.

Here is a sip debug of what happens...

Code:
    -- Executing [8001@from-internal:1] Macro("SIP/1000-091104b0", "user-callerid|SKIPTTL|") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/1000-091104b0", "AMPUSER=1000") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1000-091104b0", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1000-091104b0", "1|Set|REALCALLERIDNUM=1000") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/1000-091104b0", "AMPUSER=1000") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/1000-091104b0", "AMPUSERCIDNAME=1000") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1000-091104b0", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/1000-091104b0", "AMPUSERCID=1000") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/1000-091104b0", "CALLERID(all)="1000" <1000>") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/1000-091104b0", "REALCALLERIDNUM=1000") in new stack
    -- Executing [s@macro-user-callerid:10] ExecIf("SIP/1000-091104b0", "0|Set|CHANNEL(language)=") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/1000-091104b0", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,20)
    -- Executing [s@macro-user-callerid:20] NoOp("SIP/1000-091104b0", "Using CallerID "1000" <1000>") in new stack
    -- Executing [8001@from-internal:2] Set("SIP/1000-091104b0", "_NODEST=") in new stack
    -- Executing [8001@from-internal:3] Macro("SIP/1000-091104b0", "record-enable|1000|OUT|") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/1000-091104b0", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/1000-091104b0", "recordingcheck|20091103-001322|1257225202.28") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20091103-001322|1257225202.28: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/1000-091104b0", "") in new stack
    -- Executing [8001@from-internal:4] Macro("SIP/1000-091104b0", "dialout-trunk|2|8001||") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/1000-091104b0", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1000-091104b0", "0?sub-pincheck|s|1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1000-091104b0", "0?disabletrunk|1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/1000-091104b0", "DIAL_NUMBER=8001") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/1000-091104b0", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/1000-091104b0", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1000-091104b0", "0?nomax") in new stack
    -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/1000-091104b0", "0?chanfull") in new stack
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1000-091104b0", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/1000-091104b0", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/1000-091104b0", "outbound-callerid|2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/1000-091104b0", "0|SetCallerPres|") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/1000-091104b0", "0|Set|REALCALLERIDNUM=1000") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/1000-091104b0", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/1000-091104b0", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/1000-091104b0", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/1000-091104b0", "TRUNKOUTCID=exit") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/1000-091104b0", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/1000-091104b0", "1|Set|CALLERID(all)=exit") in new stack
    -- Executing [s@macro-outbound-callerid:13] GotoIf("SIP/1000-091104b0", "1?exit") in new stack
    -- Goto (macro-outbound-callerid,s,11)
    -- Executing [s@macro-outbound-callerid:11] MacroExit("SIP/1000-091104b0", "") in new stack
    -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/1000-091104b0", "0|AGI|fixlocalprefix") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/1000-091104b0", "OUTNUM=8001") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/1000-091104b0", "custom=SIP/test") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1000-091104b0", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/1000-091104b0", "dialout-trunk-predial-hook|") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1000-091104b0", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/1000-091104b0", "0?bypass|1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1000-091104b0", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/1000-091104b0", "SIP/test/8001|300|") in new stack
Audio is at 10.1.2.144 port 19818
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.1.4.3:5060:
INVITE sip:8001@10.1.4.3 SIP/2.0
Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK48074559;rport
From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
To: <sip:8001@10.1.4.3>
Contact: <sip:Unknown@10.1.2.144>
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 03 Nov 2009 05:13:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 312

v=0
o=root 3111 3111 IN IP4 10.1.2.144
s=session
c=IN IP4 10.1.2.144
t=0 0
m=audio 19818 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called test/8001
localhost*CLI>
<--- SIP read from 10.1.4.3:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK48074559;rport
WWW-Authenticate: Digest realm="ShoreTel",domain="sip:shoretel.com",nonce="ShoreTel:1781834944",stale=false,algorithm=md5,opaque="0"
From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
To: <sip:8001@10.1.4.3>;tag=hssUA_1781834944-111523928
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 102 INVITE
Supported: timer
User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
Contact: DefaultProfile <sip:DefaultProfile@10.1.4.3:5060>
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 10.1.4.3:5060:
ACK sip:8001@10.1.4.3 SIP/2.0
Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK48074559;rport
From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
To: <sip:8001@10.1.4.3>;tag=hssUA_1781834944-111523928
Contact: <sip:Unknown@10.1.2.144>
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Audio is at 10.1.2.144 port 19818
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.1.4.3:5060:
INVITE sip:8001@10.1.4.3 SIP/2.0
Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK2030f228;rport
From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
To: <sip:8001@10.1.4.3>
Contact: <sip:Unknown@10.1.2.144>
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="test", realm="ShoreTel", algorithm=MD5, uri="sip:shoretel.com", nonce="ShoreTel:1781834944", response="36115d8a62ccbe80e269b48e0c65428a", opaque="0"
Date: Tue, 03 Nov 2009 05:13:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 312

v=0
o=root 3111 3112 IN IP4 10.1.2.144
s=session
c=IN IP4 10.1.2.144
t=0 0
m=audio 19818 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
localhost*CLI>
<--- SIP read from 10.1.4.3:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK2030f228;rport
From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
To: <sip:8001@10.1.4.3>
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 103 INVITE
Supported: timer
User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
Contact: DefaultProfile <sip:DefaultProfile@10.1.4.3:5060>
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
localhost*CLI>
<--- SIP read from 10.1.4.3:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK2030f228;rport
From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
To: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 103 INVITE
Supported: timer
User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
Contact: DefaultProfile <sip:DefaultProfile@10.1.4.3:5060>
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
    -- SIP/test-0908be28 is ringing
localhost*CLI>
<--- SIP read from 10.1.4.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK2030f228;rport
From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
To: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 103 INVITE
Supported: timer
User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
Contact: "VM-AutoAttendant" <sip:1204@10.1.4.3:5060>
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,REFER,REGISTER,UPDATE
Content-Type: application/sdp
Content-Length: 174

v=0
o=ShoreTel 0 11 IN IP4 10.1.4.3
s=-
c=IN IP4 10.1.4.3
t=0 0
m=audio 10080 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000/1
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-15

<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 102
Peer audio RTP is at port 10.1.4.3:10080
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 102
Capabilities: us - 0x180e (gsm|ulaw|alaw|g726|g722), peer - audio=0x4 (ulaw)/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 10.1.4.3:10080
list_route: hop: <sip:1204@10.1.4.3:5060>
set_destination: Parsing <sip:1204@10.1.4.3:5060> for address/port to send to
set_destination: set destination to 10.1.4.3, port 5060
Transmitting (no NAT) to 10.1.4.3:5060:
ACK sip:1204@10.1.4.3:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK724e0f73;rport
From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
To: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
Contact: <sip:Unknown@10.1.2.144>
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/test-0908be28 answered SIP/1000-091104b0
    -- Packet2Packet bridging SIP/1000-091104b0 and SIP/test-0908be28
localhost*CLI>
<--- SIP read from 10.1.4.3:5060 --->
INVITE sip:Unknown@10.1.2.144 SIP/2.0
Via: SIP/2.0/UDP 10.1.4.3:5060;branch=z9hG4bK1787734944-113813917
Max-Forwards: 70
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,REFER,REGISTER,UPDATE
Supported: timer,replaces
From: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
To: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 2 INVITE
User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
Contact: "VM-AutoAttendant" <sip:1204@10.1.4.3:5060>
Content-Type: application/sdp
Content-Length: 186

v=0
o=ShoreTel 0 12 IN IP4 10.1.4.3
s=-
c=IN IP4 10.1.4.3
t=0 0
m=audio 10080 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000/1
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-15
a=inactive

<------------->
--- (13 headers 10 lines) ---
Sending to 10.1.4.3 : 5060 (no NAT)
Found RTP audio format 0
Found RTP audio format 102
Peer audio RTP is at port 10.1.4.3:10080
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 102
Capabilities: us - 0x180e (gsm|ulaw|alaw|g726|g722), peer - audio=0x4 (ulaw)/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 10.1.4.3:10080
localhost*CLI>
<--- Transmitting (no NAT) to 10.1.4.3:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.4.3:5060;branch=z9hG4bK1787734944-113813917;received=10.1.4.3
From: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
To: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:Unknown@10.1.2.144>
Content-Length: 0


<------------>
Audio is at 10.1.2.144 port 19818
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
localhost*CLI>
<--- Reliably Transmitting (no NAT) to 10.1.4.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.4.3:5060;branch=z9hG4bK1787734944-113813917;received=10.1.4.3
From: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
To: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:Unknown@10.1.2.144>
Content-Type: application/sdp
Content-Length: 234

v=0
o=root 3111 3113 IN IP4 10.1.2.144
s=session
c=IN IP4 10.1.2.144
t=0 0
m=audio 19818 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=inactive

<------------>
    -- Started music on hold, class 'default', on SIP/1000-091104b0
localhost*CLI>
<--- SIP read from 10.1.4.3:5060 --->
ACK sip:Unknown@10.1.2.144 SIP/2.0
Via: SIP/2.0/UDP 10.1.4.3:5060;branch=z9hG4bK1787754944-111523936
Max-Forwards: 70
From: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
To: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 2 ACK
User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
Contact: "VM-AutoAttendant" <sip:1204@10.1.4.3:5060>
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
localhost*CLI>
<--- SIP read from 10.1.4.3:5060 --->
REFER sip:Unknown@10.1.2.144 SIP/2.0
Via: SIP/2.0/UDP 10.1.4.3:5060;branch=z9hG4bK1787756944-111523938
Max-Forwards: 70
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,REFER,REGISTER,UPDATE
Supported: replaces
From: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
To: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 3 REFER
User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
Contact: "VM-AutoAttendant" <sip:1204@10.1.4.3:5060>
Refer-To: "1172" <sip:1172@10.1.4.3:5060>
Referred-By: "VM-AutoAttendant" <sip:1204@10.1.4.3:5060>
Content-Length: 0


<------------->
--- (14 headers 0 lines) ---
Call 77015734440f26db754a299a0a518727@10.1.2.144 got a SIP call transfer from caller: (REFER)!
SIP transfer to extension 1172@from-internal-xfer by 1204@10.1.4.3:5060
localhost*CLI>
<--- Transmitting (no NAT) to 10.1.4.3:5060 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.1.4.3:5060;branch=z9hG4bK1787756944-111523938;received=10.1.4.3
From: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
To: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 3 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:Unknown@10.1.2.144>
Content-Length: 0


<------------>
set_destination: Parsing <sip:1204@10.1.4.3:5060> for address/port to send to
set_destination: set destination to 10.1.4.3, port 5060
Reliably Transmitting (no NAT) to 10.1.4.3:5060:
NOTIFY sip:1204@10.1.4.3:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK2c8d9608;rport
From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
To: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
Contact: <sip:Unknown@10.1.2.144>
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: refer;id=3
Subscription-state: active
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 21

SIP/2.0 183 Ringing

---
set_destination: Parsing <sip:1204@10.1.4.3:5060> for address/port to send to
set_destination: set destination to 10.1.4.3, port 5060
Reliably Transmitting (no NAT) to 10.1.4.3:5060:
NOTIFY sip:1204@10.1.4.3:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK39284fae;rport
From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
To: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
Contact: <sip:Unknown@10.1.2.144>
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: refer;id=3
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 16

SIP/2.0 200 Ok

---
    -- Stopped music on hold on SIP/1000-091104b0
    -- Executing [h@from-internal-xfer:1] Macro("SIP/1000-091104b0", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/1000-091104b0", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("SIP/1000-091104b0", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("SIP/1000-091104b0", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("SIP/1000-091104b0", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("SIP/1000-091104b0", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("SIP/1000-091104b0", "") in new stack
  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1000-091104b0' in macro 'hangupcall'
  == Spawn h extension (from-internal-xfer, s, 1) exited non-zero on 'SIP/1000-091104b0'
Scheduling destruction of SIP dialog '77015734440f26db754a299a0a518727@10.1.2.144' in 32000 ms (Method: REFER)
  == Spawn extension (from-internal-xfer, 1172, 0) exited non-zero on 'SIP/1000-091104b0' in macro 'dialout-trunk'
  == Spawn extension (from-internal-xfer, 1172, 0) exited non-zero on 'SIP/1000-091104b0'
    -- Executing [1172@from-internal-xfer:1] Macro("SIP/1000-091104b0", "user-callerid|SKIPTTL|") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/1000-091104b0", "AMPUSER=1000") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1000-091104b0", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1000-091104b0", "0|Set|REALCALLERIDNUM=") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/1000-091104b0", "AMPUSER=1000") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/1000-091104b0", "AMPUSERCIDNAME=1000") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1000-091104b0", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/1000-091104b0", "AMPUSERCID=1000") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/1000-091104b0", "CALLERID(all)="1000" <1000>") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/1000-091104b0", "REALCALLERIDNUM=1000") in new stack
    -- Executing [s@macro-user-callerid:10] ExecIf("SIP/1000-091104b0", "0|Set|CHANNEL(language)=") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/1000-091104b0", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,20)
    -- Executing [s@macro-user-callerid:20] NoOp("SIP/1000-091104b0", "Using CallerID "1000" <1000>") in new stack
    -- Executing [1172@from-internal-xfer:2] Set("SIP/1000-091104b0", "_NODEST=") in new stack
localhost*CLI>
<--- SIP read from 10.1.4.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK2c8d9608;rport
From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
To: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 104 NOTIFY
User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
Contact: "VM-AutoAttendant" <sip:1204@10.1.4.3:5060>
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,REFER,REGISTER,UPDATE
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
    -- Executing [1172@from-internal-xfer:3] Macro("SIP/1000-091104b0", "record-enable|1000|OUT|") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/1000-091104b0", "0?check") in new stack
    -- Executing [s@macro-record-enable:2] ResetCDR("SIP/1000-091104b0", "w") in new stack
    -- Executing [s@macro-record-enable:3] StopMonitor("SIP/1000-091104b0", "") in new stack
    -- Executing [s@macro-record-enable:4] AGI("SIP/1000-091104b0", "recordingcheck|20091103-001328|1257225202.28") in new stack
localhost*CLI>
<--- SIP read from 10.1.4.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK39284fae;rport
From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
To: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 105 NOTIFY
User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
Contact: "VM-AutoAttendant" <sip:1204@10.1.4.3:5060>
Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,REFER,REGISTER,UPDATE
Content-Length: 0


<------------->
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
localhost*CLI>
<--- SIP read from 10.1.4.3:5060 --->
BYE sip:Unknown@10.1.2.144 SIP/2.0
Via: SIP/2.0/UDP 10.1.4.3:5060;branch=z9hG4bK1787794944-113813923
Max-Forwards: 70
From: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
To: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 4 BYE
User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Sending to 10.1.4.3 : 5060 (no NAT)
Scheduling destruction of SIP dialog '77015734440f26db754a299a0a518727@10.1.2.144' in 32000 ms (Method: BYE)
localhost*CLI>
<--- Transmitting (no NAT) to 10.1.4.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.4.3:5060;branch=z9hG4bK1787794944-113813923;received=10.1.4.3
From: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
To: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
CSeq: 4 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
  recordingcheck|20091103-001328|1257225202.28: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/1000-091104b0", "") in new stack
    -- Executing [1172@from-internal-xfer:4] Macro("SIP/1000-091104b0", "dialout-trunk|2|1172||") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/1000-091104b0", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1000-091104b0", "0?sub-pincheck|s|1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1000-091104b0", "0?disabletrunk|1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/1000-091104b0", "DIAL_NUMBER=1172") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/1000-091104b0", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/1000-091104b0", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1000-091104b0", "0?nomax") in new stack
    -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/1000-091104b0", "0?chanfull") in new stack
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1000-091104b0", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/1000-091104b0", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/1000-091104b0", "outbound-callerid|2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/1000-091104b0", "0|SetCallerPres|") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/1000-091104b0", "0|Set|REALCALLERIDNUM=1000") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/1000-091104b0", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/1000-091104b0", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/1000-091104b0", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/1000-091104b0", "TRUNKOUTCID=exit") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/1000-091104b0", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/1000-091104b0", "1|Set|CALLERID(all)=exit") in new stack
    -- Executing [s@macro-outbound-callerid:13] GotoIf("SIP/1000-091104b0", "1?exit") in new stack
    -- Goto (macro-outbound-callerid,s,11)
    -- Executing [s@macro-outbound-callerid:11] MacroExit("SIP/1000-091104b0", "") in new stack
    -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/1000-091104b0", "0|AGI|fixlocalprefix") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/1000-091104b0", "OUTNUM=1172") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/1000-091104b0", "custom=SIP/test") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1000-091104b0", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/1000-091104b0", "dialout-trunk-predial-hook|") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1000-091104b0", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/1000-091104b0", "0?bypass|1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1000-091104b0", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/1000-091104b0", "SIP/test/1172|300|") in new stack
Audio is at 10.1.2.144 port 19304
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.1.4.3:5060:
INVITE sip:1172@10.1.4.3 SIP/2.0
Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK64a2ecfc;rport
From: "exit" <sip:Unknown@10.1.2.144>;tag=as0e5ae63a
To: <sip:1172@10.1.4.3>
Contact: <sip:Unknown@10.1.2.144>
Call-ID: 69441aee56c2f80310352f0f599a1d57@10.1.2.144
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 03 Nov 2009 05:13:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 312

v=0
o=root 3111 3111 IN IP4 10.1.2.144
s=session
c=IN IP4 10.1.2.144
t=0 0
m=audio 19304 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called test/1172
localhost*CLI>
<--- SIP read from 10.1.4.3:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK64a2ecfc;rport
WWW-Authenticate: Digest realm="ShoreTel",domain="sip:shoretel.com",nonce="ShoreTel:1787926944",stale=false,algorithm=md5,opaque="0"
From: "exit" <sip:Unknown@10.1.2.144>;tag=as0e5ae63a
To: <sip:1172@10.1.4.3>;tag=hssUA_1787926944-111523940
Call-ID: 69441aee56c2f80310352f0f599a1d57@10.1.2.144
CSeq: 102 INVITE
Supported: timer
User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
Contact: DefaultProfile <sip:DefaultProfile@10.1.4.3:5060>
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 10.1.4.3:5060:
ACK sip:1172@10.1.4.3 SIP/2.0
Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK64a2ecfc;rport
From: "exit" <sip:Unknown@10.1.2.144>;tag=as0e5ae63a
To: <sip:1172@10.1.4.3>;tag=hssUA_1787926944-111523940
Contact: <sip:Unknown@10.1.2.144>
Call-ID: 69441aee56c2f80310352f0f599a1d57@10.1.2.144
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Audio is at 10.1.2.144 port 19304
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.1.4.3:5060:
INVITE sip:1172@10.1.4.3 SIP/2.0
Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK116e6ec8;rport
From: "exit" <sip:Unknown@10.1.2.144>;tag=as0e5ae63a
To: <sip:1172@10.1.4.3>
Contact: <sip:Unknown@10.1.2.144>
Call-ID: 69441aee56c2f80310352f0f599a1d57@10.1.2.144
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="test", realm="ShoreTel", algorithm=MD5, uri="sip:shoretel.com", nonce="ShoreTel:1787926944", response="98e4f7115557d1ef106e255365c66d52", opaque="0"
Date: Tue, 03 Nov 2009 05:13:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 312

v=0
o=root 3111 3112 IN IP4 10.1.2.144
s=session
c=IN IP4 10.1.2.144
t=0 0
m=audio 19304 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
localhost*CLI>
<--- SIP read from 10.1.4.3:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK116e6ec8;rport
From: "exit" <sip:Unknown@10.1.2.144>;tag=as0e5ae63a
To: <sip:1172@10.1.4.3>
Call-ID: 69441aee56c2f80310352f0f599a1d57@10.1.2.144
CSeq: 103 INVITE
Supported: timer
User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
Contact: DefaultProfile <sip:DefaultProfile@10.1.4.3:5060>
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
localhost*CLI>
<--- SIP read from 10.1.4.3:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK116e6ec8;rport
From: "exit" <sip:Unknown@10.1.2.144>;tag=as0e5ae63a
To: <sip:1172@10.1.4.3>;tag=hssUA_1788012944-113813927
Call-ID: 69441aee56c2f80310352f0f599a1d57@10.1.2.144
CSeq: 103 INVITE
Supported: timer
User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
Contact: DefaultProfile <sip:DefaultProfile@10.1.4.3:5060>
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
    -- SIP/test-b7d0f8d0 is ringing
localhost*CLI> exit
[root@localhost ~]#
I dialed from the Elastix to the Shoretel via 8001. The call was bridged to the Shoretel Auto Attendant. When I dialed 1172 into that AA menu the call is bridged to my Shoretel phone but the callerid name is "exit". My understanding of this is because the Shoretel sends a referral in a way that confuses the Asterisk system into thinking it is an extension local to the machine. I setup a outbound route over the sip trunk that looks like this _1XXX to dial across the sip trunk and the call is bridged but I am really trying to understand why this "exit" is present as the caller id name.

Thanks
 

Members online

No members online now.

Latest posts

Forum statistics

Threads
30,915
Messages
130,920
Members
17,595
Latest member
feparra121
Top