Asterisk Hangs up on Exchange UM Auto Attendant

system · November 27, 2010, 6:55am

Hello,
I have asterisk 1.8 installed with PBX In a Flash as well as a Microsoft Exchange Server 2010 setup for Unified Messaging. all peices are working but when i try to use the Unified Messaging Auto attendant to reach an extension within the Asterisk pbx, the call is transfered, then asterisk hangs up… not sure why. Below is a snipet of the sip debug:

[code]<— SIP read from TCP:10.0.0.5:5065 —>
REFER sip:6000@10.0.0.4:5060;transport=TCP SIP/2.0
FROM: sip:8888@10.0.0.5:5065;epid=88A5C36BD6;tag=334b1afb11
TO: sip:6000@10.0.0.4;tag=as4330bc9b
CSEQ: 1 REFER
CALL-ID: 3871ca460dfb5d3855219e4a2bda4db7@10.0.0.4:5060
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 10.0.0.5:5065;branch=z9hG4bK6a493bb0
CONTACT: sip:natasmx.natasnet.com:5065;transport=Tcp;maddr=10.0.0.5;ms-opaque=1842e78949a60f19;automata
CONTENT-LENGTH: 0
REFER-TO: sip:6000@10.0.0.4:5060;transport=TCP;user=phone
REFERRED-BY: sip:8888@10.0.0.5:5065
USER-AGENT: RTCC/3.1.0.0

<------------->
— (12 headers 0 lines) —
Call 3871ca460dfb5d3855219e4a2bda4db7@10.0.0.4:5060 got a SIP call transfer from callee: (REFER)!
SIP transfer to extension 6000@from-internal-xfer by 8888@10.0.0.5:5065

<— Transmitting (no NAT) to 10.0.0.5:5065 —>
SIP/2.0 202 Accepted
Via: SIP/2.0/TCP 10.0.0.5:5065;branch=z9hG4bK6a493bb0;received=10.0.0.5
From: sip:8888@10.0.0.5:5065;epid=88A5C36BD6;tag=334b1afb11
To: sip:6000@10.0.0.4;tag=as4330bc9b
Call-ID: 3871ca460dfb5d3855219e4a2bda4db7@10.0.0.4:5060
CSeq: 1 REFER
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:6000@10.0.0.4:5060;transport=TCP
Content-Length: 0

<------------>
set_destination: Parsing sip:natasmx.natasnet.com:5065;transport=Tcp;maddr=10.0.0.5 for address/port to send to
set_destination: set destination to 10.0.0.5:5065
Reliably Transmitting (no NAT) to 10.0.0.5:5065:
NOTIFY sip:natasmx.natasnet.com:5065;transport=Tcp;maddr=10.0.0.5 SIP/2.0
Via: SIP/2.0/TCP 10.0.0.4:5060;branch=z9hG4bK7d31d972
Max-Forwards: 70
From: “Steven Whitehead” sip:6000@10.0.0.4;tag=as4330bc9b
To: sip:8888@10.0.0.5:5065;tag=334b1afb11
Contact: sip:6000@10.0.0.4:5060;transport=TCP
Call-ID: 3871ca460dfb5d3855219e4a2bda4db7@10.0.0.4:5060
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 1.8.0
Event: refer;id=1
Subscription-state: active
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 21

SIP/2.0 183 Ringing

set_destination: Parsing sip:natasmx.natasnet.com:5065;transport=Tcp;maddr=10.0.0.5 for address/port to send to
set_destination: set destination to 10.0.0.5:5065
Reliably Transmitting (no NAT) to 10.0.0.5:5065:
NOTIFY sip:natasmx.natasnet.com:5065;transport=Tcp;maddr=10.0.0.5 SIP/2.0
Via: SIP/2.0/TCP 10.0.0.4:5060;branch=z9hG4bK230797dd
Max-Forwards: 70
From: “Steven Whitehead” sip:6000@10.0.0.4;tag=as4330bc9b
To: sip:8888@10.0.0.5:5065;tag=334b1afb11
Contact: sip:6000@10.0.0.4:5060;transport=TCP
Call-ID: 3871ca460dfb5d3855219e4a2bda4db7@10.0.0.4:5060
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX 1.8.0
Event: refer;id=1
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 16

SIP/2.0 200 Ok

-- Executing [h@from-internal-xfer:1] Macro("SIP/6000-00000000", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/6000-00000000", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/6000-00000000", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)

– Executing [s@macro-hangupcall:7] GotoIf(“SIP/6000-00000000”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] Hangup(“SIP/6000-00000000”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/6000-00000000’ in macro 'hangupcall’
Scheduling destruction of SIP dialog ‘3871ca460dfb5d3855219e4a2bda4db7@10.0.0.4:5060’ in 32000 ms (Method: REFER)
== Spawn extension (from-internal-xfer, 6000, 1) exited non-zero on ‘SIP/6000-00000000’ in macro ‘dialout-trunk’
== Spawn extension (from-internal-xfer, 6000, 1) exited non-zero on 'SIP/6000-00000000’
Scheduling destruction of SIP dialog ‘NzU0YTE2MDc2MmFiYjgyYjY3NmU3NmQwMWEwZGJkY2Y.’ in 6400 ms (Method: ACK)
set_destination: Parsing sip:6000@10.0.0.15:61716 for address/port to send to
set_destination: set destination to 10.0.0.15:61716
Reliably Transmitting (NAT) to 10.0.0.15:61716:
BYE sip:6000@10.0.0.15:61716 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK221b4413;rport
Max-Forwards: 70
From: “8888"sip:8888@nataspbx.natasnet.com;tag=as41b9a9cd
To: “Steven Whitehead"sip:6000@nataspbx.natasnet.com;tag=4e14f2e8
Call-ID: NzU0YTE2MDc2MmFiYjgyYjY3NmU3NmQwMWEwZGJkY2Y.
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.0
Proxy-Authorization: Digest username=“6000”, realm=“asterisk”, algorithm=MD5, uri=“nataspbx.natasnet.com”, nonce=””, response="689b47080249fd870a869a1d998c79a1"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<— SIP read from UDP:10.0.0.15:61716 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK221b4413;rport=5060
Contact: sip:6000@10.0.0.15:61716
To: "Steven Whitehead"sip:6000@nataspbx.natasnet.com;tag=4e14f2e8
From: "8888"sip:8888@nataspbx.natasnet.com;tag=as41b9a9cd
Call-ID: NzU0YTE2MDc2MmFiYjgyYjY3NmU3NmQwMWEwZGJkY2Y.
CSeq: 102 BYE
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0

<------------->
— (9 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘NzU0YTE2MDc2MmFiYjgyYjY3NmU3NmQwMWEwZGJkY2Y.’ Method: ACK

<— SIP read from TCP:10.0.0.5:5065 —>
SIP/2.0 200 OK
FROM: "Steven Whitehead"sip:6000@10.0.0.4;tag=as4330bc9b
TO: sip:8888@10.0.0.5:5065;tag=334b1afb11;epid=88A5C36BD6
CSEQ: 103 NOTIFY
CALL-ID: 3871ca460dfb5d3855219e4a2bda4db7@10.0.0.4:5060
VIA: SIP/2.0/TCP 10.0.0.4:5060;branch=z9hG4bK7d31d972
CONTENT-LENGTH: 0
SERVER: RTCC/3.1.0.0

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

<— SIP read from TCP:10.0.0.5:5065 —>
SIP/2.0 200 OK
FROM: "Steven Whitehead"sip:6000@10.0.0.4;tag=as4330bc9b
TO: sip:8888@10.0.0.5:5065;tag=334b1afb11;epid=88A5C36BD6
CSEQ: 104 NOTIFY
CALL-ID: 3871ca460dfb5d3855219e4a2bda4db7@10.0.0.4:5060
VIA: SIP/2.0/TCP 10.0.0.4:5060;branch=z9hG4bK230797dd
CONTENT-LENGTH: 0
SERVER: RTCC/3.1.0.0

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

<— SIP read from TCP:10.0.0.5:5065 —>
BYE sip:6000@10.0.0.4:5060;transport=TCP SIP/2.0
FROM: sip:8888@10.0.0.5:5065;epid=88A5C36BD6;tag=334b1afb11
TO: sip:6000@10.0.0.4;tag=as4330bc9b
CSEQ: 2 BYE
CALL-ID: 3871ca460dfb5d3855219e4a2bda4db7@10.0.0.4:5060
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 10.0.0.5:5065;branch=z9hG4bK2380cede
CONTENT-LENGTH: 0
USER-AGENT: RTCC/3.1.0.0

<------------->
— (9 headers 0 lines) —
Sending to 10.0.0.5:5065 (no NAT)
Scheduling destruction of SIP dialog ‘3871ca460dfb5d3855219e4a2bda4db7@10.0.0.4:5060’ in 32000 ms (Method: BYE)

<— Transmitting (no NAT) to 10.0.0.5:5065 —>
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.0.0.5:5065;branch=z9hG4bK2380cede;received=10.0.0.5
From: sip:8888@10.0.0.5:5065;epid=88A5C36BD6;tag=334b1afb11
To: sip:6000@10.0.0.4;tag=as4330bc9b
Call-ID: 3871ca460dfb5d3855219e4a2bda4db7@10.0.0.4:5060
CSeq: 2 BYE
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
[/code]

Any Ideas?

penguinpbx · July 6, 2011, 8:18pm

Try setting the __TRANSFER_CONTEXT variable to the context where your phones can be reached before dialing out to Exchange. Otherwise, it comes back to the “default” context, which is usually pretty well blocked off to prevent toll fraud.