ASTERISK-17044: Call torn down upon connection when early media 183 used

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Summary: ASTERISK-17044: Call torn down upon connection when early media 183 used

Reporter: Erik Smith (eeman) Labels:

Date Opened: 2010-11-29 15:27:58.000-0600 Date Closed: 2014-06-03 09:32:15

Priority: Major Regression? No

Status: Closed/Complete Components: Channels/chan_sip/General

Versions: Frequency of
Occurrence

Related
Issues:

Environment: Attachments: ( 0) GateWayDebugLog
( 1) gw-to-carrier-8006262001-1.pcap
( 2) gw-to-carrier-8006262001-2.pcap
( 3) issue_18399_full_log.txt
( 4) me-to-18006262001.pcap
( 5) phone-endpoint-18006262001.pcap

Description: Asterisk 1.8.1-rc1 & Asterisk 1.6.2.14
Centos 5.5

have scenario as such

Asterisk-1.8.1-rc1 -SIP-> Asterisk 1.6.2.14 -SIP-> Broadvox (Sonus Softswitch)

When calling a TF number that uses early media for their IVR (example 1-800-626-2001); once the call gets connected and the 200 OK message is received, my 1.8.1-rc1 box issues a BYE message with a HangupCauseCode of 0. I can reproduce this with several numbers that are using early media for their IVR's. Just as soon as my call gets connected to a call-center's ACD Queue I hear 1-2 seconds of the recording before the call is torn down. I have tested this using a linksys SPA-2102 ATA, A Polycom IP501, as well as a Digium FXS module and get identical results.

****** ADDITIONAL INFORMATION ******

-- Called +18006262001@GW01EEMAN
-- SIP/GW01EEMAN-00000039 is making progress passing it to SIP/202-00000038
[Nov 29 16:24:26] WARNING[25528]: chan_sip.c:12909 __set_address_from_contact: Invalid contact uri (missing sip: or sips:), attempting to use anyway
[Nov 29 16:24:26] ERROR[25528]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("Ã ", "(null)", ...): Name or service not known
[Nov 29 16:24:26] WARNING[25528]: chan_sip.c:12920 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'Ã '
-- SIP/GW01EEMAN-00000039 answered SIP/202-00000038
== Spawn extension (macro-tl-dialout-base, dial-SIP, 7) exited non-zero on 'SIP/202-00000038' in macro 'tl-dialout-base'
== Spawn extension (macro-tl-dialout-1-trunk, s, 3) exited non-zero on 'SIP/202-00000038' in macro 'tl-dialout-1-trunk'
== Spawn extension (from-inside-redir, 18006262001, 1) exited non-zero on 'SIP/202-00000038'
-- Executing [h@from-inside-redir:1] Hangup("SIP/202-00000038", "") in new stack
== Spawn extension (from-inside-redir, h, 1) exited non-zero on 'SIP/202-00000038'

Comments: By: Erik Smith (eeman) 2010-11-29 15:40:10.000-0600

example when using sip set debug -

-- Executing [dial-SIP@macro-tl-dialout-base:7] Dial("SIP/202-0000003c", "SIP/+18006262001@GW01EEMAN,60,") in new stack
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 69.64.11.6:5060:
INVITE sip:+18006262001@69.64.11.6 SIP/2.0
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK11cb4eae
Max-Forwards: 70
From: "Erik Smith" <sip:5023152516@216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001@69.64.11.6>
Contact: <sip:5023152516@216.135.89.226:5060>
Call-ID: 0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.1-rc1
Date: Mon, 29 Nov 2010 21:33:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Erik Smith" <sip:5023152516@216.135.89.226>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 184

v=0
o=root 276500518 276500518 IN IP4 216.135.89.226
s=Asterisk PBX 1.8.1-rc1
c=IN IP4 216.135.89.226
t=0 0
m=audio 12180 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

---
-- Called +18006262001@GW01EEMAN

<--- SIP read from UDP:69.64.11.6:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK11cb4eae;received=216.135.89.226
From: "Erik Smith" <sip:5023152516@216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001@69.64.11.6>;tag=as4e48411b
Call-ID: 0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="669350ed"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 69.64.11.6:5060:
ACK sip:+18006262001@69.64.11.6 SIP/2.0
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK11cb4eae
Max-Forwards: 70
From: "Erik Smith" <sip:5023152516@216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001@69.64.11.6>;tag=as4e48411b
Contact: <sip:5023152516@216.135.89.226:5060>
Call-ID: 0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.1-rc1
Content-Length: 0

---
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 69.64.11.6:5060:
INVITE sip:+18006262001@69.64.11.6 SIP/2.0
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK02aac998
Max-Forwards: 70
From: "Erik Smith" <sip:5023152516@216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001@69.64.11.6>
Contact: <sip:5023152516@216.135.89.226:5060>
Call-ID: 0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.1-rc1
Authorization: Digest username="GW01EEMAN", realm="asterisk", algorithm=MD5, uri="sip:+18006262001@69.64.11.6", nonce="669350ed", response="8c8669c1bc9665d5ebeaed71ad6618a1"
Date: Mon, 29 Nov 2010 21:33:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Erik Smith" <sip:5023152516@216.135.89.226>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 184

v=0
o=root 276500518 276500519 IN IP4 216.135.89.226
s=Asterisk PBX 1.8.1-rc1
c=IN IP4 216.135.89.226
t=0 0
m=audio 12180 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:69.64.11.6:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK02aac998;received=216.135.89.226
From: "Erik Smith" <sip:5023152516@216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001@69.64.11.6>
Call-ID: 0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060
CSeq: 103 INVITE
Server: Asterisk PBX 1.6.2.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:+18006262001@69.64.11.6>
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:69.64.11.6:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK02aac998;received=216.135.89.226
From: "Erik Smith" <sip:5023152516@216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001@69.64.11.6>;tag=as35c589d8
Call-ID: 0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060
CSeq: 103 INVITE
Server: Asterisk PBX 1.6.2.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:+18006262001@69.64.11.6>
Content-Type: application/sdp
Content-Length: 179

v=0
o=root 629951581 629951581 IN IP4 69.64.11.6
s=Asterisk PBX 1.6.2.14-rc1
c=IN IP4 69.64.11.6
t=0 0
m=audio 13978 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 69.64.11.6:13978
-- SIP/GW01EEMAN-0000003d is making progress passing it to SIP/202-0000003c
Reliably Transmitting (no NAT) to 69.64.11.6:5060:
INFO sip:+18006262001@69.64.11.6 SIP/2.0
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK1f19538a
Max-Forwards: 70
From: "Erik Smith" <sip:5023152516@216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001@69.64.11.6>;tag=as35c589d8
Contact: <sip:5023152516@216.135.89.226:5060>
Call-ID: 0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060
CSeq: 104 INFO
User-Agent: Asterisk PBX 1.8.1-rc1
Content-Type: application/dtmf-relay
Content-Length: 23

Signal=1
Duration=51

---

<--- SIP read from UDP:69.64.11.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK1f19538a;received=216.135.89.226
From: "Erik Smith" <sip:5023152516@216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001@69.64.11.6>;tag=as35c589d8
Call-ID: 0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060
CSeq: 104 INFO
Server: Asterisk PBX 1.6.2.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Reliably Transmitting (no NAT) to 69.64.11.6:5060:
INFO sip:+18006262001@69.64.11.6 SIP/2.0
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK479acf86
Max-Forwards: 70
From: "Erik Smith" <sip:5023152516@216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001@69.64.11.6>;tag=as35c589d8
Contact: <sip:5023152516@216.135.89.226:5060>
Call-ID: 0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060
CSeq: 105 INFO
User-Agent: Asterisk PBX 1.8.1-rc1
Content-Type: application/dtmf-relay
Content-Length: 23

Signal=3
Duration=40

---

<--- SIP read from UDP:69.64.11.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK479acf86;received=216.135.89.226
From: "Erik Smith" <sip:5023152516@216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001@69.64.11.6>;tag=as35c589d8
Call-ID: 0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060
CSeq: 105 INFO
Server: Asterisk PBX 1.6.2.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '4caf777a5e65552c61b28e3f2c8e2b4a@69.64.11.6' Method: OPTIONS

<--- SIP read from UDP:69.64.11.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK02aac998;received=216.135.89.226
From: "Erik Smith" <sip:5023152516@216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001@69.64.11.6>;tag=as35c589d8
Call-ID: 0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060
CSeq: 103 INVITE
Server: Asterisk PBX 1.6.2.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 179

v=0
o=root 629951581 629951582 IN IP4 69.64.11.6
s=Asterisk PBX 1.6.2.14-rc1
c=IN IP4 69.64.11.6
t=0 0
m=audio 13978 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 69.64.11.6:13978
list_route: no route
[Nov 29 16:33:38] WARNING[25528]: chan_sip.c:12909 __set_address_from_contact: Invalid contact uri (missing sip: or sips:), attempting to use anyway
[Nov 29 16:33:38] ERROR[25528]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("Ã ", "(null)", ...): Name or service not known
[Nov 29 16:33:38] WARNING[25528]: chan_sip.c:12920 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'Ã '
Transmitting (no NAT) to 69.64.11.6:5060:
ACK sip:+18006262001@69.64.11.6 SIP/2.0
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK6b67a4d8
Max-Forwards: 70
From: "Erik Smith" <sip:5023152516@216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001@69.64.11.6>;tag=as35c589d8
Contact: <sip:5023152516@216.135.89.226:5060>
Call-ID: 0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.1-rc1
Content-Length: 0

---
Reliably Transmitting (no NAT) to 69.64.11.6:5060:
BYE sip:+18006262001@69.64.11.6 SIP/2.0
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK1460ab81
Max-Forwards: 70
From: "Erik Smith" <sip:5023152516@216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001@69.64.11.6>;tag=as35c589d8
Call-ID: 0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060
CSeq: 106 BYE
User-Agent: Asterisk PBX 1.8.1-rc1
Authorization: Digest username="GW01EEMAN", realm="asterisk", algorithm=MD5, uri="sip:+18006262001@69.64.11.6", nonce="669350ed", response="42e7818855fbe4b45995bd5bc2aba30a"
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0

---
Scheduling destruction of SIP dialog '0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060' in 6400 ms (Method: INVITE)
-- SIP/GW01EEMAN-0000003d answered SIP/202-0000003c
Scheduling destruction of SIP dialog '0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 69.64.11.6:5060:
BYE sip:+18006262001@69.64.11.6 SIP/2.0
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK063298a2
Max-Forwards: 70
From: "Erik Smith" <sip:5023152516@216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001@69.64.11.6>;tag=as35c589d8
Call-ID: 0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060
CSeq: 107 BYE
User-Agent: Asterisk PBX 1.8.1-rc1
Authorization: Digest username="GW01EEMAN", realm="asterisk", algorithm=MD5, uri="sip:+18006262001@69.64.11.6", nonce="669350ed", response="42e7818855fbe4b45995bd5bc2aba30a"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---
== Spawn extension (macro-tl-dialout-base, dial-SIP, 7) exited non-zero on 'SIP/202-0000003c' in macro 'tl-dialout-base'
== Spawn extension (macro-tl-dialout-1-trunk, s, 3) exited non-zero on 'SIP/202-0000003c' in macro 'tl-dialout-1-trunk'
== Spawn extension (from-inside-redir, 18006262001, 1) exited non-zero on 'SIP/202-0000003c'
-- Executing [h@from-inside-redir:1] Hangup("SIP/202-0000003c", "") in new stack
== Spawn extension (from-inside-redir, h, 1) exited non-zero on 'SIP/202-0000003c'

<--- SIP read from UDP:69.64.11.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK1460ab81;received=216.135.89.226
From: "Erik Smith" <sip:5023152516@216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001@69.64.11.6>;tag=as35c589d8
Call-ID: 0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060
CSeq: 106 BYE
Server: Asterisk PBX 1.6.2.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

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

<--- SIP read from UDP:69.64.11.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.135.89.226:5060;branch=z9hG4bK063298a2;received=216.135.89.226
From: "Erik Smith" <sip:5023152516@216.135.89.226>;tag=as3813e5d8
To: <sip:+18006262001@69.64.11.6>;tag=as35c589d8
Call-ID: 0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060
CSeq: 107 BYE
Server: Asterisk PBX 1.6.2.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '0794caf82d792f245c6d346d37d2a924@216.135.89.226:5060' Method: INVITE
eeman*CLI> sip set debug off
SIP Debugging Disabled
By: moshe Teitelbaum (moshe) 2010-11-29 16:02:54.000-0600

i have the same issue currently using asterisk 1.4.37 as a endpoint server and 1.4.24.1 as a gateway, i have recently upgraded my end point server from 1.4.24.1 to 1.4.37 but i'm not sure it is related to that

i will upload captures from me to my carrier and the other direction
By: Paul Belanger (pabelanger) 2010-11-30 21:40:51.000-0600

Please attach a full debug log (see below)
---
We require a complete debug log to help triage the issue.

This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue:

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
By: moshe Teitelbaum (moshe) 2010-12-01 06:00:16.000-0600

uploaded debug file from my gateway using 1.4.24.1

i couldn't upload debug fill from my server running 1.4.37 getting error 1- not found (maybe the file is too big over 13 MB too much junk) if you need i will try to figure out an alternative

By: Erik Smith (eeman) 2011-01-14 05:42:53.000-0600

any update?
By: Stefan Schmidt (schmidts) 2011-01-14 09:02:46.000-0600

maybe i am complete wrong but the warning:
[Nov 29 16:33:38] WARNING[25528]: chan_sip.c:12909 __set_address_from_contact: Invalid contact uri (missing sip: or sips:), attempting to use anyway
says there is no contact header in the 200 ok message, and there is really no contact in there only in the 183 session progress.

sorry if i missunderstood the problem but it looks like this missing contact is the issue or the wrong handling of asterisk if its ok to have no contact in a 200 ok.

best regards

stefan
By: Jacob LIpstein (jacobli) 2011-01-26 15:41:35.000-0600

I have run into the same issue. Using Asterisk-1.8.2.2 and libpri-1.4.11.5.
SIP-phone (Polycom,Grandstream) -> asterisk -> PRI -> some numbers
Getting "dead air" (if option "r" is not added to the DIAL command) and disconnection after the dial timeout is expired when calling to the number mentioned above and to several more numbers:
1-888-272-6565 (AT&T), 1-800-332-3226 (Safeco).

When calling to all of numbers which are not returning a ring tone a "cause code 127" record appears in the log:

... app_dial.c:1288 in wait_for_answer: ...is proceeding passing it ...
... sig_pri.c:5080 in pri_dchannel ... PROGRESS with cause code 127 received

But most of such kind of numbers starts an IVR dialog after a short "dead air" excluding mentioned numbers.
By: Walter Doekes (wdoekes) 2011-01-27 02:11:44.000-0600

schmidts: https://issues.asterisk.org/view.php?id=18632 this looks related (1.6 does not send Contact header in 200)
By: Stefan Schmidt (schmidts) 2011-01-27 02:17:20.000-0600

thx walter. it really looks like the same problem.
By: Jacob LIpstein (jacobli) 2011-03-02 18:10:58.000-0600

The problem still exist in vers. 1.8.3
By: Erik Smith (eeman) 2011-06-22 07:02:28.517-0500

any update devs? This problem still exists in 1.8.4.2
By: Jacob LIpstein (jacobli) 2011-07-29 18:19:40.712-0500

It seems the issue is fixed in vers. 1.8.5.0

By: Matt Jordan (mjordan) 2014-06-03 09:32:15.730-0500

Closing as "Fixed" per the last comment.

If this problem is still an issue in Asterisk, please contact a bug marshal in #asterisk-bugs or comment on this issue and we'd be happy to reopen it. Thanks!