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Summary:ASTERISK-24762: not able to make outbound call from Asterisk/1.8.13.1
Reporter:Ashish (Ashish)Labels:
Date Opened:2015-02-05 23:14:59.000-0600Date Closed:2015-02-06 09:33:33.000-0600
Priority:MajorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:1.8.13.1 Frequency of
Occurrence		Constant
Related
Issues:
Environment: == Using SIP RTP CoS mark 5 -- Executing [61452377795@outgoing:1] NoOp("SIP/61500-00000018", "outgoing CALLERID(all) "ASHISH" <61500> calling EXTEN 61452377795") in new stack -- Auto fallthrough, channel 'SIP/61500-00000018' status is 'UNKNOWN' Attachments:
Description:{noformat}
== Using SIP RTP CoS mark 5
   -- Executing [61452377795@outgoing:1] NoOp("SIP/61500-00000018", "outgoing CALLERID(all) "ASHISH" <61500> calling EXTEN 61452377795") in new stack
   -- Auto fallthrough, channel 'SIP/61500-00000018' status is 'UNKNOWN'
{noformat}
Comments:By: Ashish (Ashish) 2015-02-05 23:31:39.387-0600

{noformat}
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2015.02.06 15:57:40 =~=~=~=~=~=~=~=~=~=~=~=

Reliably Transmitting (NAT) to 10.61.0.101:5060:
OPTIONS sip:61500@10.61.0.101:5060 SIP/2.0

Via: SIP/2.0/UDP 10.61.0.4:5060;branch=z9hG4bK6f1253a9;rport

Max-Forwards: 70

From: "asterisk" <sip:asterisk@10.61.0.4>;tag=as646db1b8

To: <sip:61500@10.61.0.101:5060>

Contact: <sip:asterisk@10.61.0.4:5060>

Call-ID: 1cc3fdb05f09b56939445076518e7cc6@10.61.0.4:5060

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3

Date: Fri, 06 Feb 2015 04:57:21 GMT

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

Supported: replaces, timer

Content-Length: 0




---

ipbx-mlb*CLI>

<--- SIP read from UDP:10.61.0.101:5060 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.61.0.4:5060;branch=z9hG4bK6f1253a9;rport=5060

From: "asterisk" <sip:asterisk@10.61.0.4>;tag=as646db1b8

To: <sip:61500@10.61.0.101:5060>;tag=80b95d422aace411923a46763858d8a5

Call-ID: 1cc3fdb05f09b56939445076518e7cc6@10.61.0.4:5060

CSeq: 102 OPTIONS

Contact: <sip:61500@10.61.0.101:5060>

Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE

Server: SIPPER for PhonerLite

Content-Length: 0




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

ipbx-mlb*CLI>
Really destroying SIP dialog '1cc3fdb05f09b56939445076518e7cc6@10.61.0.4:5060' Method: OPTIONS

ipbx-mlb*CLI>
Really destroying SIP dialog '802E0436-4EAB-E411-9D91-16652760B7C0@10.61.0.110' Method: REGISTER

ipbx-mlb*CLI>
Really destroying SIP dialog '005D132D-4EAB-E411-B4FA-7201B2F660D5@10.61.0.104' Method: REGISTER

ipbx-mlb*CLI>

<--- SIP read from UDP:10.61.0.101:5060 --->
INVITE sip:61452377795@10.61.0.4 SIP/2.0

Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923b46763858d8a5;rport

From: "PhonerLite" <sip:61500@10.61.0.4>;tag=1643736679

To: <sip:61452377795@10.61.0.4>

Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101

CSeq: 108 INVITE

Contact: <sip:61500@10.61.0.101:5060>

Content-Type: application/sdp

Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE

Max-Forwards: 70

Supported: 100rel, replaces, from-change

P-Early-Media: supported

User-Agent: SIPPER for PhonerLite

P-Preferred-Identity: <sip:61500@10.61.0.4>

Content-Length:   302



v=0

o=- 4097900355 1 IN IP4 10.61.0.101

s=SIPPER for PhonerLite

c=IN IP4 10.61.0.101

t=0 0

m=audio 5062 RTP/AVP 8 0 3 97 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:97 iLBC/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ssrc:2997880776

a=sendrecv


<------------->
--- (15 headers 14 lines) ---

ipbx-mlb*CLI>
Sending to 10.61.0.101:5060 (no NAT)
Using INVITE request as basis request - 8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101

ipbx-mlb*CLI>
Found peer '61500' for '61500' from 10.61.0.101:5060

ipbx-mlb*CLI>

<--- Reliably Transmitting (NAT) to 10.61.0.101:5060 --->
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923b46763858d8a5;received=10.61.0.101;rport=5060

From: "PhonerLite" <sip:61500@10.61.0.4>;tag=1643736679

To: <sip:61452377795@10.61.0.4>;tag=as43203775

Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101

CSeq: 108 INVITE

Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3

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


ipbx-mlb*CLI>
Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1c5808b3"

Content-Length: 0




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

ipbx-mlb*CLI>
Scheduling destruction of SIP dialog '8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101' in 6400 ms (Method: INVITE)

ipbx-mlb*CLI>

<--- SIP read from UDP:10.61.0.101:5060 --->
ACK sip:61452377795@10.61.0.4 SIP/2.0

Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923b46763858d8a5;rport

From: "PhonerLite" <sip:61500@10.61.0.4>;tag=1643736679

To: <sip:61452377795@10.61.0.4>;tag=as43203775

Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101

CSeq: 108 ACK

Content-Length: 0




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

ipbx-mlb*CLI>

<--- SIP read from UDP:10.61.0.101:5060 --->
INVITE sip:61452377795@10.61.0.4 SIP/2.0

Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923c46763858d8a5;rport

From: "PhonerLite" <sip:61500@10.61.0.4>;tag=1643736679

To: <sip:61452377795@10.61.0.4>

Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101

CSeq: 109 INVITE

Contact: <sip:61500@10.61.0.101:5060>

Authorization: Digest username="61500", realm="asterisk", nonce="1c5808b3", uri="sip:61452377795@10.61.0.4", response="d13bba2f0e2554ca11820f9b66786718", algorithm=MD5

Content-Type: application/sdp

Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE

Max-Forwards: 70

Supported: 100rel, replaces, from-change

P-Early-Media: supported

User-Agent: SIPPER for PhonerLite

P-Preferred-Identity: <sip:61500@10.61.0.4>

Content-Length:   302



v=0

o=- 4097900355 1 IN IP4 10.61.0.101

s=SIPPER for PhonerLite

c=IN IP4 10.61.0.101

t=0 0

m=audio 5062 RTP/AVP 8 0 3 97 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:97 iLBC/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ssrc:2997880776

a=sendrecv


<------------->
--- (16 headers 14 lines) ---

ipbx-mlb*CLI>
Sending to 10.61.0.101:5060 (NAT)
Using INVITE request as basis request - 8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101
Found peer '61500' for '61500' from 10.61.0.101:5060

ipbx-mlb*CLI>
  == Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0

ipbx-mlb*CLI>
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.61.0.101:5062

ipbx-mlb*CLI>
Looking for 61452377795 in outgoing (domain 10.61.0.4)

ipbx-mlb*CLI>
list_route: hop: <sip:61500@10.61.0.101:5060>

ipbx-mlb*CLI>

<--- Transmitting (NAT) to 10.61.0.101:5060 --->
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923c46763858d8a5;received=10.61.0.101;rport=5060

From: "PhonerLite" <sip:61500@10.61.0.4>;tag=1643736679

To: <sip:61452377795@10.61.0.4>

Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101

CSeq: 109 INVITE

Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3

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

Supported: replaces, timer

Contact: <sip:61452377795@10.61.0.4:5060>

Content-Length: 0




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

ipbx-mlb*CLI>
    -- Executing [61452377795@outgoing:1] NoOp("SIP/61500-00000017", "outgoing CALLERID(all) "ASHISH" <61500> calling EXTEN 61452377795") in new stack

ipbx-mlb*CLI>
    -- Auto fallthrough, channel 'SIP/61500-00000017' status is 'UNKNOWN'

ipbx-mlb*CLI>
Scheduling destruction of SIP dialog '8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101' in 6400 ms (Method: INVITE)

ipbx-mlb*CLI>

<--- Reliably Transmitting (NAT) to 10.61.0.101:5060 --->
SIP/2.0 603 Declined

Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923c46763858d8a5;received=10.61.0.101;rport=5060

From: "PhonerLite" <sip:61500@10.61.0.4>;tag=1643736679

To: <sip:61452377795@10.61.0.4>;tag=as105d9377

Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101

CSeq: 109 INVITE

Server: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3

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

Supported: replaces, timer

Content-Length: 0




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

ipbx-mlb*CLI>

<--- SIP read from UDP:10.61.0.101:5060 --->
ACK sip:61452377795@10.61.0.4 SIP/2.0

Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923c46763858d8a5;rport

From: "PhonerLite" <sip:61500@10.61.0.4>;tag=1643736679

To: <sip:61452377795@10.61.0.4>;tag=as105d9377

Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101

CSeq: 109 ACK

Content-Length: 0




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

ipbx-mlb*CLI>
Reliably Transmitting (no NAT) to 10.61.0.110:5060:
OPTIONS sip:61509@10.61.0.110:5060 SIP/2.0

Via: SIP/2.0/UDP 10.61.0.4:5060;branch=z9hG4bK26ced09d

Max-Forwards: 70

From: "asterisk" <sip:asterisk@10.61.0.4>;tag=as78edb60e

To: <sip:61509@10.61.0.110:5060>

Contact: <sip:asterisk@10.61.0.4:5060>

Call-ID: 3f03ecf16b30b999040cda492f2a6100@10.61.0.4:5060

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3

Date: Fri, 06 Feb 2015 04:58:07 GMT

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

Supported: replaces, timer

Content-Length: 0




---

ipbx-mlb*CLI>
Retransmitting #1 (no NAT) to 10.61.0.110:5060:
OPTIONS sip:61509@10.61.0.110:5060 SIP/2.0

Via: SIP/2.0/UDP 10.61.0.4:5060;branch=z9hG4bK26ced09d

Max-Forwards: 70

From: "asterisk" <sip:asterisk@10.61.0.4>;tag=as78edb60e

To: <sip:61509@10.61.0.110:5060>

Contact: <sip:asterisk@10.61.0.4:5060>

Call-ID: 3f03ecf16b30b999040cda492f2a6100@10.61.0.4:5060

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3

Date: Fri, 06 Feb 2015 04:58:07 GMT

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

Supported: replaces, timer

Content-Length: 0




---

ipbx-mlb*CLI>
Reliably Transmitting (NAT) to 10.61.0.104:5060:
OPTIONS sip:61504@10.61.0.104:5060 SIP/2.0

Via: SIP/2.0/UDP 10.61.0.4:5060;branch=z9hG4bK37ee2d9b;rport

Max-Forwards: 70

From: "asterisk" <sip:asterisk@10.61.0.4>;tag=as6a5cfaad

To: <sip:61504@10.61.0.104:5060>

Contact: <sip:asterisk@10.61.0.4:5060>

Call-ID: 4afe1d1e7fe19bab7f80b02218709b96@10.61.0.4:5060

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3

Date: Fri, 06 Feb 2015 04:58:08 GMT

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

Supported: replaces, timer

Content-Length: 0




---

ipbx-mlb*CLI>

<--- SIP read from UDP:10.61.0.104:5060 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.61.0.4:5060;branch=z9hG4bK37ee2d9b;rport=5060

From: "asterisk" <sip:asterisk@10.61.0.4>;tag=as6a5cfaad

To: <sip:61504@10.61.0.104:5060>;tag=802c52672aace411b6857201b2f660d5

Call-ID: 4afe1d1e7fe19bab7f80b02218709b96@10.61.0.4:5060

CSeq: 102 OPTIONS

Contact: <sip:61504@10.61.0.104:5060>

Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE

Server: SIPPER for PhonerLite

Content-Length: 0




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

ipbx-mlb*CLI>
Really destroying SIP dialog '4afe1d1e7fe19bab7f80b02218709b96@10.61.0.4:5060' Method: OPTIONS

ipbx-mlb*CLI>
Retransmitting #2 (no NAT) to 10.61.0.110:5060:
OPTIONS sip:61509@10.61.0.110:5060 SIP/2.0

Via: SIP/2.0/UDP 10.61.0.4:5060;branch=z9hG4bK26ced09d

Max-Forwards: 70

From: "asterisk" <sip:asterisk@10.61.0.4>;tag=as78edb60e

To: <sip:61509@10.61.0.110:5060>

Contact: <sip:asterisk@10.61.0.4:5060>

Call-ID: 3f03ecf16b30b999040cda492f2a6100@10.61.0.4:5060

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3

Date: Fri, 06 Feb 2015 04:58:07 GMT

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

Supported: replaces, timer

Content-Length: 0




---

ipbx-mlb*CLI>
Retransmitting #3 (no NAT) to 10.61.0.110:5060:
OPTIONS sip:61509@10.61.0.110:5060 SIP/2.0

Via: SIP/2.0/UDP 10.61.0.4:5060;branch=z9hG4bK26ced09d

Max-Forwards: 70

From: "asterisk" <sip:asterisk@10.61.0.4>;tag=as78edb60e

To: <sip:61509@10.61.0.110:5060>

Contact: <sip:asterisk@10.61.0.4:5060>

Call-ID: 3f03ecf16b30b999040cda492f2a6100@10.61.0.4:5060

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3

Date: Fri, 06 Feb 2015 04:58:07 GMT

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

Supported: replaces, timer

Content-Length: 0




---

ipbx-mlb*CLI> sip set debug on
Retransmitting #4 (no NAT) to 10.61.0.110:5060:
OPTIONS sip:61509@10.61.0.110:5060 SIP/2.0

Via: SIP/2.0/UDP 10.61.0.4:5060;branch=z9hG4bK26ced09d

Max-Forwards: 70

From: "asterisk" <sip:asterisk@10.61.0.4>;tag=as78edb60e

To: <sip:61509@10.61.0.110:5060>

Contact: <sip:asterisk@10.61.0.4:5060>

Call-ID: 3f03ecf16b30b999040cda492f2a6100@10.61.0.4:5060

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3

Date: Fri, 06 Feb 2015 04:58:07 GMT

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

Supported: replaces, timer

Content-Length: 0




---
[Feb  6 04:58:11] NOTICE[17885]: chan_sip.c:26267 sip_poke_noanswer: Peer '61509' is now UNREACHABLE!  Last qualify: 1
Really destroying SIP dialog '3f03ecf16b30b999040cda492f2a6100@10.61.0.4:5060' Method: OPTIONS

ipbx-mlb*CLI> sip set debug onf
Really destroying SIP dialog '8097975C-2AAC-E411-923A-46763858D8A5@10.61.0.101' Method: ACK

ipbx-mlb*CLI> sip set debug off

ipbx-mlb*CLI>
SIP Debugging Disabled

ipbx-mlb*CLI>
{noformat}
By: Ashish (Ashish) 2015-02-05 23:32:02.760-0600

Global Settings:
----------------
 UDP Bindaddress:        0.0.0.0:5060
 TCP SIP Bindaddress:    Disabled
 TLS SIP Bindaddress:    Disabled
 Videosupport:           No
 Textsupport:            No
 Ignore SDP sess. ver.:  No
 AutoCreate Peer:        No
 Match Auth Username:    No
 Allow unknown access:   Yes
 Allow subscriptions:    Yes
 Allow overlap dialing:  Yes
 Allow promisc. redir:   No
 Enable call counters:   No
 SIP domain support:     Yes
 Realm. auth:            No
 Our auth realm          asterisk
 Use domains as realms:  No
 Call to non-local dom.: Yes
 URI user is phone no:   No
 Always auth rejects:    Yes
 Direct RTP setup:       No
 User Agent:             Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
 SDP Session Name:       Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
 SDP Owner Name:         root
 Reg. context:           (not set)
 Regexten on Qualify:    No
 Legacy userfield parse: No
 Caller ID:              asterisk
 From: Domain:
 Record SIP history:     On
 Call Events:            Off
 Auth. Failure Events:   Off
 T.38 support:           No
 T.38 EC mode:           Unknown
 T.38 MaxDtgrm:          -1
 SIP realtime:           Disabled
 Qualify Freq :          60000 ms
 Q.850 Reason header:    No
 Store SIP_CAUSE:        No

Network QoS Settings:
---------------------------
 IP ToS SIP:             CS0
 IP ToS RTP audio:       CS0
 IP ToS RTP video:       CS0
 IP ToS RTP text:        CS0
 802.1p CoS SIP:         4
 802.1p CoS RTP audio:   5
 802.1p CoS RTP video:   6
 802.1p CoS RTP text:    5
 Jitterbuffer enabled:   No

Network Settings:
---------------------------
 SIP address remapping:  Enabled using externaddr
 Externhost:             <none>
 Externaddr:             110.142.242.138:0
 Externrefresh:          10
 Localnet:               10.61.0.0/255.255.192.0

Global Signalling Settings:
---------------------------
 Codecs:                 0x80000008050e (gsm|ulaw|alaw|g729|ilbc|h263|testlaw)
 Codec Order:            alaw:20,gsm:20,g729:20,ilbc:30
 Relax DTMF:             No
 RFC2833 Compensation:   No
 Symmetric RTP:          No
 Compact SIP headers:    No
 RTP Keepalive:          0 (Disabled)
 RTP Timeout:            0 (Disabled)
 RTP Hold Timeout:       0 (Disabled)
 MWI NOTIFY mime type:   application/simple-message-summary
 DNS SRV lookup:         Yes
 Pedantic SIP support:   No
 Reg. min duration       60 secs
 Reg. max duration:      3600 secs
 Reg. default duration:  1800 secs
 Outbound reg. timeout:  20 secs
 Outbound reg. attempts: 0
 Notify ringing state:   Yes
   Include CID:          No
 Notify hold state:      Yes
 SIP Transfer mode:      open
 Max Call Bitrate:       384 kbps
 Auto-Framing:           No
 Outb. proxy:            <not set>
 Session Timers:         Accept
 Session Refresher:      uas
 Session Expires:        1800 secs
 Session Min-SE:         90 secs
 Timer T1:               500
 Timer T1 minimum:       100
 Timer B:                32000
 No premature media:     Yes
 Max forwards:           70

Default Settings:
-----------------
 Allowed transports:     UDP
 Outbound transport:     UDP
 Context:                default
 Force rport:            No
 DTMF:                   rfc2833
 Qualify:                0
 Use ClientCode:         No
 Progress inband:        Never
 Language:
 MOH Interpret:          default
 MOH Suggest:
 Voice Mail Extension:   asterisk

By: Matt Jordan (mjordan) 2015-02-06 09:33:27.484-0600

Thanks for your comments. This does not appear to be a bug report and we are closing it. We appreciate the difficulties you are facing, but it would make more sense to raise your question in the support tracker, http://www.asterisk.org/support.