[asterisk-bugs] [JIRA] (ASTERISK-24762) not able to make outbound call from Asterisk/1.8.13.1
Matt Jordan (JIRA)
noreply at issues.asterisk.org
Fri Feb 6 09:33:35 CST 2015
[ https://issues.asterisk.org/jira/browse/ASTERISK-24762?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=224781#comment-224781 ]
Matt Jordan edited comment on ASTERISK-24762 at 2/6/15 9:32 AM:
----------------------------------------------------------------
{noformat}
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2015.02.06 15:57:40 =~=~=~=~=~=~=~=~=~=~=~=
[0KReliably Transmitting (NAT) to 10.61.0.101:5060:
OPTIONS sip:61500 at 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 at 10.61.0.4>;tag=as646db1b8
To: <sip:61500 at 10.61.0.101:5060>
Contact: <sip:asterisk at 10.61.0.4:5060>
Call-ID: 1cc3fdb05f09b56939445076518e7cc6 at 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
---
[Kipbx-mlb*CLI>
[0K
<--- 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 at 10.61.0.4>;tag=as646db1b8
To: <sip:61500 at 10.61.0.101:5060>;tag=80b95d422aace411923a46763858d8a5
Call-ID: 1cc3fdb05f09b56939445076518e7cc6 at 10.61.0.4:5060
CSeq: 102 OPTIONS
Contact: <sip:61500 at 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) ---
[Kipbx-mlb*CLI>
[0KReally destroying SIP dialog '1cc3fdb05f09b56939445076518e7cc6 at 10.61.0.4:5060' Method: OPTIONS
[Kipbx-mlb*CLI>
[0KReally destroying SIP dialog '802E0436-4EAB-E411-9D91-16652760B7C0 at 10.61.0.110' Method: REGISTER
[Kipbx-mlb*CLI>
[0KReally destroying SIP dialog '005D132D-4EAB-E411-B4FA-7201B2F660D5 at 10.61.0.104' Method: REGISTER
[Kipbx-mlb*CLI>
[0K
<--- SIP read from UDP:10.61.0.101:5060 --->
INVITE sip:61452377795 at 10.61.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923b46763858d8a5;rport
From: "PhonerLite" <sip:61500 at 10.61.0.4>;tag=1643736679
To: <sip:61452377795 at 10.61.0.4>
Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5 at 10.61.0.101
CSeq: 108 INVITE
Contact: <sip:61500 at 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 at 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) ---
[Kipbx-mlb*CLI>
[0KSending to 10.61.0.101:5060 (no NAT)
Using INVITE request as basis request - 8097975C-2AAC-E411-923A-46763858D8A5 at 10.61.0.101
[Kipbx-mlb*CLI>
[0KFound peer '61500' for '61500' from 10.61.0.101:5060
[Kipbx-mlb*CLI>
[0K
<--- 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 at 10.61.0.4>;tag=1643736679
To: <sip:61452377795 at 10.61.0.4>;tag=as43203775
Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5 at 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
[Kipbx-mlb*CLI>
[0KSupported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1c5808b3"
Content-Length: 0
<------------>
[Kipbx-mlb*CLI>
[0KScheduling destruction of SIP dialog '8097975C-2AAC-E411-923A-46763858D8A5 at 10.61.0.101' in 6400 ms (Method: INVITE)
[Kipbx-mlb*CLI>
[0K
<--- SIP read from UDP:10.61.0.101:5060 --->
ACK sip:61452377795 at 10.61.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923b46763858d8a5;rport
From: "PhonerLite" <sip:61500 at 10.61.0.4>;tag=1643736679
To: <sip:61452377795 at 10.61.0.4>;tag=as43203775
Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5 at 10.61.0.101
CSeq: 108 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
[Kipbx-mlb*CLI>
[0K
<--- SIP read from UDP:10.61.0.101:5060 --->
INVITE sip:61452377795 at 10.61.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923c46763858d8a5;rport
From: "PhonerLite" <sip:61500 at 10.61.0.4>;tag=1643736679
To: <sip:61452377795 at 10.61.0.4>
Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5 at 10.61.0.101
CSeq: 109 INVITE
Contact: <sip:61500 at 10.61.0.101:5060>
Authorization: Digest username="61500", realm="asterisk", nonce="1c5808b3", uri="sip:61452377795 at 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 at 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) ---
[Kipbx-mlb*CLI>
[0KSending to 10.61.0.101:5060 (NAT)
Using INVITE request as basis request - 8097975C-2AAC-E411-923A-46763858D8A5 at 10.61.0.101
Found peer '61500' for '61500' from 10.61.0.101:5060
[Kipbx-mlb*CLI>
[0K == 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
[Kipbx-mlb*CLI>
[0KFound 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
[Kipbx-mlb*CLI>
[0KLooking for 61452377795 in outgoing (domain 10.61.0.4)
[Kipbx-mlb*CLI>
[0Klist_route: hop: <sip:61500 at 10.61.0.101:5060>
[Kipbx-mlb*CLI>
[0K
<--- 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 at 10.61.0.4>;tag=1643736679
To: <sip:61452377795 at 10.61.0.4>
Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5 at 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 at 10.61.0.4:5060>
Content-Length: 0
<------------>
[Kipbx-mlb*CLI>
[0K -- Executing [61452377795 at outgoing:1] [1;36mNoOp[0m("[1;35mSIP/61500-00000017[0m", "[1;35moutgoing CALLERID(all) "ASHISH" <61500> calling EXTEN 61452377795[0m") in new stack
[Kipbx-mlb*CLI>
[0K -- Auto fallthrough, channel 'SIP/61500-00000017' status is 'UNKNOWN'
[Kipbx-mlb*CLI>
[0KScheduling destruction of SIP dialog '8097975C-2AAC-E411-923A-46763858D8A5 at 10.61.0.101' in 6400 ms (Method: INVITE)
[Kipbx-mlb*CLI>
[0K
<--- 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 at 10.61.0.4>;tag=1643736679
To: <sip:61452377795 at 10.61.0.4>;tag=as105d9377
Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5 at 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
<------------>
[Kipbx-mlb*CLI>
[0K
<--- SIP read from UDP:10.61.0.101:5060 --->
ACK sip:61452377795 at 10.61.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923c46763858d8a5;rport
From: "PhonerLite" <sip:61500 at 10.61.0.4>;tag=1643736679
To: <sip:61452377795 at 10.61.0.4>;tag=as105d9377
Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5 at 10.61.0.101
CSeq: 109 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
[Kipbx-mlb*CLI>
[0KReliably Transmitting (no NAT) to 10.61.0.110:5060:
OPTIONS sip:61509 at 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 at 10.61.0.4>;tag=as78edb60e
To: <sip:61509 at 10.61.0.110:5060>
Contact: <sip:asterisk at 10.61.0.4:5060>
Call-ID: 3f03ecf16b30b999040cda492f2a6100 at 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
---
[Kipbx-mlb*CLI>
[0KRetransmitting #1 (no NAT) to 10.61.0.110:5060:
OPTIONS sip:61509 at 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 at 10.61.0.4>;tag=as78edb60e
To: <sip:61509 at 10.61.0.110:5060>
Contact: <sip:asterisk at 10.61.0.4:5060>
Call-ID: 3f03ecf16b30b999040cda492f2a6100 at 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
---
[Kipbx-mlb*CLI>
[0KReliably Transmitting (NAT) to 10.61.0.104:5060:
OPTIONS sip:61504 at 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 at 10.61.0.4>;tag=as6a5cfaad
To: <sip:61504 at 10.61.0.104:5060>
Contact: <sip:asterisk at 10.61.0.4:5060>
Call-ID: 4afe1d1e7fe19bab7f80b02218709b96 at 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
---
[Kipbx-mlb*CLI>
[0K
<--- 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 at 10.61.0.4>;tag=as6a5cfaad
To: <sip:61504 at 10.61.0.104:5060>;tag=802c52672aace411b6857201b2f660d5
Call-ID: 4afe1d1e7fe19bab7f80b02218709b96 at 10.61.0.4:5060
CSeq: 102 OPTIONS
Contact: <sip:61504 at 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) ---
[Kipbx-mlb*CLI>
[0KReally destroying SIP dialog '4afe1d1e7fe19bab7f80b02218709b96 at 10.61.0.4:5060' Method: OPTIONS
[Kipbx-mlb*CLI>
[0KRetransmitting #2 (no NAT) to 10.61.0.110:5060:
OPTIONS sip:61509 at 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 at 10.61.0.4>;tag=as78edb60e
To: <sip:61509 at 10.61.0.110:5060>
Contact: <sip:asterisk at 10.61.0.4:5060>
Call-ID: 3f03ecf16b30b999040cda492f2a6100 at 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
---
[Kipbx-mlb*CLI>
[0KRetransmitting #3 (no NAT) to 10.61.0.110:5060:
OPTIONS sip:61509 at 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 at 10.61.0.4>;tag=as78edb60e
To: <sip:61509 at 10.61.0.110:5060>
Contact: <sip:asterisk at 10.61.0.4:5060>
Call-ID: 3f03ecf16b30b999040cda492f2a6100 at 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
---
[Kipbx-mlb*CLI> sip set debug on
[0KRetransmitting #4 (no NAT) to 10.61.0.110:5060:
OPTIONS sip:61509 at 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 at 10.61.0.4>;tag=as78edb60e
To: <sip:61509 at 10.61.0.110:5060>
Contact: <sip:asterisk at 10.61.0.4:5060>
Call-ID: 3f03ecf16b30b999040cda492f2a6100 at 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] [1;33mNOTICE[0m[17885]: [1;37mchan_sip.c[0m:[1;37m26267[0m [1;37msip_poke_noanswer[0m: Peer '61509' is now UNREACHABLE! Last qualify: 1
Really destroying SIP dialog '3f03ecf16b30b999040cda492f2a6100 at 10.61.0.4:5060' Method: OPTIONS
[Kipbx-mlb*CLI> sip set debug on[Kf
[0KReally destroying SIP dialog '8097975C-2AAC-E411-923A-46763858D8A5 at 10.61.0.101' Method: ACK
[Kipbx-mlb*CLI> sip set debug off
ipbx-mlb*CLI>
[0KSIP Debugging Disabled
[Kipbx-mlb*CLI>
{noformat}
was (Author: ashish):
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2015.02.06 15:57:40 =~=~=~=~=~=~=~=~=~=~=~=
[0KReliably Transmitting (NAT) to 10.61.0.101:5060:
OPTIONS sip:61500 at 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 at 10.61.0.4>;tag=as646db1b8
To: <sip:61500 at 10.61.0.101:5060>
Contact: <sip:asterisk at 10.61.0.4:5060>
Call-ID: 1cc3fdb05f09b56939445076518e7cc6 at 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
---
[Kipbx-mlb*CLI>
[0K
<--- 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 at 10.61.0.4>;tag=as646db1b8
To: <sip:61500 at 10.61.0.101:5060>;tag=80b95d422aace411923a46763858d8a5
Call-ID: 1cc3fdb05f09b56939445076518e7cc6 at 10.61.0.4:5060
CSeq: 102 OPTIONS
Contact: <sip:61500 at 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) ---
[Kipbx-mlb*CLI>
[0KReally destroying SIP dialog '1cc3fdb05f09b56939445076518e7cc6 at 10.61.0.4:5060' Method: OPTIONS
[Kipbx-mlb*CLI>
[0KReally destroying SIP dialog '802E0436-4EAB-E411-9D91-16652760B7C0 at 10.61.0.110' Method: REGISTER
[Kipbx-mlb*CLI>
[0KReally destroying SIP dialog '005D132D-4EAB-E411-B4FA-7201B2F660D5 at 10.61.0.104' Method: REGISTER
[Kipbx-mlb*CLI>
[0K
<--- SIP read from UDP:10.61.0.101:5060 --->
INVITE sip:61452377795 at 10.61.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923b46763858d8a5;rport
From: "PhonerLite" <sip:61500 at 10.61.0.4>;tag=1643736679
To: <sip:61452377795 at 10.61.0.4>
Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5 at 10.61.0.101
CSeq: 108 INVITE
Contact: <sip:61500 at 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 at 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) ---
[Kipbx-mlb*CLI>
[0KSending to 10.61.0.101:5060 (no NAT)
Using INVITE request as basis request - 8097975C-2AAC-E411-923A-46763858D8A5 at 10.61.0.101
[Kipbx-mlb*CLI>
[0KFound peer '61500' for '61500' from 10.61.0.101:5060
[Kipbx-mlb*CLI>
[0K
<--- 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 at 10.61.0.4>;tag=1643736679
To: <sip:61452377795 at 10.61.0.4>;tag=as43203775
Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5 at 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
[Kipbx-mlb*CLI>
[0KSupported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1c5808b3"
Content-Length: 0
<------------>
[Kipbx-mlb*CLI>
[0KScheduling destruction of SIP dialog '8097975C-2AAC-E411-923A-46763858D8A5 at 10.61.0.101' in 6400 ms (Method: INVITE)
[Kipbx-mlb*CLI>
[0K
<--- SIP read from UDP:10.61.0.101:5060 --->
ACK sip:61452377795 at 10.61.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923b46763858d8a5;rport
From: "PhonerLite" <sip:61500 at 10.61.0.4>;tag=1643736679
To: <sip:61452377795 at 10.61.0.4>;tag=as43203775
Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5 at 10.61.0.101
CSeq: 108 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
[Kipbx-mlb*CLI>
[0K
<--- SIP read from UDP:10.61.0.101:5060 --->
INVITE sip:61452377795 at 10.61.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923c46763858d8a5;rport
From: "PhonerLite" <sip:61500 at 10.61.0.4>;tag=1643736679
To: <sip:61452377795 at 10.61.0.4>
Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5 at 10.61.0.101
CSeq: 109 INVITE
Contact: <sip:61500 at 10.61.0.101:5060>
Authorization: Digest username="61500", realm="asterisk", nonce="1c5808b3", uri="sip:61452377795 at 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 at 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) ---
[Kipbx-mlb*CLI>
[0KSending to 10.61.0.101:5060 (NAT)
Using INVITE request as basis request - 8097975C-2AAC-E411-923A-46763858D8A5 at 10.61.0.101
Found peer '61500' for '61500' from 10.61.0.101:5060
[Kipbx-mlb*CLI>
[0K == 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
[Kipbx-mlb*CLI>
[0KFound 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
[Kipbx-mlb*CLI>
[0KLooking for 61452377795 in outgoing (domain 10.61.0.4)
[Kipbx-mlb*CLI>
[0Klist_route: hop: <sip:61500 at 10.61.0.101:5060>
[Kipbx-mlb*CLI>
[0K
<--- 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 at 10.61.0.4>;tag=1643736679
To: <sip:61452377795 at 10.61.0.4>
Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5 at 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 at 10.61.0.4:5060>
Content-Length: 0
<------------>
[Kipbx-mlb*CLI>
[0K -- Executing [61452377795 at outgoing:1] [1;36mNoOp[0m("[1;35mSIP/61500-00000017[0m", "[1;35moutgoing CALLERID(all) "ASHISH" <61500> calling EXTEN 61452377795[0m") in new stack
[Kipbx-mlb*CLI>
[0K -- Auto fallthrough, channel 'SIP/61500-00000017' status is 'UNKNOWN'
[Kipbx-mlb*CLI>
[0KScheduling destruction of SIP dialog '8097975C-2AAC-E411-923A-46763858D8A5 at 10.61.0.101' in 6400 ms (Method: INVITE)
[Kipbx-mlb*CLI>
[0K
<--- 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 at 10.61.0.4>;tag=1643736679
To: <sip:61452377795 at 10.61.0.4>;tag=as105d9377
Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5 at 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
<------------>
[Kipbx-mlb*CLI>
[0K
<--- SIP read from UDP:10.61.0.101:5060 --->
ACK sip:61452377795 at 10.61.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.61.0.101:5060;branch=z9hG4bK8097975c2aace411923c46763858d8a5;rport
From: "PhonerLite" <sip:61500 at 10.61.0.4>;tag=1643736679
To: <sip:61452377795 at 10.61.0.4>;tag=as105d9377
Call-ID: 8097975C-2AAC-E411-923A-46763858D8A5 at 10.61.0.101
CSeq: 109 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
[Kipbx-mlb*CLI>
[0KReliably Transmitting (no NAT) to 10.61.0.110:5060:
OPTIONS sip:61509 at 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 at 10.61.0.4>;tag=as78edb60e
To: <sip:61509 at 10.61.0.110:5060>
Contact: <sip:asterisk at 10.61.0.4:5060>
Call-ID: 3f03ecf16b30b999040cda492f2a6100 at 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
---
[Kipbx-mlb*CLI>
[0KRetransmitting #1 (no NAT) to 10.61.0.110:5060:
OPTIONS sip:61509 at 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 at 10.61.0.4>;tag=as78edb60e
To: <sip:61509 at 10.61.0.110:5060>
Contact: <sip:asterisk at 10.61.0.4:5060>
Call-ID: 3f03ecf16b30b999040cda492f2a6100 at 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
---
[Kipbx-mlb*CLI>
[0KReliably Transmitting (NAT) to 10.61.0.104:5060:
OPTIONS sip:61504 at 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 at 10.61.0.4>;tag=as6a5cfaad
To: <sip:61504 at 10.61.0.104:5060>
Contact: <sip:asterisk at 10.61.0.4:5060>
Call-ID: 4afe1d1e7fe19bab7f80b02218709b96 at 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
---
[Kipbx-mlb*CLI>
[0K
<--- 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 at 10.61.0.4>;tag=as6a5cfaad
To: <sip:61504 at 10.61.0.104:5060>;tag=802c52672aace411b6857201b2f660d5
Call-ID: 4afe1d1e7fe19bab7f80b02218709b96 at 10.61.0.4:5060
CSeq: 102 OPTIONS
Contact: <sip:61504 at 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) ---
[Kipbx-mlb*CLI>
[0KReally destroying SIP dialog '4afe1d1e7fe19bab7f80b02218709b96 at 10.61.0.4:5060' Method: OPTIONS
[Kipbx-mlb*CLI>
[0KRetransmitting #2 (no NAT) to 10.61.0.110:5060:
OPTIONS sip:61509 at 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 at 10.61.0.4>;tag=as78edb60e
To: <sip:61509 at 10.61.0.110:5060>
Contact: <sip:asterisk at 10.61.0.4:5060>
Call-ID: 3f03ecf16b30b999040cda492f2a6100 at 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
---
[Kipbx-mlb*CLI>
[0KRetransmitting #3 (no NAT) to 10.61.0.110:5060:
OPTIONS sip:61509 at 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 at 10.61.0.4>;tag=as78edb60e
To: <sip:61509 at 10.61.0.110:5060>
Contact: <sip:asterisk at 10.61.0.4:5060>
Call-ID: 3f03ecf16b30b999040cda492f2a6100 at 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
---
[Kipbx-mlb*CLI> sip set debug on
[0KRetransmitting #4 (no NAT) to 10.61.0.110:5060:
OPTIONS sip:61509 at 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 at 10.61.0.4>;tag=as78edb60e
To: <sip:61509 at 10.61.0.110:5060>
Contact: <sip:asterisk at 10.61.0.4:5060>
Call-ID: 3f03ecf16b30b999040cda492f2a6100 at 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] [1;33mNOTICE[0m[17885]: [1;37mchan_sip.c[0m:[1;37m26267[0m [1;37msip_poke_noanswer[0m: Peer '61509' is now UNREACHABLE! Last qualify: 1
Really destroying SIP dialog '3f03ecf16b30b999040cda492f2a6100 at 10.61.0.4:5060' Method: OPTIONS
[Kipbx-mlb*CLI> sip set debug on[Kf
[0KReally destroying SIP dialog '8097975C-2AAC-E411-923A-46763858D8A5 at 10.61.0.101' Method: ACK
[Kipbx-mlb*CLI> sip set debug off
ipbx-mlb*CLI>
[0KSIP Debugging Disabled
[Kipbx-mlb*CLI>
> not able to make outbound call from Asterisk/1.8.13.1
> -----------------------------------------------------
>
> Key: ASTERISK-24762
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-24762
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Affects Versions: 1.8.13.1
> Environment: == Using SIP RTP CoS mark 5
> -- Executing [61452377795 at 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'
> Reporter: Ashish
>
> {noformat}
> == Using SIP RTP CoS mark 5
> -- Executing [61452377795 at 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}
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