[asterisk-bugs] [JIRA] Commented: (ASTERISK-20440) No ringback towards SLAstation on outbound trunk call.
dkerr (JIRA)
noreply at issues.asterisk.org
Tue Sep 18 15:13:27 CDT 2012
[ https://issues.asterisk.org/jira/browse/ASTERISK-20440?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=197257#comment-197257 ]
dkerr commented on ASTERISK-20440:
----------------------------------
<--- SIP read from UDP:166.137.87.102:61089 --->
INVITE sip:*1 at pbx.kerrfamily.org SIP/2.0
Via: SIP/2.0/UDP 10.10.242.5:2034;branch=z9hG4bKu8cr2TDbjzwV6T7Y;rport
Contact: <sip:104 at 10.10.242.5:2034>
Max-Forwards: 70
From: "David" <sip:104 at pbx.kerrfamily.org>;tag=0EA9FBFAD87083C1C7CFFCA6BB22C2F9
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
User-Agent: Acrobits Softphone/5.2
To: <sip:*1 at pbx.kerrfamily.org>
Content-Type: application/sdp
Call-ID: D084B497A0A9430F7ABB0D3DA7A7D78AC60B20F9
CSeq: 1 INVITE
Content-Length: 240
v=0
o=- 19412 50398 IN IP4 10.10.242.5
s=mkwhemp
c=IN IP4 10.10.242.5
t=0 0
m=audio 19108 RTP/AVP 102 3 0 8 9 101
a=rtpmap:101 telephone-event/8000
a=rtpmap:102 ILBC/8000
a=fmtp:102 mode=30
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (13 headers 12 lines) ---
Sending to 166.137.87.102:61089 (NAT)
Using INVITE request as basis request - D084B497A0A9430F7ABB0D3DA7A7D78AC60B20F9
Found peer '104' for '104' from 166.137.87.102:61089
== Using SIP RTP CoS mark 5
Found RTP audio format 102
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Found audio description format ILBC for ID 102
Capabilities: us - 0x1506 (gsm|ulaw|g729|ilbc|g722), peer - audio=0x140e (gsm|ulaw|alaw|ilbc|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x1406 (gsm|ulaw|ilbc|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.10.242.5:19108
Looking for *1 in DialPlanSLA (domain pbx.kerrfamily.org)
list_route: hop: <sip:104 at 10.10.242.5:2034>
<--- Transmitting (NAT) to 166.137.87.102:61089 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.242.5:2034;branch=z9hG4bKu8cr2TDbjzwV6T7Y;received=166.137.87.102;rport=61089
From: "David" <sip:104 at pbx.kerrfamily.org>;tag=0EA9FBFAD87083C1C7CFFCA6BB22C2F9
To: <sip:*1 at pbx.kerrfamily.org>
Call-ID: D084B497A0A9430F7ABB0D3DA7A7D78AC60B20F9
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:*1 at 24.128.119.26:5060>
Content-Length: 0
<------------>
-- Executing [*1 at DialPlanSLA:1] SLAStation("SIP/104-0000004e", "SLAphone104_SLAtrunk1") in new stack
-- Called s at OutboundSLA
-- Executing [s at OutboundSLA:1] NoOp("Local/s at OutboundSLA-df6c;2", "SLA Outbound context") in new stack
-- Executing [s at OutboundSLA:2] DISA("Local/s at OutboundSLA-df6c;2", "no-password,OutboundSLA") in new stack
-- Local/s at OutboundSLA-df6c;1 answered
-- Created MeetMe conference 1023 for conference 'SLA_SLAtrunk1'
Audio is at 16394
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x1000 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 166.137.87.102:61089 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.242.5:2034;branch=z9hG4bKu8cr2TDbjzwV6T7Y;received=166.137.87.102;rport=61089
From: "David" <sip:104 at pbx.kerrfamily.org>;tag=0EA9FBFAD87083C1C7CFFCA6BB22C2F9
To: <sip:*1 at pbx.kerrfamily.org>;tag=as6420d410
Call-ID: D084B497A0A9430F7ABB0D3DA7A7D78AC60B20F9
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:*1 at 24.128.119.26:5060>
Content-Type: application/sdp
Content-Length: 334
v=0
o=root 2141099548 2141099548 IN IP4 24.128.119.26
s=Asterisk PBX 1.8.15.0
c=IN IP4 24.128.119.26
t=0 0
m=audio 16394 RTP/AVP 102 3 0 9 101
a=rtpmap:102 iLBC/8000
a=fmtp:102 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
<--- SIP read from UDP:166.137.87.102:61089 --->
ACK sip:*1 at 24.128.119.26:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.242.5:2034;branch=z9hG4bKhBQz9Nf9uEDHCZLb;rport
Max-Forwards: 70
To: <sip:*1 at pbx.kerrfamily.org>;tag=as6420d410
From: "David" <sip:104 at pbx.kerrfamily.org>;tag=0EA9FBFAD87083C1C7CFFCA6BB22C2F9
Call-ID: D084B497A0A9430F7ABB0D3DA7A7D78AC60B20F9
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '51214729 at 192_168_17_141' Method: REGISTER
Really destroying SIP dialog '1838558966 at 192_168_17_141' Method: REGISTER
-- Executing [19144994018 at OutboundSLA:1] Macro("Local/s at OutboundSLA-df6c;2", "trunkdial-failover,SIP/vitel-outbound/19144994018,SIP/voipcheap/0019144994018,vitel-outbound,voipcheap") in new stack
-- Executing [s at macro-trunkdial-failover:1] NoOp("Local/s at OutboundSLA-df6c;2", "trunk dial by 104") in new stack
-- Executing [s at macro-trunkdial-failover:2] ExecIf("Local/s at OutboundSLA-df6c;2", "0?Set(VOLUME(RX)=10)") in new stack
-- Executing [s at macro-trunkdial-failover:3] ExecIf("Local/s at OutboundSLA-df6c;2", "0?Set(VOLUME(RX)=10)") in new stack
-- Executing [s at macro-trunkdial-failover:4] GotoIf("Local/s at OutboundSLA-df6c;2", "0?1-fmsetcid,1") in new stack
-- Executing [s at macro-trunkdial-failover:5] GotoIf("Local/s at OutboundSLA-df6c;2", "0?1-setgbobname,1") in new stack
-- Executing [s at macro-trunkdial-failover:6] Set("Local/s at OutboundSLA-df6c;2", "CALLERID(num)=2034381428") in new stack
-- Executing [s at macro-trunkdial-failover:7] GotoIf("Local/s at OutboundSLA-df6c;2", "1?1-dial,1") in new stack
-- Goto (macro-trunkdial-failover,1-dial,1)
-- Executing [1-dial at macro-trunkdial-failover:1] Dial("Local/s at OutboundSLA-df6c;2", "SIP/vitel-outbound/19144994018,,rT") in new stack
== Using SIP RTP CoS mark 5
Audio is at 16560
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 64.2.142.189:5060:
INVITE sip:19144994018 at outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 24.128.119.26:5060;branch=z9hG4bK425dbaee;rport
Max-Forwards: 70
From: "Softphone" <sip:dkerr at 24.128.119.26>;tag=as5536a4cd
To: <sip:19144994018 at outbound.vitelity.net>
Contact: <sip:dkerr at 24.128.119.26:5060>
Call-ID: 18bf1819371d28b1659750b3021942e2 at 24.128.119.26:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.15.0
Date: Tue, 18 Sep 2012 19:32:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Softphone" <sip:2034381428 at 24.128.119.26>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 1439404609 1439404609 IN IP4 24.128.119.26
s=Asterisk PBX 1.8.15.0
c=IN IP4 24.128.119.26
t=0 0
m=audio 16560 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called SIP/vitel-outbound/19144994018
<--- SIP read from UDP:64.2.142.189:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 24.128.119.26:5060;branch=z9hG4bK425dbaee;received=24.128.119.26;rport=5060
From: "Softphone" <sip:dkerr at 24.128.119.26>;tag=as5536a4cd
To: <sip:19144994018 at outbound.vitelity.net>;tag=as7a338b40
Call-ID: 18bf1819371d28b1659750b3021942e2 at 24.128.119.26:5060
CSeq: 102 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="464022aa"
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 64.2.142.189:5060:
ACK sip:19144994018 at outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 24.128.119.26:5060;branch=z9hG4bK425dbaee;rport
Max-Forwards: 70
From: "Softphone" <sip:dkerr at 24.128.119.26>;tag=as5536a4cd
To: <sip:19144994018 at outbound.vitelity.net>;tag=as7a338b40
Contact: <sip:dkerr at 24.128.119.26:5060>
Call-ID: 18bf1819371d28b1659750b3021942e2 at 24.128.119.26:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.15.0
Content-Length: 0
---
Audio is at 16560
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 64.2.142.189:5060:
INVITE sip:19144994018 at outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 24.128.119.26:5060;branch=z9hG4bK7830a8bb;rport
Max-Forwards: 70
From: "Softphone" <sip:dkerr at 24.128.119.26>;tag=as5536a4cd
To: <sip:19144994018 at outbound.vitelity.net>
Contact: <sip:dkerr at 24.128.119.26:5060>
Call-ID: 18bf1819371d28b1659750b3021942e2 at 24.128.119.26:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.15.0
Proxy-Authorization: Digest username="dkerr", realm="asterisk", algorithm=MD5, uri="sip:19144994018 at outbound.vitelity.net", nonce="464022aa", response="2ea2f6381cfb62f43f540d415fb1bad4"
Date: Tue, 18 Sep 2012 19:32:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: "Softphone" <sip:2034381428 at 24.128.119.26>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 1439404609 1439404610 IN IP4 24.128.119.26
s=Asterisk PBX 1.8.15.0
c=IN IP4 24.128.119.26
t=0 0
m=audio 16560 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:64.2.142.189:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 24.128.119.26:5060;branch=z9hG4bK7830a8bb;received=24.128.119.26;rport=5060
From: "Softphone" <sip:dkerr at 24.128.119.26>;tag=as5536a4cd
To: <sip:19144994018 at outbound.vitelity.net>
Call-ID: 18bf1819371d28b1659750b3021942e2 at 24.128.119.26:5060
CSeq: 103 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:19144994018 at 64.2.142.189>
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from UDP:64.2.142.189:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 24.128.119.26:5060;branch=z9hG4bK7830a8bb;received=24.128.119.26;rport=5060
From: "Softphone" <sip:dkerr at 24.128.119.26>;tag=as5536a4cd
To: <sip:19144994018 at outbound.vitelity.net>;tag=as13109d91
Call-ID: 18bf1819371d28b1659750b3021942e2 at 24.128.119.26:5060
CSeq: 103 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:19144994018 at 64.2.142.189>
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:19144994018 at 64.2.142.189>
-- SIP/vitel-outbound-0000004f is ringing
Really destroying SIP dialog '47a723b6639149626f3ec3df6360f6e5 at 192.168.17.1' Method: REGISTER
<--- SIP read from UDP:64.2.142.189:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 24.128.119.26:5060;branch=z9hG4bK7830a8bb;received=24.128.119.26;rport=5060
From: "Softphone" <sip:dkerr at 24.128.119.26>;tag=as5536a4cd
To: <sip:19144994018 at outbound.vitelity.net>;tag=as13109d91
Call-ID: 18bf1819371d28b1659750b3021942e2 at 24.128.119.26:5060
CSeq: 103 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:19144994018 at 64.2.142.189>
Content-Type: application/sdp
Content-Length: 287
v=0
o=root 16008 16008 IN IP4 64.2.142.189
s=session
c=IN IP4 64.2.142.189
t=0 0
m=audio 18392 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 14 lines) ---
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 64.2.142.189:18392
list_route: hop: <sip:19144994018 at 64.2.142.189>
set_destination: Parsing <sip:19144994018 at 64.2.142.189> for address/port to send to
set_destination: set destination to 64.2.142.189:5060
Transmitting (NAT) to 64.2.142.189:5060:
ACK sip:19144994018 at 64.2.142.189 SIP/2.0
Via: SIP/2.0/UDP 24.128.119.26:5060;branch=z9hG4bK270f3b90;rport
Max-Forwards: 70
From: "Softphone" <sip:dkerr at 24.128.119.26>;tag=as5536a4cd
To: <sip:19144994018 at outbound.vitelity.net>;tag=as13109d91
Contact: <sip:dkerr at 24.128.119.26:5060>
Call-ID: 18bf1819371d28b1659750b3021942e2 at 24.128.119.26:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.15.0
Content-Length: 0
---
-- SIP/vitel-outbound-0000004f answered Local/s at OutboundSLA-df6c;2
-- Executing [h at OutboundSLA:1] Hangup("Local/s at OutboundSLA-df6c;2", "") in new stack
== Spawn extension (OutboundSLA, h, 1) exited non-zero on 'Local/s at OutboundSLA-df6c;2'
== Spawn extension (macro-trunkdial-failover, 1-dial, 1) exited non-zero on 'Local/s at OutboundSLA-df6c;2' in macro 'trunkdial-failover'
== Spawn extension (OutboundSLA, 19144994018, 1) exited non-zero on 'Local/s at OutboundSLA-df6c;2'
> No ringback towards SLAstation on outbound trunk call.
> ------------------------------------------------------
>
> Key: ASTERISK-20440
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-20440
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Applications/app_meetme
> Affects Versions: 1.8.15.0
> Environment: Astlinux
> Reporter: dkerr
>
> This is similar to https://issues.asterisk.org/jira/browse/ASTERISK-11549 except that the problem is with calls in the other direction. If I make an outbound call from a SLA phone using the SLAStation then there is no ringtone audible to the phone that originates the call. Both SLA stations and SLA trunks are SIP channels and for outbound I have tried both the DISA() method of connecting to a trunk (get dialtone from DISA and then enter a number to call) and directly connecting (using Dial() application instead of DISA()).
> I have turned on debug and verbose in the asterisk CLI and can observe the SIP messages and dialplan execution. A SIP RINGING message is received by asterisk from the trunk, but the message is not passed on to the SLA station. I have set progressinband=yes in the sip.conf and this made no difference.
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