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Summary:ASTERISK-14507: Not passing audio on a sip call in and out on the same peer
Reporter:Mark Murawski (kobaz)Labels:
Date Opened:2009-07-21 10:07:01Date Closed:2011-06-07 14:00:56
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) xyz
( 1) xyz.debug
Description:This worked in 1.4.x, so I'm assuming this is a bug.

Call comes in from an itsp via sip.  We then proceed to dial out that same itsp (ie: call forwarding).  The remote side answers the call, but no audio is passed.

This happens on 1.6.0.10, but it's not available as a product version.

rtp packets are zero during the call.  The asterisk box is also not behind nat.

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

Dialplan
------------
context trunkhandler_voicepulse
 _X! => {
    Dial(SIP/voicepulse-primary/15180000000,45,r);
 }
}

-------------
SIP DEBUG

<--- SIP read from UDP://64.61.93.190:5060 --->
INVITE sip:15180000000@207.255.0.0:5060 SIP/2.0
Record-Route: <sip:64.61.93.190;lr=on;ftag=as41f08a9f;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA>
Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKe5af.600cd714.0
Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKe5af.d8b096f2.0
Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK3b47ef36;rport=5060
From: "Harrisburg   PA" <sip:7170000000@64.61.93.170>;tag=as41f08a9f
To: <sip:15180000000@nycinpro01.voicepulse.net>
Contact: <sip:7170000000@64.61.93.170>
Call-ID: 64799942770c09fa39cf74347be78bf7@64.61.93.170
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 68
Remote-Party-ID: "Harrisburg   PA" <sip:7170000000@64.61.93.170>;privacy=off;screen=no
Date: Tue, 21 Jul 2009 14:47:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 410

v=0
o=root 29667 29667 IN IP4 64.61.93.170
s=session
c=IN IP4 64.61.93.170
t=0 0
m=audio 12524 RTP/AVP 0 8 3 97 111 5 7 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

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

--- (18 headers 19 lines) ---

 == Using SIP RTP CoS mark 5

Sending to 64.61.93.190 : 5060 (no NAT)

Using INVITE request as basis request - 64799942770c09fa39cf74347be78bf7@64.61.93.170

No user '7170000000' in SIP users list

Found peer 'voicepulse-primary' for '7170000000' from 64.61.93.190:5060

Found RTP audio format 0

Found RTP audio format 8

Found RTP audio format 3

Found RTP audio format 97

Found RTP audio format 111

Found RTP audio format 5

Found RTP audio format 7

Found RTP audio format 101

Peer audio RTP is at port 64.61.93.170:12524

Found audio description format PCMU for ID 0

Found audio description format PCMA for ID 8

Found audio description format GSM for ID 3

Found audio description format iLBC for ID 97

Found audio description format G726-32 for ID 111

Found audio description format DVI4 for ID 5

Found audio description format LPC for ID 7

Found audio description format telephone-event for ID 101

Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xcae (gsm|ulaw|alaw|g726|adpcm|lpc10|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)

Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)

Peer audio RTP is at port 64.61.93.170:12524

Looking for 15180000000 in trunkhandler_voicepulse (domain 207.255.0.0)

list_route: hop: <sip:64.61.93.190;lr=on;ftag=as41f08a9f;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA>

bob*CLI>
<--- Transmitting (no NAT) to 64.61.93.190:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKe5af.600cd714.0;received=64.61.93.190
Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKe5af.d8b096f2.0
Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK3b47ef36;rport=5060
Record-Route: <sip:64.61.93.190;lr=on;ftag=as41f08a9f;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA>
From: "Harrisburg   PA" <sip:7170000000@64.61.93.170>;tag=as41f08a9f
To: <sip:15180000000@nycinpro01.voicepulse.net>
Call-ID: 64799942770c09fa39cf74347be78bf7@64.61.93.170
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:15180000000@207.255.0.0>
Content-Length: 0


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

   -- Executing [15180000000@trunkhandler_voicepulse:1] NoOp("SIP/So-082dc570", "CALL FROM: Harrisburg   PA <7170000000> To: 15180000000") in new stack

   -- Goto (trunkhandler_voicepulse,15180000000,4)
   -- Executing [15180000000@trunkhandler_voicepulse:1] Dial("SIP/So-082dc570", "SIP/voicepulse-primary/15180000000,45,r") in new stack
 == Using SIP RTP CoS mark 5
Audio is at 207.255.0.0 port 16600
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.61.93.190:5060:
INVITE sip:15180000000@jfk-primary.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 207.255.0.0:5060;branch=z9hG4bK4ea13c9b;rport
Max-Forwards: 70
From: "Harrisburg   PA" <sip:7170000000@207.255.0.0>;tag=as2b32b8ed
To: <sip:15180000000@jfk-primary.voicepulse.com>
Contact: <sip:7170000000@207.255.0.0>
Call-ID: 2ccfb18a78b81f21243c9919533a4493@207.255.0.0
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.10
Remote-Party-ID: "Harrisburg   PA" <sip:7170000000@207.255.0.0>;privacy=off;screen=no
Date: Tue, 21 Jul 2009 14:47:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 765619182 765619182 IN IP4 207.255.0.0
s=Asterisk PBX 1.6.0.10
c=IN IP4 207.255.0.0
t=0 0
m=audio 16600 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
   -- Called voicepulse-primary/15180000000

<--- Transmitting (no NAT) to 64.61.93.190:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKe5af.600cd714.0;received=64.61.93.190
Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKe5af.d8b096f2.0
Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK3b47ef36;rport=5060
Record-Route: <sip:64.61.93.190;lr=on;ftag=as41f08a9f;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA>
From: "Harrisburg   PA" <sip:7170000000@64.61.93.170>;tag=as41f08a9f
To: <sip:15180000000@nycinpro01.voicepulse.net>;tag=as0d7bfe5d
Call-ID: 64799942770c09fa39cf74347be78bf7@64.61.93.170
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:15180000000@207.255.0.0>
Content-Length: 0


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

<--- SIP read from UDP://64.61.93.190:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 207.255.0.0:5060;branch=z9hG4bK4ea13c9b;rport=5060
From: "Harrisburg   PA" <sip:7170000000@207.255.0.0>;tag=as2b32b8ed
To: <sip:15180000000@jfk-primary.voicepulse.com>;tag=329cfeaa6ded039da25ff8cbb8668bd2.c244
Call-ID: 2ccfb18a78b81f21243c9919533a4493@207.255.0.0
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="207.255.0.0", nonce="4a65d64787d75518a2d3d791710b591cc603248f", qop="auth"
Server: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 64.61.93.190:5060:
ACK sip:15180000000@jfk-primary.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 207.255.0.0:5060;branch=z9hG4bK4ea13c9b;rport
Max-Forwards: 70
From: "Harrisburg   PA" <sip:7170000000@207.255.0.0>;tag=as2b32b8ed
To: <sip:15180000000@jfk-primary.voicepulse.com>;tag=329cfeaa6ded039da25ff8cbb8668bd2.c244
Contact: <sip:7170000000@207.255.0.0>
Call-ID: 2ccfb18a78b81f21243c9919533a4493@207.255.0.0
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.10
Remote-Party-ID: "Harrisburg   PA" <sip:7170000000@207.255.0.0>;privacy=off;screen=no
Content-Length: 0


---
Audio is at 207.255.0.0 port 16600
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.61.93.190:5060:
INVITE sip:15180000000@jfk-primary.voicepulse.com SIP/2.0
Via: SIP/2.0/UDP 207.255.0.0:5060;branch=z9hG4bK329318c6;rport
Max-Forwards: 70
From: "Harrisburg   PA" <sip:7170000000@207.255.0.0>;tag=as2b32b8ed
To: <sip:15180000000@jfk-primary.voicepulse.com>
Contact: <sip:7170000000@207.255.0.0>
Call-ID: 2ccfb18a78b81f21243c9919533a4493@207.255.0.0
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.0.10
Remote-Party-ID: "Harrisburg   PA" <sip:7170000000@207.255.0.0>;privacy=off;screen=no
Proxy-Authorization: Digest username="So", realm="207.255.0.0", algorithm=MD5, uri="sip:15180000000@jfk-primary.voicepulse.com", nonce="4a65d64787d75518a2d3d791710b591cc603248f", response="1ee6726abea44005f6af0001c6f4eeb1", qop=auth, cnonce="15403ca5", nc=00000001
Date: Tue, 21 Jul 2009 14:47:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 765619182 765619183 IN IP4 207.255.0.0
s=Asterisk PBX 1.6.0.10
c=IN IP4 207.255.0.0
t=0 0
m=audio 16600 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP://64.61.93.190:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 207.255.0.0:5060;branch=z9hG4bK329318c6;rport=5060
From: "Harrisburg   PA" <sip:7170000000@207.255.0.0>;tag=as2b32b8ed
To: <sip:15180000000@jfk-primary.voicepulse.com>
Call-ID: 2ccfb18a78b81f21243c9919533a4493@207.255.0.0
CSeq: 103 INVITE
Server: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0


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

bob*CLI>
<--- SIP read from UDP://64.61.93.190:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 207.255.0.0:5060;branch=z9hG4bK329318c6;rport=5060
Record-Route: <sip:64.61.93.190;lr=on;ftag=as2b32b8ed>
From: "Harrisburg   PA" <sip:7170000000@207.255.0.0>;tag=as2b32b8ed
To: <sip:15180000000@jfk-primary.voicepulse.com>;tag=as5fa2cc02
Call-ID: 2ccfb18a78b81f21243c9919533a4493@207.255.0.0
CSeq: 103 INVITE
User-Agent: VoicePulse PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:+15180000000@64.61.93.189>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 18503 18503 IN IP4 64.61.93.189
s=session
c=IN IP4 64.61.93.189
t=0 0
m=audio 14978 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

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

--- (13 headers 14 lines) ---

Found RTP audio format 0

Found RTP audio format 8

Found RTP audio format 3

Found RTP audio format 101

Peer audio RTP is at port 64.61.93.189:14978

Found audio description format PCMU for ID 0

Found audio description format PCMA for ID 8

Found audio description format GSM for ID 3

Found audio description format telephone-event for ID 101

Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)

Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)

Peer audio RTP is at port 64.61.93.189:14978

   -- SIP/voicepulse-primary-0828a910 is making progress passing it to SIP/So-082dc570

   -- Refreshing DNS lookups.

bob*CLI>
<--- SIP read from UDP://64.61.93.190:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 207.255.0.0:5060;branch=z9hG4bK329318c6;rport=5060
Record-Route: <sip:64.61.93.190;lr=on;ftag=as2b32b8ed>
From: "Harrisburg   PA" <sip:7170000000@207.255.0.0>;tag=as2b32b8ed
To: <sip:15180000000@jfk-primary.voicepulse.com>;tag=as5fa2cc02
Call-ID: 2ccfb18a78b81f21243c9919533a4493@207.255.0.0
CSeq: 103 INVITE
User-Agent: VoicePulse PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:+15180000000@64.61.93.189>
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 18503 18504 IN IP4 64.61.93.189
s=session
c=IN IP4 64.61.93.189
t=0 0
m=audio 14978 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

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

--- (13 headers 14 lines) ---

Found RTP audio format 0

Found RTP audio format 8

Found RTP audio format 3

Found RTP audio format 101

Peer audio RTP is at port 64.61.93.189:14978

Found audio description format PCMU for ID 0

Found audio description format PCMA for ID 8

Found audio description format GSM for ID 3

Found audio description format telephone-event for ID 101

Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)

Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)

Peer audio RTP is at port 64.61.93.189:14978

list_route: hop: <sip:64.61.93.190;lr=on;ftag=as2b32b8ed>

set_destination: Parsing <sip:64.61.93.190;lr=on;ftag=as2b32b8ed> for address/port to send to

set_destination: set destination to 64.61.93.190, port 5060

Transmitting (no NAT) to 64.61.93.190:5060:
ACK sip:+15180000000@64.61.93.189 SIP/2.0
Via: SIP/2.0/UDP 207.255.0.0:5060;branch=z9hG4bK71db5e89;rport
Route: <sip:64.61.93.190;lr=on;ftag=as2b32b8ed>
Max-Forwards: 70
From: "Harrisburg   PA" <sip:7170000000@207.255.0.0>;tag=as2b32b8ed
To: <sip:15180000000@jfk-primary.voicepulse.com>;tag=as5fa2cc02
Contact: <sip:7170000000@207.255.0.0>
Call-ID: 2ccfb18a78b81f21243c9919533a4493@207.255.0.0
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.0.10
Remote-Party-ID: "Harrisburg   PA" <sip:7170000000@207.255.0.0>;privacy=off;screen=no
Content-Length: 0


---

   -- SIP/voicepulse-primary-0828a910 answered SIP/So-082dc570

Audio is at 207.255.0.0 port 16484

Adding codec 0x2 (gsm) to SDP

Adding codec 0x4 (ulaw) to SDP

Adding codec 0x8 (alaw) to SDP

Adding non-codec 0x1 (telephone-event) to SDP

bob*CLI>
<--- Reliably Transmitting (no NAT) to 64.61.93.190:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKe5af.600cd714.0;received=64.61.93.190
Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKe5af.d8b096f2.0
Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK3b47ef36;rport=5060
Record-Route: <sip:64.61.93.190;lr=on;ftag=as41f08a9f;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA>
From: "Harrisburg   PA" <sip:7170000000@64.61.93.170>;tag=as41f08a9f
To: <sip:15180000000@nycinpro01.voicepulse.net>;tag=as0d7bfe5d
Call-ID: 64799942770c09fa39cf74347be78bf7@64.61.93.170
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:15180000000@207.255.0.0>
Content-Type: application/sdp
Content-Length: 313

v=0
o=root 947049890 947049890 IN IP4 207.255.0.0
s=Asterisk PBX 1.6.0.10
c=IN IP4 207.255.0.0
t=0 0
m=audio 16484 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

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

   -- Packet2Packet bridging SIP/So-082dc570 and SIP/voicepulse-primary-0828a910

bob*CLI>
<--- SIP read from UDP://64.61.93.190:5060 --->
ACK sip:15180000000@207.255.0.0:5060 SIP/2.0
Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKe5af.600cd714.2
Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKe5af.d8b096f2.2
Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK4e0af965;rport=5060
From: "Harrisburg   PA" <sip:7170000000@64.61.93.170>;tag=as41f08a9f
To: <sip:15180000000@nycinpro01.voicepulse.net>;tag=as0d7bfe5d
Contact: <sip:7170000000@64.61.93.170>
Call-ID: 64799942770c09fa39cf74347be78bf7@64.61.93.170
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 68
Remote-Party-ID: "Harrisburg   PA" <sip:7170000000@64.61.93.170>;privacy=off;screen=no
Content-Length: 0
VP-hint: loose-routed


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

--- (14 headers 0 lines) ---

Really destroying SIP dialog '22529eae241c77e7143eba8d786a64eb@127.0.0.1' Method: REGISTER

bob*CLI>
<--- SIP read from UDP://64.61.93.190:5060 --->
BYE sip:15180000000@207.255.0.0:5060 SIP/2.0
Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKf5af.81f51295.0
Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKf5af.81f51295.0
Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK091e28b1;rport=5060
From: "Harrisburg   PA" <sip:7170000000@64.61.93.170>;tag=as41f08a9f
To: <sip:15180000000@nycinpro01.voicepulse.net>;tag=as0d7bfe5d
Call-ID: 64799942770c09fa39cf74347be78bf7@64.61.93.170
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 68
Remote-Party-ID: "Harrisburg   PA" <sip:7170000000@64.61.93.170>;privacy=off;screen=no
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
VP-hint: loose-routed
VP-hint: forwarded


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

--- (16 headers 0 lines) ---

Sending to 64.61.93.190 : 5060 (no NAT)

bob*CLI>
<--- Transmitting (no NAT) to 64.61.93.190:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.61.93.190;branch=z9hG4bKf5af.81f51295.0;received=64.61.93.190
Via: SIP/2.0/UDP 64.61.93.174;rport=5060;branch=z9hG4bKf5af.81f51295.0
Via: SIP/2.0/UDP 64.61.93.170:5060;received=64.61.93.170;branch=z9hG4bK091e28b1;rport=5060
From: "Harrisburg   PA" <sip:7170000000@64.61.93.170>;tag=as41f08a9f
To: <sip:15180000000@nycinpro01.voicepulse.net>;tag=as0d7bfe5d
Call-ID: 64799942770c09fa39cf74347be78bf7@64.61.93.170
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.0.10
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


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

Scheduling destruction of SIP dialog '2ccfb18a78b81f21243c9919533a4493@207.255.0.0' in 32000 ms (Method: INVITE)

set_destination: Parsing <sip:64.61.93.190;lr=on;ftag=as2b32b8ed> for address/port to send to

set_destination: set destination to 64.61.93.190, port 5060

Reliably Transmitting (no NAT) to 64.61.93.190:5060:
BYE sip:+15180000000@64.61.93.189 SIP/2.0
Via: SIP/2.0/UDP 207.255.0.0:5060;branch=z9hG4bK1309fcae;rport
Route: <sip:64.61.93.190;lr=on;ftag=as2b32b8ed>
Max-Forwards: 70
From: "Harrisburg   PA" <sip:7170000000@207.255.0.0>;tag=as2b32b8ed
To: <sip:15180000000@jfk-primary.voicepulse.com>;tag=as5fa2cc02
Call-ID: 2ccfb18a78b81f21243c9919533a4493@207.255.0.0
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.0.10
Remote-Party-ID: "Harrisburg   PA" <sip:7170000000@207.255.0.0>;privacy=off;screen=no
Proxy-Authorization: Digest username="So", realm="207.255.0.0", algorithm=MD5, uri="sip:+15180000000@64.61.93.189", nonce="4a65d64787d75518a2d3d791710b591cc603248f", response="f6e0c5c72b5465d4ea33c08787256c2e", qop=auth, cnonce="6c905bda", nc=00000002
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

 == Spawn extension (trunkhandler_voicepulse, 15180000000, 5) exited non-zero on 'SIP/So-082dc570'

Really destroying SIP dialog '64799942770c09fa39cf74347be78bf7@64.61.93.170' Method: BYE

bob*CLI>
<--- SIP read from UDP://64.61.93.190:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 207.255.0.0:5060;branch=z9hG4bK1309fcae;rport=5060
From: "Harrisburg   PA" <sip:7170000000@207.255.0.0>;tag=as2b32b8ed
To: <sip:15180000000@jfk-primary.voicepulse.com>;tag=as5fa2cc02
Call-ID: 2ccfb18a78b81f21243c9919533a4493@207.255.0.0
CSeq: 104 BYE
User-Agent: VoicePulse PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


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

--- (10 headers 0 lines) ---

Really destroying SIP dialog '2ccfb18a78b81f21243c9919533a4493@207.255.0.0' Method: INVITE

bob*CLI>

bob*CLI>

bob*CLI> sip set debug off
sip set debug off
SIP Debugging Disabled

------

SIP Peer

 * Name       : voicepulse-primary
 Secret       : <Set>
 MD5Secret    : <Not set>
 Context      : trunkhandler_voicepulse
 Subscr.Cont. : <Not set>
 Language     :
 AMA flags    : Unknown
 Transfer mode: open
 CallingPres  : Presentation Allowed, Not Screened
 Callgroup    : 0
 Pickupgroup  : 0
 Mailbox      :
 VM Extension : asterisk
 LastMsgsSent : 32767/65535
 Call limit   : 4
 Dynamic      : No
 Callerid     : "" <>
 MaxCallBR    : 384 kbps
 Expire       : -1
 Insecure     : port,invite
 Nat          : RFC3581
 ACL          : No
 T38 pt UDPTL : No
 CanReinvite  : No
 PromiscRedir : No
 User=Phone   : No
 Video Support: No
 Text Support : No
 Trust RPID   : Yes
 Send RPID    : Yes
 Subscriptions: Yes
 Overlap dial : Yes
 DTMFmode     : rfc2833
 Timer T1     : 500
 Timer B      : 32000
 ToHost       : jfk-primary.voicepulse.com
 Addr->IP     : 64.61.93.190 Port 5060
 Defaddr->IP  : 0.0.0.0 Port 0
 Transport    : UDP
 Def. Username: So
 SIP Options  : replaces replace
 Codecs       : 0x8000e (gsm|ulaw|alaw|h263)
 Codec Order  : (none)
 Auto-Framing :  No
 100 on REG   : No
 Status       : Unmonitored
 Useragent    :
 Reg. Contact :
 Qualify Freq : 60000 ms
 Sess-Timers  : Accept
 Sess-Refresh : uas
 Sess-Expires : 1800 secs
 Min-Sess     : 90 secs
Comments:By: Leif Madsen (lmadsen) 2009-09-18 07:48:41

Are you still getting this issue with 1.6.0.16-rc1?

If so, could you provide the same output, but with the 'sip history' also enabled, along with any console output, and 'core set debug 10' enabled after turning it on in logger.conf (console => notice,warning,error,debug), followed by a 'logger reload'

Thanks!
By: Mark Murawski (kobaz) 2009-09-18 12:09:32

still not working on 1.6.0.16-rc1.. see attached new log
By: Mark Murawski (kobaz) 2009-09-18 12:16:15

whoops, didn't have debug on... uploaded debug log
By: Paul Belanger (pabelanger) 2010-06-02 13:40:28

Is this still an issue using 1.6.2?  Also, try using Progress() in you dialplan.

---
Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.6.0 and 1.6.1 branches has ended. For continued maintenance support please move to the 1.6.2 branch.

More information on this change can be found in the release announcement: http://www.asterisk.org/node/49924

By: Mark Murawski (kobaz) 2010-06-06 15:50:51

I'll test with 1.6.2
By: Paul Belanger (pabelanger) 2010-06-18 10:01:55

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.

Further information can be found at http://www.asterisk.org/developers/bug-guidelines