X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode:0

rbreidenstein · November 25, 2009, 3:10pm

X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0

I am currently having an issue where whenever the phone system makes an outbound call it will connect for what appears to be < 1 second (you can hear the other side for this breif period) and then the call disconnects and sends the CSeq: 104 BYE with the abox HangupCause. The strange thing is that I can make a call to a googlevoice number and the call works fine; but if I make the call to a direct cell phone or landline the call performs as stated above. I have talked to my SIP provider (flowroute) and they stated, as i have seen in my debugging as well, that the BYE is initiated from my asterisk server.

Is there any more debugging that I can do to help determine what maybe causing the issue?

Any thoughts and suggestions are welcome.

rbreidenstein · November 25, 2009, 4:42pm

Below is the SIP history of the call I will work on posting debug info as well.

NewChan Channel SIP/provider-name-here-000000a4 - from 7348559XXXXXXXXXX7a21cbf07
TxReqRel INVITE / 102 INVITE - INVITE
Rx SIP/2.0 / 102 INVITE / 100 Trying
Rx SIP/2.0 / 102 INVITE / 407 Proxy Authentication Required
TxReq ACK / 102 ACK - ACK
AuthResp Auth response sent for XXXXXX in realm sip.provider-name-here.com - nc
TxReqRel INVITE / 103 INVITE - INVITE
Rx SIP/2.0 / 103 INVITE / 100 Trying
Rx SIP/2.0 / 103 INVITE / 183 Session Progress
Rx SIP/2.0 / 103 INVITE / 180 Ringing
Rx SIP/2.0 / 103 INVITE / 200 OK
TxReq ACK / 103 ACK - ACK
AuthResp Auth response sent for XXXXX in realm sip.provider=name-here.com - nc
TxReqRel BYE / 104 BYE - BYE
SchedDestroy 32000 ms
Rx SIP/2.0 / 104 BYE / 200 OK
NeedDestroy Setting needdestroy because received 200 response

rbreidenstein · November 25, 2009, 5:00pm



<--- SIP read from UDP:CALLING-PHONE-PUBLIC-IP:21387 --->
INVITE sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP:5060 SIP/2.0
Via: SIP/2.0/UDP CALLING-PHONE-PUBLIC-IP:21387;branch=z9hG4bKdb900db96a753e6c3.45e8dcf2f332c44e3
Max-Forwards: 70
From: "CALLING-PHONES-DID" <sip:CALLING-PHONES-EXTENSION@ASTERISK-INTERNAL-IP:5060>;tag=c75ff149b5
To: "DIALED-NUMBER" <sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP:5060>
Call-ID: 9037d4b679fdffea
CSeq: 10630 INVITE
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "CALLING-PHONES-DID" <sip:CALLING-PHONES-EXTENSION@CALLING-PHONE-PUBLIC-IP:21387;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D20E58D>"
Supported: gruu, path, timer, 100rel, replaces
User-Agent: Aastra 6730i/2.5.0.82
Content-Type: application/sdp
Content-Length: 675

v=0
o=MxSIP 0 0 IN IP4 CALLING-PHONE-PUBLIC-IP
s=SIP Call
c=IN IP4 CALLING-PHONE-PUBLIC-IP
t=0 0
m=audio 22156 RTP/AVP 0 18 106 107 113 110 111 112 4 4 98 97 115 96 9 108 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:113 L16/16000
a=rtpmap:110 PCMU/16000
a=rtpmap:111 PCMA/16000
a=rtpmap:112 L16/8000
a=rtpmap:4 G723/8000
a=rtpmap:4 G723/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:9 G722/8000
a=rtpmap:108 G7221/16000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (14 headers 28 lines) ---
  == Using SIP RTP CoS mark 5
Sending to CALLING-PHONE-PUBLIC-IP : 21387 (no NAT)
Using INVITE request as basis request - 9037d4b679fdffea
Found peer 'CALLING-PHONES-EXTENSION' for 'CALLING-PHONES-EXTENSION' from CALLING-PHONE-PUBLIC-IP:21387

<--- Reliably Transmitting (NAT) to CALLING-PHONE-PUBLIC-IP:21387 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP CALLING-PHONE-PUBLIC-IP:21387;branch=z9hG4bKdb900db96a753e6c3.45e8dcf2f332c44e3;received=CALLING-PHONE-PUBLIC-IP
From: "CALLING-PHONES-DID" <sip:CALLING-PHONES-EXTENSION@ASTERISK-INTERNAL-IP:5060>;tag=c75ff149b5
To: "DIALED-NUMBER" <sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP:5060>;tag=as777fc4c9
Call-ID: 9037d4b679fdffea
CSeq: 10630 INVITE
Server: Asterisk PBX 1.6.2.0-rc6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="378cec73"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '9037d4b679fdffea' in 6400 ms (Method: INVITE)
e[KASTERISK-SERVERNAME*CLI> 
<--- SIP read from UDP:CALLING-PHONE-PUBLIC-IP:21387 --->
ACK sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP:5060 SIP/2.0
Via: SIP/2.0/UDP CALLING-PHONE-PUBLIC-IP:21387;branch=z9hG4bKdb900db96a753e6c3.45e8dcf2f332c44e3
Max-Forwards: 70
From: "CALLING-PHONES-DID" <sip:CALLING-PHONES-EXTENSION@ASTERISK-INTERNAL-IP:5060>;tag=c75ff149b5
To: "DIALED-NUMBER" <sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP:5060>;tag=as777fc4c9
Call-ID: 9037d4b679fdffea
CSeq: 10630 ACK
User-Agent: Aastra 6730i/2.5.0.82
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
e[KASTERISK-SERVERNAME*CLI> 
<--- SIP read from UDP:CALLING-PHONE-PUBLIC-IP:21387 --->
INVITE sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP:5060 SIP/2.0
Via: SIP/2.0/UDP CALLING-PHONES-INTERNAL-IP:5060;branch=z9hG4bKcd646e02266a8a15d.b170f36132ed0399a
Max-Forwards: 70
From: "CALLING-PHONES-DID" <sip:CALLING-PHONES-EXTENSION@ASTERISK-INTERNAL-IP:5060>;tag=c75ff149b5
To: "DIALED-NUMBER" <sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP:5060>
Call-ID: 9037d4b679fdffea
CSeq: 10631 INVITE
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest username="CALLING-PHONES-EXTENSION",realm="asterisk",nonce="378cec73",uri="sip:DIALED-NUMBER@ASTERISK-EXTERNAL-IP:5060",response="f88ae1f9b363dd4ff309b8057948ed09",algorithm=MD5
Contact: "CALLING-PHONES-DID" <sip:CALLING-PHONES-EXTENSION@CALLING-PHONES-INTERNAL-IP:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D20E58D>"
Supported: gruu, path, timer, 100rel, replaces
User-Agent: Aastra 6730i/2.5.0.82
Content-Type: application/sdp
Content-Length: 670

v=0
o=MxSIP 0 0 IN IP4 CALLING-PHONES-INTERNAL-IP
s=SIP Call
c=IN IP4 CALLING-PHONES-INTERNAL-IP
t=0 0
m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 4 4 98 97 115 96 9 108 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:113 L16/16000
a=rtpmap:110 PCMU/16000
a=rtpmap:111 PCMA/16000
a=rtpmap:112 L16/8000
a=rtpmap:4 G723/8000
a=rtpmap:4 G723/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:9 G722/8000
a=rtpmap:108 G7221/16000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
--- (15 headers 28 lines) ---
Sending to CALLING-PHONE-PUBLIC-IP : 21387 (NAT)
Using INVITE request as basis request - 9037d4b679fdffea
Found peer 'CALLING-PHONES-EXTENSION' for 'CALLING-PHONES-EXTENSION' from CALLING-PHONE-PUBLIC-IP:21387
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 106
e[KASTERISK-SERVERNAME*CLI> 
Found RTP audio format 107
e[KASTERISK-SERVERNAME*CLI> 
Found RTP audio format 113
Found RTP audio format 110
e[KASTERISK-SERVERNAME*CLI> 
Found RTP audio format 111
Found RTP audio format 112
e[KASTERISK-SERVERNAME*CLI> 
Found RTP audio format 4
Found RTP audio format 4
e[KASTERISK-SERVERNAME*CLI> 
Found RTP audio format 98
Found RTP audio format 97
e[KASTERISK-SERVERNAME*CLI> 
Found RTP audio format 115
Found RTP audio format 96
e[KASTERISK-SERVERNAME*CLI> 
Found RTP audio format 9
Found RTP audio format 108
e[KASTERISK-SERVERNAME*CLI> 
Found RTP audio format 8
Found RTP audio format 101
e[KASTERISK-SERVERNAME*CLI> 
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
e[KASTERISK-SERVERNAME*CLI> 
Found audio description format BV16 for ID 106
Found audio description format BV32 for ID 107
e[KASTERISK-SERVERNAME*CLI> 
Found audio description format L16 for ID 113
Found audio description format PCMU for ID 110
e[KASTERISK-SERVERNAME*CLI> 
Found audio description format PCMA for ID 111
Found audio description format L16 for ID 112
e[KASTERISK-SERVERNAME*CLI> 
Found audio description format G723 for ID 4
Found audio description format G723 for ID 4
e[KASTERISK-SERVERNAME*CLI> 
Found audio description format G726-16 for ID 98
Found audio description format G726-24 for ID 97
e[KASTERISK-SERVERNAME*CLI> 
Found audio description format G726-32 for ID 115
Found audio description format G726-40 for ID 96
Found audio description format G722 for ID 9
Found audio description format G7221 for ID 108
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x8103f4d (g723|ulaw|alaw|g726|slin|g729|speex|ilbc|g722|h263p|t140|siren7)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port CALLING-PHONES-INTERNAL-IP:3000
Looking for DIALED-NUMBER in 3701 (domain ASTERISK-INTERNAL-IP)
list_route: hop: <sip:CALLING-PHONES-EXTENSION@CALLING-PHONES-INTERNAL-IP:5060;transport=udp>

<--- Transmitting (NAT) to CALLING-PHONE-PUBLIC-IP:21387 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP CALLING-PHONES-INTERNAL-IP:5060;branch=z9hG4bKcd646e02266a8a15d.b170f36132ed0399a;received=CALLING-PHONE-PUBLIC-IP
From: "CALLING-PHONES-DID" <sip:CALLING-PHONES-EXTENSION@ASTERISK-INTERNAL-IP:5060>;tag=c75ff149b5
To: "DIALED-NUMBER" <sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP:5060>
Call-ID: 9037d4b679fdffea
CSeq: 10631 INVITE
Server: Asterisk PBX 1.6.2.0-rc6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP>
Content-Length: 0


<------------>
    -- Executing [DIALED-NUMBER@3701:1] e[1me[36mDiale[0m("e[1me[35mSIP/CALLING-PHONES-EXTENSION-000000ade[0m", "e[1me[35mSIP/DIALED-NUMBER@SERVICE-PROVIDER-NAMEe[0m") in new stack
e[KASTERISK-SERVERNAME*CLI> 
  == Using SIP RTP CoS mark 5
e[KASTERISK-SERVERNAME*CLI> 
    -- Called DIALED-NUMBER@SERVICE-PROVIDER-NAME
e[KASTERISK-SERVERNAME*CLI> 
    -- SIP/SERVICE-PROVIDER-NAME-000000ae is making progress passing it to SIP/CALLING-PHONES-EXTENSION-000000ad
Audio is at ASTERISK-INTERNAL-IP port 14528
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to CALLING-PHONE-PUBLIC-IP:21387 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP CALLING-PHONES-INTERNAL-IP:5060;branch=z9hG4bKcd646e02266a8a15d.b170f36132ed0399a;received=CALLING-PHONE-PUBLIC-IP
From: "CALLING-PHONES-DID" <sip:CALLING-PHONES-EXTENSION@ASTERISK-INTERNAL-IP:5060>;tag=c75ff149b5
To: "DIALED-NUMBER" <sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP:5060>;tag=as250ce587
Call-ID: 9037d4b679fdffea
CSeq: 10631 INVITE
Server: Asterisk PBX 1.6.2.0-rc6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 1351868283 1351868283 IN IP4 ASTERISK-INTERNAL-IP
s=Asterisk PBX 1.6.2.0-rc6
c=IN IP4 ASTERISK-INTERNAL-IP
t=0 0
m=audio 14528 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
e[KASTERISK-SERVERNAME*CLI> 
    -- SIP/SERVICE-PROVIDER-NAME-000000ae is ringing

<--- Transmitting (NAT) to CALLING-PHONE-PUBLIC-IP:21387 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP CALLING-PHONES-INTERNAL-IP:5060;branch=z9hG4bKcd646e02266a8a15d.b170f36132ed0399a;received=CALLING-PHONE-PUBLIC-IP
From: "CALLING-PHONES-DID" <sip:CALLING-PHONES-EXTENSION@ASTERISK-INTERNAL-IP:5060>;tag=c75ff149b5
To: "DIALED-NUMBER" <sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP:5060>;tag=as250ce587
Call-ID: 9037d4b679fdffea
CSeq: 10631 INVITE
Server: Asterisk PBX 1.6.2.0-rc6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP>
Content-Length: 0


<------------>
    -- SIP/SERVICE-PROVIDER-NAME-000000ae is making progress passing it to SIP/CALLING-PHONES-EXTENSION-000000ad
e[KASTERISK-SERVERNAME*CLI> 
    -- SIP/SERVICE-PROVIDER-NAME-000000ae answered SIP/CALLING-PHONES-EXTENSION-000000ad
Audio is at ASTERISK-INTERNAL-IP port 14528
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to CALLING-PHONE-PUBLIC-IP:21387 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP CALLING-PHONES-INTERNAL-IP:5060;branch=z9hG4bKcd646e02266a8a15d.b170f36132ed0399a;received=CALLING-PHONE-PUBLIC-IP
From: "CALLING-PHONES-DID" <sip:CALLING-PHONES-EXTENSION@ASTERISK-INTERNAL-IP:5060>;tag=c75ff149b5
To: "DIALED-NUMBER" <sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP:5060>;tag=as250ce587
Call-ID: 9037d4b679fdffea
CSeq: 10631 INVITE
Server: Asterisk PBX 1.6.2.0-rc6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP>
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 1351868283 1351868284 IN IP4 ASTERISK-INTERNAL-IP
s=Asterisk PBX 1.6.2.0-rc6
c=IN IP4 ASTERISK-INTERNAL-IP
t=0 0
m=audio 14528 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
    -- Packet2Packet bridging SIP/CALLING-PHONES-EXTENSION-000000ad and SIP/SERVICE-PROVIDER-NAME-000000ae
e[KASTERISK-SERVERNAME*CLI> 
<--- SIP read from UDP:CALLING-PHONE-PUBLIC-IP:21387 --->
ACK sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP SIP/2.0
Via: SIP/2.0/UDP CALLING-PHONE-PUBLIC-IP:21387;branch=z9hG4bK9134d90f25931b51c.88518cb1fab48b586
Max-Forwards: 70
From: "CALLING-PHONES-DID" <sip:CALLING-PHONES-EXTENSION@ASTERISK-INTERNAL-IP:5060>;tag=c75ff149b5
To: "DIALED-NUMBER" <sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP:5060>;tag=as250ce587
Call-ID: 9037d4b679fdffea
CSeq: 10631 ACK
Authorization: Digest username="CALLING-PHONES-EXTENSION",realm="asterisk",nonce="378cec73",uri="sip:DIALED-NUMBER@ASTERISK-EXTERNAL-IP:5060",response="f88ae1f9b363dd4ff309b8057948ed09",algorithm=MD5
User-Agent: Aastra 6730i/2.5.0.82
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
e[KASTERISK-SERVERNAME*CLI> 
Really destroying SIP dialog '504d9983781103ec' Method: REGISTER
e[KASTERISK-SERVERNAME*CLI> 
<--- SIP read from UDP:CALLING-PHONE-PUBLIC-IP:21387 --->
BYE sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP SIP/2.0
Via: SIP/2.0/UDP CALLING-PHONE-PUBLIC-IP:21387;branch=z9hG4bKaafc36b1bb91b9adc.f1862b2dd11bdc059
Max-Forwards: 70
From: "CALLING-PHONES-DID" <sip:CALLING-PHONES-EXTENSION@ASTERISK-INTERNAL-IP:5060>;tag=c75ff149b5
To: "DIALED-NUMBER" <sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP:5060>;tag=as250ce587
Call-ID: 9037d4b679fdffea
CSeq: 10632 BYE
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest username="CALLING-PHONES-EXTENSION",realm="asterisk",nonce="378cec73",uri="sip:DIALED-NUMBER@ASTERISK-EXTERNAL-IP",response="83aec3da389afe2f932c45debb4a75f8",algorithm=MD5
Supported: gruu, path, timer
User-Agent: Aastra 6730i/2.5.0.82
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Sending to CALLING-PHONE-PUBLIC-IP : 21387 (NAT)

<--- Transmitting (NAT) to CALLING-PHONE-PUBLIC-IP:21387 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP CALLING-PHONE-PUBLIC-IP:21387;branch=z9hG4bKaafc36b1bb91b9adc.f1862b2dd11bdc059;received=CALLING-PHONE-PUBLIC-IP
From: "CALLING-PHONES-DID" <sip:CALLING-PHONES-EXTENSION@ASTERISK-INTERNAL-IP:5060>;tag=c75ff149b5
To: "DIALED-NUMBER" <sip:DIALED-NUMBER@ASTERISK-INTERNAL-IP:5060>;tag=as250ce587
Call-ID: 9037d4b679fdffea
CSeq: 10632 BYE
Server: Asterisk PBX 1.6.2.0-rc6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
e[KASTERISK-SERVERNAME*CLI> 
  == Spawn extension (3701, DIALED-NUMBER, 1) exited non-zero on 'SIP/CALLING-PHONES-EXTENSION-000000ad'
e[KASTERISK-SERVERNAME*CLI> 
Really destroying SIP dialog '9037d4b679fdffea' Method: BYE
e[KASTERISK-SERVERNAME*CLI>

system · January 21, 2010, 5:39pm

rbreidenstein, I just posted here on the exact same problem: viewtopic.php?f=1&t=72893

Flowroute seems convinced that the issue is on my end…i’m not convinced given all of my other VoIP connections work without issue.

Did you ever able to find a solution?

HC

rbreidenstein · April 14, 2010, 9:27pm

I ended up recompiling asterisk. That worked for some reason.

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X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode:0

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