I have an HT-286 connected to an analog conference phone. Internal calls between extensions work just fine, but outgoing calls through our SIP trunk provider result in one way audio (outgoing audio is passed through, but incoming audio doesn’t make it to the phone). All other handsets in our office (combo of Snom 320s and Cisco SPA504Gs) work perfectly fine through the SIP trunk.
I wouldn’t think this is a NAT issue since the phones are talking to a server on our local network and shouldn’t be directly communicating through our firewall, unless I’m missing something?
sip.conf (setting is identical for all other handsets):
[Conference]
disallow=all
canreinvite=no
nat=no
allow=ulaw
allow=alaw
callerid="Conference Room" <220>
secret=MY PASSWORD
context = internal
type=friend
host=dynamic
subscribecontext=internalHere’s the sip trace on a failed call (call connects, but I can’t hear callee caller/callee phone numbers replaced with “NUMBER_I_AM_CALLING” and “MY_PHONE_NUMBER”):
<--- SIP read from UDP:10.2.1.16:5060 --->
INVITE sip:NUMBER_I_AM_CALLING@10.2.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.16;branch=z9hG4bK738bce32d370f062
From: "Conference Room" <sip:Conference@10.2.0.10>;tag=24e34f24310cad94
To: <sip:NUMBER_I_AM_CALLING@10.2.0.10>
Contact: <sip:Conference@10.2.1.16>
Supported: replaces, timer
Call-ID: 273b6b0b6aeb85b4@10.2.1.16
CSeq: 34108 INVITE
User-Agent: Grandstream HT287 1.1.0.42 DevId 000b82334458
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Type: application/sdp
Content-Length: 378
v=0
o=Conference 8000 8000 IN IP4 10.2.1.16
s=SIP Call
c=IN IP4 10.2.1.16
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 97 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (13 headers 18 lines) ---
Sending to 10.2.1.16:5060 (no NAT)
Using INVITE request as basis request - 273b6b0b6aeb85b4@10.2.1.16
Found peer 'Conference' for 'Conference' from 10.2.1.16:5060
<--- Reliably Transmitting (no NAT) to 10.2.1.16:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.2.1.16;branch=z9hG4bK738bce32d370f062;received=10.2.1.16
From: "Conference Room" <sip:Conference@10.2.0.10>;tag=24e34f24310cad94
To: <sip:NUMBER_I_AM_CALLING@10.2.0.10>;tag=as3ec18f36
Call-ID: 273b6b0b6aeb85b4@10.2.1.16
CSeq: 34108 INVITE
Server: Asterisk PBX 1.8.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e92f37e"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '273b6b0b6aeb85b4@10.2.1.16' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:10.2.1.16:5060 --->
ACK sip:NUMBER_I_AM_CALLING@10.2.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.16;branch=z9hG4bK738bce32d370f062
From: "Conference Room" <sip:Conference@10.2.0.10>;tag=24e34f24310cad94
To: <sip:NUMBER_I_AM_CALLING@10.2.0.10>;tag=as3ec18f36
Contact: <sip:Conference@10.2.1.16>
Call-ID: 273b6b0b6aeb85b4@10.2.1.16
CSeq: 34108 ACK
User-Agent: Grandstream HT287 1.1.0.42 DevId 000b82334458
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from UDP:10.2.1.16:5060 --->
INVITE sip:NUMBER_I_AM_CALLING@10.2.0.10 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.16;branch=z9hG4bKf33456146bdcff6b
From: "Conference Room" <sip:Conference@10.2.0.10>;tag=24e34f24310cad94
To: <sip:NUMBER_I_AM_CALLING@10.2.0.10>
Contact: <sip:Conference@10.2.1.16>
Supported: replaces, timer
Authorization: Digest username="Conference", realm="asterisk", algorithm=MD5, uri="sip:NUMBER_I_AM_CALLING@10.2.0.10", nonce="3e92f37e", response="1c05616d7ddf8ffa698863560e1f3e9d"
Call-ID: 273b6b0b6aeb85b4@10.2.1.16
CSeq: 34109 INVITE
User-Agent: Grandstream HT287 1.1.0.42 DevId 000b82334458
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Type: application/sdp
Content-Length: 378
v=0
o=Conference 8000 8001 IN IP4 10.2.1.16
s=SIP Call
c=IN IP4 10.2.1.16
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 97 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (14 headers 18 lines) ---
Sending to 10.2.1.16:5060 (no NAT)
Using INVITE request as basis request - 273b6b0b6aeb85b4@10.2.1.16
Found peer 'Conference' for 'Conference' from 10.2.1.16:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 97
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 2
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.2.1.16:5004
Looking for NUMBER_I_AM_CALLING in internal (domain 10.2.0.10)
list_route: hop: <sip:Conference@10.2.1.16>
<--- Transmitting (no NAT) to 10.2.1.16:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.1.16;branch=z9hG4bKf33456146bdcff6b;received=10.2.1.16
From: "Conference Room" <sip:Conference@10.2.0.10>;tag=24e34f24310cad94
To: <sip:NUMBER_I_AM_CALLING@10.2.0.10>
Call-ID: 273b6b0b6aeb85b4@10.2.1.16
CSeq: 34109 INVITE
Server: Asterisk PBX 1.8.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:NUMBER_I_AM_CALLING@10.2.0.10:5060>
Content-Length: 0
<------------>
-- Executing [NUMBER_I_AM_CALLING@internal:1] Set("SIP/Conference-00000011", "DYNAMIC_FEATURES=automon") in new stack
-- Executing [NUMBER_I_AM_CALLING@internal:2] Dial("SIP/Conference-00000011", "SIP/sip.integra.net-outgoing/NUMBER_I_AM_CALLING,60,KXx") in new stack
== Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 67.137.39.100:5060:
INVITE sip:NUMBER_I_AM_CALLING@ptldor17.orv.integra.net:5060 SIP/2.0
Via: SIP/2.0/UDP 209.210.16.218:5060;branch=z9hG4bK7530ced1
Max-Forwards: 70
From: "Conference Room" <sip:MY_PHONE_NUMBER@ptldor17.orv.integra.net>;tag=as2fdcf3bb
To: <sip:NUMBER_I_AM_CALLING@ptldor17.orv.integra.net:5060>
Contact: <sip:MY_PHONE_NUMBER@209.210.16.218:5060>
Call-ID: 596755814925b3a3217a7a5021cac562@ptldor17.orv.integra.net
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.8.0
Date: Fri, 06 Jan 2012 19:23:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 267
v=0
o=root 1068668542 1068668542 IN IP4 209.210.16.218
s=Asterisk PBX 1.8.8.0
c=IN IP4 209.210.16.218
t=0 0
m=audio 12012 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
---
-- Called SIP/sip.integra.net-outgoing/NUMBER_I_AM_CALLING
<--- SIP read from UDP:67.137.39.100:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 209.210.16.218:5060;branch=z9hG4bK7530ced1
From: "Conference Room" <sip:MY_PHONE_NUMBER@67.137.39.100>;tag=as2fdcf3bb
To: <sip:NUMBER_I_AM_CALLING@ptldor17.orv.integra.net:5060>
Call-ID: 596755814925b3a3217a7a5021cac562@ptldor17.orv.integra.net
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
<--- SIP read from UDP:67.137.39.100:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.210.16.218:5060;branch=z9hG4bK7530ced1
From: "Conference Room" <sip:MY_PHONE_NUMBER@67.137.39.100>;tag=as2fdcf3bb
To: <sip:NUMBER_I_AM_CALLING@ptldor17.orv.integra.net:5060>;tag=ptleorteca0.orv.integra.voip+1+45a625+74bb8f9e
Call-ID: 596755814925b3a3217a7a5021cac562@ptldor17.orv.integra.net
CSeq: 102 INVITE
WWW-Authenticate: Digest realm=" ptldor17.orv.integra.net",nonce="75f434f81ba5",stale=false,algorithm=MD5,qop="auth"
Server: DC-SIP/2.0
Organization: MetaSwitch
Supported: 100rel, resource-priority
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 67.137.39.100:5060:
ACK sip:NUMBER_I_AM_CALLING@ptldor17.orv.integra.net:5060 SIP/2.0
Via: SIP/2.0/UDP 209.210.16.218:5060;branch=z9hG4bK7530ced1
Max-Forwards: 70
From: "Conference Room" <sip:MY_PHONE_NUMBER@ptldor17.orv.integra.net>;tag=as2fdcf3bb
To: <sip:NUMBER_I_AM_CALLING@ptldor17.orv.integra.net:5060>;tag=ptleorteca0.orv.integra.voip+1+45a625+74bb8f9e
Contact: <sip:MY_PHONE_NUMBER@209.210.16.218:5060>
Call-ID: 596755814925b3a3217a7a5021cac562@ptldor17.orv.integra.net
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.8.0
Content-Length: 0
---
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 67.137.39.100:5060:
INVITE sip:NUMBER_I_AM_CALLING@ptldor17.orv.integra.net:5060 SIP/2.0
Via: SIP/2.0/UDP 209.210.16.218:5060;branch=z9hG4bK4880cd77
Max-Forwards: 70
From: "Conference Room" <sip:MY_PHONE_NUMBER@ptldor17.orv.integra.net>;tag=as2fdcf3bb
To: <sip:NUMBER_I_AM_CALLING@ptldor17.orv.integra.net:5060>
Contact: <sip:MY_PHONE_NUMBER@209.210.16.218:5060>
Call-ID: 596755814925b3a3217a7a5021cac562@ptldor17.orv.integra.net
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.8.0
Authorization: Digest username="MY_PHONE_NUMBER", realm=" ptldor17.orv.integra.net", algorithm=MD5, uri="sip:NUMBER_I_AM_CALLING@ptldor17.orv.integra.net:5060", nonce="75f434f81ba5", response="e65339f402f1d62d6de4f63d3a443c69", qop=auth, cnonce="03269289", nc=00000001
Date: Fri, 06 Jan 2012 19:23:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 267
v=0
o=root 1068668542 1068668543 IN IP4 209.210.16.218
s=Asterisk PBX 1.8.8.0
c=IN IP4 209.210.16.218
t=0 0
m=audio 12012 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
---
<--- SIP read from UDP:67.137.39.100:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 209.210.16.218:5060;branch=z9hG4bK4880cd77
From: "Conference Room" <sip:MY_PHONE_NUMBER@67.137.39.100>;tag=as2fdcf3bb
To: <sip:NUMBER_I_AM_CALLING@ptldor17.orv.integra.net:5060>
Call-ID: 596755814925b3a3217a7a5021cac562@ptldor17.orv.integra.net
CSeq: 103 INVITE
<------------->
--- (6 headers 0 lines) ---
<--- SIP read from UDP:67.137.39.100:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 209.210.16.218:5060;branch=z9hG4bK4880cd77
From: "Conference Room" <sip:MY_PHONE_NUMBER@67.137.39.100>;tag=as2fdcf3bb
To: <sip:NUMBER_I_AM_CALLING@ptldor17.orv.integra.net:5060>;tag=ptleorteca0.orv.integra.voip+1+455a39+104acdc3
Call-ID: 596755814925b3a3217a7a5021cac562@ptldor17.orv.integra.net
CSeq: 103 INVITE
Server: DC-SIP/2.0
Organization: MetaSwitch
Contact: <sip:NUMBER_I_AM_CALLING@67.137.39.100:5060;transport=udp>
Content-Length: 153
Content-Type: application/sdp
v=0
o=- 3027881742 3027881742 IN IP4 67.137.39.100
s=-
c=IN IP4 67.137.39.100
t=0 0
m=audio 29864 RTP/AVP 0
a=ptime:20
a=silenceSupp:off - - - -
<------------->
--- (11 headers 8 lines) ---
Found RTP audio format 0
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 67.137.39.100:29864
-- SIP/sip.integra.net-outgoing-00000012 is making progress passing it to SIP/Conference-00000011
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (no NAT) to 10.2.1.16:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.2.1.16;branch=z9hG4bKf33456146bdcff6b;received=10.2.1.16
From: "Conference Room" <sip:Conference@10.2.0.10>;tag=24e34f24310cad94
To: <sip:NUMBER_I_AM_CALLING@10.2.0.10>;tag=as1b98070f
Call-ID: 273b6b0b6aeb85b4@10.2.1.16
CSeq: 34109 INVITE
Server: Asterisk PBX 1.8.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:NUMBER_I_AM_CALLING@10.2.0.10:5060>
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 2003243540 2003243540 IN IP4 10.2.0.10
s=Asterisk PBX 1.8.8.0
c=IN IP4 10.2.0.10
t=0 0
m=audio 15622 RTP/AVP 0 8 101
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
<------------>
Really destroying SIP dialog '3BB039F9@ptleorteca0.orv.integra.voip' Method: OPTIONS
<--- SIP read from UDP:67.137.39.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.210.16.218:5060;branch=z9hG4bK4880cd77
From: "Conference Room" <sip:MY_PHONE_NUMBER@67.137.39.100>;tag=as2fdcf3bb
To: <sip:NUMBER_I_AM_CALLING@ptldor17.orv.integra.net:5060>;tag=ptleorteca0.orv.integra.voip+1+455a39+104acdc3
Call-ID: 596755814925b3a3217a7a5021cac562@ptldor17.orv.integra.net
CSeq: 103 INVITE
Server: DC-SIP/2.0
Organization: MetaSwitch
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event
Supported: 100rel, resource-priority
Allow: INVITE, ACK, CANCEL, BYE, REGISTER, OPTIONS, PRACK, UPDATE, SUBSCRIBE, NOTIFY, REFER, INFO, PUBLISH
Accept-Encoding: identity
Accept: application/sdp, application/simple-message-summary, message/sipfrag, application/isup, application/x-simple-call-service-info, multipart/mixed, application/broadsoft, application/vq-rtcpxr, application/media_control+xml, application/dtmf-relay, text/plain, application/x-as-feature-event+xml
Contact: <sip:NUMBER_I_AM_CALLING@67.137.39.100:5060;transport=udp>
Content-Length: 153
Content-Type: application/sdp
v=0
o=- 3027881742 3027881742 IN IP4 67.137.39.100
s=-
c=IN IP4 67.137.39.100
t=0 0
m=audio 29864 RTP/AVP 0
a=ptime:20
a=silenceSupp:off - - - -
<------------->
--- (16 headers 8 lines) ---
list_route: hop: <sip:NUMBER_I_AM_CALLING@67.137.39.100:5060;transport=udp>
set_destination: Parsing <sip:NUMBER_I_AM_CALLING@67.137.39.100:5060;transport=udp> for address/port to send to
set_destination: set destination to 67.137.39.100:5060
Transmitting (no NAT) to 67.137.39.100:5060:
ACK sip:NUMBER_I_AM_CALLING@67.137.39.100:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 209.210.16.218:5060;branch=z9hG4bK74068aad
Max-Forwards: 70
From: "Conference Room" <sip:MY_PHONE_NUMBER@ptldor17.orv.integra.net>;tag=as2fdcf3bb
To: <sip:NUMBER_I_AM_CALLING@ptldor17.orv.integra.net:5060>;tag=ptleorteca0.orv.integra.voip+1+455a39+104acdc3
Contact: <sip:MY_PHONE_NUMBER@209.210.16.218:5060>
Call-ID: 596755814925b3a3217a7a5021cac562@ptldor17.orv.integra.net
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.8.0
Content-Length: 0
---
-- SIP/sip.integra.net-outgoing-00000012 answered SIP/Conference-00000011
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 10.2.1.16:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.1.16;branch=z9hG4bKf33456146bdcff6b;received=10.2.1.16
From: "Conference Room" <sip:Conference@10.2.0.10>;tag=24e34f24310cad94
To: <sip:NUMBER_I_AM_CALLING@10.2.0.10>;tag=as1b98070f
Call-ID: 273b6b0b6aeb85b4@10.2.1.16
CSeq: 34109 INVITE
Server: Asterisk PBX 1.8.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:NUMBER_I_AM_CALLING@10.2.0.10:5060>
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 2003243540 2003243541 IN IP4 10.2.0.10
s=Asterisk PBX 1.8.8.0
c=IN IP4 10.2.0.10
t=0 0
m=audio 15622 RTP/AVP 0 8 101
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
<------------>
<--- SIP read from UDP:10.2.1.16:5060 --->
ACK sip:NUMBER_I_AM_CALLING@10.2.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 10.2.1.16;branch=z9hG4bK8982617b5c704a50
From: "Conference Room" <sip:Conference@10.2.0.10>;tag=24e34f24310cad94
To: <sip:NUMBER_I_AM_CALLING@10.2.0.10>;tag=as1b98070f
Contact: <sip:Conference@10.2.1.16>
Authorization: Digest username="Conference", realm="asterisk", algorithm=MD5, uri="sip:NUMBER_I_AM_CALLING@10.2.0.10:5060", nonce="3e92f37e", response="95512a29ce4ac027f308e1f0aba72e9a"
Call-ID: 273b6b0b6aeb85b4@10.2.1.16
CSeq: 34109 ACK
User-Agent: Grandstream HT287 1.1.0.42 DevId 000b82334458
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
<--- SIP read from UDP:67.137.39.100:5060 --->
BYE sip:MY_PHONE_NUMBER@209.210.16.218:5060 SIP/2.0
Via: SIP/2.0/UDP 67.137.39.100:5060;branch=z9hG4bKfakhdi00dgthjgohb7r1sdq9c1gr3.1
Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event
Max-Forwards: 69
Call-ID: 596755814925b3a3217a7a5021cac562@ptldor17.orv.integra.net
From: <sip:NUMBER_I_AM_CALLING@67.137.39.100:5060>;tag=ptleorteca0.orv.integra.voip+1+455a39+104acdc3
To: "Conference Room" <sip:MY_PHONE_NUMBER@ptldor17.orv.integra.net>;tag=as2fdcf3bb
CSeq: 984678647 BYE
Organization: MetaSwitch
Supported: 100rel, resource-priority
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 67.137.39.100:5060 (no NAT)
Scheduling destruction of SIP dialog '596755814925b3a3217a7a5021cac562@ptldor17.orv.integra.net' in 6400 ms (Method: BYE)
<--- Transmitting (no NAT) to 67.137.39.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.137.39.100:5060;branch=z9hG4bKfakhdi00dgthjgohb7r1sdq9c1gr3.1;received=67.137.39.100
From: <sip:NUMBER_I_AM_CALLING@67.137.39.100:5060>;tag=ptleorteca0.orv.integra.voip+1+455a39+104acdc3
To: "Conference Room" <sip:MY_PHONE_NUMBER@ptldor17.orv.integra.net>;tag=as2fdcf3bb
Call-ID: 596755814925b3a3217a7a5021cac562@ptldor17.orv.integra.net
CSeq: 984678647 BYE
Server: Asterisk PBX 1.8.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (internal, NUMBER_I_AM_CALLING, 2) exited non-zero on 'SIP/Conference-00000011'
Scheduling destruction of SIP dialog '273b6b0b6aeb85b4@10.2.1.16' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:Conference@10.2.1.16> for address/port to send to
set_destination: set destination to 10.2.1.16:5060
Reliably Transmitting (no NAT) to 10.2.1.16:5060:
BYE sip:Conference@10.2.1.16 SIP/2.0
Via: SIP/2.0/UDP 10.2.0.10:5060;branch=z9hG4bK4820f517
Max-Forwards: 70
From: <sip:NUMBER_I_AM_CALLING@10.2.0.10>;tag=as1b98070f
To: "Conference Room" <sip:Conference@10.2.0.10>;tag=24e34f24310cad94
Call-ID: 273b6b0b6aeb85b4@10.2.1.16
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.8.0
Proxy-Authorization: Digest username="Conference", realm="asterisk", algorithm=MD5, uri="sip:10.2.0.10", nonce="", response="d09887bce4bd0aea14b86d2db132eebd"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:10.2.1.16:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.0.10:5060;branch=z9hG4bK4820f517
From: <sip:NUMBER_I_AM_CALLING@10.2.0.10>;tag=as1b98070f
To: "Conference Room" <sip:Conference@10.2.0.10>;tag=24e34f24310cad94
Call-ID: 273b6b0b6aeb85b4@10.2.1.16
CSeq: 102 BYE
User-Agent: Grandstream HT287 1.1.0.42 DevId 000b82334458
Contact: <sip:Conference@10.2.1.16>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE
Supported: replaces, timer
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '273b6b0b6aeb85b4@10.2.1.16' Method: ACK