[asterisk-users] No reply to our critical packet
Andrew Joakimsen
joakimsen at gmail.com
Wed Oct 8 14:42:21 CDT 2008
-- Executing [s at macro-vmlogin:2]
VoiceMailMain("SIP/17865221569-b6b03f60", "3523782778|s") in new stack
-- <SIP/17865221569-b6b03f60> Playing 'vm-youhave' (language 'en')
app5*CLI>
<--- SIP read from 74.170.252.213:5060 --->
ACK sip:14193016245 at 74.124.208.137 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bKaf5e8319A2DD152C
From: "17865221569" <sip:17865221569 at sip02.netjdn.com>;tag=329CAFE3-451838A4
To: <sip:14193016245 at sip02.netjdn.com;user=phone>;tag=as530e3156
CSeq: 2 ACK
Call-ID: 4835bcaf-1924b231-a74bd31a at 192.168.1.54
Contact: <sip:17865221569 at 192.168.1.54>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
Proxy-Authorization: Digest username="17865221569",
realm="netjdn.com", nonce="01c73ede",
uri="sip:14193016245 at sip02.netjdn.com:5060;user=phone",
response="7b07733275a9401cc5d8bbdfcd4028b4", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
-- <SIP/17865221569-b6b03f60> Playing 'digits/1' (language 'en')
Retransmitting #2 (NAT) to 74.170.252.213:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.54;branch=z9hG4bK36ec371e46AEDA45;received=74.170.252.213
From: "17865221569" <sip:17865221569 at sip02.netjdn.com>;tag=329CAFE3-451838A4
To: <sip:14193016245 at sip02.netjdn.com;user=phone>;tag=as530e3156
Call-ID: 4835bcaf-1924b231-a74bd31a at 192.168.1.54
CSeq: 2 INVITE
User-Agent: HardenedSipServer-4.x
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:14193016245 at 74.124.208.137>
Content-Type: application/sdp
Content-Length: 291
v=0
o=root 26803 26803 IN IP4 74.124.208.137
s=session
c=IN IP4 74.124.208.137
t=0 0
m=audio 10624 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
app5*CLI>
<--- SIP read from 74.170.252.213:5060 --->
ACK sip:14193016245 at 74.124.208.137 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bKaf5e8319A2DD152C
From: "17865221569" <sip:17865221569 at sip02.netjdn.com>;tag=329CAFE3-451838A4
To: <sip:14193016245 at sip02.netjdn.com;user=phone>;tag=as530e3156
CSeq: 2 ACK
Call-ID: 4835bcaf-1924b231-a74bd31a at 192.168.1.54
Contact: <sip:17865221569 at 192.168.1.54>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
Proxy-Authorization: Digest username="17865221569",
realm="netjdn.com", nonce="01c73ede",
uri="sip:14193016245 at sip02.netjdn.com:5060;user=phone",
response="7b07733275a9401cc5d8bbdfcd4028b4", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
-- <SIP/17865221569-b6b03f60> Playing 'vm-INBOX' (language 'en')
-- <SIP/17865221569-b6b03f60> Playing 'vm-and' (language 'en')
-- <SIP/17865221569-b6b03f60> Playing 'digits/8' (language 'en')
app5*CLI>
<--- SIP read from 74.170.252.213:5060 --->
REGISTER sip:sip02.netjdn.com:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bKe9cfe68bDC7AD066
From: "17865221569" <sip:17865221569 at sip02.netjdn.com>;tag=256F3E05-5ECA57DE
To: <sip:17865221569 at sip02.netjdn.com>
CSeq: 6105 REGISTER
Call-ID: 6b5eb1f1-f6629a3-6cf6f764 at 192.168.1.54
Contact: <sip:17865221569 at 192.168.1.54>;methods="INVITE, ACK, BYE,
CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE,
REFER"
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
Authorization: Digest username="17865221569", realm="netjdn.com",
nonce="3a435c0d", uri="sip:sip02.netjdn.com:5060",
response="9e6d6128a5e6e4508dca68b29c2c277c", algorithm=MD5
Max-Forwards: 70
Expires: 30
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 74.170.252.213 : 5060 (NAT)
<--- Transmitting (NAT) to 74.170.252.213:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.54;branch=z9hG4bKe9cfe68bDC7AD066;received=74.170.252.213
From: "17865221569" <sip:17865221569 at sip02.netjdn.com>;tag=256F3E05-5ECA57DE
To: <sip:17865221569 at sip02.netjdn.com>
Call-ID: 6b5eb1f1-f6629a3-6cf6f764 at 192.168.1.54
CSeq: 6105 REGISTER
User-Agent: HardenedSipServer-4.x
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:17865221569 at 74.124.208.137>
Content-Length: 0
<------------>
app5*CLI>
<--- Transmitting (NAT) to 74.170.252.213:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.54;branch=z9hG4bKe9cfe68bDC7AD066;received=74.170.252.213
From: "17865221569" <sip:17865221569 at sip02.netjdn.com>;tag=256F3E05-5ECA57DE
To: <sip:17865221569 at sip02.netjdn.com>;tag=as5e3cef8a
Call-ID: 6b5eb1f1-f6629a3-6cf6f764 at 192.168.1.54
CSeq: 6105 REGISTER
User-Agent: HardenedSipServer-4.x
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: <sip:17865221569 at 192.168.1.54>;expires=60
Date: Wed, 08 Oct 2008 20:14:48 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
'6b5eb1f1-f6629a3-6cf6f764 at 192.168.1.54' in 32000 ms (Method:
REGISTER)
-- <SIP/17865221569-b6b03f60> Playing 'vm-Old' (language 'en')
Retransmitting #3 (NAT) to 74.170.252.213:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.54;branch=z9hG4bK36ec371e46AEDA45;received=74.170.252.213
From: "17865221569" <sip:17865221569 at sip02.netjdn.com>;tag=329CAFE3-451838A4
To: <sip:14193016245 at sip02.netjdn.com;user=phone>;tag=as530e3156
Call-ID: 4835bcaf-1924b231-a74bd31a at 192.168.1.54
CSeq: 2 INVITE
User-Agent: HardenedSipServer-4.x
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:14193016245 at 74.124.208.137>
Content-Type: application/sdp
Content-Length: 291
v=0
o=root 26803 26803 IN IP4 74.124.208.137
s=session
c=IN IP4 74.124.208.137
t=0 0
m=audio 10624 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
app5*CLI>
<--- SIP read from 74.170.252.213:5060 --->
ACK sip:14193016245 at 74.124.208.137 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bKaf5e8319A2DD152C
From: "17865221569" <sip:17865221569 at sip02.netjdn.com>;tag=329CAFE3-451838A4
To: <sip:14193016245 at sip02.netjdn.com;user=phone>;tag=as530e3156
CSeq: 2 ACK
Call-ID: 4835bcaf-1924b231-a74bd31a at 192.168.1.54
Contact: <sip:17865221569 at 192.168.1.54>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
Proxy-Authorization: Digest username="17865221569",
realm="netjdn.com", nonce="01c73ede",
uri="sip:14193016245 at sip02.netjdn.com:5060;user=phone",
response="7b07733275a9401cc5d8bbdfcd4028b4", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
app5*CLI>
<--- SIP read from 74.170.252.213:5060 --->
BYE sip:14193016245 at 74.124.208.137 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bK5ffcb43bD93CC4D6
From: "17865221569" <sip:17865221569 at sip02.netjdn.com>;tag=43A660C8-2E49305
To: <sip:14193016245 at sip02.netjdn.com;user=phone>;tag=as04d385ce
CSeq: 3 BYE
Call-ID: b4cbb064-4b8268de-f0f2c6a3 at 192.168.1.54
Contact: <sip:17865221569 at 192.168.1.54>
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
Proxy-Authorization: Digest username="17865221569",
realm="netjdn.com", nonce="3e862d54",
uri="sip:14193016245 at sip02.netjdn.com:5060;user=phone",
response="e741febb8b521e03c0cd813820cded12", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
-- <SIP/17865221569-b6b03f60> Playing 'vm-messages' (language 'en')
-- <SIP/17865221569-b6b03f60> Playing 'vm-onefor' (language 'en')
-- <SIP/17865221569-b6b03f60> Playing 'vm-INBOX' (language 'en')
-- <SIP/17865221569-b6b03f60> Playing 'vm-messages' (language 'en')
-- <SIP/17865221569-b6b03f60> Playing 'vm-opts' (language 'en')
Retransmitting #4 (NAT) to 74.170.252.213:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.54;branch=z9hG4bK36ec371e46AEDA45;received=74.170.252.213
From: "17865221569" <sip:17865221569 at sip02.netjdn.com>;tag=329CAFE3-451838A4
To: <sip:14193016245 at sip02.netjdn.com;user=phone>;tag=as530e3156
Call-ID: 4835bcaf-1924b231-a74bd31a at 192.168.1.54
CSeq: 2 INVITE
User-Agent: HardenedSipServer-4.x
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:14193016245 at 74.124.208.137>
Content-Type: application/sdp
Content-Length: 291
v=0
o=root 26803 26803 IN IP4 74.124.208.137
s=session
c=IN IP4 74.124.208.137
t=0 0
m=audio 10624 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
app5*CLI>
<--- SIP read from 74.170.252.213:5060 --->
ACK sip:14193016245 at 74.124.208.137 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bKaf5e8319A2DD152C
From: "17865221569" <sip:17865221569 at sip02.netjdn.com>;tag=329CAFE3-451838A4
To: <sip:14193016245 at sip02.netjdn.com;user=phone>;tag=as530e3156
CSeq: 2 ACK
Call-ID: 4835bcaf-1924b231-a74bd31a at 192.168.1.54
Contact: <sip:17865221569 at 192.168.1.54>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
Proxy-Authorization: Digest username="17865221569",
realm="netjdn.com", nonce="01c73ede",
uri="sip:14193016245 at sip02.netjdn.com:5060;user=phone",
response="7b07733275a9401cc5d8bbdfcd4028b4", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
app5*CLI>
<--- SIP read from 74.170.252.213:5060 --->
BYE sip:14193016245 at 74.124.208.137 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bK5ffcb43bD93CC4D6
From: "17865221569" <sip:17865221569 at sip02.netjdn.com>;tag=43A660C8-2E49305
To: <sip:14193016245 at sip02.netjdn.com;user=phone>;tag=as04d385ce
CSeq: 3 BYE
Call-ID: b4cbb064-4b8268de-f0f2c6a3 at 192.168.1.54
Contact: <sip:17865221569 at 192.168.1.54>
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
Proxy-Authorization: Digest username="17865221569",
realm="netjdn.com", nonce="3e862d54",
uri="sip:14193016245 at sip02.netjdn.com:5060;user=phone",
response="e741febb8b521e03c0cd813820cded12", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Retransmitting #5 (NAT) to 74.170.252.213:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.54;branch=z9hG4bK36ec371e46AEDA45;received=74.170.252.213
From: "17865221569" <sip:17865221569 at sip02.netjdn.com>;tag=329CAFE3-451838A4
To: <sip:14193016245 at sip02.netjdn.com;user=phone>;tag=as530e3156
Call-ID: 4835bcaf-1924b231-a74bd31a at 192.168.1.54
CSeq: 2 INVITE
User-Agent: HardenedSipServer-4.x
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:14193016245 at 74.124.208.137>
Content-Type: application/sdp
Content-Length: 291
v=0
o=root 26803 26803 IN IP4 74.124.208.137
s=session
c=IN IP4 74.124.208.137
t=0 0
m=audio 10624 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
app5*CLI>
<--- SIP read from 74.170.252.213:5060 --->
ACK sip:14193016245 at 74.124.208.137 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bKaf5e8319A2DD152C
From: "17865221569" <sip:17865221569 at sip02.netjdn.com>;tag=329CAFE3-451838A4
To: <sip:14193016245 at sip02.netjdn.com;user=phone>;tag=as530e3156
CSeq: 2 ACK
Call-ID: 4835bcaf-1924b231-a74bd31a at 192.168.1.54
Contact: <sip:17865221569 at 192.168.1.54>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
Proxy-Authorization: Digest username="17865221569",
realm="netjdn.com", nonce="01c73ede",
uri="sip:14193016245 at sip02.netjdn.com:5060;user=phone",
response="7b07733275a9401cc5d8bbdfcd4028b4", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
app5*CLI>
<--- SIP read from 74.170.252.213:5060 --->
BYE sip:14193016245 at 74.124.208.137 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bK5ffcb43bD93CC4D6
From: "17865221569" <sip:17865221569 at sip02.netjdn.com>;tag=43A660C8-2E49305
To: <sip:14193016245 at sip02.netjdn.com;user=phone>;tag=as04d385ce
CSeq: 3 BYE
Call-ID: b4cbb064-4b8268de-f0f2c6a3 at 192.168.1.54
Contact: <sip:17865221569 at 192.168.1.54>
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
Proxy-Authorization: Digest username="17865221569",
realm="netjdn.com", nonce="3e862d54",
uri="sip:14193016245 at sip02.netjdn.com:5060;user=phone",
response="e741febb8b521e03c0cd813820cded12", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Scheduling destruction of SIP dialog
'093797fb5ee3659e6df119ac796eae48 at 74.124.208.137' in 8640 ms (Method:
NOTIFY)
Reliably Transmitting (NAT) to 74.170.252.213:5060:
NOTIFY sip:17865221569 at 192.168.1.54 SIP/2.0
Via: SIP/2.0/UDP 74.124.208.137:5060;branch=z9hG4bK54fdc202;rport
From: "asterisk" <sip:asterisk at 74.124.208.137>;tag=as42840967
To: <sip:17865221569 at 192.168.1.54>
Contact: <sip:asterisk at 74.124.208.137>
Call-ID: 093797fb5ee3659e6df119ac796eae48 at 74.124.208.137
CSeq: 102 NOTIFY
User-Agent: HardenedSipServer-4.x
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 97
Messages-Waiting: no
Message-Account: sip:14193016245 at 74.124.208.137
Voice-Message: 0/0 (0/0)
---
app5*CLI>
<--- SIP read from 74.170.252.213:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 74.124.208.137:5060;branch=z9hG4bK54fdc202;rport
From: "asterisk" <sip:asterisk at 74.124.208.137>;tag=as42840967
To: <sip:17865221569 at 192.168.1.54>;tag=DED38D10-3F2880AD
CSeq: 102 NOTIFY
Call-ID: 093797fb5ee3659e6df119ac796eae48 at 74.124.208.137
Contact: <sip:17865221569 at 192.168.1.54>
Event: message-summary
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog
'093797fb5ee3659e6df119ac796eae48 at 74.124.208.137' Method: NOTIFY
-- <SIP/17865221569-b6b03f60> Playing 'vm-helpexit' (language 'en')
Retransmitting #6 (NAT) to 74.170.252.213:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.54;branch=z9hG4bK36ec371e46AEDA45;received=74.170.252.213
From: "17865221569" <sip:17865221569 at sip02.netjdn.com>;tag=329CAFE3-451838A4
To: <sip:14193016245 at sip02.netjdn.com;user=phone>;tag=as530e3156
Call-ID: 4835bcaf-1924b231-a74bd31a at 192.168.1.54
CSeq: 2 INVITE
User-Agent: HardenedSipServer-4.x
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:14193016245 at 74.124.208.137>
Content-Type: application/sdp
Content-Length: 291
v=0
o=root 26803 26803 IN IP4 74.124.208.137
s=session
c=IN IP4 74.124.208.137
t=0 0
m=audio 10624 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
app5*CLI>
<--- SIP read from 74.170.252.213:5060 --->
ACK sip:14193016245 at 74.124.208.137 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bKaf5e8319A2DD152C
From: "17865221569" <sip:17865221569 at sip02.netjdn.com>;tag=329CAFE3-451838A4
To: <sip:14193016245 at sip02.netjdn.com;user=phone>;tag=as530e3156
CSeq: 2 ACK
Call-ID: 4835bcaf-1924b231-a74bd31a at 192.168.1.54
Contact: <sip:17865221569 at 192.168.1.54>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
Proxy-Authorization: Digest username="17865221569",
realm="netjdn.com", nonce="01c73ede",
uri="sip:14193016245 at sip02.netjdn.com:5060;user=phone",
response="7b07733275a9401cc5d8bbdfcd4028b4", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
[Oct 8 16:15:04] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
4835bcaf-1924b231-a74bd31a at 192.168.1.54 for seqno 2 (Critical
Response)
[Oct 8 16:15:04] WARNING[26819]: chan_sip.c:1972 retrans_pkt: Hanging
up call 4835bcaf-1924b231-a74bd31a at 192.168.1.54 - no reply to our
critical packet.
== Spawn extension (macro-vmlogin, s, 2) exited non-zero on
'SIP/17865221569-b6b03f60' in macro 'vmlogin'
== Spawn extension (macro-vmlogin, s, 2) exited non-zero on
'SIP/17865221569-b6b03f60' in macro 'vmcenter'
== Spawn extension (macro-vmlogin, s, 2) exited non-zero on
'SIP/17865221569-b6b03f60'
Really destroying SIP dialog '4835bcaf-1924b231-a74bd31a at 192.168.1.54'
Method: ACK
On Mon, Oct 6, 2008 at 8:26 PM, Atis Lezdins <atis at iq-labs.net> wrote:
> On Tue, Oct 7, 2008 at 2:22 AM, Andrew Joakimsen <joakimsen at gmail.com> wrote:
>> The odd thing is on this particular phone it only happens when you
>> call voicemail.
>>
>> It is certainly a bug in Asterisk, not the UA. Asterisk is trying to
>> send to 192.168.1.x which obviously is not possible. Something in the
>> NAT support is not working right.
>
> Hi,
>
> You should get SIP traces to see why Asterisk is trying to reply to 192.168.1.x.
>
> To do this, enter "sip set debug on" in asterisk CLI, and post us a
> log of call reaching voicemail and disconnecting.
>
> Regards,
> Atis
>
>>
>> On Mon, Oct 6, 2008 at 3:06 PM, SIP <sip at arcdiv.com> wrote:
>>> This message is usually caused by Asterisk not receiving an ACK after
>>> about 30 seconds of attempts. There are countless misconfigured UAs and
>>> proxies out there that don't handle ACK well, so it would be nice to be
>>> able to turn this 'feature' off. What's annoying is that the explanation
>>> has always been "If we can't get an ACK, we can't send any RTP data."
>>> This is patently false, as the RTP will often work fine even if ACK
>>> handling is misconfigured (we see it all the time).
>>>
>>> But alas. As far as I can tell, there's no way to disable this check. I
>>> suppose I could code around it, but not being the world's most
>>> proficient C coder, I'm always afraid I'll break something else. ;)
>>>
>>> N.
>>>
>>>
>>> Andrew Joakimsen wrote:
>>>> I am using a Polycom 501 SIP phone behind NAT. Asterisk server is
>>>> public with no NAT... everything works on the Asterisk end just fine
>>>> EXCEPT that I can never check voice mail
>>>>
>>>> After about 30 seconds the call drops with these messagess:
>>>>
>>>> [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum
>>>> retries exceeded on transmission
>>>> 320893f1-50c13ba3-78c26164 at 192.168.1.54 for seqno 2 (Critical
>>>> Response)
>>>> [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1972 retrans_pkt: Hanging
>>>> up call 320893f1-50c13ba3-78c26164 at 192.168.1.54 - no reply to our
>>>> critical packet.
>>>>
>>>> It seems to me that the problem is the way Asterisk is handling this
>>>> "critical packet" -- of course it can not be sent to 192.168.1.54, the
>>>> phone is at that IP behind a NAT and the Asterisk server is not. I can
>>>> make any other phone call from this same phone as long as it is not
>>>> voicemail and I can be on the line for hours with no problem.
>>>>
>>>> I am really at a loss here. I have searched a bit and come up with
>>>> nothing other than blaming the UA. I know the Polycoms dont have the
>>>> best NAT support but besides this it works problem-free. It's odd I
>>>> can make a call anywhere else even for hours and not have any issues
>>>> at all but 30 seconds into a voicemail call it just drops....
>>>>
>>>>
>>>> app5*CLI> sip show peer 17865221569
>>>> app5*CLI>
>>>>
>>>> * Name : 17865221569
>>>> Secret : <Set>
>>>> MD5Secret : <Not set>
>>>> Context : blended-lcr
>>>> Subscr.Cont. : sla_stations
>>>> Language : en
>>>> AMA flags : Unknown
>>>> Transfer mode: closed
>>>> CallingPres : Presentation Allowed, Not Screened
>>>> Callgroup :
>>>> Pickupgroup :
>>>> Mailbox : 17865221569
>>>> VM Extension : 14193016245
>>>> LastMsgsSent : 0/0
>>>> Call limit : 2
>>>> Dynamic : Yes
>>>> Callerid : "" <CENSORED>
>>>> MaxCallBR : 256 kbps
>>>> Expire : 63
>>>> Insecure : no
>>>> Nat : Always
>>>> ACL : No
>>>> T38 pt UDPTL : Yes
>>>> CanReinvite : No
>>>> PromiscRedir : No
>>>> User=Phone : Yes
>>>> Video Support: No
>>>> Trust RPID : No
>>>> Send RPID : No
>>>> Subscriptions: Yes
>>>> Overlap dial : No
>>>> DTMFmode : rfc2833
>>>> LastMsg : 0
>>>> ToHost :
>>>> Addr->IP : 74.CENSORED.213 Port 5060
>>>> Defaddr->IP : 0.0.0.0 Port 5060
>>>> Reg. exten :
>>>> Def. Username: 17865221569
>>>> SIP Options : (none)
>>>> Codecs : 0x104 (ulaw|g729)
>>>> Codec Order : (g729:20,ulaw:20)
>>>> Auto-Framing: No
>>>> Status : OK (130 ms)
>>>> Useragent : PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
>>>> Reg. Contact : sip:17865221569 at 192.168.1.54
>>>>
>>>>
>>>> app5*CLI> core show version
>>>> Asterisk 1.4.21.1 built by root @ app5 on a i686 running Linux on
>>>> 2008-07-09 01:41:43 UTC
>>>>
>>>> _______________________________________________
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>>>>
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>>>>
>>>
>>>
>>> _______________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
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>>> Register Now: http://www.astricon.net
>>>
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>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
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>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Atis Lezdins,
> VoIP Project Manager / Developer,
> atis at iq-labs.net
> Skype: atis.lezdins
> Cell Phone: +371 28806004
> Cell Phone: +1 800 7300689
> Work phone: +1 800 7502835
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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