[asterisk-users] SIP/Polycom Issue, Asterisk 1.2.16, calls dropped
Timothy McKee
tmckee at sdnglobal.com
Tue Mar 20 20:37:40 MST 2007
I've been running the 8/1/2004 Head release up until a little over a
week ago. I was forced to due to a card failure to upgrade to 1.2.16
without any advance preparation or testing (most of my connections
are via satellite to all corners of the globe with high latency).
Up until the upgrade I was running with very few issues. Since the
upgrade I have been experiencing strange issues with my Polycom
SP-601 phones. My customers attempt to get their voicemail and
Asterisk drops their connection ~15 seconds after they dial VM. I
have captured a SIP debug and included it (somewhat sanitized). I'm
not a SIP guru, but I can see the 15 second timer being set and I see
repeated INVITEs being sent without any acks. OPTIONs are being sent
and acked. The remote SIP phone is 'eden-1000a' and the voicemail
extension is 9990. *This worked just fine up until the upgrade.*
Does this ring a bell with anyone out there???
Tim McKee
<tmckee at sdnglobal dot com>
SDN Global
==============================================
pbx*CLI> sip debug peer eden-1000a
SIP Debugging Enabled for IP: 10.253.4.50:5060
pbx*CLI>
<-- SIP read from 10.253.4.50:5060:
INVITE sip:9990 at hostname.company.domain;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>
CSeq: 1 INVITE
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
Contact: <sip:eden-1000a at 10.253.4.50>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 245
v=0
o=- 978307756 978307756 IN IP4 10.253.4.50
s=Polycom IP Phone
c=IN IP4 10.253.4.50
t=0 0
m=audio 2228 RTP/AVP 0 18 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
--- (14 headers 11 lines) ---
Using INVITE request as basis request -
a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
Sending to 10.253.4.50 : 5060 (NAT)
Reliably Transmitting (no NAT) to 10.253.4.50:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK8ed5192B7E6AF;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as7f808f0f
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="2584558d"
Content-Length: 0
---
Scheduling destruction of call
'a857d7ac-36f29d46-4d6ef889 at 10.253.4.50' in 15000 ms
Found user 'eden-1000a'
pbx*CLI>
<-- SIP read from 10.253.4.50:5060:
INVITE sip:9990 at hostname.company.domain;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>
CSeq: 1 INVITE
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
Contact: <sip:eden-1000a at 10.253.4.50>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 245
v=0
o=- 978307756 978307756 IN IP4 10.253.4.50
s=Polycom IP Phone
c=IN IP4 10.253.4.50
t=0 0
m=audio 2228 RTP/AVP 0 18 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
--- (14 headers 11 lines) ---
Ignoring this INVITE request
pbx*CLI>
<-- SIP read from 10.253.4.50:5060:
ACK sip:9990 at hostname.company.domain SIP/2.0
Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as7f808f0f
CSeq: 1 ACK
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
Contact: <sip:eden-1000a at 10.253.4.50>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
Max-Forwards: 70
Content-Length: 0
--- (11 headers 0 lines) ---
pbx*CLI>
<-- SIP read from 10.253.4.50:5060:
INVITE sip:9990 at hostname.company.domain;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>
CSeq: 2 INVITE
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
Contact: <sip:eden-1000a at 10.253.4.50>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="eden-1000a", realm="asterisk",
nonce="2584558d", uri="sip:9990 at hostname.company.domain;user=phone",
response="d9b3ca0769228d580b8877300d1e4ef3", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 245
v=0
o=- 978307756 978307756 IN IP4 10.253.4.50
s=Polycom IP Phone
c=IN IP4 10.253.4.50
t=0 0
m=audio 2228 RTP/AVP 0 18 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
--- (15 headers 11 lines) ---
Using INVITE request as basis request -
a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
Sending to 10.253.4.50 : 5060 (non-NAT)
Found user 'eden-1000a'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.253.4.50:2228
Peer video RTP is at port 10.253.4.50:65535
Found description format PCMU
Found description format G729
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x10c (ulaw|alaw|g729)/
video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 9990 in eden-dialout (domain
hostname.company.domain;user=phone)
list_route: hop: <sip:eden-1000a at 10.253.4.50>
Transmitting (no NAT) to 10.253.4.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990 at 172.30.42.5>
Content-Length: 0
---
-- Executing Answer("SIP/eden-1000a-4150cc98", "") in new stack
We're at 172.30.42.5 port 29816
Video is at 172.30.42.5 port 29214
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.253.4.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990 at 172.30.42.5>
Content-Type: application/sdp
ontent-Length: 235
v=0
o=root 5641 5641 IN IP4 172.30.42.5
s=session
c=IN IP4 172.30.42.5
t=0 0
m=audio 29816 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Executing VoiceMailMain("SIP/eden-1000a-4150cc98",
"1000 at eden") in new stack
-- Playing 'vm-password' (language 'en')
pbx*CLI>
<-- SIP read from 10.253.4.50:5060:
INVITE sip:9990 at hostname.company.domain;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>
CSeq: 2 INVITE
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
Contact: <sip:eden-1000a at 10.253.4.50>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="eden-1000a", realm="asterisk",
nonce="2584558d", uri="sip:9990 at hostname.company.domain;user=phone",
response="d9b3ca0769228d580b8877300d1e4ef3", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 245
v=0
o=- 978307756 978307756 IN IP4 10.253.4.50
s=Polycom IP Phone
c=IN IP4 10.253.4.50
t=0 0
m=audio 2228 RTP/AVP 0 18 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
--- (15 headers 11 lines) ---
Ignoring this INVITE request
We're at 172.30.42.5 port 29816
Video is at 172.30.42.5 port 29214
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.253.4.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990 at 172.30.42.5>
Content-Type: application/sdp
ontent-Length: 235
v=0
o=root 5641 5642 IN IP4 172.30.42.5
s=session
c=IN IP4 172.30.42.5
t=0 0
m=audio 29816 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from 10.253.4.50:5060:
INVITE sip:9990 at hostname.company.domain;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK82926abd205366FA
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>
CSeq: 2 INVITE
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
Contact: <sip:eden-1000a at 10.253.4.50>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="eden-1000a", realm="asterisk",
nonce="2584558d", uri="sip:9990 at hostname.company.domain;user=phone",
response="d9b3ca0769228d580b8877300d1e4ef3", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 245
v=0
o=- 978307756 978307756 IN IP4 10.253.4.50
s=Polycom IP Phone
c=IN IP4 10.253.4.50
t=0 0
m=audio 2228 RTP/AVP 0 18 8 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
--- (15 headers 11 lines) ---
Ignoring this INVITE request
We're at 172.30.42.5 port 29816
Video is at 172.30.42.5 port 29214
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.253.4.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990 at 172.30.42.5>
Content-Type: application/sdp
ontent-Length: 235
v=0
o=root 5641 5643 IN IP4 172.30.42.5
s=session
c=IN IP4 172.30.42.5
t=0 0
m=audio 29816 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from 10.253.4.50:5060:
ACK sip:9990 at 172.30.42.5 SIP/2.0
Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK1674aeae5EA3A4B
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
CSeq: 2 ACK
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
Contact: <sip:eden-1000a at 10.253.4.50>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
Proxy-Authorization: Digest username="eden-1000a", realm="asterisk",
nonce="2584558d", uri="sip:9990 at hostname.company.domain;user=phone",
response="d9b3ca0769228d580b8877300d1e4ef3", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
--- (12 headers 0 lines) ---
Retransmitting #1 (no NAT) to 10.253.4.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990 at 172.30.42.5>
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 5641 5641 IN IP4 172.30.42.5
s=session
c=IN IP4 172.30.42.5
t=0 0
m=audio 29816 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Retransmitting #1 (no NAT) to 10.253.4.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990 at 172.30.42.5>
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 5641 5642 IN IP4 172.30.42.5
s=session
c=IN IP4 172.30.42.5
t=0 0
m=audio 29816 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Retransmitting #2 (no NAT) to 10.253.4.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990 at 172.30.42.5>
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 5641 5641 IN IP4 172.30.42.5
s=session
c=IN IP4 172.30.42.5
t=0 0
m=audio 29816 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Retransmitting #2 (no NAT) to 10.253.4.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990 at 172.30.42.5>
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 5641 5642 IN IP4 172.30.42.5
s=session
c=IN IP4 172.30.42.5
t=0 0
m=audio 29816 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Playing 'vm-youhave' (language 'en')
-- Playing 'digits/1' (language 'en')
Retransmitting #3 (no NAT) to 10.253.4.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990 at 172.30.42.5>
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 5641 5641 IN IP4 172.30.42.5
s=session
c=IN IP4 172.30.42.5
t=0 0
m=audio 29816 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Retransmitting #3 (no NAT) to 10.253.4.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990 at 172.30.42.5>
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 5641 5642 IN IP4 172.30.42.5
s=session
c=IN IP4 172.30.42.5
t=0 0
m=audio 29816 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Playing 'vm-Old' (language 'en')
-- Playing 'vm-message' (language 'en')
-- Playing 'vm-onefor' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'vm-Old' (language 'en')
-- Playing 'vm-first' (language 'en')
Retransmitting #4 (no NAT) to 10.253.4.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990 at 172.30.42.5>
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 5641 5641 IN IP4 172.30.42.5
s=session
c=IN IP4 172.30.42.5
t=0 0
m=audio 29816 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Playing 'vm-message' (language 'en')
Retransmitting #4 (no NAT) to 10.253.4.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990 at 172.30.42.5>
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 5641 5642 IN IP4 172.30.42.5
s=session
c=IN IP4 172.30.42.5
t=0 0
m=audio 29816 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
== Parsing '/var/spool/asterisk/voicemail/eden/1000/Old/
msg0000.txt': Found
-- Playing 'vm-received' (language 'en')
-- Playing 'digits/at' (language 'en')
-- Playing 'digits/17' (language 'en')
-- Playing 'digits/hundred' (language 'en')
Retransmitting #5 (no NAT) to 10.253.4.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990 at 172.30.42.5>
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 5641 5641 IN IP4 172.30.42.5
s=session
c=IN IP4 172.30.42.5
t=0 0
m=audio 29816 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- SIP/cp-0821a7d8 is making progress passing it to IAX2/acppbx-102
Retransmitting #5 (no NAT) to 10.253.4.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990 at 172.30.42.5>
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 5641 5642 IN IP4 172.30.42.5
s=session
c=IN IP4 172.30.42.5
t=0 0
m=audio 29816 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Playing 'digits/50' (language 'en')
-- Playing 'digits/5' (language 'en')
-- Playing 'hours' (language 'en')
-- Playing '/var/spool/asterisk/voicemail/eden/1000/Old/
msg0000' (language 'en')
12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.253.4.50:5060:
OPTIONS sip:eden-1000a at 10.253.4.50 SIP/2.0
Via: SIP/2.0/UDP 172.30.42.5:5060;branch=z9hG4bK7823a1a6;rport
From: "asterisk" <sip:asterisk at 172.30.42.5>;tag=as021e29c4
To: <sip:eden-1000a at 10.253.4.50>
Contact: <sip:asterisk at 172.30.42.5>
Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4 at 172.30.42.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 20 Mar 2007 23:01:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Retransmitting #6 (no NAT) to 10.253.4.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990 at 172.30.42.5>
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 5641 5641 IN IP4 172.30.42.5
s=session
c=IN IP4 172.30.42.5
t=0 0
m=audio 29816 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Retransmitting #6 (no NAT) to 10.253.4.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK82926abd205366FA;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990 at 172.30.42.5>
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 5641 5642 IN IP4 172.30.42.5
s=session
c=IN IP4 172.30.42.5
t=0 0
m=audio 29816 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
Retransmitting #1 (no NAT) to 10.253.4.50:5060:
OPTIONS sip:eden-1000a at 10.253.4.50 SIP/2.0
Via: SIP/2.0/UDP 172.30.42.5:5060;branch=z9hG4bK7823a1a6;rport
From: "asterisk" <sip:asterisk at 172.30.42.5>;tag=as021e29c4
To: <sip:eden-1000a at 10.253.4.50>
Contact: <sip:asterisk at 172.30.42.5>
Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4 at 172.30.42.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 20 Mar 2007 23:01:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
ontent-Length: 0
---
pbx*CLI> exit
<-- SIP read from 10.253.4.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.42.5:5060;branch=z9hG4bK7823a1a6;rport
From: "asterisk" <sip:asterisk at 172.30.42.5>;tag=as021e29c4
To: <sip:eden-1000a at 10.253.4.50>;tag=9E3B7462-6F180925
CSeq: 102 OPTIONS
Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4 at 172.30.42.5
Contact: <sip:eden-1000a at 10.253.4.50>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
Content-Length: 0
--- (10 headers 0 lines) ---
Destroying call '2a1ab9c42b63a0305f6de14715f4f8f4 at 172.30.42.5'
pbx*CLI> exit
<-- SIP read from 10.253.4.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.42.5:5060;branch=z9hG4bK7823a1a6;rport
From: "asterisk" <sip:asterisk at 172.30.42.5>;tag=as021e29c4
To: <sip:eden-1000a at 10.253.4.50>;tag=9E3B7462-6F180925
CSeq: 102 OPTIONS
Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4 at 172.30.42.5
Contact: <sip:eden-1000a at 10.253.4.50>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
Content-Length: 0
--- (10 headers 0 lines) ---
Mar 20 18:01:44 WARNING[2770]: chan_sip.c:1228 retrans_pkt: Maximum
retries exceeded on transmission
a857d7ac-36f29d46-4d6ef889 at 10.253.4.50 for seqno 2 (Critical Response)
Mar 20 18:01:44 WARNING[2770]: chan_sip.c:1245 retrans_pkt: Hanging
up call a857d7ac-36f29d46-4d6ef889 at 10.253.4.50 - no reply to our
critical packet.
== Spawn extension (eden-dialout, 9990, 2) exited non-zero on 'SIP/
eden-1000a-4150cc98'
Mar 20 18:01:45 WARNING[2770]: chan_sip.c:1228 retrans_pkt: Maximum
retries exceeded on transmission
a857d7ac-36f29d46-4d6ef889 at 10.253.4.50 for seqno 2 (Non-critical
Response)
-- SIP/cp-0821a7d8 answered IAX2/acppbx-102
Destroying call 'a857d7ac-36f29d46-4d6ef889 at 10.253.4.50'
pbx*CLI> exit
<-- SIP read from 10.253.4.50:5060:
BYE sip:9990 at 172.30.42.5 SIP/2.0
Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK449f277f6319767C
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
CSeq: 3 BYE
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
Contact: <sip:eden-1000a at 10.253.4.50>
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
Proxy-Authorization: Digest username="eden-1000a", realm="asterisk",
nonce="2584558d", uri="sip:9990 at hostname.company.domain;user=phone",
response="32687f30de53796b3ad2c3283d199984", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
--- (11 headers 0 lines) ---
Transmitting (NAT) to 10.253.4.50:5060:
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK449f277f6319767C;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
pbx*CLI> exit
<-- SIP read from 10.253.4.50:5060:
BYE sip:9990 at 172.30.42.5 SIP/2.0
Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK449f277f6319767C
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at hostname.company.domain;user=phone>;tag=as789e1ad9
CSeq: 3 BYE
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
Contact: <sip:eden-1000a at 10.253.4.50>
User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127
Proxy-Authorization: Digest username="eden-1000a", realm="asterisk",
nonce="2584558d", uri="sip:9990 at hostname.company.domain;user=phone",
response="32687f30de53796b3ad2c3283d199984", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
--- (11 headers 0 lines) ---
Transmitting (NAT) to 10.253.4.50:5060:
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP
10.253.4.50;branch=z9hG4bK449f277f6319767C;received=10.253.4.50
From: "eden-1000a"
<sip:eden-1000a at hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990 at 111.111.111.111;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889 at 10.253.4.50
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
pbx*CLI> exit
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