thr3ads.net - asterisk users - [asterisk-users] SIP/Polycom Issue, Asterisk 1.2.16, calls dropped [Mar 2007]

If this information is useful, please help other people find it:
Share via:

Timothy McKee

2007-Mar-20 20:37 UTC

[asterisk-users] SIP/Polycom Issue, Asterisk 1.2.16, calls dropped

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@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>
CSeq: 1 INVITE
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
Contact: <sip:eden-1000a@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@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as7f808f0f
Call-ID: a857d7ac-36f29d46-4d6ef889@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@10.253.4.50' in 15000 ms
Found user 'eden-1000a'
pbx*CLI>
<-- SIP read from 10.253.4.50:5060:
INVITE sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>
CSeq: 1 INVITE
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
Contact: <sip:eden-1000a@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@hostname.company.domain SIP/2.0
Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF
From: "eden-1000a"  
<sip:eden-1000a@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as7f808f0f
CSeq: 1 ACK
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
Contact: <sip:eden-1000a@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@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>
CSeq: 2 INVITE
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
Contact: <sip:eden-1000a@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@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@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@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990@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@eden") in new stack
     -- Playing 'vm-password' (language 'en')
pbx*CLI>
<-- SIP read from 10.253.4.50:5060:
INVITE sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>
CSeq: 2 INVITE
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
Contact: <sip:eden-1000a@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@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990@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@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>
CSeq: 2 INVITE
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
Contact: <sip:eden-1000a@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@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990@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@172.30.42.5 SIP/2.0
Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK1674aeae5EA3A4B
From: "eden-1000a"  
<sip:eden-1000a@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
CSeq: 2 ACK
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
Contact: <sip:eden-1000a@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@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990@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@10.253.4.50 SIP/2.0
Via: SIP/2.0/UDP 172.30.42.5:5060;branch=z9hG4bK7823a1a6;rport
From: "asterisk" <sip:asterisk@172.30.42.5>;tag=as021e29c4
To: <sip:eden-1000a@10.253.4.50>
Contact: <sip:asterisk@172.30.42.5>
Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9990@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@10.253.4.50 SIP/2.0
Via: SIP/2.0/UDP 172.30.42.5:5060;branch=z9hG4bK7823a1a6;rport
From: "asterisk" <sip:asterisk@172.30.42.5>;tag=as021e29c4
To: <sip:eden-1000a@10.253.4.50>
Contact: <sip:asterisk@172.30.42.5>
Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4@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@172.30.42.5>;tag=as021e29c4
To: <sip:eden-1000a@10.253.4.50>;tag=9E3B7462-6F180925
CSeq: 102 OPTIONS
Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4@172.30.42.5
Contact: <sip:eden-1000a@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@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@172.30.42.5>;tag=as021e29c4
To: <sip:eden-1000a@10.253.4.50>;tag=9E3B7462-6F180925
CSeq: 102 OPTIONS
Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4@172.30.42.5
Contact: <sip:eden-1000a@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@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@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@10.253.4.50 for seqno 2 (Non-critical  
Response)
     -- SIP/cp-0821a7d8 answered IAX2/acppbx-102
Destroying call 'a857d7ac-36f29d46-4d6ef889@10.253.4.50'
pbx*CLI> exit
<-- SIP read from 10.253.4.50:5060:
BYE sip:9990@172.30.42.5 SIP/2.0
Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK449f277f6319767C
From: "eden-1000a"  
<sip:eden-1000a@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
CSeq: 3 BYE
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
Contact: <sip:eden-1000a@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@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889@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@172.30.42.5 SIP/2.0
Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK449f277f6319767C
From: "eden-1000a"  
<sip:eden-1000a@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
CSeq: 3 BYE
Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
Contact: <sip:eden-1000a@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@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@hostname.company.domain>;tag=D4964260-95FB99E3
To: <sip:9990@111.111.111.111;user=phone>;tag=as789e1ad9
Call-ID: a857d7ac-36f29d46-4d6ef889@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

Stuart Sheldon

2007-Mar-20 20:46 UTC

head link

[asterisk-users] SIP/Polycom Issue, Asterisk 1.2.16, calls dropped

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Are you using Answer() before VoiceMailMain()?

Stu


Timothy McKee wrote:> 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@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>
> CSeq: 1 INVITE
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> Contact: <sip:eden-1000a@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@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as7f808f0f
> Call-ID: a857d7ac-36f29d46-4d6ef889@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@10.253.4.50'
> in 15000 ms
> Found user 'eden-1000a'
> pbx*CLI>
> <-- SIP read from 10.253.4.50:5060:
> INVITE sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>
> CSeq: 1 INVITE
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> Contact: <sip:eden-1000a@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@hostname.company.domain SIP/2.0
> Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK8ed5192B7E6AF
> From: "eden-1000a"
> <sip:eden-1000a@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as7f808f0f
> CSeq: 1 ACK
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> Contact: <sip:eden-1000a@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@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>
> CSeq: 2 INVITE
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> Contact: <sip:eden-1000a@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@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@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@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990@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@eden")
> in new stack
>     -- Playing 'vm-password' (language 'en')
> pbx*CLI>
> <-- SIP read from 10.253.4.50:5060:
> INVITE sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>
> CSeq: 2 INVITE
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> Contact: <sip:eden-1000a@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@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990@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@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>
> CSeq: 2 INVITE
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> Contact: <sip:eden-1000a@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@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990@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@172.30.42.5 SIP/2.0
> Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK1674aeae5EA3A4B
> From: "eden-1000a"
> <sip:eden-1000a@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> CSeq: 2 ACK
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> Contact: <sip:eden-1000a@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@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990@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@10.253.4.50 SIP/2.0
> Via: SIP/2.0/UDP 172.30.42.5:5060;branch=z9hG4bK7823a1a6;rport
> From: "asterisk" <sip:asterisk@172.30.42.5>;tag=as021e29c4
> To: <sip:eden-1000a@10.253.4.50>
> Contact: <sip:asterisk@172.30.42.5>
> Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:9990@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@10.253.4.50 SIP/2.0
> Via: SIP/2.0/UDP 172.30.42.5:5060;branch=z9hG4bK7823a1a6;rport
> From: "asterisk" <sip:asterisk@172.30.42.5>;tag=as021e29c4
> To: <sip:eden-1000a@10.253.4.50>
> Contact: <sip:asterisk@172.30.42.5>
> Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4@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@172.30.42.5>;tag=as021e29c4
> To: <sip:eden-1000a@10.253.4.50>;tag=9E3B7462-6F180925
> CSeq: 102 OPTIONS
> Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4@172.30.42.5
> Contact: <sip:eden-1000a@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@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@172.30.42.5>;tag=as021e29c4
> To: <sip:eden-1000a@10.253.4.50>;tag=9E3B7462-6F180925
> CSeq: 102 OPTIONS
> Call-ID: 2a1ab9c42b63a0305f6de14715f4f8f4@172.30.42.5
> Contact: <sip:eden-1000a@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@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@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@10.253.4.50
> for seqno 2 (Non-critical Response)
>     -- SIP/cp-0821a7d8 answered IAX2/acppbx-102
> Destroying call 'a857d7ac-36f29d46-4d6ef889@10.253.4.50'
> pbx*CLI> exit
> <-- SIP read from 10.253.4.50:5060:
> BYE sip:9990@172.30.42.5 SIP/2.0
> Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK449f277f6319767C
> From: "eden-1000a"
> <sip:eden-1000a@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> CSeq: 3 BYE
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> Contact: <sip:eden-1000a@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@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889@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@172.30.42.5 SIP/2.0
> Via: SIP/2.0/UDP 10.253.4.50;branch=z9hG4bK449f277f6319767C
> From: "eden-1000a"
> <sip:eden-1000a@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@hostname.company.domain;user=phone>;tag=as789e1ad9
> CSeq: 3 BYE
> Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50
> Contact: <sip:eden-1000a@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@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@hostname.company.domain>;tag=D4964260-95FB99E3
> To: <sip:9990@111.111.111.111;user=phone>;tag=as789e1ad9
> Call-ID: a857d7ac-36f29d46-4d6ef889@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
> 
> 
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> -----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.3 (GNU/Linux)

iD8DBQFGAKq0K69Y+xPZrWYRAn/uAJ4rLdaVqgKP3Bd50gXCFr3NzhBixgCfeuNE
RZbCnj32FyOBFef9yzvr+F4=K61z
-----END PGP SIGNATURE-----

Possibly Parallel Threads

Search for more apparently analagous threads

asterisk users - Mar 2007 - SIP/Polycom Issue, Asterisk 1.2.16, calls dropped

[asterisk-users] SIP/Polycom Issue, Asterisk 1.2.16, calls dropped

[asterisk-users] SIP/Polycom Issue, Asterisk 1.2.16, calls dropped

Possibly Parallel Threads