[Asterisk-Users] chan_sip changes affecting ACK? - Bug?

Sat Oct 23 08:22:15 MST 2004

Olle,

No...Thank you! You are the perfect guy to look at this problem as well
since ultimately I need to switch to chan_sip2 given the outboundproxy
functionality.

My testing shows that not only stable has this issue but so does head.
That said, the problem could carry over to chan_sip2. Anyway...

I originally sent several log files from both the Siparator and Asterisk
but the message was refused from the list because of size.

Attached are 2 asterisk sip debug files. I fear that some of the
information scrolled off the screen during debug. If these don't have
enough information please let me know. When I get back to the office I
will log sip debug to a file rather than console as I was so I don't
loose anything.

If you would like to see the separator logs I will need to send them to
you directly because they are 300K a piece and go over the limit for
this list.

Thanks,

Chad

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Olle E.
Johansson
Sent: Saturday, October 23, 2004 2:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] chan_sip changes affecting ACK? - Bug?

Chad,
I need a more complete SIP debug than just one packet to try to look
into this
issue. If the device registers, both a REGISTER transaction and a
subsequent
call with the ACK - THank you!

/O
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-------------- next part --------------

 to 10.10.0.110:5060
    -- Executing Dial("SIP/101-eac7", "SIP/12534056726 at 10.10.0.5") in new stack
We're at 10.10.0.6 port 17666
Answering/Requesting with root capability 4
Answering with capability 0x2(GSM)
Answering with capability 0x8(ALAW)
Answering with non-codec capability 0x1(G723)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:12534056726 at 10.10.0.5 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK6016290f
From: "Chad Brown" <sip:101 at 10.10.0.6>;tag=as2041d236
To: <sip:12534056726 at 10.10.0.5>
Contact: <sip:101 at 10.10.0.6>
Call-ID: 1a03af943d7201bc6da3a14e3750865c at 10.10.0.6
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 21 Oct 2004 17:14:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 8649 8649 IN IP4 10.10.0.6
s=session
c=IN IP4 10.10.0.6
t=0 0
m=audio 17666 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (no NAT) to 10.10.0.5:5060
    -- Called 12534056726 at 10.10.0.5
impbx01*CLI>

Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK6016290f
From: "Chad Brown" <sip:101 at 10.10.0.6>;tag=as2041d236
Call-ID: 1a03af943d7201bc6da3a14e3750865c at 10.10.0.6
CSeq: 102 INVITE
Server: SIParator/4.1.3
To: <sip:12534056726 at 10.10.0.5>
Content-Length: 0

8 headers, 0 lines
impbx01*CLI>

Sip read:
SIP/2.0 180 Ringing
To: <sip:12534056726 at 10.10.0.5>;tag=3307367608-554546
From: "Chad Brown" <sip:101 at 10.10.0.6>;tag=as2041d236
Call-ID: 1a03af943d7201bc6da3a14e3750865c at 10.10.0.6
CSeq: 102 INVITE
Contact: <sip:e2i_cjrQXf_07gGdSVIMofwc5VYtECYEoQaBUD_SIhNyCrwe9EIMs75QA6vHgd8Xz at 10.10.0.5>
Content-Type: application/sdp
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK6016290f
Content-Length: 187

v=0
o=NexTone-MSW 1234 467212419 IN IP4 10.10.0.5
s=sip call
c=IN IP4 10.10.0.5
t=0 0
m=audio 58030 RTP/AVP 0
a=silenceSupp:off
a=ecan:b on g168
a=ptime:20
a=rtpmap:0 PCMU/8000

9 headers, 10 lines
    -- SIP/10.10.0.5-13c2 is ringing
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.0.110:5060;branch=z9hG4bK1ed3fe76
From: "Chad Brown - ext 101" <sip:101 at sip.identitymine.com>;tag=001193d886a3010e3ae0c0a4-4f179b0e
To: <sip:12534056726 at sip.identitymine.com>;tag=as33b30738
Call-ID: 001193d8-86a3001b-1cd49d4c-76746a4f at 10.10.0.110
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:12534056726 at 10.10.0.6>
Content-Length: 0

 to 10.10.0.110:5060
impbx01*CLI>

Sip read:
SIP/2.0 200 OK
To: <sip:12534056726 at 10.10.0.5>;tag=3307367608-554546
From: "Chad Brown" <sip:101 at 10.10.0.6>;tag=as2041d236
Call-ID: 1a03af943d7201bc6da3a14e3750865c at 10.10.0.6
CSeq: 102 INVITE
Contact: <sip:eJ7Tp7r12vK-plwl1R4IhUifBHIlEYbMje2Wq3qLiWpSLcSfN6MCsjkU-yOFq6kIT at 10.10.0.5>
Content-Type: application/sdp
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK6016290f
Record-Route: <sip:280dcf6a at 10.10.0.5;lr>
Content-Length: 187

v=0
o=NexTone-MSW 1234 467212419 IN IP4 10.10.0.5
s=sip call
c=IN IP4 10.10.0.5
t=0 0
m=audio 58030 RTP/AVP 0
a=silenceSupp:off
a=ecan:b on g168
a=ptime:20
a=rtpmap:0 PCMU/8000

10 headers, 10 lines
Found RTP audio format 0
Peer audio RTP is at port 10.10.0.5:58030
Found description format PCMU
Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY)
list_route: hop: <sip:280dcf6a at 10.10.0.5;lr>
list_route: hop: <sip:eJ7Tp7r12vK-plwl1R4IhUifBHIlEYbMje2Wq3qLiWpSLcSfN6MCsjkU-yOFq6kIT at 10.10.0.5>
set_destination: Parsing <sip:280dcf6a at 10.10.0.5;lr> for address/port to send to
set_destination: set destination to 10.10.0.5, port 5060
Transmitting:
ACK sip:12534056726 at 10.10.0.5 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK08fa5d64
Route: <sip:eJ7Tp7r12vK-plwl1R4IhUifBHIlEYbMje2Wq3qLiWpSLcSfN6MCsjkU-yOFq6kIT at 10.10.0.5>
From: "Chad Brown" <sip:101 at 10.10.0.6>;tag=as2041d236
To: <sip:12534056726 at 10.10.0.5>;tag=3307367608-554546
Contact: <sip:101 at 10.10.0.6>
Call-ID: 1a03af943d7201bc6da3a14e3750865c at 10.10.0.6
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 10.10.0.5:5060
    -- SIP/10.10.0.5-13c2 answered SIP/101-eac7
We're at 10.10.0.6 port 17116
Answering with preferred capability 0x4(ULAW)
Answering with capability 0x2(GSM)
Answering with capability 0x8(ALAW)
Answering with non-codec capability 0x1(G723)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.0.110:5060;branch=z9hG4bK1ed3fe76
From: "Chad Brown - ext 101" <sip:101 at sip.identitymine.com>;tag=001193d886a3010e3ae0c0a4-4f179b0e
To: <sip:12534056726 at sip.identitymine.com>;tag=as33b30738
Call-ID: 001193d8-86a3001b-1cd49d4c-76746a4f at 10.10.0.110
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:12534056726 at 10.10.0.6>
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 8649 8649 IN IP4 10.10.0.6
s=session
c=IN IP4 10.10.0.6
t=0 0
m=audio 17116 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to 10.10.0.110:5060
    -- Attempting native bridge of SIP/101-eac7 and SIP/10.10.0.5-13c2
impbx01*CLI>

Sip read:
ACK sip:12534056726 at 10.10.0.6:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.110:5060;branch=z9hG4bK4bcf39c0
From: "Chad Brown - ext 101" <sip:101 at sip.identitymine.com>;tag=001193d886a3010e3ae0c0a4-4f179b0e
To: <sip:12534056726 at sip.identitymine.com>;tag=as33b30738
Call-ID: 001193d8-86a3001b-1cd49d4c-76746a4f at 10.10.0.110
CSeq: 101 ACK
User-Agent: CSCO/7
Content-Length: 0
-------------- next part --------------
From: "Chad Brown" <sip:asterisk at 10.10.0.6>;tag=as490d60cd
Call-ID: 08b25b4e793c9fda031a818f7922c61a at 10.10.0.6
CSeq: 102 INVITE
Contact: <sip:eKvj1A8MZuPxroET_BXewOVi3uR-3Ad_liBBCrJQq8dbq7VO-p-RGl6icEsXi2TZX at 10.10.0.5>
Content-Type: application/sdp
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK4441b7b5
Record-Route: <sip:4a37c67a at 10.10.0.5;lr>
Content-Length: 187

v=0
o=NexTone-MSW 1234 467334735 IN IP4 10.10.0.5
s=sip call
c=IN IP4 10.10.0.5
t=0 0
m=audio 58032 RTP/AVP 0
a=silenceSupp:off
a=ecan:b on g168
a=ptime:20
a=rtpmap:0 PCMU/8000

10 headers, 10 lines
Found audio format UNKN
Found description format PCMU
Capabilities: us - 524302, them - 4/0, combined - 4
Non-codec capabilities: us - 1, them - 0, combined - 0
list_route: hop: <sip:4a37c67a at 10.10.0.5;lr>
list_route: hop: <sip:eKvj1A8MZuPxroET_BXewOVi3uR-3Ad_liBBCrJQq8dbq7VO-p-RGl6icEsXi2TZX at 10.10.0.5>
set_destination: Parsing <sip:4a37c67a at 10.10.0.5;lr> for address/port to send to
set_destination: set destination to 10.10.0.5, port 5060
Transmitting:
ACK sip:eKvj1A8MZuPxroET_BXewOVi3uR-3Ad_liBBCrJQq8dbq7VO-p-RGl6icEsXi2TZX at 10.10.0.5 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK4441b7b5
Route: <sip:eKvj1A8MZuPxroET_BXewOVi3uR-3Ad_liBBCrJQq8dbq7VO-p-RGl6icEsXi2TZX at 10.10.0.5>
From: "Chad Brown" <sip:asterisk at 10.10.0.6>;tag=as490d60cd
To: <sip:12534056726 at 10.10.0.5>;tag=3307489924-529483
Contact: <sip:asterisk at 10.10.0.6>
Call-ID: 08b25b4e793c9fda031a818f7922c61a at 10.10.0.6
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 10.10.0.5:5060
    -- SIP/10.10.0.5-0744 answered SIP/101-2878
We're at 10.10.0.6 port 12570
Answering with preferred capability 4
Answering with capability 2
Answering with capability 8
Answering with non-codec capability 1
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.0.110:5060;branch=z9hG4bK2ac6b76c
From: "Chad Brown - ext 101" <sip:101 at sip.identitymine.com>;tag=001193d886a300482359b5ac-0ecac181
To: <sip:12534056726 at sip.identitymine.com>;tag=as0943736e
Call-ID: 001193d8-86a30009-5dea4b03-583b7dbb at 10.10.0.110
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:12534056726 at 10.10.0.6>
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 1940 1940 IN IP4 10.10.0.6
s=session
c=IN IP4 10.10.0.6
t=0 0
m=audio 12570 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to 10.10.0.110:5060
    -- Attempting native bridge of SIP/101-2878 and SIP/10.10.0.5-0744
impbx01*CLI>

Sip read:
ACK sip:12534056726 at 10.10.0.6:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.110:5060;branch=z9hG4bK7e2ea034
From: "Chad Brown - ext 101" <sip:101 at sip.identitymine.com>;tag=001193d886a300482359b5ac-0ecac181
To: <sip:12534056726 at sip.identitymine.com>;tag=as0943736e
Call-ID: 001193d8-86a30009-5dea4b03-583b7dbb at 10.10.0.110
CSeq: 101 ACK
User-Agent: CSCO/7
Content-Length: 0

-------------- next part --------------
a=fmtp:101 0-15

12 headers, 11 lines
Using latest request as basis request
Sending to 10.10.0.110 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.10.0.110:19404
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x10c(ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
Found user '101'
Looking for 12534056726 in trusted
list_route: hop: <sip:101 at 10.10.0.110:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.0.110:5060;branch=z9hG4bK14abe6ee
From: "Chad Brown - ext 101" <sip:101 at sip.identitymine.com>;tag=001193d886a3000c13c96f5b-25455014
To: <sip:12534056726 at sip.identitymine.com>;tag=as68d0496a
Call-ID: 001193d8-86a30005-2d7ba0b0-4bf0ca5b at 10.10.0.110
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:12534056726 at 10.10.0.6>
Content-Length: 0

 to 10.10.0.110:5060
    -- Executing Dial("SIP/101-8c77", "SIP/12534056726 at 10.10.0.5") in new stack
We're at 10.10.0.6 port 12388
Answering/Requesting with root capability 4
Answering with capability 0x2(GSM)
Answering with capability 0x8(ALAW)
Answering with non-codec capability 0x1(G723)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:12534056726 at 10.10.0.5 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK00e6821d
From: "Chad Brown" <sip:asterisk at 10.10.0.6>;tag=as5c7a2a79
To: <sip:12534056726 at 10.10.0.5>
Contact: <sip:asterisk at 10.10.0.6>
Call-ID: 29fbc7297c33271446d86cfc474d0b75 at 10.10.0.6
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 20 Oct 2004 18:23:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 8669 8669 IN IP4 10.10.0.6
s=session
c=IN IP4 10.10.0.6
t=0 0
m=audio 12388 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (no NAT) to 10.10.0.5:5060
    -- Called 12534056726 at 10.10.0.5
impbx01*CLI>

Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK00e6821d
From: "Chad Brown" <sip:asterisk at 10.10.0.6>;tag=as5c7a2a79
Call-ID: 29fbc7297c33271446d86cfc474d0b75 at 10.10.0.6
CSeq: 102 INVITE
Server: SIParator/4.1.0
To: <sip:12534056726 at 10.10.0.5>
Content-Length: 0

8 headers, 0 lines
impbx01*CLI>

Sip read:
SIP/2.0 180 Ringing
To: <sip:12534056726 at 10.10.0.5>;tag=3307285355-806590
From: "Chad Brown" <sip:asterisk at 10.10.0.6>;tag=as5c7a2a79
Call-ID: 29fbc7297c33271446d86cfc474d0b75 at 10.10.0.6
CSeq: 102 INVITE
Contact: <sip:eSQaWt-bp7wmoGkGFP4SAqR9gAQrLzLqVFfwP9dFIcby_yhF8qHZG-f1JSgIwt2hb at 10.10.0.5>
Content-Type: application/sdp
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK00e6821d
Content-Length: 187

v=0
o=NexTone-MSW 1234 467130168 IN IP4 10.10.0.5
s=sip call
c=IN IP4 10.10.0.5
t=0 0
m=audio 58110 RTP/AVP 0
a=silenceSupp:off
a=ecan:b on g168
a=ptime:20
a=rtpmap:0 PCMU/8000

9 headers, 10 lines
    -- SIP/10.10.0.5-f8bf is ringing
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.10.0.110:5060;branch=z9hG4bK14abe6ee
From: "Chad Brown - ext 101" <sip:101 at sip.identitymine.com>;tag=001193d886a3000c13c96f5b-25455014
To: <sip:12534056726 at sip.identitymine.com>;tag=as68d0496a
Call-ID: 001193d8-86a30005-2d7ba0b0-4bf0ca5b at 10.10.0.110
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:12534056726 at 10.10.0.6>
Content-Length: 0

 to 10.10.0.110:5060
impbx01*CLI>

Sip read:
SIP/2.0 200 OK
To: <sip:12534056726 at 10.10.0.5>;tag=3307285355-806590
From: "Chad Brown" <sip:asterisk at 10.10.0.6>;tag=as5c7a2a79
Call-ID: 29fbc7297c33271446d86cfc474d0b75 at 10.10.0.6
CSeq: 102 INVITE
Contact: <sip:eDXTNO6dkwsoA4_uBoSao6QR6sGzyTc9suJBrvzuYOVHclaWI7YmOtc3aQUmYltVy at 10.10.0.5>
Content-Type: application/sdp
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK00e6821d
Record-Route: <sip:26998a73 at 10.10.0.5;lr>
Content-Length: 187

v=0
o=NexTone-MSW 1234 467130168 IN IP4 10.10.0.5
s=sip call
c=IN IP4 10.10.0.5
t=0 0
m=audio 58110 RTP/AVP 0
a=silenceSupp:off
a=ecan:b on g168
a=ptime:20
a=rtpmap:0 PCMU/8000

10 headers, 10 lines
Found RTP audio format 0
Peer audio RTP is at port 10.10.0.5:58110
Found description format PCMU
Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY)
list_route: hop: <sip:26998a73 at 10.10.0.5;lr>
list_route: hop: <sip:eDXTNO6dkwsoA4_uBoSao6QR6sGzyTc9suJBrvzuYOVHclaWI7YmOtc3aQUmYltVy at 10.10.0.5>
set_destination: Parsing <sip:26998a73 at 10.10.0.5;lr> for address/port to send to
set_destination: set destination to 10.10.0.5, port 5060
Transmitting:
ACK sip:12534056726 at 10.10.0.5 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK4be3d57b
Route: <sip:eDXTNO6dkwsoA4_uBoSao6QR6sGzyTc9suJBrvzuYOVHclaWI7YmOtc3aQUmYltVy at 10.10.0.5>
From: "Chad Brown" <sip:asterisk at 10.10.0.6>;tag=as5c7a2a79
To: <sip:12534056726 at 10.10.0.5>;tag=3307285355-806590
Contact: <sip:asterisk at 10.10.0.6>
Call-ID: 29fbc7297c33271446d86cfc474d0b75 at 10.10.0.6
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 10.10.0.5:5060
    -- SIP/10.10.0.5-f8bf answered SIP/101-8c77
We're at 10.10.0.6 port 10262
Answering with preferred capability 0x4(ULAW)
Answering with capability 0x2(GSM)
Answering with capability 0x8(ALAW)
Answering with non-codec capability 0x1(G723)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.0.110:5060;branch=z9hG4bK14abe6ee
From: "Chad Brown - ext 101" <sip:101 at sip.identitymine.com>;tag=001193d886a3000c13c96f5b-25455014
To: <sip:12534056726 at sip.identitymine.com>;tag=as68d0496a
Call-ID: 001193d8-86a30005-2d7ba0b0-4bf0ca5b at 10.10.0.110
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:12534056726 at 10.10.0.6>
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 8669 8669 IN IP4 10.10.0.6
s=session
c=IN IP4 10.10.0.6
t=0 0
m=audio 10262 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to 10.10.0.110:5060
    -- Attempting native bridge of SIP/101-8c77 and SIP/10.10.0.5-f8bf
impbx01*CLI>

Sip read:
ACK sip:12534056726 at 10.10.0.6:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.110:5060;branch=z9hG4bK30b1b6be
From: "Chad Brown - ext 101" <sip:101 at sip.identitymine.com>;tag=001193d886a3000c13c96f5b-25455014
To: <sip:12534056726 at sip.identitymine.com>;tag=as68d0496a
Call-ID: 001193d8-86a30005-2d7ba0b0-4bf0ca5b at 10.10.0.110
CSeq: 101 ACK
User-Agent: CSCO/7
Content-Length: 0

8 headers, 0 lines
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:10.10.0.110 SIP/2.0
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK40f5a9b2
From: "asterisk" <sip:asterisk at 10.10.0.6>;tag=as3c8e3a61
To: <sip:10.10.0.110>
Contact: <sip:asterisk at 10.10.0.6>
Call-ID: 7cde84ff2190c25f12ebe7a07a8f98d7 at 10.10.0.6
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Wed, 20 Oct 2004 18:23:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0

 (no NAT) to 10.10.0.110:5060
impbx01*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK40f5a9b2
From: "asterisk" <sip:asterisk at 10.10.0.6>;tag=as3c8e3a61
To: <sip:10.10.0.110>;tag=001193d886a3000d18ef9b5b-6e8f8846
Call-ID: 7cde84ff2190c25f12ebe7a07a8f98d7 at 10.10.0.6
CSeq: 102 OPTIONS
Server: CSCO/7
Content-Type: application/sdp
Content-Length: 233
Allow: OPTIONS,INVITE,BYE,CANCEL,REGISTER,ACK,NOTIFY,REFER

v=0
o=Cisco-SIPUA 0 0 IN IP4 10.10.0.110
s=SIP Call
c=IN IP4 10.10.0.110
t=0 0
m=audio 1 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

10 headers, 11 lines