Apparently, there is a SIP(diversionheader) field that fixes the problem
below, but I cannot find any docs or examples of how to use it in my
dialplan. Any help would be appreciated. We have a Cisco CallManager
where users forward their numbers, so PSTN->PSTN calls get this error...
-Greg
<--- SIP read from 209.253.136.204:5060 --->
INVITE sip:9723814678 at 209.33.163.37;transport=UDP SIP/2.0
From: "Cell Phone
TX"<sip:2142080740 at
209.253.136.204>;tag=501ff0a-13c4-4807c9fe-a5bac2-7f73fc7a
To: "CISTERA 9723814678"<sip:9723814678 at mcleodusa>
Call-ID:
CXC-34-68a7e0e0-501ff0a-13c4-4807c9fe-a5bac2-72ed9e9c at 209.253.136.204
CSeq: 1 INVITE
Via: SIP/2.0/UDP
209.253.136.204:5060;branch=z9hG4bK-301-4807c9fe-a5bac2-5babdc2b
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY
Supported: timer
Accept: multipart/mixed,application/media_control+xml,application/sdp
Max-Forwards: 9
Min-SE: 60
Contact: <sip:2142080740 at 209.253.136.204:5060;transport=UDP>
Content-Type: application/sdp
Content-Length: 500
v=0
o=BroadWorks 31324769 1 IN IP4 209.253.136.204
s=-
c=IN IP4 209.253.136.204
t=0 0
m=audio 24418 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=x-cxc-sess:04c2e65cf9a2aa97-1
a=x-cxc-info:cGVlci1wdWI9NjQuMTk5LjUxLjIxMDtwZWVyLXNkcD0yMDkuMjUzLjEyOS4xNTc6MjYzMDY7
a=x-cxc-info:cGVlci1yb3V0ZS10YWc9aW50ZXJuYWw7YW5jaG9yLWRzdD0yMDkuMjUzLjEzNi4yMDQ6MjQ0MTg7
a=sendrecv
<------------->
--- (14 headers 17 lines) ---
Sending to 209.253.136.204 : 5060 (no NAT)
Using INVITE request as basis request -
CXC-34-68a7e0e0-501ff0a-13c4-4807c9fe-a5bac2-72ed9e9c at 209.253.136.204
Found peer 'McLeodUSA'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 209.253.136.204:24418
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|
g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 209.253.136.204:24418
Looking for 9723814678 in default (domain 209.33.163.37)
list_route: hop: <sip:2142080740 at 209.253.136.204:5060;transport=UDP>
ns2*CLI>
<--- Transmitting (no NAT) to 209.253.136.204:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
209.253.136.204:5060;branch=z9hG4bK-301-4807c9fe-a5bac2-5babdc2b;received=209.253.136.204
From: "Cell Phone
TX"<sip:2142080740 at
209.253.136.204>;tag=501ff0a-13c4-4807c9fe-a5bac2-7f73fc7a
To: "CISTERA 9723814678"<sip:9723814678 at mcleodusa>
Call-ID:
CXC-34-68a7e0e0-501ff0a-13c4-4807c9fe-a5bac2-72ed9e9c at 209.253.136.204
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9723814678 at 209.33.163.37>
Content-Length: 0
<------------>
-- Executing [9723814678 at default:1]
Dial("SIP/4693412073-08fdbf78",
"SIP/4678 at 192.168.5.10") in new stack
Audio is at 192.168.5.14 port 13374
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.5.10:5060:
INVITE sip:4678 at 192.168.5.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.14:5060;branch=z9hG4bK4a44607b;rport
From: "Cell Phone TX" <sip:2142080740 at
192.168.5.14>;tag=as178544f0
To: <sip:4678 at 192.168.5.10>
Contact: <sip:2142080740 at 192.168.5.14>
Call-ID: 6eaabaa87a62e3f81a8effcf3cc7fce6 at 192.168.5.14
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 17 Apr 2008 22:08:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 28662 28662 IN IP4 192.168.5.14
s=session
c=IN IP4 192.168.5.14
t=0 0
m=audio 13374 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 4678 at 192.168.5.10
ns2*CLI>
<--- SIP read from 192.168.5.10:49365 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.5.14:5060;branch=z9hG4bK4a44607b;rport
From: "Cell Phone TX" <sip:2142080740 at
192.168.5.14>;tag=as178544f0
To: <sip:4678 at 192.168.5.10>;tag=16863906
Date: Thu, 17 Apr 2008 22:06:54 GMT
Call-ID: 6eaabaa87a62e3f81a8effcf3cc7fce6 at 192.168.5.14
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
ns2*CLI>
<--- SIP read from 192.168.5.10:6060 --->
INVITE sip:18005551212 at 192.168.5.14:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.10:6060;branch=z9hG4bK32426484
From: "Cell Phone TX" <sip:2142080740 at
192.168.5.10>;tag=16863908
To: <sip:18005551212 at 192.168.5.14>
Date: Thu, 17 Apr 2008 22:06:55 GMT
Call-ID: 95f1af00-1de12acd-10153-a05a8c0 at 192.168.5.10
Supported: timer
Min-SE: 1800
User-Agent: Cisco-CCM4.1
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: "Cell Phone TX"
<sip:2142080740 at 192.168.5.10>;party=calling;screen=no;privacy=off
Contact: <sip:2142080740 at 192.168.5.10:6060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 227
v=0
o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 192.168.5.10
s=SIP Call
c=IN IP4 192.168.5.10
t=0 0
m=audio 29150 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (18 headers 11 lines) ---
Sending to 192.168.5.10 : 6060 (no NAT)
Using INVITE request as basis request -
95f1af00-1de12acd-10153-a05a8c0 at 192.168.5.10
Found peer 'Publisher'
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.5.10:29150
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.5.10:29150
Looking for 18005551212 in default (domain 192.168.5.14)
list_route: hop: <sip:2142080740 at 192.168.5.10:6060>
<--- Transmitting (no NAT) to 192.168.5.10:6060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.5.10:6060;branch=z9hG4bK32426484;received=192.168.5.10
From: "Cell Phone TX" <sip:2142080740 at
192.168.5.10>;tag=16863908
To: <sip:18005551212 at 192.168.5.14>
Call-ID: 95f1af00-1de12acd-10153-a05a8c0 at 192.168.5.10
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:18005551212 at 192.168.5.14>
Content-Length: 0
<------------>
-- Executing [18005551212 at default:1]
Dial("SIP/192.168.5.10-08ff1690", "SIP/18005551212 at
McLeodUSA") in new
stack
Audio is at 209.33.163.37 port 11122
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 209.253.136.204:5060:
INVITE sip:18005551212 at bwas2.global.voip.mcleodusa.net SIP/2.0
Via: SIP/2.0/UDP 209.33.163.37:5060;branch=z9hG4bK005731d8;rport
From: "Cell Phone TX"
<sip:2142080740 at bwas2.global.voip.mcleodusa.net>;tag=as370c4a2f
To: <sip:18005551212 at bwas2.global.voip.mcleodusa.net>
Contact: <sip:2142080740 at 209.33.163.37>
Call-ID:
672fb64337c1f37018257b280c95c194 at bwas2.global.voip.mcleodusa.net
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 17 Apr 2008 22:08:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 242
v=0
o=root 28662 28662 IN IP4 209.33.163.37
s=session
c=IN IP4 209.33.163.37
t=0 0
m=audio 11122 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 18005551212 at McLeodUSA
ns2*CLI>
<--- SIP read from 209.253.136.204:5060 --->
SIP/2.0 100 Trying
From: "Cell Phone
TX"<sip:2142080740 at bwas2.global.voip.mcleodusa.net>;tag=as370c4a2f
To:
<sip:18005551212 at
bwas2.global.voip.mcleodusa.net>;tag=501ff0a-13c4-4807c9fe-a5bb83-2b952534
Call-ID:
672fb64337c1f37018257b280c95c194 at bwas2.global.voip.mcleodusa.net
CSeq: 102 INVITE
Via: SIP/2.0/UDP 209.33.163.37:5060;rport=5060;branch=z9hG4bK005731d8
Contact: <sip:18005551212 at 209.253.136.204:5060;transport=UDP>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
ns2*CLI>
<--- SIP read from 209.253.136.204:5060 --->
SIP/2.0 604 Does not exist anywhere
From: "Cell Phone
TX"<sip:2142080740 at bwas2.global.voip.mcleodusa.net>;tag=as370c4a2f
To:
<sip:18005551212 at
bwas2.global.voip.mcleodusa.net>;tag=501ff0a-13c4-4807c9fe-a5bb83-2b952534
Call-ID:
672fb64337c1f37018257b280c95c194 at bwas2.global.voip.mcleodusa.net
CSeq: 102 INVITE
Via: SIP/2.0/UDP 209.33.163.37:5060;rport=5060;branch=z9hG4bK005731d8
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
-- Got SIP response 604 "Does not exist anywhere" back from
209.253.136.204
Transmitting (no NAT) to 209.253.136.204:5060:
ACK sip:18005551212 at bwas2.global.voip.mcleodusa.net SIP/2.0
Via: SIP/2.0/UDP 209.33.163.37:5060;branch=z9hG4bK005731d8;rport
From: "Cell Phone TX"
<sip:2142080740 at bwas2.global.voip.mcleodusa.net>;tag=as370c4a2f
To:
<sip:18005551212 at
bwas2.global.voip.mcleodusa.net>;tag=501ff0a-13c4-4807c9fe-a5bb83-2b952534
Contact: <sip:2142080740 at 209.33.163.37>
Call-ID:
672fb64337c1f37018257b280c95c194 at bwas2.global.voip.mcleodusa.net
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/192.168.5.10-08ff1690' status is
'CHANUNAVAIL'
<--- Transmitting (no NAT) to 192.168.5.10:6060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP
192.168.5.10:6060;branch=z9hG4bK32426484;received=192.168.5.10
From: "Cell Phone TX" <sip:2142080740 at
192.168.5.10>;tag=16863908
To: <sip:18005551212 at 192.168.5.14>;tag=as07fb43c0
Call-ID: 95f1af00-1de12acd-10153-a05a8c0 at 192.168.5.10
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:18005551212 at 192.168.5.14>
Content-Length: 0
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
<------------>
ns2*CLI>
<--- SIP read from 192.168.5.10:6060 --->
ACK sip:18005551212 at 192.168.5.14:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.10:6060;branch=z9hG4bK32426484
From: "Cell Phone TX" <sip:2142080740 at
192.168.5.10>;tag=16863908
To: <sip:18005551212 at 192.168.5.14>;tag=as07fb43c0
Date: Thu, 17 Apr 2008 22:06:55 GMT
Call-ID: 95f1af00-1de12acd-10153-a05a8c0 at 192.168.5.10
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog
'672fb64337c1f37018257b280c95c194 at bwas2.global.voip.mcleodusa.net'
Method: INVITE
Really destroying SIP dialog
'95f1af00-1de12acd-10153-a05a8c0 at 192.168.5.10' Method: ACK
ns2*CLI>
<--- SIP read from 192.168.5.10:49365 --->
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 192.168.5.14:5060;branch=z9hG4bK4a44607b;rport
From: "Cell Phone TX" <sip:2142080740 at
192.168.5.14>;tag=as178544f0
To: <sip:4678 at 192.168.5.10>;tag=16863906
Date: Thu, 17 Apr 2008 22:06:54 GMT
Call-ID: 6eaabaa87a62e3f81a8effcf3cc7fce6 at 192.168.5.14
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
-- Got SIP response 500 "Internal Server Error" back from
192.168.5.10
Transmitting (no NAT) to 192.168.5.10:5060:
ACK sip:4678 at 192.168.5.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.14:5060;branch=z9hG4bK4a44607b;rport
From: "Cell Phone TX" <sip:2142080740 at
192.168.5.14>;tag=as178544f0
To: <sip:4678 at 192.168.5.10>;tag=16863906
Contact: <sip:2142080740 at 192.168.5.14>
Call-ID: 6eaabaa87a62e3f81a8effcf3cc7fce6 at 192.168.5.14
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/192.168.5.10-08fd84c0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/4693412073-08fdbf78' status is
'CONGESTION'
<--- Transmitting (no NAT) to 209.253.136.204:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP
209.253.136.204:5060;branch=z9hG4bK-301-4807c9fe-a5bac2-5babdc2b;received=209.253.136.204
From: "Cell Phone
TX"<sip:2142080740 at
209.253.136.204>;tag=501ff0a-13c4-4807c9fe-a5bac2-7f73fc7a
To: "CISTERA 9723814678"<sip:9723814678 at
mcleodusa>;tag=as2df2b0f6
Call-ID:
CXC-34-68a7e0e0-501ff0a-13c4-4807c9fe-a5bac2-72ed9e9c at 209.253.136.204
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9723814678 at 209.33.163.37>
Content-Length: 0
X-Asterisk-HangupCause: Network out of order
X-Asterisk-HangupCauseCode: 38
<------------>
ns2*CLI>
<--- SIP read from 209.253.136.204:5060 --->
ACK sip:9723814678 at 209.33.163.37;transport=UDP SIP/2.0
From: "Cell Phone
TX"<sip:2142080740 at
209.253.136.204>;tag=501ff0a-13c4-4807c9fe-a5bac2-7f73fc7a
To: "CISTERA 9723814678"<sip:9723814678 at
mcleodusa>;tag=as2df2b0f6
Call-ID:
CXC-34-68a7e0e0-501ff0a-13c4-4807c9fe-a5bac2-72ed9e9c at 209.253.136.204
CSeq: 1 ACK
Via: SIP/2.0/UDP
209.253.136.204:5060;branch=z9hG4bK-301-4807c9fe-a5bac2-5babdc2b
Max-Forwards: 69
Contact: <sip:2142080740 at 209.253.136.204:5060;transport=UDP>
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog
'6eaabaa87a62e3f81a8effcf3cc7fce6 at 192.168.5.14' Method: INVITE
Really destroying SIP dialog
'CXC-34-68a7e0e0-501ff0a-13c4-4807c9fe-a5bac2-72ed9e9c at
209.253.136.204'
Method: ACK
ns2*CLI>
<--- SIP read from 209.253.136.204:5060 --->
INVITE sip:9723814678 at 209.33.163.37;transport=UDP SIP/2.0
From: "Cell Phone
TX"<sip:2142080740 at
209.253.136.204>;tag=501ff0a-13c4-4807c9fe-a5bc2c-1049ecf6
To: "CISTERA 9723814678"<sip:9723814678 at mcleodusa>
Call-ID:
CXC-123-68a805e0-501ff0a-13c4-4807c9fe-a5bc2c-418ea3b8 at 209.253.136.204
CSeq: 1 INVITE
Via: SIP/2.0/UDP
209.253.136.204:5060;branch=z9hG4bK-303-4807c9fe-a5bc2c-31f108f7
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY
Supported: timer
Accept: multipart/mixed,application/media_control+xml,application/sdp
Max-Forwards: 9
Min-SE: 60
Contact: <sip:2142080740 at 209.253.136.204:5060;transport=UDP>
Content-Type: application/sdp
Content-Length: 502
v=0
o=BroadWorks 208497 1 IN IP4 209.253.136.204
s=-
c=IN IP4 209.253.136.204
t=0 0
m=audio 24424 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=x-cxc-sess:04c2e65cf9b3e3a0-1
a=x-cxc-info:cGVlci1wdWI9MjA5LjI1My4xMzQuNjM7cGVlci1zZHA9MjA5LjI1My4xMjkuMTU3OjI2MzA2Ow=a=x-cxc-info:cGVlci1yb3V0ZS10YWc9aW50ZXJuYWw7YW5jaG9yLWRzdD0yMDkuMjUzLjEzNi4yMDQ6MjQ0MjQ7
a=sendrecv
<------------->
--- (14 headers 17 lines) ---
Sending to 209.253.136.204 : 5060 (no NAT)
Using INVITE request as basis request -
CXC-123-68a805e0-501ff0a-13c4-4807c9fe-a5bc2c-418ea3b8 at 209.253.136.204
Found peer 'McLeodUSA'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 209.253.136.204:24424
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|
g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 209.253.136.204:24424
Looking for 9723814678 in default (domain 209.33.163.37)
list_route: hop: <sip:2142080740 at 209.253.136.204:5060;transport=UDP>
<--- Transmitting (no NAT) to 209.253.136.204:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
209.253.136.204:5060;branch=z9hG4bK-303-4807c9fe-a5bc2c-31f108f7;received=209.253.136.204
From: "Cell Phone
TX"<sip:2142080740 at
209.253.136.204>;tag=501ff0a-13c4-4807c9fe-a5bc2c-1049ecf6
To: "CISTERA 9723814678"<sip:9723814678 at mcleodusa>
Call-ID:
CXC-123-68a805e0-501ff0a-13c4-4807c9fe-a5bc2c-418ea3b8 at 209.253.136.204
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9723814678 at 209.33.163.37>
Content-Length: 0
<------------>
-- Executing [9723814678 at default:1]
Dial("SIP/4693412073-08fdbf78",
"SIP/4678 at 192.168.5.10") in new stack
Audio is at 192.168.5.14 port 15932
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.5.10:5060:
INVITE sip:4678 at 192.168.5.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.14:5060;branch=z9hG4bK5d9871a5;rport
From: "Cell Phone TX" <sip:2142080740 at
192.168.5.14>;tag=as09ed2d26
To: <sip:4678 at 192.168.5.10>
Contact: <sip:2142080740 at 192.168.5.14>
Call-ID: 37d8007908e2a8ee5aadcded77f3bc0d at 192.168.5.14
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 17 Apr 2008 22:08:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 28662 28662 IN IP4 192.168.5.14
s=session
c=IN IP4 192.168.5.14
t=0 0
m=audio 15932 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv