[Asterisk-Users] Caller-ID is being sent wrong. How to fix it?

Sat Feb 7 16:08:50 MST 2004

Olle E. Johansson wrote:
> Would like to see a SIP debug
> * The invite from the caller phone to Asterisk
> * The invite from Asterisk to the called phone
> 
> As well as the configs (extensions.conf and sip.conf)
> 
> Can't reproduce in my servers.
> 
> /O

OK.  Here is a call from extension 100 to extension 2288888.

Sip read:
INVITE sip:2288888 at sipproxy.enterzone.net SIP/2.0
Via: SIP/2.0/UDP 24.33.239.118:5060;branch=z9hG4bK149f400a
From: "John Fraizer 100" 
<sip:100 at sipproxy.enterzone.net>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:2288888 at sipproxy.enterzone.net>
Call-ID: 000bbe40-419b0037-5e6c82e8-57fefd2c at 24.33.239.118
Date: Sat, 07 Feb 2004 22:57:46 GMT
CSeq: 101 INVITE
User-Agent: CSCO/6
Contact: <sip:100 at 24.33.239.118:5060>
Expires: 180
Content-Type: application/sdp
Content-Length: 249
Accept: application/sdp

v=0
o=Cisco-SIPUA 21234 22236 IN IP4 24.33.239.118
s=SIP Call
c=IN IP4 24.33.239.118
t=0 0
m=audio 18846 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

13 headers, 11 lines
Using latest request as basis request
Sending to 24.33.239.118 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 524302, them - 268/0, combined - 12
Non-codec capabilities: us - 1, them - 1, combined - 1
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 
24.33.239.118:5060;branch=z9hG4bK149f400a;received=24.33.239.118
From: "John Fraizer 100" 
<sip:100 at sipproxy.enterzone.net>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:2288888 at sipproxy.enterzone.net>;tag=as0638308b
Call-ID: 000bbe40-419b0037-5e6c82e8-57fefd2c at 24.33.239.118
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2288888 at 66.35.64.38>
Proxy-Authenticate: Digest realm="asterisk", nonce="4844d22f"
Content-Length: 0

  to 24.33.239.118:5060
Border2*CLI>

Sip read:
ACK sip:2288888 at sipproxy.enterzone.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK3579a8a9
From: "John Fraizer 100" 
<sip:100 at sipproxy.enterzone.net>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:2288888 at sipproxy.enterzone.net>;tag=as0638308b
Call-ID: 000bbe40-419b0037-5e6c82e8-57fefd2c at 24.33.239.118
Date: Sat, 07 Feb 2004 22:57:46 GMT
CSeq: 101 ACK
Content-Length: 0

8 headers, 0 lines
Border2*CLI>

Sip read:
INVITE sip:2288888 at sipproxy.enterzone.net SIP/2.0
Via: SIP/2.0/UDP 24.33.239.118:5060;branch=z9hG4bK3f142fb1
From: "John Fraizer 100" 
<sip:100 at sipproxy.enterzone.net>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:2288888 at sipproxy.enterzone.net>
Call-ID: 000bbe40-419b0037-5e6c82e8-57fefd2c at 24.33.239.118
Date: Sat, 07 Feb 2004 22:57:46 GMT
CSeq: 102 INVITE
User-Agent: CSCO/6
Contact: <sip:100 at 24.33.239.118:5060>
Proxy-Authorization: Digest 
username="100",realm="asterisk",uri="sip:66.35.64.38",response="9d7ae43306bc23bb256068b8f4044017",nonce="4844d22f",algorithm=md5
Expires: 180
Content-Type: application/sdp
Content-Length: 249

v=0
o=Cisco-SIPUA 21234 22236 IN IP4 24.33.239.118
s=SIP Call
c=IN IP4 24.33.239.118
t=0 0
m=audio 18846 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

13 headers, 11 lines
Using latest request as basis request
Sending to 24.33.239.118 : 5060 (NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 524302, them - 268/0, combined - 12
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 2288888 in allaccess
list_route: hop: <sip:100 at 24.33.239.118:5060>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
24.33.239.118:5060;branch=z9hG4bK3f142fb1;received=24.33.239.118
From: "John Fraizer 100" 
<sip:100 at sipproxy.enterzone.net>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:2288888 at sipproxy.enterzone.net>;tag=as4cba15e7
Call-ID: 000bbe40-419b0037-5e6c82e8-57fefd2c at 24.33.239.118
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2288888 at 66.35.64.38>
Content-Length: 0

  to 24.33.239.118:5060

*** THIS IS WHERE IT STARTS BREAKING ***

     -- Executing Dial("SIP/100-9284", "SIP/2288888|20") in new stack
We're at 66.35.64.38 port 10990
Answering with capability 2
Answering with capability 4
Answering with capability 8
Answering with non-codec capability 1
12 headers, 11 lines
Reliably Transmitting:
INVITE sip:2288888 at 24.33.239.118 SIP/2.0
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK1aed7dc4
From: "John Fraizer 100" <sip:2288888 at 66.35.64.38>;tag=as7e10d688
To: <sip:2288888 at 24.33.239.118>
Contact: <sip:2288888 at 66.35.64.38>
Call-ID: 39a37d1505a23f201014b6f967c1b36c at 66.35.64.38
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 07 Feb 2004 22:57:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 234

v=0
o=root 12840 12840 IN IP4 66.35.64.38
s=session
c=IN IP4 66.35.64.38
t=0 0
m=audio 10990 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
  (NAT) to 24.33.239.118:5060
     -- Called 2288888
Border2*CLI>

Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK1aed7dc4
From: "John Fraizer 100" <sip:2288888 at 66.35.64.38>;tag=as7e10d688
To: <sip:2288888 at 24.33.239.118>
Call-ID: 39a37d1505a23f201014b6f967c1b36c at 66.35.64.38
Date: Sat, 07 Feb 2004 22:57:47 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:2288888 at 24.33.239.118:5060>
Content-Length: 0

10 headers, 0 lines
Border2*CLI>

Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK1aed7dc4
From: "John Fraizer 100" <sip:2288888 at 66.35.64.38>;tag=as7e10d688
To: <sip:2288888 at 24.33.239.118>;tag=000bbe40419b00544779520d-53ff391d
Call-ID: 39a37d1505a23f201014b6f967c1b36c at 66.35.64.38
Date: Sat, 07 Feb 2004 22:57:47 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:2288888 at 24.33.239.118:5060>
Content-Length: 0

10 headers, 0 lines
     -- SIP/2288888-61b6 is ringing
Transmitting (NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 
24.33.239.118:5060;branch=z9hG4bK3f142fb1;received=24.33.239.118
From: "John Fraizer 100" 
<sip:100 at sipproxy.enterzone.net>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:2288888 at sipproxy.enterzone.net>;tag=as4cba15e7
Call-ID: 000bbe40-419b0037-5e6c82e8-57fefd2c at 24.33.239.118
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2288888 at 66.35.64.38>
Content-Length: 0

  to 24.33.239.118:5060
Border2*CLI>

Sip read:
CANCEL sip:2288888 at sipproxy.enterzone.net SIP/2.0
Via: SIP/2.0/UDP 24.33.239.118:5060;branch=z9hG4bK0637d0a0
From: "John Fraizer 100" 
<sip:100 at sipproxy.enterzone.net>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:2288888 at sipproxy.enterzone.net>
Call-ID: 000bbe40-419b0037-5e6c82e8-57fefd2c at 24.33.239.118
Date: Sat, 07 Feb 2004 22:57:47 GMT
CSeq: 102 CANCEL
User-Agent: CSCO/6
Content-Length: 0
Proxy-Authorization: Digest 
username="100",realm="asterisk",uri="sip:66.35.64.38",response="0fc86a40056de27b983aac5139698ce3",nonce="4844d22f",algorithm=md5

10 headers, 0 lines
Sending to 24.33.239.118 : 5060 (NAT)
Reliably Transmitting (NAT):
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 
24.33.239.118:5060;branch=z9hG4bK3f142fb1;received=24.33.239.118
From: "John Fraizer 100" 
<sip:100 at sipproxy.enterzone.net>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:2288888 at sipproxy.enterzone.net>;tag=as4cba15e7
Call-ID: 000bbe40-419b0037-5e6c82e8-57fefd2c at 24.33.239.118
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2288888 at 66.35.64.38>
Content-Length: 0

  to 24.33.239.118:5060
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
24.33.239.118:5060;branch=z9hG4bK0637d0a0;received=24.33.239.118
From: "John Fraizer 100" 
<sip:100 at sipproxy.enterzone.net>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:2288888 at sipproxy.enterzone.net>;tag=as4cba15e7
Call-ID: 000bbe40-419b0037-5e6c82e8-57fefd2c at 24.33.239.118
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2288888 at 66.35.64.38>
Content-Length: 0

  to 24.33.239.118:5060
Reliably Transmitting:
CANCEL sip:2288888 at 24.33.239.118:5060 SIP/2.0
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK1aed7dc4
From: "John Fraizer 100" <sip:2288888 at 66.35.64.38>;tag=as7e10d688
To: <sip:2288888 at 24.33.239.118>
Contact: <sip:2288888 at 66.35.64.38>
Call-ID: 39a37d1505a23f201014b6f967c1b36c at 66.35.64.38
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

  (NAT) to 24.33.239.118:5060
   == Spawn extension (allaccess, 2288888, 1) exited non-zero on 'SIP/100-9284'
Border2*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK1aed7dc4
From: "John Fraizer 100" <sip:2288888 at 66.35.64.38>;tag=as7e10d688
To: <sip:2288888 at 24.33.239.118>;tag=000bbe40419b00544779520d-53ff391d
Call-ID: 39a37d1505a23f201014b6f967c1b36c at 66.35.64.38
Date: Sat, 07 Feb 2004 22:57:48 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:2288888 at 24.33.239.118:5060>
Content-Type: application/sdp
Content-Length: 198

v=0
o=Cisco-SIPUA 20876 3789 IN IP4 24.33.239.118
s=SIP Call
c=IN IP4 24.33.239.118
t=0 0
m=audio 18848 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

11 headers, 9 lines
Found audio format UNKN
Found audio format UNKN
Found description format PCMU
Found description format telephone-event
Capabilities: us - 524302, them - 4/0, combined - 4
Non-codec capabilities: us - 1, them - 1, combined - 1
list_route: hop: <sip:2288888 at 24.33.239.118:5060>
set_destination: Parsing <sip:2288888 at 24.33.239.118:5060> for address/port 
to send to
set_destination: set destination to 24.33.239.118, port 5060
Transmitting:
ACK sip:2288888 at 24.33.239.118:5060 SIP/2.0
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK1aed7dc4
From: "John Fraizer 100" <sip:2288888 at 66.35.64.38>;tag=as7e10d688
To: <sip:2288888 at 24.33.239.118>;tag=000bbe40419b00544779520d-53ff391d
Contact: <sip:2288888 at 66.35.64.38>
Call-ID: 39a37d1505a23f201014b6f967c1b36c at 66.35.64.38
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

  (NAT) to 24.33.239.118:5060
Border2*CLI>

Sip read:
ACK sip:2288888 at sipproxy.enterzone.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK745c6f34
From: "John Fraizer 100" 
<sip:100 at sipproxy.enterzone.net>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:2288888 at sipproxy.enterzone.net>;tag=as4cba15e7
Call-ID: 000bbe40-419b0037-5e6c82e8-57fefd2c at 24.33.239.118
Date: Sat, 07 Feb 2004 22:57:48 GMT
CSeq: 102 ACK
Content-Length: 0

8 headers, 0 lines
Border2*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK1aed7dc4
From: "John Fraizer 100" <sip:2288888 at 66.35.64.38>;tag=as7e10d688
To: <sip:2288888 at 24.33.239.118>;tag=000bbe40419b00544779520d-53ff391d
Call-ID: 39a37d1505a23f201014b6f967c1b36c at 66.35.64.38
Date: Sat, 07 Feb 2004 22:57:48 GMT
CSeq: 102 CANCEL
Server: CSCO/6
Content-Length: 0

9 headers, 0 lines
Border2*CLI>

Sip read:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK1aed7dc4
From: "John Fraizer 100" <sip:2288888 at 66.35.64.38>;tag=as7e10d688
To: <sip:2288888 at 24.33.239.118>;tag=000bbe40419b00544779520d-53ff391d
Call-ID: 39a37d1505a23f201014b6f967c1b36c at 66.35.64.38
Date: Sat, 07 Feb 2004 22:57:48 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:2288888 at 24.33.239.118:5060>
Content-Length: 0

10 headers, 0 lines
set_destination: Parsing <sip:2288888 at 24.33.239.118:5060> for address/port 
to send to
set_destination: set destination to 24.33.239.118, port 5060
Transmitting:
ACK sip:2288888 at 24.33.239.118:5060 SIP/2.0
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK1aed7dc4
From: "John Fraizer 100" <sip:2288888 at 66.35.64.38>;tag=as7e10d688
To: <sip:2288888 at 24.33.239.118>;tag=000bbe40419b00544779520d-53ff391d
Contact: <sip:2288888 at 66.35.64.38>
Call-ID: 39a37d1505a23f201014b6f967c1b36c at 66.35.64.38
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

  (NAT) to 24.33.239.118:5060

And here's a call from 2222 to 100:

Sip read:
INVITE sip:100 at ENTERZONE SIP/2.0
Via: SIP/2.0/UDP 
24.33.239.118:15060;rport;branch=z9hG4bK50A16FE0C159D811866900022D691075
From: John Fraizer <sip:2222 at ENTERZONE>;tag=2208077544
To: <sip:100 at ENTERZONE>
Contact: <sip:2222 at 24.33.239.118:15060>
Call-ID: 9E6EC3D5-C159-D811-8669-00022D691075 at 192.168.1.105
CSeq: 6445 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite build 1101
Content-Length: 294

v=0
o=2222 3695982 3695982 IN IP4 24.33.239.118
s=X-Lite
c=IN IP4 24.33.239.118
t=0 0
m=audio 18000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

11 headers, 13 lines
Using latest request as basis request
Sending to 24.33.239.118 : 15060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 524302, them - 1550/0, combined - 14
Non-codec capabilities: us - 1, them - 1, combined - 1
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 
24.33.239.118:15060;rport;branch=z9hG4bK50A16FE0C159D811866900022D691075;received=24.33.239.118
From: John Fraizer <sip:2222 at ENTERZONE>;tag=2208077544
To: <sip:100 at ENTERZONE>;tag=as223e598a
Call-ID: 9E6EC3D5-C159-D811-8669-00022D691075 at 192.168.1.105
CSeq: 6445 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:100 at 66.35.64.38>
Proxy-Authenticate: Digest realm="asterisk", nonce="1a4190c5"
Content-Length: 0

  to 24.33.239.118:15060
Border2*CLI>

Sip read:
ACK sip:100 at ENTERZONE SIP/2.0
Via: SIP/2.0/UDP 
24.33.239.118:15060;rport;branch=z9hG4bK50A16FE0C159D811866900022D691075
From: John Fraizer <sip:2222 at ENTERZONE>;tag=2208077544
To: <sip:100 at ENTERZONE>;tag=as223e598a
Contact: <sip:2222 at 24.33.239.118:15060>
Call-ID: 9E6EC3D5-C159-D811-8669-00022D691075 at 192.168.1.105
CSeq: 6445 ACK
Max-Forwards: 70
Content-Length: 0

9 headers, 0 lines
Border2*CLI>

Sip read:
INVITE sip:100 at ENTERZONE SIP/2.0
Via: SIP/2.0/UDP 
24.33.239.118:15060;rport;branch=z9hG4bKCEE291E0C159D811866900022D691075
From: John Fraizer <sip:2222 at ENTERZONE>;tag=2208077544
To: <sip:100 at ENTERZONE>
Contact: <sip:2222 at 24.33.239.118:15060>
Call-ID: 9E6EC3D5-C159-D811-8669-00022D691075 at 192.168.1.105
CSeq: 6446 INVITE
Proxy-Authorization: Digest 
username="2222",realm="asterisk",nonce="1a4190c5",response="e212a3ad53c067407873952eaaa7755f",uri="sip:100 at ENTERZONE"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite build 1101
Content-Length: 294

v=0
o=2222 3696207 3696207 IN IP4 24.33.239.118
s=X-Lite
c=IN IP4 24.33.239.118
t=0 0
m=audio 18000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

12 headers, 13 lines
Using latest request as basis request
Sending to 24.33.239.118 : 15060 (NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 524302, them - 1550/0, combined - 14
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 100 in allaccess
list_route: hop: <sip:2222 at 24.33.239.118:15060>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
24.33.239.118:15060;rport;branch=z9hG4bKCEE291E0C159D811866900022D691075;received=24.33.239.118
From: John Fraizer <sip:2222 at ENTERZONE>;tag=2208077544
To: <sip:100 at ENTERZONE>;tag=as4aff8ad3
Call-ID: 9E6EC3D5-C159-D811-8669-00022D691075 at 192.168.1.105
CSeq: 6446 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:100 at 66.35.64.38>
Content-Length: 0

  to 24.33.239.118:15060

*** AND HERE IS WHERE IT BREAKS IN THE SAME EXACT WAY ***

     -- Executing Dial("SIP/2222-ff8b", "SIP/100|20") in new stack
We're at 66.35.64.38 port 14714
Answering with capability 2
Answering with capability 4
Answering with capability 8
Answering with non-codec capability 1
12 headers, 11 lines
Reliably Transmitting:
INVITE sip:100 at 24.33.239.118 SIP/2.0
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK3e5a357c
From: "John Fraizer" <sip:100 at 66.35.64.38>;tag=as4adb7fc6
To: <sip:100 at 24.33.239.118>
Contact: <sip:100 at 66.35.64.38>
Call-ID: 31f698dc4507bca030bdcf5c1acbd0a0 at 66.35.64.38
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 07 Feb 2004 23:03:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 234

v=0
o=root 12888 12888 IN IP4 66.35.64.38
s=session
c=IN IP4 66.35.64.38
t=0 0
m=audio 14714 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
  (NAT) to 24.33.239.118:5060
     -- Called 100
Border2*CLI>

Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK3e5a357c
From: "John Fraizer" <sip:100 at 66.35.64.38>;tag=as4adb7fc6
To: <sip:100 at 24.33.239.118>
Call-ID: 31f698dc4507bca030bdcf5c1acbd0a0 at 66.35.64.38
Date: Sat, 07 Feb 2004 23:03:44 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:100 at 24.33.239.118:5060>
Content-Length: 0

10 headers, 0 lines
Border2*CLI>

Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK3e5a357c
From: "John Fraizer" <sip:100 at 66.35.64.38>;tag=as4adb7fc6
To: <sip:100 at 24.33.239.118>;tag=000bbe40419b00551296115b-34fa118e
Call-ID: 31f698dc4507bca030bdcf5c1acbd0a0 at 66.35.64.38
Date: Sat, 07 Feb 2004 23:03:44 GMT
CSeq: 102 INVITE
Server: CSCO/6
Contact: <sip:100 at 24.33.239.118:5060>
Content-Length: 0

10 headers, 0 lines
     -- SIP/100-00e0 is ringing
Transmitting (NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 
24.33.239.118:15060;rport;branch=z9hG4bKCEE291E0C159D811866900022D691075;received=24.33.239.118
From: John Fraizer <sip:2222 at ENTERZONE>;tag=2208077544
To: <sip:100 at ENTERZONE>;tag=as4aff8ad3
Call-ID: 9E6EC3D5-C159-D811-8669-00022D691075 at 192.168.1.105
CSeq: 6446 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:100 at 66.35.64.38>
Content-Length: 0

  to 24.33.239.118:15060

Here are the configs:

;
; SIP Configuration for Asterisk
;
[general]
port = 5060                     ; Port to bind to
bindaddr = 66.35.64.38          ; Address to bind to
context = default               ; Default for incoming calls
srvlookup = yes         ; Enable SRV lookups on outbound calls

[100]
type=friend
username=100
secret=secret
host=dynamic
fromuser=100
mailbox=100
context=allaccess
canreinvite=yes
dtmfmode=rfc2833
nat=yes

[2288888]
type=friend
username=2288888
secret=secret
host=dynamic
fromuser=2288888
mailbox=100
context=allaccess
canreinvite=yes
dtmfmode=rfc2833
nat=yes

[2222]
type=friend
username=2222
secret=secret
host=dynamic
fromuser=2222
context=allaccess
canreinvite=yes
dtmfmode=rfc2833
nat=yes

;
; Static extension configuration files, used by
; the pbx_config module.
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=yes

; You can include other config files, use the #include command (without the ';')
; Note that this is different from the "include" command that includes 
contexts within
; other contexts. The #include command works in all asterisk configuration 
files.
;#include "filename.conf"

; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental 
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]

[allaccess]

exten => 100,1,Dial(SIP/100,20)
exten => 100,2,Voicemail2(u100)
exten => 100,3,Hangup
exten => 100,102,Voicemail2(b100)

exten => 2288888,1,Dial(SIP/2288888,20)
exten => 2288888,2,Voicemail2(u100)
exten => 2288888,3,Hangup
exten => 2288888,102,Voicemail2(b100)

exten => 2222,1,Dial(SIP/2222,20)
exten => 2222,2,Hangup

Note: I created a VERY simple config to test with once I determined that 
there was most likely a problem with Asterisk.

Anyone see a problem other than that the invite messages are being munged by 
Asterisk?

John