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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 24.33.239.118:5060;branch=z9hG4bK149f400a
From: "John Fraizer 100" <sip:[EMAIL PROTECTED]>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
Date: Sat, 07 Feb 2004 22:57:46 GMT
CSeq: 101 INVITE
User-Agent: CSCO/6
Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:[EMAIL PROTECTED]>;tag=as0638308b
Call-ID: [EMAIL PROTECTED]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Proxy-Authenticate: Digest realm="asterisk", nonce="4844d22f"
Content-Length: 0
to 24.33.239.118:5060 Border2*CLI>
Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK3579a8a9
From: "John Fraizer 100" <sip:[EMAIL PROTECTED]>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:[EMAIL PROTECTED]>;tag=as0638308b
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 24.33.239.118:5060;branch=z9hG4bK3f142fb1
From: "John Fraizer 100" <sip:[EMAIL PROTECTED]>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
Date: Sat, 07 Feb 2004 22:57:46 GMT
CSeq: 102 INVITE
User-Agent: CSCO/6
Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:[EMAIL PROTECTED]>;tag=as4cba15e7
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
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:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK1aed7dc4 From: "John Fraizer 100" <sip:[EMAIL PROTECTED]>;tag=as7e10d688 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;tag=as7e10d688 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] Date: Sat, 07 Feb 2004 22:57:47 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=as7e10d688 To: <sip:[EMAIL PROTECTED]>;tag=000bbe40419b00544779520d-53ff391d Call-ID: [EMAIL PROTECTED] Date: Sat, 07 Feb 2004 22:57:47 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:[EMAIL PROTECTED]>;tag=as4cba15e7
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
to 24.33.239.118:5060 Border2*CLI>
Sip read:
CANCEL sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 24.33.239.118:5060;branch=z9hG4bK0637d0a0
From: "John Fraizer 100" <sip:[EMAIL PROTECTED]>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:[EMAIL PROTECTED]>;tag=as4cba15e7
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
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:[EMAIL PROTECTED]>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:[EMAIL PROTECTED]>;tag=as4cba15e7
Call-ID: [EMAIL PROTECTED]
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
to 24.33.239.118:5060 Reliably Transmitting: CANCEL sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK1aed7dc4 From: "John Fraizer 100" <sip:[EMAIL PROTECTED]>;tag=as7e10d688 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;tag=as7e10d688 To: <sip:[EMAIL PROTECTED]>;tag=000bbe40419b00544779520d-53ff391d Call-ID: [EMAIL PROTECTED] Date: Sat, 07 Feb 2004 22:57:48 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060>
set_destination: Parsing <sip:[EMAIL PROTECTED]:5060> for address/port to send to
set_destination: set destination to 24.33.239.118, port 5060
Transmitting:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK1aed7dc4
From: "John Fraizer 100" <sip:[EMAIL PROTECTED]>;tag=as7e10d688
To: <sip:[EMAIL PROTECTED]>;tag=000bbe40419b00544779520d-53ff391d
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(NAT) to 24.33.239.118:5060 Border2*CLI>
Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK745c6f34
From: "John Fraizer 100" <sip:[EMAIL PROTECTED]>;tag=000bbe40419b00532a4215e9-779f0059
To: <sip:[EMAIL PROTECTED]>;tag=as4cba15e7
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]>;tag=as7e10d688 To: <sip:[EMAIL PROTECTED]>;tag=000bbe40419b00544779520d-53ff391d Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;tag=as7e10d688 To: <sip:[EMAIL PROTECTED]>;tag=000bbe40419b00544779520d-53ff391d Call-ID: [EMAIL PROTECTED] Date: Sat, 07 Feb 2004 22:57:48 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:[EMAIL PROTECTED]:5060> Content-Length: 0
10 headers, 0 lines
set_destination: Parsing <sip:[EMAIL PROTECTED]:5060> for address/port to send to
set_destination: set destination to 24.33.239.118, port 5060
Transmitting:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK1aed7dc4
From: "John Fraizer 100" <sip:[EMAIL PROTECTED]>;tag=as7e10d688
To: <sip:[EMAIL PROTECTED]>;tag=000bbe40419b00544779520d-53ff391d
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 24.33.239.118:15060;rport;branch=z9hG4bK50A16FE0C159D811866900022D691075
From: John Fraizer <sip:[EMAIL PROTECTED]>;tag=2208077544
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]:15060>
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]>;tag=2208077544
To: <sip:[EMAIL PROTECTED]>;tag=as223e598a
Call-ID: [EMAIL PROTECTED]
CSeq: 6445 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Proxy-Authenticate: Digest realm="asterisk", nonce="1a4190c5"
Content-Length: 0
to 24.33.239.118:15060 Border2*CLI>
Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 24.33.239.118:15060;rport;branch=z9hG4bK50A16FE0C159D811866900022D691075
From: John Fraizer <sip:[EMAIL PROTECTED]>;tag=2208077544
To: <sip:[EMAIL PROTECTED]>;tag=as223e598a
Contact: <sip:[EMAIL PROTECTED]:15060>
Call-ID: [EMAIL PROTECTED]
CSeq: 6445 ACK
Max-Forwards: 70
Content-Length: 0
9 headers, 0 lines Border2*CLI>
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 24.33.239.118:15060;rport;branch=z9hG4bKCEE291E0C159D811866900022D691075
From: John Fraizer <sip:[EMAIL PROTECTED]>;tag=2208077544
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]:15060>
Call-ID: [EMAIL PROTECTED]
CSeq: 6446 INVITE
Proxy-Authorization: Digest username="2222",realm="asterisk",nonce="1a4190c5",response="e212a3ad53c067407873952eaaa7755f",uri="sip:[EMAIL PROTECTED]"
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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=2208077544
To: <sip:[EMAIL PROTECTED]>;tag=as4aff8ad3
Call-ID: [EMAIL PROTECTED]
CSeq: 6446 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
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:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK3e5a357c From: "John Fraizer" <sip:[EMAIL PROTECTED]>;tag=as4adb7fc6 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;tag=as4adb7fc6 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] Date: Sat, 07 Feb 2004 23:03:44 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=as4adb7fc6 To: <sip:[EMAIL PROTECTED]>;tag=000bbe40419b00551296115b-34fa118e Call-ID: [EMAIL PROTECTED] Date: Sat, 07 Feb 2004 23:03:44 GMT CSeq: 102 INVITE Server: CSCO/6 Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=2208077544
To: <sip:[EMAIL PROTECTED]>;tag=as4aff8ad3
Call-ID: [EMAIL PROTECTED]
CSeq: 6446 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
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
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