[Asterisk-Users] Iconnect Incomming calls
Glenn Dalgliesh
asterisk at techhat.com
Tue Oct 7 08:18:21 MST 2003
I have an IconnectHere account with a Inbound number and have setup the sip.conf to register and am recieving the call but When I answer the call it disconnect. I have tried sending the call to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon as I accept the call it disconnects. I believe it may be some type of codec issue but I am not very familiar with that layer.
Below is the .conf's & SIP debug
Thank for any help....
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = sipinbound ; Default for incoming calls
register => 1410344xxxx:yyyy at 213.137.73.176/1410344xxxx
--=-=-=-= extentions.conf-=-=-=-=-=- have also tried sip phone same results
[sipinbound]
Exten => _.,1,Dial,Zap/5-1
-=-=-=-=-=-=-=-=-=- upgraded to lastest cvs with same results
pbx1*CLI> show version
Asterisk CVS-10/03/03-13:40:08 built by root at pbx1.routerboy.com on a i686 running Linux
-=-=-=-=-=-=-=-=-=-=
pbx1*CLI>
Sip read:
INVITE sip:14103445557 at 162.33.165.198 SIP/2.0
Record-Route: <sip:14103445557 at 213.137.73.178:5060;maddr=213.137.73.176>
Via: SIP/2.0/UDP 213.137.73.176:5060;branch=7fdf8984-233992d8-bccd43ce-d4d3adac-1
Via: SIP/2.0/UDP 213.137.65.234:5060
From: <sip:4103532264 at 213.137.65.234>;tag=17B49340-1BD5
To: <sip:14103445557 at 213.137.73.178>
Date: Fri, 03 Oct 2003 20:37:52 GMT
Call-ID: 4B86D860-F51811D7-94A4DA91-FCF3ECE5 at 213.137.65.234
Supported: timer,100rel
Min-SE: 1800
Cisco-Guid: 1267048311-4111995351-2493635217-4243844325
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 9
Remote-Party-ID: <sip:4103532264 at 213.137.65.234>;party=calling;screen=yes;privacy=off
Timestamp: 1065213472
Contact: <sip:4103532264 at 213.137.65.234:5060>
Diversion: <sip:4103445557 at 213.137.65.234>;reason=unconditional
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 332
v=0
o=CiscoSystemsSIP-GW-UserAgent 7043 3136 IN IP4 213.137.65.234
s=SIP Call
c=IN IP4 213.137.65.234
t=0 0
m=audio 16826 RTP/AVP 4 18 101 19
c=IN IP4 213.137.65.234
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
23 headers, 14 lines
Using latest request as basis request
Sending to 213.137.73.176 : 5060 (non-NAT)
Found audio format ULAW
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found description format G723
Found description format G729
Found description format telephone-event
Found description format CN
Capabilities: us - 524302, them - 257/0, combined - 0
Non-codec capabilities: us - 1, them - 3, combined - 1
Sip read:
INVITE sip:14103445557 at 162.33.165.198 SIP/2.0
Record-Route: <sip:14103445557 at 213.137.73.178:5060;maddr=213.137.73.176>
Via: SIP/2.0/UDP 213.137.73.176:5060;branch=7fdf8984-233992d8-bccd43ce-d4d3adac-1
Via: SIP/2.0/UDP 213.137.65.234:5060
From: <sip:4103532264 at 213.137.65.234>;tag=17B49340-1BD5
To: <sip:14103445557 at 213.137.73.178>
Date: Fri, 03 Oct 2003 20:37:52 GMT
Call-ID: 4B86D860-F51811D7-94A4DA91-FCF3ECE5 at 213.137.65.234
Supported: timer,100rel
Min-SE: 1800
Cisco-Guid: 1267048311-4111995351-2493635217-4243844325
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 9
Remote-Party-ID: <sip:4103532264 at 213.137.65.234>;party=calling;screen=yes;privacy=off
Timestamp: 1065213472
Contact: <sip:4103532264 at 213.137.65.234:5060>
Diversion: <sip:4103445557 at 213.137.65.234>;reason=unconditional
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 332
v=0
o=CiscoSystemsSIP-GW-UserAgent 7043 3136 IN IP4 213.137.65.234
s=SIP Call
c=IN IP4 213.137.65.234
t=0 0
m=audio 16826 RTP/AVP 4 18 101 19
c=IN IP4 213.137.65.234
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
23 headers, 14 lines
Ignoring this request
Looking for 14103445557 in sipinbound
RDNIS is 4103445557
list_route: hop: <sip:14103445557 at 213.137.73.178:5060;maddr=213.137.73.176>
list_route: hop: <sip:4103532264 at 213.137.65.234:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.137.73.176:5060;branch=7fdf8984-233992d8-bccd43ce-d4d3adac-1
Via: SIP/2.0/UDP 213.137.65.234:5060
From: <sip:4103532264 at 213.137.65.234>;tag=17B49340-1BD5
To: <sip:14103445557 at 213.137.73.178>;tag=as72f8d457
Call-ID: 4B86D860-F51811D7-94A4DA91-FCF3ECE5 at 213.137.65.234
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:14103445557 at 162.33.165.198>
Content-Length: 0
to 213.137.73.176:5060
-- Executing Dial("SIP/-080e9768", "Zap/5-1") in new stack
-- Called 5-1
-- Zap/5-1 is ringing
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 213.137.73.176:5060;branch=7fdf8984-233992d8-bccd43ce-d4d3adac-1
Via: SIP/2.0/UDP 213.137.65.234:5060
From: <sip:4103532264 at 213.137.65.234>;tag=17B49340-1BD5
To: <sip:14103445557 at 213.137.73.178>;tag=as72f8d457
Call-ID: 4B86D860-F51811D7-94A4DA91-FCF3ECE5 at 213.137.65.234
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:14103445557 at 162.33.165.198>
Content-Length: 0
to 213.137.73.176:5060
-- Zap/5-1 is ringing
-- Zap/5-1 answered SIP/-080e9768
We're at 162.33.165.198 port 17288
Answering with non-codec capability 1
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.137.73.176:5060;branch=7fdf8984-233992d8-bccd43ce-d4d3adac-1
Via: SIP/2.0/UDP 213.137.65.234:5060
Record-Route: <sip:14103445557 at 213.137.73.178:5060;maddr=213.137.73.176>
From: <sip:4103532264 at 213.137.65.234>;tag=17B49340-1BD5
To: <sip:14103445557 at 213.137.73.178>;tag=as72f8d457
Call-ID: 4B86D860-F51811D7-94A4DA91-FCF3ECE5 at 213.137.65.234
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:14103445557 at 162.33.165.198>
Content-Type: application/sdp
Content-Length: 167
v=0
o=root 1387 1387 IN IP4 162.33.165.198
s=session
c=IN IP4 162.33.165.198
t=0 0
m=audio 17288 RTP/AVP 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
to 213.137.73.176:5060
Sip read:
ACK sip:14103445557 at 162.33.165.198:5060 SIP/2.0
Record-Route: <sip:14103445557 at 213.137.73.178:5060;maddr=213.137.73.176>
Via: SIP/2.0/UDP 213.137.73.176:5060;branch=b8153fe5-dbaec9a8-33f7ea42-b21b1257-1
Via: SIP/2.0/UDP 213.137.65.234:5060
From: <sip:4103532264 at 213.137.65.234>;tag=17B49340-1BD5
To: <sip:14103445557 at 213.137.73.178>;tag=as72f8d457
Date: Fri, 03 Oct 2003 20:37:52 GMT
Call-ID: 4B86D860-F51811D7-94A4DA91-FCF3ECE5 at 213.137.65.234
Max-Forwards: 9
Content-Length: 0
CSeq: 101 ACK
11 headers, 0 lines
-- Hungup 'Zap/5-1'
== Spawn extension (sipinbound, 14103445557, 1) exited non-zero on 'SIP/-080e9768'
-- Executing Dial("SIP/-080e9768", "Zap/5-1") in new stack
Sip read:
BYE sip:14103445557 at 162.33.165.198:5060 SIP/2.0
Record-Route: <sip:14103445557 at 213.137.73.178:5060;maddr=213.137.73.176>
Via: SIP/2.0/UDP 213.137.73.176:5060;branch=ad138698-b8834c00-64f04b6f-b958ea28-1
Via: SIP/2.0/UDP 213.137.65.234:5060
From: <sip:4103532264 at 213.137.65.234>;tag=17B49340-1BD5
To: <sip:14103445557 at 213.137.73.178>;tag=as72f8d457
Date: Fri, 03 Oct 2003 20:37:52 GMT
Call-ID: 4B86D860-F51811D7-94A4DA91-FCF3ECE5 at 213.137.65.234
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 9
Timestamp: 1065213477
CSeq: 102 BYE
Content-Length: 0
13 headers, 0 lines
Sending to 213.137.73.176 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.137.73.176:5060;branch=ad138698-b8834c00-64f04b6f-b958ea28-1
Via: SIP/2.0/UDP 213.137.65.234:5060
Record-Route: <sip:14103445557 at 213.137.73.178:5060;maddr=213.137.73.176>
From: <sip:4103532264 at 213.137.65.234>;tag=17B49340-1BD5
To: <sip:14103445557 at 213.137.73.178>;tag=as72f8d457
Call-ID: 4B86D860-F51811D7-94A4DA91-FCF3ECE5 at 213.137.65.234
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0
to 213.137.73.176:5060
== Everyone is busy at this time
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:213.137.73.176 SIP/2.0
Via: SIP/2.0/UDP 162.33.165.198:5060;branch=z9hG4bK66367b26
From: <sip:14103445557 at 213.137.73.176>;tag=as4299840a
To: <sip:14103445557 at 213.137.73.176>
Call-ID: 7e01e2fb05dced9e46b4815671a0e597 at 162.33.165.198
CSeq: 110 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:14103445557 at 162.33.165.198>
Event: registration
Content-length: 0
(no NAT) to 213.137.73.176:5060
Retransmitting #1 (no NAT):
REGISTER sip:213.137.73.176 SIP/2.0
Via: SIP/2.0/UDP 162.33.165.198:5060;branch=z9hG4bK66367b26
From: <sip:14103445557 at 213.137.73.176>;tag=as4299840a
To: <sip:14103445557 at 213.137.73.176>
Call-ID: 7e01e2fb05dced9e46b4815671a0e597 at 162.33.165.198
CSeq: 110 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:14103445557 at 162.33.165.198>
Event: registration
Content-length: 0
to 213.137.73.176:5060
Retransmitting #2 (no NAT):
REGISTER sip:213.137.73.176 SIP/2.0
Via: SIP/2.0/UDP 162.33.165.198:5060;branch=z9hG4bK66367b26
From: <sip:14103445557 at 213.137.73.176>;tag=as4299840a
To: <sip:14103445557 at 213.137.73.176>
Call-ID: 7e01e2fb05dced9e46b4815671a0e597 at 162.33.165.198
CSeq: 110 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:14103445557 at 162.33.165.198>
Event: registration
Content-length: 0
to 213.137.73.176:5060
Retransmitting #3 (no NAT):
REGISTER sip:213.137.73.176 SIP/2.0
Via: SIP/2.0/UDP 162.33.165.198:5060;branch=z9hG4bK66367b26
From: <sip:14103445557 at 213.137.73.176>;tag=as4299840a
To: <sip:14103445557 at 213.137.73.176>
Call-ID: 7e01e2fb05dced9e46b4815671a0e597 at 162.33.165.198
CSeq: 110 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:14103445557 at 162.33.165.198>
Event: registration
Content-length: 0
to 213.137.73.176:5060
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 162.33.165.198:5060;branch=z9hG4bK66367b26
Call-ID: 7e01e2fb05dced9e46b4815671a0e597 at 162.33.165.198
From: <sip:14103445557 at 213.137.73.176>;tag=as4299840a
To: <sip:14103445557 at 213.137.73.176>
CSeq: 110 REGISTER
Content-Length: 0
7 headers, 0 lines
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 162.33.165.198:5060;branch=z9hG4bK66367b26
Call-ID: 7e01e2fb05dced9e46b4815671a0e597 at 162.33.165.198
From: <sip:14103445557 at 213.137.73.176>;tag=as4299840a
To: <sip:14103445557 at 213.137.73.176>
CSeq: 110 REGISTER
WWW-Authenticate: DIGEST realm="deltathree.com", nonce="3f7dde70", algorithm=MD5
Content-Length: 0
8 headers, 0 lines
12 headers, 0 lines
Reliably Transmitting:
REGISTER sip:213.137.73.176 SIP/2.0
Via: SIP/2.0/UDP 162.33.165.198:5060;branch=z9hG4bK66367b26
From: <sip:14103445557 at 213.137.73.176>;tag=as4299840a
To: <sip:14103445557 at 213.137.73.176>
Call-ID: 7e01e2fb05dced9e46b4815671a0e597 at 162.33.165.198
CSeq: 111 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username="14103445557", realm="deltathree.com", algorithm="MD5", uri="sip:213.137.73.176", nonce="3f7dde70", response="6c9da63c44fce35453b1608ec7c98902"
Expires: 120
Contact: <sip:14103445557 at 162.33.165.198>
Event: registration
Content-length: 0
(no NAT) to 213.137.73.176:5060
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 162.33.165.198:5060;branch=z9hG4bK66367b26
Call-ID: 7e01e2fb05dced9e46b4815671a0e597 at 162.33.165.198
From: <sip:14103445557 at 213.137.73.176>;tag=as4299840a
To: <sip:14103445557 at 213.137.73.176>
CSeq: 111 REGISTER
Content-Length: 0
7 headers, 0 lines
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 162.33.165.198:5060;branch=z9hG4bK66367b26
Call-ID: 7e01e2fb05dced9e46b4815671a0e597 at 162.33.165.198
From: <sip:14103445557 at 213.137.73.176>;tag=as4299840a
To: <sip:14103445557 at 213.137.73.176>
CSeq: 111 REGISTER
Contact: <sip:14103445557 at 162.33.165.198>;expires="Fri, 03 Oct 2003 20:41:12 GMT"
Expires: 120
Content-Length: 0
9 headers, 0 lines
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:213.137.73.176 SIP/2.0
Via: SIP/2.0/UDP 162.33.165.198:5060;branch=z9hG4bK59b6c676
From: <sip:14103445557 at 213.137.73.176>;tag=as710c9606
To: <sip:14103445557 at 213.137.73.176>
Call-ID: 7e01e2fb05dced9e46b4815671a0e597 at 162.33.165.198
CSeq: 112 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:14103445557 at 162.33.165.198>
Event: registration
Content-length: 0
(no NAT) to 213.137.73.176:5060
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 162.33.165.198:5060;branch=z9hG4bK59b6c676
Call-ID: 7e01e2fb05dced9e46b4815671a0e597 at 162.33.165.198
From: <sip:14103445557 at 213.137.73.176>;tag=as710c9606
To: <sip:14103445557 at 213.137.73.176>
CSeq: 112 REGISTER
Content-Length: 0
7 headers, 0 lines
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 162.33.165.198:5060;branch=z9hG4bK59b6c676
Call-ID: 7e01e2fb05dced9e46b4815671a0e597 at 162.33.165.198
From: <sip:14103445557 at 213.137.73.176>;tag=as710c9606
To: <sip:14103445557 at 213.137.73.176>
CSeq: 112 REGISTER
WWW-Authenticate: DIGEST realm="deltathree.com", nonce="3f7dded9", algorithm=MD5
Content-Length: 0
8 headers, 0 lines
12 headers, 0 lines
Reliably Transmitting:
REGISTER sip:213.137.73.176 SIP/2.0
Via: SIP/2.0/UDP 162.33.165.198:5060;branch=z9hG4bK59b6c676
From: <sip:14103445557 at 213.137.73.176>;tag=as710c9606
To: <sip:14103445557 at 213.137.73.176>
Call-ID: 7e01e2fb05dced9e46b4815671a0e597 at 162.33.165.198
CSeq: 113 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username="14103445557", realm="deltathree.com", algorithm="MD5", uri="sip:213.137.73.176", nonce="3f7dded9", response="73ac8fce7657d3b459b650269a555a7e"
Expires: 120
Contact: <sip:14103445557 at 162.33.165.198>
Event: registration
Content-length: 0
(no NAT) to 213.137.73.176:5060
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 162.33.165.198:5060;branch=z9hG4bK59b6c676
Call-ID: 7e01e2fb05dced9e46b4815671a0e597 at 162.33.165.198
From: <sip:14103445557 at 213.137.73.176>;tag=as710c9606
To: <sip:14103445557 at 213.137.73.176>
CSeq: 113 REGISTER
Content-Length: 0
7 headers, 0 lines
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 162.33.165.198:5060;branch=z9hG4bK59b6c676
Call-ID: 7e01e2fb05dced9e46b4815671a0e597 at 162.33.165.198
From: <sip:14103445557 at 213.137.73.176>;tag=as710c9606
To: <sip:14103445557 at 213.137.73.176>
CSeq: 113 REGISTER
Contact: <sip:14103445557 at 162.33.165.198>;expires="Fri, 03 Oct 2003 20:42:57 GMT"
Expires: 120
Content-Length: 0
9 headers, 0 lines
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