[Asterisk-Users] Iconnect Incomming calls

Glenn Dalgliesh asterisk at techhat.com
Fri Oct 3 10:17:05 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 SIP debug
 
Thank for any help....
 
 to 162.33.165.195:5060
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=6acca267-67952546-2efdf97b-e80aa0ef-1
Via: SIP/2.0/UDP  213.137.65.234:5060
From: <sip:4103532264 at 213.137.65.234>;tag=16A2EDA0-5B2
To: <sip:14103445557 at 213.137.73.178>
Date: Fri, 03 Oct 2003 15:38:58 GMT
Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5 at 213.137.65.234
Supported: timer,100rel
Min-SE: 1800
Cisco-Guid: 2316671854-4109242839-3208043153-4243844325
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: 1065195538
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 8647 4690 IN IP4 213.137.65.234
s=SIP Call
c=IN IP4 213.137.65.234
t=0 0
m=audio 16836 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 4
Found audio format 18
Found audio format 101
Found audio format 19
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=6acca267-67952546-2efdf97b-e80aa0ef-1
Via: SIP/2.0/UDP  213.137.65.234:5060
From: <sip:4103532264 at 213.137.65.234>;tag=16A2EDA0-5B2
To: <sip:14103445557 at 213.137.73.178>
Date: Fri, 03 Oct 2003 15:38:58 GMT
Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5 at 213.137.65.234
Supported: timer,100rel
Min-SE: 1800
Cisco-Guid: 2316671854-4109242839-3208043153-4243844325
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: 1065195538
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 8647 4690 IN IP4 213.137.65.234
s=SIP Call
c=IN IP4 213.137.65.234
t=0 0
m=audio 16836 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=6acca267-67952546-2efdf97b-e80aa0ef-1
Via: SIP/2.0/UDP  213.137.65.234:5060
From: <sip:4103532264 at 213.137.65.234>;tag=16A2EDA0-5B2
To: <sip:14103445557 at 213.137.73.178>;tag=as075e701d
Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5 at 213.137.65.234
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Contact: <sip:14103445557 at 162.33.165.198>
Content-Length: 0


 to 213.137.73.176:5060
    -- Executing Dial("SIP/-0810da50", "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=6acca267-67952546-2efdf97b-e80aa0ef-1
Via: SIP/2.0/UDP  213.137.65.234:5060
From: <sip:4103532264 at 213.137.65.234>;tag=16A2EDA0-5B2
To: <sip:14103445557 at 213.137.73.178>;tag=as075e701d
Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5 at 213.137.65.234
CSeq: 101 INVITE
User-Agent: Asterisk PBX
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/-0810da50
We're at 162.33.165.198 port 13196
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=6acca267-67952546-2efdf97b-e80aa0ef-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=16A2EDA0-5B2
To: <sip:14103445557 at 213.137.73.178>;tag=as075e701d
Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5 at 213.137.65.234
CSeq: 101 INVITE
User-Agent: Asterisk PBX
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 13196 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=50f2f595-1a997f3d-5142cdf4-e4672261-1
Via: SIP/2.0/UDP  213.137.65.234:5060
From: <sip:4103532264 at 213.137.65.234>;tag=16A2EDA0-5B2
To: <sip:14103445557 at 213.137.73.178>;tag=as075e701d
Date: Fri, 03 Oct 2003 15:38:58 GMT
Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-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/-0810da50'
    -- Executing Dial("SIP/-0810da50", "Zap/5-1") in new stack
  == Everyone is busy at this time
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=f0d025bc-dd6e222a-65fe226f-29e9c47-1
Via: SIP/2.0/UDP  213.137.65.234:5060
From: <sip:4103532264 at 213.137.65.234>;tag=16A2EDA0-5B2
To: <sip:14103445557 at 213.137.73.178>;tag=as075e701d
Date: Fri, 03 Oct 2003 15:38:58 GMT
Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5 at 213.137.65.234
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 9
Timestamp: 1065195544
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=f0d025bc-dd6e222a-65fe226f-29e9c47-1
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=16A2EDA0-5B2
To: <sip:14103445557 at 213.137.73.178>;tag=as075e701d
Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5 at 213.137.65.234
CSeq: 102 BYE
User-Agent: Asterisk PBX
Contact: 
Content-Length: 0


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