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Glenn Dalgliesh

2003-Oct-03 10:17 UTC

[Asterisk-Users] Iconnect Incomming calls

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@162.33.165.198 SIP/2.0
Record-Route: <sip:14103445557@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@213.137.65.234>;tag=16A2EDA0-5B2
To: <sip:14103445557@213.137.73.178>
Date: Fri, 03 Oct 2003 15:38:58 GMT
Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@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@213.137.65.234>;party=calling;screen=yes;privacy=off
Timestamp: 1065195538
Contact: <sip:4103532264@213.137.65.234:5060>
Diversion: <sip:4103445557@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@162.33.165.198 SIP/2.0
Record-Route: <sip:14103445557@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@213.137.65.234>;tag=16A2EDA0-5B2
To: <sip:14103445557@213.137.73.178>
Date: Fri, 03 Oct 2003 15:38:58 GMT
Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@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@213.137.65.234>;party=calling;screen=yes;privacy=off
Timestamp: 1065195538
Contact: <sip:4103532264@213.137.65.234:5060>
Diversion: <sip:4103445557@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@213.137.73.178:5060;maddr=213.137.73.176>
list_route: hop: <sip:4103532264@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@213.137.65.234>;tag=16A2EDA0-5B2
To: <sip:14103445557@213.137.73.178>;tag=as075e701d
Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Contact: <sip:14103445557@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@213.137.65.234>;tag=16A2EDA0-5B2
To: <sip:14103445557@213.137.73.178>;tag=as075e701d
Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Contact: <sip:14103445557@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@213.137.73.178:5060;maddr=213.137.73.176>
From: <sip:4103532264@213.137.65.234>;tag=16A2EDA0-5B2
To: <sip:14103445557@213.137.73.178>;tag=as075e701d
Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Contact: <sip:14103445557@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@162.33.165.198:5060 SIP/2.0
Record-Route: <sip:14103445557@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@213.137.65.234>;tag=16A2EDA0-5B2
To: <sip:14103445557@213.137.73.178>;tag=as075e701d
Date: Fri, 03 Oct 2003 15:38:58 GMT
Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@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@162.33.165.198:5060 SIP/2.0
Record-Route: <sip:14103445557@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@213.137.65.234>;tag=16A2EDA0-5B2
To: <sip:14103445557@213.137.73.178>;tag=as075e701d
Date: Fri, 03 Oct 2003 15:38:58 GMT
Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@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@213.137.73.178:5060;maddr=213.137.73.176>
From: <sip:4103532264@213.137.65.234>;tag=16A2EDA0-5B2
To: <sip:14103445557@213.137.73.178>;tag=as075e701d
Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234
CSeq: 102 BYE
User-Agent: Asterisk PBX
Contact: 
Content-Length: 0


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listas iPfone

2003-Oct-03 11:18 UTC

head link

[Asterisk-Users] Iconnect Incomming calls

Hi!

I?m thinking in an incoming number from ICH

please share your sip and extensions.conf files off list, it will help me a lot.

miklos
  ----- Original Message ----- 
  From: Glenn Dalgliesh 
  To: asterisk-users@lists.digium.com 
  Sent: Friday, October 03, 2003 2:17 PM
  Subject: [Asterisk-Users] Iconnect Incomming calls


  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@162.33.165.198 SIP/2.0
  Record-Route: <sip:14103445557@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@213.137.65.234>;tag=16A2EDA0-5B2
  To: <sip:14103445557@213.137.73.178>
  Date: Fri, 03 Oct 2003 15:38:58 GMT
  Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@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@213.137.65.234>;party=calling;screen=yes;privacy=off
  Timestamp: 1065195538
  Contact: <sip:4103532264@213.137.65.234:5060>
  Diversion: <sip:4103445557@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@162.33.165.198 SIP/2.0
  Record-Route: <sip:14103445557@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@213.137.65.234>;tag=16A2EDA0-5B2
  To: <sip:14103445557@213.137.73.178>
  Date: Fri, 03 Oct 2003 15:38:58 GMT
  Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@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@213.137.65.234>;party=calling;screen=yes;privacy=off
  Timestamp: 1065195538
  Contact: <sip:4103532264@213.137.65.234:5060>
  Diversion: <sip:4103445557@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@213.137.73.178:5060;maddr=213.137.73.176>
  list_route: hop: <sip:4103532264@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@213.137.65.234>;tag=16A2EDA0-5B2
  To: <sip:14103445557@213.137.73.178>;tag=as075e701d
  Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234
  CSeq: 101 INVITE
  User-Agent: Asterisk PBX
  Contact: <sip:14103445557@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@213.137.65.234>;tag=16A2EDA0-5B2
  To: <sip:14103445557@213.137.73.178>;tag=as075e701d
  Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234
  CSeq: 101 INVITE
  User-Agent: Asterisk PBX
  Contact: <sip:14103445557@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@213.137.73.178:5060;maddr=213.137.73.176>
  From: <sip:4103532264@213.137.65.234>;tag=16A2EDA0-5B2
  To: <sip:14103445557@213.137.73.178>;tag=as075e701d
  Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234
  CSeq: 101 INVITE
  User-Agent: Asterisk PBX
  Contact: <sip:14103445557@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@162.33.165.198:5060 SIP/2.0
  Record-Route: <sip:14103445557@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@213.137.65.234>;tag=16A2EDA0-5B2
  To: <sip:14103445557@213.137.73.178>;tag=as075e701d
  Date: Fri, 03 Oct 2003 15:38:58 GMT
  Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@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@162.33.165.198:5060 SIP/2.0
  Record-Route: <sip:14103445557@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@213.137.65.234>;tag=16A2EDA0-5B2
  To: <sip:14103445557@213.137.73.178>;tag=as075e701d
  Date: Fri, 03 Oct 2003 15:38:58 GMT
  Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@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@213.137.73.178:5060;maddr=213.137.73.176>
  From: <sip:4103532264@213.137.65.234>;tag=16A2EDA0-5B2
  To: <sip:14103445557@213.137.73.178>;tag=as075e701d
  Call-ID: 8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234
  CSeq: 102 BYE
  User-Agent: Asterisk PBX
  Contact: 
  Content-Length: 0


   
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Brian Capouch

2003-Oct-03 21:10 UTC

head link

[Asterisk-Users] Iconnect Incomming calls

Glenn Dalgliesh wrote:> 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.
>  
I am having the exact same problem.

It used to work.  Of course, was it iconnecthere or asterisk?

FYI.

B.



-- 
This message has been scanned for viruses
and is believed to be clean.
Scan engine v4.2.40 for Linux.
Virus data file v4294 created Sep 18 2003
Scanning for 80178 viruses, trojans and variants.

Glenn Dalgliesh

2003-Oct-07 08:18 UTC

head link

[Asterisk-Users] Iconnect Incomming calls

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@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@pbx1.routerboy.com on a i686
running Linux
-=-=-=-=-=-=-=-=-=-
pbx1*CLI> 
Sip read: 
INVITE sip:14103445557@162.33.165.198 SIP/2.0
Record-Route: <sip:14103445557@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@213.137.65.234>;tag=17B49340-1BD5
To: <sip:14103445557@213.137.73.178>
Date: Fri, 03 Oct 2003 20:37:52 GMT
Call-ID: 4B86D860-F51811D7-94A4DA91-FCF3ECE5@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@213.137.65.234>;party=calling;screen=yes;privacy=off
Timestamp: 1065213472
Contact: <sip:4103532264@213.137.65.234:5060>
Diversion: <sip:4103445557@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@162.33.165.198 SIP/2.0
Record-Route: <sip:14103445557@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@213.137.65.234>;tag=17B49340-1BD5
To: <sip:14103445557@213.137.73.178>
Date: Fri, 03 Oct 2003 20:37:52 GMT
Call-ID: 4B86D860-F51811D7-94A4DA91-FCF3ECE5@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@213.137.65.234>;party=calling;screen=yes;privacy=off
Timestamp: 1065213472
Contact: <sip:4103532264@213.137.65.234:5060>
Diversion: <sip:4103445557@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@213.137.73.178:5060;maddr=213.137.73.176>
list_route: hop: <sip:4103532264@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@213.137.65.234>;tag=17B49340-1BD5
To: <sip:14103445557@213.137.73.178>;tag=as72f8d457
Call-ID: 4B86D860-F51811D7-94A4DA91-FCF3ECE5@213.137.65.234
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:14103445557@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@213.137.65.234>;tag=17B49340-1BD5
To: <sip:14103445557@213.137.73.178>;tag=as72f8d457
Call-ID: 4B86D860-F51811D7-94A4DA91-FCF3ECE5@213.137.65.234
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:14103445557@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@213.137.73.178:5060;maddr=213.137.73.176>
From: <sip:4103532264@213.137.65.234>;tag=17B49340-1BD5
To: <sip:14103445557@213.137.73.178>;tag=as72f8d457
Call-ID: 4B86D860-F51811D7-94A4DA91-FCF3ECE5@213.137.65.234
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:14103445557@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@162.33.165.198:5060 SIP/2.0
Record-Route: <sip:14103445557@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@213.137.65.234>;tag=17B49340-1BD5
To: <sip:14103445557@213.137.73.178>;tag=as72f8d457
Date: Fri, 03 Oct 2003 20:37:52 GMT
Call-ID: 4B86D860-F51811D7-94A4DA91-FCF3ECE5@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@162.33.165.198:5060 SIP/2.0
Record-Route: <sip:14103445557@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@213.137.65.234>;tag=17B49340-1BD5
To: <sip:14103445557@213.137.73.178>;tag=as72f8d457
Date: Fri, 03 Oct 2003 20:37:52 GMT
Call-ID: 4B86D860-F51811D7-94A4DA91-FCF3ECE5@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@213.137.73.178:5060;maddr=213.137.73.176>
From: <sip:4103532264@213.137.65.234>;tag=17B49340-1BD5
To: <sip:14103445557@213.137.73.178>;tag=as72f8d457
Call-ID: 4B86D860-F51811D7-94A4DA91-FCF3ECE5@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@213.137.73.176>;tag=as4299840a
To: <sip:14103445557@213.137.73.176>
Call-ID: 7e01e2fb05dced9e46b4815671a0e597@162.33.165.198
CSeq: 110 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:14103445557@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@213.137.73.176>;tag=as4299840a
To: <sip:14103445557@213.137.73.176>
Call-ID: 7e01e2fb05dced9e46b4815671a0e597@162.33.165.198
CSeq: 110 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:14103445557@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@213.137.73.176>;tag=as4299840a
To: <sip:14103445557@213.137.73.176>
Call-ID: 7e01e2fb05dced9e46b4815671a0e597@162.33.165.198
CSeq: 110 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:14103445557@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@213.137.73.176>;tag=as4299840a
To: <sip:14103445557@213.137.73.176>
Call-ID: 7e01e2fb05dced9e46b4815671a0e597@162.33.165.198
CSeq: 110 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:14103445557@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@162.33.165.198
From: <sip:14103445557@213.137.73.176>;tag=as4299840a
To: <sip:14103445557@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@162.33.165.198
From: <sip:14103445557@213.137.73.176>;tag=as4299840a
To: <sip:14103445557@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@213.137.73.176>;tag=as4299840a
To: <sip:14103445557@213.137.73.176>
Call-ID: 7e01e2fb05dced9e46b4815671a0e597@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@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@162.33.165.198
From: <sip:14103445557@213.137.73.176>;tag=as4299840a
To: <sip:14103445557@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@162.33.165.198
From: <sip:14103445557@213.137.73.176>;tag=as4299840a
To: <sip:14103445557@213.137.73.176>
CSeq: 111 REGISTER
Contact: <sip:14103445557@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@213.137.73.176>;tag=as710c9606
To: <sip:14103445557@213.137.73.176>
Call-ID: 7e01e2fb05dced9e46b4815671a0e597@162.33.165.198
CSeq: 112 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:14103445557@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@162.33.165.198
From: <sip:14103445557@213.137.73.176>;tag=as710c9606
To: <sip:14103445557@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@162.33.165.198
From: <sip:14103445557@213.137.73.176>;tag=as710c9606
To: <sip:14103445557@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@213.137.73.176>;tag=as710c9606
To: <sip:14103445557@213.137.73.176>
Call-ID: 7e01e2fb05dced9e46b4815671a0e597@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@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@162.33.165.198
From: <sip:14103445557@213.137.73.176>;tag=as710c9606
To: <sip:14103445557@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@162.33.165.198
From: <sip:14103445557@213.137.73.176>;tag=as710c9606
To: <sip:14103445557@213.137.73.176>
CSeq: 113 REGISTER
Contact: <sip:14103445557@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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John Croft

2003-Oct-16 01:49 UTC

head link

[Asterisk-Users] Iconnect Incomming calls

I've just been playing with Asterisk (great product, well done!) and
configuring it to work with IConnectHere and experienced the same problem.

It looked to me as if an INVITE was being sent to IConnectHere when you pick up
the incoming call which was being rejected. I guess that this may be something
to do with redirecting the audio path to connect directly to the phone?

Anyway, I "cured" the problem by setting canreinvite=false in the
[general] section. This has the side effect of setting the default behaviour to
route the audio through Asterisk, but at least it works!

I get the feeling that there should be some way of specifying this option only
for iconnecthere calls, but I can't find the appropriate incantation to
associate calls coming in because of the register=blah@sipauth.deltathree.com
line in [general] with the properties defined in the [iconnecthere] section. Can
anyone give me a clue?

John
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