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<DIV><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p>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.</o:p></SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p></o:p></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p>Below is the SIP
debug</o:p></SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p></o:p></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p>Thank for any
help....</o:p></SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p></o:p></SPAN></FONT> </DIV>
<DIV><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p> to
162.33.165.195:5060<BR>Sip read: <BR>INVITE sip:14103445557@162.33.165.198
SIP/2.0<BR>Record-Route:
<sip:14103445557@213.137.73.178:5060;maddr=213.137.73.176><BR>Via:
SIP/2.0/UDP
213.137.73.176:5060;branch=6acca267-67952546-2efdf97b-e80aa0ef-1<BR>Via:
SIP/2.0/UDP 213.137.65.234:5060<BR>From:
<sip:4103532264@213.137.65.234>;tag=16A2EDA0-5B2<BR>To:
<sip:14103445557@213.137.73.178><BR>Date: Fri, 03 Oct 2003 15:38:58
GMT<BR>Call-ID: <A
href="mailto:8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234">8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234</A><BR>Supported:
timer,100rel<BR>Min-SE: 1800<BR>Cisco-Guid:
2316671854-4109242839-3208043153-4243844325<BR>User-Agent:
Cisco-SIPGateway/IOS-12.x<BR>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK,
COMET, REFER, SUBSCRIBE, NOTIFY, INFO<BR>CSeq: 101 INVITE<BR>Max-Forwards:
9<BR>Remote-Party-ID:
<sip:4103532264@213.137.65.234>;party=calling;screen=yes;privacy=off<BR>Timestamp:
1065195538<BR>Contact: <sip:4103532264@213.137.65.234:5060><BR>Diversion:
<sip:4103445557@213.137.65.234>;reason=unconditional<BR>Expires:
180<BR>Allow-Events: telephone-event<BR>Content-Type:
application/sdp<BR>Content-Length: 332</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=CiscoSystemsSIP-GW-UserAgent 8647 4690 IN IP4
213.137.65.234<BR>s=SIP Call<BR>c=IN IP4 213.137.65.234<BR>t=0 0<BR>m=audio
16836 RTP/AVP 4 18 101 19<BR>c=IN IP4 213.137.65.234<BR>a=rtpmap:4
G723/8000<BR>a=fmtp:4 annexa=no<BR>a=rtpmap:18 G729/8000<BR>a=fmtp:18
annexb=no<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101 0-16<BR>a=rtpmap:19
CN/8000</DIV>
<DIV> </DIV>
<DIV>23 headers, 14 lines<BR>Using latest request as basis request<BR>Sending to
213.137.73.176 : 5060 (non-NAT)<BR>Found audio format 4<BR>Found audio format
18<BR>Found audio format 101<BR>Found audio format 19<BR>Found description
format G723<BR>Found description format G729<BR>Found description format
telephone-event<BR>Found description format CN<BR>Capabilities: us - 524302,
them - 257/0, combined - 0<BR>Non-codec capabilities: us - 1, them - 3, combined
- 1<BR>Sip read: <BR>INVITE sip:14103445557@162.33.165.198
SIP/2.0<BR>Record-Route:
<sip:14103445557@213.137.73.178:5060;maddr=213.137.73.176><BR>Via:
SIP/2.0/UDP
213.137.73.176:5060;branch=6acca267-67952546-2efdf97b-e80aa0ef-1<BR>Via:
SIP/2.0/UDP 213.137.65.234:5060<BR>From:
<sip:4103532264@213.137.65.234>;tag=16A2EDA0-5B2<BR>To:
<sip:14103445557@213.137.73.178><BR>Date: Fri, 03 Oct 2003 15:38:58
GMT<BR>Call-ID: <A
href="mailto:8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234">8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234</A><BR>Supported:
timer,100rel<BR>Min-SE: 1800<BR>Cisco-Guid:
2316671854-4109242839-3208043153-4243844325<BR>User-Agent:
Cisco-SIPGateway/IOS-12.x<BR>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK,
COMET, REFER, SUBSCRIBE, NOTIFY, INFO<BR>CSeq: 101 INVITE<BR>Max-Forwards:
9<BR>Remote-Party-ID:
<sip:4103532264@213.137.65.234>;party=calling;screen=yes;privacy=off<BR>Timestamp:
1065195538<BR>Contact: <sip:4103532264@213.137.65.234:5060><BR>Diversion:
<sip:4103445557@213.137.65.234>;reason=unconditional<BR>Expires:
180<BR>Allow-Events: telephone-event<BR>Content-Type:
application/sdp<BR>Content-Length: 332</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=CiscoSystemsSIP-GW-UserAgent 8647 4690 IN IP4
213.137.65.234<BR>s=SIP Call<BR>c=IN IP4 213.137.65.234<BR>t=0 0<BR>m=audio
16836 RTP/AVP 4 18 101 19<BR>c=IN IP4 213.137.65.234<BR>a=rtpmap:4
G723/8000<BR>a=fmtp:4 annexa=no<BR>a=rtpmap:18 G729/8000<BR>a=fmtp:18
annexb=no<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101 0-16<BR>a=rtpmap:19
CN/8000</DIV>
<DIV> </DIV>
<DIV>23 headers, 14 lines<BR>Ignoring this request<BR>Looking for 14103445557 in
sipinbound<BR>RDNIS is 4103445557<BR>list_route: hop:
<sip:14103445557@213.137.73.178:5060;maddr=213.137.73.176><BR>list_route:
hop: <sip:4103532264@213.137.65.234:5060><BR>Transmitting (no
NAT):<BR>SIP/2.0 100 Trying<BR>Via: SIP/2.0/UDP
213.137.73.176:5060;branch=6acca267-67952546-2efdf97b-e80aa0ef-1<BR>Via:
SIP/2.0/UDP 213.137.65.234:5060<BR>From:
<sip:4103532264@213.137.65.234>;tag=16A2EDA0-5B2<BR>To:
<sip:14103445557@213.137.73.178>;tag=as075e701d<BR>Call-ID: <A
href="mailto:8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234">8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234</A><BR>CSeq:
101 INVITE<BR>User-Agent: Asterisk PBX<BR>Contact:
<sip:14103445557@162.33.165.198><BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR> to 213.137.73.176:5060<BR> -- Executing
Dial("SIP/-0810da50", "Zap/5-1") in new stack<BR> -- Called
5-1<BR> -- Zap/5-1 is ringing<BR>Transmitting (no
NAT):<BR>SIP/2.0 180 Ringing<BR>Via: SIP/2.0/UDP
213.137.73.176:5060;branch=6acca267-67952546-2efdf97b-e80aa0ef-1<BR>Via:
SIP/2.0/UDP 213.137.65.234:5060<BR>From:
<sip:4103532264@213.137.65.234>;tag=16A2EDA0-5B2<BR>To:
<sip:14103445557@213.137.73.178>;tag=as075e701d<BR>Call-ID: <A
href="mailto:8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234">8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234</A><BR>CSeq:
101 INVITE<BR>User-Agent: Asterisk PBX<BR>Contact:
<sip:14103445557@162.33.165.198><BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR> to 213.137.73.176:5060<BR> -- Zap/5-1 is
ringing<BR> -- Zap/5-1 answered SIP/-0810da50<BR>We're at
162.33.165.198 port 13196<BR>Answering with non-codec capability 1<BR>Reliably
Transmitting (no NAT):<BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP
213.137.73.176:5060;branch=6acca267-67952546-2efdf97b-e80aa0ef-1<BR>Via:
SIP/2.0/UDP 213.137.65.234:5060<BR>Record-Route:
<sip:14103445557@213.137.73.178:5060;maddr=213.137.73.176><BR>From:
<sip:4103532264@213.137.65.234>;tag=16A2EDA0-5B2<BR>To:
<sip:14103445557@213.137.73.178>;tag=as075e701d<BR>Call-ID: <A
href="mailto:8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234">8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234</A><BR>CSeq:
101 INVITE<BR>User-Agent: Asterisk PBX<BR>Contact:
<sip:14103445557@162.33.165.198><BR>Content-Type:
application/sdp<BR>Content-Length: 167</DIV>
<DIV> </DIV>
<DIV>v=0<BR>o=root 1387 1387 IN IP4 162.33.165.198<BR>s=session<BR>c=IN IP4
162.33.165.198<BR>t=0 0<BR>m=audio 13196 RTP/AVP 101<BR>a=rtpmap:101
telephone-event/8000<BR>a=fmtp:101 0-16</DIV>
<DIV> </DIV>
<DIV> to 213.137.73.176:5060<BR>Sip read: <BR>ACK
sip:14103445557@162.33.165.198:5060 SIP/2.0<BR>Record-Route:
<sip:14103445557@213.137.73.178:5060;maddr=213.137.73.176><BR>Via:
SIP/2.0/UDP
213.137.73.176:5060;branch=50f2f595-1a997f3d-5142cdf4-e4672261-1<BR>Via:
SIP/2.0/UDP 213.137.65.234:5060<BR>From:
<sip:4103532264@213.137.65.234>;tag=16A2EDA0-5B2<BR>To:
<sip:14103445557@213.137.73.178>;tag=as075e701d<BR>Date: Fri, 03 Oct 2003
15:38:58 GMT<BR>Call-ID: <A
href="mailto:8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234">8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234</A><BR>Max-Forwards:
9<BR>Content-Length: 0<BR>CSeq: 101 ACK</DIV>
<DIV> </DIV>
<DIV><BR>11 headers, 0 lines<BR> -- Hungup 'Zap/5-1'<BR>
== Spawn extension (sipinbound, 14103445557, 1) exited non-zero on
'SIP/-0810da50'<BR> -- Executing Dial("SIP/-0810da50",
"Zap/5-1") in new stack<BR> == Everyone is busy at this time<BR>Sip read:
<BR>BYE sip:14103445557@162.33.165.198:5060 SIP/2.0<BR>Record-Route:
<sip:14103445557@213.137.73.178:5060;maddr=213.137.73.176><BR>Via:
SIP/2.0/UDP
213.137.73.176:5060;branch=f0d025bc-dd6e222a-65fe226f-29e9c47-1<BR>Via:
SIP/2.0/UDP 213.137.65.234:5060<BR>From:
<sip:4103532264@213.137.65.234>;tag=16A2EDA0-5B2<BR>To:
<sip:14103445557@213.137.73.178>;tag=as075e701d<BR>Date: Fri, 03 Oct 2003
15:38:58 GMT<BR>Call-ID: <A
href="mailto:8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234">8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234</A><BR>User-Agent:
Cisco-SIPGateway/IOS-12.x<BR>Max-Forwards: 9<BR>Timestamp: 1065195544<BR>CSeq:
102 BYE<BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR>13 headers, 0 lines<BR>Sending to 213.137.73.176 : 5060
(non-NAT)<BR>Transmitting (no NAT):<BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP
213.137.73.176:5060;branch=f0d025bc-dd6e222a-65fe226f-29e9c47-1<BR>Via:
SIP/2.0/UDP 213.137.65.234:5060<BR>Record-Route:
<sip:14103445557@213.137.73.178:5060;maddr=213.137.73.176><BR>From:
<sip:4103532264@213.137.65.234>;tag=16A2EDA0-5B2<BR>To:
<sip:14103445557@213.137.73.178>;tag=as075e701d<BR>Call-ID: <A
href="mailto:8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234">8A16D3BE-F4EE11D7-BF39DA91-FCF3ECE5@213.137.65.234</A><BR>CSeq:
102 BYE<BR>User-Agent: Asterisk PBX<BR>Contact: <BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR></o:p></SPAN></FONT> </DIV></BODY></HTML>