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Summary:ASTERISK-02140: Cisco 7960 SIP incoming call issue
Reporter:Matthew Simpson (matthewsimpson)Labels:
Date Opened:2004-07-29 14:26:11Date Closed:2004-09-25 02:12:17
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) extensions.conf
( 1) sip.conf
Description:Cisco 7960 SIP phone can make outgoing calls, but cannot receive incoming calls.  Seems to be a port problem.  Debug dump from Cisco 7960 attached.  Both asterisk and Cisco are on public IP addresses.  

****** ADDITIONAL INFORMATION ******

[12:07:52] SIPProcessUDPMessage: recv UDP message from <67.137.224.13>:<50195>:
<INVITE sip:2147649296@67.153.209.220 SIP/2.0
Via: SIP/2.0/UDP 67.137.224.13:0;branch=z9hG4bK2aca727e
From: "2022463521" <sip:2022463521@67.137.224.13:0>;tag=as399036a5
To: <sip:2147649296@67.153.209.220>
Contact: <sip:2022463521@67.137.224.13:0>
Call-ID: 38f0e703759fa812118353011dc04924@67.137.224.13
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 29 Jul 2004 19:10:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 31267 31267 IN IP4 67.137.224.13
s=session
c=IN IP4 67.137.224.13
t=0 0
m=audio 15766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
>, length=710
[12:07:52]
sippmh_parse_via: Invalid port number in Via[12:07:52]
sippmh_parse_via: Invalid port number in Via[12:07:52] Sendresponse: Error: No Via Header present in Message! returned error.
[12:07:52] SIPTaskProcessSIPMessage: Error: sipSPISendErrorResponse(400) failed.
[12:07:53] SIPProcessUDPMessage: recv UDP message from <67.137.224.13>:<50195>:
<INVITE sip:2147649296@67.153.209.220 SIP/2.0
Via: SIP/2.0/UDP 67.137.224.13:0;branch=z9hG4bK2aca727e
From: "2022463521" <sip:2022463521@67.137.224.13:0>;tag=as399036a5
To: <sip:2147649296@67.153.209.220>
Contact: <sip:2022463521@67.137.224.13:0>
Call-ID: 38f0e703759fa812118353011dc04924@67.137.224.13
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 29 Jul 2004 19:10:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 31267 31267 IN IP4 67.137.224.13
s=session
c=IN IP4 67.137.224.13
t=0 0
m=audio 15766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
>, length=710
[12:07:53]
sippmh_parse_via: Invalid port number in Via[12:07:53]
sippmh_parse_via: Invalid port number in Via[12:07:53] Sendresponse: Error: No Via Header present in Message! returned error.
[12:07:53] SIPTaskProcessSIPMessage: Error: sipSPISendErrorResponse(400) failed.
[12:07:54] SIPProcessUDPMessage: recv UDP message from <67.137.224.13>:<50195>:
<INVITE sip:2147649296@67.153.209.220 SIP/2.0
Via: SIP/2.0/UDP 67.137.224.13:0;branch=z9hG4bK2aca727e
From: "2022463521" <sip:2022463521@67.137.224.13:0>;tag=as399036a5
To: <sip:2147649296@67.153.209.220>
Contact: <sip:2022463521@67.137.224.13:0>
Call-ID: 38f0e703759fa812118353011dc04924@67.137.224.13
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 29 Jul 2004 19:10:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 31267 31267 IN IP4 67.137.224.13
s=session
c=IN IP4 67.137.224.13
t=0 0
m=audio 15766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
>, length=710
[12:07:54]
sippmh_parse_via: Invalid port number in Via[12:07:54]
sippmh_parse_via: Invalid port number in Via[12:07:54] Sendresponse: Error: No Via Header present in Message! returned error.
[12:07:54] SIPTaskProcessSIPMessage: Error: sipSPISendErrorResponse(400) failed.
[12:07:55] SIPProcessUDPMessage: recv UDP message from <67.137.224.13>:<50195>:
<INVITE sip:2147649296@67.153.209.220 SIP/2.0
Via: SIP/2.0/UDP 67.137.224.13:0;branch=z9hG4bK2aca727e
From: "2022463521" <sip:2022463521@67.137.224.13:0>;tag=as399036a5
To: <sip:2147649296@67.153.209.220>
Contact: <sip:2022463521@67.137.224.13:0>
Call-ID: 38f0e703759fa812118353011dc04924@67.137.224.13
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 29 Jul 2004 19:10:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 31267 31267 IN IP4 67.137.224.13
s=session
c=IN IP4 67.137.224.13
t=0 0
m=audio 15766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
>, length=710
[12:07:55]
sippmh_parse_via: Invalid port number in Via[12:07:55]
sippmh_parse_via: Invalid port number in Via[12:07:55] Sendresponse: Error: No Via Header present in Message! returned error.
[12:07:55] SIPTaskProcessSIPMessage: Error: sipSPISendErrorResponse(400) failed.
[12:07:56] SIPProcessUDPMessage: recv UDP message from <67.137.224.13>:<50195>:
<CANCEL sip:2147649296@67.153.209.220 SIP/2.0
Via: SIP/2.0/UDP 67.137.224.13:0;branch=z9hG4bK2aca727e
From: "2022463521" <sip:2022463521@67.137.224.13:0>;tag=as399036a5
To: <sip:2147649296@67.153.209.220>
Contact: <sip:2022463521@67.137.224.13:0>
Call-ID: 38f0e703759fa812118353011dc04924@67.137.224.13
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

>, length=373
[12:07:56]
sippmh_parse_via: Invalid port number in Via[12:07:56]
sippmh_parse_via: Invalid port number in Via[12:07:56] Sendresponse: Error: No Via Header present in Message! returned error.
[12:07:56] SIPTaskProcessSIPMessage: Error: sipSPISendErrorResponse(400) failed.
[12:07:56] SIPProcessUDPMessage: recv UDP message from <67.137.224.13>:<50195>:
<INVITE sip:2147649296@67.153.209.220 SIP/2.0
Via: SIP/2.0/UDP 67.137.224.13:0;branch=z9hG4bK2aca727e
From: "2022463521" <sip:2022463521@67.137.224.13:0>;tag=as399036a5
To: <sip:2147649296@67.153.209.220>
Contact: <sip:2022463521@67.137.224.13:0>
Call-ID: 38f0e703759fa812118353011dc04924@67.137.224.13
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 29 Jul 2004 19:10:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 31267 31267 IN IP4 67.137.224.13
s=session
c=IN IP4 67.137.224.13
t=0 0
m=audio 15766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
>, length=710
[12:07:56]
sippmh_parse_via: Invalid port number in Via[12:07:56]
sippmh_parse_via: Invalid port number in Via[12:07:56] Sendresponse: Error: No Via Header present in Message! returned error.
[12:07:56] SIPTaskProcessSIPMessage: Error: sipSPISendErrorResponse(400) failed.
[12:07:57] SIPProcessUDPMessage: recv UDP message from <67.137.224.13>:<50195>:
<CANCEL sip:2147649296@67.153.209.220 SIP/2.0
Via: SIP/2.0/UDP 67.137.224.13:0;branch=z9hG4bK2aca727e
From: "2022463521" <sip:2022463521@67.137.224.13:0>;tag=as399036a5
To: <sip:2147649296@67.153.209.220>
Contact: <sip:2022463521@67.137.224.13:0>
Call-ID: 38f0e703759fa812118353011dc04924@67.137.224.13
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

>, length=373
[12:07:57]
sippmh_parse_via: Invalid port number in Via[12:07:57]
sippmh_parse_via: Invalid port number in Via[12:07:57] Sendresponse: Error: No Via Header present in Message! returned error.
[12:07:57] SIPTaskProcessSIPMessage: Error: sipSPISendErrorResponse(400) failed.
[12:07:57] SIPProcessUDPMessage: recv UDP message from <67.137.224.13>:<50195>:
<INVITE sip:2147649296@67.153.209.220 SIP/2.0
Via: SIP/2.0/UDP 67.137.224.13:0;branch=z9hG4bK2aca727e
From: "2022463521" <sip:2022463521@67.137.224.13:0>;tag=as399036a5
To: <sip:2147649296@67.153.209.220>
Contact: <sip:2022463521@67.137.224.13:0>
Call-ID: 38f0e703759fa812118353011dc04924@67.137.224.13
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 29 Jul 2004 19:10:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 31267 31267 IN IP4 67.137.224.13
s=session
c=IN IP4 67.137.224.13
t=0 0
m=audio 15766 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
>, length=710
[12:07:57]
sippmh_parse_via: Invalid port number in Via[12:07:57]
sippmh_parse_via: Invalid port number in Via[12:07:57] Sendresponse: Error: No Via Header present in Message! returned error.
[12:07:57] SIPTaskProcessSIPMessage: Error: sipSPISendErrorResponse(400) failed.
[12:07:58] SIPProcessUDPMessage: recv UDP message from <67.137.224.13>:<50195>:
<CANCEL sip:2147649296@67.153.209.220 SIP/2.0
Via: SIP/2.0/UDP 67.137.224.13:0;branch=z9hG4bK2aca727e
From: "2022463521" <sip:2022463521@67.137.224.13:0>;tag=as399036a5
To: <sip:2147649296@67.153.209.220>
Contact: <sip:2022463521@67.137.224.13:0>
Call-ID: 38f0e703759fa812118353011dc04924@67.137.224.13
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

>, length=373
[12:07:58]
sippmh_parse_via: Invalid port number in Via[12:07:58]
sippmh_parse_via: Invalid port number in Via[12:07:58] Sendresponse: Error: No Via Header present in Message! returned error.
[12:07:58] SIPTaskProcessSIPMessage: Error: sipSPISendErrorResponse(400) failed.
[12:07:59] SIPProcessUDPMessage: recv UDP message from <67.137.224.13>:<50195>:
<CANCEL sip:2147649296@67.153.209.220 SIP/2.0
Via: SIP/2.0/UDP 67.137.224.13:0;branch=z9hG4bK2aca727e
From: "2022463521" <sip:2022463521@67.137.224.13:0>;tag=as399036a5
To: <sip:2147649296@67.153.209.220>
Contact: <sip:2022463521@67.137.224.13:0>
Call-ID: 38f0e703759fa812118353011dc04924@67.137.224.13
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

>, length=373
[12:07:59]
sippmh_parse_via: Invalid port number in Via[12:07:59]
sippmh_parse_via: Invalid port number in Via[12:07:59] Sendresponse: Error: No Via Header present in Message! returned error.
[12:07:59] SIPTaskProcessSIPMessage: Error: sipSPISendErrorResponse(400) failed.
[12:08:00] SIPProcessUDPMessage: recv UDP message from <67.137.224.13>:<50195>:
<CANCEL sip:2147649296@67.153.209.220 SIP/2.0
Via: SIP/2.0/UDP 67.137.224.13:0;branch=z9hG4bK2aca727e
From: "2022463521" <sip:2022463521@67.137.224.13:0>;tag=as399036a5
To: <sip:2147649296@67.153.209.220>
Contact: <sip:2022463521@67.137.224.13:0>
Call-ID: 38f0e703759fa812118353011dc04924@67.137.224.13
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

>, length=373
[12:08:00]
sippmh_parse_via: Invalid port number in Via[12:08:00]
sippmh_parse_via: Invalid port number in Via[12:08:00] Sendresponse: Error: No Via Header present in Message! returned error.
[12:08:00] SIPTaskProcessSIPMessage: Error: sipSPISendErrorResponse(400) failed.
[12:08:01] SIPProcessUDPMessage: recv UDP message from <67.137.224.13>:<50195>:
<CANCEL sip:2147649296@67.153.209.220 SIP/2.0
Via: SIP/2.0/UDP 67.137.224.13:0;branch=z9hG4bK2aca727e
From: "2022463521" <sip:2022463521@67.137.224.13:0>;tag=as399036a5
To: <sip:2147649296@67.153.209.220>
Contact: <sip:2022463521@67.137.224.13:0>
Call-ID: 38f0e703759fa812118353011dc04924@67.137.224.13
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

>, length=373
[12:08:01]
sippmh_parse_via: Invalid port number in Via[12:08:01]
sippmh_parse_via: Invalid port number in Via[12:08:01] Sendresponse: Error: No Via Header present in Message! returned error.
[12:08:01] SIPTaskProcessSIPMessage: Error: sipSPISendErrorResponse(400) failed.
Comments:By: Matthew Simpson (matthewsimpson) 2004-07-29 14:28:58

additional note:  I can swap in a Grandstream BT directly in place of the Cisco and everything works properly.  

I am using Cisco 7.1 SIP code, but this problem has been tested with version 6.3 and 5.3 as well with same results.
By: Matthew Simpson (matthewsimpson) 2004-07-29 14:29:44

additional note: If I plug the grandstream into the network along with the Cisco, I can place a SIP ip-->ip call bypassing asterisk to the 7960 which does work correctly.
By: Mark Spencer (markster) 2004-07-29 14:51:55

You must have something unusual in your sip configuration.  Can you please post your complete sip.conf file as an attachment?
By: Matthew Simpson (matthewsimpson) 2004-07-29 15:03:19

Here is my sip.conf
By: Mark Spencer (markster) 2004-07-29 15:13:17

Asterisk wasn't intializing the port to 5060 if you didn't explicitly say in sip.conf port=5060.  I've fixed this in CVS, or you may also simply add "port = 5060" to the top of your sip.conf.