7940 SIP Issues - Being called

Sypher · April 28, 2007, 12:17pm

Hiya,

Well because my colleage wanted to use SIP on his phone, and i don’t want to support multiple protocols i’ve decided to change my phone from SCCP to SIP. Things went well.

I can call, but they can’t call me! Pretty annoying. In the Cisco debug it says:

[quote][05:17:10:27125]
sippmh_parse_via: Invalid port number in Via[05:17:10:27127]
sippmh_parse_via: Invalid port number in Via[05:17:10:27128] Sendresponse: Error: Bad Via Header in Message! returned error.
[05:17:10:27129] SIPTaskProcessSIPMessage: Error: sipSPISendErrorResponse(400) failed.
[05:17:10:27129] SIPTaskProcessSIPMessage: Error: sippmh_is_message_complete() returned error.
[/quote]

10.0.0.151 is my client (number 500)
10.0.0.153 is my server
10.0.0.249 is my CP7940 (number 510)

Anyone knows how to resolve this? Thanks!

Sypher · April 29, 2007, 10:34am

I fiddled around with lots of settings but i still can’t call my 7940 phone…

bwilks · April 29, 2007, 11:12am

Hi,

I use the same phones; what does it say in the asterisk CLI and SIP debug.

You could also provide more info; like sip.conf extensions.conf

Sypher · April 29, 2007, 11:29am

Sip show peers:

Sip show users:

sip.conf:

[quote][510]
type=friend
username=510
secret=test
host=dynamic
dtmfmode=rfc2833
context=kantoor
canreinvite=no
nat=no
mailbox=510@kantoor
callerid=<510> [/quote]

extensions.conf:

[quote]exten => 510,1,Dial(SIP/510)
exten => 510,2,Hangup[/quote]

rtp.conf:

[quote][general]
rtpstart=199980
rtpend=200000[/quote]

sip.conf (general block)

[quote][general]
language=nl
context=kantoor
bindport=0.0.0.0
srvlookup=yes
maxexpirey=180
allow = ulaw
defaultexpirey=160
externip=1.2.3.4
localnet=10.0.0.0/255.224.0.0
nat=yes
sipdebug=no[/quote]

When i do “console dial 500@kantoor” i get the following after a while:

with nat=no in the SIP peer:

[quote]Audio is at 10.0.0.153 port 12900
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.249:5060:
INVITE sip:510@10.0.0.249:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.0.153:0;branch=z9hG4bK4e51795a;rport
From: “Testaccount” sip:500@10.0.0.153:0;tag=as020f68b5
To: sip:510@10.0.0.249:5060;user=phone;transport=udp
Contact: sip:500@10.0.0.153:0
Call-ID: 64955bbe738ace2d6e8e1cc6278618a8@10.0.0.153
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 29 Apr 2007 11:31:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 281

v=0
o=asterisk 3019 3019 IN IP4 10.0.0.153
s=session
c=IN IP4 10.0.0.153
t=0 0
m=audio 12900 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #1 (no NAT) to 10.0.0.249:5060:
INVITE sip:510@10.0.0.249:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.0.153:0;branch=z9hG4bK4e51795a;rport
From: “Testaccount” sip:500@10.0.0.153:0;tag=as020f68b5
To: sip:510@10.0.0.249:5060;user=phone;transport=udp
Contact: sip:500@10.0.0.153:0
Call-ID: 64955bbe738ace2d6e8e1cc6278618a8@10.0.0.153
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 29 Apr 2007 11:31:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 281

v=0
o=asterisk 3019 3019 IN IP4 10.0.0.153
s=session
c=IN IP4 10.0.0.153
t=0 0
m=audio 12900 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #2 (no NAT) to 10.0.0.249:5060:
INVITE sip:510@10.0.0.249:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.0.153:0;branch=z9hG4bK4e51795a;rport
From: “Testaccount” sip:500@10.0.0.153:0;tag=as020f68b5
To: sip:510@10.0.0.249:5060;user=phone;transport=udp
Contact: sip:500@10.0.0.153:0
Call-ID: 64955bbe738ace2d6e8e1cc6278618a8@10.0.0.153
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 29 Apr 2007 11:31:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 281

v=0
o=asterisk 3019 3019 IN IP4 10.0.0.153
s=session
c=IN IP4 10.0.0.153
t=0 0
m=audio 12900 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #3 (no NAT) to 10.0.0.249:5060:
INVITE sip:510@10.0.0.249:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.0.153:0;branch=z9hG4bK4e51795a;rport
From: “Testaccount” sip:500@10.0.0.153:0;tag=as020f68b5
To: sip:510@10.0.0.249:5060;user=phone;transport=udp
Contact: sip:500@10.0.0.153:0
Call-ID: 64955bbe738ace2d6e8e1cc6278618a8@10.0.0.153
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 29 Apr 2007 11:31:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 281

v=0
o=asterisk 3019 3019 IN IP4 10.0.0.153
s=session
c=IN IP4 10.0.0.153
t=0 0
m=audio 12900 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv[/quote]

with nat=yes in the sip peer:

[quote]Audio is at 10.0.0.153 port 19636
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.0.0.249:5060:
INVITE sip:510@10.0.0.249:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.0.0.153:0;branch=z9hG4bK0328cfca;rport
From: “Testaccount” sip:500@10.0.0.153:0;tag=as79325b0e
To: sip:510@10.0.0.249:5060;user=phone;transport=udp
Contact: sip:500@10.0.0.153:0
Call-ID: 2cdc03ff15499dc30cb3b2aa218b2bf9@10.0.0.153
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 29 Apr 2007 11:33:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 281