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Summary:ASTERISK-00770: Software SIP phone can not register when computer has no gateway
Reporter:kenalker (kenalker)Labels:
Date Opened:2004-01-09 19:52:00.000-0600Date Closed:2011-06-07 14:04:40
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
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Description:A software SIP phone running on a computer that has no gateway defined can not register with Asterisk.

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

1) When client computer and PBX computer are in same subnet, there is no need to have a gateway, and in fact, this is preferred in many cases for security reasons.  However, this breaks SIP with Asterisk.

2) Problem is consistent when using SJphone 1.10 build 187c.exe on Windows 2000 5.00.2195 SP4.

3) Here is what is seen from the Asterisk CLI with "sip debug" enabled (note the "VIA" line missing IP address of computer running software SIP phone):

Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP :5060
From: <sip:kensoftphone@192.168.64.164>;tag=3366426445
To: <sip:kensoftphone@192.168.64.164>;tag=as55662497
Call-ID: DE7B0FBC-DCAE-4303-AAC0-2A4BB8C50397@local.agent
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:kensoftphone@192.168.64.164>
Proxy-Authenticate: Digest realm="asterisk", nonce="457ba4a7"
Content-Length: 0

to 192.168.64.193:5060

4) Here is error log in /var/log/asterisk/messages:
Jan  9 16:50:05 WARNING[5126]: File chan_sip.c, Line 3931 (check_via): '' is not a valid host

5) Here is what is seen from the Asterisk CLI with "sip debug" enabled after a gateway is added to the computer running the software SIP phone.  SIP phone can now register with Asterisk.

Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.64.193:5060
From: <sip:kensoftphone@192.168.64.164>;tag=3366659760
To: <sip:kensoftphone@192.168.64.164>;tag=as54972e93
Call-ID: DE7B0FBC-DCAE-4303-AAC0-2A4BB8C50397@local.agent
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 120
Contact: <sip:kensoftphone@192.168.64.164>;expires=120
Date: Sat, 10 Jan 2004 00:53:58 GMT
Content-Length: 0

to 192.168.64.193:5060

6) SIP phone installed and all params left as default except the following (all found under "SJphone Options" "SIP" tab):
[X] Use local outbound proxy
Proxy IP address: 192.168.64.164:5060
Caller ID: sip:kensoftphone@192.168.64.164
Account: kensoftphone
Password: ********
Comments:By: kenalker (kenalker) 2004-01-09 20:00:01.000-0600

I am running Asterisk CVS date 1/7/2004.
out of "uname -a":
Linux pbx 2.4.24 #2 Thu Jan 8 05:29:42 PST 2004 i686 unknown
By: jrollyson (jrollyson) 2004-01-10 20:22:43.000-0600

This looks like a *client* error - the client is sending a null VIA header - it should have its own IP there if some sort of SIP proxy/gateway isn't utilized.
By: Brian West (bkw918) 2004-01-10 20:27:34.000-0600

Yep I agree with jrollyson this isn't asterisk fault.  It can only work with what you give it... and if you don't give it a via header (from the client) its isn't going to work.  So SJphone is to blame here.
By: Brian West (bkw918) 2004-01-11 04:05:09.000-0600

Network related issue and/or issue with SJPhone.