asterisk users - Nov 2004 - RE: IAXys or IAX Softphones cannot call SIP phones

I've found the problem.  


Sip read:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 172.31.30.3:5060;branch=z9hG4bK658d8eed;rport
From: ""Nik Martin IAX"
<sip:105@172.31.30.3>;tag=as40e79563
<<<<<<<<<<<<<<<<<
To: <sip:nmartin@172.31.30.7:5060;user=phone>
Call-ID: 250ce5ee131823bc1c3a69430e686b4c@172.31.30.3
Date: Wed, 03 Nov 2004 15:21:52 GMT
Warning: 399 Bad Request - 'Malformed/Missing FROM: field'
<<<<<<<<<<<<<<<<<<
CSeq: 102 INVITE
Content-Length: 0


Prior to upgrading, I had caller ID strings in IAX.Conf like:

Callerid="Nik Martin <105>"

Now, evedintly these must look like:

Callerid "Nik Martin" <105>

This was probably an error on my part, but prior to upgrading, it seemed to
work fine. I don't remember if callerd id was actually being sent from IAX
phones or not, but it worked.


-----Original Message-----
From: Nik Martin [mailto:nmartin@radiancetech.com] 
Sent: Wednesday, November 03, 2004 9:29 AM
To: 'asterisk-users@lists.digium.com'
Subject: IAXys or IAX Softphones cannot call SIP phones


I recently upgraded to the latest asterisk, and everything was working fine.
Just recently, my softphones or IAXys cannot call any SIP phones.  This
happens whether the IAX calls originate on my LAN, or outside the network.
The Cisco SIP phones can call the IAX phones fine. I thought it was codec
related, but in iax.conf, I've disallowed everything but alaw.  Here's
the
SIP debug.  Thanks for any insight.

This is a firefly (latest thirdparty release) softphone calling a SIP phone
(Cisco 7960):




pbxMobile*CLI> sip debug
SIP Debugging Enabled
    -- Accepting AUTHENTICATED call from 172.31.30.20, requested format 1024,
actual format = 4
    -- Executing Dial("IAX2/nikko@172.31.30.20:4569/8",
"SIP/nmartin|20|tT")
in new stack We're at 172.31.30.3 port 11950 Answering/Requesting with root
capability 4 Answering with preferred capability 0x8(ALAW) 12 headers, 9
lines Reliably Transmitting: INVITE sip:nmartin@172.31.30.7:5060;user=phone
SIP/2.0
Via: SIP/2.0/UDP 172.31.30.3:5060;branch=z9hG4bK658d8eed;rport
From: ""Nik Martin IAX"
<sip:105@172.31.30.3>;tag=as40e79563
To: <sip:nmartin@172.31.30.7:5060;user=phone>
Contact: <sip:105@172.31.30.3>
Call-ID: 250ce5ee131823bc1c3a69430e686b4c@172.31.30.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 03 Nov 2004 15:16:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 182

v=0
o=root 15333 15333 IN IP4 172.31.30.3
s=session
c=IN IP4 172.31.30.3
t=0 0
m=audio 11950 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
 (NAT) to 172.31.30.7:5060
    -- Called nmartin
pbxMobile*CLI>

Sip read:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 172.31.30.3:5060;branch=z9hG4bK658d8eed;rport
From: ""Nik Martin IAX"
<sip:105@172.31.30.3>;tag=as40e79563
To: <sip:nmartin@172.31.30.7:5060;user=phone>
Call-ID: 250ce5ee131823bc1c3a69430e686b4c@172.31.30.3
Date: Wed, 03 Nov 2004 15:21:52 GMT
Warning: 399 Bad Request - 'Malformed/Missing FROM: field'
CSeq: 102 INVITE
Content-Length: 0