[Asterisk-Users] New Head Appears to Break SIP to iConnect

[Greg Blakely]

 Folks,

        I have to admit that I MAY have changed something (at someone's
advice) on a previous CVS head (May 28), but I'm not sure.  I think that
it had to do with changing "digest realm," but that may be a different
issue. At any rate, I had both incoming and outgoing with iConnectHere.

        Now, I made exactly ONE change:  I upgraded to the CVS head
dated 7/30.  I still have outgoing SIP via iconnect, but the incoming
just hangs, and finally times out to an iconnect intercept recording
(Your Call Cannot be completed).

I've done a 'sip debug,' and it appears that I've got the old 407 Error:

(209.98.47.209 is me; 213.xxx.xxx.xxx is iconnect)

lizzie*CLI>

Sip read:
INVITE sip:16126053676 at 209.98.47.209:5060 SIP/2.0
Via: SIP/2.0/UDP 213.137.73.140:5060;maddr=213.137.73.183
Via: SIP/2.0/UDP
213.137.73.41:5060;branch=22b27932-ad4142ba-5399fdb0-571ef11f-1
Via: SIP/2.0/UDP 213.137.65.239:5060;received=213.137.65.239
To: <sip:16126053676 at 213.137.73.179>
From: <sip:9529496767 at 213.137.65.239>;tag=31062DE8-21CE
Call-ID: 359C5C75-E5BB11D8-9120D756-77435478 at 213.137.65.239
CSeq: 101 INVITE
Contact: <sip:9529496767 at 213.137.65.239:5060>
Record-Route: <sip:16126053676 at 213.137.73.140:5060;maddr=213.137.73.183>
Record-Route:
<sip:9529496767.22b27932-ad4142ba-5399fdb0-571ef11f at 213.137.65.239:5060;
maddr=213.137.73.41>
Content-Type: application/sdp
Content-Length: 148

v=0
o=CiscoSystemsSIP-GW-UserAgent 5851 2446 IN IP4 213.137.65.239
s=SIP Call
c=IN IP4 213.137.65.239
t=0 0
m=audio 18698 RTP/AVP 4 0 8 2 101

13 headers, 6 lines
Using latest request as basis request
Sending to 213.137.73.140 : 5060 (non-NAT)
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 2
Found RTP audio format 101
Peer RTP is at port 213.137.65.239:0
Capabilities: us - 0xc(ULAW|ALAW), peer -
audio=0x1d(G723|ULAW|ALAW|G726)/video=0x0(EMPTY), combined -
0xc(ULAW|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
0x1(G723)
Found peer 'iconnect'
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 213.137.73.140:5060;maddr=213.137.73.183
Via: SIP/2.0/UDP
213.137.73.41:5060;branch=22b27932-ad4142ba-5399fdb0-571ef11f-1
Via: SIP/2.0/UDP 213.137.65.239:5060;received=213.137.65.239
From: <sip:9529496767 at 213.137.65.239>;tag=31062DE8-21CE
To: <sip:16126053676 at 213.137.73.179>;tag=as702accc9
Call-ID: 359C5C75-E5BB11D8-9120D756-77435478 at 213.137.65.239
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:16126053676 at 209.98.47.209>
Proxy-Authenticate: Digest realm="asterisk", nonce="60c6600e"
Content-Length: 0


 to 213.137.73.140:5060
Scheduling destruction of call
'359C5C75-E5BB11D8-9120D756-77435478 at 213.137.65.239' in 15000 ms
lizzie*CLI>

Sip read:
ACK sip:16126053676 at 209.98.47.209:5060 SIP/2.0
Via: SIP/2.0/UDP 213.137.73.140:5060;maddr=213.137.73.183
Via: SIP/2.0/UDP
213.137.73.41:5060;branch=22b27932-ad4142ba-5399fdb0-571ef11f-1
From: <sip:9529496767 at 213.137.65.239>;tag=31062DE8-21CE
To: <sip:16126053676 at 213.137.73.179>;tag=as702accc9
Call-ID: 359C5C75-E5BB11D8-9120D756-77435478 at 213.137.65.239
CSeq: 101 ACK
Content-Length: 0


8 headers, 0 lines
lizzie*CLI>
[root at lizzie asterisk]#