[Asterisk-Users] ATA-186 and fake ring

Thu Mar 13 00:10:46 MST 2003

On 2003-03-12 at 09:44, you wrote:

> > Who is generating this ringback?  The ATA or asterisk?

> Find out by doing  a trace.  If you're using callprogress, then you should
> see a 180 Ringing sent to the ATA when we detect ringing on the FXO.  If
> you're not using call progress, then we should not be sending 180 ringing.

We're not using callprogress (at least it's not set in zapata.conf).  
I also tried explicitly setting it to 'no', reloading, and trying again.
No change.

I do get a 180 Ringing.

I am dialing from my ATA-186 to 18189950699, a telco busy test.  Yet
all I hear is a ring, though on one call I heard a short blip of busy
before the ring started.

[See attachment for the trace output]

> You can also use the new "debug channel Zap/1-1" to see if the FXO is
> ringing.

If I do a 'show channel' on it, I get:
Zap/21-1  (intrunk    6197474525   1   ) Ringing AppDial       (Outgoing Line)
SIP/0054-b2bc  (outtrunk   8189950699   2   )    Ring Dial          Tor/g1/BYEXTENSION

That 6197474525 is strange.  That's probably the DNIS used on the 
last incoming call to that channel, but has nothing to do with my 
outgoing call.  I was calling from 6193640054 (SIP 0054).

My definition in sip.conf is:

[0054]
type=friend
insecure=yes
secret=myownsecret
callerid="Jim Gottlieb <(619) 364-0054>"
;  dynamic binding seems to time-out; try defaultip
host=dynamic
defaultip=192.168.40.90
; need to set the following so we can use voicemail and other DTMF apps
dtmfmode=rfc2833

-------------- next part --------------
[I dialed...]

Sip read: > 
INVITE sip:18189950699 at 198.51.175.9;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.40.90:5060
From:  <sip:0054 at 198.51.175.9;user=phone>;tag=850095511
To:  <sip:18189950699 at 198.51.175.9;user=phone>
Call-ID: 3623648208 at 192.168.40.90
CSeq: 1 INVITE
Contact:  <sip:0054 at 192.168.40.90:5060;user=phone;transport=udp>
User-Agent: Cisco ATA  v2.15 ata18x (020927a)
Expires: 300
Content-Length: 247
Content-Type: application/sdp

v=0
o=0054 5680 5680 IN IP4 192.168.40.90
s=ATA186 Call
c=IN IP4 192.168.40.90
t=0 0
m=audio 16384 RTP/AVP 0 4 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

11 headers, 11 lines
Interface is eth0
IP Address is 198.51.175.9
Using latest request as basis request
Sending to 192.168.40.90 : 5060 (non-NAT)
Capabilities: us - 14, them - 13, combined - 12
Non-codec capabilities: us - 1, them - 1, combined - 1
Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.40.90:5060
From: <sip:0054 at 198.51.175.9;user=phone>;tag=850095511
To: <sip:18189950699 at 198.51.175.9;user=phone>;tag=06e4b177
Call-ID: 3623648208 at 192.168.40.90
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact: <sip:18189950699 at 198.51.175.9;user=phone>
Proxy-Authenticate: Digest realm="asterisk", nonce="028f4554"
Content-Length: 0

 to 192.168.40.90:5060
Sip read: > 
ACK sip:18189950699 at 198.51.175.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.40.90:5060
From: <sip:0054 at 198.51.175.9;user=phone>;tag=850095511
To: <sip:18189950699 at 198.51.175.9;user=phone>;tag=06e4b177
Call-ID: 3623648208 at 192.168.40.90
CSeq: 1 ACK
User-Agent: Cisco ATA  v2.15 ata18x (020927a)
Content-Length: 0

8 headers, 0 lines
Sip read: > 
INVITE sip:18189950699 at 198.51.175.9;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.40.90:5060
From:  <sip:0054 at 198.51.175.9;user=phone>;tag=850095511
To:  <sip:18189950699 at 198.51.175.9;user=phone>
Call-ID: 3623648208 at 192.168.40.90
CSeq: 2 INVITE
Contact:  <sip:0054 at 192.168.40.90:5060;user=phone;transport=udp>
User-Agent: Cisco ATA  v2.15 ata18x (020927a)
Proxy-Authorization: Digest username="0054",realm="asterisk",nonce="028f4554",ur
i="sip:18189950699 at 198.51.175.9",response="a8dbe8f8d6faee139756514c82cad48f"
Expires: 300
Content-Length: 247
Content-Type: application/sdp

v=0
o=0054 5686 5686 IN IP4 192.168.40.90
s=ATA186 Call
c=IN IP4 192.168.40.90
t=0 0
m=audio 16384 RTP/AVP 0 4 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

12 headers, 11 lines
Using latest request as basis request
Sending to 192.168.40.90 : 5060 (non-NAT)
Capabilities: us - 14, them - 13, combined - 12
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 18189950699 in sip
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.40.90:5060
From: <sip:0054 at 198.51.175.9;user=phone>;tag=850095511
To: <sip:18189950699 at 198.51.175.9;user=phone>;tag=5b1ade90
Call-ID: 3623648208 at 192.168.40.90
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:18189950699 at 198.51.175.9;user=phone>
Content-Length: 0

 to 192.168.40.90:5060
We're at 198.51.175.9 port 53614
Answering with capability 4
Answering with capability 8
Answering with non-codec capability 1
Transmitting (no NAT):
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.40.90:5060
From: <sip:0054 at 198.51.175.9;user=phone>;tag=850095511
To: <sip:18189950699 at 198.51.175.9;user=phone>;tag=5b1ade90
Call-ID: 3623648208 at 192.168.40.90
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:18189950699 at 198.51.175.9;user=phone>
Content-Type: application/sdp
Content-Length: 211

v=0
o=root 5860 5860 IN IP4 198.51.175.9
s=session
c=IN IP4 198.51.175.9
t=0 0
m=audio 53614 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 TELEPHONE-EVENT/8000
a=fmtp:101 0-16

 to 192.168.40.90:5060
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.40.90:5060
From: <sip:0054 at 198.51.175.9;user=phone>;tag=850095511
To: <sip:18189950699 at 198.51.175.9;user=phone>;tag=5b1ade90
Call-ID: 3623648208 at 192.168.40.90
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Contact: <sip:18189950699 at 198.51.175.9;user=phone>
Content-Length: 0

 to 192.168.40.90:5060

[At this pont I was hearing ringback tone.  Then I hung up...]

Sip read: > 
CANCEL sip:18189950699 at 198.51.175.9;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.40.90:5060
From:  <sip:0054 at 198.51.175.9;user=phone>;tag=850095511
To:  <sip:18189950699 at 198.51.175.9;user=phone>;tag=5b1ade90
Call-ID: 3623648208 at 192.168.40.90
CSeq: 2 CANCEL
User-Agent: Cisco ATA  v2.15 ata18x (020927a)
Content-Length: 0

8 headers, 0 lines
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.40.90:5060
From: <sip:0054 at 198.51.175.9;user=phone>;tag=850095511
To: <sip:18189950699 at 198.51.175.9;user=phone>;tag=5b1ade90
Call-ID: 3623648208 at 192.168.40.90
CSeq: 2 CANCEL
User-Agent: Asterisk PBX
Contact: <sip:18189950699 at 198.51.175.9;user=phone>
Content-Length: 0

 to 192.168.40.90:5060