[Asterisk-Users] sip channel driver causes asterisk to crash when talking to quintum A800

[John Todd]

Make that three of us.  However, Asterisk isn't crashing, it's merely 
"locking up" with my ATA-186, but not my Cisco 7960's.  My own debug 
included below.


JT

>On Sat, 7 Jun 2003, shido wrote:
>
>  > This is the sip debug when the call went through........
>>
>[snip]
>
>
>Funnily enough I've been looking at the same problem.  Will get a
>chance to look a bit more tomorrow.
>
>Steve


   SIP is acting poorly with my ATA-186 devices, and I can't narrow 
down exactly why.  This is on code from about an hour ago, with a 
complete cvs update; make clean; make; make install .

- Asterisk starts
- various phones REGISTER - this works fine
- test: calls from my 7960 to either line on my ATA-186 work fine
- test: calls from my 7960 to any other destination work fine (IAX, Zap, etc.)
- all of my phones are behind the same NAT, if that matters


- the first call I try to place out of my ATA-186 fails (to any 
destination; my example uses a call to the 7960) but I see the 
included sip debug information on my console.  No more SIP debugging 
information appears past that point.  It is as if the ATA-186 for 
some reason "kills" Asterisk, where it did not before. 

- After that point, all other SIP calls from any other device fail, 
and looking at tethereal I see that there are no replies to new SIP 
REGISTER requests, either.  I can type "stop now" or "stop 
gracefully" and the system will not stop.  I have to manually killall 
to get asterisk to die.

- I backed out to a version from June 3 21:18 and all dial modes work 
correctly with exactly the same /etc/asterisk/* files, so it is a 
change in Asterisk and not in the phones.





*CLI> show version
Asterisk CVS-06/07/03-01:40:15 built by root at foo.fox-den.com on a 
i686 running Linux
*CLI>
*CLI> sip debug
SIP Debugging Enabled
Sip read:
INVITE sip:2203 at 206.181.87.10;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.1.25:5060
From: <sip:2204 at 206.181.87.10;user=phone>;tag=2961659159
To: <sip:2203 at 206.181.87.10;user=phone>
Call-ID: 1413586497 at 10.0.1.25
CSeq: 1 INVITE
Contact: <sip:2204 at 10.0.1.25:5060;user=phone;transport=udp>
User-Agent: Cisco ATA 186  v2.16 ata18x (030401a)
Expires: 300
Content-Length: 243
Content-Type: application/sdp

v=0
o=2204 23257 23257 IN IP4 10.0.1.25
s=ATA186 Call
c=IN IP4 10.0.1.25
t=0 0
m=audio 16386 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

11 headers, 11 lines
Using latest request as basis request
Sending to 10.0.1.25 : 5060 (non-NAT)
Capabilities: us - 14, them - 268, combined - 12
Non-codec capabilities: us - 1, them - 1, combined - 1

*CLI>
*CLI>
*CLI> show channels
         Channel  (Context    Extension    Pri )   State Appl. 
Data          
0 active channel(s)
*CLI>
*CLI>
*CLI> sip show channels
Peer             Username    Call ID      Seq (Tx/Rx)  Lag      Jitter  Format
10.0.1.25        (None)      1852710522@  00101/00002  00000ms  0000ms  0
1 active SIP channel(s)
*CLI>



Configuration for ATA-186 line 1:

[2204]
type=friend
username=2204
secret=somepassword
mailbox=2203
host=dynamic
context=intern
canreinvite=no
dtmfmode=rfc2833
nat=1



For reference, here is the SIP debug for a functional call from a 
7960 on the same version of Asterisk code (2203 = 7960, 2204 = 
ATA-186 line 1)

*CLI>
Sip read:
INVITE sip:2204 at 206.181.87.10 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.15:5060
From: "2203" <sip:2203 at 206.181.87.10>;tag=0002b9eb0ef400c3289c4132-36211630
To: <sip:2204 at 206.181.87.10>
Call-ID: 0002b9eb-0ef420f0-6ad91f45-1c1c55ad at 10.0.1.15
Date: Sat, 07 Jun 2003 19:19:33 GMT
CSeq: 101 INVITE
User-Agent: CSCO/4
Contact: <sip:2203 at 10.0.1.15:5060>
Expires: 180
Content-Type: application/sdp
Content-Length: 241
Accept: application/sdp
Remote-Party-ID: "2203" 
<sip:2203 at 10.0.1.15>;party=calling;id-type=subscriber;privacy=off;screen=no

v=0
o=Cisco-SIPUA 14891 19200 IN IP4 10.0.1.15
s=SIP Call
c=IN IP4 10.0.1.15
t=0 0
m=audio 23764 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

14 headers, 11 lines
Using latest request as basis request
Sending to 10.0.1.15 : 5060 (non-NAT)
Capabilities: us - 14, them - 268, combined - 12
Non-codec capabilities: us - 1, them - 1, combined - 1