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

[Daryl Jones]

I experienced the exact same symptoms but didn't have the confidence
to post my experience to this list because of my lack of experience with
Asterisk.  I restored the June 1 version from CVS and the problem went away.
There's definitely a problem in code since June 1.


On Sat, 7 Jun 2003, John Todd wrote:

> - 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
>
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