[Asterisk-Users] miserable time with Cisco ATA186

[Matthew Simpson]

I'm having a horrible experience getting a Cisco ATA-186 to work with *.

I can make calls from the ATA with no problems.  However, incoming calls
make the ATA ring once, and then the call is disconnected.  I have no
problems with my Sipura 2000 or my Grandstream phones.

I am running 2.16.1 sip code on the ATA 186.  Neither * nor the ATA is
behind a NAT.  They are both on public IP addresses right next to each other
on the same subnet.

SIP Debug shows [munged being the IP address]:

Answering/Requesting with root capability 4
Answering with preferred capability 0x8(ALAW)
Answering with capability 0x1(G723)
Answering with capability 0x2(GSM)
Answering with capability 0x10(G726)
Answering with capability 0x20(ADPCM)
Answering with capability 0x40(SLINR)
Answering with capability 0x80(LPC10)
Answering with capability 0x100(G729A)
Answering with capability 0x200(SPEEX)
Answering with capability 0x400(ILBC)
Answering with capability 0x800(UNKN)
Answering with capability 0x1000(UNKN)
Answering with capability 0x2000(UNKN)
Answering with capability 0x4000(UNKN)
Answering with capability 0x8000(UNKN)
Answering with non-codec capability 0x1(G723)
12 headers, 20 lines
Reliably Transmitting:
INVITE sip:8664113278 at munged SIP/2.0
Via: SIP/2.0/UDP munged:0;branch=z9hG4bK304da88f
From: munged
To: munged
Contact: munged
Call-ID: 29cc2fe50f4e9c827dcc7e57676564b7 at munged
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 04 Jun 2004 02:26:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 461

v=0
o=root 284 284 IN IP4 munged
s=session
c=IN IP4 munged
t=0 0
m=audio 14466 RTP/AVP 0 8 4 3 2 5 10 7 18 110 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=rtpmap:110 SPEEX/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


This Retransmits several times and then the call is scheduled for
destruction.  The "CANCEL" sip messages seem to fail also, as they are
retransmitted many times.  I'm running the ATA conf from:
http://www.fnords.org/~eric/asterisk/ata-186.shtml

Any ideas?