[Asterisk-Users] miserable time with Cisco ATA186

[Eric Wieling]

Perhaps, but *I* at least had decent luck with 2.16.1.  I suspect he has
allow=all and the codec that ends up being used is G723.1 and then, of
course, everything goes to hell.


On Thu, 2004-06-03 at 22:59, brian k. west wrote:
> because 2.16.1 has some bugs.. you need 2.16.2 or higher.
> 
> bkw
> 
> ----- Original Message ----- 
> From: "Matthew Simpson" <matthew at symatec-computer.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Thursday, June 03, 2004 8:43 PM
> Subject: [Asterisk-Users] miserable time with Cisco ATA186
> 
> 
> > 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?
> >
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> 
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-- 
          Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."