[Asterisk-Users] miserable time with Cisco ATA186

[Timothy R. McKee]

Noticed that he has ALAW set as the preferred codec on the ATA.  I'd suggest
testing with allow of ulaw only, then try turning on other codecs.  We know
that one works well. 



====================================================================
Timothy R. McKee


-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Eric Wieling
Sent: Thursday, June 03, 2004 23:36
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] miserable time with Cisco ATA186

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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-- 
          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."

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