[Asterisk-Users] miserable time with Cisco ATA186

[brian k. west]

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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