You have allow=all in sip.conf. For testing set disallow=all and allow=ulaw ONLY!
On Thu, 2004-06-03 at 21:43, Matthew Simpson wrote: > 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:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP munged:0;branch=z9hG4bK304da88f > From: munged > To: munged > Contact: munged > Call-ID: [EMAIL PROTECTED] > 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? > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- 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." _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users