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
