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@munged SIP/2.0 Via: SIP/2.0/UDP munged:0;branch=z9hG4bK304da88f From: munged To: munged Contact: munged Call-ID: 29cc2fe50f4e9c827dcc7e57676564b7@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?
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:8664113278@munged SIP/2.0 > Via: SIP/2.0/UDP munged:0;branch=z9hG4bK304da88f > From: munged > To: munged > Contact: munged > Call-ID: 29cc2fe50f4e9c827dcc7e57676564b7@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? > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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."
because 2.16.1 has some bugs.. you need 2.16.2 or higher. bkw ----- Original Message ----- From: "Matthew Simpson" <matthew@symatec-computer.com> To: <asterisk-users@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 eachother> 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@munged SIP/2.0 > Via: SIP/2.0/UDP munged:0;branch=z9hG4bK304da88f > From: munged > To: munged > Contact: munged > Call-ID: 29cc2fe50f4e9c827dcc7e57676564b7@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? > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
If I turn allow=ulaw on only, asterisk tries to use it a=rtpmap:0 PCMU/8000 but the ATA says it doesn't have it: Answering/Requesting with root capability 4 Answering with non-codec capability 0x1(G723) If I turn allow=alaw on only or with allow=ulaw, asterisk sends it, the ATA says it has it [alaw], but it still won't negotiate it. I think the stupid ATA is just determined to use G723 no matter what... I have LBRCodec set to 3 which should have it try to use G729, but it still tries to use G723. The AudioMode setting has a parameter bit to "Enable G711 only", but I'm not sure how that bit thing works. Either the default 0x00150015 or the recommended 0x00140014 fails. [btw, bit 1 should be "1" to enable G711 only, if someone can help me there]. I'm seriously about to punt this thing into the garbage. Help! thanks, matt> From: "Timothy R. McKee" <tim@baseworx.net> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] miserable time with Cisco ATA186 > Date: Fri, 4 Jun 2004 00:04:22 -0400 > Reply-To: asterisk-users@lists.digium.com > > Noticed that he has ALAW set as the preferred codec on the ATA. I'dsuggest> testing with allow of ulaw only, then try turning on other codecs. Weknow> that one works well. > > > > ===================================================================> Timothy R. McKee > > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Eric Wieling > Sent: Thursday, June 03, 2004 23:36 > To: asterisk-users@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 > >
Hi Matt, On the ATA, set TxCodec=2 and RxCodec=2 (G.711u). Also, set AudioMode=0x00160016 , which will force G.711 . After saving, reload the /dev page to be sure that these values are set as expected. In Asterisk, allow=ulaw only. If it still doesn't work, use the NPrintf field and prserv, Ethereal, or Asterisk itself to trace the SIP. The problem may not be (only) codec negotiation. Don't throw out your ATA. We have several in our network, and they have better voice quality and fewer glitches than our other adapters. --Stewart> ----- Original Message ----- > From: "Matthew Simpson" <matthew@symatec-computer.com> > To: <asterisk-users@lists.digium.com> > Sent: Friday, June 04, 2004 8:52 AM > Subject: Re: [Asterisk-Users] miserable time with Cisco ATA186 > > > > If I turn allow=ulaw on only, asterisk tries to use it > > > > a=rtpmap:0 PCMU/8000 > > > > but the ATA says it doesn't have it: > > > > Answering/Requesting with root capability 4 > > Answering with non-codec capability 0x1(G723) > > > > If I turn allow=alaw on only or with allow=ulaw, asterisk sends it, the > ATA > > says it has it [alaw], but it still won't negotiate it. > > > > I think the stupid ATA is just determined to use G723 no matter what... I > > have LBRCodec set to 3 which should have it try to use G729, but it still > > tries to use G723. The AudioMode setting has a parameter bit to "Enable > > G711 only", but I'm not sure how that bit thing works. Either the default > > 0x00150015 or the recommended 0x00140014 fails. [btw, bit 1 should be "1" > > to enable G711 only, if someone can help me there]. > > > > I'm seriously about to punt this thing into the garbage. > > > > Help! > > > > thanks, > > matt