We are currently using asterisk with that voip router so I can assure that it's just a matter of configuration, not codecs.
It seems that you have a nat issue ... can you explain better you configuration ? Is the dryteck connected to a public ADSL line ? Is the asterisk box listening on a public ip ? Hello louis, Friday, May 28, 2004, 11:37:50 AM, you wrote: lg> I am unable to get a my Draytek working with our Asterisk server. I can lg> make/recieve calls but get no audio. I have tried the various codecs at the lg> Vigor end but still getting nothing. I looked at sip debug (below) but am lg> new to Asterisk and don't really know what I am looking for. Asterisk works lg> fine with XLITE so I know my installation is ok. lg> Sip read: lg> INVITE sip:[EMAIL PROTECTED] SIP/2.0 lg> Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-Ifn-9746 lg> From: phone1 <sip:[EMAIL PROTECTED]:5060>;tag=eSJ-4736 lg> To: <sip:[EMAIL PROTECTED]> lg> Call-ID: [EMAIL PROTECTED] lg> CSeq: 1 INVITE lg> Contact: <sip:[EMAIL PROTECTED]> lg> Max-Forwards: 70 lg> User-Agent: DrayTek UA-1.0 lg> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE lg> Content-Type: application/sdp lg> Content-Length: 290 lg> v=0 lg> o=phone2 5972727 56415 IN IP4 192.168.1.1 lg> s=SIP Call lg> c=IN IP4 192.168.1.1 lg> t=0 0 lg> m=audio 10116 RTP/AVP 18 0 8 4 2 101 lg> a=rtpmap:18 G729/8000 lg> a=rtpmap:0 pcmu/8000 lg> a=rtpmap:8 pcma/8000 lg> a=rtpmap:4 g723/8000 lg> a=rtpmap:2 g726/8000 lg> a=rtpmap:101 telephone-event/8000 lg> a=fmtp:101 0-15 lg> 12 headers, 13 lines lg> Using latest request as basis request lg> Sending to 192.168.1.1 : 5060 (non-NAT) lg> Found RTP audio format 18 lg> Found RTP audio format 0 lg> Found RTP audio format 8 lg> Found RTP audio format 4 lg> Found RTP audio format 2 lg> Found RTP audio format 101 lg> Peer RTP is at port 192.168.1.1:0 lg> Found description format G729 lg> Found description format pcmu lg> Found description format pcma lg> Found description format g723 lg> Found description format g726 lg> Found description format telephone-event lg> Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - lg> audio=0x11d(G723|ULAW|ALAW|G726|G729A)/video=0x0(EMPTY), combined - lg> 0xc(ULAW|ALAW) lg> Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - lg> 0x1(G723) lg> Found user 'phone1' lg> Looking for 90800500005 in sip lg> list_route: hop: <sip:[EMAIL PROTECTED]> lg> Transmitting (no NAT): lg> SIP/2.0 100 Trying lg> Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-Ifn-9746 lg> From: phone1 <sip:[EMAIL PROTECTED]:5060>;tag=eSJ-4736 lg> To: <sip:[EMAIL PROTECTED]>;tag=as71701551 lg> Call-ID: [EMAIL PROTECTED] lg> CSeq: 1 INVITE lg> User-Agent: Asterisk PBX lg> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER lg> Contact: <sip:[EMAIL PROTECTED]> lg> Content-Length: 0 lg> to 192.168.1.1:5060 lg> We're at 192.168.0.250 port 13586 lg> Answering with capability 0x2(GSM) lg> Answering with capability 0x4(ULAW) lg> Answering with capability 0x8(ALAW) lg> Answering with non-codec capability 0x1(G723) lg> Reliably Transmitting (no NAT): lg> SIP/2.0 200 OK lg> Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-Ifn-9746 lg> From: phone1 <sip:[EMAIL PROTECTED]:5060>;tag=eSJ-4736 lg> To: <sip:[EMAIL PROTECTED]>;tag=as71701551 lg> Call-ID: [EMAIL PROTECTED] lg> CSeq: 1 INVITE lg> User-Agent: Asterisk PBX lg> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER lg> Contact: <sip:[EMAIL PROTECTED]> lg> Content-Type: application/sdp lg> Content-Length: 265 lg> v=0 lg> o=root 24864 24864 IN IP4 192.168.0.250 lg> s=session lg> c=IN IP4 192.168.0.250 lg> t=0 0 lg> m=audio 13586 RTP/AVP 3 0 8 101 lg> a=rtpmap:3 GSM/8000 lg> a=rtpmap:0 PCMU/8000 lg> a=rtpmap:8 PCMA/8000 lg> a=rtpmap:101 telephone-event/8000 lg> a=fmtp:101 0-16 lg> a=silenceSupp:off - - - - lg> to 192.168.1.1:5060 mars*CLI>> lg> Sip read: lg> ACK sip:[EMAIL PROTECTED] SIP/2.0 lg> Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-YQM-30118 lg> From: phone1 <sip:[EMAIL PROTECTED]:5060>;tag=eSJ-4736 lg> To: <sip:[EMAIL PROTECTED]>;tag=as71701551 lg> Call-ID: [EMAIL PROTECTED] lg> CSeq: 1 ACK lg> Max-Forwards: 70 lg> User-Agent: DrayTek UA-1.0 lg> Content-Length: 0 lg> 9 headers, 0 lines mars*CLI>> lg> Sip read: lg> BYE sip:[EMAIL PROTECTED] SIP/2.0 lg> Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-eSd-21367 lg> From: phone1 <sip:[EMAIL PROTECTED]:5060>;tag=eSJ-4736 lg> To: <sip:[EMAIL PROTECTED]>;tag=as71701551 lg> Call-ID: [EMAIL PROTECTED] lg> CSeq: 2 BYE lg> Max-Forwards: 70 lg> User-Agent: DrayTek UA-1.0 lg> Content-Length: 0 lg> 9 headers, 0 lines lg> Sending to 192.168.1.1 : 5060 (non-NAT) lg> Transmitting (no NAT): lg> SIP/2.0 200 OK lg> Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-eSd-21367 lg> From: phone1 <sip:[EMAIL PROTECTED]:5060>;tag=eSJ-4736 lg> To: <sip:[EMAIL PROTECTED]>;tag=as71701551 lg> Call-ID: [EMAIL PROTECTED] lg> CSeq: 2 BYE lg> User-Agent: Asterisk PBX lg> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER lg> Contact: <sip:[EMAIL PROTECTED]> lg> Content-Length: 0 lg> to 192.168.1.1:5060 lg> Destroying call '[EMAIL PROTECTED]' mars*CLI>> lg> _______________________________________________ lg> Asterisk-Users mailing list lg> [EMAIL PROTECTED] lg> http://lists.digium.com/mailman/listinfo/asterisk-users lg> To UNSUBSCRIBE or update options visit: lg> http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alessio mailto:[EMAIL PROTECTED] _______________________________________________ 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
