I am unable to get a my Draytek working with our Asterisk server. I can make/recieve calls but get no audio. I have tried the various codecs at the Vigor end but still getting nothing. I looked at sip debug (below) but am new to Asterisk and don't really know what I am looking for. Asterisk works fine with XLITE so I know my installation is ok. Sip read: INVITE sip:90800500005@192.168.0.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-Ifn-9746 From: phone1 <sip:phone1@192.168.0.250:5060>;tag=eSJ-4736 To: <sip:90800500005@192.168.0.250> Call-ID: diY-24872@192.168.1.1 CSeq: 1 INVITE Contact: <sip:phone1@192.168.1.1> Max-Forwards: 70 User-Agent: DrayTek UA-1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Type: application/sdp Content-Length: 290 v=0 o=phone2 5972727 56415 IN IP4 192.168.1.1 s=SIP Call c=IN IP4 192.168.1.1 t=0 0 m=audio 10116 RTP/AVP 18 0 8 4 2 101 a=rtpmap:18 G729/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:4 g723/8000 a=rtpmap:2 g726/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 13 lines Using latest request as basis request Sending to 192.168.1.1 : 5060 (non-NAT) Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 2 Found RTP audio format 101 Peer RTP is at port 192.168.1.1:0 Found description format G729 Found description format pcmu Found description format pcma Found description format g723 Found description format g726 Found description format telephone-event Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x11d(G723|ULAW|ALAW|G726|G729A)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found user 'phone1' Looking for 90800500005 in sip list_route: hop: <sip:phone1@192.168.1.1> Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-Ifn-9746 From: phone1 <sip:phone1@192.168.0.250:5060>;tag=eSJ-4736 To: <sip:90800500005@192.168.0.250>;tag=as71701551 Call-ID: diY-24872@192.168.1.1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:90800500005@192.168.0.250> Content-Length: 0 to 192.168.1.1:5060 We're at 192.168.0.250 port 13586 Answering with capability 0x2(GSM) Answering with capability 0x4(ULAW) Answering with capability 0x8(ALAW) Answering with non-codec capability 0x1(G723) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-Ifn-9746 From: phone1 <sip:phone1@192.168.0.250:5060>;tag=eSJ-4736 To: <sip:90800500005@192.168.0.250>;tag=as71701551 Call-ID: diY-24872@192.168.1.1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:90800500005@192.168.0.250> Content-Type: application/sdp Content-Length: 265 v=0 o=root 24864 24864 IN IP4 192.168.0.250 s=session c=IN IP4 192.168.0.250 t=0 0 m=audio 13586 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 192.168.1.1:5060 mars*CLI> Sip read: ACK sip:90800500005@192.168.0.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-YQM-30118 From: phone1 <sip:phone1@192.168.0.250:5060>;tag=eSJ-4736 To: <sip:90800500005@192.168.0.250>;tag=as71701551 Call-ID: diY-24872@192.168.1.1 CSeq: 1 ACK Max-Forwards: 70 User-Agent: DrayTek UA-1.0 Content-Length: 0 9 headers, 0 lines mars*CLI> Sip read: BYE sip:90800500005@192.168.0.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-eSd-21367 From: phone1 <sip:phone1@192.168.0.250:5060>;tag=eSJ-4736 To: <sip:90800500005@192.168.0.250>;tag=as71701551 Call-ID: diY-24872@192.168.1.1 CSeq: 2 BYE Max-Forwards: 70 User-Agent: DrayTek UA-1.0 Content-Length: 0 9 headers, 0 lines Sending to 192.168.1.1 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-eSd-21367 From: phone1 <sip:phone1@192.168.0.250:5060>;tag=eSJ-4736 To: <sip:90800500005@192.168.0.250>;tag=as71701551 Call-ID: diY-24872@192.168.1.1 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:90800500005@192.168.0.250> Content-Length: 0 to 192.168.1.1:5060 Destroying call 'diY-24872@192.168.1.1' mars*CLI>
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:90800500005@192.168.0.250 SIP/2.0 lg> Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-Ifn-9746 lg> From: phone1 <sip:phone1@192.168.0.250:5060>;tag=eSJ-4736 lg> To: <sip:90800500005@192.168.0.250> lg> Call-ID: diY-24872@192.168.1.1 lg> CSeq: 1 INVITE lg> Contact: <sip:phone1@192.168.1.1> 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:phone1@192.168.1.1> 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:phone1@192.168.0.250:5060>;tag=eSJ-4736 lg> To: <sip:90800500005@192.168.0.250>;tag=as71701551 lg> Call-ID: diY-24872@192.168.1.1 lg> CSeq: 1 INVITE lg> User-Agent: Asterisk PBX lg> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER lg> Contact: <sip:90800500005@192.168.0.250> 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:phone1@192.168.0.250:5060>;tag=eSJ-4736 lg> To: <sip:90800500005@192.168.0.250>;tag=as71701551 lg> Call-ID: diY-24872@192.168.1.1 lg> CSeq: 1 INVITE lg> User-Agent: Asterisk PBX lg> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER lg> Contact: <sip:90800500005@192.168.0.250> 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:90800500005@192.168.0.250 SIP/2.0 lg> Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-YQM-30118 lg> From: phone1 <sip:phone1@192.168.0.250:5060>;tag=eSJ-4736 lg> To: <sip:90800500005@192.168.0.250>;tag=as71701551 lg> Call-ID: diY-24872@192.168.1.1 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:90800500005@192.168.0.250 SIP/2.0 lg> Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-eSd-21367 lg> From: phone1 <sip:phone1@192.168.0.250:5060>;tag=eSJ-4736 lg> To: <sip:90800500005@192.168.0.250>;tag=as71701551 lg> Call-ID: diY-24872@192.168.1.1 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:phone1@192.168.0.250:5060>;tag=eSJ-4736 lg> To: <sip:90800500005@192.168.0.250>;tag=as71701551 lg> Call-ID: diY-24872@192.168.1.1 lg> CSeq: 2 BYE lg> User-Agent: Asterisk PBX lg> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER lg> Contact: <sip:90800500005@192.168.0.250> lg> Content-Length: 0 lg> to 192.168.1.1:5060 lg> Destroying call 'diY-24872@192.168.1.1' mars*CLI>> lg> _______________________________________________ lg> Asterisk-Users mailing list lg> Asterisk-Users@lists.digium.com 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:afoc@interconnessioni.it
Hello louis, Friday, May 28, 2004, 6:32:33 PM, you wrote: lg> Hi Alessio lg> Thank you for the reply. Our configuration is as follows lg> Asterisk Server 192.168.0.250 is on our LAN lg> Vigor 192.168.1.1 connects to the LAN VPN (vigor to vigor) lg> Laptop 192.168.1.10 with XLite I can suggest this: turn off xlite on the laptop, then reset the vigor that's on the side of the laptop. I can guess it will work then, I found similar problems some time ago. It seems that the vigor voip ports are only working if there are no sip clients behind the ethernet port, maybe it's some kind of port redirection issue. I can also say that vigor support is, in my experience, quick and very helpful. Hope it helps ! lg> The Vigor and Laptop both register with Asterisk using their correct private lg> ip's ie 192.168.1.1 and 192.168.1.10 lg> I can make and recieve calls fine on the Laptop but not on the Vigor. lg> I have yet to try placing the Asterisk server on a public IP address but I lg> may try this tomorrow when I am back in the office. Any ideas? I have a lg> standard SIP.CONF with no special config options but I may be missing lg> something. lg> Many Thanks lg> Louis Guadagno lg> Network Manager lg> Practical Law Company -- Best regards, Alessio mailto:afoc@interconnessioni.it