[Asterisk-Users] Asterisk with Draytek 2600V

Alessio Focardi afoc at interconnessioni.it
Fri May 28 03:40:02 MST 2004


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 at 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 at 192.168.0.250:5060>;tag=eSJ-4736
lg> To: <sip:90800500005 at 192.168.0.250>
lg> Call-ID: diY-24872 at 192.168.1.1
lg> CSeq: 1 INVITE
lg> Contact: <sip:phone1 at 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 at 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 at 192.168.0.250:5060>;tag=eSJ-4736
lg> To: <sip:90800500005 at 192.168.0.250>;tag=as71701551
lg> Call-ID: diY-24872 at 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 at 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 at 192.168.0.250:5060>;tag=eSJ-4736
lg> To: <sip:90800500005 at 192.168.0.250>;tag=as71701551
lg> Call-ID: diY-24872 at 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 at 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 at 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 at 192.168.0.250:5060>;tag=eSJ-4736
lg> To: <sip:90800500005 at 192.168.0.250>;tag=as71701551
lg> Call-ID: diY-24872 at 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 at 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 at 192.168.0.250:5060>;tag=eSJ-4736
lg> To: <sip:90800500005 at 192.168.0.250>;tag=as71701551
lg> Call-ID: diY-24872 at 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 at 192.168.0.250:5060>;tag=eSJ-4736
lg> To: <sip:90800500005 at 192.168.0.250>;tag=as71701551
lg> Call-ID: diY-24872 at 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 at 192.168.0.250>
lg> Content-Length: 0


lg> to 192.168.1.1:5060
lg> Destroying call 'diY-24872 at 192.168.1.1'
mars*CLI>>


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-- 
Best regards,
 Alessio                            mailto:afoc at interconnessioni.it





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