[Asterisk-Users] Asterisk with Draytek 2600V

[louis g]

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 at 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 at 192.168.0.250:5060>;tag=eSJ-4736
To: <sip:90800500005 at 192.168.0.250>
Call-ID: diY-24872 at 192.168.1.1
CSeq: 1 INVITE
Contact: <sip:phone1 at 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 at 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 at 192.168.0.250:5060>;tag=eSJ-4736
To: <sip:90800500005 at 192.168.0.250>;tag=as71701551
Call-ID: diY-24872 at 192.168.1.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:90800500005 at 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 at 192.168.0.250:5060>;tag=eSJ-4736
To: <sip:90800500005 at 192.168.0.250>;tag=as71701551
Call-ID: diY-24872 at 192.168.1.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:90800500005 at 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 at 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 at 192.168.0.250:5060>;tag=eSJ-4736
To: <sip:90800500005 at 192.168.0.250>;tag=as71701551
Call-ID: diY-24872 at 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 at 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 at 192.168.0.250:5060>;tag=eSJ-4736
To: <sip:90800500005 at 192.168.0.250>;tag=as71701551
Call-ID: diY-24872 at 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 at 192.168.0.250:5060>;tag=eSJ-4736
To: <sip:90800500005 at 192.168.0.250>;tag=as71701551
Call-ID: diY-24872 at 192.168.1.1
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:90800500005 at 192.168.0.250>
Content-Length: 0


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