Look, I’ll put my cards on the table. I purchased two Grandstream ATAs and spent some time installing Asterisk on my server. I justified this expendature of time and money by saying “Once this is working you’ll just be able to pick up the phone and speak to your mother whenever you like for free without any need to turn on any computers, dear”. I have now been at my mother-in-law’s in Estonia for several days and after many hours or effort it is no closer to working. I am starting to get “If we’d just used that Skype thing like my mother suggested then it would be working by now” noises. I need to get this fixed before I go back to the UK in a few days and I am running out of ideas. If someone could suggest something which I could test or adjust then it would be a great help. I can adjust my server and router via an ssh connection.
New info since first posting:
Estonian ATA connects to commercial SIP server (www.voipcheap.co.uk) and works perfectly. Therefore, no problem with ATA or mother-in-law’s ISP.
If I remove the ATA and connect direct to the internet, X-Lite exhibits exactly the same symptoms (i.e. works fine with commercial server, registers with my server, but won’t actually make a call).
Handytone has a config box which claims to change the RTP port. It defaults to 5004, but changing it to 10000 makes no difference.
I suspect that this is a NAT problem and I have read voip-info.org/wiki-Asterisk+ … +solutions!
Setup: Asterisk 1.0.8 (latest stable on Gentoo) behind NAT router on IP 81.187.172.195 (internal IP is 10.42.42.1)
Handytone 488 inside router works just fine (connects to server and calls demo numbers)
Handytone 486 outside router on 84.15.82.106 (dynamic) doesn’t work
Both Handytones register with Asterisk OK
server*CLI> sip show peers
Name/username Host Dyn Nat ACL Mask Port Status
486Tallinn/486T 84.52.18.106 D N 255.255.255.255 5060 Unmonitored
488/488 10.42.42.30 D 255.255.255.255 5060 OK (5 ms)Refering to the afformentioned NAT guide, I suspect that I am a type 3 installation. I therefore need some port fowarding and some NAT config.
In sip.conf I have:
nat=yes
externip=81.187.172.195
fromdomain=huskydog.org.uk
localnet=10.42.42.0/255.255.255.0
I have forwarded 5060, 5004, 10000-10010 for all protocols (stupid router doesn’t allow port ranges to be opened, hence limited RTP range, hence)
rtpstart=10000
rtpend=10010
When I try to make a call to the demo system from the outside Handytone, all I get is an beeping tone.
sip debug says
Sip read:
INVITE sip:10.42.42.1@mail.huskydog.org.uk SIP/2.0
Via: SIP/2.0/UDP 84.52.18.106;branch=z9hG4bKd976a10f550ba22e
From: "Ljudmilla" <sip:486Tallinn@mail.huskydog.org.uk>;tag=36b99140642f9ae5
To: <sip:10.42.42.1@mail.huskydog.org.uk>
Contact: <sip:486Tallinn@84.52.18.106>
Supported: replaces, timer
Call-ID: 7f9d87b30897cf15@84.52.18.106
CSeq: 57404 INVITE
User-Agent: Grandstream HT487 1.0.8.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Type: application/sdp
Content-Length: 307
v=0
o=486Tallinn 8000 8000 IN IP4 84.52.18.106
s=SIP Call
c=IN IP4 84.52.18.106
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 97
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:20
13 headers, 15 lines
Using latest request as basis request
Sending to 84.52.18.106 : 5060 (non-NAT)
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 84.52.18.106;branch=z9hG4bKd976a10f550ba22e;received=84.52.18.106;rport=5060
From: "Ljudmilla" <sip:486Tallinn@mail.huskydog.org.uk>;tag=36b99140642f9ae5
To: <sip:10.42.42.1@mail.huskydog.org.uk>;tag=as5c62fcdd
Call-ID: 7f9d87b30897cf15@84.52.18.106
CSeq: 57404 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:10.42.42.1@81.187.172.195>
Proxy-Authenticate: Digest realm="huskydog.org.uk", nonce="29eda9b7"
Content-Length: 0
to 84.52.18.106:5060
Scheduling destruction of call '7f9d87b30897cf15@84.52.18.106' in 15000 ms
Found user '486Tallinn'
server*CLI>
Sip read:
ACK sip:10.42.42.1@mail.huskydog.org.uk SIP/2.0
Via: SIP/2.0/UDP 84.52.18.106;branch=z9hG4bKd976a10f550ba22e
From: "Ljudmilla" <sip:486Tallinn@mail.huskydog.org.uk>;tag=36b99140642f9ae5
To: <sip:10.42.42.1@mail.huskydog.org.uk>;tag=as5c62fcdd
Contact: <sip:486Tallinn@84.52.18.106>
Call-ID: 7f9d87b30897cf15@84.52.18.106
CSeq: 57404 ACK
User-Agent: Grandstream HT487 1.0.8.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Length: 0
11 headers, 0 lines
server*CLI>
Sip read:
INVITE sip:10.42.42.1@mail.huskydog.org.uk SIP/2.0
Via: SIP/2.0/UDP 84.52.18.106;branch=z9hG4bK86ef5382cd2018a2
From: "Ljudmilla" <sip:486Tallinn@mail.huskydog.org.uk>;tag=36b99140642f9ae5
To: <sip:10.42.42.1@mail.huskydog.org.uk>
Contact: <sip:486Tallinn@84.52.18.106>
Supported: replaces, timer
Proxy-Authorization: Digest username="486Tallinn", realm="huskydog.org.uk", algorithm=MD5, uri="sip:1000@mail.huskydog.org.uk", nonce="29eda9b7", response="71a7c5221fcf489eb974d3cd512f9824"
Call-ID: 7f9d87b30897cf15@84.52.18.106
CSeq: 57405 INVITE
User-Agent: Grandstream HT487 1.0.8.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Type: application/sdp
Content-Length: 307
v=0
o=486Tallinn 8000 8001 IN IP4 84.52.18.106
s=SIP Call
c=IN IP4 84.52.18.106
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 97
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=ptime:20
14 headers, 15 lines
Using latest request as basis request
Sending to 84.52.18.106 : 5060 (NAT)
Found user '486Tallinn'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 97
Peer audio RTP is at port 84.52.18.106:5004
Found description format PCMU
Found description format PCMA
Found description format G723
Found description format G729
Found description format G726-32
Found description format iLBC
Capabilities: us - 0x410 (g726|ilbc), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x410 (g726|ilbc)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for 10.42.42.1 in demo
Reliably Transmitting (NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 84.52.18.106;branch=z9hG4bK86ef5382cd2018a2;received=84.52.18.106;rport=5060
From: "Ljudmilla" <sip:486Tallinn@mail.huskydog.org.uk>;tag=36b99140642f9ae5
To: <sip:10.42.42.1@mail.huskydog.org.uk>;tag=as5c62fcdd
Call-ID: 7f9d87b30897cf15@84.52.18.106
CSeq: 57405 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:10.42.42.1@81.187.172.195>
Content-Length: 0
to 84.52.18.106:5060
server*CLI>
Sip read:
ACK sip:10.42.42.1@mail.huskydog.org.uk SIP/2.0
Via: SIP/2.0/UDP 84.52.18.106;branch=z9hG4bK86ef5382cd2018a2
From: "Ljudmilla" <sip:486Tallinn@mail.huskydog.org.uk>;tag=36b99140642f9ae5
To: <sip:10.42.42.1@mail.huskydog.org.uk>;tag=as5c62fcdd
Contact: <sip:486Tallinn@84.52.18.106>
Proxy-Authorization: Digest username="486Tallinn", realm="huskydog.org.uk", algorithm=MD5, uri="sip:1000@mail.huskydog.org.uk", nonce="29eda9b7", response="9304bc4846bacd836836186a88f802de"
Call-ID: 7f9d87b30897cf15@84.52.18.106
CSeq: 57405 ACK
User-Agent: Grandstream HT487 1.0.8.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE
Content-Length: 0
12 headers, 0 lines
Destroying call '7f9d87b30897cf15@84.52.18.106'
server*CLI> Note: User-Agent says Grandstream HT487, but device definately has HandyTone-486 written on it.
HELP!!
Thanks