UKVoIPTalk.com

I've been getting reports that people can't dial my voip.co.uk or Sipgate
numbers, and a few test calls reveal:

Sending to 217.10.79.23 : 5060 (non-NAT)
Found RTP audio format 8
<snip loads of this>
Found RTP audio format 10
Peer audio RTP is at port 217.10.66.71:12052
Found description format PCMA
<snip loads of this>
Found description format L16
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x5fe (gsm
ulaw|alaw|g726|adpcm|slin|lpc10|g729|ilbc)/video=0x0 (nothing), combined -
0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Looking for 640xxxx in default (domain 85.189.113.174)
Reliably Transmitting (no NAT) to 217.10.79.23:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
217.10.79.23;branch=z9hG4bK7a1b.e8753713.0;received=192.168.254.2
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK7a1b.407746b2.0
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK16462595;rport=5060
From: "0x9x4x4x4x3" <sip:0x9x4x4x4x3@217.10.66.71>;tag=as639fe80a
To: <sip:44191640xxxx@217.10.79.8>;tag=as56ddf6fc
Call-ID: 14efe9852138a7c928aef19e7554857a@217.10.66.71
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


[default] isn't where I want the calls to go - it should be
[incoming_sipgate] or [incoming_voipcouk] as appropriate. The trunks each
have the relevant context in sip.conf.

Restarting Asterisk will get incoming calls working again for a few minutes:


<-- SIP read from 217.10.79.23:5060:
INVITE sip:640xxxx@85.189.113.174 SIP/2.0
Record-Route: <sip:217.10.79.23;lr=on>
Record-Route: <sip:217.10.79.8;ftag=as027acffd;lr=on>
Via: SIP/2.0/UDP 217.10.79.23;branch=z9hG4bKa8f4.94b368a1.0
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKa8f4.26f50137.0
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK78e3fb9a;rport=5060
From: "0x9x4x4x4x3" <sip:0x9x4x4x4x3@217.10.66.71>;tag=as027acffd
To: <sip:44191640xxxx@217.10.79.8>
Contact: <sip:0x9x4x4x4x3@217.10.66.71>
Call-ID: 763bb79f09c49a3c4a56ff2f59f587b3@217.10.66.71
CSeq: 102 INVITE
User-Agent: sipgate asterisk
Max-Forwards: 15
Date: Mon, 07 May 2007 21:26:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 426

v=0
o=root 15612 15612 IN IP4 217.10.66.71
s=session
c=IN IP4 217.10.66.71
t=0 0
m=audio 19300 RTP/AVP 8 0 3 97 18 111 5 7 10
a=rtpmap:8 PCMA/8000
<snip loads of this>
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

--- (18 headers 20 lines) ---
Using INVITE request as basis request -
763bb79f09c49a3c4a56ff2f59f587b3@217.10.66.71
Sending to 217.10.79.23 : 5060 (non-NAT)
Found peer 'sipgate'
Found RTP audio format 8
<snip loads of this>
Found RTP audio format 10
Peer audio RTP is at port 217.10.66.71:19300
Found description format PCMA
<snip loads of this>
Found description format L16
Capabilities: us - 0x2 (gsm), peer - audio=0x5fe (gsm|ulaw|alaw|g726|adpcm
slin|lpc10|g729|ilbc)/video=0x0 (nothing), combined - 0x2 (gsm)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Looking for 640xxxx in incoming_sipgate (domain 85.189.113.174)
list_route: hop: <sip:217.10.79.23;lr=on>
list_route: hop: <sip:217.10.79.8;ftag=as027acffd;lr=on>
Transmitting (no NAT) to 217.10.79.23:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
217.10.79.23;branch=z9hG4bKa8f4.94b368a1.0;received=217.10.79.23
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKa8f4.26f50137.0
Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK78e3fb9a;rport=5060
From: "0x9x4x4x4x3" <sip:0x9x4x4x4x3@217.10.66.71>;tag=as027acffd
To: <sip:44191640xxxx@217.10.79.8>
Call-ID: 763bb79f09c49a3c4a56ff2f59f587b3@217.10.66.71
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:640xxxx@85.189.113.174>
Content-Length: 0


But usually within 10 minutes, they've stopped working. I've followed the
advice at [url]http://forums.digium.com/viewtopic.php?p=49136[/url] and changed my
insecure = very to insecure = port,invite. Even when it isn't working, the
context for the peer is right, it's just that calls don't go to the right
place:

westogre*CLI> sip show peer sipgate

* Name : sipgate
Secret : <Set>
MD5Secret : <Not set>
Context : incoming_sipgate


Can anyone suggest anything, short of putting everything in [default]?

--
<http://ale.cx/> (AIM:troffasky) (UnSoEsNpEaTm@ale.cx)
22:31:15 up 9 days, 30 min, 1 user, load average: 0.52, 0.45, 0.32
09 f9 11 02 9d 74 e3 5b d8 41 56 c5 63 56 88 c0

alexd wrote:
[color=blue]
> Can anyone suggest anything, short of putting everything in [default]?[/color]

The answer to my own question is, "fix the iptables port forwarding rules on
your gateway so that forwarded traffic originates from it's actual source
address rather than that of the gateway". So Asterisk was behaving exactly
to spec - the port-forwarded SIP traffic appeared to Asterisk to be coming
from the gateway rather than from Sipgate or voip.co.uk, so Asterisk sent
all the inbound calls to default because it has no idea [in SIP terms] what
the gateway is.

--
<http://ale.cx/> (AIM:troffasky) (UnSoEsNpEaTm@ale.cx)
21:43:01 up 10 days, 23:42, 2 users, load average: 0.56, 0.34, 0.26
09 f9 11 02 9d 74 e3 5b d8 41 56 c5 63 56 88 c0