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)
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