Hi,<br>
<br>
I am experiencing a hard to solve problem with my VoIP provider. I can
make calls without any problem but I can not receive any. Actually,
calls arive to * but the phone just does not ring. I believe must
be a problem with NAT but I think I have a good config:<br>
- Extensions have nat=always and qualify=yes<br>
- Have introduced in sip.conf Externip and localnet<br>
- ADSL modem/router is redirected to my * server<br>
- With sip debug I can see the call arrives<br>
Am I misssing something that someone else can see?<br>
<br>
Appreciate any hint. Thanks<br>
====================================<br>
ASTERISK VERSION: <br>
Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q<br>
<br>
SIP DEBUG CAPTURE<br>
<pre><-- SIP read from <a href="http://62.22.20.194:5060">62.22.20.194:5060</a>:
INVITE sip:34700758288001@87.218.175.120:5060 SIP/2.0
Record-Route: <sip:<a href="http://62.22.20.194">62.22.20.194</a>;ftag=08ff6000ff05ff10ff00000e0c4effff;lr>
Via: SIP/2.0/UDP <a href="http://62.22.20.194">62.22.20.194</a>;branch=z9hG4bK90bf.b9c560e1.0
Via: SIP/2.0/UDP <a href="http://62.22.20.207:5060">62.22.20.207:5060</a>;branch=z9hG4bK-10ff6000ff05ff10ff00000e0c4effff
From: <<a href="mailto:sip:690351498@62.22.20.207">sip:690351498@62.22.20.207</a>;user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff
To: <sip:34700758288001@62.22.20.194:5060;user=phone>
Call-ID: d2c76000bf05c0108000000e0c4ef4b3@siphit-1
CSeq: 1 INVITE
Contact: <<a href="mailto:sip:690351498@62.22.20.207">sip:690351498@62.22.20.207</a>;user=phone>
Max-Forwards: 9
User-Agent: MERA MSIP v.1.0.2
Cisco-Guid: 908093991-393679323-3151091529-1429652222
Content-Type: application/sdp
Content-Length: 216
v=0
o=- 1153435071 1153435071 IN IP4 <a href="http://62.22.20.207">62.22.20.207</a>
s=-
c=IN IP4 <a href="http://62.22.20.207">62.22.20.207</a>
t=0 0
m=audio 59320 RTP/AVP 18 4 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--- (14 headers 10 lines)---
Using INVITE request as basis request - d2c76000bf05c0108000000e0c4ef4b3@siphit-1
Sending to <a href="http://62.22.20.194">62.22.20.194</a> : 5060 (non-NAT)
Found peer 'Peoplecall'
Reliably Transmitting (NAT) to <a href="http://62.22.20.194:5060">62.22.20.194:5060</a>:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP <a href="http://62.22.20.194">62.22.20.194</a>;branch=z9hG4bK90bf.b9c560e1.0;received=<a href="http://62.22.20.194">62.22.20.194</a>
Via: SIP/2.0/UDP <a href="http://62.22.20.207:5060">62.22.20.207:5060</a>;branch=z9hG4bK-10ff6000ff05ff10ff00000e0c4effff
From: <<a href="mailto:sip:690351498@62.22.20.207">sip:690351498@62.22.20.207</a>;user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff
To: <sip:34700758288001@62.22.20.194:5060;user=phone>;tag=as476d14de
Call-ID: d2c76000bf05c0108000000e0c4ef4b3@siphit-1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <<a href="mailto:sip:34700758288001@87.218.175.74">sip:34700758288001@87.218.175.74</a>>
Proxy-Authenticate: Digest realm="asterisk", nonce="008d23b0"
Content-Length: 0
---
Scheduling destruction of call 'd2c76000bf05c0108000000e0c4ef4b3@siphit-1' in 15000 ms
asterisk1*CLI>
<-- SIP read from <a href="http://62.22.20.194:5060">62.22.20.194:5060</a>:
ACK sip:34700758288001@87.218.175.120:5060 SIP/2.0
Via: SIP/2.0/UDP <a href="http://62.22.20.194">62.22.20.194</a>;branch=z9hG4bK90bf.b9c560e1.0
From: <<a href="mailto:sip:690351498@62.22.20.207">sip:690351498@62.22.20.207</a>;user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff
Call-ID: d2c76000bf05c0108000000e0c4ef4b3@siphit-1
To: <sip:34700758288001@62.22.20.194:5060;user=phone>;tag=as476d14de
CSeq: 1 ACK
User-Agent: OpenSer (1.0.0 (i386/linux))
Content-Length: 0
--- (8 headers 0 lines)---
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to <a href="http://62.22.20.194:5060">62.22.20.194:5060</a>:
REGISTER sip:<a href="http://sip.peoplecall.com">sip.peoplecall.com</a> SIP/2.0
Via: SIP/2.0/UDP <a href="http://87.218.175.74:5060">87.218.175.74:5060</a>;branch=z9hG4bK4a6abe4f;rport
From: <<a href="mailto:sip:34700758288001@sip.peoplecall.com">sip:34700758288001@sip.peoplecall.com</a>>;tag=as79fdfc26
To: <<a href="mailto:sip:34700758288001@sip.peoplecall.com">sip:34700758288001@sip.peoplecall.com</a>>
Call-ID: <a href="mailto:1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1">1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1</a>
CSeq: 421 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="34700758288001", realm="<a href="http://sip.peoplecall.com">sip.peoplecall.com</a>", algorithm=MD5, uri="sip:<a href="http://sip.peoplecall.com">sip.peoplecall.com
</a>", nonce="44c0059db2d71f523aeb30399a54a4a32d8aeed6", response="ee782a37bae7eed1a0a881147c733ede", opaque=""
Expires: 120
Contact: <<a href="mailto:sip:34700758288001@87.218.175.74">sip:34700758288001@87.218.175.74</a>>
Event: registration
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from <a href="http://62.22.20.194:5060">62.22.20.194:5060</a>:
SIP/2.0 200 OK
Via: SIP/2.0/UDP <a href="http://192.168.1.104:5060">192.168.1.104:5060</a>;branch=z9hG4bK4a6abe4f;rport=5060
From: <<a href="mailto:sip:34700758288001@sip.peoplecall.com">sip:34700758288001@sip.peoplecall.com</a>>;tag=as79fdfc26
To: <<a href="mailto:sip:34700758288001@sip.peoplecall.com">sip:34700758288001@sip.peoplecall.com</a>>;tag=555271b30cfd40f8a3b4837b054360a3.975d
Call-ID: <a href="mailto:1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1">1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1</a>
CSeq: 421 REGISTER
Contact: <sip:34700758288001@192.168.1.104:5060>;expires=120
Server: OpenSer (1.0.0 (i386/linux))
Content-Length: 0
--- (9 headers 0 lines)---
Scheduling destruction of call '<a href="mailto:1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1">1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1</a>' in 32000 ms
Destroying call 'd2c76000bf05c0108000000e0c4ef4b3@siphit-1'
asterisk1*CLI> sip no debug
SIP Debugging Disabled</pre><br>