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<DIV dir=ltr align=left><SPAN class=295471709-06062005><FONT face=Arial
size=2>Hi All,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=295471709-06062005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=295471709-06062005><FONT face=Arial
size=2> I'm trying to connect to a SIP carrier who never connected with
Asterisk. </FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=295471709-06062005><FONT face=Arial size=2>I
managed to connect with a sipura phone or a grandstream, no
problem.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=295471709-06062005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=295471709-06062005><FONT face=Arial
size=2> When I configure asterisk, I'm able to send out calls to the
carrier no problems,<BR>however, receiving calls doesn't work, and I keep
getting the following messages:</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=295471709-06062005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=295471709-06062005><FONT face=Arial
size=2><-- SIP read from 69.xx.xx.xx:5060:<BR>INVITE
sip:s@10.0.0.200:5060;maddr=10.0.0.200 SIP/2.0<BR>Record-Route:
<sip:83555501@69.xx.xx.xx:5060;maddr=69.xx.xx.xx>,
<sip:83555501@69.xx.xx.xx:5062;maddr=69.xx.xx.xx><BR>Via: SIP/2.0/UDP
69.xx.xx.xx:5060, SIP/2.0/UDP 69.xx.xx.xx:5062, SIP/2.0/UDP 69.xx.xx.xx:5062,
SIP/2.0/UDP
69.xx.xx.xx:5081;branch=z9hG4bKB18439727BD34C17BE5095ACF45BE007<BR>To:
<sip:83555501@69.xx.xx.xx:5060><BR>From: Sason
<sip:grouphone0@69.xx.xx.xx:5081>;tag=KmsqOjY5LjIwLjYxLjIxOTo1MDYyOzY5LjIwLjYxLjIxOTo1MDgx<BR>CSeq:
1 INVITE<BR>Call-ID: <A
href="mailto:bc1e6d746b7c0e4df@192.168.1.3">bc1e6d746b7c0e4df@192.168.1.3</A><BR>Contact:
<sip:grouphone0@69.xx.xx.xx:5081><BR>Max-Forwards: 70<BR>Content-Type:
application/sdp<BR>Content-Length: 386</FONT></SPAN></DIV>
<DIV> </DIV>
<DIV dir=ltr align=left><SPAN class=295471709-06062005><FONT face=Arial
size=2>v=0<BR>o=- 8000 1 IN IP4 69.xx.xx.xx<BR>s=-<BR>c=IN IP4
69.xx.xx.xx<BR>t=0 0<BR>m=audio 31060 RTP/AVP 4 18 0 8 2 15 99
101<BR>a=sendrecv<BR>a=rtpmap:4 G723/8000/3<BR>a=rtpmap:18
G729/8000/3<BR>a=rtpmap:0 PCMU/8000/3<BR>a=rtpmap:8 PCMA/8000/3<BR>a=rtpmap:2
G726-32/8000/3<BR>a=rtpmap:15 G728/8000/3<BR>a=rtpmap:99
iLBC/8000/3<BR>a=fmtp:99 mode=20<BR>a=ptime:60<BR>a=rtpmap:101
telephone-event/8000/3<BR>a=fmtp:101 0-11</FONT></SPAN></DIV>
<DIV> </DIV>
<DIV dir=ltr align=left><SPAN class=295471709-06062005><FONT face=Arial
size=2>--- (11 headers 18 lines)---<BR>Using INVITE request as basis request -
<A
href="mailto:bc1e6d746b7c0e4df@192.168.1.3">bc1e6d746b7c0e4df@192.168.1.3</A><BR>Sending
to 69.xx.xx.xx : 5060 (NAT)<BR>Found peer 'sip-devices'<BR>Reliably Transmitting
(no NAT) to 69.xx.xx.xx:5060:<BR>SIP/2.0 407 Proxy Authentication
Required<BR>Via: SIP/2.0/UDP 69.xx.xx.xx:5060, SIP/2.0/UDP 69.xx.xx.xx:5062,
SIP/2.0/UDP 69.xx.xx.xx:5062, SIP/2.0/UDP
69.xx.xx.xx:5081;branch=z9hG4bKB18439727BD34C17BE5095ACF45BE007<BR>From: Sason
<sip:grouphone0@69.xx.xx.xx:5081>;tag=KmsqOjY5LjIwLjYxLjIxOTo1MDYyOzY5LjIwLjYxLjIxOTo1MDgx<BR>To:
<sip:83555501@69.xx.xx.xx:5060>;tag=as6343d6ca<BR>Call-ID: <A
href="mailto:bc1e6d746b7c0e4df@192.168.1.3">bc1e6d746b7c0e4df@192.168.1.3</A><BR>CSeq:
1 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER, NOTIFY<BR>Contact: <sip:s@10.0.0.200><BR>Proxy-Authenticate:
Digest realm="asterisk", nonce="162720d1"<BR>Content-Length:
0<BR> <BR>---<BR>Scheduling destruction of call <A
href="mailto:'bc1e6d746b7c0e4df@192.168.1.3'">'bc1e6d746b7c0e4df@192.168.1.3'</A>
in 15000 ms<BR>Retransmitting #1 (no NAT) to 69.xx.xx.xx:5060:<BR>SIP/2.0 407
Proxy Authentication Required<BR>Via: SIP/2.0/UDP 69.xx.xx.xx:5060, SIP/2.0/UDP
69.xx.xx.xx:5062, SIP/2.0/UDP 69.xx.xx.xx:5062, SIP/2.0/UDP
69.xx.xx.xx:5081;branch=z9hG4bKB18439727BD34C17BE5095ACF45BE007<BR>From: Sason
<sip:grouphone0@69.xx.xx.xx:5081>;tag=KmsqOjY5LjIwLjYxLjIxOTo1MDYyOzY5LjIwLjYxLjIxOTo1MDgx<BR>To:
<sip:83555501@69.xx.xx.xx:5060>;tag=as6343d6ca<BR>Call-ID: <A
href="mailto:bc1e6d746b7c0e4df@192.168.1.3">bc1e6d746b7c0e4df@192.168.1.3</A><BR>CSeq:
1 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER, NOTIFY<BR>Contact: <sip:s@10.0.0.200><BR>Proxy-Authenticate:
Digest realm="asterisk", nonce="162720d1"<BR>Content-Length:
0<BR> <BR>---<BR>Retransmitting #2 (no NAT) to 69.xx.xx.xx:5060:<BR>SIP/2.0
407 Proxy Authentication Required<BR>Via: SIP/2.0/UDP 69.xx.xx.xx:5060,
SIP/2.0/UDP 69.xx.xx.xx:5062, SIP/2.0/UDP 69.xx.xx.xx:5062, SIP/2.0/UDP
69.xx.xx.xx:5081;branch=z9hG4bKB18439727BD34C17BE5095ACF45BE007<BR>From: Sason
<sip:grouphone0@69.xx.xx.xx:5081>;tag=KmsqOjY5LjIwLjYxLjIxOTo1MDYyOzY5LjIwLjYxLjIxOTo1MDgx<BR>To:
<sip:83555501@69.xx.xx.xx:5060>;tag=as6343d6ca<BR>Call-ID: <A
href="mailto:bc1e6d746b7c0e4df@192.168.1.3">bc1e6d746b7c0e4df@192.168.1.3</A><BR>CSeq:
1 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER, NOTIFY<BR>Contact: <sip:s@10.0.0.200><BR>Proxy-Authenticate:
Digest realm="asterisk", nonce="162720d1"<BR>Content-Length:
0</FONT></SPAN></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><SPAN class=295471709-06062005><FONT face=Arial size=2>Any idea what may be
causing this ?</FONT></SPAN></DIV>
<DIV><SPAN class=295471709-06062005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=295471709-06062005><FONT face=Arial size=2>The configuration is
using AMP, and it looks as following:</FONT></SPAN></DIV>
<DIV><SPAN class=295471709-06062005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=295471709-06062005><FONT face=Arial size=2>[root@ipbx root]#
cat /etc/asterisk/sip.conf<BR>; Note: If your SIP devices are behind a NAT and
your Asterisk<BR>; server isn't, try adding "nat=1" to each peer
definition to<BR>; solve translation problems.</FONT></SPAN></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><SPAN class=295471709-06062005><FONT face=Arial
size=2>[general]</FONT></SPAN></DIV>
<DIV><SPAN class=295471709-06062005><FONT face=Arial size=2>port =
5060 ; Port to bind
to (SIP is 5060)<BR>bindaddr = 0.0.0.0 ; Address to bind to
(all addresses on machine)<BR>externip =
62.219.XXX.XXX<BR>disallow=all<BR>allow=ulaw<BR>allow=alaw<BR>context =
from-sip-external ; Send unknown SIP callers to this context<BR>callerid =
Unknown<BR>nat = yes</FONT></SPAN></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><SPAN class=295471709-06062005><FONT face=Arial size=2>#include
sip_nat.conf<BR>#include sip_additional.conf<BR></FONT></SPAN></DIV>
<DIV><SPAN class=295471709-06062005><FONT face=Arial size=2>[root@crystalclear
root]# cat
/etc/asterisk/sip_additional.conf<BR>register=TollIPdemo1:somesecret@sipdevice.FQDN.net</FONT></SPAN></DIV>
<DIV><SPAN class=295471709-06062005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=295471709-06062005><FONT face=Arial
size=2>[sip-devices]<BR>username=TollIPdemo1<BR>type=friend<BR>secret=somesecret<BR>host=sipdevice.FQDN.net<BR>fromuser=TollIPdemo1<BR>context=from-pstn<BR>canreinvite=no<BR>callerid=TollIPdemo1<BR></FONT></SPAN></DIV>
<DIV><SPAN class=295471709-06062005><FONT face=Arial size=2>Any information
would be highly appreciated.</FONT></SPAN></DIV>
<DIV><SPAN class=295471709-06062005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=295471709-06062005><FONT face=Arial size=2>Nir
S</DIV></FONT></SPAN>
<DIV><SPAN class=295471709-06062005><FONT face=Arial
size=2> </DIV></FONT></SPAN></BODY></HTML>