[Asterisk-Users] Issue with SIP inter-op

[Nir Simionovich]

Hi All,
 
  I'm trying to connect to a SIP carrier who never connected with Asterisk. 
I managed to connect with a sipura phone or a grandstream, no problem.
 
  When I configure asterisk, I'm able to send out calls to the carrier no
problems,
however, receiving calls doesn't work, and I keep getting the following
messages:
 
<-- SIP read from 69.xx.xx.xx:5060:
INVITE sip:s at 10.0.0.200:5060;maddr=10.0.0.200 SIP/2.0
Record-Route: <sip:83555501 at 69.xx.xx.xx:5060;maddr=69.xx.xx.xx>,
<sip:83555501 at 69.xx.xx.xx:5062;maddr=69.xx.xx.xx>
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
To: <sip:83555501 at 69.xx.xx.xx:5060>
From: Sason
<sip:grouphone0 at 69.xx.xx.xx:5081>;tag=KmsqOjY5LjIwLjYxLjIxOTo1MDYyOzY5LjIwLj
YxLjIxOTo1MDgx
CSeq: 1 INVITE
Call-ID: bc1e6d746b7c0e4df at 192.168.1.3
Contact: <sip:grouphone0 at 69.xx.xx.xx:5081>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 386
 
v=0
o=- 8000 1 IN IP4 69.xx.xx.xx
s=-
c=IN IP4 69.xx.xx.xx
t=0 0
m=audio 31060 RTP/AVP 4 18 0 8 2 15 99 101
a=sendrecv
a=rtpmap:4 G723/8000/3
a=rtpmap:18 G729/8000/3
a=rtpmap:0 PCMU/8000/3
a=rtpmap:8 PCMA/8000/3
a=rtpmap:2 G726-32/8000/3
a=rtpmap:15 G728/8000/3
a=rtpmap:99 iLBC/8000/3
a=fmtp:99 mode=20
a=ptime:60
a=rtpmap:101 telephone-event/8000/3
a=fmtp:101 0-11
 
--- (11 headers 18 lines)---
Using INVITE request as basis request - bc1e6d746b7c0e4df at 192.168.1.3
Sending to 69.xx.xx.xx : 5060 (NAT)
Found peer 'sip-devices'
Reliably Transmitting (no NAT) to 69.xx.xx.xx:5060:
SIP/2.0 407 Proxy Authentication Required
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
From: Sason
<sip:grouphone0 at 69.xx.xx.xx:5081>;tag=KmsqOjY5LjIwLjYxLjIxOTo1MDYyOzY5LjIwLj
YxLjIxOTo1MDgx
To: <sip:83555501 at 69.xx.xx.xx:5060>;tag=as6343d6ca
Call-ID: bc1e6d746b7c0e4df at 192.168.1.3
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:s at 10.0.0.200>
Proxy-Authenticate: Digest realm="asterisk", nonce="162720d1"
Content-Length: 0
 
---
Scheduling destruction of call 'bc1e6d746b7c0e4df at 192.168.1.3' in 15000 ms
Retransmitting #1 (no NAT) to 69.xx.xx.xx:5060:
SIP/2.0 407 Proxy Authentication Required
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
From: Sason
<sip:grouphone0 at 69.xx.xx.xx:5081>;tag=KmsqOjY5LjIwLjYxLjIxOTo1MDYyOzY5LjIwLj
YxLjIxOTo1MDgx
To: <sip:83555501 at 69.xx.xx.xx:5060>;tag=as6343d6ca
Call-ID: bc1e6d746b7c0e4df at 192.168.1.3
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:s at 10.0.0.200>
Proxy-Authenticate: Digest realm="asterisk", nonce="162720d1"
Content-Length: 0
 
---
Retransmitting #2 (no NAT) to 69.xx.xx.xx:5060:
SIP/2.0 407 Proxy Authentication Required
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
From: Sason
<sip:grouphone0 at 69.xx.xx.xx:5081>;tag=KmsqOjY5LjIwLjYxLjIxOTo1MDYyOzY5LjIwLj
YxLjIxOTo1MDgx
To: <sip:83555501 at 69.xx.xx.xx:5060>;tag=as6343d6ca
Call-ID: bc1e6d746b7c0e4df at 192.168.1.3
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:s at 10.0.0.200>
Proxy-Authenticate: Digest realm="asterisk", nonce="162720d1"
Content-Length: 0
 
Any idea what may be causing this ?
 
The configuration is using AMP, and it looks as following:
 
[root at ipbx root]# cat /etc/asterisk/sip.conf
; Note: If your SIP devices are behind a NAT and your Asterisk
;  server isn't, try adding "nat=1" to each peer definition to
;  solve translation problems.
 
[general]
port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
externip = 62.219.XXX.XXX
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
nat = yes
 
#include sip_nat.conf
#include sip_additional.conf

[root at crystalclear root]# cat /etc/asterisk/sip_additional.conf
register=TollIPdemo1:somesecret at sipdevice.FQDN.net
 
[sip-devices]
username=TollIPdemo1
type=friend
secret=somesecret
host=sipdevice.FQDN.net
fromuser=TollIPdemo1
context=from-pstn
canreinvite=no
callerid=TollIPdemo1

Any information would be highly appreciated.
 
Nir S
 
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