Nir Simionovich wrote: > 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:[EMAIL PROTECTED]:5060;maddr=10.0.0.200 SIP/2.0 > Record-Route: <sip:[EMAIL PROTECTED]:5060;maddr=69.xx.xx.xx>, > <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5060> > From: Sason > <sip:[EMAIL > PROTECTED]:5081>;tag=KmsqOjY5LjIwLjYxLjIxOTo1MDYyOzY5LjIwLjYxLjIxOTo1MDgx > CSeq: 1 INVITE > Call-ID: [EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]> > Contact: <sip:[EMAIL PROTECTED]: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 - [EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]> > 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:[EMAIL > PROTECTED]:5081>;tag=KmsqOjY5LjIwLjYxLjIxOTo1MDYyOzY5LjIwLjYxLjIxOTo1MDgx > To: <sip:[EMAIL PROTECTED]:5060>;tag=as6343d6ca > Call-ID: [EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]> > CSeq: 1 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY > Contact: <sip:[EMAIL PROTECTED]> > Proxy-Authenticate: Digest realm="asterisk", nonce="162720d1" > Content-Length: 0 > > --- > Scheduling destruction of call '[EMAIL PROTECTED]' > <mailto:'[EMAIL PROTECTED]'> 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:[EMAIL > PROTECTED]:5081>;tag=KmsqOjY5LjIwLjYxLjIxOTo1MDYyOzY5LjIwLjYxLjIxOTo1MDgx > To: <sip:[EMAIL PROTECTED]:5060>;tag=as6343d6ca > Call-ID: [EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]> > CSeq: 1 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY > Contact: <sip:[EMAIL PROTECTED]> > 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:[EMAIL > PROTECTED]:5081>;tag=KmsqOjY5LjIwLjYxLjIxOTo1MDYyOzY5LjIwLjYxLjIxOTo1MDgx > To: <sip:[EMAIL PROTECTED]:5060>;tag=as6343d6ca > Call-ID: [EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]> > CSeq: 1 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY > Contact: <sip:[EMAIL PROTECTED]> > 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: > > [EMAIL PROTECTED] 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 > [EMAIL PROTECTED] root]# cat /etc/asterisk/sip_additional.conf > register=TollIPdemo1:[EMAIL PROTECTED] > > [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. > The sip-devices friend has a secret, thus Asterisk requires authentication. Using type=friend when setting up a connection to a service provider is not recommended. See all the examples for other service providers on the Wiki.
I would recommend that you add another peer, with the same host name after this entry. Asterisk matches the last one in sip.conf. In this peer, add "insecure=very" to disable authentication. Regards, /Olle ---- Astricon - the Asterisk User's conference - Madrid June 15-17 http://www.astricon.net/europe/ - Register today! _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users