Re: [asterisk-users] Registration of Snom 320 phone withAsterisk 1.4.13

I guess the problem is that * sends the response to port 5060, while the phone listens on port 2xxx for an answer.

CS

-----Ursprüngliche Nachricht-----
Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Im Auftrag von Jason White
Gesendet: Montag, 29. Oktober 2007 07:46
An: asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] Registration of Snom 320 phone withAsterisk 1.4.13

Here are more details:

The phone and the Asterisk box are behind the same router (the Asterisk machine is 192.168.0.2 and the phone is 192.168.0.4).

A ping command works:

jason@jdc:~$ ping -c 10 192.168.0.4
PING 192.168.0.4 (192.168.0.4) 56(84) bytes of data.
64 bytes from 192.168.0.4: icmp_seq=1 ttl=64 time=0.500 ms
64 bytes from 192.168.0.4: icmp_seq=2 ttl=64 time=0.491 ms
64 bytes from 192.168.0.4: icmp_seq=3 ttl=64 time=0.493 ms
64 bytes from 192.168.0.4: icmp_seq=4 ttl=64 time=0.495 ms
64 bytes from 192.168.0.4: icmp_seq=5 ttl=64 time=0.495 ms
64 bytes from 192.168.0.4: icmp_seq=6 ttl=64 time=0.493 ms
64 bytes from 192.168.0.4: icmp_seq=7 ttl=64 time=0.493 ms
64 bytes from 192.168.0.4: icmp_seq=8 ttl=64 time=0.495 ms
64 bytes from 192.168.0.4: icmp_seq=9 ttl=64 time=0.505 ms
64 bytes from 192.168.0.4: icmp_seq=10 ttl=64 time=0.492 ms

--- 192.168.0.4 ping statistics ---
10 packets transmitted, 10 received, 0% packet loss, time 9005ms rtt min/avg/max/mdev = 0.491/0.495/0.505/0.014 ms jason@jdc:~$

However, the phone never appears to receive the responses from Asterisk to its register requests. The error on the phone is:
[2]29/10/2007 17:02:59: Transport Error: Pending packet 1046807: generating fake
[2]29/10/2007 17:02:59: Registrar 5549@192.168.0.2 timed out

From /etc/asterisk/sip.conf:

[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to A
realm=192.168.0.2
context = default ;Default for incoming calls [5549] disallow=all allow=ulaw allow=alaw allow=gsm type=friend ;(inbound and outbound calls accepted) secret=localphone ; obvious password for testing host=dynamic callerid=Jason White <5549> dtmfmode=auto
mailbox=5549 ;(Asterisk VM-system's mailbox #)

The output from sip set debug is attached, as captured earlier by the script command.

Asterisk version 1.4.13, Debian GNU/Linux Sid (up to date); this phone has successfully registered with external Asterisk servers.

Suggestions are much appreciated.

_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-users