[asterisk-users] At whit's end was 'DHCP Failure screws up system '

[Eric Wieling]

Remove the qualify= option from sip.conf.  Also make sure the DISABLE 
CDP in the Polycom's boot menu.

Doug Lytle wrote:
> Hey everybody,
> 
> I'm still having issues with this system.  The phones won't stay 
> registered for more then a few minutes.  They're bouncing up and down.  
> I'm able to ping the phones just fine.  What I've done so far:
> 
> Power cycled all phones and verified
> Power cycled all switches
> Checked the ARP tables on the routers/phone system (Seems to be okay)
> Upgraded Asterisk to 1.4.19.2
> 
> Wireshark shows UDP checksum errors, but from what I can see on Google, 
> this may be normal.
> 
> If I am on one of the phones when it goes AWOL, the call is not 
> interrupted, but as soon as I hang up, I can't use it.
> 
> Any other suggestions?
> 
> 
> Captured a sip debug as one of the extensions was dropping:
> 
> 
> 
> Reliably Transmitting (no NAT) to 10.10.10.198:5060:
> OPTIONS sip:4247 at 10.10.10.198 SIP/2.0
> Via: SIP/2.0/UDP 10.10.10.15:5060;branch=z9hG4bK122e04b7;rport
> From: "asterisk" <sip:asterisk at 10.10.10.15>;tag=as1ed0ad8f
> To: <sip:4247 at 10.10.10.198>
> Contact: <sip:asterisk at 10.10.10.15>
> Call-ID: 5080b35c0caca7a06943a01e77f95f2c at 10.10.10.15
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Tue, 20 May 2008 14:47:40 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
> 
> 
> ---
> Retransmitting #1 (no NAT) to 10.10.10.198:5060:
> OPTIONS sip:4247 at 10.10.10.198 SIP/2.0
> Via: SIP/2.0/UDP 10.10.10.15:5060;branch=z9hG4bK122e04b7;rport
> From: "asterisk" <sip:asterisk at 10.10.10.15>;tag=as1ed0ad8f
> To: <sip:4247 at 10.10.10.198>
> Contact: <sip:asterisk at 10.10.10.15>
> Call-ID: 5080b35c0caca7a06943a01e77f95f2c at 10.10.10.15
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Tue, 20 May 2008 14:47:40 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
> 
> ---
> [May 20 10:47:41] NOTICE[3900]: chan_sip.c:15863 sip_poke_noanswer: Peer 
> '4247' is now UNREACHABLE!  Last qualify: 39
> 
> 
> 
> _______________________________________________
> -- 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
> 
> 

-- 
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, 
T-1, PRI, Frame Relay, Linux, and network design.  Based near 
Birmingham, AL.  Now accepting clients worldwide.