Doug Lytle
2008-May-20 15:51 UTC
[asterisk-users] At whit's end was 'DHCP Failure screws up system '
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
Patrick
2008-May-20 16:17 UTC
[asterisk-users] At whit's end was 'DHCP Failure screws up system '
On Tue, 2008-05-20 at 11:51 -0400, 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.Not sure if this helps but iirc I've seen checksum issues on an Asterisk & DHCP box that I was able to get rid of by turning off some of the checksum offloading with ethtool. I have these new settings on the box where the errors used to occur (so errors are gone now with these settings): rx-checksumming: off tx-checksumming: on scatter-gather: on tcp segmentation offload: on udp fragmentation offload: off generic segmentation offload: off Regards, Patrick
Eric Wieling
2008-May-20 17:33 UTC
[asterisk-users] At whit's end was 'DHCP Failure screws up system '
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.