Re: [asterisk-users] At whit's end was 'DHCP Failure screws up system '
Anthony Francis wrote: > Did you ever try turning off all phones, flushing the lease table and > bringing the phones back up? > > Yes, It made no difference. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- 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
Re: [asterisk-users] At whit's end was 'DHCP Failure screws up system '
Eric Wieling wrote: > Doug Lytle wrote: > >> Eric Wieling wrote: >> >>> Remove the qualify= option from sip.conf. Also make sure the DISABLE >>> CDP in the Polycom's boot menu. >>> >>> >> That didn't help and CDP is off by default, the phones still couldn't >> receive/send calls when in this state. I've sent an employee out to >> grab a replacement NIC. Hopefully this will fix things. >> > > Based on the SIP poke message you pasted in an earlier message, the > qualify= option you used is virtually guaranteed to cause SIP poke problems. > > Did you ever try turning off all phones, flushing the lease table and bringing the phones back up? ___ -- 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
Re: [asterisk-users] At whit's end was 'DHCP Failure screws up system '
SIP poke does NOT just measure network latency. It also measures the PHONE latency. Asterisk sends a SIP OPTIONS packet to the phone, the phone responds, Asterisk measures how long it took. Most phones seem to make responding to OPTIONS packets a low priority. A phone busy doing a registration, accepting a call, placing a call, etc and can easily cause the phone to take 2000ms or more to respond to the OPTIONS packet. Remember, most of these phones have so little general CPU power they can easily be overwhelmed by SIP traffic. Most phones have a dedicated chip to handle audio encoding/decoding, so audio would not normally be affected by the general CPU being bogged down. Doug Lytle wrote: > Eric Wieling wrote: >> Doug Lytle wrote: >> >> Based on the SIP poke message you pasted in an earlier message, the >> qualify= option you used is virtually guaranteed to cause SIP poke problems. >> > Understood, wouldn't it also indicate that, when putting 2 phones and > the phone system on it's own little switch and I still see the SIP poke > messgages that it's something to do with either the Polycoms or the system? > > Just struggling to understand what is going on. > > On the norm, we've never had anything above 60ms. Taking out the > qualify worked briefly. FOP showed statuses fine. The they started > dropping out and phones were unable to receive/make calls. > > Also, I replace the NIC and it made no difference and using 2 NICs (1 > for 192.x.x.x and one for the 10.10.10.x networks) instead of 2 IPs on > one interface made no difference either. > > I guess I go back to plan A and replace the machine. --- 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. ___ -- 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
Re: [asterisk-users] At whit's end was 'DHCP Failure screws up system '
Eric Wieling wrote: > Doug Lytle wrote: > > Based on the SIP poke message you pasted in an earlier message, the > qualify= option you used is virtually guaranteed to cause SIP poke problems. > Understood, wouldn't it also indicate that, when putting 2 phones and the phone system on it's own little switch and I still see the SIP poke messgages that it's something to do with either the Polycoms or the system? Just struggling to understand what is going on. On the norm, we've never had anything above 60ms. Taking out the qualify worked briefly. FOP showed statuses fine. The they started dropping out and phones were unable to receive/make calls. Also, I replace the NIC and it made no difference and using 2 NICs (1 for 192.x.x.x and one for the 10.10.10.x networks) instead of 2 IPs on one interface made no difference either. I guess I go back to plan A and replace the machine. Thanks again ___ -- 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
Re: [asterisk-users] At whit's end was 'DHCP Failure screws up system '
Doug Lytle wrote: > Eric Wieling wrote: >> Remove the qualify= option from sip.conf. Also make sure the DISABLE >> CDP in the Polycom's boot menu. >> > > That didn't help and CDP is off by default, the phones still couldn't > receive/send calls when in this state. I've sent an employee out to > grab a replacement NIC. Hopefully this will fix things. Based on the SIP poke message you pasted in an earlier message, the qualify= option you used is virtually guaranteed to cause SIP poke problems. ___ -- 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
Re: [asterisk-users] At whit's end was 'DHCP Failure screws up system '
Eric Wieling wrote: > Remove the qualify= option from sip.conf. Also make sure the DISABLE > CDP in the Polycom's boot menu. > That didn't help and CDP is off by default, the phones still couldn't receive/send calls when in this state. I've sent an employee out to grab a replacement NIC. Hopefully this will fix things. Thank you for your input though! ___ -- 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
Re: [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:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 10.10.10.15:5060;branch=z9hG4bK122e04b7;rport > From: "asterisk" ;tag=as1ed0ad8f > To: > Contact: > Call-ID: [EMAIL PROTECTED] > 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:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 10.10.10.15:5060;branch=z9hG4bK122e04b7;rport > From: "asterisk" ;tag=as1ed0ad8f > To: > Contact: > Call-ID: [EMAIL PROTECTED] > 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. ___ -- 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
Re: [asterisk-users] At whit's end was 'DHCP Failure screws up system '
Patrick wrote: > 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 > Thanks Patrick. I've narrowed the issue down a bit, I put 2 Polycom phones and the phone system on it's own switch, the phones register and a few seconds later they are bouncing. I'm guessing that there is an issue with the machine. I've got a spare at the office and will get it setup and put it into place tomorrow. I'm guessing that it's a NIC issue, but would rather put the spare into place. ___ -- 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
Re: [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 ___ -- 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
[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:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.10.10.15:5060;branch=z9hG4bK122e04b7;rport From: "asterisk" ;tag=as1ed0ad8f To: Contact: Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.10.10.15:5060;branch=z9hG4bK122e04b7;rport From: "asterisk" ;tag=as1ed0ad8f To: Contact: Call-ID: [EMAIL PROTECTED] 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
