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support at drdos.info
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 Posted: Tue May 20, 2008 10:51 am Post subject: [asterisk-users] At whit's end was 'DHCP Failure screws up s |
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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 |
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eric at fnords.org
Guest
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| Posted: Tue May 20, 2008 12:33 pm Post subject: [asterisk-users] At whit's end was 'DHCP Failure screws up s |
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Remove the qualify= option from sip.conf. Also make sure the DISABLE
CDP in the Polycom's boot menu.
Doug Lytle wrote:
| Quote: | 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
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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. |
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support at drdos.info
Guest
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| Posted: Tue May 20, 2008 1:12 pm Post subject: [asterisk-users] At whit's end was 'DHCP Failure screws up s |
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Eric Wieling wrote:
| Quote: | 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! |
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eric at fnords.org
Guest
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| Posted: Tue May 20, 2008 2:06 pm Post subject: [asterisk-users] At whit's end was 'DHCP Failure screws up s |
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Doug Lytle wrote:
| Quote: | Eric Wieling wrote:
| Quote: | 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. |
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support at drdos.info
Guest
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| Posted: Tue May 20, 2008 2:38 pm Post subject: [asterisk-users] At whit's end was 'DHCP Failure screws up s |
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Eric Wieling wrote:
| Quote: | 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 |
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eric at fnords.org
Guest
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| Posted: Tue May 20, 2008 3:55 pm Post subject: [asterisk-users] At whit's end was 'DHCP Failure screws up s |
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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:
| Quote: | Eric Wieling wrote:
| Quote: | 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. |
|
| Back to top |
|
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anthonyf at rockynet.com
Guest
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| Posted: Wed May 21, 2008 8:56 am Post subject: [asterisk-users] At whit's end was 'DHCP Failure screws up s |
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Eric Wieling wrote:
| Quote: | Doug Lytle wrote:
| Quote: | Eric Wieling wrote:
| Quote: | 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? |
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support at drdos.info
Guest
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| Posted: Wed May 21, 2008 9:50 am Post subject: [asterisk-users] At whit's end was 'DHCP Failure screws up s |
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Anthony Francis wrote:
| Quote: | 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." |
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