Re: [asterisk-users] Timeout when dialing dead peer
Stefan Schmidt s...@sil.at writes: if i understand you right you have one server (peer) where thousands of devices are connected and every device is registered to asterisk, and so every options packet will come from asterisk to this device, right? If you have a sip routing server like ser, the server itself could do a Nat keep alive check, and could drops the invite coming from asterisk if the peer isnt reachable. If these devices arent registered to asterisk why do you think that there will be so much options Packets? if you have one peer this will get only one Options packet per minute. I have decided to go with the qualify solution as you recommend. I have set the qualify frequency to 1800 seconds; that way I should get the lowered t1 timing with very few qualify requests. It looks like it works, thanks! /Benny ___ -- 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] Timeout when dialing dead peer
Benny Amorsen schrieb: Stefan Schmidt s...@sil.at writes: if i understand you right you have one server (peer) where thousands of devices are connected and every device is registered to asterisk, and so every options packet will come from asterisk to this device, right? If you have a sip routing server like ser, the server itself could do a Nat keep alive check, and could drops the invite coming from asterisk if the peer isnt reachable. If these devices arent registered to asterisk why do you think that there will be so much options Packets? if you have one peer this will get only one Options packet per minute. I have decided to go with the qualify solution as you recommend. I have set the qualify frequency to 1800 seconds; that way I should get the lowered t1 timing with very few qualify requests. It looks like it works, thanks! You can also set the overall timeout with timerb: ;timerb=32000 ; Call setup timer. If a provisional response is not received ; in this amount of time, the call will autocongest ; Defaults to 64*timert1 regards klaus ___ -- 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] Timeout when dialing dead peer
Klaus Darilion klaus.mailingli...@pernau.at writes: ;timerb=32000 ; Call setup timer. If a provisional response is not received ; in this amount of time, the call will autocongest ; Defaults to 64*timert1 Thanks! Will try that. Just what I was looking for! /Benny ___ -- 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] Timeout when dialing dead peer
Benny Amorsen schrieb: A regular Dial(somepeer) to a SIP peer which doesn't reply at all seems to not time out, or at least have a very long time out. We have a set up where we can dial two different peers, a primary and a backup peer. If the first one dies completely, so that no error messages (no ICMP unreachables or anything) are returned, Asterisk does not continue in the dial plan but just gets stuck on that one Dial(). I can of course put a time out in the Dial(), but then one call will turn into two calls if the person at the other end is too slow to answer their phone, so this is not very handy. It is possible that qualify would help, but it is not a very nice answer -- Asterisk's use of SIP OPTIONs is non-standard, and it can impose a significant load on the peer. What kind of client cant handle one packet per minute without getting a higher load? The interval asterisk sends an Options packet is 60 seconds and the default timeout is 2 s for this packet. So i believe this coudnt be a problem, or do you have a problem with the peer when a second invite arrives during an active call? It would be good if Asterisk would give up after not receving any reply after a configurable interval. What your are searching for is called Sip T1 Timeout and i´ve seen that in asterisk 1.6.1.x you can set this timeout in sip.conf. Iam not sure about changing this in other versions. /Benny best regards steve ___ -- 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] Timeout when dialing dead peer
There is a timeout function in the Dial command. The folks who wrote the command obviously felt that setting a programmatic limit on this would cause somebody a problem. If you expect a reply from your SIP peer in 30 seconds, just do Dial(SIP/peer,30) and the line will disconnect in 30 seconds. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan Schmidt Sent: Monday, June 08, 2009 7:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Timeout when dialing dead peer Benny Amorsen schrieb: A regular Dial(somepeer) to a SIP peer which doesn't reply at all seems to not time out, or at least have a very long time out. We have a set up where we can dial two different peers, a primary and a backup peer. If the first one dies completely, so that no error messages (no ICMP unreachables or anything) are returned, Asterisk does not continue in the dial plan but just gets stuck on that one Dial(). I can of course put a time out in the Dial(), but then one call will turn into two calls if the person at the other end is too slow to answer their phone, so this is not very handy. It is possible that qualify would help, but it is not a very nice answer -- Asterisk's use of SIP OPTIONs is non-standard, and it can impose a significant load on the peer. What kind of client cant handle one packet per minute without getting a higher load? The interval asterisk sends an Options packet is 60 seconds and the default timeout is 2 s for this packet. So i believe this coudnt be a problem, or do you have a problem with the peer when a second invite arrives during an active call? It would be good if Asterisk would give up after not receving any reply after a configurable interval. What your are searching for is called Sip T1 Timeout and i´ve seen that in asterisk 1.6.1.x you can set this timeout in sip.conf. Iam not sure about changing this in other versions. /Benny best regards steve ___ -- 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 ___ -- 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] Timeout when dialing dead peer
Danny Nicholas schrieb: There is a timeout function in the Dial command. The folks who wrote the command obviously felt that setting a programmatic limit on this would cause somebody a problem. If you expect a reply from your SIP peer in 30 seconds, just do Dial(SIP/peer,30) and the line will disconnect in 30 seconds. which will not work in the situation as benny wrote, when the primary peers doesnt answer to any request coming from asterisk. so you will have an 30 second timeout. what i mean is the Sip internal timeout how long a peer is able to answer to this sip packet, which per default is 30 seconds. if you set the dial timeout lower than this sip timeout you will have a lower waiting time, but as benny said, if the client answer too slow its not handy to use. -Original Message- snip Benny Amorsen schrieb: If the first one dies completely, so that no error messages (no ICMP unreachables or anything) are returned, Asterisk does not continue in the dial plan but just gets stuck on that one Dial(). I can of course put a time out in the Dial(), but then one call will turn into two calls if the person at the other end is too slow to answer their phone, so this is not very handy. /snip i made a mistake in asterisk ver. 1.6.2.b2 you are able to setting the sip timers of your own see the sip.conf sample from this version below: ;--- SIP timers ; These timers are used primarily in INVITE transactions. ; The default for Timer T1 is 500 ms or the measured run-trip time between ; Asterisk and the device if you have qualify=yes for the device. ; ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts ; Defaults to 100 ms ;timert1=500; Default T1 timer ; Defaults to 500 ms or the measured round-trip ; time to a peer (qualify=yes). ;timerb=32000 ; Call setup timer. If a provisional response is not received ; in this amount of time, the call will autocongest ; Defaults to 64*timert1 best regards steve ___ -- 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] Timeout when dialing dead peer
Stefan Schmidt s...@sil.at writes: What kind of client cant handle one packet per minute without getting a higher load? It isn't a client. It handles thousands of connected devices, so it'll be handling perhaps 50 OPTIONS packets every second if I go the qualify route. What your are searching for is called Sip T1 Timeout and i´ve seen that in asterisk 1.6.1.x you can set this timeout in sip.conf. Iam not sure about changing this in other versions. If you're talking about t1min, AFAIK that only applies to monitored devices, i.e. those with qualify=yes. /Benny ___ -- 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] Timeout when dialing dead peer
Benny Amorsen schrieb: Stefan Schmidt s...@sil.at writes: What kind of client cant handle one packet per minute without getting a higher load? It isn't a client. It handles thousands of connected devices, so it'll be handling perhaps 50 OPTIONS packets every second if I go the qualify route. if i understand you right you have one server (peer) where thousands of devices are connected and every device is registered to asterisk, and so every options packet will come from asterisk to this device, right? If you have a sip routing server like ser, the server itself could do a Nat keep alive check, and could drops the invite coming from asterisk if the peer isnt reachable. If these devices arent registered to asterisk why do you think that there will be so much options Packets? if you have one peer this will get only one Options packet per minute. if you just have an rtp routing server or something similar you should have a look at ser / openser/opensip for handling these devices directly. What your are searching for is called Sip T1 Timeout and i´ve seen that in asterisk 1.6.1.x you can set this timeout in sip.conf. Iam not sure about changing this in other versions. If you're talking about t1min, AFAIK that only applies to monitored devices, i.e. those with qualify=yes. /Benny i am talking about t1max which is per rfc definition 64xt1min. Which is normally 32000 milliseconds. If you set this down to 15 seconds the timeout would be half than now, but could cause problems with very slow clients. The qualify options only takes affect on t1min when it set to yes. Then t1min would be set to the average qualify value. As i said i think qualify would be the right solution for you. I have a server running with more than 1600 peers, all with qualify on and notify traffic is around 200 pps in the night with no calls and aroung 6kpps (also with rtp traffic) on high load without taking any affect of the system. Our Ser server have a constant load of 600 pps but that is a proxy build for doing nothing else than routing sip packets. best regards steve ___ -- 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
