Can someone tell me why qualify=yes is required in sip.conf, before you can
detect the dialled status of a channel? If I don't have qualify=yes against
a peer, than checking ${DIALSTATUS} has no effect. Asterisk will just keep
trying until the dial timeout expires.
Why can't asterisk actually LOOK at the SIP response returned by the SIP
proxy and do something with it? Why can't it detect a failure to connect (ie
proxy down) and do something based on that? Do the Asterisk developers realise
how damn hard this makes it to build any kind of reliability into Asterisk????
Doug
-----Original Message-----
From: Warren Burstein [mailto:warren@softov.co.il]
Sent: Sunday, December 11, 2005 11:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] outgoing calls that last an unreasonably
longtime
Simone Cittadini wrote:
> Warren Burstein ha scritto:
>
>>
>>
>> What is frustrating is that the cdr file shows the dst as T rather
>> than as the phone number dialed. I realize that AbsoluteTimout
>> causes it to jump to the T extension, but it would help to know who
>> the user dialed (asking a week later isn't going to get any useful
>> information out of the user). It's not in the log file, either -
>> would increasing the log level help here?
>>
>>
> I don't know how this AbsolutTimeout works, anyway I put all the info
> I need in variables before the actual Dial, then in the h extension I
> call SetUserField() (or whatever is called), helps me keeping track of
> reasons for non-terminated calls ...
I am not using the userfield for anything so that sounds like a good
idea. It's SetCDRUserField by the way.
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Ok, what's the deal with qualify in sip.conf. The docs on the voip wiki at: http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+qualify state that it can take either yes, no, of a number which represents how long in milliseconds between polling. I set it to 1000, (ie qualify=1000), did a reload, and it's obviously polling at 60 seconds, not 1 seconds, as evidenced from an ngrep trace. I'm guessing the wiki is wrong, because it also says the default polling period is 2s, not 60s. Can someone update that? Can ANYONE fix those pages on the voip wiki? Doug.
In that case, if I set qualify=1000, and it still polls every 60s, then how can it consider it unreachable at 1000ms? Doug. -----Original Message----- From: Joshua Colp [mailto:jcolp@digium.com] Sent: Mon 8/14/2006 1:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] SIP Qualify ----- Original Message ----- From: Douglas Garstang [mailto:dgarstang@oneeighty.com] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-users@lists.digium.com] Sent: Mon, 14 Aug 2006 20:24:35 -0300 Subject: [asterisk-users] SIP Qualify > Ok, what's the deal with qualify in sip.conf. The docs on the voip wiki at: > > http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+qualify > > state that it can take either yes, no, of a number which represents how long > in milliseconds between polling. I set it to 1000, (ie qualify=1000), did a > reload, and it's obviously polling at 60 seconds, not 1 seconds, as > evidenced from an ngrep trace. The value that qualify takes is the maximum time to accept before considering the device unreachable. If I set qualify to 200ms, and my device's qualify time is 250ms then the device will be considered unreachable. > I'm guessing the wiki is wrong, because it also says the default polling > period is 2s, not 60s. > > Can someone update that? Can ANYONE fix those pages on the voip wiki? Yes, you can. voip-info.org is a wiki that anyone can update/change/modify. > Doug. Joshua Colp Digium _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Had to switch to top posting..... changed mail clients.... :) Ok, I follow. Thanks. Doug. -----Original Message----- From: Jason Parker [mailto:jparker@digium.com] Sent: Mon 8/14/2006 7:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] SIP Qualify If you're gonna top post...so am I. I think you misunderstand what qualify is/does. It appears that you believe that qualify=1000 means that it'll send out a qualify packet every 1000ms. This isn't an unreasonable assumption, but it is wrong. The qualify=1000 means that Asterisk will wait 1000ms for the device to respond to the qualify packet. If after 1000ms there is no "yes, I'm here" packet, then it will be considered UNREACHABLE. Qualify packets are sent out at a set interval, which, as you can see, is 60 seconds. If the device was previously determined to be UNREACHABLE, the qualify packets will then be sent out every 10 seconds instead. ----- Original Message ----- From: Douglas Garstang <dgarstang@oneeighty.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Sent: Monday, August 14, 2006 6:42:26 PM GMT-0800 Subject: RE: [asterisk-users] SIP Qualify In that case, if I set qualify=1000, and it still polls every 60s, then how can it consider it unreachable at 1000ms? Doug. -----Original Message----- From: Joshua Colp [mailto:jcolp@digium.com] Sent: Mon 8/14/2006 1:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] SIP Qualify ----- Original Message ----- From: Douglas Garstang [mailto:dgarstang@oneeighty.com] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-users@lists.digium.com] Sent: Mon, 14 Aug 2006 20:24:35 -0300 Subject: [asterisk-users] SIP Qualify > Ok, what's the deal with qualify in sip.conf. The docs on the voip wiki at: > > http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+qualify > > state that it can take either yes, no, of a number which represents how long > in milliseconds between polling. I set it to 1000, (ie qualify=1000), did a > reload, and it's obviously polling at 60 seconds, not 1 seconds, as > evidenced from an ngrep trace. The value that qualify takes is the maximum time to accept before considering the device unreachable. If I set qualify to 200ms, and my device's qualify time is 250ms then the device will be considered unreachable. > I'm guessing the wiki is wrong, because it also says the default polling > period is 2s, not 60s. > > Can someone update that? Can ANYONE fix those pages on the voip wiki? Yes, you can. voip-info.org is a wiki that anyone can update/change/modify. > Doug. Joshua Colp Digium _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Yes, it might be a problem in our situation. We have three Asterisk boxes in a 'cluster'. The sip.conf is identical on all three. In that case, all three of the Asterisk boxes in our cluster are going to send sip options messages to the phones, which is silly. Only the Asterisk box that a phone is registered on needs to send the sip notify messages. The rest are a waste. I'm not sure how we'd work around this. We may just have to make do with the caller of an unavailable phone getting ringback until the dial timeout occurs. Doug. -----Original Message----- From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com] Sent: Mon 8/14/2006 9:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [asterisk-users] SIP Qualify Qualify does what the name implies "qualifies the connection' It pools every 60s but it calculates he time it took for the packet to reach the end device. If the endpoint has a latentcy > than the qualify parameter, * considers the endpoint unreachable. This does not however address the point you made in another post about RINGING before the INVITE. It is still possible to have a phone go dead in the 60sec between qualify re-checks. There are several post in history about qualify and it sending LARGE amounts of traffic to endpoints. I think it was John Todd that was the OP on the subject IIRC. SNIP _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Only the Asterisk box that a phone is registered on WILL send the sip notify messages. The others will have no idea where to send them, and will not do so. - Brad -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Douglas Garstang Sent: Tuesday, August 15, 2006 12:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] SIP Qualify Yes, it might be a problem in our situation. We have three Asterisk boxes in a 'cluster'. The sip.conf is identical on all three. In that case, all three of the Asterisk boxes in our cluster are going to send sip options messages to the phones, which is silly. Only the Asterisk box that a phone is registered on needs to send the sip notify messages. The rest are a waste. I'm not sure how we'd work around this. We may just have to make do with the caller of an unavailable phone getting ringback until the dial timeout occurs. Doug. -----Original Message----- From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com] Sent: Mon 8/14/2006 9:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [asterisk-users] SIP Qualify Qualify does what the name implies "qualifies the connection' It pools every 60s but it calculates he time it took for the packet to reach the end device. If the endpoint has a latentcy > than the qualify parameter, * considers the endpoint unreachable. This does not however address the point you made in another post about RINGING before the INVITE. It is still possible to have a phone go dead in the 60sec between qualify re-checks. There are several post in history about qualify and it sending LARGE amounts of traffic to endpoints. I think it was John Todd that was the OP on the subject IIRC. SNIP _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.