Hi all, This issue is giving me a lot of grief with my customers. I have 5 asterisk servers running in production, each one with almost 70 simultaneous calls at peak hour. Most of my customers complain that their calls drop after 20 seconds or so. After running through my cdr's, I see that the number of 20 second calls is MUCH larger than any other number. (see below) billsec count(*) 1 924 2 841 3 725 4 812 5 779 6 681 7 644 8 630 9 613 10 515 11 522 12 516 13 557 14 527 15 507 16 457 17 456 18 467 19 424 20 1644 21 365 22 382 23 353 24 382 25 379 26 370 27 350 28 337 29 302 30 291>30 lotsI am running Asterisk 1.6.1.14 on Debian Lenny. My servers are Dell Power Edge 1950. I use canreinvite=no, and all of my servers are on public IP's. All of my customers use Grandstream GXW4004 telephony adapters. It is a hard issue to debug because it does not happen always. I will try to obtain a sip trace from a dropped call, but until then, any pointers, opinions or even guesses would be much appreciated!! Thanks in advance, Alex
At 20:07 4/19/2010, Alejandro Recarey wrote: >Hi all, > >This issue is giving me a lot of grief with my customers. I have 5 >asterisk servers running in production, each one with almost 70 >simultaneous calls at peak hour. Most of my customers complain that >their calls drop after 20 seconds or so. Many people have had this type of problem: http://www.Google.com/#q=Asterisk+20+seconds > >After running through my cdr's, I see that the number of 20 second >calls is MUCH larger than any other number. (see below) > >billsec count(*) >1 924 >2 841 >3 725 >4 812 >5 779 >6 681 >7 644 >8 630 >9 613 >10 515 >11 522 >12 516 >13 557 >14 527 >15 507 >16 457 >17 456 >18 467 >19 424 >20 1644 >21 365 >22 382 >23 353 >24 382 >25 379 >26 370 >27 350 >28 337 >29 302 >30 291 >>30 lots > > >I am running Asterisk 1.6.1.14 on Debian Lenny. My servers are Dell >Power Edge 1950. I use canreinvite=no, and all of my servers are on >public IP's. All of my customers use Grandstream GXW4004 telephony >adapters. > >It is a hard issue to debug because it does not happen always. I will >try to obtain a sip trace from a dropped call, but until then, any >pointers, opinions or even guesses would be much appreciated!! > >Thanks in advance, > >Alex > >-- >_____________________________________________________________________ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- >New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
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Doug, thanks for the help, already looked it up, but it does not seem to be a NAT issue (which is what most posters suggest when googling) Danny, those are billsec durations, the call has been established and media is being passed for 20 seconds. Thanks again! Alex
--- On Mon, 4/19/10, Alejandro Recarey <alexrecarey at gmail.com> wrote:> their calls drop after 20 seconds or so. > All of my customers use Grandstream GXW4004 > telephony > adapters.Check out the "early dial" feature in the Grandstream products (if you enabled it) and play with the "pedantic" option. You might want to take a look at this: https://issues.asterisk.org/view.php?id=14652
Alejandro Recarey schrieb:> Doug, thanks for the help, already looked it up, but it does not seem > to be a NAT issue (which is what most posters suggest when googling) > > Danny, those are billsec durations, the call has been established and > media is being passed for 20 seconds. > > Thanks again! > > Alex > >Hi, How do you dial the users? direct with the peername or something like exten at ipofpeer ? i know this problem when dialing a patton ISDN ata without an extension. The call is established but when the T1 sip timeout fires the call gets disconnected. Maybe you could do some sip debugging and watch for resend sip messages. best regards steve -- F?r weitere Fragen stehen wir gerne unter voip at sil.at oder 059944 - 2440 zur Verf?gung. Mit freundlichen Gr?ssen -- Stefan Schmidt Sysadmin/VOIP // voip at sil.at // Tel 059944-2440// ------------------------------------------------- SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // -------------------------------------------------
Like the poster below said, do a sip debug on a call and see which end sends the bye message or ends the call and go from there. That should give you some sort of clue as to who is having a timer issue. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Stefan Schmidt Sent: Wednesday, April 21, 2010 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls drop after 20 seconds Alejandro Recarey schrieb:> Doug, thanks for the help, already looked it up, but it does not seem > to be a NAT issue (which is what most posters suggest when googling) > > Danny, those are billsec durations, the call has been established and > media is being passed for 20 seconds. > > Thanks again! > > Alex > >Hi, How do you dial the users? direct with the peername or something like exten at ipofpeer ? i know this problem when dialing a patton ISDN ata without an extension. The call is established but when the T1 sip timeout fires the call gets disconnected. Maybe you could do some sip debugging and watch for resend sip messages. best regards steve -- F?r weitere Fragen stehen wir gerne unter voip at sil.at oder 059944 - 2440 zur Verf?gung. Mit freundlichen Gr?ssen -- Stefan Schmidt Sysadmin/VOIP // voip at sil.at // Tel 059944-2440// ------------------------------------------------- SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // ------------------------------------------------- -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users