Hello: I have found the following problems with outgoing calls with asterisk, compiled with an updated CVS on 22 Oct. 1.- Problem with retries: Whenever I set the MaxRetries parameter, to something greater than 0 in a call-fille, Asterisk ignores the RetryTime parameter and retries every file in the outgoing folder when a new call-file is copied into that folder. So, if I make a call placing a call-file in the outgoing folder, nobody answers and my call-file stays in the outgoing folder waiting for 'RetryTime' seconds (say 120 seconds), if another call-file is placed in the outgoing folder the previous call is retried inmediately before the RetryTime finishes. 2.- Problem with high-volume calls: When I put lots of call-files in then outgoing folder, from time to time asterisk show an error in Master.csv. The error is OutgoingSpoolFailed, and shows no info about the call, just "failed" as the extension. I move one call-file every second, but for several minutes. How can I avoid this two problems? Thanks in advance, Robert T. _________________________________________________________________ Entra de visita en las decenas de tiendas del nuevo MSN Compras. Compara los precios antes de comprar. http://www.msn.es/compras/
Hi- As far as problem 2 is concerned, asterisk should have no problem with that call rate- one call per second, as long as the first call on a particular channel is complete before you initiate another on that same channel. I have had problems trying to initiate more than about 15 calls (on different channels) at exactly the same time, and find I have to stagger the starts a bit. I have a call generation script (very simple) to generate call load for testing, if that's what you're trying to accomplish. It's good for generating huge call volumes for IVR testing. Let me know if you need it! Scott M. Stingel Emerging Voice Technology Inc. Email: scott "at" evtmedia.com URL: www.evtmedia.com> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Asterisk List > Sent: Friday, December 26, 2003 11:58 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Problems with outgoing calls > > > Hello: > > I have found the following problems with outgoing calls with > asterisk, > compiled with an updated CVS on 22 Oct. > > 1.- Problem with retries: > Whenever I set the MaxRetries parameter, to something greater > than 0 in a > call-fille, Asterisk ignores the RetryTime parameter and > retries every file > in the outgoing folder when a new call-file is copied into > that folder. > So, if I make a call placing a call-file in the outgoing > folder, nobody > answers and my call-file stays in the outgoing folder waiting for > 'RetryTime' seconds (say 120 seconds), if another call-file > is placed in the > outgoing folder the previous call is retried inmediately before the > RetryTime finishes. > > 2.- Problem with high-volume calls: > When I put lots of call-files in then outgoing folder, from > time to time > asterisk show an error in Master.csv. The error is > OutgoingSpoolFailed, and > shows no info about the call, just "failed" as the extension. > I move one > call-file every second, but for several minutes. > > How can I avoid this two problems? > > Thanks in advance, > Robert T. > > _________________________________________________________________ > Entra de visita en las decenas de tiendas del nuevo MSN > Compras. Compara los > precios antes de comprar. http://www.msn.es/compras/ > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > >
Thanks Scott, but I already have a script that creates a call-load. That's how I have found the "OutgoingSpoolFailed" error and the other error with calls being retried without waiting to their retry time. Has anybody found this problem? Robert T. From: "Scott Stingel" <scott@evtmedia.com> Date: Fri, 26 Dec 2003 13:44:18 -0000>I have a call generation script (very simple) to generate >call load for testing, if that's what you're trying to accomplish. It's >good for generating huge call volumes for IVR testing. >Let me know if you need it!_________________________________________________________________ Reserva y planifica tu viaje online. http://www.msn.es/Viajes/
Hallo. I am living oin Germany and having two ISDN BRI Lines available. Capi driver! I need a Sip Gateway and a H 323 Gateway. About H.323, there should be a full implementation of H.450. Which software is available that gives me a Sip and a H.323 Gateway to enter my PSTN with a BRI (digital line)? Is there anywhere I can find a board? (Forum) Thanks in advance Bj?rn
> Hallo. > > I am living oin Germany and having two ISDN BRI Lines available. Capi > driver! > > I need a Sip Gateway and a H 323 Gateway. > About H.323, there should be a full implementation of H.450. > > Which software is available that gives me a Sip and a H.323 Gateway to > enter > my PSTN with a BRI (digital line)? > > > Is there anywhere I can find a board? (Forum) > > Thanks in advance > > > Bj?rn >Hello Bj?rn, You can find alot of information about Asterisk at: http://www.voip-info.org/tiki-index.php?page=Asterisk There are several folks on this email list that have implementations of Asterisk in Germany. Gru? aus Friedrichshafen, Robert
> >I have a call generation script (very simple) to generate > >call load for testing, if that's what you're trying to accomplish. > It's > >good for generating huge call volumes for IVR testing. > >Let me know if you need it!I would be interested in the script. If it is small, maybe you could post it to share on the list? Thanks, Kevin
Hi, Our setup is: Asterisk 1.0.7 running on Debian 2.4.27-2-386 TE110P card ISDN 30 (UK E1 PRI) When making outgoing calls to the PSTN using call files I get the following problems: 1. No hangup detection - have to wait for time-out 2. No pickup detection - the dial-plan starts as soon as the line rings All solutions I have read seem to be based on the assumption that a X100P card is being used with analogue lines. Any help on this would be much appreciated. Dr Roy Gardner Director www.psycle.com Tel: 01948 780120 Mob: 07713 985657 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060911/e389358b/attachment-0001.htm