[asterisk-users] Persistentmembers (Not working with restart)
Hello All, I currently have an Asterisk Box, running a callcenter with 04 queues. I set queues.conf with "persistentmembers=yes" in the general section as follows: [general] monitor-type = MixMonitor persistentmembers = yes However when I perform any kind of restart in the Asterisk application, all agents are considered unavailable after that. Though when performing reload, agents keep their status as it was before the reload. Is there any where else that I should set dynamic agents as persistent members to keep their status after a asterisk restart?? Thanks, Marco ___ -- 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
[asterisk-users] RES: DSS1 vs SS7
Hello All, I have an Asterisk Box currently running 1.6.0, dahdi drivers and libss7 with a TE120P (01 E1) card for the past 02 months. I have it connected to a Cellular Operator switch (MSC), and it is working perfectly. Traffic is still quite low, but increasing as we start to use it for new applications everyday. I have made some stress call tests, using all available CICs at once, and had no problem at all. Congrats to the perfect development of the SS7 support to Mr. Fredrickson. Hopefully soon we'll have MAP support as well. Marco -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Matthew Fredrickson Enviada em: sexta-feira, 22 de agosto de 2008 11:49 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] DSS1 vs SS7 Kevin P. Fleming wrote: > Alex Balashov wrote: > >> Some carriers now do offer private SS7 instead of ISDN. But there is >> absolutely no reason why you should be doing this with Asterisk. >> Asterisk-SS7 is quite tenuous at best. Unless you have some specific >> reason to be using it, don't. > > Actually, SS7 support in Asterisk 1.6.0 appears to be quite solid, and > it is being used in a quite a number of production deployments. Thanks for the plug Kevin! :-) Yeah, actually, if you guys want to know more there's an asterisk-ss7 mailing list. Asterisk-1.6.0 with libss7 is being used in many successful and high traffic installations around the world. The current record (that I have been told of) is an installation doing over 100,000 calls per day. So try to beat that ;-) Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RES: GotoIftime
Hello Nhadie, I had a very similar situation. My solution, even tough might not look very wise, solved my problem the way I needed. I repeated the GotoIftime command in the next line in my extensions.conf . Like this: GotoIfTime(22:00-23:59|*|30|jul?test,s,1) GotoIfTime(00:00-02:00|*|31|jul?test,s,1) Rgs, Marco Cordeiro -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Nhadie Enviada em: quarta-feira, 30 de julho de 2008 16:47 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: [asterisk-users] GotoIftime Hi How cn i define in GotoIfTime from day 1 extending to day 2? e.g July 30 2200 up to July 31 0200 I'm thinking like this: GotoIfTime(22:00-02:00|*|30-31|jul?test,s,1) but i think that means July 30 22:00-2:00 and July 31 22:00-2:00 which looks like an invalid entry for the time. is it possible in asterisk? regards, nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RES: GXW 4108 asterisk configuration
I have an Asterisk running with both GXW4008 (FXS) and GXW4108 (FXO). The FXS Gateway works perfectly, no problem so far. The FXO Gateway (GXW4108) also works fine. The configuration for local settings in Brazil was quite easy, however, I still not able to make Caller ID to work. I'm setting as DTMF Caller ID type, but still not working. Let us know what kind of problem you have, maybe I can help you out. Marco -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Doug Enviada em: quarta-feira, 18 de junho de 2008 15:57 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] GXW 4108 asterisk configuration What EXACTLY is the problem? There is a bug in these units that won't let you put punctuation in the extension name. Also I am having problems with a unit that disrupts the network. Still haven't found out what exactly is going on. This is with a GXW4024. At 13:19 6/18/2008, Jorge Valdes, wrote: >-BEGIN PGP SIGNED MESSAGE- >Hash: SHA1 > >Nelson Granados wrote: >> GXW 4108 asterisk configuration >> >> Dear, >> >> I'm having problems with the configuration of this >> gateway(GrandStream GXW 4108), I used the instructions from >> GrandStream but it doesn’t work. Someone has a good configuration >> for this gateway? >> >> >> Thanks in advance, >> >> >> Nelson >> >> >> -- >> >> ___ >> -- 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 >Yeah... it took me some digging but finally got it working by >following the instructions provided by Grandstream. What is it exactly >the problem? > >- -- >Jorge Valdes > >-BEGIN PGP SIGNATURE- >Version: GnuPG v1.4.6 (GNU/Linux) >Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > >iD8DBQFIWVGXvseWMACI1MYRAnZEAJ9saY6ogO8eEwmYqVCThwp0ODjGpACZAaoE >ZZsys6XMvUGShDHmuESS4Mk= >=en2Y >-END PGP SIGNATURE- > > >___ >-- 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 ___ -- 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
