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hello all, does anyone has experiencie using asterisk with a digium CARD using G729 managing 120 concurrent calls with SIP and/or H323??? I wanna know if Asterisk is stable doing this....because we wanna implement it in some locations!! Thanks All!! Sebastian. -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ide) > 5. RE: STILL NO AUDIO (Eric Wieling) > 6. Re: STILL NO AUDIO (Holger Schurig) > 7. RE: Mac OS X installer for Asterisk (Wallingford, Ted) > 8. Re: PhoneGaim? (creslin@digium.com) > 9. Re: BroadVoice problems? (Chris Shaw) > 10. RE: STILL NO AUDIO (Sebastian Nocetti) > 11. Re: TDM400P Internal Extenion Config (Jason Williams) > 12. IP Phone recommendation (Yiannis Costopoulos) > 13. Re: Cheap PoE switches/injectors? (asteriskstuff@ziplip.com) > 14. RE: STILL NO AUDIO (Sebastian Nocetti) > > --__--__-- > > Message: 1 > Date: M...

is h323 per user based working??? I have setup this: [User1] type=user host=xx.xx.xx.xx context=international incominglimit=30 But all calls from xx.xx.xx.xx are not routed to context international, it is working????? I am using chan_h323 Thanks!! Sebastian Nocetti. --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 2004-11-30 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041221/b8799c1c/attach...

hello all, I am having a trouble with Audio using h.323 channel... I am doing this Call comes into cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and send call to a SoftSwitch that routes the call, I can see log debug telling me, CALLED XXX, and then RINGING, and I can hear ring tones... but when call is answered, I DONT HEAR ANYTHING... I am using lastest ASTERISK

Does Asterisk support Radius accounting?.... -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Lunes, 17 de Noviembre de 2003 12:08 p.m. Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users digest, Vol 1 #1912 - 11 msgs Send Asterisk-Users mailing list

how many calls can be handled with Asterisk Calling Card app? somebody knows that? --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.744 / Virus Database: 496 - Release Date: 2004-08-24 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041004/498fd5f8/attachment.htm

Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip: 64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome message and resend call to Cisco 3600 that have 4 analog lines connected... but after cisco play welcome message and when

Hi, I'm planning to use a Asterisk with Digium E1 cards, I understand that using a codec such as G.729 can be very CPU demanding. What are the real advantages of using a codec such as G.729 ? Bandwidth only ? Using no compression wouldn't increase the scalability of my asterisk PBX ? This is considering I have no bandwidth issues in my network. Thanks

Hi, does anybody manage to get music-on-hold with inaccess oh323 driver? Statement like : exten => 10,1,Dial(OH323/xx,mt) works (dials the xx number) but no music is heared. Also, if I put 'r' (ringback) it doesn't work either. With chan_h323 I got this functionality but this driver had some other problems (call transfer don't work).... Thanx in advance, Victor...

...m.com http://lists.digium.com/mailman/listinfo/asterisk-users --__--__-- Message: 2 Date: Mon, 17 Nov 2003 16:33:10 -0500 From: Jeremy McNamara <jj@nufone.net> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Radius on * Reply-To: asterisk-users@lists.digium.com Sebastian Nocetti wrote: >Does Asterisk support Radius accounting?.... > > No and there is absolutely no need for it to. RADIUS is not anything that should have ever been deployed in a VoIP environment. There are many methods to talk directly to a database, why add another layer of complexity and...

Hi, I want to use G.723.1 on *, I read it is supported in Pass Through mode, but I don't understand whats the meaning of that. I have a GW 5300 and an ATA 186 and I want to place calls to PSTN. I setup this config: [general] port = 5060 bindaddr = xx.xx.xx.xx context = sip tos=throughput maxexpirey=360 defaultexpirey=120 [gw5300] type=friend insecure=yes host=xx.xx.xx.xx

I tried a lot of times to get it worked, but I cant obtain audio using SIP<->chan_h323 or chan_h323<->SIP I tried disbling FastStart without good results... What's the problem? I need to do BRIDGE between SIP and H.323!! help!! Sebastian.- -------------- next part -------------- An HTML attachment was scrubbed... URL: