search for: interband

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:

...select specifically the HF energy around blurred edges, in order not to have edges with aliasing.Is it possible to distinguish this HF energy from other ones: noise, little details,...? I think it is possible as the SPIHT algorithm tends to give such results.Does someone can explain me where in the interband dependencies (parent->child coeffs trees) do this HF energy is taken into account? The additional but secondary problem that will arise, is that if I manage to define 2 or 4 entropy words for HF energy for blurred edges, then it will unbalance my current Huffman tree, so I will have to write an...

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 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

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

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

An example for Radius is calling cards.. I can use * for this kind of service... With platforms that use Radius Server. -----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 07:16 p.m. Para: asterisk-users@lists.digium.com Asunto:

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