Re: [asterisk-users] CALLERPRES() with Queue

Hi, Bump to see if anyone can help us too. Really this is a problem. I don't want to show the caller id number and name to the Agent in certain conditions. Changing the CID will mess the CDR/Queue log and this is not the acceptable behavior. In the Dial app, everything is OK. Alexandre Em 06-

[asterisk-users] Set Queue to ring in only one member

members... Any hints ? Best regards, Alexandre Aldeia Digital -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

[asterisk-users] Call-limit and internal transfer

Hi, I set the sip.conf parameter call-limit=1 to limit outbound calls and 'disable' call waiting. But internally, I want to enable transfers. If the call-limit=1, the transfers fails. Any help ? Thanks all, Alexandre ___ --Bandwidth and Colocation pr

Re: [Asterisk-Users] Re: Asterisk bandwidth tuning?

from mi experience, at 16kbps the audio becomes so distored that is very difficult to understand the other party. --- [EMAIL PROTECTED] wrote: Message: 1 Date: Wed, 19 Jan 2005 15:05:53 -0200 From: "alexandre::aldeia digital" <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] R

Re: [Asterisk-Users] Re: Asterisk bandwidth tuning?

18 - 22 Kbps my dream! I have asterisk -> INTERNET -> asterisk connection with IAX2 and I try iLBC, gsm, g729 and speex and the minimun bandwidth was 38 Kbps for 1 channel. What the parameters do you set to have this rate ??? Thank you. Miguel Ruiz Velasco Sobrino wrote: I have an installat

Re: [Asterisk-Users] internet bandwidth (comparing overhead)

Hi, I like to know why iLBC and GSM generate a 40-50kbps bandwidth Is very high, if compared with your calculations for other codecs(G723.1 / 17kbps and G729 / 24 Kbps). Alexandre Kanuri, Seshu (Company IT) wrote: /SNIP/ Some corrections are needed: 6.3kbps of G723.1 will become around 17kbps

Re: [Asterisk-Users] X100P & Analog PBX - not RING and not answer (SOLVED!)

Hi again, I solve my problem... Puting some debugs in wcfxo a discover that MINPEGTIME is too high to capture the rings from PBX. Reducing your value, everything is functions perfectly. Alexandre alexandre::aldeia digital wrote: Hi, I connect a X100p in a Analog PBX extension. If I want to call

[Asterisk-Users] X100P & Analog PBX - not RING and not answer

Hi, I connect a X100p in a Analog PBX extension. If I want to call a analog extension (e.g.: using a softphone), the asterisk pick up the extension and dial perfectly. If I call the extension where where the X100Pp is connected (inside the company), the asterisk doesn't answer the call. I do: mo

Re: [Asterisk-Users] Firefly 1.9.6 released (thanks)

Hi Tim, - responds to POKE messages; qualify=yes should now work in iax.conf Very, very thanks to implement this !!! I finally solved my problem for calling a registered client behind NAT... Alexandre ___ Asterisk-Users mailing list [EMAIL PROTE

Re: [Asterisk-Users] Calling IAX client behind NAT

:07 -0200, alexandre::aldeia digital <[EMAIL PROTECTED]> wrote: I try to explain again: Client(Firefly) --> NAT --> INTERNET --> * (public IP) --> SIPURA iax.conf: [teste] type=friend host=dynamic secret= callerid="9955" qualify=yes This won't work with Fir

Re: [Asterisk-Users] Calling IAX client behind NAT

al host=dynamic notransfer=yes trunk=no canreinvite=no callerid="Test User" <3006> -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of alexandre::aldeia digital Sent: Thursday, 21 October 2004 11:23 PM To: Asterisk Users Mailing List - Non-C

Re: [Asterisk-Users] Calling IAX client behind NAT

ge- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Thursday, 21 October 2004 10:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Calling IAX client behind NAT Tell the client to register every 60 seconds. alexa

Re: [Asterisk-Users] Calling IAX client behind NAT

ng IAX client behind NAT Try port forwarding 4569 on the client side. All the claims of "IAX doesn't require prt forwarding" on the wiki site confused the hell out of me as I couldn't get incoming calls working. Cheers alex -----Original Message- From: alexandre::aldeia dig

[Asterisk-Users] Calling IAX client behind NAT

Hi, Sorry if this is a stupid question. Can I call a dynamic registered IAX client localized behind a NAT ? Client(Firefly,DIAX) --> NAT --> INTERNET --> * (public IP) --> SPA-2000 (192.168...) The client can make calls to asterisk normally. But I can't call the client(lol...:). With a tunnel (