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
members...
Any hints ?
Best regards,
Alexandre
Aldeia Digital
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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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
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
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] Re: Asterisk bandwidth tuning
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
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
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:
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
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Asterisk-Users mailing list
[EMAIL
: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=snip
callerid=9955
qualify=yes
This won't work with Firefly because qualify is broken in Firefly
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
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 digital [mailto:[EMAIL PROTECTED]
Sent: 21 October 2004
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.
alexandre::aldeia digital
: [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-Commercial Discussion
Subject: Re: [Asterisk-Users] Calling IAX client behind NAT
Hi,
But this is not my problem.
Asterisk have
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