Im getting these messages when making calls from a sip extension/other asterisk
peer out my pri.
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/1-1]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/1-1]
It happens whenever i send dtmf. I have all of my devices set to inband, be it
All,
Im having a problem with ReceiveFax where its generating a ton of these
messages the entire time the receivefax app is running receiving my fax.
[Apr 7 22:16:06] ERROR[26918]: channel.c:2520 __ast_read: ast_read() called
with no recorded file descriptor.
Im running on Centos 5.2 with al
> Message: 13
> Date: Tue, 24 Feb 2009 15:13:41 -0500
> From: Mike
> Subject: Re: [asterisk-users] Aastra phones
> To: asterisk-ad...@hulber.com, 'Asterisk Users Mailing List -
> Non-Commercial Discussion'
> Message-ID: <001501c996bc$63001e60$29005b...@ca>
> Content-Type: text/plain; char
>
> Hi,
>
>
>
> I`ve been toying with an Aastra phone (9143i) wondering if it could be a
> good alternative to to the more expensive Polycom phones.
>
>
>
> One thing which I can't figure out, although it certainly looks simple, is
> to update the firmware though FTP (not TFTP). I have s
I don't see why you couldn't use asterisk in a setup that large. It would
require a number of servers, and SER to handle the registrations, and call
routing and use asterisk for what its good at, ivr/vm.
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Are you not sending the exten to iaxmodem like this:
exten => xx,1,dial(IAX2/xxx/${EXTEN},30,r)
I had the same problem, routing wouldnt work, until i passed it the did like
above.
> Message: 3
> Date: Mon, 18 Feb 2008 15:26:31 +0200
> From: Louwrens Benad? <[EMAIL PROTECTED]>
> Subjec
I gave up on the rxfax business as it never worked for me. I use iaxmodem and
hylafax and it works perfectly, every single time i use it. inbound or outbound
doesnt matter.
I have not read about anyone using iaxmodem and hylafax having any issues. and
its fairly easy to setup. Took me about 1 h
I have a site running asterisk 1.2.8 with a hand full of polycoms and
grandstream 2Kxp's. When a call is missed and you look at the missed call logs
on either, its has the persons exten, not the incoming caller id. Any ideas?
\\\|///\\ ~ ~ // ( @ @ )--oOOo-(_)-oOOo—_
I have been looking at the rhino r1t1, and digium single t1, and the sangoma, but from what i read they all sound like good products. Anyone have anything bad about any one of them? I am leaning towards the sangoma as it seems to have a better following here on the lists. Any advice is appreciate
I use the following in rc.local for setting tos bits using iptables:iptables -A POSTROUTING -t mangle -p udp -m udp --sport 1:2 -j DSCP --set-dscp 0x2eWorks like a champ!> 1. RE: Setting QOS settings in asterisk and/or CentOS?> (Redouane Doumer)> De : BerkHolz, Steven [mailto:[EM
It is totally nat, first try to port map the ports through your firewall, on the network page set the rtp and sip ports plus the nat ip to use. I had the exact same problem and this was the only solution.
Or add the following to your config for the phone:
nat.mediaPortStart="5004" nat.signalPor
All,
I have this issue happening on 2 seperate asterisk boxes, it happen from version 1.2.4 i am currently running version 1.2.7.
What happens is i will be on a call, and all of a sudden I will hear a fast busy, the person that i was talking to can still hear me fine. It doesn't really matte
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