Displaying 6 results from an estimated 6 matches for "mccrari".
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mccrary
2004 Nov 22
2
Unknown number CID on SIP phone
Hello,
I'm a new Asterisk user and I hope I haven't missed something, but I
can't seem to find an answer to this issue. I have a Cisco SIP
gateway terminating calls into a 7960 phone. The issue I would like to
fix is if I have an incoming call without an ANI, such as directly from
my TDM phone switch, Asterisk says the call is coming from the IP
address of the Cisco gateway,
2005 Mar 21
2
G726-16 passthrough...
Hello,
I'm wondering if anyone has benn able to successfully get g726-16
passthrouhg to work? I am wanting to use this codec instead of g729 as
I'm running out of DSPs using a high complexity codec on the Ciscos. I
would think it would work just as g729 does, which has been working fine
for me, but it does not. G726-32 does work great however, but it's like
Asterisk doesn't
2005 Mar 14
2
qualify and NAT....
Hello,
I'm trying to run an ATA behind a NAT device, and am confused on exactly
what the qualify config option does, other than send NOTIFY packets.
Outbound calls work fine, but inbound calls do not go through. With
qualify=yes and nat=yes, my show sip peers looks like:
7771111001/7771111001 10.0.0.10 D N 255.255.255.255
1222 OK (36 ms)
So, it has established a
2005 Jan 24
0
Need some help with G729 passthru
I'm trying to get Asterisk to pass thru calls using the G729 codec.
I've got a 7960 phone and my gateway is an AS5400. I got the following
messages when debugging SIP (7778881000 is the 7960):
WARNING[1872]: channel.c:2115 ast_channel_make_compatible: No path to
translate from SIP/7778881000-2874(4) to SIP/as5400-35c1(256)
WARNING[1872]: app_dial.c:1002 dial_exec: Had to drop call
2005 Feb 04
0
Specify a codec in dial plan?
Hello,
Earlier I was having problems getting g729 pass-thru to work correctly.
I was able to fix this by putting allow=g729 as the first allow
statement directly under the configuration of the user. I also put
allow=ulaw in there below it. It seems the second "allow" statement is
totally ignored, as asterisk only negotiates codecs based on the first
statement.
It looks as if asterisk
2004 Nov 23
2
Re: Asterisk-Users Digest, Vol 4, Issue 300
Andrew Thompson wrote:
> You should be able to set the inbound callerid from the switch/gateway
> to a specific unknown in sip.conf file with just a callerid= line.
>
> The place I looked on the wiki didn't show a specific description for
> the callerid= line, but that's what I thought I read for it somewhere.
>
>