Unable to make outgoing calls with Internode

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Surely there must be someone here who can help me with this problem. I have spent weeks trying to get this damned service to work with no
luck. I have incoming calls working, but no outgoing. If get outgoing
working then incoming don't work. I have sent this problem to this list a couple of times with little or
no response, and I _really_ need some help to sort it out. I have an asterisk 1.8 server running on FreeBSD 8.1, and another
FreeBSD 8.1 running as a firewall/gateway with PF. I have a nodephone service…

Asterisk Users 4.6 years ago 3 Answers

AEL Eswitches

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On Wednesday 09 February 2011 13:31:36 Thiago Maluf wrote:
> Hi List,
>
> Would someone can to explain me the main difference in SWITCHES or
> ESWITCHES in AEL.
>
> context default {
> switches {
> DUNDi/e164;
> IAX2/box5;
> };
> eswitches {
> IAX2/context@${CURSERVER};
> };
> }; A switch evaluates variables at load time. An eswitch evaluates variables
locally at query time. An lswitch sends the variables intact through to
the switch backend.

Asterisk Users 4.6 years ago 0 Answers

Error loading module ��Է�Vi.so

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Just recently I noticed that my Asterisk 1.8 server is giving the
following error at startup:
[Feb 9 17:48:56] WARNING[7968]: loader.c:387 load_dynamic_module: Error
loading module '��Է�Vi': /usr/lib/asterisk/modules/��Է�Vi.so: cannot
open shared object file: No such file or directory I have checked the modules directory and there are no files with
strange characters that I can see. The only extra modules I have are
the G729 codec and Lumenvox speech recognition. Both are loaded and
working. All expected functionality for Asterisk is working so I really
do not know what that module may…

Asterisk Users 4.6 years ago 0 Answers

Defining what an extension should do after the Dial() command returns busy.

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We have a customer who wants to forward an extension to their cell phone,
if and only if that extension is "unavailable", or when the Dial() command
times out. However, should the Dial() command return "busy" it should go
to voicemail instead. As far as I know, the dialplan doesn't support this. Certainly not
natively or in any particularly easy or obvious way, and I can't find
anything on voip-info.org to suggest that there is.

Asterisk Users 4.6 years ago 3 Answers

SIP MESSAGE outside calls - state of the art?

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I have a mobile phone (UTStarCom GF-210) that uses SIP MESSAGE to send
"SMS" messages over VoIP. My Asterisk 1.4 installation drops these
messages and returns a failure condition to the phone: [Feb 9 10:17:22] WARNING[11960]: chan_sip.c:9859 receive_message: Received message to from "Display Name" ;tag=87739132, dropped it...
Content-Type:text/plain; charset=UTF-8
Message: test message (Packet trace shows a SIP MESSAGE, answered by a 405.) ...and apparently is unable to originate them either; SendText, which
looks as though it ought to be the right way to send them, produces (in
the context of a call, since…

Asterisk Users 4.6 years ago 3 Answers

Reliably getting sip extension name from channel variables

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Hi We're using asterisk 1.4.17 debian package soon moving to 1.8 rpm
package. When using MixMonitor to do call recordings, for outbound calls I have
been using the channel variable SIPURI to get the originating SIP
extension name. I have now stumbled across a few files where the SIP
extension name must be incorrect when cross referencing the call with
other sources (such as the channel shown in the cdr). So, a couple of questions I'm throwing out there:
Why would the Channel variable have a different SIP extension as part of
it's…

Asterisk Users 4.6 years ago 0 Answers

dial option 'g' not working

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Hi, I'm trying to get my dialplan to continue executing in the current context
after a third-party is called and hangs up. It seems like it should be
straightforward but it's not working. Here's what I have in extensions.conf: exten => 333,1,Answer()
exten => 333,n,Playback(hello)
exten => 333,n,Dial(SIP/19992223333@sipcarrier,,g)
exten => 333,n,Playback(hello)
exten => 333,n,Playback(hello)
exten => 333,n,Playback(hello)
exten => 333,n,Hangup() The 9992223333 number is dialed, but after that party hangs up, there's just
dead air. No hello's are played and nothing seems to be happening. What am I doing wrong? Thanks,

Asterisk Users 4.6 years ago 0 Answers