Problem with SendDTMF

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Hello, I am having a problem with SendDTMF - it is not sending the numbers
properly during the phone call.. I want the numbers key to to be
pressed/sent automatically after 3 seconds during a phone call. I use software phone to test it... when I dialed 501, I cant hear anything
for about 10 seconds (this is because of SendDTMF) and then I can hear
the operator saying to enter the numbers but SendDTMF already did it?! Asterisk server are connected to voip.ms provider. I have spent many hours trying to get to work, how…

Asterisk Users 3.4 years ago 1 Answer

Grandstream 1.0.3.30 BETA Firmware

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If you are using Grandstream GXP 21XXX and 14XX phones and you are doing
any kind of remote firmware updates or config updates DO NOT use the
1.0.3.30 BETA version. We have found a bug in it that causes HTTP updates
to not work if you are using a domain name and not an IP address for
pulling configs and firmware. It can put you in a state where you can't
update the configs or firmware without direct web or telnet contact to the
phone. Thanks Bryant

Asterisk Users 3.4 years ago 0 Answers

Asterisk ACL

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Hi We are trying to accept inbound calls from a SIP provider who sends us calls from various IP's within a given subnet but they are failing every time with the following message on the console. chan_sip.c:20006 handle_request_invite: Call from '' to extension '' rejected because extension not found Our understanding is that the deny line blocks every IP and the following permit line then allows calls from the specified subnet but it seems that the peer is never matched when a calls hits the server.
It's almost as if there should be a setting somewhere that we…

Asterisk Users 3.5 years ago 7 Answers

in sip.conf or dialplan or db?

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On Tuesday 27 March 2012, Roland wrote:
> I am setting up my dialplan with quite some outbound numbers. We have a
> block of 100 DID's, for which some of them will go direct to specific
> phones. I am struggling how to solve this, so I am searching for a little
> advice. These are my concerns.
>
> I could set the DID in the sip.conf using something like:
>
> callerid="137-Roland" <31229253137>
>
> 137 would be my extention number here.
> .....
>…

Asterisk Users 3.5 years ago 1 Answer

Park and PARKINGDYNAMIC

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I have been trying to get the dynamic parking working. For some reason when I park a call using this method the console says it is
using the default parking context not the one I am trying to specidfy. It
also is playing the parked extension to the caller. I am transfering the
call to an extension that is doing a goto to the context below. Any ideas
or examples on how to make this work. What I need to be able to do is have
multiple parking lots using the same extension pools but…

Asterisk Users 3.6 years ago 2 Answers

Sporadic one way audio problem

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Hi all again, I've got a problem with sporadic one way audio calls, which means
sometimes I can't hear the calling party (call is established, but audio
is missing). Today I received ~90 calls, one of them got this problem. I've got two networks involved, without NAT: - 192.168.1.X, in there one nic of my server and all the phones
- a private net to my provider, in there a nic of my server and the voice
switch of my provider My sip.conf:
[general]
port=5060
bindaddr=0.0.0.0
language=de
allowguest=no
;echocancel=yes

Asterisk Users 3.7 years ago 1 Answer

MWI for non-subscribed Realtime peers?

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Hi, I have an Asterisk behind an OpenSIPS proxy. The proxy handles registrations and also SIP SUBSCRIBE for MWI. The Asterisk are configured to send NOTIFY to the proxy even when the SUBSCRIBE haven't been received. I can configure a user in sip.conf that works: [az5134939706]
type=friend
host=xxx.xxx.xxx.xxx (IP of proxy)
port=5060
nat=no
mailbox=1234@customer
subscribemwi=no
defaultuser=az5134939706
Every time a voicemail has been left in the mailbox 1234@customer, a NOTIFY is sent off to the proxy. Remember, the peer doesn't register or send SUBSCRIBE to Asterisk, but "subscribemwi=no" forces NOTIFY to be…

Asterisk Users 3.8 years ago 1 Answer

2 same sip extension number on 2 asterisk - call not passing on certain condition

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Hi list, something crazy here. 2 asterisk on 2 different place (1.4 and 1.8) both
having an extension [115], one as type peer (caller side 1.4) and one as
friend (callee side 1.8). Phones from both location connect to Asterisk
from LAN. Router are Linux boxes. Connection between the 2 sites is done like this: On the callee side [115] ;callee
type=friend
host=dynamic
secret=otherSecret
context=local
nat=no
canreinvite=no
qualify=no
dtmfmode=rfc2833
allow=all
call-limit=1
busy-level=1
allow=all [Caller]
type=peer
host=voip1.domain.net
deny=0.0.0.0/0.0.0.0
permit=xxx.xxx.xxx.xxx
context=myAccess

Asterisk Users 3.8 years ago 0 Answers

Calling an independent gateway from asterisk

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Amar, In general, gateways don't register. They are simply defined as a peer and calls are routed to them in the dialplan. When I do this I usually use the local channel to get to the
dialing contexts. Get in touch if you need a more detailed example Bruce Ferrell On 11/14/2011 10:01 PM, Amar Akshat wrote:
> Hello,
> I have a testing scenario at hand. I want to make a call from Asterisk
> CLI or AMI to an external network gateway. Is this possible.
> Let me explain the use case.

Asterisk Users 3.9 years ago 0 Answers