Matt Hoskins wrote:
Hello,
Kia ora,
I am currently trying to set up pjsip realtime and would like to have
outbound-publish, inbound-publication, and asterisk-publication sorcery
object types in ODBC realtime. Is that currently supported? I know that
some object types are known working and
I have two machines on the internet. Box A and Box B.
Box A has a SIP trunk to the world, Making calls box A works fine
I have audio to my cell and all works.
I defined a SIP trunk between box B and A. tried to make a call originating
from box B - to box A and then over the SIP trunk to my cell.
Hello
Le 17/02/2015 17:00, Administrator TOOTAI a écrit :
Hi,
as stated in the documentation, it's allowed to set
FAXOPT(faxdetect)=yes/no to allow fax detection.
It's done (see below) but still fax detection :-( Extension 300 is
hylafax with iaxmodem.
On the upper Asterisk gw it's the
But the phone rings - so its routed - just no audio.
The ringing is SIP signaling. The audio is RTP data. See if the audio
is getting routed with a sniffer. Maybe use one codec that both clients
support.
Adrian Serafini
--
I solved the issue by not answering the call as I assume others have done.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator
TOOTAI
Sent: Wednesday, February 18, 2015 12:50 PM
To:
Excellent. I was using ast-13.1.0 with no luck. I upgraded to 13.2.0 and
have made it further, but am having a little difficulty. The
outbound-publish object types seems to be working in realtime now. But
the asterisk-publication object is only reading from sorcery.conf. I know
you said that
Justin Killen wrote:
Joshua,
If I'm understanding this correctly, you're saying that the Playback
is failing because it isn't connected to anything on the other end,
because the Dial() failed. When the channel is created on the
OutgoingSpoolFailed extension, what context is it created in, one
Joshua,
If I'm understanding this correctly, you're saying that the Playback is failing
because it isn't connected to anything on the other end, because the Dial()
failed. When the channel is created on the OutgoingSpoolFailed extension,
what context is it created in, one of the origin legs?
Hi all
Is there a relation between the above?
I'm having a problem where I suspect my internet access provider (through
whom I go to a SIP trunk provider) have got MTU size problems.
My asterisk (1.8.11.0) is constantly going into the situation where a
TimerFD error is spammed in the CLI, load
What are the cons, if any, of enabling a jitterbuffer?Â
Memory and latency.
--
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Hello,
I am currently trying to set up pjsip realtime and would like to have
outbound-publish, inbound-publication, and asterisk-publication sorcery
object types in ODBC realtime. Is that currently supported? I know that
some object types are known working and others are not. I was curious
Hello people
What are the cons, if any, of enabling a jitterbuffer?
We are currently using version 1.8
Thanks in advance
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
I just want to make a SIP call from 192.168.1.3 to 192.168.1.4; or not
even a call. Ring? Beep? Ping? Some sort of hello world connection.
192.168.1.1 netgear router
192.168.1.2 asterisk (vicidial)
192.168.1.3 ubuntu client
192.168.1.4 mac OSX client (not shown)
Do I
On Tue, 17 Feb 2015 08:28:31 -0600, Matthew Jordan wrote:
Asterisk attempts to look up who the OPTIONS request is for, using the
username portion of the request URI. Make sure you have a matching
extension for what your upstream provider is sending you, and chan_sip
will respond with a 200
Hi,
I **think** that I have user of thufir101, because I get a 200 response
below, but I also get a 404. It seems to depend on how I send the ip
address/fqdn?
tleilax*CLI
tleilax*CLI sip show users
Username Secret Accountcode
Def.Context ACL
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