Yes they did.
On 3/26/12, SamyGo wrote:
> Good to know, hope our replies did some help :)
>
> On Thu, Mar 22, 2012 at 7:39 PM, James Mutuku wrote:
>
>> Hi,
>>
>> Thanks for the support. Issue solved. Somehow the routes on the fxo
>> gw were not working.
>>
>>
>>
>> On 3/21/12, James Mutuku wro
At first I was like "*DIAL(SIP/92${NUM}@SIP_PROVIDER)*" But Danny and Sam
are right, the information is incomplete to give any answer at all !!
BR/
Sammy
On Fri, Mar 23, 2012 at 8:16 PM, Lutgring, Sam wrote:
> The short answer is yes you can. Now the longer answer is give us more
> detail if y
Good to know, hope our replies did some help :)
On Thu, Mar 22, 2012 at 7:39 PM, James Mutuku wrote:
> Hi,
>
> Thanks for the support. Issue solved. Somehow the routes on the fxo
> gw were not working.
>
>
>
> On 3/21/12, James Mutuku wrote:
> > Hi,
> >
> > I have configured a route on the fxo
Hello All,
I am Asterisk user, and right now I have some troubles about Asterisk As Client
settings.
Here are my envrionments:
Asterisk-1.8.5.0
---
Server Settings(IP:172.16.70.121)
extensions.conf
[from-in
Bingo, it was the "r" option!
Thank you
Leandro
2012/3/25
> Do you have "r" in your dial string?
> If yes remove that
> -Original Message-
> From: Leandro Dardini
> Sender: asterisk-users-boun...@lists.digium.com
> Date: Sun, 25 Mar 2012 11:35:45
> To: Asterisk Users Mailing List - No
Do you have "r" in your dial string?
If yes remove that
-Original Message-
From: Leandro Dardini
Sender: asterisk-users-boun...@lists.digium.com
Date: Sun, 25 Mar 2012 11:35:45
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Reply-To: Asterisk Users Mailing List - Non-Commer
As far as I know, this is not the general tendency of any B2BUA that generates
such media independently. However, I could be mistaken.
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Atlanta, GA 30030
Tel: +1-678-954-0671
Web: http://www.evaristesys.com/, http
I want to have the early media to pass from the provider down to the soft
phone because it contains important information about the call, like "Your
call cannot go through, please try your call again " ... The provider is
giving this info via early media, just after the 183 SESSION PROGRESS.
Leand
I think I may have misunderstood your initial question, sorry.
You are looking for Asterisk to directly pass through the early media from
upstream? Why would it do that?
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Atlanta, GA 30030
Tel: +1-678-954-0671
We
The asterisk box has only one interface. I am capturing all the traffic on
the box and the only audio traffic is from the provider to the asterisk box.
Obviously if I set progressinband=yes, then I get the ringing tone from the
asterisk box, but no the audio from the provider I was looking for.
L
Are you absolutely sure that nothing is coming out, even on a different
interface than the one on which you are capturing? Are you capture on the
Asterisk server and not the receiving host?
Secondly, are you absolutely positive that something is supposed to be coming
out? 183 does not logical
All NAT and firewall problems are already been excluded. All peers are on
public IP address and no firewall is active between them. The missing
routing of the audio path to the peer has been checked with tcpdump ...
nothing is coming out from the asterisk box.
Leandro
2012/3/25 Alex Balashov
>
I assume you have ruled out NAT and firewall issues?
Between those two, 99% of the reasons why something may not be routed somewhere
correctly are accounted for.
If you don't know, your best bet is to take a packet capture or SIP debug on
the Asterisk server and find out where that early media
Hello,
I have a problem with premature media and inband progress audio. I am using
the latest 1.8.10.1 and this is the setup:
soft phone --- asterisk --- SIP provider
The number I call is giving back some hints via inband audio I am not able
to ear from the soft phone. They stop on the asterisk a
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