On Mon, Aug 28, 2017, at 05:45 AM, Benoit Panizzon wrote:
> Hello List
>
> > I work at an SIP Provider and we have added and SBC in front of our
> > Voice Switch to protect it.
>
> Well using two peers for incomming and outgoing calls solve the
> previous issue.
>
> Now I have a new one.
>
>
hi
the issue still the same i have 2 trunks whe i configure the first in
x-lite and the second in my server or my ip-phone snom320 directly
from x-lite i can call my trunk without issue but when i try ti call from
snom320 to x-lite or from my server asterisk using extension in x-lite the
thanks for your response
i noticed that when i active the voicemail in the IP-phone where the number
0033149xx is configured i can call this number without issue
the server asterisk and the ip-phone where the number is configured are in
the same network 192.168.1.X
Using SIP RTP TOS bits
i noticed that when i active the voicemail in the IP-phone where the number
0033149xx is configured i can call this number without issue
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/FD/0033149xx == Begin MixMonitor Recording
SIP/101-010d
--
I am making some assumptions, but assuming the 217.195.xx.xxx is your
provider, you are getting this back from them:
Got SIP response 556 No address found back from 217.195.xx.xxx:5060
Are you sure that 0033149xx is the format the provider is expecting?
You might try enabling SIP debug on
So you are saying that it resolved the issue to activate voicemail on the
device that sits past your trunk provider? That confuses me a little, but
if your calls are working, that's great news.
On Fri, Mar 20, 2015 at 1:44 PM Salaheddine Elharit
salah.elharit...@gmail.com wrote:
i noticed that
Apologize for following up to my own question, but wanted to mention that
some toll free numbers with ivrs work fine. Only run into issues with
certain numbers like the test number in my previous email.
Any ideas?
On Fri, May 13, 2011 at 10:26 AM, Gaurav P
gaurav.lists+asterisk-us...@gmail.com
3 sep 2009 kl. 00.27 skrev John A. Sullivan III:
On Wed, 2009-09-02 at 21:31 +, Ott Rose wrote:
i have posted this before but was unable to resolve it. i have some
new info so i figured i would try again. the trace from bandwidth.com
are below. they are telling me that the ip that is
On Wed, 2009-09-02 at 21:31 +, Ott Rose wrote:
i have posted this before but was unable to resolve it. i have some
new info so i figured i would try again. the trace from bandwidth.com
are below. they are telling me that the ip that is bold should be our
ip not bandwidth.com. i have
-simple.co.nz
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] outbound calls not ringing
Generally with FreePBX the ring options are set in the General Options -
you can set the Dial options which are normally tr, but I guess that
isn't working for you.
The SIP files you
Have you tried putting a (,r) on your Dial command (dial
dahdi/1/18005551212,60,r) ?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ott Rose
Sent: Wednesday, August 19, 2009 8:55 AM
To: asterisk-users@lists.digium.com
On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
I put a post on here about my issues with outbound calls not ringing
but i haven't resolved it. so i am trying again.
When i dial any outside number i dont get a ring tone at all. when the
person picks up and starts to talk i can hear them
we are using Aastra 57i
i don't see that setting. where is it at?
From: jsulli...@opensourcedevel.com
To: asterisk-users@lists.digium.com
Date: Wed, 19 Aug 2009 11:07:21 -0400
Subject: Re: [asterisk-users] outbound calls not ringing
On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
I
sip.conf
On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote:
we are using Aastra 57i
i don't see that setting. where is it at?
From: jsulli...@opensourcedevel.com
To: asterisk-users@lists.digium.com
Date: Wed, 19 Aug 2009 11:07:21 -0400
Subject: Re: [asterisk-users] outbound calls
is loaded it will append your additions to the end of
;that extension.
;
#include sip_custom_post.conf
From: jsulli...@opensourcedevel.com
To: asterisk-users@lists.digium.com
Date: Wed, 19 Aug 2009 12:17:15 -0400
Subject: Re: [asterisk-users] outbound calls not ringing
sip.conf
On Wed, 2009-08
:21 -0400
Subject: Re: [asterisk-users] outbound calls not ringing
On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote:
I put a post on here about my issues with outbound calls not
ringing
but i haven't resolved it. so i am trying again.
When i dial any outside
...@opensourcedevel.com
To: asterisk-users@lists.digium.com
Date: Wed, 19 Aug 2009 12:17:15 -0400
Subject: Re: [asterisk-users] outbound calls not ringing
sip.conf
On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote:
we are using Aastra 57i
i don't see that setting. where
On Mon, Aug 3, 2009 at 8:20 AM, Guillaume Yziquel
guillaume.yziq...@citycable.ch wrote:
Hello.
I've set up and configured an Asterisk server to make SIP phone calls to
external classic phones.
However, it happens that after 15 or 30 seconds, the phone call drops.
The SIP session still
Steve Totaro a écrit :
On Mon, Aug 3, 2009 at 8:20 AM, Guillaume Yziquel
guillaume.yziq...@citycable.ch wrote:
Hello.
I've set up and configured an Asterisk server to make SIP phone calls to
external classic phones.
However, it happens that after 15 or 30 seconds, the phone call drops.
On Tue, 2009-07-28 at 13:33 -0500, Tom Poe wrote:
Any vitelity customers with pbxinaflash boxes? I'm able to call
in-house, but failing to make outbound calls. My assigned server at
vitelity is not reachable. I can ping to my ISP OK.
Any help appreciated. Such as actually how to make
John A. Sullivan III wrote:
On Tue, 2009-07-28 at 13:33 -0500, Tom Poe wrote:
Any vitelity customers with pbxinaflash boxes? I'm able to call
in-house, but failing to make outbound calls. My assigned server at
vitelity is not reachable. I can ping to my ISP OK.
Any help appreciated.
On Mon, Apr 24, 2006 at 04:46:29PM +0530, Ajit spake thusly:
Hi,
I wish to use the manager API to make an outbound call to a sip
url,subsequently play a prompt and hangup.Any hints on how to acheive
this/feasability will be much appreciated.
I'm no expert, but it looks simple enough to me -
Thanks
Inserting a w did resolve the problem. I saw another post from
today where somebody else is having the same problem with a
TDM2400P. Hopefully someday Asterisk will be coded to wait for a dial tone.
nb
On 4/19/06, Time Bandit [EMAIL PROTECTED] wrote:
When dialing an outbound number,
When dialing an outbound number, sometimes all the digits are not dialed
properly on the outside line. In the dial plan I added a SayDigits to the
outbound rule and it properly reads back the entire number that was entered
on the phone before dialing.
Is asterisk dialing too quickly, is
Mike Raley wrote:
Hi all, a noob here, I am trying to get outbound calls through
asterisk working with Broadvoice.
I have consulted the following two online tutorials:
http://www.broadvoice.com/support_install_asterisk.html
http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice
in
Frank wrote:
I've been looking through the archives and have not been able to find anyone
with a similar problem but perhaps I'm not searching in the right places. The
problem is that my outbound call sometimes go though and sometimes don't. If
someone can point me in the right direction it
Thats amazing! Worked like a charm...any explanations as to why this happens?
On Wednesday 19 January 2005 03:21 am, Matt Riddell wrote:
Frank wrote:
I've been looking through the archives and have not been able to find
anyone with a similar problem but perhaps I'm not searching in the
Frank wrote:
Thats amazing! Worked like a charm...any explanations as to why this happens?
Basically some connections require you to wait a little bit before
dialling the number. Without the w's it dials straight away. With them
it pauses and then dials.
On Wednesday 19 January 2005 03:21
On Tue, 2005-01-18 at 05:20 -0500, Frank wrote:
I've been looking through the archives and have not been able to find anyone
with a similar problem but perhaps I'm not searching in the right places. The
problem is that my outbound call sometimes go though and sometimes don't. If
someone can
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