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.
>
> Th
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.
The SBC in use needs to match incomming calls from the asterisk with
the call id u
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
call
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 184
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 th
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
-- SIP/FD-
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
hello list
i have an issue related to outbound calls i can contact all the number
except on number given by our provider in trunk
the issue just when i configure my trunk in our server but when i configure
the trunk directly in x-lite i can contact this number without issue
below the cli
== U
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>
Hi All,
I'm using Asterisk 1.8.2 with outbound calls using Google Voice. I've been
having issues calling several toll free numbers where the call 'is ringing'
but never transitions to 'answered'. These are toll free numbers which are
typically answered by an ivrs where you enter eg. a conference b
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
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 cha
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
changed every setting that i can see and nothing fixes thi
n...@e-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
p_custom.conf
>> #include sip_additional.conf
>>
>> ;sip_custom_post.conf If you have extra parameters that are needed for
>> a
>> ;extension to work to for example, those go here. So you have
>> extension
>> ;1000 defined in your system you start by cre
meter that is needed.
> ;When the sip.conf 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:
When the sip.conf 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 r
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
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 a
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 th
Subject: [asterisk-users] outbound calls not ringing
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
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 fine. it sounds great. How do I
start to troubleshot t
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 p
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
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 seems valid, but no sound comes through any more.
How would you go through to troubleshoot thi
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 appr
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
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 email contact with
support at vitelity. They're not resp
Hi,
I am just trying to but an Asterisk server between a German Telecom
(Deutsche Telekom) PMUX and a Siemens HiCom. Inbound Calls (Telekom ->
HiCom) work like charm. Outbound calls however all end up being
congested:
-- Executing Dial("Zap/50-1", "Zap/g1/06151343|180|WT") in new
st
Greetings all-
Long story short - I find myself suddenly running a Asterisk PBX after old PBX
suddenly died. Fortunately, I had been "playing" with Asterisk (via Trixbox)
on a server in consideration of replacing our aged Merlin Legend - so over
the course of last weekend, I brought my testbed
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
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.
Regards,
Ajit
___
--Bandwidth and Colocation provided by Easynews.com --
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 nu
> 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,
Hello All,
Phone: Cisco 7960G
Asterisk 1.2.7.1
libpri 1.2.2
zaptel 1.2.5
OS: Fedora Core 4
TDM2400P w/8FXO
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 read
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
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 an effort to make o
My issue is dialing out to a local prefix does not always connect. The telco
"were sorry, your call does not come thru..." message is received. If I dial
my cell phone (a 201 prefix vs. a 758 prefix) then my cell phone rings every
time. My clients are Xlite on a Mac and Xpro on a pocketpc.
I have
Hi,
I'm using the macro below in extensions.conf for most of my outbound
calls. One issue with my current configuration is that when I make an
outbound call it doesn't properly detect that my PSTN line (Zap/1) is
busy with another call and then overflow to my outbound IAX
connections. I think 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 am,
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:
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 will
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
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 will be highly a
Hi All,
With the help and patience of this forum, I have been able to set my
asterisk box to make outbound
calls to iconnecthere. My intention is to make two such calls and bridge
them( three way calling) . Based on a earlier suggestion, I have created two
accounts with iconnect and have succes
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