Hi All,
Can someone please recommend a FXS bank that support a minimum of 12 ports. I
would prefer an IP connection to Asterisk rather than a USB or physical card.
If an IP style is not available I'll consider a USB type. A card is not an
option.
Your recommendations would be greatly
Hello,
I am able to get the call recording file path of each call in the CDR. How
can I get the realtime call recording streaming?
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Hi,
We have the following network architecture :
UAC1-kamailioVoipSwitch-PSTN--Phone1
(Sip Client)
Now UAC1 calls Phone1 and everything is ok. If UAC1 hangs up session
is terminated cleanly.
But if Phone1 hangs up the BYE message which
Hi Sammy go
Can you help me with my problem
I have asterisk 1.8 and i am using asterisk-gui 2.0, and in asterisk-gui
2.0 the voice prompt menu which is used for custom voice recording for IVR
is not working and not recording. Can u tell me how to defualt this feature.
thanks
--
Please share if anyone has encountered this cdr issue with call transfer.
On Fri, May 18, 2012 at 5:32 PM, [Digital^Dude] ®
millennium@gmail.comwrote:
Hello,
I'm using attended call transfer in asterisk 1.8.11.0 on a CentOS machine.
Each CDR entry of calls that are transferred is
Hi List,
I am trying to add new SIP account in new file additional_sip.conf. I read
in Wiki there is API command UpdateConfig which is used to update , add and
delete any entry from configure files. I am using PHP to make new entry in
additional_sip.conf. Below is the code which I tryed
Hi,
1- try putting absolute filepath in source and destination field.
2- verify that the permissions of the files you're changing.
Regards,
Sammy.
On Mon, May 21, 2012 at 5:10 PM, virendra bhati virbh...@gmail.com wrote:
Hi List,
I am trying to add new SIP account in new file
Hi,
Can you check if there is any transcoding involved with these calls, or
maybe some NAT handling needs to be done by asterisk so it's not stepping
out of the media-path !?
Regards,
Sammy
On Mon, May 21, 2012 at 5:03 PM, David Wessell da...@ringfree.biz wrote:
I am attempting to get an
I have update sammy but no luck
?php
$socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
if (!$socket)
{
$done=0;
} else {
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: admin\r\n);
fputs($socket, Secret: admin\r\n\r\n);
fputs($socket, Action:
All g711 calls, and the only nat is on the endpoint.
Snom M9 Phone (behind nat) - PBX (Public IP) - LCR Trunk (Public IP)
- SIP Provider (Public IP)...
I'm expecting the LCR trunk to get out of the media path and connect
the PBX with the SIP Provider
Thanks
David
On Mon, May 21, 2012 at
On 05/20/2012 11:29 PM, Danny Dias wrote:
I have a question regarding DPMA for Digium Phones, if i install the
DPMA on my Asterisk Server A, and then, i move the phone to register
into another Asterisk Server B, can i install for free another DPMA
license for my digium phones on this second
On 05/20/2012 11:48 PM, Danny Dias wrote:
By the way, the DPMA is only available for Asterisk Certified, is that
right? is there any problem for an Asterisk production server on a
customer with Asterisk OpenSource 1.8.5 to migrate to Asterisk
Certified? What is exactly an Asterisk Certified? Do
No one have an idea ?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: 2012-05-19 15:27
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] IAX2 passing back and forth
There was a nice thread on this back in April.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: Monday, May 21, 2012 9:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re:
Hello,
I am using asterisk 1.4 and I have the following issue: I would like to
turn off the voicemail option that asks you to press 1 (to accept this
message), 2 ( to listen to it) and 3 (to rerecord your message)when
you make the record since they are not in my native language. Does
That's what the ,s switch on voicemail() is for.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bogdan
Sent: Monday, May 21, 2012 9:52 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk
Since this didn't ring a bell with anybody, it must not be related to asterisk.
Digging further into my code, I may have found the reason for this. Note to
others who may find this post later, looks like asterisk can take Originate
actions as fast as you can give it, look other places..:)
--
I am using that but does not seem to work.
On 05/21/2012 05:53 PM, Danny Nicholas wrote:
That's what the ,s switch on voicemail() is for.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bogdan
Sent: Monday,
Your other option would be to set CHANNEL(language) to your native
language. The ,s option only suppresses the initial greeting; you are
wanting to suppress the after recording instructions.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
To turn of the after recording messages, in voicemail.conf change review=yes to
review=no
--
Mehmet Avcioglu
meh...@activecom.net
On May 21, 2012, at 6:11 PM, Danny Nicholas wrote:
Your other option would be to set CHANNEL(language) to your native
language. The ,s option only suppresses
I will do that. Thank you very much for your answer!
Best Regards,
Bogdan
On 05/21/2012 06:11 PM, Danny Nicholas wrote:
Your other option would be to set CHANNEL(language) to your native
language. The ,s option only suppresses the initial greeting; you are
wanting to suppress the after
On 05/21/2012 07:03 AM, David Wessell wrote:
I am attempting to get an asterisk server to step out of the media
path, but am running into a brick wall. Can someone assist? Here's my
setup..
Ultimate SIP Provider --- LCR Trunk (Asterisk 1.6) PBX (Asterisk 1.8).
In order to be able to
On 05/21/2012 06:13 PM, Mehmet Avcioglu wrote:
To turn of the after recording messages, in voicemail.conf change review=yes to
review=no
Hello,
That did it!!! I still have a thank you at the end but I assume that
is because of the default language set to English. Thank you very much
for
Gradstream has a 24 port solution: GXW4024
Sangoma has a new solution for up to 50 ports: Vega5000
On Mon, 2012-05-21 at 06:04 +, Klaverstyn, David C wrote:
Hi All,
Can someone please recommend a FXS bank that support a minimum of 12
ports. I would prefer an IP connection to
We use Adtran Total Access boxes to convert PSTN to SIP.Xorcom has some
PSTN/SIP USB boxes which people seem to love.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Monday, May 21,
Hi Kevin,
Thank you. Here's the requested information.
1) The Trunk is running 1.6.2.9. Also it's running a2billing.
2) The PBX is running asterisk 1.8.12.0 along with FreePBX.
3) I did directmedia on the trunk and canreinvite on the pbx since
they were different versions.
Thansk
David
On Mon,
On 05/21/2012 11:46 AM, David Wessell wrote:
Hi Kevin,
Thank you. Here's the requested information.
1) The Trunk is running 1.6.2.9. Also it's running a2billing.
2) The PBX is running asterisk 1.8.12.0 along with FreePBX.
3) I did directmedia on the trunk and canreinvite on the pbx since
they
Thanks Sammy, I think I'll stop using SIP realtime.
Regards,
Ricardo.
On Mon, May 21, 2012 at 5:14 AM, SamyGo govoi...@gmail.com wrote:
Hello Ricardo,
The reason why your asterisk refused the calls from phone registering on
SIP proxy is that it only gets INVITE of the call from: a user
So I need directmedia set in sip.conf on the LCR trunk.
1) Do I need it in the individual trunk settings for each pbx? Or is
in sip.conf enough?
2) Do I need anything on the pbx side that we are hoping to transfer media to?
3) How long into the call before the media is transferred over?
Thanks
On 05/21/2012 12:54 PM, David Wessell wrote:
So I need directmedia set in sip.conf on the LCR trunk.
1) Do I need it in the individual trunk settings for each pbx? Or is
in sip.conf enough?
You say 'in sip.conf' multiple times, but that's far too vague to mean
anything. sip.conf is a
Hello,
I'm using asterisk v1.8 with a standard scenario, A Sip call from A to B
through asterisk :
A --SIP-- ASTERISK --SIP-- B
The asterisk extension is :
exten = _X.,1,Dial(SIP/B/${EXTEN},600)
exten = _X.,n,Hangup()
When B send a 404 back to the asterisk, the asterisk sends a 503 to A. It
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