-- Forwarded message --
From: Peter Kunz munged
Been there, many, many times.
http://xkcd.com/806/
Look at this comic, you will laugh, I guarantee it!
Thanks Peter!
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Does it working good with RFC standard? Or where can I get a crack version?
Thanks
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New to Asterisk? Join us for a live introductory webinar every Thurs:
Hi Jigar
I am facing issue while generating a dial plan for the following case:
all caller should be asked a code to enter than All the callers should be
connected one extension.
Try DISA component, and then use MeetMe component if you want callers to go
to conference or Dial component if you
Actually it is bad only when received on cell phones. Today I listened to
the same voices on a Cisco 7942 and they were great. I actually enjoyed
listening to them. Not bad on X-Lite either. Previously I was mostly
listening to them only through cell phones. So it means it is because of the
I totally agree with Steve's wise advice. One should at least give himself a
week learning asterisk fundamentals and related Linux basics before jumping
into creating dialplans or setting up Telecom systems. Asterisk's official
book's first few chapters cover all the basics which every asterisk
Thanks Kevin to verify this. This would really solve a very big problem for
me as E1-T1 conversions has been a big part of my work lately, with no
satisfactory and reliable solution yet. I'll propose this card to my client
and would love to try it.
Zeeshan A Zakaria
--
www.ilovetovoip.com
Do you recommend using wav files instead? Will there be any downside of
using wav?
Zeeshan A Zakaria
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www.ilovetovoip.com
www.pbxforall.com (beta)
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Well, the downside to wav files is the disk i/o. Asterisk will and does
translate the audio frames from ulaw to whatever other codec.
Sent from my iPhone
On Oct 24, 2010, at 9:42 AM, Zeeshan Zakaria zisha...@gmail.com wrote:
Do you recommend using wav files instead? Will there be any
Paul,
Further to my last, I think I found another small related issue with IAX which
is generating the following error:
[Oct 24 14:42:12] ERROR[15589]: netsock2.c:94 ast_sockaddr_stringify_fmt:
getnameinfo(): ai_family not supported
To reproduce this issue, setup a phone in iax.conf or your
I fiddled with the demo version of swift a year or so ago and I had better
sound quality if I used the non-8khz versions and had app_swift or asterisk
convert it for me (not sure, giving app_swift a regular version seemed to
JustWork(tm)
On Sat, Oct 23, 2010 at 3:24 PM, Zeeshan Zakaria
Hello,
My setup is :
phone - PSTN/ISDN - Patton SN4638 --- Asterisk
(Asterisk is in 1.6.1.18, Patton in 5.3)
When I call the Asterisk, I can read from console that :
- the call comes in,
- the line MusicOnHold(,10) in my diaplan is reached and played,
- I see RTP packets coming in
2010/10/24 Olivier oza_4...@yahoo.fr
Hello,
My setup is :
phone - PSTN/ISDN - Patton SN4638 --- Asterisk
(Asterisk is in 1.6.1.18, Patton in 5.3)
When I call the Asterisk, I can read from console that :
- the call comes in,
- the line MusicOnHold(,10) in my diaplan is
2010/10/21 Zakir Mahomedy z...@mayfair2000.com
Hi
I wonder if anyone could give some light on SIP NAT.
I've having a friken headache with SIP NAT 1 way audio.
Client - NAT - NAT - Server
Client can hear users from server side
but server cant hear client.
Ive tried every
Hi all,
I'm being requested to deploy an IVR service using SS7.
I've deployed Asterisk before using ISDN connection, but never with SS7.
Can anyone explain me the different between using ISDN and SS7 ? What need I do
now to change to use SS7 ?.
Many thanks,
Giang
--
2010/10/24 Olivier oza_4...@yahoo.fr
2010/10/14 Danny Nicholas da...@debsinc.com
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
*Sent:* Thursday, October 14, 2010 3:34 PM
*To:*
Adding an Answer() before MusicOnHold made it works.
Thanks for everyone that helped !
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My asterisk machine has 2 nic's. One nic (ETH0) is connected to a cable
modem. The other nic (ETH1) is connected to an internal lan. The
internal lan also has access to the internet.
The cable service, Time-Warner RoadRunner, is great when up, but is not
reliable. And sip connections are
Hi cary,
Can you recommend me what add-on vendors I should use ?
Can a open source solution such as chan_ss7 or libss7 support many conncurrent
calls (for example 240 calls) ?
Thanks
From: Cary Fitch ca...@usawide.net
To: Asterisk Users Mailing List -
I do not have knowledge of the SS7 vendors for Asterisk. Using redundant
56k data channels, we handle calls via 6 DS3s (672 X 6 calls) from the PSTN
on a commercial telephone switch, with no issues at all.
SS7 can support any number of simultaneous calls depending only on the
bandwidth of the
On Sun, Oct 24, 2010 at 10:06 AM, Nic Colledge n...@njcolledge.net wrote:
Further to my last, I think I found another small related issue with IAX
which is generating the following error:
Do you mind collecting a debug log [1]? Having some issues reproducing this.
[1]
Paul,
I made a debug log of the register and unregister process for a single Zoiper
client using IAX and have emailed it direct to you.
The error shows in the file as:
[Oct 24 19:07:32] ERROR[1403] netsock2.c: getnameinfo(): ai_family not supported
Thanks,
Nic.
-Original Message-
From:
1.6.2.13, sip.conf:
[155]
type=friend
context=longdistance
callerid=Admin 155
secret=test
host=dynamic
dtmfmode=rfc2833
allow=all
defaultuser=155-trust
On aastra:
Basic SIP Authentication Settings
Screen Name
Phone Number 155
Caller ID 155
Hi all,
I used to configure each of my sip clients with a unique identifier via
setvar. These clients were all configured as friends.
However, now that I've got some Polycom phones, which MUST be peers, I am
unable to define this variable.
For example, this works:
[friend-client]
context =
Evening,
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
Thanks
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On Sun, Oct 24, 2010 at 6:23 PM, Stephen Reese rsre...@gmail.com wrote:
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
I wrote one last week:
http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/
Also: http://www.davidvossel.com/?p=28
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Hi all,
Please, anybody that have some knowllege about E1 configuration could give
some guidance about it?
I trying to set an Asterisk with E1 CAS signalling and everything looks good,
but when I try to go out with calls I receive the follow message:
== Using SIP RTP CoS mark 5--
On Sun, Oct 24, 2010 at 7:06 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Sun, Oct 24, 2010 at 6:23 PM, Stephen Reese rsre...@gmail.com wrote:
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
I wrote one last week:
Forget it !!
After several attempts, I have solved !!!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
From: flaviormira...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Sun, 24 Oct 2010 22:28:16 -0200
Subject: [asterisk-users] E1 configuration
On Sun, Oct 24, 2010 at 9:24 PM, Stephen Reese rsre...@gmail.com wrote:
On Sun, Oct 24, 2010 at 7:06 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Sun, Oct 24, 2010 at 6:23 PM, Stephen Reese rsre...@gmail.com wrote:
Has anyone seen a how-to on getting Asterisk to work with Google
Anything on this guys?
I am sure someone had the need to record the HOLD time or maybe it is
already being recorded somewhere?
Any thoughts are appreciated.
Thanks,
Bruce
On Wed, Oct 20, 2010 at 3:30 AM, Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
We are using Queuemetrics but it
On Sunday, October 24, 2010 05:23:13 pm Stephen Reese wrote:
Evening,
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
Thanks
For Google Voice, I use an ipKall number for the inbound trunk. Here are the
relevant sections of my extensions.conf:
;
I am running Asterisk on Ubuntu 2.6.32-25-server with asterisk
1.6.2.5-0ubuntu1 and dahdi 2.2.1-0ubuntu2.
The machine has a passive HCF-based PCI ISDN card and an Astribank 8
attached. The ISDN card works fine.
r...@servaction:~# lsusb
Bus 001 Device 002: ID 04b4:8613 Cypress Semiconductor Corp.
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