Hi
Does anyone used asterisk as a operator phone,with multiple lines
and features like transfer forward and etc.I used chan_alsa driver to
make asterisk as SIP Phone,but it has limitation,we cant make or receive
multiple calls,and will not able to do any features like transfer
forward
On Thu, 2011-07-21 at 17:13 -0600, Mike Diehl wrote:
Hi all,
I've got a strange problem with a customer's phones.
They've got a bunch of Grandstreams that seem to be rock solid... until
7:00pm. At 7:00, some of the phones become unavailable, and stay down. Call
quality is solid almost
hi,
i'm trying build asterisk rpm
normal compilation is ok but rpm building always fail
centos6/asterisk 1.8.5.0
any ideas?
gcc -o recno/rec_utils.o -c recno/rec_utils.c -MD -MT recno/rec_utils.o
-MF .recno_rec_utils.o.d -MP -pthread -Wall -D__DBINTERFACE_PRIVATE -I.
-I.. -Iinclude -Ihash
On Friday 22 July 2011 1:42:33 am Ishfaq Malik wrote:
On Thu, 2011-07-21 at 17:13 -0600, Mike Diehl wrote:
Hi all,
I've got a strange problem with a customer's phones.
They've got a bunch of Grandstreams that seem to be rock solid... until
7:00pm. At 7:00, some of the phones become
Is anybody else seeing this at all ?
--
Thanks, Phil
- Original Message -
Just received a call and on checking messages I now see:
ERROR[14824] pbx.c: Function MASTER_CHANNEL not registered
Grrr, looks like time to go back to 1.8.3 as all the apps and
functions exist in
Forwarded Message
From: Ishfaq Malik i...@pack-net.co.uk
To: Mike Diehl mdi...@diehlnet.com
Subject: Re: [asterisk-users] Strange network issue
Date: Fri, 22 Jul 2011 09:55:53 +0100
On Fri, 2011-07-22 at 02:53 -0600, Mike Diehl wrote:
On Friday 22 July 2011 2:42:12 am Ishfaq
On 22/07/2011 5:43 AM, Israel Gottlieb wrote:
On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming
kpflem...@digium.com mailto:kpflem...@digium.com wrote:
On 07/21/2011 04:34 PM, Joaquin Sosa wrote:
On Mon, Jul 18, 2011 at 07:58, Steve
Daviesdavies...@gmail.com
Hi!
I'm using ael language and I need to pick up a call from outside to an internal
number.
for example:
i'm 120
the phone 100 rings, it's a call from outside.
now I pick up the call with: *8100
and I would expect to answer the call but the response is Declined
the Puckup code is below:
_*8X!
I think I have solved with the following code:
_*8X! = {
PickUpChan(SIP/${EXTEN:2});
Hangup();
}
thanks
From: Alessio
Sent: Friday, July 22, 2011 11:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Pickup(${EXTEN:2}); not works from outside
Hi!
I'm using ael
Benoit Panizzon benoit.paniz...@imp.ch writes:
Is there a way to get asterisk not to invent a CALLERID(name) if there is
none?
Id did try to set ${CALLERID(name)=} but that resulted in From: sip...
and the displaying of this empty string on the subscribers phone.
I believe you have hit
I see, thank you for explaning. The reason for my concern is, we are
sometimes having DTMF issues on outbound calls. It seems when the user
(Polycom) enters digits, they are not being recognized by the other end.
On Thu, Jul 21, 2011 at 5:17 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On
I have a call trace of one of these calls...and this seems strange:
asterisk sends on INVITE
a=fmtp:101 0-16
then 183 Session progress is sent back with:
a=fmtp:101 0-16
then asterisk sends 183 Session progress with:
a=fmtp:127 0-16
OK is sent back with:
a=fmtp:101 0-16
then asterisk sends OK
Most likely if DTMF is not recognized on the far end, it would be an
incompatibility setting of DTMF support or bug on either UAC and UAS.
Wireshark trace at both end will help you understand the issue.
On Fri, Jul 22, 2011 at 8:12 PM, vip killa vipki...@gmail.com wrote:
I see, thank you for
How can we wireshark a trace on the remote end? It is a peer such as Level3
or Dash
On Fri, Jul 22, 2011 at 9:15 AM, Jesie Paluca jesie.pal...@gmail.comwrote:
Most likely if DTMF is not recognized on the far end, it would be an
incompatibility setting of DTMF support or bug on either UAC and
On Thu, Jul 21, 2011 at 7:13 PM, Mike Diehl mdi...@diehlnet.com wrote:
Hi all,
I've got a strange problem with a customer's phones.
They've got a bunch of Grandstreams that seem to be rock solid... until
7:00pm. At 7:00, some of the phones become unavailable, and stay down.
Call
quality
Hi all,
I have a major issue with a codec renegotiation in an asterisk 1.4.33.1
setup, which leads me to ask a general question about asterisk 1.4.X codec
negotiation: asterisk can support a re-negotiation of a codec on the fly
through a re-Invite? If my SIP provider sends me a re-invite changing
Asterisk does not support changing codecs on the fly.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matteo Campana
Sent: Friday, July 22, 2011 10:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On Fri, Jul 22, 2011 at 4:46 PM, Eric Wieling ewiel...@nyigc.com wrote:
** **
Asterisk does not support changing codecs on the fly.
And why asterisk sends 200 OK to the provider, if does not support its
re-invite?
M.
** **
*From:* asterisk-users-boun...@lists.digium.com
From: Eric Wieling ewiel...@nyigc.com
Sent: Friday, July 22, 2011 10:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Question about codec re-negotiation in
asterisk 1.4.X
Asterisk supports reinvites (if reinvites are enabled in sip.conf), just not
changing codecs in the middle of the call. If anyone has managed to get it
to work, I'd love to hear about it.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Eric
With 1.8.x I use.
exten = Process,1,Set(SIP_CODEC=ulaw)
And the system kicks the call over to ulaw. Now this is just prior to the
answer so I don't know if it meets your criteria. But it works great to enforce
inline T.30 audio faxes. I also use the f/F option T.38 or T.30 on recevie
Ah, we do not use 1.8 yet.I've been unable to get 1.8 to transcode between
g722 and ulaw. I assume it is a config issue.
Does your (pre-answer) example change the codec for BOTH legs of the call or
just the incoming leg or outgoing leg? When I was referring to a call I
meant both legs
Hey all
I am looking at pulling out the macro command from all of my dial plans.
This is due to the long term phase out of macros as per the upgrade
documentation. I would like to get feed back from others on how they
might go about this. Based on what I see it looks like Gosub / Return is
From: Bryant Zimmerman brya...@zktech.com
Sent: Friday, July 22, 2011 11:21 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Phase out macro command.
Hey all
I am looking at pulling out the macro command from all of my dial plans.
Michael,
It looks like your problem is caused by a phone with a non-standard
SDP session version implementation. The phone is sending an INVITE
with SDP that contains an a=sendonly line. Asterisk should respond
with an OK that contains an a=recvonly line, but it responds with
a=sendrecv
Anyone have any configuration experience connecting Asterisk 1.8 to the PSTN
via SIP on a Taqua 7000 switch?
My local carrier recently upgraded software and changed their configs so that
signalling and media are on different cards (and hence different IP addresses),
and it's causing issues.
I
My provider has always sent the SIP control info from one IP and the media
packets from another. As long as your firewall passes the data there should be
no problem. I did not have to do anything special in my configuration. This is
using ABE which is based on 1.4.
--
Jim Dickenson
On Fri, Jul 22, 2011 at 10:10:01AM +0200, marek cervenka wrote:
hi,
i'm trying build asterisk rpm
normal compilation is ok but rpm building always fail
centos6/asterisk 1.8.5.0
any ideas?
gcc -o recno/rec_utils.o -c recno/rec_utils.c -MD -MT recno/rec_utils.o
-MF
They've got a bunch of Grandstreams that seem to be rock solid... until
7:00pm. At 7:00, some of the phones become unavailable, and stay down. Call
quality is solid almost all the time. But right at 7:00, things go bad.
Only
some of the phone lines go down and they stay down until the
I read Kevin's piece in asterisk-announce about the new numbering scheme,
and saw in svn-commits some tagging of 10.0.0-beta1.
Perhaps I'm thick (I hope not!), but I really can't see why calling the
next version 10.0.0 is any better than calling it 2.0.0!
I'm surprised not to have seen ANY talk
I agree, the numbering seems to make no sense. Oh well it's just an
arbitrary measurement of non-progress anyway
On Fri, Jul 22, 2011 at 3:44 PM, Tony Mountifield t...@mountifield.orgwrote:
I read Kevin's piece in asterisk-announce about the new numbering scheme,
and saw in svn-commits some
I thought it was going to be 1.10.0
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Friday, July 22, 2011 2:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 10.0.0 better
Hi Tony,
On 07/22/2011 09:44 PM, Tony Mountifield wrote:
I read Kevin's piece in asterisk-announce about the new numbering scheme,
and saw in svn-commits some tagging of 10.0.0-beta1.
Totally missed that one. Just did a quick browse.
Perhaps I'm thick (I hope not!), but I really can't see
Maybe just a typo ? Misplaced dots between all those 1's and 0's ...
Maybe we should call it version 12 instead of 1100 ;-)
Am 22.07.2011 21:50, schrieb Danny Nicholas:
I thought it was going to be 1.10.0
*From:*asterisk-users-boun...@lists.digium.com
Since this change I started measuring temperature in Rankine. Its now
592.67 degrees here (south NJ).
On Fri, Jul 22, 2011 at 3:44 PM, Tony Mountifield t...@mountifield.org wrote:
I read Kevin's piece in asterisk-announce about the new numbering scheme,
and saw in svn-commits some tagging of
On Fri, Jul 22, 2011 at 2:58 PM, Norbert Zawodsky norb...@zawodsky.atwrote:
Maybe just a typo ? Misplaced dots between all those 1's and 0's ...
Maybe we should call it version 12 instead of 1100 ;-)
Am 22.07.2011 21:50, schrieb Danny Nicholas:
I thought it was going to be 1.10.0
No,
Maybe they are trying to live down their bad press from 1.6 and 1.8 by
abandoning the 1.X schema.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Friday, July 22, 2011 3:27 PM
To: Asterisk Users Mailing List -
Kevin P. Fleming: The versions all go to ten. Look, right across the
board, ten, ten, ten and...
Asterisk Users: Oh, I see. And most open source projects upgrade to
two?
Kevin P. Fleming: Exactly.
Asterisk Users: Does that mean it's better? Is it any better?
Kevin P. Fleming: Well, it's eight
The Asterisk Development Team is pleased to announce the first beta release of
Asterisk 10.0.0-beta1. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
With the release of the Asterisk 10 branch, the preceding '1.' has been removed
from the
On Fri, 2011-07-22 at 10:58 -0700, Dave Platt wrote:
They've got a bunch of Grandstreams that seem to be rock solid... until
7:00pm. At 7:00, some of the phones become unavailable, and stay down.
Call
quality is solid almost all the time. But right at 7:00, things go bad.
Only
I have some terminals that have phone lines.
One of my tech had an idea of using IAXmodem or something similar to use
existing PRI/DAHDI Trucks for dial out via the asterisk/Linux console.
Anybody ever heard of doing this?
I would think maybe would use iaxmodem maybe and a shell
Hello,
I am wondering if there is a way to drop SIP packets for generic
transactions? For example, only SIP PEERs are allowed to call in and receive
ACK or Declined rather that those inviting a call who are not PEERs at all.
Currently my Asterisk setup sends, *SIP/2.0 603 Declined *to any
On 07/22/2011 07:32 PM, Bruce B wrote:
Hello,
I am wondering if there is a way to drop SIP packets for generic
transactions? For example, only SIP PEERs are allowed to call in and
receive ACK or Declined rather that those inviting a call who are not
PEERs at all.
Currently my Asterisk setup
On 07/22/11 18:13, William Stillwell wrote:
I have some terminals that have phone lines.
One of my tech had an idea of using IAXmodem or something similar to use
existing PRI/DAHDI Trucks for dial out via the asterisk/Linux console.
Anybody ever heard of doing this?
I would think maybe would
Thanks for the input. I am really surprised. But yes, I want exactly what
firewall does, DROP packet instead of REJECTING it.
So, you are saying that one has to tamper the SIP stack to add the option to
not respond to un-trusted sources?
I really thought Asterisk might have this built in as a
Asterisk does not expose low-level control of its SIP stack. It's something
intended to be configured and used at the application level.
If you really want to do this without a firewall, put a Kamailio proxy in front
of your Asterisk install and drop things as you see fit. But why go through
On 11-07-22 07:32 PM, Bruce B wrote:
Hello,
I am wondering if there is a way to drop SIP packets for generic
transactions? For example, only SIP PEERs are allowed to call in and receive
ACK or Declined rather that those inviting a call who are not PEERs at all.
Currently my Asterisk setup
Paul,
Won't that just send a 403 Forbidden?
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
On Jul 22, 2011, at 9:48 PM, Paul Belanger pabelan...@digium.com wrote:
On 11-07-22 09:51 PM, Alex Balashov wrote:
Paul,
Won't that just send a 403 Forbidden?
I believe so, but I was proposing a different SIP message then 603
Declined. As you mentioned, a firewall is the real solution if OP wants
to drop packets.
Asterisk is a B2BUA, not a firewall.
--
Paul
Robert thanks for weighing in.
So, you are saying that FreeSwitch on it's own can tackle issues like this
without the need of OpenSIPs? Can you elaborate please?
Thanks
On Sat, Jul 23, 2011 at 12:17 AM, Robert-iPhone rhuddles...@gmail.comwrote:
I like to put mine on 3389
hahaha just
50 matches
Mail list logo
