Anyone using the French DID provider num2sip? Could you share the
sip.conf setup?
Thanks,
sean
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
To deal with google dropping xmpp for voice, I've gotten a callcentric
number. The cc number connects to asterisk, and all works fine. Then I
set up the cc number as the gvoice forwarding number. If I'm on the
gvoice site, I can make a call and it will ring my cc number and then
the outside
On 04/26/2014 04:42 PM, Joshua Colp wrote:
Sean Darcy wrote:
I can't reach digium.com or asterisk.org. Did I miss the memo?
I have opened a ticket with IT. I'll keep the list apprised when the
problem is isolated and resolved.
Cheers,
Thanks.
Works fine today, FWIW.
sean
Asterisk-11.9.0, Fedora 20:
res_calendar_caldav.so = (Asterisk CalDAV Calendar Integration)
[Apr 27 10:49:13] ERROR[4255]: res_calendar_ews.c:911 load_module:
Exchange Web Service calendar module require neon = 0.29.1, but neon
0.30.0: Library build, IPv6, Expat 2.1.0, zlib 1.2.8, GNU TLS
On 04/27/2014 01:37 PM, Sean Darcy wrote:
On 04/26/2014 04:42 PM, Joshua Colp wrote:
Sean Darcy wrote:
I can't reach digium.com or asterisk.org. Did I miss the memo?
I have opened a ticket with IT. I'll keep the list apprised when the
problem is isolated and resolved.
Cheers,
Thanks
On 11.9.0:
-- Accepting AUTHENTICATED call from 111.xxx.yyy.zzz:
-- requested format = speex,
-- requested prefs = (),
-- actual format = ulaw,
-- host prefs = (silk16|ulaw|gsm|g722),
-- priority = mine
-- Executing
I can't reach digium.com or asterisk.org. Did I miss the memo?
sean
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
On 04/15/2014 06:52 PM, Kai-Uwe Jensen wrote:
Oops, had it wrong. Here's how it works for me:
[callcentric-template](!)
type=friend
context=from-callcentric
fromdomain=callcentric.com http://callcentric.com
defaultuser=1777xxx
fromuser=1777xxx
secret=password
insecure=port,invite
On 04/16/2014 05:42 PM, Josh Metzger wrote:
Try starting Asterisk with the -f option. It will NOT fork into the
background so you will see all messages on startup (including any that
might not end up in the log file). Search for DAHDI errors which will
likely be there.
Also, if you configure
On 04/14/2014 11:47 AM, Kelvin Chua wrote:
wild guess would be a conflict on host= setting.
there might be another entity on your sip.conf which have type=friend
and host=callcentric.com or host=204.11.192.161
Kelvin Chua
On Mon, Apr 14, 2014 at 8:01 AM, Sean Darcy seandar...@gmail.com wrote
On 04/15/2014 05:24 PM, Sean Darcy wrote:
On 04/14/2014 11:47 AM, Kelvin Chua wrote:
wild guess would be a conflict on host= setting.
there might be another entity on your sip.conf which have type=friend
and host=callcentric.com or host=204.11.192.161
Kelvin Chua
On Mon, Apr 14, 2014 at 8:01
On 11.9, trying to set up a callcentric peer:
sip debug:
--- SIP read from UDP:204.11.192.161:5060 ---
INVITE sip:1777myccid@10.10.11.180:5060 SIP/2.0
v: SIP/2.0/UDP
204.11.192.161:5060;branch=z9hG4bK-6104e46ef4249814d16a2ffb990d
f: sip:calling number@66.193.176.35;tag=3606475083-968127
Here's my cmd:
originate MOTIF/8447/+12122064...@voice.google.com extension s@greeting
Greeting:
[greeting]
exten= s,1,Wait(2)
same=n,Background(hello)
same=n,Wait(3)
I can see the call go out (also in, since testing on one our own
numbers), but [greeting] never executes.
I'm
I'm used to seeing fraudulent attempts to authenticate, But now I'm
getting them from the server itself.
I have an asterisk server behind a firewalled router. The local subnet
is 10.10.10.0/24, the server is 10.10.10.100.
Now I'm seeing in the log lots of:
Failed to authenticate device
On 10/08/2013 03:29 PM, Adrian Serafini wrote:
The qualify is on for the peer. It is failing to reply to the requested
SIP status. Maybe it is on wifi, screen goes off, wifi follows, zoiper
iax stack doesn't re-reg with the asterisk.
[Oct 8 18:14:14] NOTICE[510]: chan_iax2.c:11071
Using zoiper on a nexus 4, asterisk 11.5.1, sometimes we see failed
authentication. The secret seems correct, so we can't figure out why
we're getting failed authentication. But at the same time the device
shows as registered:
[Oct 8 18:14:14] NOTICE[510]: chan_iax2.c:11071
On 09/30/2013 12:09 PM, Sean Darcy wrote:
On 09/28/2013 11:11 AM, Asghar Mohammad wrote:
Hi,
If you post your configuration someone may help you.
On Sat, Sep 28, 2013 at 5:03 PM, Sean Darcy seandar...@gmail.com
mailto:seandar...@gmail.com wrote:
On 09/27/2013 09:08 PM, Sean Darcy wrote
On 09/28/2013 11:11 AM, Asghar Mohammad wrote:
Hi,
If you post your configuration someone may help you.
On Sat, Sep 28, 2013 at 5:03 PM, Sean Darcy seandar...@gmail.com
mailto:seandar...@gmail.com wrote:
On 09/27/2013 09:08 PM, Sean Darcy wrote:
We have zoiper connected over iax
On 09/27/2013 09:08 PM, Sean Darcy wrote:
We have zoiper connected over iax to asterisk in Sydney. The call is to
asterisk in New York. The caller in NZ can hear clearly. Nothing in NY.
Here's the sydney server:
-- Accepting AUTHENTICATED call from zoiperipaddr:
requested format
We have zoiper connected over iax to asterisk in Sydney. The call is to
asterisk in New York. The caller in NZ can hear clearly. Nothing in NY.
Here's the sydney server:
-- Accepting AUTHENTICATED call from zoiperipaddr:
requested format = speex,
requested prefs = (),
-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Darcy
Sent: Monday, September 09, 2013 7:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] iax2: two users can't authenticate from same ip
address
On 09/09
On 09/10/2013 05:27 PM, Joshua Colp wrote:
Sean Darcy wrote:
On 09/10/2013 12:15 PM, Joshua Colp wrote:
Sean Darcy wrote:
Maybe a different question would be helpful. Let's assume no NAT; the
server is directly connected with an FQDN. Two iax devices register.
Does asterisk assign them
On 09/10/2013 12:15 PM, Joshua Colp wrote:
Sean Darcy wrote:
Maybe a different question would be helpful. Let's assume no NAT; the
server is directly connected with an FQDN. Two iax devices register.
Does asterisk assign them different ports?
Asterisk does not assign ports. The IAX2 channel
On 09/09/2013 08:04 AM, Julian Beach wrote:
Hello Sean,
Sunday, September 8, 2013, 11:25:24 PM, you wrote:
The problem is that once a phone has used the server, no other phone can
use it. The servers sees all the phones as having the same ip address
(though different ports).
This sounds
On 09/09/2013 11:08 AM, Joshua Colp wrote:
Sean Darcy wrote:
On the server each device has type=friend.
I do notice that peer home has the standard iax port 4569. The other
peers are assigned 1026, 1027 and 1028. How are these ports assigned?
The actual configuration entries (minus password
On 09/09/2013 01:54 PM, Joshua Colp wrote:
Sean Darcy wrote:
home is from the home machine, which registers with the server:
register = home:pwhome@serverip
[home]
type=friend
insecure=port,invite
secret=pwhome ; same secret as on server
context=incoming
host=serverip
You aren't specifying
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Darcy
Sent: Monday, September 09, 2013 3:30 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] iax2: two users can't authenticate from same ip
address
Dial(IAX2/home-14358, IAX2/gn
I'm trying set up asterisk on an amazon instance in Sydney. It's to use
for our kids in Sydney to connect with their friends in the States.
We've found iax works better than sip with these distances. But we now
have weird problem: everybody has a cell phone, and it's much
cheaper/better to
On 09/07/2013 10:33 AM, Tony Mountifield wrote:
In article 522a934d.8010...@gmail.com,
Sean Darcy seandar...@gmail.com wrote:
On 09/06/2013 07:08 PM, Steve Edwards wrote:
On Fri, 6 Sep 2013, Sean Darcy wrote:
I'm not sure asterisk is even listening for the packets:
[root@asterisk
On 09/07/2013 01:26 PM, Tony Mountifield wrote:
In article l0fkfp$4ua$1...@ger.gmane.org,
Sean Darcy seandar...@gmail.com wrote:
On 09/07/2013 10:33 AM, Tony Mountifield wrote:
In article 522a934d.8010...@gmail.com,
Sean Darcy seandar...@gmail.com wrote:
On 09/06/2013 07:08 PM, Steve Edwards
I'm marching forward trying to get asterisk running on a amazon EC2
instance, Fedora 19.
If I start it from the terminal all works. I can login as user
asterisk and start asterisk.
But if I try to use systemctl to start it automatically I get the error
it doesn't have the permission to
I have 11.4.0 on an Amazon EC2 instance. SIP works fine, but I can't get
iax to work.
I've opened 4569 in the EC2 Security Group.
I'm using the zoiper client. Using tcpdump I can see the zoiper packets
coming in on 4569, but nothing shows on the asterisk cli.
Frame 33: 79 bytes on wire (632
On 09/06/2013 07:08 PM, Steve Edwards wrote:
On Fri, 6 Sep 2013, Sean Darcy wrote:
I'm not sure asterisk is even listening for the packets:
[root@asterisk ~]# netstat -apnt | grep 4569
[root@asterisk ~]#
'-t' meand TCP. IAX is UDP.
My bad:
netstat -apnu | grep 4569
udp0 0
Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but
no success:
[Jun 10 16:18:22] WARNING[4090][C-000a]: channel.c:6164
ast_channel_make_compatible_helper: No path to translate from
SIP/ng- to Motif/+12025551...@voice.google.com-da3c
[Jun 10 16:18:22]
On 06/10/2013 05:24 PM, Sean Darcy wrote:
Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but
no success:
[Jun 10 16:18:22] WARNING[4090][C-000a]: channel.c:6164
ast_channel_make_compatible_helper: No path to translate from
SIP/ng- to Motif/+12025551
On 06/10/2013 05:24 PM, Sean Darcy wrote:
Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but
no success:
[Jun 10 16:18:22] WARNING[4090][C-000a]: channel.c:6164
ast_channel_make_compatible_helper: No path to translate from
SIP/ng- to Motif/+12025551
I'm showing a lot of these on the console. I'm not using any database.
Where would this be coming from?
sean
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
and then play
the dtmf tones and bridge the call to your extension afterwards.
yves
Am 07.06.2013 17:51, schrieb Sean Darcy:
I'm trying to call a conference service, wait 10 seconds, then
send the passcode.
I've tried ww:
Dial(SIP/18005551212ww12345
I'm trying to call a conference service, wait 10 seconds, then send the
passcode.
I've tried ww:
Dial(SIP/18005551212ww12345#@sip.com,60,r)
The sip channel didn't like that. Added 'p' , still no help.
I tried D:
Dial(SIP/18005551...@sip.com,60,rD(12345#)
The dtmf is sent too soon. I
On 06/07/2013 01:17 PM, Yves A. wrote:
This would be possible with an agi...
the agi can wait for silence or 10 seconds, as u like and then play the
dtmf tones and bridge the call to your extension afterwards.
yves
Am 07.06.2013 17:51, schrieb Sean Darcy:
I'm trying to call a conference
On 05/16/2013 10:07 AM, sean darcy wrote:
On 05/16/2013 09:41 AM, sean darcy wrote:
I have a call on gv over motif. I try to bridge it to another call over
motif, but a different gv account, and I get congestion.
motif only handles one 1 channel at a time??
sean
More:
Two different motif
I have a call on gv over motif. I try to bridge it to another call over
motif, but a different gv account, and I get congestion.
motif only handles one 1 channel at a time??
sean
--
_
-- Bandwidth and Colocation Provided
On 05/16/2013 09:41 AM, sean darcy wrote:
I have a call on gv over motif. I try to bridge it to another call over
motif, but a different gv account, and I get congestion.
motif only handles one 1 channel at a time??
sean
More:
Two different motif sections. Two different xmpp sections.
xmpp
I've set up google voice to chat with me:
Forwards calls to:
me@gmail.com
and xmpp:
[general]
debug=no; Enable debugging (disabled by
default).
autoprune=yes ; Auto remove users from buddy
list. Depending on your
I rebooted our server Fedora 17 today, and now asterisk won't start;
asterisk[1063]: segfault at 0 ip 7f117aee122d sp 7fffbc398990
error 4 in libiksemel.so.3.1.1[7f117aed8000+d000]
iksemel is required for motif and xmpp.
I downloaded the iksemel source and rebuilt. No luck.
Any help
I know you that GV won't respect CALLERID from motif, but is there a way
have GV show Unknown on outgoing calls. I don't want to have people
think my GV number is really my number.
sean
--
_
-- Bandwidth and Colocation
Some calls I get from google voice, I just send myself an email about
the call and want to hangup. But I can't seem to make gv know I've hung up.
extensions.conf:
same = n,GoToIf($[${CALLERID(num)}=office]?email)
.
same = n(email),System(/usr/local/bin/emailme)
same =
On 03/07/2013 09:48 AM, Joshua Colp wrote:
sean darcy wrote:
Some calls I get from google voice, I just send myself an email about
the call and want to hangup. But I can't seem to make gv know I've
hung up.
extensions.conf:
same = n,GoToIf($[${CALLERID(num)}=office]?email
On 02/13/2013 09:39 AM, Matthew Jordan wrote:
On 02/12/2013 06:48 PM, sean darcy wrote:
On 02/12/2013 05:37 PM, Rusty Newton wrote:
Original Message -
From: sean darcy seandar...@gmail.com
Can I throw A and B into a confbridge and then add C? Create a new
channel that grabs
On 02/12/2013 05:37 PM, Rusty Newton wrote:
Original Message -
From: sean darcy seandar...@gmail.com
Can I throw A and B into a confbridge and then add C? Create a new
channel that grabs the A - B channel? Or is there a more straight
forward way to do this?
The Asterisk
I had motif working two days ago but now:
Executing [1171@internal:1] Dial(DAHDI/1-1, Motif/1171) in new stack
[Feb 12 20:56:18] ERROR[7794][C-0001]: chan_motif.c:1762
jingle_request: Unable to determine endpoint name and target.
motif.conf:
[11XX](!)
transport=google-v1
disallow=all
I'd like to have an extension join a call. That is, C can join A and
B, just as if it were an analog extension phone.
ChanSpy works, sort of. The problem is that once A or B hangs up, the
channel is gone. With an analog extension, C would remain connected with
B if A hung up.
Can I throw A
On 12/11/2012 10:12 PM, Mitul Limbani wrote:
snom m9 dect ip
But it's 2-3 x the price!
sean
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
I have an asterisk server at home. I'm looking to replace my internal
phones with sip cordless (DECT) phones. I'm now looking at the Siemens
A510IP base ($90 ) and A510H handset ($40) and the Grandview DP715 base
($80) and DP710 handset ($45).
The Siemens has a feature were I can also use a
On 12/11/2012 04:37 PM, Roy Abshire wrote:
That is true about the A580.
I don't like the interface much to check messages.
Besides that every time I go to dial a number...it always uses the first
digit pressed to go into phone mode..so I have to press the first digit
twice...
I would test
I'd like to see on cli what happens on executing remote commands. For
instance:
asterisk -rx originate
Motif/gvoice/12026668...@voice.google.com,,rL(5000)) extension s@default
Now I get on cli, verbose 10:
-- Remote UNIX connection
-- Remote UNIX connection disconnected
Any way to see
On 11/06/2012 09:45 PM, Michael L. Young wrote:
- Original Message -
From: sean darcy seandar...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Tuesday, November 6, 2012 7:51:04 PM
Subject: [asterisk-users] 11.0.1: more sip registry woes
Upgrade to 11. This worked on 10.X.X
Upgrade to 11. This worked on 10.X.X
sip.conf:
register=myusername:password@nyc.teliax.net
telnet nyc.teliax.net 5060
Trying 8.14.120.23...
Connected to nyc.teliax.net.
Escape character is '^]'.
sip show registry
Hostdnsmgr Username Refresh
State
On Wed, Oct 10, 2012 at 8:06 AM, Deepesh D deep.d2...@gmail.com wrote:
Hello,
How do I use the asterisk application 'Transfer' to transfer a SIP
call from one asterisk to another?
I have the following scenario. I have two asterisk servers S1 and S2.
There is a third asterisk server C1 which
On 10/09/2012 07:40 AM, Steve Underwood wrote:
On 10/09/2012 12:28 AM, Brett Lehrer wrote:
How many fax and voice calls (which codecs for tha latter ones ?) are on
average using your DSL line ?
1. Previously, I experienced failures during the process of converting
incoming PDF documents into
On 10/08/2012 05:15 PM, Asterisk Development Team wrote:
The Asterisk Development Team is pleased to announce the first release candidate
of Asterisk 11.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
All interested users
10.9.0. I'm trying to have a setup where hitting # sends the called
party to the confbridge. I've set GOTO_ON_BLINDXFR:
CLI dialplan show globals
.
GOTO_ON_BLINDXFR=tel-incoming^confbridge^1
(Also tried tel-incoming,confbridge,1 and using | )
but it doesn't work:
Dial(DAHDI/1-1,
So here's what I used:
$['x${CALLERID(num)}'='x2024324321']
And that worked!
On 10/05/2012 08:28 AM, Richard Kenner wrote:
I'm getting a parsing error with the folllowing:
same=n,GoSubIf($[${CALLERID(num)} = 2024324321]?other,1($
{thisexten}):)
WARNING[11356]:
I'm getting a parsing error with the folllowing:
same=n,GoSubIf($[${CALLERID(num)} = 2024324321]?other,1(${thisexten}):)
WARNING[11356]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '=', expecting $end; Input:
= 2024324321
^
[Oct 4 21:53:35]
I'm building asterisk 11 beta 2. I've been using silk a lot. I don't see
silk listed in menuselect as a codec. But I also don't see an asterisk
11 silk codec on http://downloads.digium.com/pub/telephony/codec_silk.
Do we use the asterisk 10 codec_silk.so ?
sean
--
On 09/25/2012 11:49 AM, Jonathan Rose wrote:
Jonathan Rose wrote:
Sean Darcy wrote:
I'm building asterisk 11 beta 2. I've been using silk a lot. I
don't
see
silk listed in menuselect as a codec. But I also don't see an
asterisk
11 silk codec on
http://downloads.digium.com/pub/telephony
I've installed 10.6.0-rc2 on two machines. On one of the machines (but
not the other) /tmp gets filled with:
...
-rw---. 1 asterisk asterisk 53661696 Jul 7 23:46
core.PBX-2012-07-07T23:46:10-0400
-rw---. 1 asterisk asterisk 53891072 Jul 7 23:48
On 07/10/2012 11:44 AM, Matthew Jordan wrote:
- Original Message -
From: sean darcy seandar...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Tuesday, July 10, 2012 10:42:20 AM
Subject: [asterisk-users] 10.6.0-rc2: tmp full of core.PBX
I've installed 10.6.0-rc2 on two machines
, 2012 at 3:21 PM, sean darcy seandar...@gmail.com
mailto:seandar...@gmail.com wrote:
[home_outgoing]
type=friend
transport=tcp
secret=
fromuser=office_incoming
host=dynamic
disallow=all
allow=ulaw
It's because you're using fromuser as your username setting
I'm trying to set the callerid on a SIP call:
same=n,Set(CALLERID(all)=test2023214321)
same=n,Dial(SIP/home_outgoing/150)
-- Executing [202454@from-test-sip:3] Set(SIP/sip-test-0019,
CALLERID(all)=test2023214321) in new stack
-- Executing [202454@from-test-sip:4]
On 04/20/2012 02:05 PM, Asterisk Development Team wrote:
The Asterisk Development Team has announced the releases of:
DAHDI-Linux 2.6.1
DAHDI-Linux 2.5.1
DAHDI-Tools 2.6.1
DAHDI-Tools 2.5.1
DAHDI-Linux-Complete 2.6.1+2.6.1
DAHDI-Linux-Complete 2.5.1+2.5.1
These releases are
On 04/21/2012 12:00 PM, Shaun Ruffell wrote:
On Sat, Apr 21, 2012 at 10:26:30AM -0400, sean darcy wrote:
On 04/20/2012 02:05 PM, Asterisk Development Team wrote:
The Asterisk Development Team has announced the releases of:
DAHDI-Linux 2.6.1
DAHDI-Linux 2.5.1
DAHDI-Tools 2.6.1
DAHDI
, sean darcy wrote:
We found this morning we had no SIP connection to another site. sip
show registry on the main site gave no authentication. sip show
peers on the other site showed the peer unspecified.
The odd part about this: doing sip reload on the main site made it all
work. Nothing else
We found this morning we had no SIP connection to another site. sip show
registry on the main site gave no authentication. sip show peers on
the other site showed the peer unspecified.
The odd part about this: doing sip reload on the main site made it all
work. Nothing else was changed.
On 04/09/2012 08:51 PM, Barry Miller wrote:
On Mon, Apr 09, 2012 at 06:21:40PM -0400, sean darcy wrote:
I've cut and pasted from the digium fax admin manual:
exten = send,1,NoOp( SENDING FAX )
exten = send,n,Wait(6)
exten = send,n,Set(GLOBAL(FAXCOUNT)=$[ ${GLOBAL(FAXCOUNT)} + 1
On 04/07/2012 04:20 PM, Noah Engelberth wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Saturday, April 07, 2012 4:10 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users
Trying to use gtalk:
-- Executing [andy@ipkall:2] Dial(SIP/ipkall-,
gtalk/andy-gtalk/+1xxxyyyz...@voice.google.com) in new stack
[Apr 1 10:41:53] ERROR[2416]: chan_gtalk.c:1934 gtalk_request: No XMPP
client to talk to, us (partial JID) : andy-gtalk
gtalk.conf
[general]
I'm setting up res_fax to use with an iax provider. I'm calling over
PSTN to the provider. When I stand at our fax machine (Brother), I can
see the call come in, and it appears to set up correctly. What is odd,
however, is that asterisk drops off while the fax machine is still
sending. I've
traffic originated by your provider
while happily NATing the traffic originated by your Asterisk.
It is also a good idea to have qualify=yes in your SIP peers' settings
to keep these NAT tables on the firewall updated for incoming SIP traffic.
-Vladimir
On 3/9/2012 9:15 PM, sean darcy wrote
I'm trying to move the asterisk server to an Amazon Web instance. We
have teliax for our sip provider. I'd like for our DID lines to be
connected to a users cell phone.
Seems simple enough, but I'm getting the dreaded one-way audio, even
with nat=yes everyplace I can think of.
The dialplan
On 03/09/2012 04:16 PM, sean darcy wrote:
I'm trying to move the asterisk server to an Amazon Web instance. We
have teliax for our sip provider. I'd like for our DID lines to be
connected to a users cell phone.
Seems simple enough, but I'm getting the dreaded one-way audio, even
with nat=yes
, sean darcy wrote:
I'm trying to move the asterisk server to an Amazon Web instance. We
have teliax for our sip provider. I'd like for our DID lines to be
connected to a users cell phone.
Seems simple enough, but I'm getting the dreaded one-way audio, even
with nat=yes everyplace I can think
panel?
Sent from my iPhone
On Mar 10, 2012, at 7:16 AM, sean darcyseandar...@gmail.com wrote:
On 03/09/2012 04:16 PM, sean darcy wrote:
I'm trying to move the asterisk server to an Amazon Web instance. We
have teliax for our sip provider. I'd like for our DID lines to be
connected to a users cell
On 02/29/2012 02:30 AM, Zohair Raza wrote:
You want to allow single IP or whole subnet ?
Regards,
Zohair Raza
On Wed, Feb 29, 2012 at 4:44 AM, sean darcy seandar...@gmail.com
mailto:seandar...@gmail.com wrote:
An outside device can't register:
WARNING: getnameinfo(): ai_family
An outside device can't register:
WARNING: getnameinfo(): ai_family not supported
WARNING: chan_sip.c:14456 parse_register_contact: Domain
'69.xxx.yyy.zzz:5060' disallowed by contact ACL (violating IP )
sip.conf:
[general]
...
alwaysreject=yes
dynamic_exclude_static = yes
allowguest=no
On 02/17/2012 03:28 AM, Frank Church wrote:
Should a Linksys Sipura 2102 be configured with nat=yes even if it is on
the local network?
I have been having some troubles with a Linksys Sipura 2100 series,
which suffers from NO AUDIO after a few calls.. Because it is on the
same subnet as
On 02/16/2012 12:30 PM, Kevin P. Fleming wrote:
On 02/11/2012 06:59 PM, Bruce B wrote:
If your server is open to the internet and in SIP general section you
have nat=no and in peers you have nat=yes or vice versa then it's
possible to enumerate your extension. Because Asterisk responds with
I've been lurking on the dev discussion on creating nat=auto. It all
leads me to think there's no reason to use nat=no.
We have about 60 internal sip extensions connected to an multihomed
asterisk box where the external ip is not nat'ed. Each of the internal
sip contexts has nat=no. On
Nope. No templates. And sterisk is running as root.
sean
On 01/26/2012 10:50 PM, Jim DeVito wrote:
Are you by chance using templates (!) In your sip.con? Ive had access
denied errors befor when running as non root.
- Original message -
I've just upgraded from 1.8.8.0 to 10.1.0-rc2.
I've just upgraded from 1.8.8.0 to 10.1.0-rc2. Now I'm getting a flood of:
WARNING[5100]: db.c:295 ast_db_put: Couldn't execute statment: SQL logic
error or missing database
AFAIK, I'm not doing any database puts (or gets). There were no such
warnings in 1.8.8.0.
What do I need to do to
On 01/06/2012 05:00 PM, Tom Poe wrote:
Just installed asterisknow 1.6. I can access freepbx. I need to test
system on my LAN. Which softphone is best to use? I'm running ubuntu on
Dell optiplex G260 desktop at home. I'm hoping to setup basic IP PBX for
incoming/outgoing calls. No video.
Tom
On Sat, Jan 7, 2012 at 9:34 AM, Gilles codecompl...@free.fr wrote:
On Sat, 07 Jan 2012 09:27:29 -0500, sean darcy seandar...@gmail.com
wrote:
But what really made us choose linphone was you use it on android/iphone.
That has been a huge plus. As a bonus, you can use any degegistered
smartphone
On 1/4/2012 2:26 PM, Lefteris Zafiris wrote:
Works beautifully. Amazing job Lefteris. Thanks.
The best result I got in probability was 0.9725632 by saying, hello. I
think there is some non-phonetic logic built-in as well. I tried, 1, 2 and
I got 0.86534226 in accuracy. While I tried 1, 2, 3,
On 1/4/2012 4:37 AM, Jayesh Labade wrote:
Please help me..
Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com mailto:jayesh.lab...@gmail.com
On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.com
mailto:jayesh.lab...@gmail.com wrote:
Hello Experts,
I have
On 01/01/2012 11:34 PM, sean darcy wrote:
I'm trying to setup a simple tcp sip connection based on the toronto
osaka example in the Asterisk book.
On the remote box (osaka) (1.8.9.0-rc1):
[toronto]
type=friend
transport=tcp
secret=welcome
context=toronto_incoming
host=dynamic
disallow=all
On 01/02/2012 11:21 AM, sean darcy wrote:
On 01/01/2012 11:34 PM, sean darcy wrote:
I'm trying to setup a simple tcp sip connection based on the toronto
osaka example in the Asterisk book.
On the remote box (osaka) (1.8.9.0-rc1):
[toronto]
type=friend
transport=tcp
secret=welcome
context
On 01/02/2012 11:30 AM, sean darcy wrote:
On 01/02/2012 11:21 AM, sean darcy wrote:
On 01/01/2012 11:34 PM, sean darcy wrote:
I'm trying to setup a simple tcp sip connection based on the toronto
osaka example in the Asterisk book.
On the remote box (osaka) (1.8.9.0-rc1):
[toronto]
type
I'm trying to setup a simple tcp sip connection based on the toronto
osaka example in the Asterisk book.
On the remote box (osaka) (1.8.9.0-rc1):
[toronto]
type=friend
transport=tcp
secret=welcome
context=toronto_incoming
host=dynamic
disallow=all
allow=ulaw
sip show peer toronto
* Name
Trying to set up a simple sip - sip connection over tcp. Home : 10.0.0 -
Office: 1.8.8.0
Home sip.conf:
register =
tcp://office-going-to-home:password@office-ipaddr/home-coming-from-office
[home-coming-from-office] ; receives calls
type=friend
transport=tcp
dtmfmode=rfc2833
On 12/26/2011 10:05 PM, sean darcy wrote:
I've now set up tcp to connect for some home-office connections.
Home is 10.0.0, office is 1.8.8.0.
The home sip device is home-going-to-office, office device:
office-coming-from-home - home ip is 10.10.11.180
-- Called SIP/home-going-to-office/166
101 - 200 of 488 matches
Mail list logo
