We have a need for up to a dozen UK mobile numbers to be forwarded to
a UK landline. I know that I can just forward them, but was wondering
if anyone knew of any deals / contracts with a UK mobile operator that
would lessen the cost.
At the moment we are looking at going with Vodafone .
Thanks
Hello,
I need to capture calee's audio in real-time in order to capture operator
messages (I've written sound recognition software that works with Jack:
http://github.com/Motiejus/SoundPatty/).
Jack does the following:
Incoming call audio - audio in to jack, audio out from jack -
current Asterisk
Hello,
I saw that Asterisk don't calcultate fine the ANSWEREDTIME for me.
I want that when ANSWEREDTIME =~ 5.6 become 6 and if ANSWEREDTIME= 10.3
become 10
because, now, if ANSWEREDTIME =~ 15.9, it become 15! it isn't correct
I could manipulate the app_dial.c to have my own result.
But do
Update:
I thought this may be the solution:
*CLI core set chanvar SIP/poly1-ab23jadf234 JACK_HOOK(manipulate) on
(For 1.6.2 it's *dialplan*set chanvar SIP/poly1-ab23jadf234
JACK_HOOK(manipulate) on)
Source: voip-info.org%20http://www.voip-info.org/wiki/view/Asterisk+cmd+jack
The command opens
Hello,
I am working on getting the following to work and I couldn't find it in the
documentation I did read. Where should I look or does someone have an
example how I can do it?
Current situation:
Incoming call - 3 SIP phones + 2 mobile phones ring - if mobile phone goes
to voicemail the call is
Hi guys,
Having issue with getting CDR to write to MS-SQL via ODBC.
cdr_odbc: Connected to freetds-connector
cdr_odbc: Error in PREPARE -1
cdr_odbc: Query FAILED Call not logged!
== Spawn extension (cisco, ##, 2) exited non-zero on
'IAX2/ast-507
Isql
Mark Scholten wrote:
Situation I want:
Incoming call - 3 SIP + 2 mobile phones ring - if the call is answered by
a mobile phone the person picking up the call needs to press 1 (or another
http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe
Doug
--
Ben Franklin quote:
Those who
and it happened again, I've attached kernel logs from dahdi restart on paste-bin
http://pastebin.com/drg3WD20
to fix this problem, I have to:
stop dahdi and asterisk - /etc/init.d/dahdi stop
remove all dahdi modules - rmmod wct4xxp, dahdi_echocan_mg2, dahdi
load modules - modprobe dahdi,
Hello list,
I am trying to solve a problem and after unsucessfully chasing forums
and google for some hours, I turn to you in hope of a solution. I feel
it's just a configuration issue but I just can't get my head wrapped
around it.
The situation is basically this: I have an Asterisk
I have two Asterisk boxe. One is running 1.6 and the other 1.2
The users on the 1.2 system press # plus a local 7 digit number to place
local calls through the trunk to the 1.6 box.
For some reason this dial pattern fails right away with unavailable. There
is no activity in the CLI. Other
Hi list,
Anyone have successfully compiled amr codec for asterisk 1.6.1.X ?
I still have no problem compiling and playing with it on Asterisk 1.4.X.
I have used the following patch :
https://asteriskvideo.svn.sourceforge.net/svnroot/asteriskvideo/amr/
Hare is what i get while loading
Which 1.6 are you running? I dropped my 1.6.1.6 back to 1.4.30 because my
other 2 1.4.30 boxes wouldn't talk to it properly.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 8:23
Thank you Danny, but it says in the link that it's an iptables issue, though
i allowed everything on this network interface and even stopped iptables but
still i have this issue.
2010/5/4 Danny Nicholas da...@debsinc.com
See if this helps
http://www.voipuser.org/forum_topic_3921.html
I am sorry it is not 1.6 but 1.4.30 (argh.. to many boxes)
The other box is running 1.2.1
Thanks,
David
On Wed, May 5, 2010 at 9:28 AM, Danny Nicholas da...@debsinc.com wrote:
Which 1.6 are you running? I dropped my 1.6.1.6 back to 1.4.30 because
my other 2 1.4.30 boxes wouldn’t talk to it
Set verbose to 5 and see if you get a CLI output.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Nickel
Sent: Wednesday, May 05, 2010 8:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hi all,
I've just bought some SPA922. First time with this hardware for me.
I see no LAN tab in its web GUI where I can setup NAT for PC conected to its
LAN ethernet port.
However, when I connect a PC to that port, SPA922 works as bridge.
Anybody can confirm SPA922 can NAT a PC connected to its
This is a little over my head, but the message indicates that you don't have
a fully authorized connection. Can you post the iax.conf snippets relevant
to the call?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of khalid
Hello list,
as I am trying to write a complex macro for my users i have the problem,
that the appdata field in the extensions table is to small for all my macro
parameters.
I am using the DB definition from
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions
so appdata is limited
Nothing..goes directly to The person you are calling is unavailable.
On Wed, May 5, 2010 at 9:46 AM, Danny Nicholas da...@debsinc.com wrote:
Set verbose to 5 and see if you get a CLI output.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
Ok - you have to be getting something or you wouldn't get that message. You
are looking at CLI on the 1.2 or 1.4 box? If you're looking at the 1.4 side,
you won't see anything until a connection is made (although you should see
some kind of credential reject or something??)
_
From:
Hi Guys,
first of all, thanks Danny for your support trying to help is a big help
itself.
so the thing is:
from pbx1 to pbx2 which was able to leave VM, it was set up like this:
exten = 8021,1,Dial(IAX2/pbx2/${EXTEN},30,tTWwr)
but from pbx2 to pbx1 which was not able to leave VM, it was setup
On 5 May 2010, at 14:39, Sebastian Milioto wrote:
However, when I connect a PC to that port, SPA922 works as bridge.
Anybody can confirm SPA922 can NAT a PC connected to its LAN port? Does exist
such LAN tab for setting up parameters as port forwarding?
(by the way, version is 5.1.15(a).
I am on the 1.2 box and see nothing with the verbose cranked up. I do see
the following when tailing the asterisk full log during the calls:
May 5 11:09:46 DEBUG[26538] chan_sip.c: Setting NAT on RTP to 0
May 5 11:09:46 DEBUG[26538] chan_sip.c: Checking SIP call limits for device
3000
May 5
On Wednesday 05 May 2010 06:51:48 Neeraj Chand wrote:
I can connect to the database and run via isql, and also use func_odbc,
etc with res_odbc configured with the same database / freetds, but I
cannot write CDRs.
Are you writing to the database with func_odbc, or just reading? My gut says
However, when I connect a PC to that port, SPA922 works as bridge.
Exactly. The SPA9x2 has a 2-port switch; no NAT, no routing (unlike
the SPA2102, etc).
I think the 5.1 series is the latest firmware for the 922; the the
942, there is 6.1.5a.
Luki
--
In my system (Asterisk 1.4.30) I found that if I have some playback() or
saydigit() before dial(), the billsec in CDR count all the time includes
the playback time. For example, if I dial a number, listen the playback,
then just hangup before the call get answered, the CDR show me the time
== Registered translator 'amrtolin' from format unknown to slin, cost 4000
== Registered translator 'lintoamr' from format slin to unknown, cost 32002
Probably shouldn't be listing it as unknown
Have you tried using that AMR codec beyond commands in the asterisk cli?
Did the patch apply
It says in the readme from that link you provided:
This patch adds AMR-NB support to Asterisk 1.4
(for Asterisk 1.6 check out asterisk 1.6 branch and use the
asterisk-1.6-AMR.patch patch (provided by Ivelin Ivanov))
Did you use the 1.6 branch and patch ??
I'll have to
Anyone have any experience with a Japanese local VoIP termination
supplier?
I've emailed a few companies looking to setup some PSTN to SIP and SIP
to PSTN termination, but no luck so far.
Thanks,
Adrian
--
_
--
Ok. I'm confused. I was interpreting what you wrote to say that you are
doing this:
1. pick up sip phone attached to pbx1 (1.2 box)
2. dial #5551212
3. command dial(iax2/trunk/5551212,30,r) gets executed on 1.2 box
4. 1.4 box should fall into _XXX and do DAHDI dial?
Maybe a rtp.conf problem - normal values are 1-2.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Lamanna
Sent: Wednesday, May 05, 2010 12:01 PM
To: Asterisk Users Mailing List - Non-Commercial
Hi,
I'm having a problem trying to get a Cisco 7965 phone registered on
Asterisk 1.4.26.
As we know, Cisco now, for security reasons, has made the phone ports
non-symmetric, in that it sends out UDP requests on a high port and
receives them on a different port.
It seems that, even with 'nat' set
DeadAGI is deprecated in Asterisk 1.6.x !
2010/4/9 Danny Nicholas da...@debsinc.com
Do the call in a context and have the context run the script as a
DeadAGI.
[call_and_do]
- exten = s,1,Dial…
- exten = h,1,Deadagi(…)
--
*From:*
On Wed, May 5, 2010 at 10:16 AM, Danny Nicholas da...@debsinc.com wrote:
Maybe a rtp.conf problem - normal values are 1-2.
I haven't even gotten to the RTP stage, it won't even register on the SIP side
because responses are being sent back to the wrong SIP signaling port.
-- James
Regular AGI with SIGHUP detection?
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DeadAGI
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mickael
Monsieur
Sent: Wednesday, May 05, 2010 12:36 PM
To: Asterisk
probe with this:
www.siptraffic.com
Our company have a lot of experience with this company through a lot of
routes and simply they are the best we know in quality/price rate.
They ask you for 200$ initially, but they work perfectly! The only problem
is that they never give a CID number.
If you
On 05-05-2010 18:00, Jian Gao wrote:
In my system (Asterisk 1.4.30) I found that if I have some playback() or
saydigit() before dial(), the billsec in CDR count all the time includes
the playback time. For example, if I dial a number, listen the playback,
then just hangup before the call get
Hi List,
If we have a scenario where a customer is using a telephone and their WAN
link goes down for example the channel in asterisk stays marked as in use
and this affects the subscribe also.
*CLI core show channels
Channel Location State Application(Data)
Hi all,
I am trying to connect to a softphone application using an Iax channel on
Asterisk 1.4.30. I can do outbound calls, from softphone to asterisk, but
not inbound from asterisk to softphone.
I get the following Debug:
--
I have a question about the blind transfer using ##. This works great on our
cordless phone, but there have been occasions that we can't transfer using
##. I was able to reproduce the issue by doing the following:
1) Call in from the outside line,
2) Ask the operator to transfer me to an
Hi list!
I have this configuration for sending T38 faxes to my T38 fax termination
provider:
T38modem -- hylafax -- Asterisk-SIP-Extension -- T38 termination provider
-- T.30 termination to PSTN
We are experiencing 2 problems with this (if you want configuration files,
it won't be a problem,
Your interpretation is right ownvery weird problem. The problem is when
i dial #551212 there is absolutely no activity in the CLI. It is almost like
there is a conflict somewhere.
On Wed, May 5, 2010 at 12:40 PM, Danny Nicholas da...@debsinc.com wrote:
Ok. I’m confused. I was
From 1.2 CLI, do dialplan show _...@default - this will tell you if your
expected context is valid (may not work on 1.2, I started this ride at 1.4
and therefore have no backward knowledge).
_
From: asterisk-users-boun...@lists.digium.com
I set: sip debug peer 3000 (my test extension) and dialed #3643873
Here is the output:
-- SIP read from 192.168.1.59:17456:
INVITE sip:%233643...@192.168.2.10 sip%3a%25233643...@192.168.2.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.59:17456
;branch=z9hG4bK-d8754z-e210db3acd1ed62b-1---d8754z-;rport
James,
I'm assuming your talking SIP here.
It's not the Contact header that is important here but the Via header.
Responses should be going back to whatever port is specified there.
Contact header is used for incoming requests.
Via header is used for responses to outgoing requests.
7965:
2010/5/4 sean darcy seandar...@gmail.com
On 5/4/2010 7:32 AM, Miguel Amez wrote:
App_fax? I didn't hear about that. What's that?
Could you please explain that a little bit better?
I'm experiencing some troubles with T38modem and would like to solve on
the better way.
regards,
ad...@3a.hu wrote:
On 05-05-2010 18:00, Jian Gao wrote:
In my system (Asterisk 1.4.30) I found that if I have some playback() or
saydigit() before dial(), the billsec in CDR count all the time includes
the playback time. For example, if I dial a number, listen the playback,
then just
It doesnt seem to like the _X. . What is this suppose represent?
Thanks
On Wed, May 5, 2010 at 5:53 PM, Danny Nicholas da...@debsinc.com wrote:
From 1.2 CLI, do “dialplan show _...@default – this will tell you if your
expected context is valid (may not work on 1.2, I started this ride at 1.4
I've got two 1.6.2 asterisk boxes. I'd like to be able to set up two
separate sip connections. But when I try that I get:
chan_sip.c:12671 check_auth: username mismatch, have one-sip-peer,
digest has another-sip-peer
Looking around I found this in a 2007 bug report on version 1.4.4,
Hi!
apps like playback do an implicit answer and this fires up the billsec
counter.
OK, here is my dialplan:
exten = _011X.,n,Playback(this-call-will-end-in)
exten =
_011X.,n,Dial(SIP/${ext...@${ldtrunk1},60,L(${ms}:3))
Is there any way that Asterisk will record the correct
Hi!
I set: sip debug peer 3000 (my test extension) and dialed #3643873
Your X-Lite softphone actually calls %233643873 and not #3643873.
You would need to check the SIP RFCs in order to find out if Asterisk is
behaving correctly here by not decoding %23 as #.
In the meanwhile you could try
On Wed, May 5, 2010 at 5:05 PM, David White david.wh...@watchguard.comwrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com on behalf of James Lamanna
Sent: Wed 5/5/2010 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users]
Are there any CLI commands to free this up or any other ways without having
to restart asterisk.
Did you try soft hangup channel? Or set an RTP timeout to avoid
abandoned channels?
Luki
--
_
-- Bandwidth and Colocation
---
Message: 10
Date: Wed, 5 May 2010 10:26:34 -0500
From: Tilghman Lesher tles...@digium.com
Subject: Re: [asterisk-users] CDR to MS-SQL via ODBC issue
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Wednesday 05 May 2010 18:29:26 Neeraj Chand wrote:
---
Message: 10
Date: Wed, 5 May 2010 10:26:34 -0500
From: Tilghman Lesher tles...@digium.com
Subject: Re: [asterisk-users] CDR to MS-SQL via ODBC issue
To:
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