G.Day!
Thanks for the response!
i've tryed to do this, but in /var/spool/hylafax/log/xferfaxlog
I read this:
09/06/11 09:04 CALL00108 ttyIAXfax
+39.06.456789 0 0 0:00:09 0:00:09 Failure to
receive silence (synchronization failure).
Hi,
I'm looking for a PCIe card with 1 ISDN2 connection which works with Asterisk
Could anybody give me an advise which card I can use?
Regards,
Arjan Kroon
Mobillion.
--
_
-- Bandwidth and Colocation Provided by
what do you mean exactly?! One what?! What do you plan to accomplish?!
Do you mean a 1 Port ISDN BRI Board?! Difficult to find, and thus boards
are really expensive, not under 400.- € inkluding DSP Processors.
I advise you taking Gentoo Linux, getting asterisk on it and put a
single Port HFC-S
Hi Tamar,
Yes, I mean 1 Port ISDN BRI PCIe board.
We need an PCIe board, because the board don't provide PCI slots, only PCIe
slots.
It doesn't matter which distribution we use.
But I will look at Sangoma.
Best Regards,
Arjan
Mobillion BV
-Oorspronkelijk bericht-
Van:
On 09/06/2011 09:08 AM, Arjan Kroon | Mobillion wrote:
Hi,
I'm looking for a PCIe card with 1 ISDN2 connection which works with Asterisk
Could anybody give me an advise which card I can use?
Both Sangoma and Digium have PCIe ISDN cards although a single BRI port
might be a bit of a
Hi list,
I have 20channels PRI line from Airtel. and I have 30channels digium PRI
card. and I am using asterisk1.4 with goautodial.
I need to configure DID's for every extension with sip.
which Parameters need to add in asterisk and which parameters enable from
Airtel PRI line for this DID's. can
On 09/05/2011 03:05 AM, Hans Witvliet wrote:
On Fri, 2011-09-02 at 11:33 -0500, Kevin P. Fleming wrote:
On 09/01/2011 04:39 PM, Hans Witvliet wrote:
From the asterisk-bible and the wiki's i learned that it is possible to
let asterisk do some of the presense-info by means of the jabber.conf
On 04/09/11 02:51 PM, Tamer Higazi wrote:
the 3rd edition is available, but that book covers every thing to run
the asterisk PBX.
You can read the 3rd edition online at
http://ofps.oreilly.com/titles/9780596517342/
HTH!
Leif.
--
Leif Madsen
http://www.oreilly.com/catalog/asterisk
--
On Tuesday 06 September 2011, mahesh katta wrote:
Hi list,
I have 20channels PRI line from Airtel. and I have 30channels digium PRI
card. and I am using asterisk1.4 with goautodial.
I need to configure DID's for every extension with sip.
which Parameters need to add in asterisk and which
2011/9/6 Leif Madsen leif.mad...@asteriskdocs.org
On 04/09/11 02:51 PM, Tamer Higazi wrote:
the 3rd edition is available, but that book covers every thing to run
the asterisk PBX.
You can read the 3rd edition online at http://ofps.oreilly.com/**
titles/9780596517342/
Thanks For reply A.J sir.
no result with this sir
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone
On Tuesday 06 September 2011, mahesh katta wrote:
Thanks For reply A.J sir.
no result with this sir
OK. no result isn't very helpful.
I'm presuming you've got internal calls between SIP extensions working
correctly, and something happens (even if it's not exactly what you want)
when you
On Tue, Sep 06, 2011 at 01:05:02PM +0200, Patrick Lists wrote:
Both Sangoma and Digium have PCIe ISDN cards although a single BRI port
might be a bit of a challenge:
[snip]
You could also find an Intel (formerly Eicon) Diva Server card and use
it with chan_capi (must be a Server card,
Hello list
i want to use pickup with sip and astersik 1.4
i configured all the inbound calls in 1 sip phone 224 and want to pickup the
calls using 222 SIP
Could you please see the code below and tell me what is wrong
NB when i make *8+ok i can pickup the call but i want to specify the
On Mon, Sep 05, 2011 at 02:41:50PM +0200, Jonas Kellens wrote:
I read some information and examples on the net, but they all show how
you login to the AMI, give an action and receive a response. The end.
I guess you just re-run the script every time you want the action to be
executed.
How
On Tue, Sep 06, 2011 at 04:43:39PM +, salaheddine elharit wrote:
[asterisk 1.4]
[agents]
exten = _2XX,1,Dial(SIP/${EXTEN})
exten = _*8XXX,1,PickupChan(SIP/${EXTEN:2})
SIP/222 is not a channel but an extension. See:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup
--
Daniel
ok thanks for you response
how can i do in order to fix this issue
regards
2011/9/6 Daniel Tryba dan...@tryba.nl
On Tue, Sep 06, 2011 at 04:43:39PM +, salaheddine elharit wrote:
[asterisk 1.4]
[agents]
exten = _2XX,1,Dial(SIP/${EXTEN})
exten = _*8XXX,1,PickupChan(SIP/${EXTEN:2})
Hello Leandro,
Can you tell me a short example about how can i use what you gave me for
instance suppose i want to use { txjitter, DBL, { .d8 =
stats.txjitter, }, }, how can i set it in CDR variable like mine:
exten = h,n,set(CDR(ljitt)=${CHANNEL(rtpqos,audio,local_jitter)})
Thank
I'm having annoying errors trying to get configure working.
tar xvzf /usr/local/src/asterisk-1.8.6.0.tar.gz
cd asterisk-1.8.6.0
./configure
I get complaints related to pwlib / ptlib...
checking for openr2_chan_new in -lopenr2... no
checking /root/pwlib/include/ptlib.h usability... no
checking
On 09/06/2011 03:08 PM, David Backeberg wrote:
There seems to also be a problem with CentOS 6 in general that I have
not found a package that actually provides /usr/bin/ptlib-config. I
copied that binary over from a CentOS 5 install to see if I could get
my original error to clear.
Here's THAT
On 02/09/11 11:27 PM, Joseph wrote:
In asterisk 1.4 I had:
exten = s,n,Answer()
exten = s,n,SetMusicOnHold(default)
But in 1.6 1.8 I think don't need to use: SetMusicOnHold(default)
(beside it is deprecated) as it is default.
In 1.6 and UP I think it is: Set(CHANNEL(musicclass)= Can somebody
First, have you tried ./configure --help?
Next try the --with-pwlib parameter
Somewhere in the list, make sure your YUM paths are happy.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent:
On Tue, Sep 6, 2011 at 4:28 PM, Kevin P. Fleming kpflem...@digium.com wrote:
This is a bug in the configure script, but in the meantime, you should be
able to use --without-pwlib to avoid it, as long as you aren't trying to
build chan_h323.
Thanks much.
I was trying
./configure
On 09/06/2011 04:09 PM, David Backeberg wrote:
On Tue, Sep 6, 2011 at 4:28 PM, Kevin P. Flemingkpflem...@digium.com wrote:
This is a bug in the configure script, but in the meantime, you should be
able to use --without-pwlib to avoid it, as long as you aren't trying to
build chan_h323.
2011/9/6 Esteban Cacavelos estebancacave...@gmail.com
2011/9/6 Leif Madsen leif.mad...@asteriskdocs.org
On 04/09/11 02:51 PM, Tamer Higazi wrote:
the 3rd edition is available, but that book covers every thing to run
the asterisk PBX.
You can read the 3rd edition online at
Hello list, I am a beginner at asterisk. I want to access my asterisk
box from my laptop, on a different network (mobile hotspot). The
asterisk box doesn't have a static ip, how do I connect with it using
ssh or other such programs?
Thanks for your guidance guys. It is highly appreciated.
--
Google for IP-tunneling.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A Dunor
Sent: Tuesday, September 06, 2011 7:17 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Beginner Question: Remote
Hello list. Just another beginner question. I am trying to setup a basic
home phone system. I ordered a TDM410 card, which came with 4 fxo ports.
I want the home phone system to be able to initiate and receive calls.
Can it be done with this card with just one type(no fxs) of ports? If it
can
Thanks for the speedy pointer Danny.
On 9/6/2011 8:27 PM, Danny Nicholas wrote:
Google for IP-tunneling.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A Dunor
Sent: Tuesday, September 06, 2011 7:17 PM
To:
yes, sangoma card is good.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
Date: Tue, 6 Sep 2011 10:38:44 +0200
From: th9...@googlemail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
hi:
yes, 4fxo is enough for that. four fox means that you have 4 PSTN line, do you
really need
4 fxos? you have to use fxs or sip as extensions for pick up the call and make
calls.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP).
website:
On 06/09/11 05:14 PM, Kevin P. Fleming wrote:
I was trying
./configure --disable-chan_ooh323
and that was not making a difference.
It won't, for two reasons: Asterisk modules can't be selected/deselected
via the configure script (menuselect is used for that), and chan_ooh323
doesn't use
Hi team,
I am trying to find solution to hangup b-party call after 1 min with out
disconnecting the call of a-party but following dial plan which is disconnect
both the calls.
Please suggest me the solution.
[TB]
exten = _X.,1,Wait(${INCOMING_WAIT})
exten =_X.,2,Verbose(TB)
exten
There could be as easy solutions as using teamviewer or use tools like
Hamachi used in combination with dyn-dns etc. IP-tunneling I guess needs
static public IPs for the sake of completing the route.
On Wed, Sep 7, 2011 at 5:30 AM, A Dunor alsta...@gmail.com wrote:
Thanks for the speedy pointer
On Tue, Sep 06, 2011 at 08:17:01PM -0400, A Dunor wrote:
Hello list, I am a beginner at asterisk. I want to access my asterisk
box from my laptop, on a different network (mobile hotspot). The
asterisk box doesn't have a static ip, how do I connect with it using
ssh or other such
We're using FreePBX and I'm wondering if it's possible to add to the
login/logout macros a command to execute an AGI/Command to launch an
external process for that.
Thanks.
On Fri, Aug 12, 2011 at 2:30 PM, Alex Vishnev alex9...@gmail.com wrote:
you can monitor queue_log file for ADDMEMBER or
See this link:
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL
You'll find similar pages where you can setup to store queue logs/events(as
Alex mentioned) in MySQL DB and further do your triggers or functions on
them.
On Wed, Sep 7, 2011 at 10:46 AM, Michael
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