Zhang Shukun wrote:
hi, all
in my test,it shows Playback will answer the call automaticly, but i
don't want to so.
i will use answer function to answer the call. could you help me ?
core show application Playback
Regards
Hans
--
I've just encountered an odd problem with our Digium TE410P card and was
wondering if anyone has experienced something similar before.
We utilize all 4 ports with 2 of them connected to the PSTN as E1 with the
second 2 ports connecting to a device which accepts T1. We are essentially
acting as an
Le 22/02/2010 09:28, Conor McTernan a écrit :
I've just encountered an odd problem with our Digium TE410P card and
was wondering if anyone has experienced something similar before.
There is one similar request on this list from a few weeks back iirc
We quickly removed the card and checked the
I find there are only few mails other than mine, which hasn't been replied
to. Have I put the question in the wrong mailing list ?
Sorry for being impatient.
--
Jayesh Jayan
The box said Requires Windows 95, NT, or better, so I installed Linux.
Visit my homepage @ http://www.jayeshjayan.com
Yes, when I added t38pt_usertpsource=yes to the NAT'ed fax everything
works!
Big thanks Johann!
On Sun, 21 Feb 2010 17:22:40 +0100, Magnus Benngård wrote:
t38pt_usertpsource=yes seems to do the trick, switches to T38 and fax
seems to go through (cant be 100% sure, the fax i am sending
22 feb 2010 kl. 07.23 skrev Tilghman Lesher:
open audio {tcp|udp} hostname portno
close audio
If you design something now, I would strongly suggest that we stop using
audio as an attribute. Each call will have multiple media streams - and
already have. You need to be able to select which
Pedro Santos wrote:
Does any one put a HFC-S card working in nt ptp mode?
I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno if
that helps.
/Per Jessen, Zürich
--
http://www.spamchek.com/ - your spam is our business.
--
On 22 February 2010 10:26, Per Jessen p...@computer.org wrote:
Pedro Santos wrote:
Does any one put a HFC-S card working in nt ptp mode?
I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno if
that helps.
/Per Jessen, Zürich
Not meaning to hi-jack this thread, I’m not
I have a business problem that is killing me. I do SIP2SIP, only. I place a
call after receiving the incoming request, and I need to send a Hangup(Code)
to the caller, based on the result of the outbound leg. How can I do that in
Asterisk? Is that even possible at all?
I can use Hangup(code), but
On 22 Feb 2010, at 11:16, CDR wrote:
I have a business problem that is killing me. I do SIP2SIP, only. I
place a call after receiving the incoming request, and I need to
send a Hangup(Code) to the caller, based on the result of the
outbound leg. How can I do that in Asterisk? Is that
Folks, I am strugging with Asterisk 1.4 Vs 1.6 differences over AMI
Originate? Here is the pastebin... http://pastebin.ca/1805594
Not sure why the local channel won't send to context while the remote
channel does. Worked fine in 1.4 but 1.6.1 has issues.
Any help?
Ritesh
--
Hi all,
Is there a way to deny call transfers to certain extensions?
Thanks,
Ahmed Ossama
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options
On 2/22/2010 10:26 AM, Per Jessen wrote:
Pedro Santos wrote:
Does any one put a HFC-S card working in nt ptp mode?
I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno if
that helps.
/Per Jessen, Zürich
I have use this howto
On 2/22/2010 7:36 AM, Tzafrir Cohen wrote:
On Sun, Feb 21, 2010 at 07:55:39PM +, Pedro Santos wrote:
Does any one put a HFC-S card working in nt ptp mode?
Which version of Asterisk do you use? Which channel driver?
I have use this howto
Jayesh Jayan wrote:
I find there are only few mails other than mine, which hasn't been
replied to. Have I put the question in the wrong mailing list ?
Not the wrong mailing list, but most likely nobody has any answers for
you. I personally have never used the application.
Doug
--
Doug,
Thank you for your update.
Google results also reveal very less number of users for this app.
By the if we assume it is some other app, and we have to change the default
keypad settings, how do we go about changing it ? Do we have to alert the
code ?
--
Jayesh Jayan
The box said
On Mon, Feb 22, 2010 at 12:22:39PM +, Pedro Santos wrote:
On 2/22/2010 10:26 AM, Per Jessen wrote:
Pedro Santos wrote:
Does any one put a HFC-S card working in nt ptp mode?
I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno if
that helps.
/Per
Hi,
Does anybody have any experience with asterisk where are four PCIe cards
are used in one server (TE420).
So you can have max 4 * 4 * 30 channels = 480 channels used.
Regards,
Arjan Kroon
Mobillion BV
--
_
--
Jayesh Jayan wrote:
By the if we assume it is some other app, and we have to change the
default keypad settings, how do we go about changing it ? Do we have
to alert the code ?
I'm guessing that you'd have to modify the code.
Not knowing this particular application and not having any
On 22 February 2010 13:02, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
Nowadays (as of Asterisk 1.6.0) BRI support is included in Asterisk. The
zaphfc driver, though, is still not included in DAHDI. It's maintained,
though. The version included in the Debian packages is taken from
Thank you, Doug.
Let me try in that direction.
--
Jayesh Jayan
The box said Requires Windows 95, NT, or better, so I installed Linux.
Visit my homepage @ http://www.jayeshjayan.com
On Mon, Feb 22, 2010 at 6:48 PM, Doug Lytle supp...@drdos.info wrote:
Jayesh Jayan wrote:
By the if we
Follow-me will most likely be your best bet for this trick. Say you have
extensions 100, 101 and 102. 100 is the receptionist, 101 is sales and 102
is the boss, who doesn't want to be disturbed. If you set up followme on
102 to go to voicemail or whatever, 102 won't ring.
-Original
On Wed, Feb 17, 2010 at 6:53 PM, Tilghman Lesher tles...@digium.com wrote:
Oh, right, priLOCALdialplan. What's in CALLERID(num) ? Legitimate characters
for the PSTN are numbers (and ABCD) only, so other characters are invalid,
making them candidates for usage in modifying prilocaldialplan.
Not wit four - but two of them in a single core 3GHz machine worked
flawlessly doing only switching and IVR without codec conversion.
Many will suggest that you split your lines on two machines to to
prevent a total loss when a machine fails. This will add some work on
setup but maybe save you
Hi,
We are now using 2 PCI cards (TE410) in all our server without any problem.
Because we want to reduce the power consumention of the complete server-park,
we though to put 4 PCIe cards in 1 server.
We have a redundancy of our servers, so machine fails is not a great issue.
Regards,
Arjan
I do not remember and issues we have between 1.4 and 1.6. When going to your
pastebin I get this:
Sorry, an error has occurred. Reason: That is an invalid ID, or the post has
expired.
Can you post what your ami packets contain?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
On Mon, Feb 22, 2010 at 8:20 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
Hi,
Does anybody have any experience with asterisk where are four PCIe cards are
used in one server (TE420).
So you can have max 4 * 4 * 30 channels = 480 channels used.
I would recommend calling
On 2/22/2010 1:02 PM, Tzafrir Cohen wrote:
On Mon, Feb 22, 2010 at 12:22:39PM +, Pedro Santos wrote:
On 2/22/2010 10:26 AM, Per Jessen wrote:
Pedro Santos wrote:
Does any one put a HFC-S card working in nt ptp mode?
I've got an HFC-PCI (single channel)
Here is the entire thing including problem statement, CLI, and AMI
responses.
http://pastebin.ca/1805792
Ritesh
On Mon, Feb 22, 2010 at 8:31 PM, Jim Dickenson dicken...@cfmc.com wrote:
I do not remember and issues we have between 1.4 and 1.6. When going to
your pastebin I get this:
Sorry,
Hiya - quick question..
When an external call is answered by an extension and the person answering the
call wants to forward it to a different extension, is there any way to change
the caller ID when the call is transferred?
If someone is transferring a call to me, I see the caller ID of the
On Sun, Feb 21, 2010 at 10:04 PM, Sean Brady sbr...@gtfservices.com wrote:
I do get choppy audio when playing recordings occasionally. I haven’t had
time to figure that one out, but I haven’t put it into production yet.
You just said you're getting unexplained choppiness.
You also just said
On 02/20/2010 01:53 AM, jonas kellens wrote:
I have read on this list that people do not get a reply if they ask
stupid questions.
Is this then a stupid question that I ask ?
If nobody has ever combined extensions.conf and realtime in a way that
I want to do, I wanna hear it too. Even if
What you need to do is set a channel variable with callerid(num) from the
external number, then reset callerid(num) whenever you do an internal dial
to transfer - something like this
[from-pstn]
Exten = s,1,answer
Exten = s,n,Set(passcallID=callerid(num))
[transfer]
Exten =
David Backeberg wrote:
Timers are built on the premise that they have access to either a real
timing device, or unobstructed access to a processor which clicks
through a proc cycle at a pre-determined rate. Once you break those
rules, don't be surprised when the timers stop working, and 'bad
On 19 February 2010 15:28, Steve Davies davies...@gmail.com wrote:
[snip]
I just upgraded to the new bootblock and 3.2.2 firmware, and these
phones will now talk video to other devices. Nothing in the changelogs
indicates why, but there is a definite jump up from the previous
release of this
On 22 Feb 2010, at 15:38, Danny Nicholas wrote:
What you need to do is set a channel variable with callerid(num) from the
external number, then reset callerid(num) whenever you do an internal dial
to transfer - something like this
[from-pstn]
Exten = s,1,answer
Exten =
On Mon, Feb 22, 2010 at 10:51 AM, Jonathan Addleman j...@redowl.ca wrote:
David Backeberg wrote:
Timers are built on the premise that they have access to either a real
timing device, or unobstructed access to a processor which clicks
through a proc cycle at a pre-determined rate. Once you
The ID at dial/transfer time is what you are stuck with.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Will Payne
Sent: Monday, February 22, 2010 10:00 AM
To: Asterisk Users Mailing List - Non-Commercial
Jason,
Thanks for that, but I am still getting an error. I run rpmbuild using this
command
rpmbuild --bb ~/localrpms/SPECS/dahdi-linux-kmod.spec --target=i686 --define
kversion `uname -r`
but it fails with this error message.
make[1]: Leaving directory `/usr/src/kernels/2.6.18-128.el5-i686'
Hi,
looking for your valued input on suitable suggestions for high quality VoIP
DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and
looking to a new manufacturer.
--
Thanks, Phil
--
_
-- Bandwidth
On 22 February 2010 15:59, Will Payne w...@teambadger.co.uk wrote:
On 22 Feb 2010, at 15:38, Danny Nicholas wrote:
What you need to do is set a channel variable with callerid(num) from the
external number, then reset callerid(num) whenever you do an internal dial
to transfer - something like
looking for your valued input on suitable suggestions for high quality VoIP
DECT
phones. I am having real issues with my Snom M3s and Asterisk 1.6 and looking
to a new manufacturer.
We've been using the Siemens Gigaset range for a few years now (specifically
C475IP and S685IP). Not had any
On Mon, Feb 22, 2010 at 5:18 PM, --[ UxBoD ]-- ux...@splatnix.net wrote:
Hi,
looking for your valued input on suitable suggestions for high quality VoIP
DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and
looking to a new manufacturer.
--
Thanks, Phil
RTX3080 is
On Mon, Feb 22, 2010 at 11:06 AM, David Backeberg dbackeb...@gmail.comwrote:
On Mon, Feb 22, 2010 at 10:51 AM, Jonathan Addleman j...@redowl.ca
wrote:
David Backeberg wrote:
Timers are built on the premise that they have access to either a real
timing device, or unobstructed access to a
On 22 February 2010 16:18, --[ UxBoD ]-- ux...@splatnix.net wrote:
Hi,
looking for your valued input on suitable suggestions for high quality VoIP
DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and
looking to a new manufacturer.
--
Thanks, Phil
We use the snom
I have always heard that the less cards in a single system the better.
Why not try two Sangoma A108DE cards (8 ports each). Also make sure you
have hardware echo cancellation on the cards for this number of ports.
On Mon, 2010-02-22 at 14:20 +0100, Arjan Kroon | Mobillion wrote:
Hi,
On Monday 22 February 2010 03:49:48 Olle E. Johansson wrote:
22 feb 2010 kl. 07.23 skrev Tilghman Lesher:
open audio {tcp|udp} hostname portno
close audio
If you design something now, I would strongly suggest that we stop using
audio as an attribute. Each call will have multiple media
Jerry Geis wrote:
I am trying to find out how I can tell the length of a string actually
CALLERID(num) in the dialplan.
How is that done?
If need to test the length of the CALLERID(num) if its less the 10 digits I
need to set it to a known value or insert 0's at the beginning until it
Tzafrir Cohen wrote:
On Mon, Feb 22, 2010 at 12:22:39PM +, Pedro Santos wrote:
On 2/22/2010 10:26 AM, Per Jessen wrote:
Pedro Santos wrote:
Does any one put a HFC-S card working in nt ptp mode?
I've got an HFC-PCI (single channel) running in NT ptp mode. Dunno
if
Interesting thread recently about virtual servers...
I'm thinking of doing something similar - right now looking at Containers
(lxc) rather than proper virtualisation though, however it got me
thinking of a poor mans virtualisation solution...
This would assume you have a real server to start
Running Asterisk trunk with Siemens Gigaset S685IP, no normal problems,
just some with connected-line, probaly me, who is not smart enough. :(
Sound is great, use them both at our WAN and NAT'et at my home, DTMF
working as a clock... what more can I say?
On Mon, 22 Feb 2010 16:43:04 -,
On Mon, Feb 22, 2010 at 12:57:30PM -0500, Leif Madsen wrote:
Jerry Geis wrote:
I am trying to find out how I can tell the length of a string actually
CALLERID(num) in the dialplan.
How is that done?
If need to test the length of the CALLERID(num) if its less the 10 digits I
need
On Mon, 22 Feb 2010, --[ UxBoD ]-- wrote:
Hi,
looking for your valued input on suitable suggestions for high quality
VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk
1.6 and looking to a new manufacturer.
Siemens Gigaset over M3's anyday. Nicer displays, bigger
- Gordon Henderson gordon+aster...@drogon.net wrote:
On Mon, 22 Feb 2010, --[ UxBoD ]-- wrote:
Hi,
looking for your valued input on suitable suggestions for high
quality
VoIP DECT phones. I am having real issues with my Snom M3s and
Asterisk
1.6 and looking to a new
Gordon Henderson wrote:
Interesting thread recently about virtual servers...
I'm thinking of doing something similar - right now looking at Containers
(lxc) rather than proper virtualisation though, however it got me
thinking of a poor mans virtualisation solution...
This would assume
Hi,
The last few times I have installed trunk versions of asterisk on Ubuntu I have
seen this error after doing a make config for asterisk.
install: cannot stat `contrib/init.d/etc_default_asterisk': No such file or
directory
The init.d links then fail to work properly (e.g. /etc/init.d/asterisk
Hello,
We used to recommend a commercial software but client is a small callcenter who
cannot afford something big.
Would you recommend something open-source which could work for a 40-seater?
Thank you,
Tudor
www.sunabasarabia.com
Moldova 11c/min
I have followed the instructions on voip-info.org for Realtime SIP
peers, but I get this notice :
[Feb 22 20:05:32] NOTICE[15298]: chan_sip.c:15889
handle_request_register: Registration from
'sip:test...@192.168.1.150;transport=UDP' failed for '192.168.1.105' -
No matching peer found
The CLI
Hi all,
I'm trying to get moh working on * version 1.4.4. I've setup a test
extension that answers the call and runs the musiconhold command with
the appropriate class name.
All I get on the phone is silence. The console tells me that moh
started and immediately stopped, but it complains
Little fault in my mailing :
The CLI shows :
[Feb 22 19:58:23] == Parsing '/etc/asterisk/extconfig.conf': [Feb 22
19:58:23] Found
[Feb 22 19:58:23] == Binding voicemail to
mysql/Asterisk/voicemail_users
[Feb 22 19:58:23] == Binding sipusers to mysql/Asterisk/sip_buddies
[Feb 22 19:58:23]
On Mon, Feb 22, 2010 at 2:20 PM, Mike Diehl mdi...@diehlnet.com wrote:
Hi all,
I'm trying to get moh working on * version 1.4.4. I've setup a test
I don't know the answer, but are you really using 1.4.4? If so,
consider taking some time to review the security and feature
improvements over the
David Backeberg wrote:
On Mon, Feb 22, 2010 at 2:20 PM, Mike Diehl mdi...@diehlnet.com wrote:
Hi all,
I'm trying to get moh working on * version 1.4.4. I've setup a test
I don't know the answer, but are you really using 1.4.4? If so,
consider taking some time to review the security
The problem was that I had a different value for 'name' and 'username'.
How can I have the 'name' different from the 'username' ??? Why do these
2 need to be the same ??
Jonas.
On Mon, 2010-02-22 at 20:36 +0100, jonas kellens wrote:
Little fault in my mailing :
The CLI shows :
[Feb 22
Hello Jonas:
Change this parameter, if you are using Mysql.
[general]
dbhost = 127.0.0.1
dbname = Asterisk
dbuser = asteriskuser
dbpass = asteriskpasswd
dbport = 3306
*dbsock = /var/lib/mysql/mysql.sock*
cheersss...
2010/2/22 jonas kellens jonas.kell...@telenet.be
The problem was that I had a
I have a connection of Asterisk with Avaya by H.323 and so far everything
worked well because only sent to Avaya. Now, the matter is that from Avaya will
send me an IVR calls to capture credit card information, the link is active on
Avaya 23 channels which is not how to configure Asterisk for
Dear Juan,
thank you for your answer. The reason why registration failed was a
mismatch between the 'username'-field and the 'name'-field.
If I put both values to the same, it works... But why do these 2 need to
be the same ? I would rather have a different 'name' and
'username'-parameter.
Good day all!
I have an issue which has plagued me for quite sometime now...and as I close
in on its cause, I have reached a point where additional info would be
greatly helpful!
When a SIP device dials another SIP device...Asterisk connects the calls and
displays the channel information.
If one
I think Vicidial, works great.
Regards.
2010/2/22 Apa Minerala apaminer...@yahoo.com
Hello,
We used to recommend a commercial software but client is a small callcenter
who cannot afford something big.
Would you recommend something open-source which could work for a 40-seater?
Thank
GnuDialer
*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-Soporte PostgreSQL
*-www.jqmicrosistemas.com
*-809-849-8087
*---*
From: juanch...@gmail.com
Date: Mon, 22
On 02/22/2010 11:13 AM, jonas kellens wrote:
I have followed the instructions on voip-info.org for Realtime SIP
peers, but I get this notice :
[Feb 22 20:05:32] NOTICE[15298]: chan_sip.c:15889
handle_request_register: Registration from
'sip:test...@192.168.1.150;transport=UDP' failed for
On Mon, 2010-02-22 at 16:13 -0500, JT wrote:
Is this something that is fixed in an update? (Currently running 1.2)
Yes... modern versions of Asterisk support SIP session timers. (If I
remember correctly, Asterisk 1.2 could tear down a call based on lack of
RTP data, but I never found it worked
Hello everybody.
I have a provider that has 3 asterisk boxes which I must balance my
calls against. At the moment, I route different destinations to
different boxes but this causes lots of problems.
Without resorting to OpenSER or other proxies (as my provider also
uses IAX), is there a way I
I had a system running on Xen in test. I had terrible echo problems with a
SPA3000. As a reference, I swapped to bare metal machine and although I still
had echoing, the echoing was much closer to the original sound. The Xen server
was idle apart from the AsteriskNOW installation. So, this lead
Forgive the possibly stupid question, but do these problems you describe
apply equally to the dom0 as to any domU's in a xen system? I used to
think not, but now I'm starting to realize that I'm probably mistaken...
Dom0 is still a virtual machine, so I would say so.
--
On 02/19/10 08:54, Olle E. Johansson wrote:
17 feb 2010 kl. 19.12 skrev Joseph:
Does the sort order matter in sip.conf file?
I know sort order might effect:
allow=ulaw
allow=alaw
but does it matter where I place: insecure=invite ?
The reason I'm asking is that I've loaded almost two
On Mon, 22 Feb 2010, Roderick A. Anderson wrote:
Gordon Henderson wrote:
Interesting thread recently about virtual servers...
I'm thinking of doing something similar - right now looking at Containers
(lxc) rather than proper virtualisation though, however it got me
thinking of a poor mans
On 100222 1313, JT wrote:
When a SIP device dials another SIP device...Asterisk connects the calls and
displays the channel information.
If one of those SIP devices hangs up, Asterisk receives the hangup notice
and disconnects the call/channel.
However - what does Asterisk do when the network
On Mon, 22 Feb 2010, Alejandro Recarey wrote:
I have a provider that has 3 asterisk boxes which I must balance my
calls against. At the moment, I route different destinations to
different boxes but this causes lots of problems.
[snip]
Is there any way I can balance calls between all of
On Mon, Feb 22, 2010 at 6:13 PM, Benoit maver...@maverick.eu.org wrote:
There is one similar request on this list from a few weeks back iirc
Oh, I'd looked through the archives/googled etc. but could not find
anything similar. I'll take another stab at the archives.
We quickly removed the
On Mon, 22 Feb 2010, Steve Edwards wrote:
dial(iax2/isp${MATH(${EPOCH}%3):0:1}/${EXTEN})
Improving on myself...
Using the decimal portion of UNIQUEID (the number of channels
created by this instance of Asterisk) would be better than EPOCH.
--
Thanks in advance,
Ian Murray wrote:
Forgive the possibly stupid question, but do these problems you describe
apply equally to the dom0 as to any domU's in a xen system? I used to
think not, but now I'm starting to realize that I'm probably mistaken...
Dom0 is still a virtual machine, so I would say so.
Hi!
looking for your valued input on suitable suggestions for high quality
VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk
1.6 and looking to a new manufacturer.
Define high quality.
Anyone here used any of these below with Asterisk?
* NEC AP300 and NEC DECT C124 or
Kirill 'Big K' Katsnelson wrote:
The caveat here is that it is perfectly normal NOT to transmit any RTP
data in case of long silence. This is why the SIP timers were introduced
in the first place: there is no correct way to detect when the client is
going away, as no activity is a good
On 100222 1818, Kevin P. Fleming wrote:
Kirill 'Big K' Katsnelson wrote:
The caveat here is that it is perfectly normal NOT to transmit any RTP
data in case of long silence. This is why the SIP timers were introduced
in the first place: there is no correct way to detect when the client is
23 feb 2010 kl. 03.18 skrev Kevin P. Fleming:
Kirill 'Big K' Katsnelson wrote:
The caveat here is that it is perfectly normal NOT to transmit any RTP
data in case of long silence. This is why the SIP timers were introduced
in the first place: there is no correct way to detect when the
23 feb 2010 kl. 01.47 skrev Kirill 'Big K' Katsnelson:
On 100222 1313, JT wrote:
When a SIP device dials another SIP device...Asterisk connects the calls and
displays the channel information.
If one of those SIP devices hangs up, Asterisk receives the hangup notice
and disconnects the
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