The FSV-4PFS as shipped will not switch Ethernet - it switches pins 1,2,4,5.
Craig
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of FailSafe Inc.
Sent: Tuesday, 2 September 2008 11:27 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] PRI Splitter
[442033553]
user=442033553
type=pusers
secret=1234
host=dynamic
context=users
nat=yes
make it context=stations , i am assuming this is how your DID provider
is sending u calls ?
Let us know if your DID provider is just sending calls to your ip
address or you are registering asterisk server with
Hi,
Is there any way of achieving what I have mentioned in my previous
email. Scenario:
I am recording all calls in queue. I want to save file in a way that I
can identify the agent for whom the recording ahs been made. The saved
file name should have something related to agent id or
Dear All,
Issue resolved. The problem is not in either libpri, iax, zaptel or
any other.
The problem is in Telco. Outgoing has been blocked due to billing :)
Now it is working perfectly.
As i already mentioned that incoming call was working fine.
Shariq
On 9/4/08, Richard Lyman [EMAIL
Hello,
I would like to show you that when using Dial L( x [: y ][: z ]) option
via AGI the Dial content is truncated in the first colon [:y].
In other words, note below that the error output shows a truncation in the
first colon - No such host: 1001,,L(32000
AGI Rx EXEC Dial
On Thu, 4 Sep 2008, Tharanga wrote:
Hi folks,
Can some one recommend a good video phone for asterisk (SIP.IAX2). I need
better quality, duarability. and should support various video codec's
.(Codec auto negotiation support id prefferble)
I suspect that the choices are so limited right now
Alex,
Unfortunately these two setting didn't change the behaviour either... Could
it be a bug in the 1.4.13 version I use?
Thanks,
Best regards,
Tobias
Date: Wed, 03 Sep 2008 03:27:26 -0500
From: Alejandro Kauffmann [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Asterisk Queue's
To: Asterisk
Geraint Lee wrote:
I've used several hitachi dmp330's they work great, roam between
wireless access points with no loss of audio or connection for that matter.
it will be a great shame if hitachi stop producing them, they are the
most reliable wireless sip phones i've come accross... stay
I have several asterisk servers running a couple of different versions of 1.4.
One of our severs in California is running 1.4.18 with the Dial Plan in
Realtime mySQL. This server is storing voicemails in the database connecting
via odbc. There are approximately 900 sip users registered
Greetings list,
I finally got round to upgrading a few of our 1.2 servers to 1.4 over the last
few days. Most of the changes in config files went without a hitch, but this
one bothers me:
ERROR[15836]: config.c:750 process_text_line: Future versions of Asterisk will
treat a #include of a file
The issue isn't so much when the FAX leaves the PRI card, but when the fax goes
from TDM to IP. If the FAX is going from one PRI card to another PRI card,
there should be no problem with faxing, but when you start trying to run faxes
over IP is when you will most likely start having problems.
Hello list,
I found a nice application that I want to try called ValetParking. However,
I can only find the source code (app_valetparking.c) to this, and no
installation instructions. Can anyone tell me how I compile this application
to use as a module in Asterisk 1.4?
Thanks,
Best regards,
Hi,
I'm receiving this :
[Aug 28 02:25:16] NOTICE[5895] chan_sip.c: Received SIP subscribe for peer
without mailbox: 9163
I've read this :
http://lists.digium.com/pipermail/asterisk-users/2008-May/211701.html
I typed this:
asterisk -rx reload
asterisk -rx voicemail show users
... and got :
Olivier wrote:
Hi,
I'm receiving this :
[Aug 28 02:25:16] NOTICE[5895] chan_sip.c: Received SIP subscribe for
peer without mailbox: 9163
I've read this :
http://lists.digium.com/pipermail/asterisk-users/2008-May/211701.html
I typed this:
asterisk -rx reload
asterisk -rx voicemail show
Hi,
I am trying to test the ilbc codec on asterisk.
allow=ilbc
disallow=all
using zoiper on two extensions, i set codec on zoiper to ilbc and
disabled other codecs
tested a call, looked at the channel:
NativeFormats: 0x400 (ilbc)
WriteFormat: 0x40 (slin)
ReadFormat: 0x40 (slin)
Hy Craig,
Can you elaborate on that? In our setup we have it doing just that and
it works without a glitch.
Regards,
Igor H.
Craig Guy wrote:
The FSV-4PFS as shipped will not switch Ethernet – it switches pins 1,2,4,5.
Craig
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
My current config:
pstn - audiocodes fxo gateway - asterisk - xlite
every fxo ports are registered with asterisk
I have this extensions.conf
exten = 111,1,answer
exten = 111,n,dial(sip/fxo1)
exten = 111,n,hangup
If we dial 111 by xlite, I could hear pstn dialing tone.
I have a Grandstream GXP1200 and eager to try this codec. I've heard
good things about the quality.
Anyone tried it with asterisk?
I can't until 1.6 is released.
I have used G.722 with Asterisk many times. If you have more specific
questions about it and Asterisk, I would be happy to
Dear Sir,
Please find below the error that we are getting when enabling 'sip set
debug'.
localhost*CLI
--- Reliably Transmitting (no NAT) to 83.202.82.39:5060 ---
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 83.202.82.39:5060
;branch=z9hG4bK5ac79f249887f915005b5d34415b1a56;received=83.202.82.39
From:
On Thu, Sep 4, 2008 at 4:44 PM, ACL [EMAIL PROTECTED] wrote:
My current config:
pstn - audiocodes fxo gateway - asterisk - xlite
every fxo ports are registered with asterisk
I have this extensions.conf
exten = 111,1,answer
exten = 111,n,dial(sip/fxo1)
exten = 111,n,hangup
If we dial
Hi everyone,
I'm trying to get calls to record with the following setup:
Using phpagi originate is called from a web application:
$asm-originate(Local/ . $row['extension'] . @sip-standard,
$row['phone_number'], sip-standard, 1, , , 7000);
The agent being called is extension Local/[EMAIL
2008/9/4 Rob Hillis [EMAIL PROTECTED]
Olivier wrote:
Hi,
I'm receiving this :
[Aug 28 02:25:16] NOTICE[5895] chan_sip.c: Received SIP subscribe for
peer without mailbox: 9163
I've read this :
http://lists.digium.com/pipermail/asterisk-users/2008-May/211701.html
I typed this:
On Thu, 04 Sep 2008 00:42:21 -0500, Karl Fife wrote:
Has anyone seen or done an XML phone application integration using
Z-Wave or Zigbee (or legacy Crestron) for Office or Home automation,
lighting thermostatic control, alarm systems etc?
If you've seen Crestron systems you may know that they
I am about to setup a new Asterisk box which only uses SIP.
I used to simply use menuselect with Zaptel and choose the tools
that Asterisk required to exist and ztdummy.
Now with Dahdi, I am reading
http://svn.digium.com/view/dahdi/tools/tags/2.0.0-rc2/UPGRADE.txt?view=co
and I understand I no
Hi List;
About logs existed under the folder /var/log/asterisk/, I would like to know
the following:
1) How to enable/disable the messages log?
2) When messages log happen? Based on error or running application?
3) What difference between messages log and even log?
4) queue_log to be used for
On Sep 4, 2008, at 6:25 AM, Steve Repo wrote:
Specifically my questions are,
[1] The quality of voice between g722 and say GSM or 729
I suppose that it's sort of subjective, but I think it sounds
_awesome_. It's a huge difference in quality to me. You just need to
try it out. :)
[2]
On Sep 4, 2008, at 4:12 AM, z_gringo wrote:
I have several asterisk servers running a couple of different
versions of 1.4. One of our severs in California is running
1.4.18 with the Dial Plan in Realtime mySQL. This server is storing
voicemails in the database connecting via
upgrading from zaptel to dahdi, with a TDM400P:
Is /etc/dahdi/system.conf the same as /etc/zaptel.conf? As I read the
system.conf.sample, no echo canceller need be specified if there's a
hardware ec. Can I just rename zaptel.conf?
What about zapata.conf? Is this just renamed
On Thursday 04 September 2008 03:15:51 selmak se wrote:
AGI Rx EXEC Dial SIP/1001,,L(32000:2:1)
[Sep 4 11:04:20] WARNING[18100]: chan_sip.c:2907 create_addr: No such
host: 1001,,L(32000
The issue is that internally, the application argument delimiter in 1.4 is
actually the pipe
On Thursday 04 September 2008 04:25:27 Chris Bagnall wrote:
I finally got round to upgrading a few of our 1.2 servers to 1.4 over the
last few days. Most of the changes in config files went without a hitch,
but this one bothers me:
ERROR[15836]: config.c:750 process_text_line: Future versions
On Thursday 04 September 2008 05:55:48 Nhadie wrote:
Hi,
I am trying to test the ilbc codec on asterisk.
allow=ilbc
disallow=all
using zoiper on two extensions, i set codec on zoiper to ilbc and
disabled other codecs
tested a call, looked at the channel:
NativeFormats: 0x400 (ilbc)
I'd also be more sold on it if it had half the features of the GXP2000
(which is only a little over half the price).
Sure, but if only half of the features in the GXP2000 actually work,
what is the point of them? I'd take a stable phone with less features
over one that has lots of features
Hi all,
I am having a strange problem with my asterisk server. When i dial an
outside tollfree number, if there is a menu for example press 1 for support,
press 2 for sales etc, after pressing any given option as the system begins
to transfer me the call hangs up. I have tried it so many times on
Olivier wrote:
Now that root cause is found, would you say that warnings or CLI
should have been different ?
Obviously, MWI subscriptions must come from SIP hardphones (at least
those supporting MWI feature).
So in this case, Received SIP subscribe for peer without mailbox:
9163 rather
Hi guys,
I am trying to configure an asterisk server for our office.
Asterisk 1.4.17 SIP only
The problem appears when the call comes from external point to our
internal network. So when the server receives the call the channel is
answered and the remote user hears prompt which invite him to
It will do so by default if you have a valid
/etc/asterisk/indications.conf (only used for inband tones like after an
Answer())
eng. Anatoli Marinov wrote:
Hi guys,
I am trying to configure an asterisk server for our office.
Asterisk 1.4.17 SIP only
The problem appears when the call comes
On Thu, Sep 4, 2008 at 7:57 PM, Peder @ NetworkOblivion
[EMAIL PROTECTED] wrote:
I'd also be more sold on it if it had half the features of the GXP2000
(which is only a little over half the price).
Sure, but if only half of the features in the GXP2000 actually work,
what is the point of them?
On Thu, Sep 4, 2008 at 1:48 PM, Gordon Henderson
[EMAIL PROTECTED] wrote:
On Thu, 4 Sep 2008, Tharanga wrote:
Hi folks,
Can some one recommend a good video phone for asterisk (SIP.IAX2). I need
better quality, duarability. and should support various video codec's
.(Codec auto negotiation
On Thu, 4 Sep 2008 21:12:52 +0530, Steve Repo wrote:
On Thu, Sep 4, 2008 at 7:57 PM, Peder @ NetworkOblivion
[EMAIL PROTECTED] wrote:
I'd also be more sold on it if it had half the features of the GXP2000
(which is only a little over half the price).
Sure, but if only half of the features in
Is there any special option which I should enable to activate these tones?
My progressinband is yes and I cal Dial app with r option it it right?
2008/9/4 Eric ManxPower Wieling [EMAIL PROTECTED]:
It will do so by default if you have a valid
/etc/asterisk/indications.conf (only used for
Everyone Interested,
The FSV-4PFS Order page has options for which pins you would like switched.
The default choice is T1/E1/POTS Pins (Pins 1,2,4,5)
Other possible choices are:
Ethernet (Pins 1,2,3,6)
and
All 8 Pins
Igor - you and I spoke before you ordered your devices. I knew that you
I use it with n800 device - nokia internet tablet and standard nokia
soft phone I have video call. The codec that I use is h263 and it
works great.
2008/9/4 Steve Repo [EMAIL PROTECTED]:
On Thu, Sep 4, 2008 at 1:48 PM, Gordon Henderson
[EMAIL PROTECTED] wrote:
On Thu, 4 Sep 2008, Tharanga
Hey, all. I haven't really gotten deep into Asterisk since 1.0.x, and I'm
afraid I've forgotten a fair bit. One big thing that I've forgotten is
the syntax, etc., for extensions.conf. Where do I find that? I'm looking
for stuff about commands, syntax for commands, variables, etc. Is there a
Steve Repo wrote:
I agree! I bought a GXP1200 (business class phone) and it's buggy.
Can't use the message button (404 not found).. and some other features
(404 not found). I have requested help from Grandstream and so far
nothing.
I've never heard of that problem, ar eyou sure the 404
As a result of:
http://lists.digium.com/pipermail/svn-commits/2008-July/036122.html
I am seeing these messages when upgrading for 1.6b9 to 1.6rc4. Is there
something I should be doing to address this warning?
[Sep 4 13:31:34] WARNING[2954]: chan_iax2.c:1200 __send_lagrq: I was
supposed to
This has nothing to do with the progressinband setting and you should
never use the r option.
eng. Anatoli Marinov wrote:
Is there any special option which I should enable to activate these tones?
My progressinband is yes and I cal Dial app with r option it it right?
2008/9/4 Eric
Ken D'Ambrosio wrote:
Hey, all. I haven't really gotten deep into Asterisk since 1.0.x, and I'm
afraid I've forgotten a fair bit. One big thing that I've forgotten is
the syntax, etc., for extensions.conf. Where do I find that? I'm looking
for stuff about commands, syntax for commands,
On Thu, 4 Sep 2008, Steve Repo wrote:
Dlink has launched one in india.
http://www.techgadgets.in/misc-gadgets/2008/13/d-link-gvc-3000-ip-videophone-and-glv-540-ip-phones-announced-in-india/
That's even uglier than the Grandstream ;-)
And why does it remind me of the microsoft un-natural
A cheaper alternative would be the voip wiki.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Sep 4, 2008, at 12:13 PM, Mark
On Thu, 4 Sep 2008 08:48:47 -0500, Russell Bryant wrote:
Asterisk should work fine with any phone that supports that codec.
Personally, I have only used it with Polycom phones. Also, again,
Asterisk 1.4 only supports G722 passthrough, where as Asterisk 1.6 has
full support.
Any plans to
On Thu, 4 Sep 2008, Thomas Kenyon wrote:
Steve Repo wrote:
I agree! I bought a GXP1200 (business class phone) and it's buggy.
Can't use the message button (404 not found).. and some other features
(404 not found). I have requested help from Grandstream and so far
nothing.
I've never heard
On Thursday 04 September 2008 12:59:33 MFH wrote:
As a result of:
http://lists.digium.com/pipermail/svn-commits/2008-July/036122.html
I am seeing these messages when upgrading for 1.6b9 to 1.6rc4. Is there
something I should be doing to address this warning?
[Sep 4 13:31:34]
I was on the call at the time and was not experiencing any apparent
problems.
As I was responding I did some further investigation and saw the
messages even when there wasn't an active call (so I thought). I looked
at the active IAX channels:
[Sep 4 14:27:14] WARNING[2956]:
So as I understand the only thing that I can do is to set up
indications.conf. Ok I will try it tomorrow and will write again with
my results.
Thanks a lot.
2008/9/4 Eric ManxPower Wieling [EMAIL PROTECTED]:
This has nothing to do with the progressinband setting and you should
never use the
http://www.asterisk.org/zaptel-to-dahdi is empty. Is there anyplace else
besides the README's and Upgrade.txt's for config info on updating?
sean
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Why is it an option if it should never be used?.
Thanks,
Steve Totaro
On Thu, Sep 4, 2008 at 1:56 PM, Eric ManxPower Wieling [EMAIL PROTECTED]
wrote:
This has nothing to do with the progressinband setting and you should
never use the r option.
eng. Anatoli Marinov wrote:
Is there any
The setup is as follows: SIP phone registers via international link to Asterisk
Box 1 and calls mean't for termination on Asterisk Box 2 via Zaptel Channels
need to be hairpinned from Box 1 to 2. How is sip.conf configured on Box 1 and
2 so that we don't get an error: Failed to authenticate
I do not know but I could not set it up. :) bad luck maybe.
2008/9/4 Steve Totaro [EMAIL PROTECTED]:
Why is it an option if it should never be used?.
Thanks,
Steve Totaro
On Thu, Sep 4, 2008 at 1:56 PM, Eric ManxPower Wieling [EMAIL PROTECTED]
wrote:
This has nothing to do with the
Shaun Wingrin wrote:
The setup is as follows: SIP phone registers via international link
to Asterisk Box 1 and calls mean't for termination on Asterisk Box 2
via Zaptel Channels need to be hairpinned from Box 1 to 2. How is
sip.conf configured on Box 1 and 2 so that we don't get an error:
The setup is as follows: SIP phone registers via international link
to Asterisk Box 1 and calls mean't for termination on Asterisk Box 2
via Zaptel Channels need to be hairpinned from Box 1 to 2. How is
sip.conf configured on Box 1 and 2 so that we don't get an error:
Failed to
And he can use Vidoe with SIP?
As I know that SIP still does not support video.
Regards
Bilal
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bilal ghayyad schrieb:
And he can use Vidoe with SIP?
As I know that SIP still does not support video.
Of course *SIP* supports video. :-)
It's *Asterisk* which mainly supports voice.
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH -
sean darcy wrote:
http://www.asterisk.org/zaptel-to-dahdi is empty. Is there anyplace else
besides the README's and Upgrade.txt's for config info on updating?
No, the content that was supposed to be there was put into the
Zaptel-to-DAHDI.txt files in the Asterisk 1.4 and Asterisk 1.6
Hi,
Can anyone please comment on what the issue may be with this. I am
trying to set up an Polycom IP601 with multiple buddy icons displaying
endpoint status.
I am using a polycom IP601, sip 2.2.2.0084
In the phone1.cfg file I set:
attendant attendant.uri=4158149992 attendant.reg=1/
Using
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To UNSUBSCRIBE or update options visit:
Michael Graves wrote:
On Thu, 4 Sep 2008 08:48:47 -0500, Russell Bryant wrote:
Asterisk should work fine with any phone that supports that codec.
Personally, I have only used it with Polycom phones. Also, again,
Asterisk 1.4 only supports G722 passthrough, where as Asterisk 1.6 has
Marcelo Freitas schrieb:
Please try again with a better mail client which is able to get
those MIME parts right. :-P
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
A cheaper alternative would be the voip wiki.
http://www.voip-info.org/tiki-
index.php?page=Asterisk%20config%20extensions.conf
Unfortunately, as advised by other asterisk users,
http://www.voip-info.org is sometimes really not that up-to-date.
However, that does not mean that we should give
I believe that this is what I need to enable more than one buddy icon?
Can you please point me in the right direction. Only the polycom
screen, I can only see 1 buddy icon despite having 2 speed dial
entries.
I have been able to successfully turned on presence (which is the term
used
Hi Sir,
For this call i did not do anything except just call the extension
exten = 100,1,Dial(SIP/100|20|t|M(setmusiconhold,moh-100))
that's how i dial the extension, does musiconhold make asterisk
uncompress? but during the call i did not use music on hold. whereelse
should i look at?
Hello!
I have a question of the M(X) option of the Dial, In this M(X), the X
represented the Macro which could be run. Would you tell me that could it
run more than one Macro in this option? And How to do it ?
Thanks
Good morning,
Into the libpri 1.4.5 announcement, it is stated that This version of
libpri retains the ability to operate in this mode, but it is now a
configurable option which defaults to being 'off'. The next releases of
Asterisk will have configuration options to turn this behaviour on if
the
No plan for Asterisk to support video?
What kind of benifit I can get when I have video phone registers with Asterisk?
Regards
Bilal
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Is anybody using Call Waiting works on Linksys 3102? Does it work?
If I'm on the phone, I can hear a notification 'beep', but when I put first
caller on hold the line is busy.
Linksys registers to Asterisk.
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Michael Graves wrote:
On Thu, 4 Sep 2008 08:48:47 -0500, Russell Bryant wrote:
Asterisk should work fine with any phone that supports that codec.
Personally, I have only used it with Polycom phones. Also, again,
Asterisk 1.4 only supports G722 passthrough, where as Asterisk 1.6 has
I'm using a similar feature on 550 and 650 phones, also running 2.2.2.
I've never used the attendant option to do it, though, so I'm not
sure how it differs from what I'm doing. Instead, on the phones that
are allowed to do this, I have the following in their XML config. You
could just as easily
It's not perfect, because it
doesn't display DND or queue login/pause status, but it's better than
nothing.
James, on a different note, is it true that at this stage, we can never
get any queue login status/light on Polycom phone?
I posted a query a few days ago but I have got 0 reply.
Any
As best i could figure it out, I've installed dahdi and rc4.
My TDM400P doesn't answer fxo or fxs.
/etc/dahdi/system.conf:
loadzone = us
defaultzone=us
fxoks=1,2
fxsks=4
/etc/asterisk/chan_dahdi.conf:
[house-phones]
context=internal ; Uses the [internal] context in extensions.conf
H...this bears making some calls to Polycom. They've been very good
to me recently. Very approachable. I think that they're really trying
to deal better with the Asterisk community.
Michael
On Thu, 04 Sep 2008 21:42:12 -0500, Darrick Hartman wrote:
Michael Graves wrote:
On Thu, 4 Sep 2008
I believe DEVSTATE() in 1.6 (backported to 1.4 in various places) will
let you arbitrarily control BLF, so you could control it in the
dialplan when an agent logs in or out (or pauses, or whatever).
Separately, you might be able to use sipsak (http://sipsak.org/) to
construct a SIP message that
Sorry, needed to add one more note. To clarify, my agent phones have a
speed dial assigned for their login, and another to pause/unpause. I
could then use DEVSTATE to enable or disable the light next to their
speed dial button based on their status. I can't use it to update
anything on the LCD
Sorry, needed to add one more note. To clarify, my agent phones have a
speed dial assigned for their login, and another to pause/unpause. I
could then use DEVSTATE to enable or disable the light next to their
speed dial button based on their status. I can't use it to update
anything on the
SIP supporst video :) I am sure because I use it.
2008/9/5 bilal ghayyad [EMAIL PROTECTED]:
And he can use Vidoe with SIP?
As I know that SIP still does not support video.
Regards
Bilal
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