Hi,
Can any body tell me
(i) Does digium TE-120P card can be installed on Redhat linux 9i (2.4.20-8)
kernel
(ii) It is written in documentation that TE120P card be installed only
above 2.6.xx . So, which is the best suited one for it( 2.6.15 or 2.6.18 os
some other release)
(iii)
Two reccomendations:
1) Enable nat for the SIP channels of the phones in SIP.conf.
Or
2) If all the remote phones are in the same location, an IPSEC tunnel
between the remote router, and your Asterisk machine.
Jason.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Am Samstag, den 23.06.2007, 09:52 -0300 schrieb Ronaldo Z. Afonso:
Hi all,
Does anybody know any USB phone that I can use as an IAX Client?
The USB phones I saw on the market just behave like an additional
sound card, with some control buttons perhaps, and those will not work
without a
I already noticed the hisax problem, so I removed the module from the
modules directory so it cannot be loaded anymore. Are you referring to this
driver in specific, or other misdn specific driver.
BTW it seems that messages from the list have about 2 days delay, that is
why I did not see the
Dear sir
I have setup Avaya with mediant with asterisk
[sip_phone]---[ * ]---[mediant]---E1-trunk--[Avaya]---[analog_phone]
This is my configuration when i call from SIP phone i got ringing sound of
phone but whn i call from analog_phone behind avaya i didn't get ring sound of
Hi,
I'm having some trouble setting up a Xorcom Bri 4 port. I have compiled
asterisk and zaptel using the Bristuff bristuff-0.3.0-PRE-1y-g patches.
So I'm running zaptel-1.2.17.1 and asterisk-1.2.18.
The problem I'm having is that for one I get no LEDs showing if the unit is in
TE and NT
Hi
iam using asterisk 1.2 version
I have purchased g729 license from Digium
when iam making calls, iam getting this error ?
Jun 25 14:41:45 NOTICE[4424]: frame.c:183 __ast_smoother_feed: Dropping
extra frame of G.729 since we already have a VAD frame at the end
any help
ram
Siemens GigaSet SL75
On 6/25/07, Michelle Dupuis [EMAIL PROTECTED] wrote:
We're looking at a large wifi phone deployment, and we're looking for
wifi phones that:
1. Are SIP compliant (Asterisk friendly)
2. Provision capable (ideally TFTP of own MAC address)
3. Industrial quality (no cheap
In fact, Dial() doesn't return instantly like it should, in the case it
is used with ENUM. Dial application using the ENUMLOOKUP function
doesn't skip to the next priority like it was expected, if destination
server doesn't answer to the INVITE messages sent by our server.
For example, in the
Hi
On Mon, Jun 25, 2007 at 06:38:37PM +1000, Nathan Dennis wrote:
Hi,
I'm having some trouble setting up a Xorcom Bri 4 port. I have compiled
asterisk and zaptel using the Bristuff bristuff-0.3.0-PRE-1y-g patches.
So I'm running zaptel-1.2.17.1 and asterisk-1.2.18.
The problem I'm
On Mon, Jun 25, 2007 at 12:10:04PM +0530, [EMAIL PROTECTED] wrote:
Hi,
Can any body tell me
(i) Does digium TE-120P card can be installed on Redhat linux 9i
(2.4.20-8) kernel
Why do you keep starting a new thread and not bother following up to
answers in existing threads?
--
Hello everybody.
I have a analog line in the office and a ISDN (with mISDN) line.
I want to call outside from the analog line, but when this is busy, I want
to call outside the second call from the ISDN line.
That my extensions.conf:
[general]
static=yes
writeprotect=yes
[SOME]
exten =
Hi Kyle,
You need to set up a inbound route from DID=skype1 and tell him where to finish.
Something like:
exten = skype1,1,Set(FROM_DID=skype1)
exten = skype1,n,Goto(ext-local,1000,1)
Hope it helps.
Best Regards,
Hugo Picão
Link Consulting - RedesSegurança
Tel: 213 100 182
Av. Duque de
Hi all,
On one of our client, I must to install an asterisk over a hi ability
cluster. I have no experience with clusters an linux neither asterisk.
Someone has installed an asterisk in a hi-ability enbviroment?
How do you install the cluster?
Witch solution did you use?
Witch is the best
MM == Marco Mouta [EMAIL PROTECTED] writes:
MM Siemens GigaSet SL75
The SL75 is DECT, not Wifi.
Apart from that, was it really necessary to quote 20 lines and add a
ridiculous 15 line disclaimer telling me that I'm not allowed to read
the message?
/Benny
voip crazy wrote:
Hi all,
On one of our client, I must to install an asterisk over a hi ability
cluster. I have no experience with clusters an linux neither asterisk.
Someone has installed an asterisk in a hi-ability enbviroment?
How do you install the cluster?
Witch solution did you use?
Disable Voice Activity Detection
ram wrote:
Hi
iam using asterisk 1.2 version
I have purchased g729 license from Digium
when iam making calls, iam getting this error ?
Jun 25 14:41:45 NOTICE[4424]: frame.c:183 __ast_smoother_feed:
Dropping extra frame of G.729 since we
KO, thank you very much.
i will try it.
2007/6/25, Steve Totaro [EMAIL PROTECTED]:
You could combine your two contexts or use goto.
Instead of:
[outgoing_RTB]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[outgoing_RDSI]
exten
I would say High Availability,
sorry for my english.
Any High availiability solution for asterisk?
VoipCrazy
2007/6/25, Steve Totaro [EMAIL PROTECTED]:
voip crazy wrote:
Hi all,
On one of our client, I must to install an asterisk over a hi ability
cluster. I have no experience with
I don't think my cards are bad, but maybe there is a problem with the
one. It has been two weeks since I put my ticket in with Digium...and
still no word. I am starting to get frustrated.
On 6/22/07, Daniel Hazelbaker [EMAIL PROTECTED] wrote:
Alex,
I had this problem with a new
Dear all
I have confusion how to asterisk genrate tone and what
ringing code use default 180 or 183 i have setup asterisk with mediant 2000
with avaya
[asterisk]-[mediant 2000][Avaya]
when i call from avaya side to --- asterisk i don't got ringback Sound so
You could combine your two contexts or use goto.
Instead of:
[outgoing_RTB]
exten =_9,1,Dial(ZAP/g1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten =_9,102,Hangup()
[outgoing_RDSI]
exten =_9,1,Dial(mISDN/1/${EXTEN},45,twW)
exten =_9,2,Hangup()
exten
On 6/25/07, Karl J. Vesterling [EMAIL PROTECTED] wrote:
Disable Voice Activity Detection
yes i have disabled at my eyebeam, still i see this error
iam using 1.2.18
ram
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- Original Message -
From: Darrick Hartman (lists) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, June 24, 2007 11:25 AM
Subject: Re: [asterisk-users] inband DTMF for g729
Gang Chen wrote:
On 6/22/07, Gary
There is a whole wiki page on the subject. Google is your friend.
http://www.google.com/search?q=high+availability+asteriskstart=0ie=utf-8oe=utf-8client=firefox-arls=org.mozilla:en-US:official
This is what I am currently playing with:
http://linux-ha.org
voip crazy wrote:
I would say High
Any High availiability solution for asterisk?
VoipCrazy
http://www.bicomsystems.com/files/projects/serverware/SERVERware.pdf
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Benny Amorsen schrieb:
MM Siemens GigaSet SL75
The SL75 is DECT, not Wifi.
Apart from that, was it really necessary to quote 20 lines and add a
ridiculous 15 line disclaimer telling me that I'm not allowed to read
the message?
There is a GigaSet SL75 WLAN.
Thanks Benny...
Let me give it a try...
Cheers,
Nitesh
Benny Amorsen wrote:
ND == Nitesh Divecha [EMAIL PROTECTED] writes:
ND Hello All, Recently I added some Nokia N95 customers and it worked
ND pretty good. Now the customers are complaining about the dialing
ND rules...
Greetings!
Due to high workload, I am currently checking and responding to e-mail twice
daily at 12:00 PM EST and 9:00PM EST.
If you require urgent assistance (please ensure it is urgent) that cannot wait
until either 12:00 PM or 9:00 PM, please contact me via phone at: 305-338-3867.
Thank
On 6/25/07, satish patel [EMAIL PROTECTED] wrote:
I have confusion how to asterisk genrate tone and what
ringing code use default 180 or 183 i have setup asterisk with mediant 2000
with avaya
I'm assuming that you're talking SIP here... typically, if Asterisk
receives a 180
Turn off debug
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Friday, June 22, 2007 3:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: [asterisk-users] 1.4.5
I am seeing a peculiar message on my console screen
Yes, it does.
---
Matthew Fredrickson
Software Engineer
Digium, Inc.
On Jun 23, 2007, at 7:48 PM, Ronaldo Z. Afonso wrote:
Hi all,
I was reading an IAX RFC, or a kind of, and it mentioned something
about
Call Path Optimization. Does Asterisk provide such a feature?
Thanks.
Ronaldo.
I can't find reference to TFTP for provisioning - does this phone support
it?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marcus Franke
Sent: Monday, June 25, 2007 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
This is the required dial plan:
0+61|XXX.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Withnall
Sent: Friday, June 22, 2007 5:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] international numbers...
Using
Dude you got to be freaking kidding me - are you really sending this
email to everyone who posts on the Asterisk list?
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL PROTECTED]
So, incoming calls on zap work just as I expect them - an intro is played, the
caller hits 1 for sale 2 for support or dials an extension. I'm using the
privacy option for all extensions. When calls come in from zap, they caller
is played the priv-recordintro recording, they say their name,
Hi guys,
sorry for the long e-mail, i'm only trying to give as much information
as i think is relevant to my problem (console log, sip.conf and
extension.conf parts).
i've been practicing with callback for a while, but i'm at a dead end.
I hope somebody can help me to move on.
i have
Hi list:
Is there any way or an idea of how to made a global queue policy. I need
to have a Global Policy or a common policy to many queues.
What i need is:
I have 20 agents they are members of 5 queues, i have a last recent
strategy for all the queues, the problem is that the
Senad Jordanovic wrote:
Any High availiability solution for asterisk?
VoipCrazy
http://www.bicomsystems.com/files/projects/serverware/SERVERware.pdf
Is this free?
Benefits over opensource packages?
Thanks,
Steve Totaro
___
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Thanks for reply dear
See i am going to explain my setup here
[asterisk]-[Mediant 2000]E1--[Avaya media g/w]
1) This is my setup i am useing asterisk 1.2.14 and this setup working fine but
one issuse is when i call from asterisk to avaya phone i got ringback tone in
my sip
Hi there,
I've asked this question to the BSD group too, but I'd like to know
whether anybody
else had similar experiences on Linux 2.6.20 etc.??
FreeBSD 6.2
Asterisk 1.4.5 (and 1.4.3 from ports)
Sip phone - PBX(*) -IAX2-VROUTER(*)- SIP-Voip provider
(SPA901 SPA922 phones)
We've see a
exten = +61242110,1,Goto(0${EXTEN:3},1))
Gary Mensenares wrote:
This is the required dial plan:
0+61|XXX.
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Kevin
Withnall
MF == Marcus Franke [EMAIL PROTECTED] writes:
MF There is a GigaSet SL75 WLAN.
MF http://gigaset.siemens.com/shc/0,1935,hq_en_0_122755_rArNrNrNrN,00.html
MF Hmm, I did not see any DECT SL75..
You are indeed correct, and I apologise.
I was thinking of the SL37; how I messed them up I don't
Steve Totaro wrote:
Senad Jordanovic wrote:
Any High availiability solution for asterisk?
VoipCrazy
http://www.bicomsystems.com/files/projects/serverware/SERVERware.pdf
Is this free?
Benefits over opensource packages?
Thanks,
Steve Totaro
Steve... (and anyone else)I made
On 09:52, Sat 23 Jun 07, Ronaldo Z. Afonso wrote:
Hi all,
Does anybody know any USB phone that I can use as an IAX Client?
Thanks.
For what I know, an USB phone is just an USB sound device
with a phonelike piece of plastic to hold the mic and
speaker.
So you can use it with every softphone
What does the cdr table you created in oracle look like ?
Tim.
On 20 Jun 2007, at 13:37, Everton Goularth wrote:
Hi All,
Thank's for your hint Tim Panton
I could connect my asterisk machine to my oracle machine.
I used unixODBC-2.2.11.tar.gz,
Ryan Goldberg wrote:
So, incoming calls on zap work just as I expect them - an intro is played,
the
Ah, ignore all that- it had to do with caller id being empty vs unknown or
something of that nature - at any rate some problem I can solve myself. I
jumped the gun by posting.
Ryan
Hello,
I've been racking my brain over this for much of the day so I thought
the list would probably be more helpful. A few days ago I upgraded
from Asterisk 1.2 to Asterisk 1.4.5. Everything appeared to be working
properly.
However, on the first business day, we realized that when transferring
Hi, I'm going to forward SIP request to special outbound SIP proxy with none
SIP port.
I have this in my sip.conf
[sip_proxy-out]
type=peer ; we only want to call out, not be called
username=408
host=192.168.0.95
outboundproxy=192.168.0.74
port=9097
I want a
To: [EMAIL
On Mon, 2007-06-25 at 12:51 -0500, falz wrote:
Hello,
I've been racking my brain over this for much of the day so I thought
the list would probably be more helpful. A few days ago I upgraded
from Asterisk 1.2 to Asterisk 1.4.5. Everything appeared to be working
properly.
However, on the
Hi,
I specific
outboundproxyport=9097
in version 1.4.4, but it doesn't work. It still connects sip port 5060.
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On 6/25/07, Carlos Chavez [EMAIL PROTECTED] wrote:
On Mon, 2007-06-25 at 12:51 -0500, falz wrote:
However, on the first business day, we realized that when transferring
calls (not using call parking, using the built in transfer buttons on
a Cisco 7960) would not work. This error would
Hi,
Is there a way to set the status of layer 2 in a BRI circuit to
either permanent or call. I have searched the wiki and can't find any info
on the subject but seem to recall a post a couple of years ago detailing the
process.
Thanks
Fadge
-Original Message-
From: [EMAIL
I think what Jared recommended, looking at the sip messaging, will help
you here. He means to type sip debug in the asterisk CLI and look for
hints that SDP is being specified in the conversation. If it IS being
specified, then check into NAT/firewall issues, as he recommended also.
Mojo
Hello All,
I apologize if this question has already been answered but how do you
transfer a call to a cell phone or another land line outside the PBX?
Setup
I have two pots lines into my current Asterisk Box. I have an outsides
sales guy who wants to work off his cell phone or transfer his
Looks like
outboundproxyport
doesn't support in 1.4.4
If you set the port, then it conflit with the one in To URI with host.
I saw the code for outboundproxyport from the source, but is it a bug?
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vant output of show dialplan.
Note that the sip calls come in on extension 666.
it's cursed,
Thanks much in advance.
Ryan
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Hi
Welcome to the Asterisk users mailing list.
When writing a message to the list, don't just reply to an existing
message.
When replying to a message from the list, please don't quote irrelevant
content.
And now to answer your question:
On Mon, Jun 25, 2007 at 08:16:09PM +0100, asterisk
Dear All
I have this setup
[asterisk][mediant2000]---E1 Trunk--[Avaya]
When i call from avaya to asterisk i got long ringing tone then hangup but
when i call from asterisk to avaya i got 4 ringback and then hangup with this
error
*CLI Jun
Hi all
On the Asterisk website in the blog its announced that in a next release
Asterisk would support dynamic DUNDi weitht values. I've installed Asterisk
1.4.4 (via aptitude install) but this doesn't seem to work. Has somebody some
experience with this or know whether this feature is already
Hi list:
I'm having the next problem, it appear that the application ChanIsAvail
is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS.
I add my dialplan and the output to the cli.
THanks.
In the example i'm dialing from extension SIP/112
My DialPlan Secction:
Dude you got to be freaking kidding me - are you really sending this
email to everyone who posts on the Asterisk list?
No, most likely he has an autoreply/vacation/out-of-office message enabled.
I would expect us to get more of them. Just be thankful he is on digest
mode!
John Faubion
Hi,
I ma using Asterisk 1.2.18 FreePBX 2.2.1. I have assigned every
users in office with Polycom with 2 extensions as below
555
8555
I have configured Follow-me to ring when the users doesn't picks the phone on
line 1(555) after 10 seconds then ring the line 2(8555). But this is
On Mon, 25 Jun 2007, Marcus Franke wrote:
Benny Amorsen schrieb:
MM Siemens GigaSet SL75
The SL75 is DECT, not Wifi.
Apart from that, was it really necessary to quote 20 lines and add a
ridiculous 15 line disclaimer telling me that I'm not allowed to read
the message?
There is a
Hello list,
AstPligg is a new Digg-like website devoted to * and VoIP news.
At the moment, it's in beta stage and very basic - no fancy custom
templates. It allows posting new stories, comments on stories, RSS feeds
and tags. Still, it can be very useful, as the number of * sites and blogs
I have two pots lines into my current Asterisk Box. I have an outsides
sales guy who wants to work off his cell phone or transfer his calls
from his extension and the main sales extensions. How can I do this right?
Do it right? You really haven't provided enough information to make the
right
Great! Another one. With such a catchy name too!
On Tue, 2007-06-26 at 01:42 +0200, lenz wrote:
Hello list,
AstPligg is a new Digg-like website devoted to * and VoIP news.
At the moment, it's in beta stage and very basic - no fancy custom
templates. It allows posting new stories, comments
i believe www.voipango.de sell them to US
On 6/26/07, Nick Seraphin [EMAIL PROTECTED] wrote:
On Mon, 25 Jun 2007, Marcus Franke wrote:
Benny Amorsen schrieb:
MM Siemens GigaSet SL75
The SL75 is DECT, not Wifi.
Apart from that, was it really necessary to quote 20 lines and add a
Greetings!
Due to high workload, I am currently checking and responding to e-mail twice
daily at 12:00 PM EST and 9:00PM EST.
If you require urgent assistance (please ensure it is urgent) that cannot wait
until either 12:00 PM or 9:00 PM, please contact me via phone at: 305-338-3867.
Thank
I am using VoiceOne http://voiceone.it/ as my management interface.
I am not 100% sure when it started, but my CDR is now full of s as
the DST instead of the actual dialed number.
As I understand it - it is because it is being recorded in the CDR
while in a macro (as below).
Is there any work
On 6/25/07, Alvaro Parres [EMAIL PROTECTED] wrote:
I'm having the next problem, it appear that the application ChanIsAvail
is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS.
I add my dialplan and the output to the cli.
This isn't really a problem with ChanIsAvail... it's
On 6/25/07, Hendrik Visage [EMAIL PROTECTED] wrote:
We've see a situation where the IAX2 appears to loose/drop the voice
data to be sent to the
SIP side of things. This happens semi intermittently, but we can
reliably regenerate it
at 40 alaw calls (even on a dedicated 1G network) and also
On 6/25/07, Andre Wangler [EMAIL PROTECTED] wrote:
On the Asterisk website in the blog its announced that in a next release
Asterisk would support dynamic DUNDi weitht values. I've installed Asterisk
1.4.4 (via aptitude install) but this doesn't seem to work. Has somebody
some experience with
Hello Friends,
I have successfully being able to initiate a automatic Call with AMI that
leads me to a Extension XXX.
In my extension.conf I have: exten = XXX,1,ChanSpy(SIP/).
The problem that I have is to listen to a Specific channel that's using SIP.
I tried out this:
exten =
John
thanks for the input.
forget about my right way ok!
by the way selling does not depend on the amount of lines you have and
we are very productive trust me
I have seen a million dollar corp work off four lines so your statement
is quite vague...
Otis
John Faubion wrote:
Hi,
I have been looking for an example of accomplishing this, but I've been
unable to locate something similar to what I'm trying to do.
Here's the scenario:
Users caller ID is set to their internal extension (200-250). This is set in
sip.conf for each user. Each user has a local DID as well
Yes. I have so
On 6/25/07, Nick Seraphin [EMAIL PROTECTED] wrote:
Is this strictly a European phone? I can't find anyone who is selling
them in the US... at least not a company I've ever heard of or dealt with
before.
Tried Amazon.com, voipsupply.com, Tech Data, and 3 pages of Google
6I am using asterisk 1.4.5 and storing cdr in mysql . Here's example
of one cdr generated
2007-06-26 00:44:28 682345xxx 6823456xxx s
macro-dialout-trunk SIP/343684-09544f20 SIP/provider-0938de98
Dial SIP/provider/1386734|300| 28 5 60 1
0
I've tried timing faxes two ways:
From a fax machine on a station port of an AltiGen PC/PBX served by an MCI
PRI calling back into the same PRI and reaching a RightFax server on a
station port behind the AltiGen.
From the same fax machine on the same station port of the AltiGen PC/PBX
served by
Dear ALL
I have asterisk with sip and it is integrated with avaya
through mediant
[*]-[mediant 2000]-E1--[Avaya]
Now i want to call transfer feature in asterisk means transfer call from one
phone 2 another phone how could it possible with asterisk
Regrads
Any recommendations on an economical layer 3 switch? Preferably something that
you have hands on experience with connecting to IP phones with attached PCs?
Specifically I need the ability to set the VLAN in the phone to tag voice
packets and to set a native VLAN on a per port basis on the
On 6/26/07, Marty Mastera [EMAIL PROTECTED] wrote:
Any recommendations on an economical layer 3 switch? Preferably
something that you have hands on experience with connecting to IP phones
with attached PCs? Specifically I need the ability to set the VLAN in the
phone to tag voice packets and
Brandon,
I wanted to show my support this module as well. I would appreciate
information on how to obtain the finished product and or help for beta
testing.
Kenny
On Wed, 2007-04-11 at 14:11 -0500, Brandon Kruse wrote:
I wrote a very extensive plugin for cacti to monitor asterisk.
It
This is due to changes in cdr in asterisk 1.4.5 so in all outgoing
dial from macro it shows 's' in cdr . Could this be a bug ? I doubt if
it was intended to be that way .
On 25/06/07, Troy - Purple Oranges [EMAIL PROTECTED] wrote:
I am using VoiceOne http://voiceone.it/ as my management
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