Hi Michael;
You tried iaxcomm pro as JIM is complainning from the
crashs.
PLease advise.
Regards
Bilal
-
iaxcomm pro??
On Tue, 24 Jul 2007 19:40:45 -0400, Jim Archer
[EMAIL PROTECTED]
wrote:
I tried several and had very poor luck with each I
tried. These
included
IaxComm,
Idefisk is now renamed to zoiper . http://www.zoiper.com/ :)
On 26/07/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Mittwoch, den 25.07.2007, 12:13 -0700 schrieb bilal ghayyad:
Hi BaharatSamaria;
Thanks for your kindly email.
Are (Xlite and phoner) IAX or SIP? From where I can
Am Mittwoch, den 25.07.2007, 12:13 -0700 schrieb bilal ghayyad:
Hi BaharatSamaria;
Thanks for your kindly email.
Are (Xlite and phoner) IAX or SIP? From where I can
download them (Xlite and phoner)?
I googled for xlite. One of the first matches was a wiki page on
voip-info.org, which in
Dear All
The setup is te110p with an 8 channels PRI to make and receive all calls.
SIP phones throughout the company.
TDM400p with 4 FXS modules to send/receive faxes and make credit card
transactions.
I have an analogue phone on the tdm400p for testing.
I can receive calls to the exten. There
On Thu, Jul 26, 2007 at 05:25:30PM +0530, [EMAIL PROTECTED] wrote:
Hi,
I am facing problem in configuring D-channel. I did the following
configuration for TE-120P card
/etc/zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
On 7/25/07, Jaswinder Singh wrote:
Idefisk/zoiper softphone is for IAX2 and it works fine almost
everytime . However there is more variety in sip softphones .
I think zoiper is much better than other iax2 softphones .
Feature wise you are quite right that Zoiper is pretty neat.
But
Sangoma gives EXCELLENT technical support.
I would suggest you try there first.
The few problems I have had with installation were addressed promptly
and when driver fixes proved necessary, corrected in short order.
Also the cards have a 5 year warranty!
John Novack
Nhadie Ramos wrote:
Hi,
Do you have any extension in default context of your extensions.conf
file to accept incoming calls ?
It must be something like;
exten = 12345678,1,Answer()
exten = 12345678,2,Playback(Welcome)
...
12345678 = The DID number you are calling to reach E1
Idris
-Original Message-
From:
On Thu, 2007-07-26 at 11:06 -0500, Jay Moore wrote:
So here is my question:
In this format: 1|2|3|4|5|6,
1 - ?
2 - ?
3 - queue in question?
4 - agent answering the queue?
5 - queue event?
6 - queue event info?
Is that correct? What are options 1 and 2? Times of some sort I'm
PS. Check this out:
http://bugs.digium.com/print_bug_page.php?bug_id=2471
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671
___
--Bandwidth and Colocation Provided by
Hello Marco,
On 7/27/07, Marco Mouta [EMAIL PROTECTED] wrote:
hi,
The
VoiceMailhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2application
uses
*/usr/sbin/sendmail* to mail voicemail messages to users. This can be any
sendmail-compatible MTA. In practice you can use
Hello all,
Where can I find the complete list of default Asterisk (telephone) numbers
and maybe the other special numbers that are need to be preserve and not use
for setting up own dial plan?
Thank you.
GNUbie
___
--Bandwidth and Colocation Provided
GNUbie wrote:
Hello Tzafrir,
On 7/27/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
You need to have a package that provides /usr/sbin/sendmail . While you
can get away with using nullmailer or ssmtp (that don't spool mail
locally), I would recommend you to install postfix or exim, so a
Hello Tzafrir,
On 7/27/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
You need to have a package that provides /usr/sbin/sendmail . While you
can get away with using nullmailer or ssmtp (that don't spool mail
locally), I would recommend you to install postfix or exim, so a
temporary problem won't
On Fri, Jul 27, 2007 at 08:55:01AM +0800, GNUbie wrote:
Hello all,
I am running Asterisk-1.4.5 on my Debian GNU/Linux Etch here and I want to
send the voicemails as attachment to e-mails and delete the voicemails from
my PBX once it has been sent. But, I don't have a running MTA here even
hi,
The
VoiceMailhttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+voicemail2application
uses
*/usr/sbin/sendmail* to mail voicemail messages to users. This can be any
sendmail-compatible MTA. In practice you can use
Sendmailhttp://sendmail.org/,
Postfix http://postfix.org/, Exim
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
What do you get with:
iax2 show registry
- --
Cheers,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://feeds.feedburner.com/AsteriskNews (Daily Asterisk
Hello all,
I am running Asterisk-1.4.5 on my Debian GNU/Linux Etch here and I want to
send the voicemails as attachment to e-mails and delete the voicemails from
my PBX once it has been sent. But, I don't have a running MTA here even on
the PBX itself. I just want to send the e-mails to my
FERNANDO VILLARROEL wrote:
Hello list, i need help.
My problem is that when I want to call (using ISDN
phone or internal SIP client) via the Sip provider a
real phone number (ISDN phone or internal SIP
Asterisk SIP ), I get a ring tone. When I
now decide to hang up (e.g. if
Hi Peder,
You tried blanking the caller ID field and it didn't work?
i.e., exten = ...,n,Set(CALLERID(all)=)
It worked for me, although my media gateway was not a Cisco one.
Whether SetCallerPres() will work depends entirely on what it
accomplishes. Does it just alter the cosmetic From:
Hello Jay,
Sounds like quite a complicated set up. Most queue statistics packages
will break your callers down depending on which queue they were actually
answered in (or hung up on).
If you want your stats listed as if the callers were in a single queue,
you can sign up for a FREE
Is there more than one display IE in the original ISDN setup message coming
from the Telco?
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oscar Patricio
Sent: Thursday, July 26, 2007 5:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
That is correct. The X100P only detects voltage drop, not polarity
reversal.
Walter Willis wrote:
i am have x100p and not work fine, no detect polarity, and much problems
with asterisk 1.2 to up.
:S
On 7/25/07, Mike Wright [EMAIL PROTECTED] wrote:
John Novack wrote:
Mike
Grandstream GXP-2000 with firmware 1.1.4.18 (beta) fixes an issue where
the phone did not send rtp keepalives when on mute (resulting in
disconnect from tech support hold and concalls)
A side effect seems to be that Asterisk pops the following warning on
the console...
Jul 26 14:06:35
Some time in the next two weeks, Digium will be shutting down our FTP
server, located at ftp.digium.com, and begin using only the existing
HTTP server on the same system instead.
We have decided to only offer our public downloads over the HTTP
protocol, not the FTP protocol, primarily for reasons
Are sites listed by IP or DN. If IP, dumb question but did it change? If DN,
can you resolve it from the respective boxea?
Dave Bour
Desktop Solution Center
905.381.0077
[EMAIL PROTECTED]
For those who just want it to work...
Giving you complete IT peace of mind.
(Sent via Blackberry -
Jared Smith wrote:
On Thu, 2007-07-26 at 09:37 -0500, Jay Moore wrote:
My boss would like some statistics on how many calls are answered out of
specific queues during a given time period, and I'm not sure how exactly
to obtain those stats.
It sounds like you've got quite the queue
i am have x100p and not work fine, no detect polarity, and much problems
with asterisk 1.2 to up.
:S
On 7/25/07, Mike Wright [EMAIL PROTECTED] wrote:
John Novack wrote:
Mike Wright wrote:
Just purchased a Motorola Wildcard X100P ...
but the button pressed generates no tone; on button
On 7/26/07, Matt Hoppes wrote:
I would agree... intended to send that to biz, sorry.
I see that you also sent it to the biz-list.
And if you fail the lie detector test how about agreeing to a
full boycott of your service or at least a M.L.D.P. (mailing
list death penalty :-) ?
--
On 7/26/07, satish patel [EMAIL PROTECTED] wrote:
Dear all
I have asterisk with SNOM SIP phone i want to confrance to
my users how to configure confranceing in asterisk meetme.conf is fine but
is there any otherway to confranceing
If the End device support conference still
Matt wrote:
I can think of no reason to ever need to do this.
You must not peer with Level3, or with anyone who peers with Level3 via IAX :)
Why would anyone want to send traffic/calls to Level3? A search of the
mailing list archives is all that is needed to know that. 8-)
I didn't think
Not likely.
#1, I have a public IP on that firewall.
#2. If I block 4569 at our firewall, then it goes from closed to
stealth. If I forward the port, it goes from stealth to closed.
The iaxping tool (http://www.bpvn.com/asterisk/iaxping.zip) has no
problems pinging the box from the lan, and our
what if your internet provider is blocking inbound 4569 ?
--
On 7/26/07, Michael Munger wrote:
Dear All:
I have several boxes that up and running just great, then we changed
internet equipment due to a lightning strike, now all my inbound IAX
connections (iax2 show peers) have
On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote:
;; dialtone in the background isn't there any more
;; dialed '305'
;; everything from here is exactly as expected.
OK, I missed this in the first email you sent... Asterisk is playing
dialtone *on top* of the background message the first
matt,
I just had the same problem... does your CLI report 'unable to
create channel Zap/#'
post the CLI output to help us determine the problem.
daveC
Matt Scott wrote:
Dear All
The
setup is te110p with an 8 channels PRI to make and receive all calls.
SIP phones throughout
Hi!
Thank you all for the info!
But I think I haven't explained my scenario well enough.
I am not relaying the calls to SIP.
What happens is the following (the scenario is: a call started from an
ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the
remote party):
1. A
Hi guys,
Is there any option to retrieve queue stats (particulary am interested
in the time of currently longest waiting caller) from the dialplan?
Thank, Alex
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users
Hi Saqib,
Architecture is depend on what service you want to deliver.
Voip is more cheaper then pstn for interoffice connectivity.
But consider regulatory issue before using it.
visit http://www.voip-info.org/wiki-Asterisk for complete detail.
Regards
Nasir iqbal
On Wed, 2007-07-25 at
Dear all
I have asterisk 1.2 with mediant2k i have create SIP Trunk from
asterisk 2 mediant and my PRI terminated on Mediant 2000 E1 port now everthing
is fine but problem is when i call to somebody outside and he/she disconnect my
phone but my asterisk counitine ringing my SIP
I am experiencing a change in behaviour of my Queues in 1.4.9 vs 1.4.8.
I do not pass the 'n' option to any call to Queue() in my dialplan. Yet
since I upgraded to 1.4.9, I have occasionally seen this on my console:
-- Nobody picked up in 2 ms
-- Exiting on time-out cycle
That log
features.conf
On 7/26/07, GNUbie [EMAIL PROTECTED] wrote:
Hello all,
Where can I find the complete list of default Asterisk (telephone) numbers
and maybe the other special numbers that are need to be preserve and not use
for setting up own dial plan?
Thank you.
GNUbie
Hello Mark,
On 7/27/07, Mark Burrows [EMAIL PROTECTED] wrote:
Can someone suggest a starting point on learning Linux?
First of all, welcome to the community! =)
I may consider myself as an experienced systems/network administrator but
with Asterisk and telephony, I am still newbie to
Hi,
the first thing I did with Asterisk is listening to
`demo-congrats' by Xlite on the same machine. This works
perfectly. The config files are those shipped with the
package.
Now I want to listen to it over ISDN/Capi but I don't
succeed.
My `capi.conf' is like show in many tutorial on the
I just saw this on my console:
[Jul 26 11:36:30] WARNING[8667] file.c: File vm-duration does not exist in
any format
[Jul 26 11:36:30] WARNING[8667] file.c: Unable to open vm-duration (format
0x4 (ulaw)): No such file or directory
Thinking I might have lost a file during a fsck or something, I
Hello list, i need help.
My problem is that when I want to call (using ISDN
phone or internal SIP client) via the Sip provider a
real phone number (ISDN phone or internal SIP
Asterisk SIP ), I get a ring tone. When I
now decide to hang up (e.g. if
nobody answers), the called telephone
Does anybody know if SetCallerPres works on calls via SIP through a
Cisco gateway? We are trying to mark outbound calls as anonymous and we
set it to prohib, but calls still show outbound callerid. We are SIP
from * to the Cisco gateway and then PRI outbound. If we strip the
callerid num,
HI All,
I’m new to Asterisk and also to Linux. I have a large IVR project that I’m
about to embark on. I’m new to programming; new to Linux and new to
Asterisk. I think I’m about to climb a steep learning curve. I have an
existing IVR which is getting on for nine years old and is no longer
Btw are the phones behind NAT ?? Also you can try some softphone and make
sure that this problem is caused by snom phones or some other factors ..
On 25/07/07, Michael J. Liberatore [EMAIL PROTECTED]
wrote:
I thought it was the fios service but now I realize it's the snom 360!
It doesn't hang
What do you get with:
iax2 show registry
homer*CLI iax2 show registry
Host UsernamePerceived Refresh State
64.85.162.136:456906*** 68.XX.XX.XX:4569 300 Registered
is that bad?
Patrick
Hopefully that helps clarify things!
It does immensely. Thanks a ton!
Jay
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To UNSUBSCRIBE or update options visit:
Andrew,
Could you elaborate on how you configure the MWI of the mobile device to
use asterisk voicemail?
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
___
--Bandwidth and Colocation Provided by
This may not be the best solution for you, but it's the only one I can
speak for. We use QueueMetrics for our queue information and
reporting. There is a small cost for it, but it is worth every
penny.
On 7/26/07, Jay Moore [EMAIL PROTECTED] wrote:
Greetings, list!
My boss would like some
mailing list archives is all that is needed to know that. 8-)
We've never had any issues with L3 and are very happy.
I didn't think that Level3 supported IAX connections. If you are using
an ITSP that uses Level3, I would hope the ITSP would be using inband
DTMF on SIP for their
I would agree... intended to send that to biz, sorry.
On 7/25/07, Anthony Francis [EMAIL PROTECTED] wrote:
Matt wrote:
If you have been affected by the SunRocket / ALLO folding issue,
ChiliTech would like to extend our hand to you to help you in this
time. We will transfer your numbers
David Boyd wrote:
On Wed, 2007-07-25 at 13:02 -0500, Eric ManxPower Wieling wrote:
Short Answer: No.
Long Answer: Maybe. If you can get your device to send inband DTMF and
tell Asterisk you are using INFO or RFC2833 DTMF, then Asterisk should
just pass the DTMF as audio. Then if the call
Dear All:
I have several boxes that up and running just great, then we changed
internet equipment due to a lightning strike, now all my inbound IAX
connections (iax2 show peers) have unknown status. If I log into the
remote boxes, it says Request sent.
The authentications haven't changed at all,
Thanks for the replies.
I decided to go with the USB channel bank. I hope everything will go
alright.
Lars
--
Let's not complicate our relationship by trying to communicate with each
other.
___
--Bandwidth and Colocation Provided by
Dear all
I have asterisk with SNOM SIP phone i want to confrance to my
users how to configure confranceing in asterisk meetme.conf is fine but is
there any otherway to confranceing
-
Got a little couch potato?
Check out fun summer
Oscar Patricio wrote:
3. The CONNECT that is sent from span 2 to the PBX does not have the
Display IE. The asterisk strips this IE from the CONNECT message.
This is an incorrect statement; 'strips' would imply that Asterisk is
just forwarding the CONNECT message from one PRI to the other, but
also have a look on
http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
On Thu, 2007-07-26 at 20:57 -0600, Al lists wrote:
features.conf
On 7/26/07, GNUbie [EMAIL PROTECTED] wrote:
Hello all,
Where can I find the complete list of default Asterisk
I can think of no reason to ever need to do this.
You must not peer with Level3, or with anyone who peers with Level3 via IAX :)
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To UNSUBSCRIBE or
Andrew Kohlsmith wrote:
On Thursday 26 July 2007 12:51:06 pm Eric Chamberlain wrote:
Could you elaborate on how you configure the MWI of the mobile device to
use asterisk voicemail?
yes, please explain. SMSing the phone doesn't light MWI, unless you get
access to the raw SMSC, as all the
Yes, it is a Blue Digium IAXy.
It is on my local LAN , so the Linksys SIP is working fine.
It was just a surprising discovery since Digium's owner defined IAX2, specified
that there can be no in band DTMF and then Disgium
left this out of the IAXy.
I believe that they assumed that it would
I seem to be missing digits with a PRI.
I added dtmf logging in logger.conf
This does not happen a-lot but it does happen a number of times over the
day.
I have watched a few times for calls coming in and the logger
only showed me 09 instead of 209.
I contacted my provider they checked it out
I'll take either
Actually now that I have had a chance to think about what I did (sorry
bad week here). Yes, I will admit I did patrionize the users list...
sorry if I offended anyone. I just figured I'd try to help any
SunRocket users out that may not be on the biz list.If you review
On Thu, 2007-07-26 at 09:37 -0500, Jay Moore wrote:
My boss would like some statistics on how many calls are answered out of
specific queues during a given time period, and I'm not sure how exactly
to obtain those stats.
It sounds like you've got quite the queue setup (although I don't
Yes it's possible.
It's also possible to have Asterisk try and find the person in the field
and either connect the call or deliver the message.
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Mark,
Welcome to the club. Learning Linux can be a daunting task. After working
with it for the last decade, I am still learning. My best recommendation is to
play with it on a test box, and post questions to a related community forum
if you get stuck on something. If you are looking for
James FitzGibbon wrote:
Looking back at my logs, there are semi-regular instances of this error
message. In a default setup, it's only used if the message is more than
2 minutes long, which I guess most of my user's VMs aren't.
This is my fault; we have a pending list of sounds to be
Grab a network trace (with e.g. Wireshark) and look at the payload type
and lengths of the RTP keepalive messages - if you post this information
to the list I'm sure someone will comment on what's happening.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hi all,
Has anyone made up custom ring tones for the Polycom SIP phones? We use
different rings for different lines, but the ones it comes with are all very
similar. In the interesting of sharing, here's one I made up for paging:
PAGE_BEEP se.pat.ringer.13.name=Page Beep
Yes, have them all meet in the cafeteria for brunch.
On 7/26/07, satish patel [EMAIL PROTECTED] wrote:
Dear all
I have asterisk with SNOM SIP phone i want to confrance to my
users how to configure confranceing in asterisk meetme.conf is fine but is
there any otherway to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Patrick Buller wrote:
What do you get with:
iax2 show registry
homer*CLI iax2 show registry
Host UsernamePerceived Refresh State
64.85.162.136:456906*** 68.XX.XX.XX:4569 300 Registered
On Thu, 2007-07-26 at 09:37 -0500, Jay Moore wrote:
My boss would like some statistics on how many calls are answered out of
specific queues during a given time period, and I'm not sure how exactly
to obtain those stats.
It sounds like you've got quite the queue setup (although I don't
Hi,
I am facing problem in configuring D-channel. I did the following
configuration for TE-120P card
/etc/zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
/etc/asterisk/zaptel.conf
group=1
Greetings, list!
My boss would like some statistics on how many calls are answered out of
specific queues during a given time period, and I'm not sure how exactly
to obtain those stats. Here's how our queue system works.
1) Call comes in and enters our 'ring' queue where the phones ring for
Jay,
You could try ASTassistant. It has Queue information at a glance.
http://www.astassistant.com
- Original Message -
From: Jay Moore [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, July 26, 2007 7:37 AM
On Thursday 26 July 2007 12:51:06 pm Eric Chamberlain wrote:
Could you elaborate on how you configure the MWI of the mobile device to
use asterisk voicemail?
yes, please explain. SMSing the phone doesn't light MWI, unless you get
access to the raw SMSC, as all the email gateways just mangle
On 2007 Jul 26 (Thu) at 09:32:00 -0400 (-0400), Jared Smith wrote:
:On Wed, 2007-07-25 at 15:06 -0700, Peter Hessler wrote:
: ;; dialtone in the background isn't there any more
: ;; dialed '305'
: ;; everything from here is exactly as expected.
:
:OK, I missed this in the first email you sent...
Mark Burrows wrote:
HI All,
I’m new to Asterisk and also to Linux. I have a large IVR project that
I’m about to embark on. I’m new to programming; new to Linux and new to
Asterisk. I think I’m about to climb a steep learning curve. I have an
existing IVR which is getting on for
I have just started working with Asterisk and have run into a road block
concerning IAX and an inbound DID from callwithus.com. I am getting
nowhere and I don't really know how to isolate the problem. The asterisk
version is 1.2.7 on ubuntu, sits behind a firewall with iptables. I can
connect
Hi James,
I have one posting for the Cisco7970 ringtone, which you can adapt for
the Polycom. It's here: http://www.voipphreak.ca/archives/349
I also have another one I posted for the Polycom Ringtones with a
bunch of tunes. It's here:
http://www.voipphreak.ca/archives/78
Hope these help :)
Mark Burrows wrote:
HI All,
Im new to Asterisk and also to Linux. I have a large IVR project that
Im about to embark on. Im new to programming; new to Linux and new to
Asterisk. I think Im about to climb a steep learning curve. I have an
existing IVR which is getting on for
Greetings, list!
My boss would like some statistics on how many calls are answered out of
specific queues during a given time period, and I'm not sure how exactly
to obtain those stats. Here's how our queue system works.
1) Call comes in and enters our 'ring' queue where the phones ring for
That's actually a good idea.
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, July 26, 2007 9:23 PM
Subject: Re: [asterisk-users] Asterisk Conference Call
Yes, have them all
Hi,
I am facing problem in configuring D-channel. I did the following
configuration for TE-120P card
/etc/zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
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