Nope. It isn't active. I even factory reseted the phone but still the
same. One more piece of information: it works just fine in 1.2b1. I
beginning to think it could be a bug in 1.2b2.
Any other ideas/suggestions?
Thanks,
Waldo
On Nov 5, 2005, at 9:10 PM, C F wrote:
You sure that the DND (
On Sat, Nov 05, 2005 at 02:18:00PM -0700, David D. Dixon wrote:
> I've previously run AAH (installing from the ISO and tar), but this time I'm
> doing my own install and am having problems getting AMP to work right. Any
> time I try to modify the configuration, I get an error like this:
>
> War
On Sat, Nov 05, 2005 at 07:29:18PM +0100, FaberK wrote:
> Fedora Core 3
> kernel-0-2.6.9-1.667 and kernel-2.6.12-1.1380 (same results)
> Sangoma 102
> Concerning udev, I've read that it uses hotplug and if I'm not wrong,
> I remember that zaptel got conflicts with hotplug. But maybe I'm
> confusing
Thanks it worked. However I had one other less obvious problem.
While playing around with my card I executed
/sbin/ztcfg - with fxsls settings once upon time and that was
my obstacle.
Doing that again with fxsks settings fixed problem.
Does it mean that ztcfg is some kind of configuration utili
On Saturday 05 November 2005 23:24, Dustin Goodwin wrote:
> I do have an older card. I believe I purchased it before the FXO module
> started shipping. Can I do anything to resolve or this hw problem with
> the card?
Contact Digium and get the carrier card replaced under warranty.
-A.
___
On Saturday 05 November 2005 23:22, Branko Samardzic wrote:
> I am pretty new to * but it seems that hang-up detection is hot topic. Does
> anyone know how to setup my * for proper detection of hang-up condition.
> PSTN lines I use for testing are just plain ones (don't even have CID).
> Whatever I
On Saturday 05 November 2005 23:03, Gary Eck wrote:
> "Brand new " cards, from a recommended Digium distributor.
What rev? It'll say "Freshmaker Rev x" on the blue board. I *think* it was
corrected around Rev E or F IIRC, but I don't know for certain.
> I spoke with Digium and the distributor
Hello all,
I have [EMAIL PROTECTED] working pretty well, and have been trying to fix
a few things. Today I tried to make Asterisk send voicemails to my
e-mail account. I installed Webmin, which worked fine, and tried a few
things that I read on line, mainly involving the sendmail application
and f
I do have an older card. I believe I purchased it before the FXO module
started shipping. Can I do anything to resolve or this hw problem with
the card?
- Dustin -
Andrew Kohlsmith wrote:
On Saturday 05 November 2005 22:33, Gary Eck wrote:
I have popping with FSO modules only on channel
Hi,
I am pretty new to * but it seems that hang-up detection is hot topic. Does
anyone know how to setup my * for proper detection of hang-up condition.
PSTN lines I use for testing are just plain ones (don't even have CID).
Whatever I setup (busydetect, callprogress, hanguponpolarity etc...) didn
"Brand new " cards, from a recommended Digium distributor.
I spoke with Digium and the distributor tech support on it - gave them
serial numbers, board rev numbers, etc. No one mentioned this.
Do you know what the current version of the board is?
Thanks
-Original Message-
From: [EMA
On Saturday 05 November 2005 22:33, Gary Eck wrote:
> I have popping with FSO modules only on channel 1 - the other 3 channels
> are clear.
That was corrected a long time ago. You must have an older rev TDM400 carrier
card.
-A.
___
--Bandwidth and Col
I have popping with FSO modules only on channel 1 - the other 3 channels
are clear.
Moved slots, there are no shared interrupts, replaced the TDM400 card
and all modules - still the same result.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dustin
Goodwi
I tried configuring vbuzzer on a Azacall 200... Works perfectly
Not on * .. HELP HELP HELP
- Original Message -
From: "Hitesh Sharma" <[EMAIL PROTECTED]>
To:
Sent: Saturday, November 05, 2005 8:25 PM
Subject: [Asterisk-Users] Fw: Inbound Calls on Asterisk from VBuzzer
> To Add Her
To Add Here is the VBuzzer Peer Definition
[vbuzzer]
type=peer ; we only want to call out, not be called
context=default
port=80
username=<>
secret=<>
host=vbuzzer.com
fromuser=<>
fromdomain=vbuzzer.com
disallow=all
allow=ulaw
allow=alaw
allow=g729
User-agent=vbuzzer/1.0
Insecure=yes
Hi
Any one got Inbound Calls from VBuzzer working on Asterisk
I am tried it hard and will be bald in few hours
The Call comes in... But Gets a 407 Authentication Required from Asterisk
Here is the SIP Log
Call Comes in from VBu
You sure that the DND (Do Not Disturb) button is not active on the UIP200?
On 11/4/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> I am running * 1.2b2 with some UIP200 phones and a bunch of X-Pro
> phones.
>
> All phones register fine with * and I can place outbound calls with
> no problem.
>
>
If you are using SIP then you can do this by making sure that
canreinvite is set to yes in sip.conf for those 2 sip clients and:
1. No transcoding is taking place (you are using the same codec on both ends).
2. You don't have anything in the dial command that forces asterisk to
keep the stream (li
My phones have some settings, which I do not understand. Maybe somebody
can help me here:
1. NAT
Nat traversal enabled, stun, disabled
Nat addr stun01.sipphone.com
Nat ttl 30
I played with Nat traversal:
enabled: the phone has NO audio, well it was just a white noise
On Saturday 05 November 2005 19:54, Paul wrote:
> I have no problem with origination but I was trying to use their
> termination services and the log says all calls are rejected. I used the
> setup they emailed me back in June. If anybody here is using their
> termination sucessfully I would apprec
I have no problem with origination but I was trying to use their
termination services and the log says all calls are rejected. I used the
setup they emailed me back in June. If anybody here is using their
termination sucessfully I would appreciate you sharing the config
snippets. Leave out yo
I get the same thing too.. Happens quite often for me. Its just
something I have come to live with with voipjet..
-Gerard
Garth Summey wrote:
Don't think there is anything wrong with your setup. We get the same
thing... Maybe they're down, but I would like a third opinion...
G
Michaël Gau
Don't think there is anything wrong with your setup. We get the same
thing... Maybe they're down, but I would like a third opinion...
G
Michaël Gaudette wrote:
Hi,
I`ve just tried the Voipjet 0.25$ test account, following everything the web
site told me to do (see below).
When I dial a loc
Hi,
I`ve just tried the Voipjet 0.25$ test account, following everything the web
site told me to do (see below).
When I dial a local canadian number, or even their own example (the New York
public library) the call seems to be accepted, but before it does anything I
get two lines following the "u
In CVS Head from both 2005-08-28 and 2005-11-03, I've created a SIP peer
and user with the same settings. When this peer/user sends a SIP invite
to the asterisk system running realtime, it gets a 404 not found back,
and the realtime system says it's unable to find a match for
:5060. However if I
I've
previously run AAH (installing from the ISO and tar), but this time I'm doing my
own install and am having problems getting AMP to work right. Any time I
try to modify the configuration, I get an error like
this:
Warning:
fopen(/etc/asterisk/vm_general.inc): failed to open stream: P
How can one extend the time allowed for registration?
In order not to get time out errors?
Christian
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman
On Saturday 05 November 2005 15:55, Dustin Goodwin wrote:
> I have read a lot online about interrupt related cracking/popping noise
> issues on Digium cards. The weird thing is I experience it with the FXO
> port on my TDM card but not on the FXS. Does this make any sense? I had
> assumed an interr
Here you go - place it in <~/Library/Application Support/BBEdit/
Language Modules>. It's not complete, but I add new keywords to it as
I go along. It is also case-sensitive (my preference - you can turn
this off).
AsteriskCodelessLanguageModule.plist
Description: Binary data
I'd like to
I have read a lot online about interrupt related cracking/popping noise
issues on Digium cards. The weird thing is I experience it with the FXO
port on my TDM card but not on the FXS. Does this make any sense? I had
assumed an interrupt problem. But if the FXS port is working why would
the FXO
Hello,
can somebody tell me if following is possible and if yes, how?
Assume we have a Asterisk server to which two VoIP clients are connected
over the internet (i.e. not internally). Now I would like to avoid
having the connection run over the server but would like the server to
tell the cli
They have two servers, try the other one and see what you get.
voip-co1.teliax.com
voip-co2.teliax.com
They might be on the same subnet, but it seems to me they had a
problem recently with an upstream link and one server was working
reasonably well while the other was not. Just a thought.
--
I'm interested.
Thanks,
Waldo
On Nov 5, 2005, at 2:54 PM, Anthony Rodgers wrote:
And, I couple of times now I have offered to post a BBEdit language
module to the wiki, but have no idea where to put it.
Last chance for anyone who's interested...
Regards,
--
Anthony Rodgers
Business Sys
I've been scouring the mailing list archives for an answer to this, and
cannot find one. I'm hoping someone else out there has run into this.
I'm running a Lucent TNT with TAOS 11.0.2 and trying to get it to
process VOIP calls via Asterisk. The TNT is currently accepting dialup
calls and functio
And, I couple of times now I have offered to post a BBEdit language
module to the wiki, but have no idea where to put it.
Last chance for anyone who's interested...
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www
I use traceroute to get the hop list and then compare some average ping
times. This gives a clearer picture and sometimes shows the average ping
time to the ultimate destination looking a lot better than what
traceroute shows. In this case it came out worse. I pinged both IP's 15
times and then
Use a line like this in your dialplan. I'll post a sample out of mine.
exten => _NXX,1,Dial,IAX2/[EMAIL PROTECTED]/1780${EXTEN}
That line is setup so any 7 digit numbers will be marked as belonging to
1-780.
Darren Wiebe
[EMAIL PROTECTED]
Jason Brashear wrote:
How do you set it up so
The strange thing about Teliax latency is they appear to be located in
Denver and it's about 55ms round trip to the Denver router on Level3
network. But then between the L3 router in Denver and the Teliax host
appears to be another 30ms round trip. Which from a distance point of
view makes no s
Fedora Core 3
kernel-0-2.6.9-1.667 and kernel-2.6.12-1.1380 (same results)
Sangoma 102
Concerning udev, I've read that it uses hotplug and if I'm not wrong,
I remember that zaptel got conflicts with hotplug. But maybe I'm
confusing (terrible headache!)
Thanks a lot!
2005/11/5, Tzafrir Cohen <
Hello everybody...
I´m having huge problems getting Asterisk registered to my german VoIP
Provider.
I get the following error message:
NOTICE[8308]: chan_sip.c:5247 sip_reg_timeout: -- Registration .
timed out, trying again (attempt ###)
Now my setup is DSL and IPCop as my router. My guess is
i think, TDMoE is not supported/developed anymore. This is known bug.
Franz Wu wrote:
Hi all
my system 1:
celeron 1.2GHz + intel 810e (asus TUW-LA) + 256MB SDRAM
onboard vga (intel 810e chipset)
RTL8100 NIC
debian sarge 3.1r0a / kernel 2.6.8-2-686
asterisk / libpri / zaptel from CVS HEAD @ 2005
How do you set it up so that you don't have to dial you area code ie 512 ?
-J
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-user
Maybe someone can explain how the Digium
Nightly works.
At ftp://ftp.digium.com/pub/nightly - As far as
I can tell, there is a file posted regardless if any changes were made -
True?
The strange part is the file dates inside the "tar"
- I would expect dates that were more current.
For
[EMAIL PROTECTED] wrote:
> I tend to agree with you, my experience with Teliax has been decent,
> and getting better. If only I could get to them at under 20ms though,
> right now my latency is about 75ms whereas voipjet comes through at
> 19ms.
Where are you located?
--
Cheers,
Matt Riddell
Thanks Armin, this version is working, but I still have an undefined symbol
in another module:
...
[pbx_wilcalu.so]Nov 5 18:51:12 WARNING[11348]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol:
ast_pthread_create
Nov 5 18:51:12 WARNING[11348]: loader.c:
I tend to agree with you, my experience with Teliax has been decent,
and getting better. If only I could get to them at under 20ms though,
right now my latency is about 75ms whereas voipjet comes through at
19ms.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] O
Please try chan_capi-cm CVS HEAD on sourceforge.net
Armin
On Sat, 5 Nov 2005 [EMAIL PROTECTED] wrote:
> Hello all,
>
> I've been using chan_capi-cm-0.6 as CAPI channel driver, the driver was
> working fine until I've reinstalled asterisk last week. I retrieved the
> latest asterisk version from
Hello
Try the CVS version it work's for me
Thierry
[EMAIL PROTECTED]
Tel : +33 (0)3 90 40 06 75
Fax: +33 (0)3 90 40 06 76
http://www.widevoip.com
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De la part
> de [EMAIL PROTECTED]
> Envoyé : samedi 5 novembr
Hello all,
I've been using chan_capi-cm-0.6 as CAPI channel driver, the driver was
working fine until I've reinstalled asterisk last week. I retrieved the
latest asterisk version from cvs and then build and installed it.
When rebuilding the capi channel driver with the latest asterisk headers I
r
> Rich Adamson wrote:
> > ; Calls directed to Teliax.com
> >
> > exten => _1NX,1,Set(CallerIDnum=4024325395|a)
> > exten => _1NX,2,Set(CallerIDname=NPI|a)
> > exten => _1NX,3,Dial(IAX2/teliax
On Sat, Nov 05, 2005 at 01:59:43PM +0100, FaberK wrote:
> Hi guys,
> in my 3rd asterisk installation, I have a problem with zaptel modules.
> I use the CVS.
> Instead of obtaining, for example, zaptel.o I got zaptel.ko.
> What is the reason?
That you use kernel 2.6. Do you have kernel 2.4 or 2.6?
There is a fairly steep cost to implement that function, and the majority
of the smaller itsp's don't want to impact their margins with it.
> oh and one tech from them support said they will be handling all cname soon
> .. like ETA 1 week
>
> can't wait to get my FBI na
No !
Asterisk should send the invite request to sip proxy .
Harry
--- Walter Willis <[EMAIL PROTECTED]> a écrit :
> the ser an asterisk run in the same box???
>
> redirect host + port :)
>
>
>
>
> 2005/11/4, harry gaillac <[EMAIL PROTECTED]>:
> >
> > Hello,
> >
> >
> > I wish to setup this
On Saturday 05 November 2005 02:54, Tzafrir Cohen wrote:
> Asterisk here falls under "desktop": Much like interactive desktop apps
> it need timely response rather than more time at the CPU.
I am not so sure; asterisk is a multithreaded app, yes, but I would think that
kernel preemption would be
Rich Adamson wrote:
; Calls directed to Teliax.com
exten => _1NX,1,Set(CallerIDnum=4024325395|a)
exten => _1NX,2,Set(CallerIDname=NPI|a)
exten => _1NX,3,Dial(IAX2/teliaxout/${EXTEN})
This f
Jason Brashear wrote:
OK I am exhausted.
I can't seem to figure out how to send a caller ID along with a
Outbound call.
Can you believe that I got Vonage to reset my Cisco ATA for $15.00
I then canceled my account!
Well I was with them for over two years, now I am running Asterisk like the
b
Stefan Reuter schrieb:
...
So having a look at Asterisk 1.2-beta2 is probably the way to go.
Great! Yes, this will solve my problem.
Let's upgrade ...
Thanks!
Roger.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users maili
> I want to track the ringing event of the outgoing channel.
> Unfortunatelly the link event is fired not before connect.
I suppose you are still running Asterisk 1.0.x.
For Asterisk 1.0.x i know of no clean solution for that problem.
Asterisk 1.2 introduced the Dial event that is triggered before
We have been using Teliax (www.teliax.com) for a while now and have 3
accounts with them (one for each of our asterisk servers). They've had their
ups and downs but have been working to improve their support and now we are
now able to speak with someone during their business hours (8-5PM MST).
Th
Stefan Reuter schrieb:
To propose the best solution we must know more about your actual use
case.
Thanks for your answer!
I want to track the ringing event of the outgoing channel.
Unfortunatelly the link event is fired not before connect.
Thus, I see first the incoming channel (SIP or IAX)
AFAIK, most of VOIP providers ignore callerid from ATA and substitute it with a caller id on their records.
On Fri, 2005-11-04 at 01:11 -0600, Jason Brashear wrote:
OK I am exhausted.
I can't seem to figure out how to send a caller ID along with a
Outbound call.
Can you believe that I go
I'm not sure if a failed qualify will affect your active call but you
might want to try to use the qualifysmoothing variable in iax.conf.
This won't "disqualify" a peer for a single bad sample.
;qualify=yes; Make sure this peer is alive
;qualifysmoothing = yes
Hi guys,
in my 3rd asterisk installation, I have a problem with zaptel modules.
I use the CVS.
Instead of obtaining, for example, zaptel.o I got zaptel.ko.
What is the reason?
Like that, also all the other zaptel kernel modules got the same extension.
Also, zaptel do not create /proc/zaptel/1 and r
On Sat, 2005-11-05 at 13:42 +0100, Roger Schreiter wrote:
> Now I wonder, whether I can rely on that scheme.
> I assume, the timestamp part can be different, e.g.
> if between the creation of the incoming channel and
> the creation of the outgoing channel the system clock
> switches to the next sec
You have to add an extension for "asterisk" because it gets the
messege from [EMAIL PROTECTED] and then chops off the domain.
eg...
exten => asterisk,1,Voicemailmain(${EXTEN})
On 11/4/05, Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
> I got Snom to work to flash the MWI and the stutter tone.
>
>
Hi,
the uniqueid obviously consists of a timestamp part
and an continously incremented part, separated by a dot.
The two channels of a call in most cases have the
same number before the dot (timestamp) and consecutive
numbers after the dot.
Now I wonder, whether I can rely on that scheme.
I ass
I have several clients that need DIDs in Dubai. I need to be able to set
these up urgently. If anyone on this list can help, please advise.
Thanks in advance,
Neil
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lists
Pleasants
Sent: 05 November 2005 01:59
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP extension calls itself intermittently
Intermittently Ill get calls from my only SIP extension
hey 1.2 b2 hs bugs.
On 10/18/05, Adam Moffett <[EMAIL PROTECTED]> wrote:
- Original Message -> *From:* Rob Fugina [EMAIL PROTECTED]>> *To:* asterisk-users@lists.digium.com
> asterisk-users@lists.digium.com>> *Sent:* Monday, October 17, 2005 5:14 PM> *Subject:* [Aster
yes i got my mainstream * with teliax
no problem..
keep it ,ulaw,alaw, no jitter and ,open your ports, all good for 9 months !
On 10/19/05, Rich Adamson <[EMAIL PROTECTED]> wrote:
For those on the list using iax with teliax.com, the intermitant one-wayaudio problem that I reported to them re
i use teliax for primary and bell for failover.
no neeed yet for failover ;)
they got theyre shit together...On 11/3/05, Jason Brashear <[EMAIL PROTECTED]> wrote:
Thank you Gleim I will look into that.
-Jason
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf
Hi,
thanks for the reply jim.
could you elaborate a little please.
thank you
John
On Sat, 5 Nov 2005 05:42:45 -0500, Jimmy Smith wrote
> bet its a difference in register timer..
>
> On 11/5/05, John Fraser <[EMAIL PROTECTED]> wrote:
Hi all,
>
> I am using a TE405P with an E1 from the t
oh and one tech from them support said they will be handling all cname soon .. like ETA 1 week
can't wait to get my FBI name out...
On 11/5/05, Rich Adamson <[EMAIL PROTECTED]> wrote:
> OK I am exhausted.> I can't seem to figure out how to send a caller ID along with a> Outbound call. < snip >>>
bet its a difference in register timer..
On 11/5/05, John Fraser <[EMAIL PROTECTED]> wrote:
Hi all, I am using a TE405P with an E1 from the telco. I am getting the allcircuits busy now message about 5% of the time on outbound calls from sipsoft phones. Has there ever been a resolution to this pr
Hi,
how can I know in the dialplan, whether and when I received
the ringing event?
Imho, the only way is to parse all events using the manager
and to forward this information by an application to the
dialplan. The application would have to be called on
connect or on hangup.
Hints for a more sim
Hi all,
I am using a TE405P with an E1 from the telco. I am getting the all
circuits busy now message about 5% of the time on outbound calls from sip
soft phones. Has there ever been a resolution to this problem?
suggestions please
thanks
John Fraser
_
You need to download and install oh323. I don't think it comes
pre-installed.
I am using [EMAIL PROTECTED] and had to install specifically for it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angus Comber
Sent: Friday, November 04, 2005 22:58
To: Aster
On Fri, Nov 04, 2005 at 05:53:37PM +0100, Anders Svensson wrote:
> Hi!
>
> How do I uninstall AMP and FOP from my Asterisk?
>
apt-get/aptitude could do the trick had you used debs
(rapid.dotsrc.org/amportal )
Hopefully one day inside Debian as well, though the developers of AMP
don't consider t
> OK I am exhausted.
> I can't seem to figure out how to send a caller ID along with a
> Outbound call.
< snip >
>
> Anyway, Outbound Caller ID Hos is this done?
> I now use VoicePulse as my provider.
I'm not a VoicePulse user, so not sure if they even support it.
Here's how I do it with teli
On Sun, Oct 30, 2005 at 10:02:24AM -0500, David Bandel wrote:
> Perhaps a good enhancement would be a syntax checker for the various
> .conf files.
There is a vim syntax file floating around. Also an emacs mode.
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |
On Thu, Nov 03, 2005 at 09:08:07PM -0500, [EMAIL PROTECTED] wrote:
> [EMAIL PROTECTED] wrote on 11/03/2005 03:33:06 PM:
>
> > Might be worth it to read the stuff in /usr/src/asterisk/doc and in
> > particular the README.asterisk.conf file. Lots of other good stuff
> > in that directory as well. (N
81 matches
Mail list logo
