The basic incoming and out going works fine.
Trying to create a basic dial plan and asterisk hangs up on me.
First issue:
I am only hearing part of the recording when I call in
Second issue;
Before I am even able to choose an option of hear the rest of the recording,
asterisk
PS,
I have zero FX gear. I am 100% SIP
Brad
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Welcome to the Asterisk Users mailing list,
On Fri, Jul 20, 2007 at 02:11:08AM -0400, BSumrall wrote:
PS,
I have zero FX gear. I am 100% SIP
Huh?
This nmust be related somehow to the PRI Busy problem thread you've
answered to, otherwise you wouldn't have replied to it, right?
If you want
I have SNOM phone and in my phone there is a transfer button but whn i use
transfer key and enter another party number i got hangup so is there any
configuration for Dial() t, T option is there any need to specifiy t or T
option in dial plan
Andrew Joakimsen [EMAIL PROTECTED] wrote: On
You should be running the latest Zaptel LibPRI both of which
recently have been updated. We run a similar configuration and have
not seen this problem with the upgrade.
Even after upgrading Zaptel to the latest (1.2.19) from 1.2.17.1 there is the
same problem. Libpri-1.2.5 was already at
Lenz (?), you are not the only one! It took about five days for Anthony's
reply to reach me.
--Don
Don Kelly
PCF Corp
Real Support for your Virtual Office
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Dear all
I have 2 E1 card on my asterisk and i want to route call
with fix pattern like if anyone dial mobile number like 9818875535 so it will
use PRI 1 and rest of the world goes through PRI 2 means whn number prefix
98XX then call goes through specified E1 is it
On Thu, Jul 19, 2007 at 08:32:35PM -0700, Jay Wilton wrote:
Hello,
I am trying to debug a machine that segfaults. A core file
is produced like /tmp/core.4545 . The command and error:
gdb /usr/sbin/asterisk -c /tmp/core.4545
GNU gdb 6.3-debian
...CUT
This GDB was configured as
On Fri, Jul 20, 2007 at 01:41:42AM -0500, Don Pobanz wrote:
You should be running the latest Zaptel LibPRI both of which
recently have been updated. We run a similar configuration and have
not seen this problem with the upgrade.
Even after upgrading Zaptel to the latest (1.2.19) from
Jason Parker wrote:
I'd wager that you're using the wrong path for the licenses.
I believe the correct path is something like /var/opt/asterisk/licenses/ -
it's whatever Asterisk has ast_config_AST_VAR_DIR set to, with /licenses/ at
the end.
The easiest way to tell, is to find the
Mojo with Horan Company, LLC wrote:
Sorry that this is unrelated but, Bruce, do you double-click to send
your messages? Just curious.
Sorry that this is unrelated but, Mojo with Horan, do you wake up each
morning and think of a meaningful question to ask someone, such as the
above, every
You have to first uninstall your Asterisk1.2 like this--
First you have to stop your asterisk...using--
1. killall -9 asterisk or killall -9 safe_asterisk, whichever you are using.
2. After that you have to remove the zaptel driver.
For that just run this command and see which are running--
#
There is one thing,
just forget that your phone is a snom phone or whatever...
simply to make a blind call transfer press #8, according to the my
feature.conf, default it is #, or you can assign it any, then after pressing
that you will listen a IVR transfer and dial the desired number followed
Hello,
I'm partner manager at snom. The SI-90 and SI-120 ARE NOT snom phones.
Our Indian joint venture is selling these!
These phones may not be called snom, that's why their name is SI-90 and
SI-120.
The phones are not enginieered and developed in Germany as the normal
snom300 series.
HI
Here is my info:
Asterisk - 1.2.10 with zaptel 1.2.7, 10 queues with 7 sip agents
this asterisk box is connected to another asterisk box using 5 IAX trunk to
load balance no of calls on each IAX trunk (g729 over trunk). Suddenly my
cli start flooding with message: Maximum trunk data space
Martin Smith wrote:
Hey folks,
So I'm trying to get Festival() working on 1.2.17. I'm trying to use
app_festival:
Here's the show dialplan output from that extension:
'3378' = 1. Answer()
[pbx_config]
2. Festival(Hello Asterisk caller. How is your day?)
On 7/20/07, Andrew Joakimsen [EMAIL PROTECTED] wrote:
Do you have any patches against your Asterisk, Zaptel or Kernel?
Actually are you using anything but the factory Kernel?
I'm using an older Slackware. The problem came in March or so with
1.2.14 I think.
Besides that I just wouldn't advise
Priorityjumping is totally ignored by my asterisk (tested 1.4.4 and
1.4.7.1 on FreeBSD 6.2)
[general]
priorityjumping=yes
With n+101:
exten = 1337,1,Dial(SIP/zytek,5,Ttj)
exten = 1337,102,Dial(SIP/zytek,${RINGTIME},${OPTIONS})
exten = 1337,n,Hangup
-- Executing [EMAIL PROTECTED]:1]
Did you read UPGRADE.txt? Priority jumping was deprecated in 1.2. I
assume it was removed from 1.4.
Jakub Głazik wrote
Priorityjumping is totally ignored by my asterisk (tested 1.4.4 and
1.4.7.1 on FreeBSD 6.2)
[general]
priorityjumping=yes
With n+101:
exten =
Dear all,
I have an IVR set up with the dialplan below.
After recording the first voicemail the remaining part of the context is
not executed the call was terminated by asterisk...
WHAT CAN I DO TO GET THE REMAINING PART OF MY DIALPLAN EXECUTED AFTER
THE VOICEMAIL IS RECORDED.
[netchange]
On Fri, 2007-07-20 at 08:46 +0100, Bruce McAlister wrote:
Mojo with Horan Company, LLC wrote:
Sorry that this is unrelated but, Bruce, do you double-click to send
your messages? Just curious.
Sorry that this is unrelated but, Mojo with Horan, do you wake up each
morning and think
Is it possible to bridge a media stream, lets say created by VLC on to an
Asterisk channel?
What I would ideally like to do is - When mobile dial into my Asterisk
server, follow through some security/prompts then, through the dialplan
launch VLC as an external application. VLC would connect to an
are there any good softphone on PDA window mobile 2003 / 5.0 ?
tried sjphone, sound quality is unacceptable.
Mario
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You can also give the our T.38 stack a try.
http://www.attractel.com/t38.html
Chris Childress
AsteriskGuru.com
Andrew Joakimsen wrote:
I have already tried to contact to persons from Digium and I did not
receive a response.
I was wondering if there is any plan to support fully faxing in
Dear all,
I have an IVR set up with the dialplan below.
After recording the first voicemail the remaining part of the context is
not executed the call was terminated by asterisk...
WHAT CAN I DO TO GET THE REMAINING PART OF MY DIALPLAN EXECUTED AFTER
THE VOICEMAIL IS RECORDED.
[netchange]
Hi everybody,
One of my customers wants to buy IP Phones and Asterisk solution, but his
requirement is if he'll not be happy with Asterisk, his phones should be
able to work with other IP PBX systems as well, so that he doesn't have to
buy new phones again. After all IP Phones is the main
Tell your users to exit voicemail by pressing # instead of hanging up.
Goke Aruna wrote:
Dear all,
I have an IVR set up with the dialplan below.
After recording the first voicemail the remaining part of the context is
not executed the call was terminated by asterisk...
WHAT CAN I DO TO
I'd bet the emails are addressed to the list and the original sender,
both, so for the original person they appear twice, but everyone on the
list gets them a single time. I haven't seen any duplicates.
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
David Boyd wrote:
On Fri, 2007-07-20 at 08:46 +0100, Bruce McAlister wrote:
Mojo with Horan Company, LLC wrote:
Sorry that this is unrelated but, Bruce, do you double-click to send
your messages? Just curious.
Sorry that this is unrelated but, Mojo with Horan, do you wake up each
I am sending an email to the mailer list.
Not following any thread?
Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Friday, July 20, 2007 2:38 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Novice needs help
Thank you,
Not that I want the user to hangup...I want the user to continue and
that is why i have the priority 4 on context pdata.
Thanks
Goksie
Eric ManxPower Wieling wrote:
Tell your users to exit voicemail by pressing # instead of hanging up.
Goke Aruna wrote:
Dear all,
I have an
On Fri, 2007-07-20 at 08:55 -0400, Martin Smith wrote:
I'd bet the emails are addressed to the list and the original sender,
both, so for the original person they appear twice, but everyone on the
list gets them a single time. I haven't seen any duplicates.
Martin Smith, Systems Developer
On Thu, Jul 19, 2007 at 02:40:35PM -0800, Mojo with Horan Company, LLC wrote:
Sorry that this is unrelated but, Bruce, do you double-click to send
your messages? Just curious.
Both copies have the same ID. And both were sent through the gmane
newsgroups gatewas. For some reason the order of
On Fri, Jul 20, 2007 at 09:01:27AM -0400, BSumrall wrote:
I am sending an email to the mailer list.
Not following any thread?
Brad
This email was a reply to my message, and hence appeared properly
threaded to it.
Look for BSumrall in
Bruce sorry for the top post, but your last two messages have not come
in twice Go figure...
db
On Fri, 2007-07-20 at 13:37 +0100, Bruce McAlister wrote:
David Boyd wrote:
On Fri, 2007-07-20 at 08:46 +0100, Bruce McAlister wrote:
Mojo with Horan Company, LLC wrote:
Sorry that
Hi,
how compatible a phone is to a specific phone system also depends from the
required feature set. Basic call should work between most phones and
systems.
If you're interested on snom interoperability I would like to point your
attention to the following page:
Only to continue on this thread (becouse this is start in other meail).
The 1.4.X. unicall patch is working well, only with one problem: There
is a problem hen reciving calls with no Caller ID.
Thanks.
On 6/9/07, Moises Silva [EMAIL PROTECTED] wrote:
Alvaro...
Hum..., I never have
Search at mfcr2.c this:
case MFCR2_PROT_MEXICO:
And add the next line after that line:
mfcr2-group_i_end_of_ANI_restricted = R2_SIGI_12;
This will help you on calls that have the restricted flag on the ANI only.
(Nextel). But not on no caller id calls.
I don't know if steve can help
On Fri, 20 Jul 2007, Martin Smith wrote:
I'd bet the emails are addressed to the list and the original sender,
both, so for the original person they appear twice, but everyone on the
list gets them a single time. I haven't seen any duplicates.
I've seen list duplicated and sometimes
On Fri, 2007-07-20 at 02:08 -0400, BSumrall wrote:
My dial plan of issues…..
exten = s,1,Answer(60)
exten = s,2,Background(otherwise-press)
exten = s,1,Playback(digits/1)
exten = s,2,Goto(default,s,1)
exten = s,1,Playback(digits/2)
exten = s,2,Goto(default,s,1)
I'm not sure why you have
Hi Philipp -
Since the list was switched over to API-Digital almost
every message I get is older than a week. Coincidence?
Is anyone else having trouble?
Well, this is now the third active thread on this subject, but I guess
you won't see this message for a while. Has anyone dissected the
I have written a script that is executed using ExternalIVR(). I am running
in to performance issues when I have four or more simultaneous calls running
this script. I'm running on a P4 2.8 with 512M, all calls are GSM coming in
over IAX from an asterisk box that acts as a switch and handles all
At 07:22 7/20/2007, Chris Childress wrote:
You can also give the our T.38 stack a try.
http://www.attractel.com/t38.html
Software? Hardware? Integration? Prices?
Can't make a decision without enough info.
Chris Childress
AsteriskGuru.com
Andrew Joakimsen wrote:
I have already
Not until the topic looks me in the face. Sorry I wasn't up earlier in
the morning to explain my question, but the rest of the list did for me.
I was seeing two messages _every_ time you posted one. Like I said, I
was just curious if that could have been the reason; it would have been
an
Hi Anthony,
I used their services for 3 months (signed up on the pay-as-you-go plan -
Unlimited channels) and all my minutes were rounded up to the next cent in my
CDR ...
Regards,
Marcelo
-- Original message --
From: Al Bochter [EMAIL PROTECTED]
Anthony,
So you
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
/tmp/core.4545 is not a core dump: File format not
recognized
So what is that file?
file core.4545
core.4545: ELF 32-bit LSB core file Intel 80386, version 1
(SYSV), SVR4-style, SVR4-style, from 'asterisk'
The box was rebooted before I had a
On 7/20/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Did you read UPGRADE.txt? Priority jumping was deprecated in 1.2. I
assume it was removed from 1.4.
According to UPGRADE.txt, the default in the absence of priorityjumping=
changed from yes in 1.2 to no in 1.4:
* In previous
I am able to use sox to convert audio files from ulaw to
wav (MS ADPCM), is there a way, using sox or another
command line tool, to convert them to g726 ?
( g726-32 since it is supported by * )
tia,
-baji.
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Yes it is a local call. 778 was added to the lower mainland to ease the
pressure on 604 numbers due to the explosion in mobile/fax lines being added.
If you need a 604DID (for vanity purposes or whatever) you could try the
provider I use. I'm not sure if I can post their information on the
i am have x100P clone, and install asterisk 1.4 and out call normaly and
hangup in xlite to zap but call to asterisk for zap channel nop pass to
xlite and the caller hangup the asterisk not detect.
what is the problem ???
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On Fri, 2007-07-20 at 15:33 -0500, Walter Willis wrote:
i am have x100P clone, and install asterisk 1.4 and out call normaly
and hangup in xlite to zap but call to asterisk for zap channel nop
pass to xlite and the caller hangup the asterisk not detect.
The X100P (and it's numerous clones)
Is this replacing Astricon this year?
If so it looks like a pretty poor showing in comparison to Astricon
Dallas last year.
Cheers,
Dean
From: Carl Ford [mailto:[EMAIL PROTECTED]
Sent: Wednesday, 18 July 2007 9:09 AM
To: Dean Collins
Jared Smith wrote:
On Fri, 2007-07-20 at 02:08 -0400, BSumrall wrote:
My dial plan of issues…..
exten = s,1,Answer(60)
exten = s,2,Background(otherwise-press)
exten = s,1,Playback(digits/1)
exten = s,2,Goto(default,s,1)
exten = s,1,Playback(digits/2)
exten = s,2,Goto(default,s,1)
Dean Collins wrote:
Is this replacing Astricon this year?
If so it looks like a pretty poor showing in comparison to Astricon
Dallas last year.
No it is not, as is clearly evidenced by the fact that the Astricon
website is still up, with the same content it had before this
announcement, and
I have a customer that has recently upgraded their network and now
their Aastra 9133i phones are loosing their connection to the Asterisk
server. They were using an external Asterisk server and now we have
installed a new internal server with Asterisk 1.4.8 on a SIP/IAX
implementation
On Fri, 2007-07-20 at 17:01 -0400, Dean Collins wrote:
Is this replacing Astricon this year?
Nope... this isn't meant to replace AstriCon in the slightest. As I
understand it, the two conferences have different focuses...
Obviously AstriCon is the Asterisk users' conference and is meant to
On Fri, Jul 20, 2007 at 10:09:33AM -0700, Jay Wilton wrote:
--- Tzafrir Cohen [EMAIL PROTECTED] wrote:
/tmp/core.4545 is not a core dump: File format not
recognized
So what is that file?
file core.4545
core.4545: ELF 32-bit LSB core file Intel 80386, version 1
(SYSV),
On Fri, Jul 20, 2007 at 04:56:45PM -0400, Jared Smith wrote:
On Fri, 2007-07-20 at 15:33 -0500, Walter Willis wrote:
i am have x100P clone, and install asterisk 1.4 and out call normaly
and hangup in xlite to zap but call to asterisk for zap channel nop
pass to xlite and the caller hangup
I'm looking for 24 or 48 port IEEE802.3af POE injector.
Any recommendation?
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On 7/20/07, Thomas Kenyon wrote:
convert file.g729 file.g726-32 in the asterisk CLI works here.
as does file.g726-16 (but not 24 or 40).
The weird thing is, it doesn't seem to transcode from ulaw/alaw but
works fine from g729/gsm.
thank you ! now I have another command to experiment
Hi,
I have to install an Asterisk PBX for a customer and he wants something like
logic supply's fanless computers. Can anybody advise about how good will
they work, are they compatible with the Asterisk system? I'll also be
installing a sangoma 4 port FXO card in it.
--
Zeeshan A Zakaria
look my zapata.conf
[channels]
context=default
switchtype=national
signalling=fxs_ks
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
Hi,
I have a Dell Power Edge server planning yo buy Sangoma A101D card.
To configure with my Asterisk 1.2.18 zaptel-1.2.17.1 Free-PBX setup.
So I wanted to know the steps any issue which I may come accross if any.
I have googled have some docs handy wrt Trixbox-2.2. Just wanted
On 7/20/07, Carlos Chavez [EMAIL PROTECTED] wrote:
I have a customer that has recently upgraded their network and now
their Aastra 9133i phones are loosing their connection to the Asterisk
server. They were using an external Asterisk server and now we have
installed a new internal
exten = _98XX,1,Dial(ZAP/(your preferred E1)
exten = _,1,Dial(ZAP/(second E1)
On 7/20/07, satish patel [EMAIL PROTECTED] wrote:
Dear all
I have 2 E1 card on my asterisk and i want to route
call with fix pattern like if anyone dial mobile number like 9818875535
Please, unsuscriber, this group.
regars
Nestor Castillo
- Mensaje original
De: [EMAIL PROTECTED] [EMAIL PROTECTED]
Para: asterisk-users@lists.digium.com
Enviado: viernes, 20 de julio, 2007 11:00:04
Asunto: asterisk-users Digest, Vol 36, Issue 61
Send asterisk-users mailing list
Hi Zeeshan -
I have to install an Asterisk PBX for a customer and he wants something like
logic supply's fanless computers. Can anybody advise about how good will
they work, are they compatible with the Asterisk system? I'll also be
installing a sangoma 4 port FXO card in it.
Have you
On 7/20/07, I wrote:
On 7/20/07, Thomas Kenyon wrote:
convert file.g729 file.g726-32 in the asterisk CLI works here.
as does file.g726-16 (but not 24 or 40).
The weird thing is, it doesn't seem to transcode from ulaw/alaw but
works fine from g729/gsm.
thank you ! now I have
Hi Norman -
To add to what Edgar said, yes, use linux-ha. It works nicely in
combination with DRBD. DRBD uses a dedicated network interface on
each box with a crossover cable between the two. It does a block
level copy of the entire filesystem, so you have two machines that are
identical.
Hi Arun -
Asterisk - 1.2.10 with zaptel 1.2.7, 10 queues with 7 sip agents
this asterisk box is connected to another asterisk box using 5 IAX trunk to
load balance no of calls on each IAX trunk (g729 over trunk). Suddenly my
cli start flooding with message: Maximum trunk data space exceeded
If you're using the Snom transfer button, you don't need to do
anything in features.conf. In extensions.conf, just make sure that
the dial() command used to call the snom phone uses the 't' flag.
THIS IS INCORRECT!
The options t and T are for DTMF based transfers. You do not need any
I'm looking for 24 or 48 port IEEE802.3af POE injector.
Any recommendation?
Yes. For the price of one of those multi-port injectors, you can come
close to the price of a new Netgear or 3Com PoE switch. The injectors
typically add power to the unused pairs (mode B PoE). This means you
can't
You have to first uninstall your Asterisk1.2 like this--
First you have to stop your asterisk...using--
1. killall -9 asterisk or killall -9 safe_asterisk, whichever you are using.
In my experience, you don't need to do this step. In fact, you can
keep the old asterisk running, compile and
IEEE802.3af uses same 4 wire as data.
thats what Polycom uses.
the way i'm seeing it we are better off with poe switch(looking at the
price).
On 7/20/07, Noah Miller [EMAIL PROTECTED] wrote:
I'm looking for 24 or 48 port IEEE802.3af POE injector.
Any recommendation?
Yes. For the price of
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