Hi Mike,
since you are using snom phones I have 2 question for you maybe you can
help me (no one else could till now):
1 - sometimes the message "bad gateway" appears on my phone display.
2 - after a call the phone led do not turn off so anyone else does not
know if I'm busy or not.
Have you
Hi, I'm trying to use ChanIsAvail to build a resilient 'dialout' macro.
The logic is simple; try Zap/g1 (a group of two E1s), and if that
fails, try locating a channel via DUNDi. Here's a massively cut down version
to illustrate the problem I'm having.
macro dialout ( dest ) {
ChanIsAvail(
Aren't Aastra due to release new phones and some form of
operator/reception addon? The Aastra user/admin guides are a lot more
up2date that they used to be. I'd like Aastra to add a GSM codec to
their phone and have a more regular firmware release schedule. I agree
with the list below though tha
Have you tried the #freepbx IRC channel or the freepbx mailing list?
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Arnilo S. Baluyos (Mailing Lists)
Sent: 23 January 2007 01:57
To: Asterisk Users Mailing List - Non-Commercial Discussi
Hi,
I have a working asterisk 1.4.0 with Mysql Realtime configuration, and today
I encountered this error.
Now, I have no acces to any information in mysql realtime, so nothing work
now !
[Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime:
Everything is fine.
[Jan
Hi
anyone have a sample of shorewall configuration for add a TC/QoS
on IAX2 traffic ?
Thanks for your help
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Hi List,
Does anyone know where I can get support for the digium forums ? my user ID and
pass just stoped working as of yesterday. The forums say to go to asterisk.org
for any password issues. I am able to log in there with out any issues. For
some reason when I try to log in to the forums it wo
Ciao,
I have an Asterisk 1.2.9.1 box with a beronet dualBRI
(install-misdn-queue) on a Debian distro. I'm experiencing problem with
some Telecom Italia lines: some people cannot choose menu selections or
extensions after hearing intro message. Is there anybody who knows if
there is a particula
I'm running Asterisk SVN-trunk-r51353... for some reason even if i set nat=yes
in the sip.conf for a device when i do a show sip peers it shows N for nat. Is
this a bug or am i doing somthing wrong here. I'm basically having a problem
right now where i can call in/out of asterisk and talk fine
Thanks AT.
On 1/24/07, Axel Thimm <[EMAIL PROTECTED]> wrote:
On Tue, Jan 23, 2007 at 06:20:08PM +0100, Axel Thimm wrote:
> On Tue, Jan 23, 2007 at 03:54:49PM +0200, Tzafrir Cohen wrote:
> > On Tue, Jan 23, 2007 at 02:48:07PM +0100, Axel Thimm wrote:
> >
> > >
> > > If I call asterisk -r as root
Hi,
i can see AOC messages on the asterisk console. Can i sendtext() them to the
caller or put them in cdr?
Regards, Andreas.
_
Need a new job? Check out XtraMSN Careers http://xtramsn.co.nz/careers
I ran into this problem with an early batch of IP650s. Polycom's
firmware version 2.0.3b made this issue go away.
Thanks,
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.shelto
Can Asterisk support vxml?
Can i work with Asterisk and vxml?
Is there any AGI framework that can use vxml?
Thanks
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On 1/22/07, nik600 <[EMAIL PROTECTED]> wrote:
Hi
i want to put an AGI script in a queue, to serve once at time the callers.
Example:
Queue (8 callers waiting)
Agi script / IVR (serving the caller)
can i do that?
Thanks
any idea?
___
--Bandwidth a
Hi
i use a lot of Grandstream GXP2000 with BLF
How to set up on the same key BLF blinking call interception?
So that someone is able to take a call that is destinated to another user
phone
Thanks bye
___
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Dovid B wrote:
> Hi List,
> Does anyone know where I can get support for the digium forums ? my
> user ID and pass just stoped working as of yesterday. The forums say
> to go to asterisk.org for any password issues. I am able to log in
> there with out any issues. For some reason when I try to log
I have a client who has a Panasonic Hybrid system. They are taking in
another company as a building tenant and the tenant will be on a new 12
station Asterisk system. This new asterisk system will have 4 FXO ports
plus ITSP. The two systems will be separate except that they should tie
together f
Its a problem in your database. something might have corrupted...be
prepared to load a backup
Gregory Duchatelet wrote:
Hi,
I have a working asterisk 1.4.0 with Mysql Realtime configuration, and
today I encountered this error.
Now, I have no acces to any information in mysql realtime
Where do I find more out in regards to the echo-cancelling component you
mentioned?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: <[EMAIL PROTECTED]>
To:
Sent: Tuesday, January 23, 2007 8:08 PM
Subject: asterisk-users Digest, Vo
Where do I find more out in regards to the echo-cancelling component you
mentioned?
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: <[EMAIL PROTECTED]>
To:
Sent: Tuesday, January 23, 2007 8:08 PM
Subject: asterisk-users Digest, Vo
Sorry my mistake it is in Snom 360 firmware 3.60b and higher, not 320.
-Original Message-
From: Mike Hammett [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 24, 2007 7:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] RE: Snom 320 echo
Where do I find more out in re
Yes, IAX <--> IAX works fine
it's when it's Zap(in) ->Asterisk -> Zap(out)
Every call that I pass through to that outgoing Zap channel (a dial out to a
mobile phone) fails to hang up.
CallerID is working -- with this sangoma card, which seems to need...
cidsignalling=v23
cidstart=polarity
...to
Which panasonic system?
I'm assuming you are talking about the TDA line. If so get a IP
Gateway card on the TDA system, that card uses h323, then configure it
with asterisk as h323, or my favorite, get a PRI card on the TDA sysem
(unless it's a TDA50 then the option is not available), and a T1 car
From what I heard:
Aastra is coming out with a few new phones 53i, 55i, and 57i and sidecar
On 1/24/07, Lee Archer <[EMAIL PROTECTED]> wrote:
Aren't Aastra due to release new phones and some form of
operator/reception addon? The Aastra user/admin guides are a lot more
up2date that they used to
We also use Polycom IP650 phones. They are assigned to our customer
service department. Each SIP interface is a member of our customer
service Queue in Asterisk.
The behavior we see is that the SIP interface in the queue will
sometimes not release from the in-use state. Connecting to the
i
Our 650s are running 2.0.3b. The problem still exists for us. We see
the devices as members of our customer service queue stick on 'in-use'
in the Queue application while the device has no active SIP channel and
will accept calls. Removing 'call-limit' from the sip.conf in Asterisk
for the d
Hi All,
I'm running 1.2.9.1. I can prune sip realtime peers and users and iax
realtime peers but no command to prune iax realtime users. Was this
implemented in a later version?
Thanks.
JR
--
JR Richardson
Engineering for the Masses
___
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Hi All,
I'm running 1.2.9.1, is setvar= implemented in iax.conf in a later
version of asterisk? If so, which one?
Thanks.
JR
--
JR Richardson
Engineering for the Masses
___
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I'm trying to wean my self using the Agent channel in Asterisk. The main
reason I use it is for the callback acknowledgement , where the user
presses # to finally acknowledge the call. I have implemented this in the
Dialplan using the Macro in the docs. This works well as long as the user
enter
On Wed, 24 Jan 2007 09:11:20 +
Gavin Hamill <[EMAIL PROTECTED]> wrote:
> Processing does not continue to the NoOp or Dial - what am I doing
> wrong? I've also tried with the 'j' option to 'jump to priority n+101'
> even though I'm using AEL, but it's made no difference.
For the benefit of th
Hello,
I'm trying to make my asterisk box to act as a telco, in order to reproduce a
US environment in europe.
Our telco provider is giving us those settings:
ESF
B8ZF
Inbound = E&M Immediate
Outbound sig =Wink Start
Yield to Glare = Yes
Those trunks are using CAS for signaling.
I have tried m
Lee Jenkins wrote:
Hi all,
I've just setup a sip line with Telasip and when they route the calls to
my asterisk box, they include an extension along with the context that
is defined in sip.conf for that DID.
At first, I couldn't figure why they were getting 404 error from my
asterisk box,
Sure -- this is for my Sangoma a200
-=-=-=-
Zapata.conf
(currently supporting CallerID)
[channels]
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
rxgain=1.0
txgain=1.0
musiconhold=default
callwaiting=no
transfer=yes
usecallingpres=yes
cadence=200,200,200,4000
usecallerid=yes
calleri
On Jan 24, 2007, at 10:20 AM, Thinselin, Vincent wrote:
Hello,
I'm trying to make my asterisk box to act as a telco, in order to
reproduce a US environment in europe.
Our telco provider is giving us those settings:
ESF
B8ZF
Inbound = E&M Immediate
Outbound sig =Wink Start
Yield to Glare =
James Fromm wrote:
The behavior we see is that the SIP interface in the queue will
sometimes not release from the in-use state. Connecting to the
interface from another SIP device and immediately hanging up will clear
the state.
The phones in question are configured with one line that will ex
If you use a Vegastream gateway on the actual incoming ISDN circuits then
you won't even need to touch the Panasonic to integrate both systems.
Regards
Scott Pinhorne
VoxIT Limited
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: 24 January 20
Michiel van Baak wrote:
On 17:36, Thu 18 Jan 07, Lee Jenkins wrote:
Michiel van Baak wrote:
exten=>999,1,Queue(support,tr|||60)
and never put it back. From there it was a downward spiral ;)
If you remove the r, does that fix the issue ?
Yes. It did. Still not sure why it didn't work or
On 1/24/07, JR Richardson <[EMAIL PROTECTED]> wrote:
Hi All,
I'm running 1.2.9.1. I can prune sip realtime peers and users and iax
realtime peers but no command to prune iax realtime users. Was this
implemented in a later version?
Thanks.
JR
From what I could dig up, it looks like you can
Can anyone point me to info on how to change the functionality of the
SIP (7.4) 7960's. We previously had an SCCP firmware on the phone and
the users want the phone to work like it used too. Here are some examples:
The users do not want to push the new call softkey or the speaker button
in order
Yeah, we don't use Buddy Watch. We don't use call-limit because we want
call-limit. We use it because it's the only way, that I'm aware of, to
get the SIP channel driver to monitor the state of the member SIP
interface. We use autopause=yes and ringinuse=no in our customer
service queue conf
any idea, how to do something like this, but in correct/functional form?
;-)
Set(foo=System(echo "${EXTEN}" | tr "[:upper:]" "[:lower:]"))
${EXTEN} is "SomeStrinG"
${foo} output should bee "somestring"
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Has anyone had any experience using FXO and FXS gateways to extend
legacy PBX extensions to remote users? I have a customer who needs to
do this, but wants seamless, two way communication, with a SIP server
and without the need for 2-stage dialing. If anyone has any experience
with a solution ple
> > ESF
> > B8ZF
> > Inbound = E&M Immediate
> > Outbound sig =Wink Start
> > Yield to Glare = Yes
> >
> >
> > In zaptel.conf, when having something like
> > span=5,0,0,cas,b8zs
> > and in zapata-channels something like
> > signalling=featb
> try
> em_w: E & M Wink Start
>
Jerry is right - you ne
> The users do not want to push the new call softkey or the speaker button
> in order to dial a call. They want to be able to just begin dialing the
> number.
>
> The users do not want to push the answer softkey after they pickup the
> handset in order to answer a call.
Doesn;t it answers when yo
> Can Asterisk support vxml?
> Can i work with Asterisk and vxml?
>
> Is there any AGI framework that can use vxml?
>
It seems like support is still a bit limited, but evidently it is
available:
http://www.voip-info.org/wiki/view/VoiceXML
HtH,
MC
___
Hello everybody,
I was wondering if anybody knows how to make channel IDs different if all
call are coming from the same host:
core show channels
Channel Location State Application(Data)
SIP/sip-ny1.stanapho [EMAIL PROTECTED]:4 Up Playback()
SIP/sip-ny1.stanapho [
another reason, why better is to completely avoid ci$co phones when used
with anything other than callmanager ;-)
Yehavi Bourvine +972-8-9489444 wrote:
The users want the transfer softkey on the screen while on a call.
Currently it is acessable via the more softkey.
I've asked Cisco whe
I'm trying get this working. I've looked through the list, and can't see
how to get AMD to print out more. I have it call and say Hello like I
normally would. I've tried to say more and less doesn't seem to matter.
After I hangup it does recognize hangup. Here's logging during an attempt
where
Hi all,
We're planning an Asterisk implementation consisting of two SC1425 Dell
Servers using Digium T1 cards and SATA drives. The problem I'm having
is the only PCI slot shares an IRQ with the SATA controller. Any
altering of one device's IRQ takes the other device's IRQ with it in
lockstep.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Halliday
Sent: Wednesday, January 24, 2007 11:56 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] setting up AMD
I'm trying get this working. I've looked through the list, and can't
see how to
I have one system that is crashing everytime a call is parked and I have
tried recompiling the asterisk, checking out the latest SVN of 1.2 and
modifying the configuration. I have identified what I think is the error and
have back traces but since this is occurring on only one system I want to
kno
Hello.
I am running asterisk 1.2.14 on a Dell poweredge with a Digium FXO/FXS card
connected to 6 analog lines and using Linksys spa942 phones.
My users are complaining of randomly disconnected calls, and when I watch
the log (debug warning,notice,error), I don't see any cause. It looks like
ast
now I have amd.conf set to this:
initial_silence = 3700
greeting = 2500
after_greeting_silence = 1200
total_analysis_time = 6000
min_word_length = 100
between_words_silence = 50
maximum_number_of_words = 4
silence_threshold = 860
The resulting log is this:
Jan 24 18:53:04 DEBUG[31555] chan_sip.c
Cory, it's called dialplan magic it realy depends what PBX it is, not
all of them allow dial plan magic. But it is possible on most pbxes.
On 1/24/07, Cory Andrews <[EMAIL PROTECTED]> wrote:
Has anyone had any experience using FXO and FXS gateways to extend
legacy PBX extensions to remote users
I disagree on this, you will have to create a dialplan in the
panasonic to tell it when to go over the ISDN circuit.
On 1/24/07, Scott Pinhorne <[EMAIL PROTECTED]> wrote:
If you use a Vegastream gateway on the actual incoming ISDN circuits then
you won't even need to touch the Panasonic to integ
Hi,
In the system, there are realtime and realtimeupdate to access data
in realtime model. Does it include realtimeinsert and realtimedelete
such that they can be used to manipulate the database more completely?
___
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Hello Asterisk fans,
I try to connect an analog modem to Asterisk. The modems are connected
e.g. to alarm systems or a cash terminals (POS). As PSTN-Interface I'm
using a Wildcard TE110P (E1).
Is it possible to connect the modems to an ATA?
Which ATA I should use for that scenarios?
Cheers
Noc Phibee wrote:
Hi
i use a lot of Grandstream GXP2000 with BLF
How to set up on the same key BLF blinking call interception?
So that someone is able to take a call that is destinated to another user
phone
It's called "Call Pickup".
http://www.voip-info.org/wiki-PBX+Call+Pickup
__
Hi list,
Does anyone know any ways to have mutiple parking lots? I've got a pbx
that 2 customers share, both need their own, and then have lights on
the phone flash when they park the call (snom phones). Any ideals I'm
not thinking of?!?
Any help would be great!
Thanks
Ron
_
I ran into this problem with an early batch of IP650s. Polycom's firmware
version 2.0.3b made this issue go away.
Speaking of Polycom firmware, anyone have an up to date source for the
stuff? The site I ordered from took down their FTP site that had it.
:(
-Kenneth
Hmm... not too sure what's up with this one. I've only used AMD with
Zap channels, so I don't know if there are any hidden gotchas with using
SIP.
Has anyone else used app_amd with SIP calls?
-MC
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Halliday
S
Hi all,
Has anyone faced an issue when sending DTMF from FXS of one SPA3K to FXO of
another SPA3K through asterisk?
Im not able to send it properly. Wanna be sure if its an issue faced by
all..
If you have a fix for it, pls guide me.
Thanks
Dan
___
On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote:
> In the meanwhile, use IAX, which understands about NAT pretty well.
> If you have multiple SIP phones on a LAN behind a NATing router, just
> put a small asterisk box on the LAN. It can manage your hairpin
> calls internally, save you ba
On Tue, Jan 23, 2007 at 02:01:43PM -0600, Tom wrote:
> Has anyone found a high quality wireless headset that works well with
> Cisco 7960 IP phones on an asterisk system?
>
> I tried the vxxi offering but the sound quality was pretty bad.
>
> Since these are pricey, I don't want to sample blindl
Hi,
I'm experiencing an issue with my x86_64 machine containing a Hauppauge
PVR-500 (ivtv) and a Digium TDM400p (wctdm, part of zaptel) PCI cards.
Independently of each other both cards work fine, but once the wctdm driver
is loaded and mythtv tries to record something on the PVR-500 the wctdm
dr
From: Brad Templeton <[EMAIL PROTECTED]>
On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote:
> In the meanwhile, use IAX, which understands about NAT pretty well.
> If you have multiple SIP phones on a LAN behind a NATing router, just
> put a small asterisk box on the LAN. It can manage y
C F wrote:
Which panasonic system?
I'm assuming you are talking about the TDA line. If so get a IP
Gateway card on the TDA system, that card uses h323, then configure it
with asterisk as h323, or my favorite, get a PRI card on the TDA sysem
(unless it's a TDA50 then the option is not available)
Hi,
On Thu, 2007-01-25 at 08:03 +0100, Stefan van der Eijk wrote:
> Hi,
>
> I'm experiencing an issue with my x86_64 machine containing a
> Hauppauge PVR-500 (ivtv) and a Digium TDM400p (wctdm, part of zaptel)
> PCI cards. Independently of each other both cards work fine, but once
> the wctdm dri
There is an SVN branch with this feature:
http://svn.digium.com/view/asterisk/team/oej/multiparking/
I had hope this would be a feature added to Asterisk 1.4, but fail to
see it on the changelog.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ron
McCarth
I had the exact same problem, removing the hardware echo fix the problem but
this is not a solution for a production system. I'm now using another brand
of hardware.
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Webster,
Andrew
Envoyé : 23 janvie
Hello everybody,
I was wondering if anybody knows how to make channel IDs different if all
call are coming from the same host:
core show channels
Channel Location State Application(Data)
SIP/sip-ny1.stanapho [EMAIL PROTECTED]:4 Up Playback()
SIP/sip-ny1.stanapho [
On 1/25/07, Matteo Brancaleoni <[EMAIL PROTECTED]> wrote:
Hi,
On Thu, 2007-01-25 at 08:03 +0100, Stefan van der Eijk wrote:
> Hi,
>
> I'm experiencing an issue with my x86_64 machine containing a
> Hauppauge PVR-500 (ivtv) and a Digium TDM400p (wctdm, part of zaptel)
> PCI cards. Independently
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