hi,
This is the new update_call_counter() which works fine for me:
/*! \brief update_call_counter: Handle call_limit for SIP users
* Note: This is going to be replaced by app_groupcount
* Thought: For realtime, we should propably update storage with inuse
counter... */
static int update_call_co
Hello Everyone,
I’ve recently been attempting to add a soft phone for
us to use outside our office setup. The box is a Asterisks / Fonality
server. I’ve added a new entry in the sip.conf for a
‘siptest’ user, and this allows the sip phone itself to login to
the server great. The p
>> Might the SPA-841 be crashing and rebooting? With the current
>> firmware (v. 3.1.4) I often see my phone hang and flash all its lights
>
> Really? For me the 841 is a quite stable phone. Out of the 15 we have
> in the office neither one crashed in the past 3 months. And they are
> used heavil
Hi all
Does any one give me Good calling rates to UAE
i have huge traffic going to come soon
contact me offline
ram
___
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Code Lover wrote:
Hi all,
I was trying to use G.723.1 codec for my terminator as Pass through.
But when the second party pickup phone the call is going dropted
automatically with the following error:
No path to translate from SIP/123456-fca7(1) to SIP/myterminator.com-ff11(4)
Dec 1 10:54:39 WA
With regards to #3, the advantage of putting in a PRI is that you get
23 or 24 channels that are absolutely reliable and have signal/voice
quality that is essentially as good as it gets. A PRI has got to be one
of the most reliably engineered basic services that a telephone company
can offer. With
You can leave the stuff in callback.agi the way it is.
[enhanced-outgoing]
exten => _1XX,1,Dial(SIP/000.000.000.000/${EXTEN})
exten => _1XX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten => _1XX,3,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten => _1XX,4,Dial(IAX2/[EMAI
We are currently an ISP offering broadband
and wireless internet connections. We are planning to start offering
VoIP services to our current customers. We have decided to use Asterisk
as our PBX software.
If I may, I have a multi-part question.
1) If we do use Asterisk, what would be a
goo
Does anyone have any experience with SellVoip.net?
Their DID pricing and termination pricing seems pretty good. How
is their service? How easy are they to contact should any problems arrise?
-jglucky
___
--Bandwidth and Colocation provided by Easy
Hi all
I have setup with [EMAIL PROTECTED]
with Local extentions and trunk to make out going calls
but i see some echo problem when my friend authenticate with my server and calling out side
so how can i make the call directly interact with VoIP provider after authenticated with My server
r
> Might the SPA-841 be crashing and rebooting? With the current
> firmware (v. 3.1.4) I often see my phone hang and flash all its lights
Really? For me the 841 is a quite stable phone. Out of the 15 we have
in the office neither one crashed in the past 3 months. And they are
used heavily. The pho
accessline.biz
Had them for over a year. $6.95/mo
snacktime wrote:
On 12/2/05, Rich Adamson <[EMAIL PROTECTED]> wrote:
I know this isn't an * question, but...
Does anyone know of a Fax over IP Provider?
Client looking to dump 3 regular fax lines, and bring broadcast faxes
in house (aroun
"Dave Morrow" <[EMAIL PROTECTED]> writes:
>Hi all, I have been plagued by an issue with my SPA-841 phones. The
>issue is that frequently, usually after a period of inactivity on the
>phone, an incoming call will be missed by the phone. The call works,
>cause the caller ends up at
David woodhouse,
I have basically done the following:
cvs -d :pserver:[EMAIL PROTECTED]:/home/cvs login
password anoncvs
cvs -d :pserver:[EMAIL PROTECTED]:/home/cvs co chan_bluetooth
compiled asterisk
chkconfig bluetooth on
service bluetooth start
hcitool dev will show the bluetooth dongle MAC
Try the XLite softphone - it's Free:
http://www.xten.com/index.php?menu=download
You can also buy a fancier version called EyeBeam..
Sorry can't help w/ your Linux problem...
H
On 12/2/05, Vladimir Montealegre <[EMAIL PROTECTED]> wrote:
> Hello to all, i have now two questions
>
> the first is,
Title: Linksys SPA-841 Missing Calls
Hi all, I have been plagued by an issue with my SPA-841 phones. The issue is that frequently, usually after a period of inactivity on the phone, an incoming call will be missed by the phone. The call works, cause the caller ends up at voicemail, but the
Hello,
I hope, somone can help me.
When I try to register an ata (sipura 2002 for example), it is successfull, when
a device is installed since a few weeks on the asterisk.
It isn´t successfull, when it is a completly new device, I added to the
asterisk.
Have someone any idea?
Many thanks!
Hello Joe,
I asked the same question. It is probably a
combination of things, hardware issues like PCI bus latency, plus an issue of
interest. I suspect the big companies have managed to cover this, but also
because they charge a lot for their equipment.
Being open source, yes you could w
Hi,
Same result with dial:
-- Executing DeadAGI("SIP/205-0231", "b") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/b
-- AGI Script Executing Application: (Dial) Options: (IAX2/24)
Dec 3 01:16:52 WARNING[20212]: chan_iax2.c:2747 create_addr: No such host:
24
Dec 3 01:16:5
I had the same problem today (2pm PDT), arising when I try to install
zaptel that I got by issuing
# svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel
The following is rather dumb but does it:
1) remove all header files in /usr/include/asterisk
2) install asterisk, zaptel and libpri ava
Hi,
I'm testing a Sangoma analogue card with two modules cards installed
(2 x FXO and 2 x FXS total)
I'm actually doing the testing so far in a new install of [EMAIL PROTECTED] 2.1.
The kernel is seeing the card but I'm struggling with my understanding of the
zaptel.conf file and possibly the wanpi
On 12/2/05, Jess Coburn <[EMAIL PROTECTED]> wrote:
> I didn't see it there last time I loaded [EMAIL PROTECTED] but maybe something
> has changed recently, this was about 2 months ago. I'm probably going to
> attempt to tackle it this weekend.
>
>
> On 12/2/05, Mike Dent <[EMAIL PROTECTED]> wrote:
Dear All,
In one specific solution, I am using chan_h323 with Asterisk 1.2 for
H323/PSTN gatewy.codec G729.
Checking the logs, I am getting this error:
Dec 3 01:21:32 WARNING[2655] channel.c: Prodding channel
'H323/ip$202.83.196.25:59216/17312' failed
What does it mean? I should be loosin
I have been wondering about echo canceling, it seems to be one of the major problems people have with asterisk. I've gotten it to acceptable levels (using mark2 aggressive), but everything I've read indicates that the echo canceling software isn't very effective.
My question would be, what do
On Fri, 02 Dec 2005 14:42:06 -0800, JP Carballo wrote:
>Bill Michaelson wrote:
>
>> Just curious...
>>
>> Is there anyone out there who has given this outfit money and actually
>> received any service from them?
>>
>>
I used them for a while earlier this year. When the service worked it
was pret
Alejandro Vargas wrote:
2005/12/2, ram <[EMAIL PROTECTED]>:
if you are using
AMP
go to trunk
and start regitering your account
Humm... what I'm trying to do, and what is this thread subject, is to
connect asterisk-to-asterisk.
Then... I go to trunks, create a new iax trunk, invent
Bill Michaelson wrote:
Just curious...
Is there anyone out there who has given this outfit money and actually
received any service from them?
I only have a 1800 number from them, but no problems so far.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look lik
Hi,
When you say it has a higher priority what does that mean?? Does that
mean that a call to extension 600 always goes to the higher priority
unless it is busy?
Thanks
Andy Kuo wrote:
Hi,
The one in [big-business] has higher priority than the one in
[small-business]
Included context h
On Fri, 2005-12-02 at 16:32, [EMAIL PROTECTED] wrote:
> >> After working on the problem for
> >> several days, I finally built a new box and installed Asterisk 1.2 on
> >> it. Using this new 1.2 box I no longer see the "Maximum retries
> >> exceeded on call" warnings on the console but still experi
I didn't see it there last time I loaded [EMAIL PROTECTED] but maybe something has changed recently, this was about 2 months ago. I'm probably going to attempt to tackle it this weekend.
On 12/2/05, Mike Dent <[EMAIL PROTECTED]> wrote:
On 12/2/05, Jess Coburn <[EMAIL PROTECTED]> wrote:
> Guys,>>
On 12/2/05, Rich Adamson <[EMAIL PROTECTED]> wrote:
>
> > I know this isn't an * question, but...
> >
> > Does anyone know of a Fax over IP Provider?
> >
> > Client looking to dump 3 regular fax lines, and bring broadcast faxes
> > in house (around 700 per day) in house. Their broadcast fax servic
I see that MVOX makes a pair of low cost speakerphone units; the MV100
is USB connected and the MV900 is Bluetooth enabled. Does anyone have
an experience with these?
I'd like to know if the USB model works well with soft phones. And also
if the bluetooth model adds an additional latency to the ca
On Fri, 2005-12-02 at 16:07, Chuck Bunn wrote:
> Hi,
>
> If I have an extension in a context and I have another context with the
> same extension and I include the second context in the first does this
> cause a conflict or does Asterisk know that there is a 600 extension in
> each context
Simple.GSM is a voice codec. It is not designed in any way to compress music. It is _only_ designed to work with voice. You will therefore never be able to hear good quality music over a GSM codec. No matter how much you try.
sorry.MarkOn 11/21/05, Christoph Rothe <[EMAIL PROTECTED]> wrote:
> I know this isn't an * question, but...
>
> Does anyone know of a Fax over IP Provider?
>
> Client looking to dump 3 regular fax lines, and bring broadcast faxes
> in house (around 700 per day) in house. Their broadcast fax service
> is currently outsourced.
>
> Can anyone point me in the co
Hi,
The one in [big-business] has higher priority than the one in [small-business]Included context has lower priority.
Hope this helps.
Andy
On 12/2/05, Chuck Bunn <[EMAIL PROTECTED]> wrote:
Hi,If I have an extension in a context and I have another context with thesame extension and I include
On Fri, December 2, 2005 22:50, Francesco Peeters said:
> On Fri, December 2, 2005 21:45, Kristof Hardy said:
>> Francesco Peeters wrote:
>>> Does anybody have any experience in this?
>>> I am using * 1.2 BRIstuffed 0.3.0 Pre1
>>
>> No experience on that, but there's an updated bristuff (0.3.0pre1b
On Fri, December 2, 2005 21:45, Kristof Hardy said:
> Francesco Peeters wrote:
>> Does anybody have any experience in this?
>> I am using * 1.2 BRIstuffed 0.3.0 Pre1
>
> No experience on that, but there's an updated bristuff (0.3.0pre1b),
> maybe try that one?
>
> This is 1 issue that's fixed:
> -
What version firmware are you running on your Cisco Phones? We are running
Asterisk 1.2 with the 7.4 firmware. The latest is 7.5 but there are some
strange things that happen with this firmware. If I were you I would try a
different firmware on the phones. Hope this helps.
Jeremiah
Help! I'
> you need to recompile zaptel drivers... they use the kernel
> headers to build and since you change the kernel headers by
> upgrading your kernel... time to recompile
> cd /usr/src/zaptel
> make clean; make install
Sorry to say, I've already tried that after each kernel recompile with
differen
I read that it was supposed to integrate with X10 modules and the that the
@home was reference to home automation.
That being said, I have never seen any X10 specific functionality.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---
> On Fri, 2005-12-02 at 14:22, [EMAIL PROTECTED] wrote:
>> Help! I've encountered some problems with Asterisk that IÂm unable to
>> solve. We have been running Asterisk version 1.0.9 for many months
>> using a few local network connected Cisco 7960 phones as SIP clients.
>> All our phones are curr
Hi,
In Asterisk 1.2 according to the wiki and I quote:
"It is now possible to use multi-digit extensions in the exit context
for a queue (although you should not have overlapping extensions, as
there is no digit timeout). This means that the EXITWITHKEY event in
queue_log can now contain a ke
"Message not sent "
Right, not the comma separator
I still have 2 small questions:
1) I have the following warnings:
WARNING[2943]: chan_zap.c:10916 setup_zap: Ignoring :callreturn
WARNING[2943]: res_musiconhold.c:124 spawn_mp3: /usr/share/asterisk/mohwav is
not a valid directory
WARNING[2943
Hi,
If I have an extension in a context and I have another context with the
same extension and I include the second context in the first does this
cause a conflict or does Asterisk know that there is a 600 extension in
each context
[big-business]
exten => 600,1,Dial(ZAP/1,20)
include =>
i know the hylafax but i dont have idea if this work over ip, if the pc
receive a fax you can program the hylafax to send the fax received via email
or send fax via whfc to the hylafax server
- Original Message -
From: "Kurth Bemis" <[EMAIL PROTECTED]>
To:
Sent: Friday, December 02, 2
On Fri, 2005-12-02 at 14:22, [EMAIL PROTECTED] wrote:
> Help! I've encountered some problems with Asterisk that Im unable to solve.
> We have been running Asterisk version 1.0.9 for many months using a few local
> network connected Cisco 7960 phones as SIP clients. All our phones are
> current
I know this isn't an * question, but...
Does anyone know of a Fax over IP Provider?
Client looking to dump 3 regular fax lines, and bring broadcast faxes
in house (around 700 per day) in house. Their broadcast fax service
is currently outsourced.
Can anyone point me in the correct direction? A
Hi:
Once i have seen the post of Darren Wiebe of
suggestion of a callback configuration in
extensions.conf and it was like this:
[callback]
exten =>
_.,1,AGI(callback.agi,LAKEVIEW,1234567890,9998,,meetme,enhanced-outgoing)
But i didnt know what to add in meetme and
enhanced-outgoing contexts.
Francesco Peeters wrote:
Does anybody have any experience in this?
I am using * 1.2 BRIstuffed 0.3.0 Pre1
No experience on that, but there's an updated bristuff (0.3.0pre1b),
maybe try that one?
This is 1 issue that's fixed:
- chan_zap/libpri fixes (stuck B channels)
cheers
> Subject: Re: [Asterisk-Users] Linksys SPA-941 Admin Guide
John Novack <[EMAIL PROTECTED]> wrote:
> Is the 941 available yet?
Yes, we've received 2 for evaluation from VOIPsupply.com.
> What improvements have they made.
>From an external view, it's a drastically different phone. It looks
more
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Chris Bagnall wrote:
> Hello all,
>
> I recently upgraded the kernel on one of the phone servers I have at home
> (dual Xeon 2.4) from 2.6.11 to 2.6.14 in the usual way, copying the .config
> file across and building the new kernel. Now ztdummy is ref
Hi!
> The first result is ok (-1) but not the second and the third.
> Why do I get different results for the same command?
Hm... u might want to try this:
# Check
print "EXEC ChanIsAvail IAX/24\n";
$result = ;
print "VERBOSE \"$result\" 0\n";
$result = ;
# Check
print "EXEC ChanIsAvail IAX/24
My HFC-PCI card is losing connection with the ISDN PSTN for minutes at a
time on a very regular basis...
This is of course unacceptable for any PBX...
The Dutch PSTN disconnects the D-channel once a minute.
About one minute every 5/10 minutes the D channel goes down, and doesn't
get beyond F6 (DE
Steve,
Thanks for that, now I know it can be done, do you have any references
as to how it is accomplished. Pointers are fine, Im more than happy to
RTFM and see what I can work out, but im having trouble locating it :)
Many thanks,
Jo
Steve Rawlings wrote:
- Original Message - From
Steve Blair & Jeremy Koski: THANK YOU
sntp_mode: unicast
solved the problem. We're using v7.4.
--
Michael Coburn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Blair
Sent: Tuesday, November 29, 2005 6:09 PM
To: Asterisk Users Mailing List - Non-C
On 12/2/05, Chuck Bunn <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Is it possible to get to a menu system while in a call queue.
Yes, set the context in queues.conf. Then put the extensions you want
available to callers in the queue in that context.
Chris
___
I am also seeing this in my logs.On 11/30/05, Aaron Daniel <[EMAIL PROTECTED]> wrote:
We just upgraded our current asterisk cluster to the release version ofAsterisk 1.2.0. Strange enough, out of the 11000+ calls, only 720 (andcounting) have a disposition of FAILED in the cdr's. These 720+ have
on
Hi,
Is it possible to get to a menu system while in a call queue. I want
users to be able to hit the '*' and be able to goto a menu system from a
queue if they so desire. I thought the following would do this but no
dice...
*
extension.conf
[general]
#include macros.incl
#in
On Fri, 2005-12-02 at 05:59, Code Lover wrote:
> Hi,
>
> Do you know from where i can buy g723 codec. for g729 i can buy it
> from digium.com. But Please let me know from where i can get g723
> codec.
>
> And the codecs purchasing can solved my problem?
>
>
> --
> Thank You,
> Code Lover
Becom
I followed all the steps for setting up bluetooth
headset.
I call into my laptop, I head the ringing on my headset, I see ringing
on the console.
When I answer the headset I get MANY errors/warnings on screen about
chan_bluetooth/chan_bluetooth.c:685 sco_thread: wrote <48 to sco
and I head NO
Help! I've encountered some problems with Asterisk that Im unable to solve. We
have been running Asterisk version 1.0.9 for many months using a few local
network connected Cisco 7960 phones as SIP clients. All our phones are
currently internal so there is no NAT involved. We were not having a
Hi all,
I would like to have two asterisk servers in a "cluster". From what I
understand using a mysql database I can store all of my peer/user
information in the db and share this between servers. I can then take my
polycom phone and register it to both of the asterisk servers at the
same ti
- Original Message -
From: "Jo Knight" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, December 01, 2005 10:32 PM
Subject: [Asterisk-Users] Asterisk as a gateway to Index PBX
Hi,
Is it possible to have an Asterisk act as a gateway to
On 12/2/05, Jess Coburn <[EMAIL PROTECTED]> wrote:
> Guys,
>
> I'm curious if it's possible to asterisk at home and the sangoma T1 cards
> together. I realize asteriskathome is traditionally used for at home, but
> I'd like to use it in a small office with a T1 and our hardware is a Sangoma
> card.
This is the output at the CLI when this happend.
-- Accepted AUTHENTICATED TBD call from 201.128.234.38 -- Accepting DIAL from 201.128.234.38, formats = 0x4 -- Executing Dial("
IAX2/[EMAIL PROTECTED]/5", "IAX2/111|120|tT") in new stack -- Called 111 -- Call accepted by 201.153.202
At 10:16 AM -0600 12/2/05, Kevin P. Fleming wrote:
Matthew Simpson wrote:
Is there a way in asterisk to configure a sip invite timeout ? It
seems to be about 30 seconds right now which is too long. I would
like to have asterisk return congestion if a host does not respond
to an invite withi
Hello all,
I recently upgraded the kernel on one of the phone servers I have at home
(dual Xeon 2.4) from 2.6.11 to 2.6.14 in the usual way, copying the .config
file across and building the new kernel. Now ztdummy is refusing to run, and
gives the following errors in dmesg:
ztdummy: Unknown symbo
I was experimenting with ael and first thing I tried to do was move the
inclusions for the default context form the extensions.conf file to the
extensions.ael file
Can a context that is defined in extensions.conf be included by the ael
parser?
Just asking in case anyone has already discovered thi
On Fri, Dec 02, 2005 at 12:05:08PM -0600, Kevin P. Fleming wrote:
> Louis-David Mitterrand wrote:
>
> >to Asterisk Extension Language (AEL) style.
> >I haven't found anything in the docs, wiki or examples about it.
>
> I don't believe hints are supported in AEL at this time.
Thanks for the heads
Hi list:
I'm having problem with some DIAXY ATA FROM DIGIUM, I have 3 of them in different points, all of them register
to a central asterisk server. If i call from any of the ATA's to Asterisk or Asterisk's to ATAs. But when any ATA's want to talk
to another ATA's.. The ATA's rings, but when
On Thu, Dec 01, 2005 at 06:51:54PM -0800, John Todd wrote:
#exec /usr/bin/curl -s
http://webserver.domain.com/privatefiles/username-to-numbers >
/etc/asterisk/username-to-numbers
#include username-to-numbers
Nice. However, what happens if curl takes longer than expected? your
reload waits
Dean Collins wrote:
> who's done it? and how much money are they talking about? I've been
> looking to pay for something like that for a while.
-Original Message-
From: Neil Stratford [mailto:[EMAIL PROTECTED]
Sent: 24 November 2005 09:30
To: John Martin; [EMAIL PROTECTED]
Subject: Re: Fwd
Louis-David Mitterrand wrote:
to Asterisk Extension Language (AEL) style.
I haven't found anything in the docs, wiki or examples about it.
I don't believe hints are supported in AEL at this time.
___
--Bandwidth and Colocation provided by Easynews.c
Could you send it patch please.
On 11/30/05, Paradise Dove <[EMAIL PROTECTED]> wrote:
btw, i've patched this part of code and now its working fine for me.i'm going to upload it.Paradise Dove
On 11/30/05, Kevin Hanson <[EMAIL PROTECTED]> wrote:> Paradise Dove wrote:>> >>Yes with version 1.2. I h
How could a CVS update fix an error you have made during installation?
Steve
René Enskat [Teamware GmbH] wrote:
so is there a solution in the next cvs udpate?
*Von:* René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED]
*Gesendet:* Donnerstag, 1. Dezember 2005 14:47
*An:* 'asterisk-users@
Tony Mountifield wrote:
In article <[EMAIL PROTECTED]>, John Daragon <[EMAIL PROTECTED]> wrote:
Hi;
I've been looking for an arbitrary way of discovering when the last
user has left a Meetme conference...
It occurred to me that I could launch an agi script to keep watch over
the conference an
Hello,
I am trying to convert my hint priorities from the old style:
exten => 2130,hint,SIP/0146472130
to Asterisk Extension Language (AEL) style.
I haven't found anything in the docs, wiki or examples about it.
How should I do it?
--
Sigs have been known to cause cancer in California.
_
(ENGLISH VERSION AT THE END)
Hola lista:
Requiero saber si alguien tiene un cliente o empresa donde se encuentren montado algun
Asterisk como PBX de tamaño mediano (al menos unas 50 extensiones). Esto para dar una
demostracion a un cliente mio que esta interesado en invertir en Asterisk.
Has anyone encountered 'bad' cdr logging in * 1.2?
Since upgrading to 1.2 (bristuffed) and asterisk-addons 1.2, sometimes
the clid is 'messed' up. I use AMP to look at the reports, but when I
look in the cdr database, it's the same, here's an example:
2/12/2005 15:06:02 "Tech:" <ÀB > ÀB 2 e
Dave Morrow wrote:
> Can anyone help with;
>
> Dec 2 12:11:56 WARNING[2562]: res_musiconhold.c:421 spawn_mp3: Found no
> files in '/var/lib/asterisk/mohmp3'
> Dec 2 12:11:56 WARNING[2562]: res_musiconhold.c:488 monmp3thread:
> Unable to spawn mp3player
> Dec 2 12:20:16 WARNING[2562]: res_musico
Before 1.2.0 I used Mark2 with AGGRESSIVE turned on
I would recommend switching to KB or MG in 1.2.0, we have done this with
very good results (using KB now)
Jared Armstrong
-Original Message-
From: Patrick Fortin [mailto:[EMAIL PROTECTED]
Sent: Friday, December 02, 2005 9:17 AM
To:
In article <[EMAIL PROTECTED]>, John Daragon <[EMAIL PROTECTED]> wrote:
> Hi;
>
> I've been looking for an arbitrary way of discovering when the last
> user has left a Meetme conference...
>
> It occurred to me that I could launch an agi script to keep watch over
> the conference and do something
Title: Music on Hold Error
Can anyone help with;
Dec 2 12:11:56 WARNING[2562]: res_musiconhold.c:421 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3'
Dec 2 12:11:56 WARNING[2562]: res_musiconhold.c:488 monmp3thread: Unable to spawn mp3player
Dec 2 12:20:16 WARNING[2562]: res_mus
@home by no means means it just for the home - its Asterisk nothing more, nothing less. I don't think the @home designation was meant to limit it by perception. I read somewhere it was called @home for another reason, anyone know more?
RegardsRobOn 12/2/05, Jess Coburn <[EMAIL PROTECTED]> wrote:
Th
Thanks Saul,
What you do to get the Sangoma to install and how'd you go about compiling the zaptel source did you just download zaptel and extra RPMs? I'm by no means a linux guru...
Jess
On 12/2/05, Saul Diaz <[EMAIL PROTECTED]> wrote:
Jess Coburn wrote:> Guys,>> I'm curious if it's possible to
Jess Coburn wrote:
Guys,
I'm curious if it's possible to asterisk at home and the sangoma T1
cards together. I realize asteriskathome is traditionally used for at
home, but I'd like to use it in a small office with a T1 and our
hardware is a Sangoma card. I know all I need to do to get the
Guys,
I'm curious if it's possible to asterisk at home and the sangoma T1 cards together. I realize asteriskathome is traditionally used for at home, but I'd like to use it in a small office with a T1 and our hardware is a Sangoma card. I know all I need to do to get the sangoma working is recomp
Hello All I'm bought VoiceTronix Card (Openswitch), it's bad card and resaller (www.telephonyware.com) give me are old card (for one year old). than, after that, my card is fault. I didn't received any help from telephoneware or voicetronix. I don't like voicetronix and telephoneware. i notic
> Just wandering what solution worked to eliminate echo on your setup.
>
> I am trying every solutions I can find on the wiki and none is working
> perfectly.
>
> We have asterisk 1.2.0
> 3 x digium TDM400P
> 30 Snom320 + 5 Snom360
>
> For now the best setup I have is using Mark2 Echo cancel.
Hi;
I've been looking for an arbitrary way of discovering when the last
user has left a Meetme conference...
It occurred to me that I could launch an agi script to keep watch over
the conference and do something when the user count reaches zero... And
of course, I can do that directly from the d
> > > Ouch ... error while writing audio data: : Broken pipe
> >
> > If you are talking about the Ouch message, yes lots of people have seen
> > the error and its usually the result of some misconfiguration in one of
> > your files (likely zapata.conf).
>
> Correct me if I'm wrong, but isn't that
Hello,
extensions.conf:
[mytest-in]
exten => 1,1,NoOp(${MYVAR1})
exten => 1,n,Wait(20)
exten => 1,n,Hangup()
[mytest-out]
exten => 1,1,NoOp(${MYVAR1})
exten => 1,n,Dial(Zap/g1/06111,10,H|g)
my test dial.out file:
Channel: Local/[EMAIL PROTECTED]
MaxRetries: 0
Context: mytest-in
Extension: 1
Hi list,
I have a problem with a SIP trunk on my * box: I can
originate calls but I can’t receive them.
The * box is behind a modem-router and as a private
address.
I think about a NAT problem but I don’t know
how to resolve it.
I have included some debug and configuration.
Th
Matthew Simpson wrote:
Is there a way in asterisk to configure a sip invite timeout ? It seems
to be about 30 seconds right now which is too long. I would like to
have asterisk return congestion if a host does not respond to an invite
within 5 seconds.
Asterisk 1.2 will use a T1 timer (retr
I am using http://www.gmane.com/ with my newsreader.
You still have to be a list member to post.
You can then turn on the vacation option in the list manager to stop
receiving emails.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
-
Is there a way in asterisk to configure a sip invite timeout ? It seems
to be about 30 seconds right now which is too long. I would like to
have asterisk return congestion if a host does not respond to an invite
within 5 seconds.
___
--Bandwidth and
we should be getting a limited number in a couple of weeks time.
Proper stocks will be arriving in January - www.provu.com
Paul.
Senad Jordanovic wrote:
[EMAIL PROTECTED] wrote:
There is a review on the homepage at http://voipspeak.net
It has been available for a few weeks, it
One simple way to overcome this problem would do to make an attended
transfer to check whether the receiving person is available and willing to
take the call, and then an unattended transfer to discharge the operator
of the call.
l.
On Fri, 02 Dec 2005 16:21:39 +0100, James Armstrong
<[E
This must be similar to a problem I have seen here. Some times the main
operator's phone will stop ringing when a call comes in on the queue
while the other phones still ring. I have to reset her phone which
causes a re-login to get it working again. It must stop after she does
an attended tran
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