- Johansson Olle E <[EMAIL PROTECTED]> wrote:
> No. It's certainly possible but at this time there's no interaction
> between
> the RTP clients, the various channel drivers.
I believe this is incorrect; all the RTP-using channel drivers supply
'ast_rtp_bridge' as their native bridge method
- jan sarin <[EMAIL PROTECTED]> wrote:
> Does anyone know when thease will be released and what they will cost
> when released? Thanks!
They will be released when they are ready, which will probably be no later than
the end of this week. They will cost the same as the TE406P and TE411P, both
hi
this site now also has a wiki, and the whole site will be migrated to
a wiki-only after some time
so just add your stuff :)
roy
On Jun 20, 2006, at 6:15 PM, Roy Sigurd Karlsbakk wrote:
hi all
I just setup a new site, perhaps soon a wiki, to collect what's out
there of useful bac
Did You try to specify extension i context ?
Dial(IAX2/myiax2peer/[EMAIL PROTECTED]/extension)
fragment from:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial
"
[iaxhost1-out]
exten => 99205,1,Dial(IAX2/value3:[EMAIL PROTECTED]/99105)
; A more complicated extension example:
exten =
- unplug <[EMAIL PROTECTED]> wrote:
> In my configuration below, I use realtime architecture in our
> system. I have one device attached to each asterisk server. There
> is
> no record when I issue "sip show users or sip show registry in CLI.
> I
> wonder how can I know who is registered
> On Tue, Jun 20, 2006 at 10:33:29PM -0700, Gabriel Afana wrote:
> > Is this possible?
> >
> > I checked the Wiki and googled it and couldn't find the answer
>
> voicemail.conf is read and written to by several components.
>
> Asterisk's standard config parser reads it and has no problem
> unders
On 6/21/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
- Johansson Olle E <[EMAIL PROTECTED]> wrote:
> No. It's certainly possible but at this time there's no interaction
> between
> the RTP clients, the various channel drivers.
I believe this is incorrect; all the RTP-using channel driver
In general, you are talking of distributed conferencing, which in SIP
it was tried once to standardize but never reached anything. It is
just not commercially popular, i guess.
Now, this doesn't mean that it cannot be done or that it has not been
done ... but it is "propietary" implementations. A
Yep, cmd setCDRUserField will do this for you assuming you have the field set up. I'd be keen to hear if anyone has a way of achieving the same thing across multiple user fields to save having to explode multiple values out of a single user field seperately.
SimonOn 6/20/06, trixter aka Bret McDane
Is it the way for asterisk realtime system?
register:
UA1 --register--> asterisk1 -> store user information in DB
UA2 --register--> asterisk2 > store user information in DB
UA1 --invite-UA2---> asterisk1 > asterisk1 query UA2 information in DB
->asterisk1 -invite--> U
Hi,
Here's an extract from LogWatch:
- Kernel Begin
WARNING: General Protection Faults in these executables
asterisk : 1 Time(s)
-- Kernel End -
Asterisk killed himself too after some extensives
> Andreas Sikkema wrote:
> >
> > Hi,
> >
> > To combine two sources of CDR's I want Asterisk to save the
> SIP callid for
> > all calls. I know there's a variable that contains the SIP
> CallID value,
> > but is this the callid value of the incoming INVITE message or the
> > outgoing
> > message
Hi,
I am also testing asterisk with H323, with the channel included in the
latest sources. It works ( i had some problems with media
configuration when calling from an SJPhone ... but it seems more an
SJPhone problem than asterisk).
I also bridged from SIP to H323 ... it works fine.
I have a que
Hello,
Could someone please help refer me to a
resource where I can find material on how to write IVR applications. I am using
[EMAIL PROTECTED] ver. 2.8.
Thanks
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Asterisk-Users mailing lis
Hi,
I've been asked if it is possible to
allow a user to listen in on another users call for training purposes.
I know there are ways to monitor zap channels with apps like zapscan
but I don't think this would be appropriate for these users. Can
I do:
Call comes in for user on ext 3210
User on
Andreas Sikkema wrote:
Andreas Sikkema wrote:
Hi,
To combine two sources of CDR's I want Asterisk to save the
SIP callid for
all calls. I know there's a variable that contains the SIP
CallID value,
but is this the callid value of the incoming INVITE message or the
outgoing
message? Are the
Hi,
You can try ChanSpy http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy.
Idris
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 21, 2006
12:23 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Monitor
a particular SIP cal
Hallo group members
Could You tell me a h.323 soft phone that runs well with asterisk.
I tried the following so far, but in general I cannot compile them (fc.3) or I
cannot configure them to run with asterisk:
http://www.pacphone.com/downloads/PacPhoneSetup117.exe - cannot configure
http://www.sj
On my Asterisk server it (chan_h323) gets 2-3 deadlocks every hour
regardless of openh323/pwlib and asterisk versions (since the
channel_h323 was not updated for a long time). The load is about 25-30
simultaneous calls (from h323 to zaptel, IAX and SIP).
I have another Asterisk server. There's abo
I'm wanting to capture the zap channel that a sip channel has connected to.
I came across the ${BRIDGEPEER} variable documented on the wiki, and if
I show channel SIP/ when a call is connected I can see
BRIDGEPEER as one of the channel variables.
However ${BRIDGEPEER} is not set when I want i
BTW, do you mean this function will be included in next release? When
will be the next release available?
On 6/21/06, unplug <[EMAIL PROTECTED]> wrote:
Is it the way for asterisk realtime system?
register:
UA1 --register--> asterisk1 -> store user information in DB
UA2 --register--> asteris
Anyone know why with Chanspy if I dial a specific extension and press #
I get a random agent and not the one I dialed?
Idris AVCI wrote:
Hi,
You can try ChanSpy
http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy.
Idris
---
Doh, I am so stupid.
The macro is executing in the zap channel !
Julian.
Julian Lyndon-Smith wrote:
I'm wanting to capture the zap channel that a sip channel has connected to.
I came across the ${BRIDGEPEER} variable documented on the wiki, and if
I show channel SIP/ when a call is connected
Ok,
So here's some information I've, to this point, left out. I applied
the patch to allow * to be pressed in queues to park a call (*270).
After reverting the patch the system seems stable. So, it almost
seems like the crash is directly related to the number of times
someone parks a call that
Hello,
We have problems with Asterisk and Sipura SPA-2002.
SPA is behind the NAT. Asterisk has nat=yes.
Sometimes call doesn't hangup when user finish the call and hangup the
headset.
In this case during all conversation SIP packets contains
Call-ID: [EMAIL PROTECTED]
but the final BYE packet
I had problems with sjphone ... same version as yours.
Finally, i managed to solve it by:
- in sjphone, media channels settings: untick "Use remote codec
preferences" and "Open audio streams after remote opened" ... it was
trial-error ... now it works (to Echo and Sip<->H323 call).
- in asterisk,
On 6/21/06, Matt <[EMAIL PROTECTED]> wrote:
Ok,
So here's some information I've, to this point, left out. I applied
the patch to allow * to be pressed in queues to park a call (*270).
After reverting the patch the system seems stable. So, it almost
seems like the crash is directly related to t
Hi,
after a few of upgrades, I noticed these messages in full debug log:
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan
Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring prilocaldialplan
Jun 21 12:58:11 WARNING[27273
Dear
I am using [EMAIL PROTECTED] , and I have 2 hard disks on the
system ,how can I put the database (CDR) on the second hard disk .
Regards
M. Khaled
Chehab
Monitoring & Operationg Engineer
Xplorium
Tel: +961 1
868686
Fax: +961 1
808810
e-mail: [EMAIL PROTECTED]
On 6/21/06, BJ Weschke <[EMAIL PROTECTED]> wrote:
On 6/21/06, Matt <[EMAIL PROTECTED]> wrote:
> Ok,
> So here's some information I've, to this point, left out. I applied
> the patch to allow * to be pressed in queues to park a call (*270).
> After reverting the patch the system seems stable. S
On Wed, Jun 21, 2006 at 01:57:36PM +0200, Mimmus wrote:
> Hi,
> after a few of upgrades, I noticed these messages in full debug log:
>
> Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype
> Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan
> Jun 21 12:58:11 WARNING[27273]
Does anyone know why this row:
exten => s,2,GotoIf($[${CALLERIDNAME:0:${LEN(${RGPREFIX})}} !=
${RGPREFIX}]?4:3)
generate this error:
ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_NE,
expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:
!=
^
?
I was unable t
Are there any traditional telephone set-looking handset for use with a
softphone? All the options I've found are the headset type. I'm looking
for something more traditional--it should look like a small deskset,
cellphone or cordless phone, perhaps with a dial pad and a couple of buttons
that int
Thomas Kenyon wrote:
> Doug Lytle wrote:
>
>> Thomas Kenyon wrote:
>>
>>> Is it neccesary to upgrade Zaptel at the same time as upgrading
>>> asterisk.
>>>
>>>
>>>
>> I do as a matter or course. Libpri, Zaptel, Asterisk, Asterisk-addons
>> and Sounds.
>>
>> Doug
>>
>>
> The
Do you have that patch/commit message from svn-commits where this
code was introduced so we can track it back?
On 6/21/06, Matt <[EMAIL PROTECTED]> wrote:
On 6/21/06, BJ Weschke <[EMAIL PROTECTED]> wrote:
> On 6/21/06, Matt <[EMAIL PROTECTED]> wrote:
> > Ok,
> > So here's some information I've
Hi,
> Does anyone know why this row:
> exten => s,2,GotoIf($[${CALLERIDNAME:0:${LEN(${RGPREFIX})}} !=
> ${RGPREFIX}]?4:3)
took me some squinting, but the parantheses seem correct - so I presume
the Asterisk parser can't cope with that convoluted an expression (using
a function within a variabl
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Tzafrir Cohen
>
> On Wed, Jun 21, 2006 at 01:57:36PM +0200, Mimmus wrote:
> > Hi,
> > after a few of upgrades, I noticed these messages in full debug log:
> >
> > Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring
> switchty
> "Kevin" == Kevin P Fleming <[EMAIL PROTECTED]> writes:
Kevin> I've just reviewed the code and this should be working
Kevin> properly... please do a 'set debug 3' and enable the 'debug'
I've found the problem. That's because I've loaded app_queue.so before
chan_sip.so in modules.conf.
--
It fuzzed on the first chunk, so I wiped the source directory, re
un-tarred asterisk so it was clean, and then manually applied the
changes.. doing removes and adds where necessary. So, unfortunately
no, I do not have the information, though I suppose I could run it
again.
Keep in mind, also, th
Hi!I've 2 asteriskAtHome;
How can I copy one database where are put all the sip authentificated registration to another one database on one other asteriskAthome so I've always the same Sip registrated and if one linux falls down I can run the other one without problems?
Which files must I copy?th
Ok,
Here's another bizarre one (no strange curve balls to throw this time :P).
I have several mp3 files of some easy listening music that I pulled
off some CDs we have. They sounds fine and are at a nice volume
level.
When I run this script I wrote:
(I run it by doing ./script < filename)
mpg12
> Hi
>
> How Can asterisk work as sip and h323 protocol in the same time ,and how
is
> the
> conversion protocol works .
>
> Please if u know send me how to active h323 protocol or the conversion
> protocol
>
*
No employee or agent is authorized to
"Mimmus" <[EMAIL PROTECTED]> writes:
> Hi,
> after a few of upgrades, I noticed these messages in full debug log:
>
> Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype
> Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan
> Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignori
Voip Supply has a number of USB handsets available... see http://www.voipsupply.com/index.php?cPath=95_258 I'm not affiliated with them beyond being a happy customer.
There are also a few USB handsets that look like desk phones. I have seen these offered in the $50-70 range, some even with sp
A quick Google returned this USB desk phone for $32http://www.inkjetcartridge.com/deskphone.html.
Michael
On Wed, 21 Jun 2006 08:19:44 -0400, klubarpop wrote:
>Are there any traditional telephone set-looking handset for use with a
>softphone? All the options I've found are the hea
On Tuesday 20 June 2006 17:28, Karl J. Vesterling wrote:
> I could care less about the color LCD, or browsing the web on the
> phone, but a good speakerphone built into a cordless WiFi is
> definitely a requirement that needs to be met before I purchase one.
Agreed, although every single manufactu
Are you asking about the internal Asterisk database? Or the
configuration files?
You don't need to copy anything about sip registration. Just either
copy the .conf files from /etc/asterisk (so copy all of /etc/asterisk
over). Or copy the MySQL database over and have it recreate the
files.
O
We use the AU-100 from (some Chinese company)
They work OK.
Our only issue is no having an external ringer.
We originally went with these because our Dell soundcards made the microphone
input sound awful.
USB sound devices have their own sound card on board.
--
--
Steven
http://www.glimasouth
> No, IMHO does it appear when you issue a reload command on
> the CLI. Because this options need a complete *-restart.
Yes, they appears when I issue a reload.
I will check if there are also when I restart.
Thanks
DV
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Which files must I copy?then..I'll use a ssh scritp for this, I want only
know which files I must copy...
the MySQL files are usually in "/var/lib/mysql". The databse you want
to copy is "asterisk"
hth
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Has anybody patched sucessfully cdr_addon_mysql.c , cause I get error . And does work cdrtool web interface for you? Maybe you can give me some advices
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To UNSUBSCRIBE
MVox makes a BlueTooth enabled speakerphone that works nicely with
softphones, but is not inexpensive. Details can be found here
http://www.mvox.com/
Cory J Andrews
VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
++
voice - 716.630.1555 X22
email - [EMAIL PROTECTED]
I'm using realtime for voicemail users, and for reasons that I don't yet
understand, when it doesn't get used for a while (like overnight), the first
connection attempt of the day will display this on the console.
Jun 21 07:54:00 ERROR[8112]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql:
Unknown
(Try again from the proper email address)
--Rob
-Original Message-
From: Rob Thomas
Sent: Thursday, 22 June 2006 12:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] syntax error
That's freePBX or AMP code that we've since fixed - The re
Third time's the charm.. (Email server is sending from wrong address!)
--Rob
-Original Message-
From: Rob Thomas
Sent: Thursday, 22 June 2006 12:22 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] syntax error
That's freePBX or AMP code tha
And I'll resend this one too. Silly scalix.
--Rob
-Original Message-
From: Rob Thomas
Sent: Thursday, 22 June 2006 12:19 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] zapata.conf: recent changes?
Looks like you've stopped compiling libpr
On Wed, Jun 21, 2006 at 04:06:07PM +0200, Mimmus wrote:
>
> > No, IMHO does it appear when you issue a reload command on
> > the CLI. Because this options need a complete *-restart.
> Yes, they appears when I issue a reload.
> I will check if there are also when I restart.
Restart? Who needs a r
Hi
i will forward a call to a remote server (only for one account)
is this sintax correct?
exten => 33347563,1,Dial(SIP/[EMAIL PROTECTED])
thanks
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Try Ekiga,
It works as and is stabil as well.
Wednesday, June 21, 2006, 1:53:33 PM, you wrote:
> I had problems with sjphone ... same version as yours.
> Finally, i managed to solve it by:
> - in sjphone, media channels settings: untick "Use remote codec
> preferences" and "Open audio streams af
Hi!
Has anyone used this box together with Asterisk?
I have a hard time finding information about this product. I have no manual and Telseys support does not answer any e-mails and you cannot download them on their homepage.
If anyone one has any information about configuration via tftp I wo
Hello -I am working on creating an Asterisk call center queueing application which will be an addition to our hosted services product.I currently have the functionality I need developed within Asterisk, but am falling short on finding a solution to provide the customer with a user interface into th
Sorry to be contrarian..but I bought the MV900 (from you actually) and I'm not really impressed with it. Prior to it I had been using the Phoenix Audio Duet which is a vastly superior device, albeit lacking the bluetooth capability.
As soon as the new Polycom Communicator is available I expec
Hallo Cesc
Cesc writes:
> I had problems with sjphone ... same version as yours.
> Finally, i managed to solve it by:
> - in sjphone, media channels settings: untick "Use remote codec
> preferences" and "Open audio streams after remote opened" ... it was
> trial-error ... now it works (to Echo
Just saw this message back to me...I didn't realize my message was
posted and I ended up posting again. Oops!
Anyway, she *is* able to receive calls. She gets a fast busy when trying
to dial anything.
I know we had her do speed tests on her DSL the end of last year but I
don't remember the outcom
> From: Rob Thomas
>
> That's freePBX or AMP code that we've since fixed - The
> replacement line is
>
> exten =>
> s,2,GotoIf($["${CALLERID(name):0:${LEN(${RGPREFIX})}}" !=
> "${RGPREFIX}"]?4:3) ; check for old prefix
Yes, ok. I'm gradually fixing all the code using Asterisk 1.2 syntax.
> (
Try these settings in zapata.conf:
echocancel=64
echotraining=800
echocancelwhenbridged=yes
rxgain=3.2
txgain=-3.2
with default KB1 echo canceller in zconfig.h
This setup was working fairly okay for me for about a year or so.
Also, notice that at the first seconds of the call you may hear some
> From: Rob Thomas
>
> Looks like you've stopped compiling libpri. All those options
> that are being ignored, are being ignored because they're for
> PRI, and you don't have PRI support in zaptel.
Uh?
If I don't have PRI support in zaptel, how are my 80 employees calling their
homes now?!
:-)
Close but not quite. Try below:
1. Setup sip.conf in the remote server to direct the call to the correct context
[incoming]
host=(xxx.yyy.zzz.xxx) IP of the sending servertype=friend context=(context that is holding the exten for the user) allow=ulaw
2. Setup extensions.conf o
On 15:12, Wed 21 Jun 06, Tzafrir Cohen wrote:
> On Wed, Jun 21, 2006 at 01:57:36PM +0200, Mimmus wrote:
> > Hi,
> > after a few of upgrades, I noticed these messages in full debug log:
> >
> > Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype
> > Jun 21 12:58:11 WARNING[27273] chan_za
I am looking to make a linux application that will use a SIP or IAX
clinet to connect to my Asterisk server and make calls.
I would like it to be written in C, but beggers can't be choosers. Any
information that would help me with my development would be appreciated.
If you know of a project tha
Hi - I have been developing an auto-dialling application similar to Voiceshot, Call-em-all etc. The one thing I am now struggling to get working is the ability to leave a message on an answering machine or cell phone voicemail. I am using app_amd.c and while it works well for some phones it is
Yeah customer feedback on these types of devices
ranges from love to hate, it boils down to individual user preference.
>From what I have seen, the Skype API is pretty tightly built into the
Communicator, I am not certain if it will be able to be used with any softphone,
that remains to be
Look in the sip.conf (or whatever) and make sure the "context" specifies
a context that allows outgoing calls.
Leah Newmark wrote:
Just saw this message back to me...I didn't realize my message was
posted and I ended up posting again. Oops!
Anyway, she *is* able to receive calls. She gets a fa
I am assuming you are talking about the Skype handsets here... Which
soft phones do these work with? Any linux ones?
W
Michael Graves wrote:
Voip Supply
has a number of USB handsets available... see
http://www.voipsupply.com/index.php?cPath=95_258 I'm not affiliated
with them beyond being a
Yes I want only copy the asterisk database, and I want that the authentificated sip registrations, work on the second asteriskAtHome like in the first,if I make this;my problem is:I want use softphones or wi_fi cell in 2 different asteriskAtHome, becouse I can't register sip on more asteriskAtHome,
Anyone can help with
this?
cli.c:49:30:
asterisk/version.h: No such file or directorycli.c: In function
`handle_version':cli.c:414: error: `ASTERISK_VERSION' undeclared (first use
in this function)cli.c:414: error: (Each undeclared identifier is reported
only oncecli.c:414: error: for eac
On Jun 21, 2006, at 5:19 AM, klubarpop wrote:
Are there any traditional telephone set-looking handset for use with a
softphone? All the options I've found are the headset type. I'm
looking
for something more traditional--it should look like a small deskset,
cellphone or cordless phone, perh
I like the TC400P card, how many T1s will that take? or is it just a
Daughter card on the TE4xx ? How many channels can it transcode?
On 6/20/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
Hi,
I just read a pressrelease from VON that Digium will soon be releaseing
a couple of new cards. What
nobody uses avaya phones with asterisk?
On 6/20/06, Erick Perez <[EMAIL PROTECTED]> wrote:
Hi, I setup my tftp to send SIP configurations (the bin files) to the
avaya phone. When it finished loading and rebooting it asked for the
extension and the password and the asterisk ip address. I had to i
I am looking to make a linux application that will use a SIP or IAX
clinet to connect to my Asterisk server and make calls.
I would like it to be written in C, but beggers can't be choosers. Any
information that would help me with my development would be appreciated.
If you know of a project tha
Does a solution exist that I am overlooking that may provide the
functionality I am after?
I don't understand why Queue-Metrics will not do what you need? We
run it and it does everything you just said you wanted to do.
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Anyone can help with
this?
cli.c:49:30:
asterisk/version.h: No such file or directorycli.c: In function
`handle_version':cli.c:414: error: `ASTERISK_VERSION' undeclared (first use
in this function)cli.c:414: error: (Each undeclared identifier is reported
only oncecli.c:414: error: for ea
I’m stumped on this one and any help would be greatly
appreciated.
I’m just trying to get my Polycom 601 to have multiple
extensions on it. For example, on line 1 I want extension 21, on line 2 I
want extension 22, and on line 3 I want extension 23. Ideally I’d
actuall
> Did you see this after a reload?
> Asterisk will ignore some settings when doing a reload.
> Only a restart will pickup changes to the settings mentioned
> in your mail.
True. In fact, after a restart, I don't see any WARNING.
Thanks
DV
___
--Bandwi
In reading this problem and the solution I wonder if it would be better,
(or acceptable to the users?) to send the received Hylafax fax to a
(network attached) printer of their choice. That way you eliminate the
need for special equipment (the fax machine) ?
I did a bit of research and found
Hello Steve,
Thank you for being so active helping people with Unicall problems,
I am sure a lot of us appreciate this.
I could tweak a little bit more with your software versions from the
download site and I got half of the problems solved. Now I am able to
set the loglevel to 255 and tha
Not sure why you were told the Clarisys i750H is
not shipping, we've been working with these for quite some time, and have seen
good availability on them.
Thanks
Cory J AndrewsVOIPSupply.com454 Sonwil
DriveBuffalo, NY 14225++voice - 716.630.1555
X22email - [EMAIL
With few exceptions a USB phone is just and audio device to the host PC. Most
will work with any soft phone. The Phoenix Duet and MV900 that I have used both
worked equally well with
Asterisk, Sip Phone/Gizmo, FWD, Firefly and Skype.
There are some "Skype Certified" hardware devices appearing.
I am trying to configure Asterisk to work with my packet8
subscription. And after sniffing the traffic between my ATA and
Packet8, I have noticed that when I call a PSTN line, the ATA issues
an SIP INVITE with TEL in the To URI. For example tel:NXX;phone-
context=+1NXX.
Can Asterisk
Also check the dialplan on the ATA as well. Maybe its the way she is
dialing the number that is causing the issue.
On 6/21/06, Leah Newmark <[EMAIL PROTECTED]> wrote:
Just saw this message back to me...I didn't realize my message was
posted and I ended up posting again. Oops!
Anyway, she *is*
Hi List, I have the following in my extensions.conf. For some reason if the user enters a room that does not exist instead of going to the next pri. it just says room invalid and dumps the call. Can it be a bug ? Exten => _*5XXX,1,MeetMe(${EXTEN:1},D) Exten => _5XXX,1,MeetMe(${EXTEN},cMrpsq)
Al,
Are you doing voice broadcasting –
that is, delivering a pre-recorded message, possibly giving a live caller other
options? Just curious. I’ve been working on a
voice-broadcasting application myself and I’ve had mixed success with
app_amd.c. It does work very well in some cases,
I have multiple (20+) Snom 360 phones communicating with asterisk
1.2.7.1. Almost regularly (daily) and in some cases ongoing 9every 10
minutes the phones ask for password and id the account they are seeking
the password for. If I hit the X key the phone continues operating
normally. Has anyone
I'm getting some of these errors:
ERROR[7244] chan_sip.c: Got SUBSCRIBE for extensions without hint. Please
add hint to 464 in context from-internal
All these extensions are not local (sip) but other analog phones attached to
a legacy PBX downstream.
Any idea?
Thanks
--
Domenico Viggiani
you can either use the call_limit for each internal or if you wish
something centralized ( for example a maximum total of 30 concurrent
calls ) you can use the superdial macro
http://www.voip-info.org/wiki/view/Superdial+macro
Patrick wrote:
On Tue, 2006-06-20 at 09:20 +0200, bram kortleven
Hello--
It's been a while since I wrote any updates about AEL/AEL2 to the users
list, and I thought it might be worthwhile to update everyone on what is
going on in respects to AEL.
What the heck is AEL? The Asterisk Extension Language. A higher level
language for extensions.conf, which will appe
I was testing using trxtel for outbound toll free because I have an issue on my
PRI where it will not handle early media. (IVRs that
play as a ringback tone)
There was a bug that was supposed to fix this Q4 of 2005, but I never saw any
relief for it.
voip.trxtel.com has the same issue, so at le
That's not the problem. The contexts are all fine, and the problem fixes
itself when it feels like it. I am almost positive it's her connection.
The asterisk coding on my end is fine. She has the same setup as the
other 20 employees and they all work fine.
She was running tests on dsltools.com. Is
Anybody else able to help...?
On 6/19/06, John Klimek <[EMAIL PROTECTED]> wrote:
Ahh, good catch. I've changed the context to be "incoming-bv" (to
match my context in extensions.conf), but I still get the same exact
phone message...
Also, is it normal to see "REGISTER attempt 1 to
[EMAIL PROTE
we're also seeing a similar problem with 1.2.9.1 (previously 1.2.7.1,
without this error). our manager interface is used a fair amount from
FOP, and reloads occur staggered throughout the day on changes (but not
super often).
when the problem occurs, we see a lot of agent and queue function
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