On Wed, Aug 08, 2007 at 10:28:05AM +0600, Kate Kretz wrote:
sorry, I meant RFC 3856, sip presence, not sip regitration
Twinkle 1.1 (new in that version. Released only about a month ago), linphone.
Kphone should also support it, but I so far failed to get it
authenticated with my Asterisk server.
Hi,
I am running asterisk PBX ( digium TE120P card configured) on one
system. It is connected to E1 card running application on the other system.
After establishing sync between two card, I am able to place call from sip
phone to E1 card running application. I want to pass the callerid, when
2007/8/7, mitcheloc [EMAIL PROTECTED]:
Ollvier,
You could use the Firefox plug-in for Snap. It will auto detect
numbers on a webpage and make them dialable.
Cheers,
Mitchel
I'm waiting for Snap internationalized version to use it again ;-)))
Anyway, auto detect implies some pattern
2007/8/7, Dean Collins [EMAIL PROTECTED]:
Mitchel, he's not looking for a click to dial solution - he wants to
implement some form of click on his website so people can call him.
At the end of the day most people aren't going to have it configured
correctly etc and you should really use web
Hello,
My goal is to provision C450IP or S450IP models.
Has anyone a hint to provision them from configuration files ?
Usually, we use dedicated menu embedded inside Gigaset handset.
An http server also exists but I couldn't find any dhcp-tftp combination to
configure them.
Any clue ?
Regards
Flash Operator Panel would do it.
Also the Aastra 55i phones with the expansion module, which has 36 lines on
it should work, but you will need to cofigure your Asterisk for Shared Line
Appearances (also called Bridged Line Appearance) for the Busy Lamp Field
(BLF) to work. The Aastra 55i would
You can set the caller-id in many different ways but the easiest in by
setting it in the sip.conf profile for the extension.
So you can just add a line like this to your sip.conf under the extension:
callerid=Your Name 5554441212
Hope this helps..
Regards,
Todd R.
--
Prestige Messaging
Live
When I tried it, when a user login at a phone, it replaced any
previously logged one.
hope that help
Implant them with RFIDs.
Thanks,
Steve
Tattoos and barcode scanners.
PaulH
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Zoiper is using the callto: tag (see
http://www.zoiper.com/downloads/Zoiper_2.0_Biz_Manual.pdf page 46.) It
also works without the extension= in there. (I will update the
information today to show all the ways we support)
the SIP wouldnt work if the user wants to use IAX instead.
Am Dienstag, den 07.08.2007, 07:47 +0200 schrieb Olivier:
So no proper logoff between logins, right ?
As I will apply free sitting in school environment, chances are phones
would then remain logged-in several hours or days between another user
logs in.
My thoughts are focused on finding
Price. They are good cards, just bells and whistles plus the Echo
cancellation on the a101d. Ask Sangoma, their must have a reason for
still selling them ;-)
Gavin.
On 07/08/07, Rory Campbell-Lange [EMAIL PROTECTED] wrote:
Hi Gavin
Many thanks for the note. For what reason do you recommend
Hi,
The case I'm working on is :
- a call comes from PSTN to a given extension (say 122)
- a user picks the call up (dialing *8122) from another extension (say 240)
using a SIP hardphone
- the hardphone (he one with 240 extension) displays the dialed string (here
*8122) instead of original
i have problem with pass-through faxing
with this scenario
hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen
virtual) - linksys ATA
i can fax to fax2mail on hylafax
but after upgrade asterisk2 to 1.4 faxing is not working
hylafax(iaxmodem) - iax - asterisk1 1.2.22 -
Hi all,
I am seeing on my logs this message:
Jun 13 09:14:51 DEBUG[4944] chan_zap.c: Monitor doohicky got event Event 160
on channel 3
Jun 13 09:14:51 DEBUG[4944] chan_zap.c: Monitor doohicky got event Event 160
on channel 3
(repeated much more then what I will show here).
I see that it
Dear all
I have asterisk setup now what happend when i dial 4 digit
number my asterisk wait for few digit why when i press # key it is dialing fast
but without # wait for few number is there any configuration for dialplan
I have setup asterisk with avaya system
Hi,
is anyone on the list using the Siemens Openstage phones together with
asterisk?
If yes, is it possible to use the programmable keys of these phones
together with Asterisk?
Thanks for any hints,
Stefan
--
in-put GbR - Das Linux-Systemhaus
Hi Drew. Thanks for the tips. My Line 1 works as I'd like it to, and I could be
wrong, but I don't think changing the dialplan there will help. I really just
want to be able to dial local phone calls (7 digits) and have it go out the
SPA3102, without having to dial twice. This is a snip what I
Hi to all,
I'm using asterisk 1.4.9 with chan_h323.
When someone in the H323-VoIP cloud dial 1234 this number is assigned to my
asterisk-machine, so the VoiceGW forward the flow to my machine, asterisk
though the dialplan can delivery the call to a particular SIP phone...this
is ok...
I can also
I believe X-Lite v3 (and EyeBeam) from Counterpath both support 3856 as
presence user agents.
N.
Kate Kretz wrote:
sorry, I meant RFC 3856, sip presence, not sip regitration
On 8/7/07, *Tzafrir Cohen* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
On Tue, Aug 07, 2007 at
Dear all
i need this feature in asterisk whn 2 party calling that time
i pickup call and listen conversation of that party spoofing like is it
possible in asterisk
Rgds
satish patel
-
Choose the right car based on your needs. Check
I'm confused is it for a single installation? Why bother messing around
- just install a softphone and set it up right.
If it's for a deployment to multiple sites is the web app commercial? If
so then buy a,license for one of the java softphone solutions - there's
a few free and non free
On Wednesday August 08 2007 8:28 am, Tim Johnson wrote:
Hi Drew. Thanks for the tips. My Line 1 works as I'd like it to, and I
could be wrong, but I don't think changing the dialplan there will help. I
really just want to be able to dial local phone calls (7 digits) and have
it go out the
Apologies if this is a resend, but I've sent this 12 hours ago and still
can't see it on the list.
Hi,
I've just started to setup my phones with Buddy watch. Basically, it all
works fine when using the simple example on the wiki:
exten = 123,hint, SIP/some_sip_reg
exten =
Hi Sanchal,
115 in your case is just DIALLED NUMBER and it will be searched by you E1
trunk.
If you want change your CALLERID, you would insert one default or would
insert one to each user.
the command is the same sendt by Todd:
callerid=Your Name 5554441212
but you can work with function
On 8/7/07, raviprakash sunkara [EMAIL PROTECTED] wrote:
Hello Russell,
Nice To meet U and Good Morning. I got u r mail-Id from
http://www.asterisk.org/node/48325
Recently i started the SLA configuration. But i didn't understand the
Flow of its Functionality
One of the My Client Ask to
asterisk*CLI show channels
Channel Location State Application(Data)
Zap/3-1 (None) Up Bridged Call(Zap/47-1)
Zap/47-1 [EMAIL PROTECTED] Up Dial(ZAP/g1/2105||TWK)
Zap/25-1 (None) Up Bridged
satish patel said
I have asterisk setup now what happend
when i dial 4 digit number my asterisk wait for few digit why
when i press # key it is dialing fast but without # wait for
few number is there any configuration for dialplan
This part of the dial plan looks
On 8/8/07, arkda [EMAIL PROTECTED] wrote:
I've been digging around and I haven't found a way to do this, but I have
a feeling I'll feel like an idiot because it's something I'm over looking.
Normally if I need to specify an additional option (such as different
language sound files) or I'm
i have only one single 16XX dialplan for reached to avaya system then why i
have to wait for more digit
satish patel
Don Pobanz [EMAIL PROTECTED] wrote: satish patel said
I have asterisk setup now what happend
when i dial 4 digit number my asterisk wait for few digit
This is part f the phones dial plan. Our aastra phones do the same
thing. Most phones allow you to configure the dial plan on them.
satish patel wrote:
i have only one single 16XX dialplan for reached to avaya system then
why i have to wait for more digit
satish patel
*/Don Pobanz
Hi,
I don't have this answer but would be curious to know its price for
reseller.
Any clue ?
Regards
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Is it normal for a PRI to reset the inactive B channels periodically(like once
every hour). I'm seeing on my asterisk console successful restarts, just
curious as this is all new to me.
This e-mail, facsimile, or letter and any files or attachments transmitted
Patrick wrote:
Hi all,
Anyone have an idea which version of spandsp, libunicall, libmfcr2,
libsupertone, app_rxfax/app_txfax and chan_unicall I should use for the
latest asterisk 1.2?
Would that be the ones listed below?
http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre4.tgz
On Wed, 2007-08-08 at 09:29 -0500, Jeremy Mann wrote:
Is it normal for a PRI to reset the inactive B channels
periodically(like once every hour). I’m seeing on my asterisk console
successful restarts, just curious as this is all new to me.
Yes, that's normal. You can disable it (as I usually
Absolutely normal, yes.
-Darren
--
Darren Nickerson
Senior Sales Support Engineer
Telephony Depot
www.telephonydepot.com
+1.215.825.8710 ext 8106 (office)
+1.215.243.8335 (fax)
- Original Message -
From: Jeremy Mann
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi,
Is it possible to write a function in Asterisk, that returns a value? Sort
of like any programming language allows?
For example, I`d like function ReturnSipReg to return the right
SipRegistration to dial, based on some value so that I could use it in my
dial plan:
i.e:
exten =
Hi,
When Asterisk receives SIP INVITE packets, it tries to match the packet
to a section on sip.conf, so that it can know what context of the
dialplan should be used, what codec's are allowed, etc. (what else does
it do here?)
I would like to know what is exactly the order for this matching
On Wednesday 08 August 2007 11:24:45 am Mike wrote:
Is it possible to write a function in Asterisk, that returns a value? Sort
of like any programming language allows?
Digium has taken the stance that it's better to set arbitrary variable names
to arbitrary values rather than allow what many
But what if I wanted to write my own custom application for one specific
purpose, I can't set a return value? It's not possible at all?
Let me put it this way then, if I needed to have some processing all done in
the same Asterisk priority (in my case, I want to use the hint priority
but I need
Dean,
It's for tens of single user : a couple of users at a time on as many
locations I can get !
I've got a contact with an ISV with sells directories (with coporate
charting capabilities).
Today, its software is mainly used to edit and display charts and
directories.
In directory use, it
You can add the header the vendor is suggesting in asterisk as follows;
exten = #,1,SipAddHeader(P-Asserted-Identity:
sip:${CALLERID(num)[EMAIL PROTECTED])
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: Wednesday, August
Hello,
I installed Asterisk on Dell Precision workstation and configured with
sample configuration.
I have two TDM400 board and one with 4xFXO and second one 4xFXS module
installed.
I made call to telephone line connected to FXO port and never seen callerid
on those lines.
I tested
Mike wrote:
Hi,
Is it possible to write a function in Asterisk, that returns a value?
Sort of like any programming language allows?
For example, I`d like function ReturnSipReg to return the right
SipRegistration to dial, based on some value so that I could use it in
my dial plan:
I can be a bit slow sometimes, but you said that it's not possible, and on
the other hand told me to write my own function (which appears to contradict
the first statement).
Your example of the use of a function is exactly what I need (Create a
function and Dial(SIP/${MY_FUNKY_NEW_FUNC(ooga)})) ,
2007/8/8, Steve Underwood [EMAIL PROTECTED]:
Patrick wrote:
Hi all,
Anyone have an idea which version of spandsp, libunicall, libmfcr2,
libsupertone, app_rxfax/app_txfax and chan_unicall I should use for the
latest asterisk 1.2?
Would that be the ones listed below?
On Wednesday 08 August 2007 11:41:38 am Mike wrote:
But what if I wanted to write my own custom application for one specific
purpose, I can't set a return value? It's not possible at all?
Not possible, to my knowledge.
Let me put it this way then, if I needed to have some processing all done
On Wednesday August 08 2007 12:10 pm, Mike wrote:
I can be a bit slow sometimes, but you said that it's not possible, and on
the other hand told me to write my own function (which appears to
contradict the first statement).
Your example of the use of a function is exactly what I need (Create
Asterisk Users,
Has anybody use Voicepulse Connect for Asterisk?
I am trying to cover all my bases because in the past, I got burned with
poor quality of service, along with failed DTMF tones with 3 different SIP
Providers (Vitelity, Broadvoice, and Teliax).
I am running Asterisk 1.2.13
On 8/8/07, Mike [EMAIL PROTECTED] wrote:
I'd be most thankful for some link to a page that shows how to write such
a
function in Asterisk.
There is a test application in the source tree (not built by default I
believe), but it doesn't look like anyone has made an equivalent sample
function.
On Wednesday 08 August 2007 12:10:47 pm Mike wrote:
I can be a bit slow sometimes, but you said that it's not possible, and on
the other hand told me to write my own function (which appears to
contradict the first statement).
That's because I'm a little slow today... I thought you were asking
You are looking for the AGI: http://www.voip-info.org/wiki-Asterisk+AGI
Anthony
James FitzGibbon wrote:
On 8/8/07, *Mike* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
I'd be most thankful for some link to a page that shows how to
write such a
function in Asterisk.
There is
Olivier,
I think you are getting confused. Call me on 212-203-4357 and I'll
answer your questions but basically I think you are doing this the wrong
way.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
John,
Voicepulse Connect has been great to me. I've been using it for over a year
now, and do not have any major complaints, except that there are no
printable receipts for credit card transactions. SIP is also the preferable
protocol, as IAX seems to have some issues. Customer service is usually
Wes,
What kind of service outages did you experienced?
This would use for my office and I cannot afford for any dropped calls or
poor audio quality, when talking to customers.
-John
From: Wes Baehr [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
On Wed, Aug 08, 2007 at 11:34:49AM -0400, Andrew Kohlsmith wrote:
On Wednesday 08 August 2007 11:24:45 am Mike wrote:
Is it possible to write a function in Asterisk, that returns a value? Sort
of like any programming language allows?
Digium has taken the stance that it's better to set
On Wednesday 08 August 2007 1:17:24 pm Jay R. Ashworth wrote:
Digium has taken the stance that it's better to set arbitrary variable
names to arbitrary values rather than allow what many would consider the
perfectly accepted method of using a $? type of return code in addition
to any
On Wed, 2007-08-08 at 17:08 +, John Meksavan wrote:
Wes,
What kind of service outages did you experienced?
This would use for my office and I cannot afford for any dropped calls or
poor audio quality, when talking to customers.
My experience with Voicepulse has been good
I'd like to be able to generate a Manager Event from the dialplan but
can't seem to find a way to do it.
Alternatively, trigger and Event when a record in AstDB gets changed.
Can anyone point me in the right direction? Thanks.
By way of explanation, I've a app that connects to astmanproxy
Thanks for all the replies, after some thinking AGI seems like the way to go
(writing a function in C would certainly work, but I want to avoid anything
that makes upgrading to newer version of Asterisk a potential pain. Let's
say using C is plan B).
So, I wrote (well, plagarized directly from
On Wed, 2007-08-08 at 23:55 +0900, Balgansuren Batsukh wrote:
Hello,
I installed Asterisk on Dell Precision workstation and configured with
sample configuration.
I have two TDM400 board and one with 4xFXO and second one 4xFXS module
installed.
I made call to telephone line connected
Have you checked out UserEvent:
http://www.voip-info.org/wiki/view/Asterisk+cmd+UserEvent
Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
-Original Message-
From: [EMAIL PROTECTED]
I have a queue setup to 'roundrobin' (NOT roundrobin with memory). I
have three agents. We'll call them 101, 102, and 103.
When a call comes in.. I want it to always try 101 if no answer try
102.. if no answer try 103, etc.
However, what it is doing is... it will ring 101... if 101 answers,
On Wednesday 08 August 2007 1:39:34 pm Mike wrote:
exten = 12345,1,AGI(agi-helloworld.agi)
AGI is an application, and you've called it.
exten = 12345,1,Noop(${AGI(agi-helloworld.agi)})
AGI is not a function. You cannot nest applications like that. The NoOp
application cannot call another
Asterisk 1.2.13 - Evolution PBX from Intuitive Voice Technologies
We have an installation of 35 SIP phones (Polycom 501) and
one receptionist phone (Polycom 601). I have 15 of the 501s
set up to accept a Page. From what I understand, the Page
is done using the asterisk page application that
On 8/8/07, Mike [EMAIL PROTECTED] wrote:
So, I wrote (well, plagarized directly from the Web) a simple Perl program
that prints Hello World. I call it using this:
exten = 12345,1,AGI(agi-helloworld.agi)
Seems to work (I'm not expecting anything, really, just no Asterisk
error).
When I
SLA is not BLF.
The only thing you need to configure to have BLF is adding hint priority to
your dial plan.
On 8/8/07, James Collier [EMAIL PROTECTED] wrote:
Flash Operator Panel would do it.
Also the Aastra 55i phones with the expansion module, which has 36 lines
on
it should work, but
Hi Olivier,
Hi,
I don't have this answer but would be curious to know its price for
reseller.
Any clue ?
no, I'm sorry. We're only responsible for the configuration of the
devices. Our client will buy all the necessary hardware. I will ask him
about the prices, but these will be end
Clarify this, what you are trying to achieve?
To see if handsets are being used or not?
Or to see if any trunk is being used or not and share it?
These are 2 different concepts, first is BLF you can have your asterisk to
provide that information with hint priority, and the second one is SLA.
On
AH! Thanks, I've been thinking that apps and functions were interchangeable,
hoping that I could return values with functions. Now that this is very
clear in my mind (at least I think it is) I'll go and write a function.
Might as well ask this before I go out, not find my answer and come back
Ideally you specify the ftp server via DHCP. Then you don't have to
touch the phone. If your files are right on the server, you just plug in
the phone and go. That's how we do it and it's a breeze.
Scott
Doug wrote:
At 21:02 8/1/2007, Doug, wrote:
At 16:49 8/1/2007, Douglas Garstang wrote:
Hmm beginning of the end of free trixbox by the sounds of it.
It was good while it lasted but time to download the latest iso while
it's still available by the sounds of it.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
I'm using Page application with Polycom 501 and 601 and have not seen these
issue,
i would check firmware on 601 and play with couple different firmware.
are you checking if the chanavail before sending the Page?
On 8/8/07, Bill Andersen [EMAIL PROTECTED] wrote:
Asterisk 1.2.13 - Evolution
Hello,
I am from Mongolia and when I use telephone set with callerid it display
callerid.
Yes, our phone company charge for callerid service.
Balgaa
- Original Message -
From: Steve Murphy [EMAIL PROTECTED]
To: Balgansuren Batsukh [EMAIL PROTECTED]; Asterisk Users Mailing List -
At 10:17 AM 8/8/2007, you wrote:
Digium has taken the stance that Structured Programming is a Bad Idea?
While it seems that way in hindsight, I'd guess that no thought was
put into dialplan programming when it was started and by the time
someone realized it was wrong, the person in charge said,
Just got mail from them saying my NY DID will be deactivated in few days .
Funny thing is their site is still showing orderable DID's of same area
code . Anybody else got this ?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
Bill Andersen wrote:
Asterisk 1.2.13 - Evolution PBX from Intuitive Voice Technologies
We have an installation of 35 SIP phones (Polycom 501) and
one receptionist phone (Polycom 601). I have 15 of the 501s
set up to accept a Page. From what I understand, the Page
is done using the
Mail list wrote:
Just got mail from them saying my NY DID will be deactivated in few days
. Funny thing is their site is still showing orderable DID's of same
area code . Anybody else got this ?
Wow. That is totally unacceptable.
Are they going to give you the option of porting the DID?
On Wed, Aug 08, 2007 at 01:34:56PM -0400, Andrew Kohlsmith wrote:
On Wednesday 08 August 2007 1:17:24 pm Jay R. Ashworth wrote:
Digium has taken the stance that it's better to set arbitrary variable
names to arbitrary values rather than allow what many would consider the
perfectly
Yes they are co-operating to port DID to another provider and they have
given time till august 23 so DID will continue to work till then but they
are not providing any substitute DID though ( i dont expect that ) but
atleast they should partially refund amount for remaining days ( i dont
expect
This should be configured in phone system instead of asterisk :) .
On 08/08/2007, Michael Rice [EMAIL PROTECTED] wrote:
This is part f the phones dial plan. Our aastra phones do the same
thing. Most phones allow you to configure the dial plan on them.
satish patel wrote:
i have only one
Quoting Mail list [EMAIL PROTECTED]:
In general how painful has this sort of thing been to people so far ?
I am pretty hesitant to put any sort of number like that on
letterhead, website etc., when there might be doubt about having it
long term when its provided by a small company. It seems
Why not use
1-Ruby RAGI
2-http://adhearsion.com/
or similar tools which overcome Asterisk dial plan limitations?
-E
On 8/8/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Wednesday 08 August 2007 1:39:34 pm Mike wrote:
exten = 12345,1,AGI(agi-helloworld.agi)
AGI is an application, and
One of the things that we've done is get a standard PSTN line in place that
rings down to the VoIP lines. In smaller shops there's a single copper line,
in larger shops they might have a T1/PRI. It's obviously more expensive than
pure VoIP lines, but the stability of the number is solid; You know
Quoting Alex Robar [EMAIL PROTECTED]:
surely you wouldn't do this where you are getting voip numbers so you
can have local numbers in other areas. Having an analog or other
rung through like that would be impossible in most cases and hugely
expensive where actually possible.
One of
If you cannot afford any dropped calls or poor audio quality, you need a PRI
or POTS connection. It doesn't matter how great the carrier is, the Internet
is an unreliable medium.
2-3 times VoicePulse has had issues with incomings calls ringing busy. Once
incoming calls were all garbled on my end,
Jon,
No, not at all - Sorry, that's not what I meant. Indeed, a local extension
would be quite prohibitively expensive.
What we tend to do with people who require out-of-area calling ability is
grab a toll free DID from a bit of a bigger or more stable provider. Here in
Ontario, Canada, we've
Quoting Alex Robar [EMAIL PROTECTED]:
Jon,
No, not at all - Sorry, that's not what I meant. Indeed, a local extension
would be quite prohibitively expensive.
What we tend to do with people who require out-of-area calling ability is
grab a toll free DID from a bit of a bigger or more stable
Matt wrote:
I have a queue setup to 'roundrobin' (NOT roundrobin with memory). I
have three agents. We'll call them 101, 102, and 103.
When a call comes in.. I want it to always try 101 if no answer try
102.. if no answer try 103, etc.
However, what it is doing is... it will ring 101...
Mike wrote:
Thanks for all the replies, after some thinking AGI seems like the way
to go (writing a function in C would certainly work, but I want to
avoid anything that makes upgrading to newer version of Asterisk a
potential pain. Let's say using C is plan B).
So, I wrote (well,
What we tend to do with people who require out-of-area calling ability is
grab a toll free DID from a bit of a bigger or more stable provider. Here in
Ontario, Canada, we've had great success with Unlimitel for providing toll
free DIDs.
I have run across that name before as well - anyone
On Thu, 2007-08-09 at 01:55 +0900, Balgansuren Batsukh wrote:
Hello,
I am from Mongolia and when I use telephone set with callerid it display
callerid.
Yes, our phone company charge for callerid service.
Balgaa
Balgaa--
If you have tried all combinations of cidsignalling/cidstart,
Wes, I'm working through some issues with IAX and Voicepulse right now.
It was regarding dropped inbound calls. I was able to put my server
into a different location though, and so far the issues have disappeared
so hopefully it was a network problem somewhere between us.Just
curious
Folks, I have somewhat of a serious issue here. My music on hold
mysteriously stopped working. I have made no changes to my Asterisk box
in the past month and up until an hour ago, MoH was working fine (as far
as I know).
CLI:
-- Started music on hold, class 'default', on channel
That is why you need to start posting info about the providers at
http://www.bochterservices.com/phpbb/
so everyone knows
This is a FREE SERVICE provided by Bochter Services and it is not going
away any time soon.
There will be more added by your request
Best regards,
Al Bochter
I've had MOH die probably 4-5 times in the last 2+ years and the only
way to get it back is to stop * and restart it. Reloading MOH or just
doing a regular reload doesn't work. I have to actually do a stop now
and then asterisk to get it to work again. * restarts and MOH works
fine. No
Hi,
Here is my first step (call it a proof of concept) in using the hint
priority with dynamic values.
Background - this works
exten = 12345,hint,SIP/12345-1
To make this a little dynamic, I used a web page to return to me the value
of the sip registration. In other words,
I had a lot of problems with garbled IAX calls (even when calling into just
the IVR). The problem was compacted when I would bridge an incoming IAX call
to an outgoing SIP call, though that may be a fault of Asterisk. Since using
SIP everything has been working perfectly. I never had any real
Hi all,
Is there a way to have this command to record a mixed file instead of
one for in and one for out? I have set the Mix parameter to 1, but it is
still generating two files. I would prefer it to have the in and out
files mixed. Thnx.
___
Peder,
Unfortunately, this did not work. Any other thoughts?
Jay
Peder @ NetworkOblivion wrote:
I've had MOH die probably 4-5 times in the last 2+ years and the only
way to get it back is to stop * and restart it. Reloading MOH or just
doing a regular reload doesn't work. I have to
Wes Baehr wrote:
I had a lot of problems with garbled IAX calls (even when calling into
just the IVR). The problem was compacted when I would bridge an incoming
IAX call to an outgoing SIP call, though that may be a fault of
Asterisk. Since using SIP everything has been working perfectly. I
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