On Wed, Aug 18, 2010 at 3:59 PM, Geraint Lee gera...@gmail.com wrote:
This is what I ended up doing, working fine now.
Cheers
On 18 August 2010 08:52, Nasir Iqbal na...@ictinnovations.com wrote:
Avoid to use MySQL dialplan application, instead write an AGI script for
this purpose
LOL, I
Hi list
I have a requirement that I just don't know how to address - I don't think
its strange but can't find any pointers anywhere.
I have a user that wishes to have a multi phone divert. By that I mean
calls made to his extension say Ext200 can be redirected to a different
extension say
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paddy Grice
Sent: 19 August 2010 08:21
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Calling Line Identity - any ideas
Hi list
I have a requirement that I just don't
Converting with sox works well as followed :
sox -V intro.wav -c 1 -r 8000 intro2.wav
To convert with asterisk convert, I needed to use an absolute path :
asterisk -rx file convert /var/lib/asterisk/moh/folder/intro.wav
/var/lib/asterisk/folder/intro.alaw
All works well.
Jonas.
On
I'd have to say off the top of my head that this should already work
as long as the trunk you're sending calls to the outside world over
has the CallerID setting set AND probably sendrpid=yes...in the sip
configuration for both of those items...past that, I could dig a
bit
Cheers,
Sherwood
Are your calls coming in over IAX, SIP, or DAHDI/ZAP? this will make a
big difference in the way I'd go about trying to figure out how to
resolve the issue. If it's SIP, try doing a network capture of all SIP
and RTP traffic in and out of the server and then try to replicate the
problem. also,
Hi Sherwood
Maybe my miss-understanding sip.conf I will try and see what happens - but I
don't understand how sip would know which CLID to send to each sip
endpoint - internal or external.
BTW this is all to get return of missed calls working.
I don't know if this makes it any clearer -
An
Hi,
Does anyone has an idea how to tell asterisk to use codec A for first 50
calls and then codec B for rest of the calls.
Thanks,
Deepika
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New
I'll see if I can make it a little clearer for you...
When ext 123 makes a call to another SIP device through the server but
not requiring a middleman (i.e. a third party provider that allows you
to dial to parties outside the immediate control of your PBX) to
accomplish the call, what happens is
On Thu, Aug 19, 2010 at 3:14 AM, Deepika Nijhawan
deepika.nijha...@oxygen8.com wrote:
Hi,
Does anyone has an idea how to tell asterisk to use codec A for first 50
calls and then codec B for rest of the calls.
Thanks,
Deepika
--
On 19 Aug 2010, at 09:14, Deepika Nijhawan wrote:
Does anyone has an idea how to tell asterisk to use codec A for first 50
calls and then codec B for rest of the calls.
You could create two separate trunks, one for each codec?
S
--
On Thu, Aug 19, 2010 at 3:37 AM, Steve Howes steve-li...@geekinter.net wrote:
On 19 Aug 2010, at 09:14, Deepika Nijhawan wrote:
Does anyone has an idea how to tell asterisk to use codec A for first 50
calls and then codec B for rest of the calls.
You could create two separate trunks, one
Hi,
here is a trick for you!
exten = s,1,Dial(SIP/Ext400Local/${EXTEN}/home-context)
[home-context]
exten = s,1,Set(CALLERID(num)=44112233445566)
exten = s,1,Dial(SIP/TheWorld/441234567890)
Regards
On Thu, Aug 19, 2010 at 12:21 PM, Paddy Grice pa...@wizaner.com wrote:
Hi list
I have a
little syntax mistake, try this
exten = s,1,Dial(SIP/Ext400Local/${ext...@home-context)
[home-context]
exten = s,1,Set(CALLERID(num)=44112233445566)
exten = s,n,Dial(SIP/TheWorld/441234567890)
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-- Bandwidth and Colocation
On Thu, Aug 19, 2010 at 4:11 AM, Nasir Iqbal na...@ictinnovations.com wrote:
little syntax mistake, try this
exten = s,1,Dial(SIP/Ext400Local/${ext...@home-context)
[home-context]
exten = s,1,Set(CALLERID(num)=44112233445566)
exten = s,n,Dial(SIP/TheWorld/441234567890)
--
I would like to figure out why but can't really switch back now it works
since to replicate the problem... whatever it may be... i'd need to leave it
running live and wait for the live system to die... which obviously isn't
what i really want to happen :)
On 19 August 2010 08:11, Sherwood McGowan
On Thu, Aug 19, 2010 at 4:27 AM, Geraint Lee gera...@gmail.com wrote:
I would like to figure out why but can't really switch back now it works
since to replicate the problem... whatever it may be... i'd need to leave it
running live and wait for the live system to die... which obviously isn't
On 18/08/10 17:10, Gordon Henderson wrote:
... using it as a tool and understanding what it does...
So one part of it's toolset identifys valid SIP accounts - and I was under
the impression that alwaysauthreject=yes was supposed to stop this...
However, it sends a request for a highly
(I've just had 30GB of sipvicious traffic sent to my hosted servers in a
12-hour period - it came from what looked like a VPS host in France -
trivially firewalled out, but even dropping the packets didn't stop the
flood! It's so badly written it appears to just ignore any return codes
that
i do this by having 2 peers setup, one has a call limit of 10 and uses g729,
the rest of the calls get sent to the second peer which uses ulaw.
all calls attempt peer 1 if there's channels available it uses it if not it
just moves through the dialplan to use the second one.
On 19 August 2010
On Thu, 19 Aug 2010, Dana Harding wrote:
(I've just had 30GB of sipvicious traffic sent to my hosted servers in a
12-hour period - it came from what looked like a VPS host in France -
trivially firewalled out, but even dropping the packets didn't stop the
flood! It's so badly written it
Dear All,
i have a problem with 3g calling in asterisk with ISDN support .
i tried it with the help of H324M gw .
can any one tell that how i configure H324M gw .
i fallow the bellow link
http://www.voip-info.org/wiki/view/Asterisk+H324M
http://sip.fontventa.com
--
Hello,
Is there a way to capture the answering machine message when the dialer
detects the answering machine.
Thanks
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New to Asterisk? Join us for a live
Ow...
I have =no commented, so I guess =yes is default??
; Normally, CDR's are not closed out until after all extensions are finished
; executing. By enabling this option, the CDR will be ended before
executing
; the h extension so that CDR values such as end and billsec may be
; retrieved
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada
Subject: Re: [asterisk-users] CDR variables
Ow...
I have =no commented, so I guess =yes is default??
; Normally, CDR's are not closed out until after all extensions are
Ok. And how will we do for getting sip inbound calls from different ips and
sending them to dahdi.
Thanks,
D
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New to Asterisk? Join us
Hello,
Is it possible to set a variable in dialpan when the someone calls a
particular DID number so that i can use that variable for calls coming to
that number only.
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tino
Subject: [asterisk-users] setting variable for a DID number
Hello,
Is it possible to set a variable in dialpan when the someone calls a
particular DID number so that i can use
Hello list,
A calls B, B transfers to C, A speaks with C.
Anyone knows how I can record a conversation where the call is transfered ?!
The recording of the call (which begins when B answers) stops when A and
C are connected together.
Can I keep the recording going ?!
Kind regards,
Has anyone found a way to detect a loop condition in the dialplan.?? We
had a condition where this filled up 47 PRI channels in an NFAS
group connected to our media gateway... and endless loop if you will..
Thanks
--
_
--
Just a reminder: AstriCon is coming up in October in Washington, DC
(http://www.astricon.net/
) and we're looking forward to seeing you there!
We're getting to the deadline for Innovation Awards for this year.
What's an Innovation Award? The Innovation Award is designed to
recognize
Hi,
Never tried it, but you can take a look to the AUDIOHOOK_INHERIT
function that allows MixMonitor to continue the recording in the same
file after the transfer.
http://www.voip-info.org/wiki/view/Asterisk+func+AUDIOHOOK_INHERIT
Cheers,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium
But when i call my DID number following dialplans are being executed. What
i need is to set a variable with one value for one DID number and set the
same variable with another value for another DID number. Also any contexts
should be able to use this variable.
-
NoOp(SIP/5070-5407,
I want to know about asterisk and openBTS
If anybody made any test and experience...
Thanks
--
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New to Asterisk? Join us for a live introductory webinar
On 19/08/10 18:20, equis software wrote:
I want to know about asterisk and openBTS
If anybody made any test and experience...
I saw a presentation a few months ago where one of the openBTS project
founders talked about one early system they set up on a very small and
remote Pacific island
May be he was David Burguess, another founder is Harvind Samra ...
Do you know about any Equipment working?
On Thu, Aug 19, 2010 at 2:27 PM, Alan Lord (News) alansli...@gmail.comwrote:
On 19/08/10 18:20, equis software wrote:
I want to know about asterisk and openBTS
If anybody made any
If I set a call-limit field on a peer in users.conf..
I am seeing that it seems to affect other peers too?
I am running Asterisk 1.4.18 has someone seen this issue.
Peer 1 has call-limit=5
Peer 2 has call-limit=20...
In the SIP trace I see when Peer2 hits 5, Asterisk sends back a 480
On Thu, Aug 19, 2010 at 10:53 AM, equis software
equissoftw...@gmail.com wrote:
May be he was David Burguess, another founder is Harvind Samra ...
Do you know about any Equipment working?
The primary piece of equipment consists of the USRP made by Ettus
Research along with driver support
On Wednesday 18 August 2010 16:52:38 Kathryn Jones wrote:
I must not be receiving them properly, since I can't make it work. I just
can't see why :P.
My asterisk version is 1.6.2.6. Like I said before, for outgoing .call
files WaitExten works fine, it's on incoming calls that I cannot receive
On 19/08/10 18:20, equis software wrote:
I want to know about asterisk and openBTS
If anybody made any test and experience...
This island runs it's GSM network on OpenBTS: http://www.niueisland.com/
This was the place he presented about.
Read the blog here:
On Thu, Aug 19, 2010 at 12:37 PM, Alan Lord (News) alansli...@gmail.com wrote:
On 19/08/10 18:20, equis software wrote:
I want to know about asterisk and openBTS
This island runs it's GSM network on OpenBTS: http://www.niueisland.com/
This was the place he presented about.
Read the blog
Thanks for your reply :)
I added Answer to my dialplan:
exten = xxx,1,Answer()
exten = xxx,n,Background(welcome)
exten = xxx,n,WaitExten(7)
exten = _X,1,AGI(agi.php)
exten = _X,n,PlayBack(vm-tocallnumber)
exten = _X,n,Dial(SIP/voiptrunk/${NUM})
exten = t,1,Noop(*timeout*)
exten =
John: August 1 is the deadline???
On Thu, Aug 19, 2010 at 11:58 AM, John Todd jt...@digium.com wrote:
Just a reminder: AstriCon is coming up in October in Washington, DC (
http://www.astricon.net/
) and we're looking forward to seeing you there!
We're getting to the deadline for
El 19/08/10 15:07, Kathryn Jones escribió:
Thanks for your reply :)
I added Answer to my dialplan:
exten = xxx,1,Answer()
exten = xxx,n,Background(welcome)
exten = xxx,n,WaitExten(7)
exten = _X,1,AGI(agi.php)
exten = _X,n,PlayBack(vm-tocallnumber)
exten = _X,n,Dial(SIP/voiptrunk/${NUM})
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina
Subject: Re: [asterisk-users] WaitExten() always times out
snip
Til gave you the answer; When you call out the other end controls timing.
Put a waitexten(5,m) in front of
I am making a web interface so users can manage their voicemail. The
only problem I have is that since the Web server and Asterisk run as
different users I need to run some commands through Asterisk so I can
manipulate the voicemail files.
I know that from the CLI I can user the
Sorry for the delay - I lost this message in the middle of a digest.
I tried Answer(2000) and was getting an annoying warning:
[Aug 15 17:20:11] WARNING[15516]: channel.c:1044 __ast_queue_frame:
Exceptionally long voice queue length queuing to DAHDI/1-1
So I changed it back to Wait(2).
I'll try
On Thu, Aug 19, 2010 at 3:14 AM, Deepika Nijhawan
Does anyone has an idea how to tell asterisk to use codec A for first
50 calls and then codec B for rest of the calls.
On Thu, 19 Aug 2010, Sherwood McGowan wrote:
the easiest way I can think of is to use a global variable that you
On Thu, 19 Aug 2010, Carlos Chavez wrote:
I am making a web interface so users can manage their voicemail.
The only problem I have is that since the Web server and Asterisk run as
different users I need to run some commands through Asterisk so I can
manipulate the voicemail files.
On 18 August 2010 08:52, Nasir Iqbal na...@ictinnovations.com wrote:
Avoid to use MySQL dialplan application, instead write an AGI script
for this purpose
On Wed, Aug 18, 2010 at 3:59 PM, Geraint Lee gera...@gmail.com wrote:
This is what I ended up doing, working fine now. Cheers
On
Hi,
there's still no conceivable reason
What can be? except performance! (as asterisk has to create one additional
leg and bridge it) Which is very conceivable to those who are dealing with
high load traffic.
And what will be the option, if other outgoing call
requires different custom CLI
hi, all
i want to use g729 codec for set up a call. so i donwloaded the
so file from web site: http://asterisk.hosting.lv/#bin
and install it properly.
*CLI
*CLI core show translation
Translation times between formats (in microseconds) for one
second of data
Source Format
- Zhang Shukun bit...@gmail.com wrote:
== Using SIP RTP CoS mark 5
[Aug 20 18:23:21] NOTICE[14543]: chan_sip.c:8454 process_sdp: No
compatible codecs, not accepting this offer!
Could you tell me what 's wrong?
sarcasmBesides the use of an illegal G.729 implementation? /sarcasm
Are
On Thu, Aug 19, 2010 at 5:56 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Thu, 19 Aug 2010, Carlos Chavez wrote:
I am making a web interface so users can manage their voicemail.
The only problem I have is that since the Web server and Asterisk run as
different users I need to
On Thu, Aug 19, 2010 at 9:44 PM, Tim Nelson tnel...@rockbochs.com wrote:
- Zhang Shukun bit...@gmail.com wrote:
== Using SIP RTP CoS mark 5
[Aug 20 18:23:21] NOTICE[14543]: chan_sip.c:8454 process_sdp: No
compatible codecs, not accepting this offer!
Could you tell me what 's wrong?
On Thu, Aug 19, 2010 at 10:28 PM, Zhang Shukun bit...@gmail.com wrote:
== Using SIP RTP CoS mark 5
[Aug 20 18:23:21] NOTICE[14543]: chan_sip.c:8454 process_sdp: No
compatible codecs, not accepting this offer!
Could you tell me what 's wrong?
On Thu, Aug 19, 2010 at 7:46 PM, Nasir Iqbal na...@ictinnovations.com wrote:
Hi,
there's still no conceivable reason
What can be? except performance! (as asterisk has to create one additional
leg and bridge it) Which is very conceivable to those who are dealing with
high load traffic.
And
On Thu, 19 Aug 2010, Tino wrote:
But when i call my DID number following dialplans are being executed.
What i need is to set a variable with one value for one DID number and
set the same variable with another value for another DID number. Also
any contexts should be able to use this
On Thu, Aug 19, 2010 at 6:05 PM, Steve Edwards
asterisk@sedwards.com wrote:
On 18 August 2010 08:52, Nasir Iqbal na...@ictinnovations.com wrote:
Avoid to use MySQL dialplan application, instead write an AGI script for
this purpose
On Wed, Aug 18, 2010 at 3:59 PM, Geraint Lee
Un-top-posting...
Un-top-posting...
On 08/17/2010 09:00 AM, Tino wrote:
I would like to send sms to some external phone numbers from my asterisk
server. Is it possible to send sms via softphones like X-Lite ? . Any
tips regarding this will be helpful
On Wed,
On Thu, Aug 19, 2010 at 11:01 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Thu, 19 Aug 2010, Tino wrote:
But when i call my DID number following dialplans are being executed.
What i need is to set a variable with one value for one DID number and set
the same variable with another
Hi Sherwood
I actually do want dynamic CLID as I tried to make clearer
I don't know if this makes it any clearer -
An internal call from Ext123 should send 123 as the CLID to SIP/Ext400
but should
send 442071110123 to SIP/TheWorld but an external call from
44123455667788 should
send
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