Hi all,
I made it easier, AMI was not required, it can be solved directly in the
dialplan:
same = n,Dial(SIP/${TRUNK}/${ARG2}${NUM},60,M(inject^${CALLERNUMBER}))
[macro-inject]
same = s,1,Originate(Local/trunk@injectWarning,app,Playback,Message-Callee)
same =
Hello all,
This may sound an odd question but if you initiate a call using AMI does it
adhere to what has been defined in the dial plan or do we have to write the
logic into the AMI call ?
--
Thanks, Phil
--
_
--
Please ignore as this was a user error!
--
Thanks, Phil
- Original Message -
Hello all,
This may sound an odd question but if you initiate a call using AMI
does it adhere to what has been defined in the dial plan or do we
have to write the logic into the AMI call ?
--
Thanks,
Ok.
Asterisk sends the rtpmap info for the codec.
Is it possible to remove this from the 200 OK sent by Asterisk?
Possible direction I should look.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
Hello All,
I have a problem with Fax For Asterisk, the Successful Rate when sending Fax
are very Low especially when we send the Fax just once. Now I’m trying to
modify the dialplan so it will keep trying to send the fax for maximum 5 times
at once and it only retry if the Sending Status
Am 19.12.11 14:26, schrieb Zoel Hairi:
Hello All,
I have a problem with Fax For Asterisk, the Successful Rate when sending Fax
are very Low especially when we send the Fax just once. Now I’m trying to
modify the dialplan so it will keep trying to send the fax for maximum 5
times at
On 12/19/2011 07:15 AM, William Scott wrote:
Ok.
Asterisk sends the rtpmap info for the codec.
Is it possible to remove this from the 200 OK sent by Asterisk?
Possible direction I should look.
No, it is not, at least not without patching the Asterisk source code
(which of course you are
sorry. i put the wrong the dialplan. it already RetryAttempt2 in it.
exten = s,n(RetryAttempt3),Wait(6)
exten = s,n,NoOp( SENDING FAX RETRY ATTEMPT 2 : ${FAXFILE} :
${FAXOPT(error)} )
exten = s,n,SendFAX(${FAXFILE})
;zoel : Add Retry Attempt 3
exten = s,n,GotoIf($[${FAXOPT(error)} =
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Stefan Schmidt
Sent: Monday, December 19, 2011 8:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sending Fax
Hi,
On a recently updated system , I'm now reading lines as this one
(never noticed them before):
[Dec 19 19:01:52] WARNING[10828]: chan_dahdi.c:11302 pri_dchannel:
Ring requested on unconfigured channel 0/0 span 4
My setup is:
Asterisk 1.6.1.18
Libpri 1.4.12
Dahadi 2.5.0.2
My card is a
Hi all,
I succeed in injecting audio into one channel by mean of ChanSpy, but this
audio cannot be listened correctly. I am using softphones for Android and
iPhone and there are so many cuts so that they cannot understand what is said
in the audio file. Is this because the total RTP bandwidth
Hi,
I've recently met weird behaviour on 2 different and newly upgraded
libpri1.4.12/2dahdi2.5 systems (at the moment, I can't correctly
describe the symptoms but that's another story).
For various reasons, this lead me to wonder which Dahdi/Libpri
combination/version is the most widely used on
On a recently updated system , I'm now reading lines as this one
(never noticed them before):
[Dec 19 19:01:52] WARNING[10828]: chan_dahdi.c:11302 pri_dchannel:
Ring requested on unconfigured channel 0/0 span 4
My setup is:
Asterisk 1.6.1.18
Libpri 1.4.12
Dahadi 2.5.0.2
My card is a
Hi everyone,
Since three weeks ago, we have been getting A LOT of 603 Declined calls
from iCall. I called a few times and their support is either non-responsive
(they never call back) or can't fix the issue. I am wondering if everyone
else is experiencing the same thing or is it because we
Hi All,
Because I am pretty sure we have people in this DL from India, I was hoping
to get the 100% accurate information, is it legal to make calls from any
coutry to Indian mobile phones through an Asterisk server based in India?
--
Khalid Touati
Network Administrator
--
The Asterisk Development Team has announced the release of Asterisk
1.6.2.22.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.22 corrects two flaws in sip.conf.sample
related
to AST-2011-013:
* The sample
On 12/19/2011 4:08 PM, Asterisk Development Team wrote:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.22
or for the non-404-version:
http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/ChangeLog-1.6.2.22
;p
--
Jeremy Kister
http://jeremy.kister.net./
SIP in India is illegal.
Nick.
On Mon, Dec 19, 2011 at 3:06 PM, khalid touati khalidtou...@gmail.com wrote:
Hi All,
Because I am pretty sure we have people in this DL from India, I was hoping
to get the 100% accurate information, is it legal to make calls from any
coutry to Indian mobile
On Mon, 19 Dec 2011, Nick Khamis wrote:
SIP in India is illegal.
What about IAX, Skype, VPN, etc?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline
Hello,
I have a SIP provider whom I may want to have multiple trunks with, rather than
just adding more channels to the individual trunk. I have discussed the matter
with them they have told me that the only way that they identify which trunk
should be used for each call is simply by the
It seems quite unlikely that the presence of
an 'a=rtpmap' line in the SDP for G.729 is going to cause a call to have any
problems.
Thanks for the reply.
I'll expand on the scenario...
This particular ATA does not send 'a=rtpmap' for any codec.
When talking to a Asterisk PBX everything
Hello all,
I have a system with FreePBX, and as far as I can tell it does not provide a
means to limit the number of simultaneous inbound calls on a SIP trunk.
Therefore I suspect that I'll need to do some manual dialplan manipulation.
Essentially I will have 1 (or possibly 2) SIP trunk(s)
On Mon, 19 Dec 2011, Douglas Mortensen wrote:
I have a system with FreePBX, and as far as I can tell it does not
provide a means to limit the number of simultaneous inbound calls on a
SIP trunk. Therefore I suspect that I’ll need to do some manual dialplan
manipulation.
The GROUP() and
I could be wrong but this sounds like a NAT issue rather SIP related packet
issue. You are not receiving a response back is what I get a lot of times
when my NAT is not setup properly. Call goes on for 10 or 20 second (I try
the echo application and it hangs up before I get to talk) and then cuts
On 20 December 2011 12:51, Bruce B bruceb...@gmail.com wrote:
I could be wrong but this sounds like a NAT issue rather SIP related packet
issue.
I looked at this to start with. Spent sometime comparing addresses and
ports between successful and failure packets. Couldn't see any ports
that
On Tuesday 20 Dec 2011, Steve Edwards wrote:
On Mon, 19 Dec 2011, Nick Khamis wrote:
SIP in India is illegal.
What about IAX, Skype, VPN, etc?
The only thing that is not permitted is bridging Internet calls with the
Indian PSTN. In fact, that too is allowed if you have a VoIP licence
Right check out Cordia.LT
Sent from my iPhone 4S
On Dec 19, 2011, at 9:58 PM, Raj Mathur (राज माथुर) r...@linux-delhi.org
wrote:
On Tuesday 20 Dec 2011, Steve Edwards wrote:
On Mon, 19 Dec 2011, Nick Khamis wrote:
SIP in India is illegal.
What about IAX, Skype, VPN, etc?
The only
Can you register with Eyebeam to VSP and have it work? Make sure you are on
the exact same network as the ATA when making this test. This should
isolate the NAT issue.
On Mon, Dec 19, 2011 at 9:27 PM, William Scott will...@magicwilly.infowrote:
On 20 December 2011 12:51, Bruce B
Thank you Raj,
so with VOIP license calls can go beyond our pbx to PSTN (india), right, if
so this what i needed to know to call Indian cellphone from US (or other
countries)
On Mon, Dec 19, 2011 at 10:03 PM, Robert-IPhone rhuddles...@gmail.comwrote:
Right check out Cordia.LT
Sent from my
On Tuesday 20 Dec 2011, khalid touati wrote:
Thank you Raj,
so with VOIP license calls can go beyond our pbx to PSTN (india),
right, if so this what i needed to know to call Indian cellphone
from US (or other countries)
If your objective is to originate calls in the US (using whatever
On 20 December 2011 14:22, Bruce B bruceb...@gmail.com wrote:
Can you register with Eyebeam to VSP and have it work? Make sure you are on
the exact same network as the ATA when making this test. This should isolate
the NAT issue.
Great tip.
Eyebeam dosen't send a rtpmap for known codecs
AFAIK you can add exterin= in sip.conf for each trunk, correct me if i'm
wrong.
2011/12/20 Douglas Mortensen d...@impalanetworks.com
Hello,
** **
I have a SIP provider whom I may want to have multiple trunks with, rather
than just adding more channels to the individual trunk. I have
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Zoel Hairi
Sent: Monday, December 19, 2011 11:21 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Sending Fax
May I ask why do you need different IP addresses to source calls? I mean,
its not a common practice, would like to understand the idea behind it.
*José Pablo Méndez
*
On Mon, Dec 19, 2011 at 11:07 PM, Anton Kvashenkin
anton.juga...@gmail.comwrote:
AFAIK you can add exterin= in
This list is a great resource and I thank all the Asterisk Guru's who
actively contribute to it.
In Leif Madsen's AstriCon 2010 talk titled 5 Things You Didn't
Know Asterisk Could
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