On Monday 06 Jul 2015, Luca Bertoncello wrote:
John Kiniston johnkinis...@gmail.com schrieb:
The easiest solution may be to strip the leading zero's off your caller
ID before your caller enters the Voicemail app to leave you a message.
ExecIf(REGEX(^[0][0].
On Monday 06 Jul 2015, Luca Bertoncello wrote:
Zitat von A J Stiles asterisk_l...@earthshod.co.uk:
Yes. You should definitely be using A-law for calls to the Outside
World.
Well, I wanted to change these settings, but I'm not sure, where I
have to do that...
I think in the users.conf
On Monday 06 Jul 2015, Luca Bertoncello wrote:
Well, but for voice quality, which codec is better?
alaw or gsm?
A-law is better for voice quality (sorry, thought my original explanation was
obvious). But note that if the destination is a mobile phone, GSM will be
used anyway, at least for
On Sunday 05 Jul 2015, Luca Bertoncello wrote:
Hi list!
I noticed that when the phone of my wife calls the gsm codec will be used,
but if someone calls the phone, alaw will be used:
Could someone explain me why?
Second question: I think, ulaw/alaw are better then gsm, isn't it?
If so, how
On Monday 06 Jul 2015, Luca Bertoncello wrote:
So, I think, I should try to force the using of alaw for this phone,
is it right?
Usually we don't call mobile phones from our landline...
Yes. You should definitely be using A-law for calls to the Outside World.
If you use a different codec,
On Friday 03 Jul 2015, Jerry Geis wrote:
Ok digging deaper... I was always trying to run the session as
su myuser -c asterisk -fn
This does not seem to work.
If I login as myuser and run asterisk fn it worked... I got a lot of
crackly noise that I normally dont have
but it worked.
Any
On Wednesday 24 Jun 2015, tux john wrote:
hello everyone.
i am using asterisk 11.16 in my home office and i am using fax to email
with it. i am quite happy with the way it works, no problems at all. when
a fax arrives in a particular DID then the system sends it with mailutils
to my email
On Wednesday 24 Jun 2015, Ivan Demkovitch wrote:
Hello team!
I’m planning to add fax functionality to my PBX. From research it seems
that there is 2 options: spandsp and Digium. I lean towards Digium app,
licensing is fine. However, they don’t have download for v13 Should I just
download
On Friday 19 Jun 2015, asterisk wrote:
Hi,
Long story short - I have an ancient Britsh Telecom phone attached to my
Asterisk PBX via Dahdi. It works beautifully, receiving calls, and the
call quality is excellent. However, dialling out is impossible, as
Asterisk consistently mis-reads the
On Friday 19 Jun 2015, Ivan Demkovitch wrote:
Hi again!
Also, given my setup below, how do I send caller id to my cell?
SIP/83@callcentric is my cell, when I get incoming call when someone dials
into Asterisk - I just see public calcentric’s DID number. I want to send
a number of who CALLED
On Thursday 18 Jun 2015, Greg Woods wrote:
I have found several places where it is explained how to do this, and I
have got the following setup, but it is not working (the provider claims
they are not getting a proper caller ID setting from me).
I have a number of extensions that are
On Monday 15 Jun 2015, lu...@sulweb.org wrote:
Hello all,
I'm new here and I'm interested in building a small PBX with asterisk at
home. I have one single PSTN line and ethernet cabling in place. I
already have fairly decent PC that I can use (AMD FX 8350 16GB of RAM
and RAID 10 SATA
On Monday 15 Jun 2015, Ivan Demkovitch wrote:
Hello group!
I’m new to Asterisk but got one running finally :)
Now I’m trying to solve following problem. I have company Automated
Attendant and each employee have SIP phone at home, SIP phone in office,
cell phone.
I want all those 3
On Thursday 11 Jun 2015, Luca Bertoncello wrote:
Well, I decided to do that, since I have my Asterisk reachable from
Internet just for my cellphone and I want to avoid that someone guess
my password (random and long, but it's of course possible to guess
with a brute force attack) and call
On Thursday 11 Jun 2015, tux john wrote:
Hello everyone. i am running an asterisk server and i would like to have
the contacts from google. so every inbound call with fetch the caller ID
from google contacts and present it to my screen.
This is really three problems, as follows:
(1)
On Thursday 11 Jun 2015, Luca Bertoncello wrote:
Now my problem is to check in my dialplan if the peer, that originate
the call, is reachable, and if not, to give an error...
Is there any function to know if the peer is reachable?
The peer that *originated* the call *must* be reachable, by
On Wednesday 10 Jun 2015, Luca Bertoncello wrote:
I'm very sorry to write that, but these answers are really NOT helpful...
I searched two days long how can I check it and didn't found anything
useful...
Could someone suggest me a way to check if my Asterisk is an Open
Relay that accept
On Tuesday 09 Jun 2015, Luca Bertoncello wrote:
Now, I tried to register the user of my cellphone using a PC, as my
cellphone was already registered.
And Asterisk accepted this registration... :(
Did you actually reboot the server, as opposed to simply reloading your
firewall configuration
On Monday 08 Jun 2015, Luca Bertoncello wrote:
Hi again, list!
I know, I'm really annoying the list... :)
Everyone has to start somewhere; and at least you aren't asking hundreds of
questions in one go, including some which come under the heading of Don't
even think about trying to set this
On Tuesday 02 Jun 2015, Carlos Chavez wrote:
Ia had a server overload today because someone did a call forward
to their own extension. To do a call forward I write a key called CFWD
with the extensión number and number to dial . The main script tests if
the key/value exists and dials
On Wednesday 13 May 2015, Olivier wrote:
2015-05-06 17:51 GMT+02:00 Tech Support aster...@voipbusiness.us:
I believe that when you choose to store voicemails using IMAP, it applies
to all of your users which may not be what you want to do.
Yes.
These days, voicemail storage type is still
On Friday 01 May 2015, Robert Broyles wrote:
We love our Digium phones and DPMA - but we really need it to work on
our Realtime Platform. Otherwise we lose all the cool features and they
are just standard SIP phones.
Anyone working on a solution for this? Or anyone from Digium see this on
On Wednesday 22 Apr 2015, pankaj pandey wrote:
Hi All,
I am running Asterisk 11.17.1 on Ubuntu 11.10 and i am getting segfault
error very frequently. Due to this my asterisk server dies and i am
getting the following following error in /var/log/kern.log ,
Apr 22 14:21:03 pp kernel: [
** THIS IS NOT WHERE YOUR REPLY BELONGS **
On Wednesday 25 Mar 2015, Salaheddine Elharit wrote:
tnaks for your response but the number dialed exist and i can call this
number when i configure the trunk directly in x-lite and i call call also
this number from my cell phone .
any help
thanks
On Wednesday 18 Mar 2015, Rizwan H Qureshi wrote:
Hi All,
I have to forward incoming call on PRI back out to PRI but I need the
original Callerid to passthrough. Is it possible with DAHDI PRI cards
without involving the service provider?
Thanks
It depends who your service provider is!
Any
On Thursday 12 Mar 2015, Thufir wrote:
I'm testing Asterisk at home, crummy connection. Skype works fine for
me, but every SIP client, even without using Asterisk, fails to connect.
That's ok.
Is swapping out SIP for Skype a big deal?
Stay away from Skype! It is a toxic, proprietary
On Tuesday 10 Mar 2015, janani m wrote:
Thank You .
But now i get solved with that error since I had some mistakes in
installing googletts.agi
Now when calling from my softphone i have written dialplan with an AGI
script to convert from text to speech.
It get executed without error but
On Monday 09 Mar 2015, janani m wrote:
The Error Which I face I have attached.
I need a clarification of Why I face this error and how to overcome this.
Anybody know Please help..
That's a very common error and what it means is, the AGI script
/var/lib/asterisk/agi-bin/googletts.agi
On Friday 27 Feb 2015, ricky gutierrez wrote:
the problem is that my pbx all incoming calls using only the channel
gsm 1 , the idea is that an incoming call to channel 1 is passed to
channel 2
Ah. *Incoming* calls are not something that is within your control; they have
already been routed
On Monday 02 Mar 2015, Mordechay Kaganer wrote:
When a particular server gets about 500 concurrent calls, the sound quality
begins to degrade, the sound plays slowly and with clicks. As far as i
understand, it's because asterisk is unable to send the voice stream in
time i.e. the server is
On Thursday 26 Feb 2015, ricky gutierrez wrote:
Hi A J , I have a sangoma gsm gateway 4channels , not use chan dahdi
O.K. So what does your existing Dial() statement in extensions.conf look
like?
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list,
On Wednesday 25 Feb 2015, ricky gutierrez wrote:
I have a gw wiht 4 port gsm , my provider gives me 4 lines and one of
them is the main , the problem is that all my incoming calls using
this number and is always busy , and the other three are always free,
it is possible that the call is
On Wednesday 25 Feb 2015, Thufir wrote:
On Fri, 20 Feb 2015 13:05:56 -0700, Harry McGregor wrote:
Hypothetical: lag, choppy connection, dropped calls. Of course, I'd
start with checking logs. How would I establish that the problem is that
(some) of the ports aren't gigabit?
Any port with
On Tuesday 17 Feb 2015, Justin Killen wrote:
Hi,
I copied a setup from an older 1.8.5 installation to an 11.15 installation,
and I'm having problems getting call files to work.
. stuff deleted .
Whenever I try to copy this callfile into /var/spool/asterisk/outgoing/ I
get these 3
On Thursday 05 Feb 2015, jg wrote:
Calling from ServerB to ServerA works, but not vice versa. The only odd
thing that appears to me is the different perceived port on ServerA.
ServerA*CLI iax2 show registry
Host dnsmgr Username PerceivedRefresh State
On Monday 02 Feb 2015, spartan1...@hushmail.com wrote:
Hi, I'm connecting 2 Asterisk servers with an IAX2 trunk. Trunk works
fine in testing, no problems there but the Internet at server-A is an
on-demand system that is based on the amount of http/https traffic
going through it (or if the link
On Wednesday 28 Jan 2015, Ethy H. Brito wrote:
Hi all
WE have some users that turns off their phones when they are not at home.
We see the warning message:
Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
just after the Dial() command and a
Everyone
On Monday 26 Jan 2015, Antonio Gómez Soto wrote:
Hi,
does anyone have a recommendation for a SIP phone, which
allows dialing from a phonebook, and hiding the dialed number
from the end users? Also from the call history of course.
It seems Mitel can do this, and I have a use case where
On Monday 19 Jan 2015, ricky gutierrez wrote:
Hi list, I write on the list looking for help, buy a openvox gw gsm
for four channels and I'm a little disappointed with the support
openvox, for some reason , The call doesn´t get trough
support tells me it was my asterisk server, but does not
On Wednesday 07 Jan 2015, Stefan Viljoen wrote:
Hi all
I have a strange issue with 1.8.11.0 on a production Asterisk machine at
our head office, and the same issue with a production machine at a branch
office.
Every now and then, on the head office machine, ODBC CEL and CDR logging
will
On Tuesday 09 Dec 2014, Daniel Gonzalez wrote:
Hi,
Let's say I do:
Set(data=xxx,yyy)
Gosub(my-sub,s,1(${data}))
My subroutine will only receive xxx for ARG1. How can I pass a literal
with a comma to a single argument in a subroutine?
(The point is: when calling the subroutine I do
On Tuesday 09 Dec 2014, Daniel Gonzalez wrote:
Hi,
Let's say I do:
Set(data=xxx,yyy)
Gosub(my-sub,s,1(${data}))
My subroutine will only receive xxx for ARG1. How can I pass a literal
with a comma to a single argument in a subroutine?
(The point is: when calling the subroutine I do
On Thursday 27 Nov 2014, Control Oye wrote:
Hi,
I need dialplan to set INCOMING call forwarding during lunch break to my
secretary.
I want that I can set call forwarding by dialing an extension number to
turn it ON or OFF.
I am using asterisk 11.
What you need to do is, set a global
On Wednesday 26 Nov 2014, Antoine Megalla wrote:
Hi,
I looked for asterisk in /usr/sbin using the commands ls and find and
whereis and it was not there.
I know that the process is killed because when I start asterisk using the
command asterisk -c it starts and then it exits and the
On Thursday 20 Nov 2014, Jayson Baker wrote:
Mailbox continues to be missing most times. Touching (or rm'ing) the file
in /var/spool/asterisk/voicemail does nothing until a core restart now
then as soon as the phone registers the light is sync'ed. MySQL or CURL,
doesn't matter, anything
On Friday 21 Nov 2014, Andrew Colin wrote:
Hi All
We have a strange issue with our hosted asterisk server running on Debian
Internal calls btween extensions using g729 give one way audio
As soon as we change the codec to ALAW the issues goes away.
Any ideas how to fix this?
Outbound
On Friday 21 Nov 2014, Andrew Colin wrote:
I am using the free g729
OK, so there shouldn't be any licencing problems (unless for some reason your
Asterisk is wanting to use the paid-for g.729 aot the Free one. Look at the
CLI output very, very carefully to see if this might be happening).
** THIS IS NOT WHERE YOUR REPLY BELONGS **
On Wednesday 19 Nov 2014, Jayson Baker wrote:
On Wed, Nov 19, 2014 at 3:31 PM, Steve Edwards asterisk@sedwards.com
wrote:
Please don't top-post.
On Wed, 19 Nov 2014, Jayson Baker wrote:
This same issue has happened on
** THIS IS NOT WHERE YOUR REPLY BELONGS **
Which part of THIS IS NOT WHERE YOUR REPLY BELONGS do you not understand?
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
** THIS IS NOT WHERE YOUR REPLY BELONGS **
On Thursday 20 Nov 2014, Jayson Baker wrote:
On Thu, Nov 20, 2014 at 9:56 AM, A J Stiles asterisk_l...@earthshod.co.uk
wrote:
** THIS IS NOT WHERE YOUR REPLY BELONGS **
Which part of THIS IS NOT WHERE YOUR
On Thursday 16 Oct 2014, Stephan Alz wrote:
Hello
I have a simple 1 channel goip gateway
(http://www.voip-info.org/wiki/view/GoIP).
The incoming and outgoing calls work with Asterisk except the caller ID for
the outgoing calls. I think I have exhausted all possible options
regarding
On Friday 10 Oct 2014, Thorsten Göllner wrote:
Hi,
I have Asterisk 11 with DAHDI (Sangoma E1-Card) running on Ubuntu 12.04
LTS. Asterisk and DAHDI-Drivers are installed from source.
When doing an apt-get upgrade the system packages will be update but
sometimes Asterisk is broken. Which
On Tuesday 23 Sep 2014, Brahim Abidar wrote:
hi everyone,
actually i want to release an IVR system using PHPAGI API , in this IVR i
want to get value from the user.
I already used get_data defined in phpagi but they are not able to get the
value given by the user and store it in a php
On Tuesday 23 Sep 2014, Steve Edwards wrote:
For some applications, storing recorded audio (prompts and caller
recordings) as a BLOB in MySQL has advantages.
So, once I have the audio in the database, how can I play it?
Creating temporary files seems so tacky.
Is there another way to
On Monday 22 Sep 2014, Yuriy Gorlichenko wrote:
Hello I have an issue wit MixMonitor. I need to record only answered calls,
so I set b option for this but calls still recording even call no
answered My asterisk version 12.5.1, at my other servers with older
versions of asterisk (11.8 for
THIS IS NOT WHERE YOUR REPLY BELONGS
On Monday 22 Sep 2014, Yuriy Gorlichenko wrote:
2014-09-22 12:12 GMT+04:00 A J Stiles asterisk_l...@earthshod.co.uk:
On Monday 22 Sep 2014, Yuriy Gorlichenko wrote:
Hello I have an issue wit MixMonitor. I need to record only answered
calls
On Thursday 18 Sep 2014, motty cruz wrote:
Hello, I would to allow users to place calls overseas such as India and
Malaysia but only with a security code. if they don't have a security code
I want to be able to drop the calls.
can someone point me to a right direction to achieve this goal?
On Wednesday 17 Sep 2014, Rizwan H Qureshi wrote:
Hi All,
I have a GSM to VoIP gateway (specifically yeaster TG400) which I am trying
to use for kind of a call intercept between two GSM users. Call comes
through one SIM and goes out through another Sim with our Asterisk in
between to log the
On Wednesday 17 Sep 2014, Anurag Rana wrote:
in dialplan:
exten=h,n,NoOp(${DIALLEDPEERNUMBER)
variable ${DIALLEDPEERNUMBER} is returning null.
Suggestions please?
Thanks
Anurag Rana
http://newbie42.blogspot.in/
Asterisk has it mis-spelled as DIALEDPEERNUMBER (sic).
Try
exten =
On Wednesday 17 Sep 2014, Anurag Rana wrote:
Oh, Sorry My mistake, I misspelled it in mail.
It is already ${DIALEDPEERNUMBER}, still returning null.
Anurag Rana
http://newbie42.blogspot.in/
Hmm. I've looked a bit further. According to the documentation,
${DIALEDPEERNUMBER} is set by a
On Wednesday 17 Sep 2014, Anurag Rana wrote:
Thanks, That worked. :)
Anurag Rana
http://newbie42.blogspot.in/
Good; it's always nice to hear that someone has got something working!
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to
(this is not where your reply belongs)
On Monday 15 Sep 2014, Rainer Piper wrote:
Hi Patrick,
github done ;-)
what is HTH ???
HTH == Hope That Helps.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod
On Thursday 11 Sep 2014, rafa alfurqan wrote:
Hi,
Could anyone help me to tell me about how to install and using phpmyadmin
to remotely access asterisk mysql database?
I'm using asterisk 11.0.1 on ubuntu 10.04
and mysql-server version is 5.1.73-0ubuntu0.10.04.1 (ubuntu)
really need
On Sunday 07 Sep 2014, Anurag Rana wrote:
Hi,
I created a dummy dialplan where I ask the user to enter the age.
[macro-age]
exten = s,1,Background(my/age) ;;Play recorded message to enter age
exten = s,n,WaitExten(10)
exten = _XX,1,Set(AGE=${EXTEN});; this line is not
On Monday 08 Sep 2014, Anurag Rana wrote:
@A J Stiles : If you could provide an example as you said, It would be very
nice. Thanks.
This is excerpted from a dialplan application I wrote. It's actually a PIN
entry but should be usable for any general purpose application. Sound files
On Thursday 04 Sep 2014, motty cruz wrote:
Hi All,
I see this kind of attack on our Asterisk Server, do you know how to block
that IP?
Instead of blocking unwanted IPs, you should be permitting only wanted IPs.
--
AJS
Note: Originating address only accepts e-mail from list! If replying
On Thursday 04 Sep 2014, motty cruz wrote:
Hi A J,
believe me, I wish i do as you suggested, however I have a few extensions
outside the office with dynamic IPs, so that is not a possibility.
If you know what ISPs they are using, then you can allow just those ISPs'
address ranges. That will
On Tuesday 02 Sep 2014, Jonas Kellens wrote:
On 02-09-14 11:34, Steven Howes wrote:
On 2 Sep 2014, at 09:03, Jonas Kellens jonas.kell...@telenet.be
mailto:jonas.kell...@telenet.be wrote:
So just before hanging up, I add a custom SIP-header :
exten = s,n,SIPAddHeader(X-My-Hangup:
On Tuesday 02 Sep 2014, Nick Awesome wrote:
Hello guys.
Have 2 external numbers that required registration on provider server,
trunk1: 73432260005@80.75.132.66
trunk2: 73432260050@80.75.132.66
Thing is I can’t figure out how to route them to different IVRs
by default Asterisk can’t
On Monday 01 Sep 2014, Chandran Manikandan wrote:
Hi All,
I would like to Setup own IP PBX Server for our office.
I need to connect our all branch office with head quarter through local
extensions.
I need to receive and make call from our branch office and head quarter
using own DID numbers.
On Thursday 21 Aug 2014, bilal ghayyad wrote:
Hello;
I am facing a trouble with A2Billing when using analogue lines because the
channels are not closing properly when dialing happen through A2Billing
(it seems the dialing scenario including the hangup is not handled
properly through
On Tuesday 12 Aug 2014, Olivier wrote:
Hello,
A couple of questions in relation with Asterisk 12 on Debian Wheezy.
1. Can paquet libpjproject-dev (from wheezy-backport) be installed as
the sole binary to add PJSIP stack to Asterisk 12 (compiled from
source) ?
2. When compiling
On Monday 11 Aug 2014, Farid Fadaie wrote:
Hello,
Full disclosure: my name is Farid Fadaie and I'm in charge of BitTorrent
Bleep (a private P2P SIP-based messaging application in early alpha)
http://blog.bittorrent.com/2014/07/30/building-an-engine-for-decentralized-
communications/
I
On Friday 08 Aug 2014, Gergo Csibra wrote:
Hi,
back in the old analog telephony days there was digital PBX-es and
digital system phonesets. This phonesets have had many individual
illuminatable buttons connected with extensions. The PBX can show on
the buttons if some extension is ringing
On Wednesday 06 Aug 2014, I wrote:
I'm trying -- unsuccessfully! -- to configure an inbound trunk with
Simwood, and I was hoping someone on this list might have managed to do
this.
I have configured some numbers to route to a SIP endpoint
%e164@customer's server
and convinced the
I'm trying -- unsuccessfully! -- to configure an inbound trunk with Simwood,
and I was hoping someone on this list might have managed to do this.
I have configured some numbers to route to a SIP endpoint
%e164@customer's server
and convinced the customer to open up UDP ports 5060 and 1 -
On Monday 04 Aug 2014, upendra wrote:
Hi,
i wanted to know that if i have a message indicator SIP phone , then MWI
will work in ELASTIX ??
Let me know the Details of MWI and how test it.
As long as the message waiting indicator can be controlled via SIP messages,
it should Just Work in
(This is not where your reply belongs)
On Friday 01 Aug 2014, Sameer Rathod wrote:
Hi Matthew,
I know that no one is bounded to solve the issue for me.
I am new to asterisk that's why asking for help only. Pardon me if I did
something wrong.
Please let me know where do I get config.log
On Thursday 31 Jul 2014, James Thomas wrote:
Is the quality the same incoming from mobile as outgoing to mobile?
It's a one-way trunk (outgoing only).
Anyway, I've now fixed it, with help from the trunk provider. Details to
follow in a separate message.
--
AJS
Note: Originating address
I have now fixed this issue, and am posting this for the benefit of anyone else
who may be suffering with a similar problem.
It was, as I suspected all along, a subtle misconfiguration at this end.
The fix was to give the SIP trunk its own configuration stanza in sip.conf as
follows;
I'm having a problem with a new SIP trunk.
Calls within the UK to fixed lines are fine, but calls to mobiles have
noticeably poorer audio quality.
I thought it might have been a codec issue; we have used G.726 for internal
and external calls (over primary ISDN and GSM). So I tried allowing
On Saturday 19 Jul 2014, Norman Molhant wrote:
I tried many things on our FreePBX box and found out
the problem seems somehow linked with the customer's
extension (or phone number), not his inbound route
(changing the latter has no effect on the problem).
Creating a new extension with
On Friday 18 Jul 2014, Haley,Scott A wrote:
I have this working but I have one problem. I need to grab values from
variables that I have set in the calling context to dial. How would I do
that.
I think you need to prefix your variable names with *two* underscores, to make
them indefinitely
On Friday 18 Jul 2014, Haley,Scott A wrote:
That worked. I had to use the *two* underscores in the agi script where I
was setting the values. Thanks.
Glad you got it working in the end!
I always like to use plenty of NoOp() statements to make sure the variables
I'm setting are correct,
On Wednesday 16 Jul 2014, Haley,Scott A wrote:
I have a need to issue a dial command to a number:
same = n,Dial(${DIALGROUP1},${TIMER1},t)
After a number of seconds, let's say 10 seconds. I want to dial another set
of numbers while continuing to ring, or interrupting the first group of
On Thursday 10 Jul 2014, Ismael Gil wrote:
Hi there,
In one of my asterisk installation, there is a Digium E1 pri card
connected. The asterisk and card are working properly.
The problem we have is that when a storm occurs in the area, the card
stops working, and E1 lines connected not
On Tuesday 01 Jul 2014, andrew Colin wrote:
Hi Guys
Does anyone know of any good cdr rating software.
I am looking for something that I can pull reports by extension.
Not a full billing solution like a2billing.
Have you thought of rolling your own? It's not hard to write a program in
On Wednesday 02 Jul 2014, Sameer Rathod wrote:
Hi,
I am facing issue in bypassing asterisk for audio call
can anyone help in packet to packet bridging I had posted the logs in
previous mail
If required again then please let me know
Then why are you replying to a thread which, evidently
On Wednesday 02 Jul 2014, Andrew Colin wrote:
Can you try maybe assist with this, as I have tried for ages and still cant
get it right.
Firstly, have you got CDR working and writing to some sort of database? We
use cdr_mysql; although the more modern recommendation is to use cdr_odbc
(which
On Monday 30 Jun 2014, sylvain GOTRI wrote:
Hi ,
I have asterisk 1.8.5 installed on Centos 6. Now I want to configure my
PBX to work in my network. I see that I can do this with asterisk files
or use database like mysql to do it (realtime)
I want to know what is the best way and what can be
On Thursday 19 Jun 2014, thufir wrote:
http://www.voip-info.org/wiki/view/Asterisk+CRM+Integration
lists a few options. I'm looking for, literally, the simplest FOSS CRM
for click to dial functionality, but don't know where to start.
thanks,
Thufir
The Free version of SugarCRM is
On Thursday 05 Jun 2014, Mojtaba wrote:
My scenario is (2)
After doing some tests with my own hardware, I'm now convinced that this is
actually normal behaviour: As far as Asterisk is concerned, a call is deemed
answered as soon as the hardware seizes the line. It is only not answered
if
On Wednesday 04 Jun 2014, Mojtaba wrote:
Thank you for your replying.
Is there any way so that i could found the far end user pick up phone? I
could use Wait() function in dialplan but i dont how long (secend)
should be wait!
Thanks with Regards.Mojtaba
I'm confused now. Please describe
On Wednesday 04 Jun 2014, Mojtaba wrote:
Hello Experts.
Im working with Asterisk PBXand freeswitch PBX.
I have a challenge with FXO card in Asterisk and i could not solve it yet.
I hope you could guide me in this regards.
When i want route the call to FXO channels, Before the callee answer
On Friday 16 May 2014, Jayson Devor wrote:
Hello Everyone,
We have an order for SMS messaging. Can you gents and ladies be kind enough
to
disclose if SMS is possible using Asterisk? What is a quick way to test a
`Hello World`
to my cell. Finally, do all service providers support SMS
On Wednesday 30 Apr 2014, [Digital^Dude] ® wrote:
make gives this:
codec_amr.c: In function 'amrtolin_sample':
codec_amr.c:227: error: 'AST_FORMAT_AMRNB' undeclared (first use in this
function)
codec_amr.c:227: error: (Each undeclared identifier is reported only once
codec_amr.c:227:
On Thursday 24 Apr 2014, Mikael Fredin wrote:
I will look into netcat as well, thank you
There's not much to look into, really! It's just a command-line tool for
connecting STDIN and STDOUT to a network socket.
$ echo -e WIBBLE\nWIBBLE\nWIBBLE | nc somehost.co.uk 3245
will send
WIBBLE
WIBBLE
On Wednesday 23 Apr 2014, CDR wrote:
Dear friends
I filed a bug
https://issues.asterisk.org/jira/browse/ASTERISK-23656
but I am wondering if somebody can figure a workaround. I am stuck
trying to deliver an application.
The case is this: A Record is executed and an immediate Playback
On Wednesday 23 Apr 2014, Steve Edwards wrote:
On Tue, 22 Apr 2014, A J Stiles wrote:
...so absolutely *do not* pay money for a solution, and *do* insist on
the Source Code and Modification Rights.
Even an obvious and simple solution has value if it exceeds the OP's skill
set or the value
On Thursday 24 Apr 2014, binary dreamer wrote:
hello everyone.
I am running asterisk and all of my CDRs are in the default csv.
the system is so limited to ram (only 256) and I cannot run MySQL or any
other program to give CDRs a fancy view.
at the moment the only other software running is
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