On Tuesday 29 Mar 2016, Rizwan H Qureshi wrote:
> Hi Everyone,
> I need to develop a service which tells me whether a given phone number is
> in service and is valid or not. It can be international number. This is
> basically to clean the list of leads we have. Is there any service which
> can give
On Thursday 24 Mar 2016, Tony Mountifield wrote:
> In article <201603241343.24128.asterisk_l...@earthshod.co.uk>,
> A J Stiles wrote:
> > When placing a call over a SIP channel to a mobile phone, if the phone is
> > engaged, it does not return a BUSY status straig
On Thursday 24 Mar 2016, Mamadou NGOM wrote:
> Hello,
> I am asking if it is possible to left from a version to another one of
> asterisk without reinstalling it. I would like to say for example is
> there a linux command which allows us to left version 12 to 13. Passage
> from a version to an ot
I'm not sure if this is an Asterisk thing, a handset thing or a telco thing,
so please be gentle with me if this is not the right place to ask .
When placing a call over a SIP channel to a mobile phone, if the phone is
engaged, it does not return a BUSY status straightaway. Rather, I get a
On Wednesday 23 Mar 2016, Olivier wrote:
> I'm thinking about something to delegate provisionning to end users:
> a new employee joins the company, the system I'm after let him enter his
> own name himself, once for all.
This is generally good, because it means less work for you :) Just be
care
and after the timeout, announce the remaining
letters); two presses within timeout clears the name entered so far and
starts again from scratch.
So, for example, someone might enter:
2 [set current letter to "A"]
(user waits for timeout) [Store and say "A". Set current letter t
On Monday 21 Mar 2016, somsad khan wrote:
> Hello guys,
>
> I need some help.
>
> I have a client coming who wants to assign 5 different numbers to one
> virtual employee SIP phone at his desk or softphone (Zoiper).
>
> which I can assign for the incoming or outgoing both.
>
> but the problem i
On Thursday 10 Mar 2016, Joshua Colp wrote:
> I wrote:
> > I can't seem to find a definitive answer on this, and I really don't want
> > to risk breaking a production server to find out; so I am going to try
> > asking this here, and maybe anyone else in the same situation searching
> > the archive
I can't seem to find a definitive answer on this, and I really don't want to
risk breaking a production server to find out; so I am going to try asking this
here, and maybe anyone else in the same situation searching the archives
sometime in future will find the answer I get.
Can you use variab
On Wednesday 02 Mar 2016, Ryan, Travis wrote:
> I am wondering what the best solution is for initiating a call from Outlook
> Contacts. I imagine something that would start the call from the outlook
> card (or similar) and then dial the user's extension and the contact's
> phone number and place th
On Wednesday 17 Feb 2016, Sonny Rajagopalan wrote:
> OK. Let me ask this. Is anything else necessary, except choosing TCP as the
> preferred protocol on the client, to make TCP w Asterisk work? At the
> moment, I have only changed one line in pjsip.conf from my working UDP
> setup:
>
> [transport-
On Wednesday 17 Feb 2016, imperium broadcast wrote:
> I kinda have it working with chan_sip.
>
> Dial(SIP/+${EXTEN}\;phone-context=+44@10.10.10.10;user=phone)
> But it doesn't include the user=phone at the end when dialling out.
>
> "To: ".
>
> even adding
> usereqphone=yes
> to the sip.conf doe
On Wednesday 17 Feb 2016, Goke Aruna wrote:
> Hello all,
> Can someone recommend what hardware to use for a 1000 analogue line
> capacity asterisk PABX?
>
> Regards
A PCI express card with four primary rate ISDN ports, each linked up to a
channel bank, will give you 120 analogue lines. So you w
On Thursday 04 Feb 2016, Marek Červenka wrote:
> hi,
>
> is there way to get SQL schema for Asterisk 13.7.0 without alembic?
> thanks
Assuming you already have Asterisk up and running, you can just use
$ mysqldump -d -uroot DATABASE TABLE1 TABLE2 TABLE3 ...
will print (on STDOUT, so you can ju
On Wednesday 27 Jan 2016, James Cloos wrote:
> I gave up switching my edge asterisk to pjsip at least twice because I
> couldn't figure out how to configure it properly for a dynamic ip. And
> I sent a note to one of the lists at least on the 2nd attempt.
>
> That install doesn't need nat for si
On Wednesday 27 Jan 2016, Marek Červenka wrote:
> Dne 27.1.2016 v 13:14 A J Stiles napsal(a):
> > On Wednesday 27 Jan 2016, Marek Červenka wrote:
> >> hi,
> >>
> >> i have strange problem with asterisk 13 mixmonitor, recording to wav
> >> (centos6)
On Wednesday 27 Jan 2016, Marek Červenka wrote:
> hi,
>
> i have strange problem with asterisk 13 mixmonitor, recording to wav
> (centos6)
> when the system is under load, there are sometimes missing syllable
>
> there arent BIG spikes on cpus
> recordings are to ramdisk (/dev/shm)
>
> any hints
On Monday 25 Jan 2016, waqas.mehmood90 wrote:
> I am working on asterisk ivr .i am facing problrm in crontab.when i run
> example it give bash 5:command not found then i check and found that no
> crontab for root user kindly guide me please
Hello, is that the vet? One of my animals is poorly. Wh
On Thursday 21 Jan 2016, Jerry Geis wrote:
> >Not really. Very little info to go on so far. You need to give us
> >more detail of what you are doing with AGI and AMI.
>
> Sorry - let me try again...
>
>
> I am basically doing the following:
> 1) calling a phone SIP/401 upon answer run an AGI for
On Thursday 21 Jan 2016, Jerry Geis wrote:
> I am using the AMI interface to start calls.
>
> At one point I have a 10 second delay "Wait(10)" in the dialplan...
> During this time it "seems" that if I then connect with the manager
> interface
> and place a call that nothing happens till the 10 se
On Wednesday 20 Jan 2016, bilal ghayyad wrote:
> Hello List;
> I am facing a trouble with a sip trunk on asterisk 1.4 and asterisk 1.8 and
> I am getting the following debug, can someone advise me about the
> solution: <--- SIP read from Provider_IP_Address:5083 --->INVITE
> . [stuff deleted] .
On Monday 18 Jan 2016, Matthew Murphy wrote:
> Hi everyone,
>
> I am getting a segmentation fault (seems to occur randomly) using Asterisk
> 13.7.0-rc2 with PJProject 2.4.5. It appears to be something that
> libmysqlclient is complaining about when doing a query in
> ps_endpoint_id_ips. We are usi
On Wednesday 13 Jan 2016, waqas.mehmood90 wrote:
> How to get user extention no in agi php scrip from which he's calling on
> ivr i am using cid and able to get his name but not his extention no
> please help me
Within the dialplan, what you are looking for would be ${CALLERID(num)} . So
you cou
On Fri, Oct 16, 2015 at 12:45 PM, Bryant Zimmerman wrote:
>> I have a project that is requiring the use of MS SQL from asterisk. I
>> get an error when the pjsip contact tries to update the contact table.
>>
>> [Oct 15 21:34:55] WARNING[3033]: res_odbc.c:649
>> ast_odbc_prepare_and_execute: SQL
On Tuesday 03 Nov 2015, sean darcy wrote:
> On 11/01/2015 12:38 PM, sean darcy wrote:
> > I'm not getting any ringing when I use option r with Dial:
> >
> > Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com,,rTt") in
> > new stack
> >
> > Otherwise all works. The call goes through, good audio.
>
I recently had a nightmare building up some servers with these OpenVox cards.
Although I have used them successfully in the past, the chan_extra driver
building process has always been highly temperamental (although to be fair,
they have always worked fine once any necessary tweaking was done).
On Friday 25 Sep 2015, Ryan, Travis wrote:
> I've not used analog for quite some time. It seems it's not possible in
> asterisk to spoof a phone number/name on an analog call?
Probably not if you are using an analogue FXO connection to the exchange;
because there is no standardised way of communi
On Thursday 10 Sep 2015, 陳伯濤 wrote:
> Hi,
>
> I install Asterisk app_mp4, and use mp4save to record mp4 video file, then
> we can play the recorded mp4 file by using mp4play. But the recorded mp4
> file can not be played by MS media player or Quick Time Player. And we
> download mp4 file from inte
On Wednesday 02 Sep 2015, Avanish Shahi wrote:
> Now I’m trying to solve following problem. I have a requirement that
> each employee should have SIP phone at home, SIP phone in office,
> cell phone with same user.
>
>
> I want all those 3 phones to be “one extension”. So, if someone calls
> our
On Wednesday 12 Aug 2015, Jonas Kellens wrote:
> Hello
>
> I was wondering of it is possible to have Queue Agents with the same
> priority (penalty) but with a certain order ?
>
> So I have 20 Agents.
>
> Agent 1 till Agent 10 has penalty 1.
>
> Agent 11 till Agent 15 has penalty 2.
> (only con
On Friday 07 Aug 2015, Tech Support wrote:
> All;
>
> I know that there is no way to determine an exact number, or even a
> close number, but does anyone know a ballpark figure of how many Asterisk
> deployments are out there worldwide? How about the percentage of Asterisk
> PBX's compared to
On Thursday 06 Aug 2015, Jerry Geis wrote:
> I am looking for a push to talk solution does anyone know of a good
> PTT phone one that works with asterisk.
Um . Asterisk supports full-duplex telephony, so there's no need for any
of that "over to you, roger and out" business -- you can actuall
On Monday 03 Aug 2015, Eric Klein wrote:
> Hi all,
>
> Strange request, I have a customer where we are putting an Asterisk PBX in
> front of a legacy (non-VoIP) PBX. One of the requirements it that the
> Asterisk PBX have 2 PRI ports (on towards the legacy PBX and one towards
> the carrier) with t
On Saturday 01 Aug 2015, Murthy Gandikota wrote:
> Hi All
>
> Has anyone used Asterisk for a Call Center operation? What I mean is: given
> a list of phone numbers, can Asterisk dial each number, play a message and
> accept some DTMF?
Yes it can, very easily. But before you go too far, you need
On Wednesday 29 Jul 2015, Murthy Gandikota wrote:
> Hi All,As
> Downloaded latest version of Asterisk from www.asteriskwin32.com and
> installed on Windows 7.
Why?
Trying to get Asterisk to run on Windows is like trying to teach a gerbil to
bark. It's an extraordinary effort, and the result is
On Wednesday 15 Jul 2015, Luca Bertoncello wrote:
> But it seems, that I found the problem, adding:
>
> disallow=all
> allow=g729
>
> to the configuration of the peer for this number...
You need the following;
disallow=all
allow=alaw
in the configuration for *every* device. There is literally
On Thursday 16 Jul 2015, Thyda ENG wrote:
> I would like to see how can we config the asterisk to enable calling to
> multiple SIP number at the same time?
If you want to have a number that will call several phones when dialled, you
can do it in the Dial() command. The following example refers t
On Friday 10 Jul 2015, Thyda ENG wrote:
> Dear Sir,
>
> Does the asterisk support SMS feature ?
> If it does how can we config that ?
> I am waiting for your reply,Thank.
You need a suitable GSM card. We have used the OpenVox G400P / E400E series.
This has a facility for sending SMS directly v
On Wednesday 08 Jul 2015, Andrew Colin wrote:
> Hi Guys
>
>
>
> I am trying to write a macro for a call return so for example
>
> Anyone in the company transfers a call to another extension and it is not
> answered etc it must return to the person who did the transfer
>
> I have got it working
On Wednesday 08 Jul 2015, Thyda ENG wrote:
> Hi,
>
> I am new to asterisk, I have set up the asterisk server and successfully I
> could make the dialplan between 2 SIPs but when there are more than two
> sips calling each other, my dialplan seems doing the wrong routing to the
> sip. Do i need to
On Monday 06 Jul 2015, Luca Bertoncello wrote:
> John Kiniston 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]."
> > ${CALLERID(NUM)})?Set(CALLERID(nu
On Monday 06 Jul 2015, Luca Bertoncello wrote:
> Zitat von A J Stiles :
> > 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
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 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 s
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 wor
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
> downlo
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 C
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
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 sho
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 d
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 thos
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) Access
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 u
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, b
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 acc
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 t
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 :
> > 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 a compilation tim
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 o
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
> than
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!
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 p
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 e
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 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 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 Thursday 26 Feb 2015, ricky gutierrez wrote:
> Hi A J , I have a sangoma gsm gateway "4"channels , 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 reply
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 tr
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 w
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 thes
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
> 80.
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
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
>
>
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 whe
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 n
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
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 subro
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 subro
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
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
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).
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?
>
> Out
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, anythin
** 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
>
> wrote:
> > ** THIS IS NOT WHERE YOUR REPLY BELONGS **
> >
> > Which part of "THIS IS
** 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 Wednesday 19 Nov 2014, Jayson Baker wrote:
> On Wed, Nov 19, 2014 at 3:31 PM, Steve Edwards
>
> wrote:
> > Please don't top-post.
> >
> > On Wed, 19 Nov 2014, Jayson Baker wrote:
> > This same issue has happened on 1.8 as well. A
On Tuesday 28 Oct 2014, NACHIAPPAN, SUBBAIAH (SUBBAIAH) wrote:
> Hello,
>
> I am new to Asterisk forum :).
>
> I have a requirement of terminating 3G Mobile originated calls (coming
> through 3G-MSC) to EPBX Phones via Asterisk PBX.
>
> Setup:
>
> Mobile > Mobile Switching Center ( 3G)--
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
> regardi
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.
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 wa
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 var
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 :
> > On Monday 22 Sep 2014, Yuriy Gorlichenko wrote:
> > > Hello I have an issue wit MixMonitor. I need to record only answered
> > &g
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
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 goa
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
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
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