Am 24.05.12 23:46, schrieb bilal ghayyad:
> Thanks for all for the help and kindly reply.
>
> One last point that will help me alot:
>
> I am thinking to have 4 Servers running Asterisk and 2 Servers to be for
> database. The load to be distributed on the 4 Asterisk Servers with ability
> to be
Hello Michael,
Thanks a lot for your immediate help. After applying patch MixMonitor
started works normally,
I can understand that this can be Happen in asterisk 10.4 but as a stable
and Long support version 1.8.12.0 this should not happen. I got same error
in both version.
Anyways this patch so
On 5/23/12 2:42 AM, Danny Dias wrote:
Can i delete like this:
rm -rf /var/spool/asterisk/voicemail/voicemailcontextcustomer/300/INBOX/*.*
Is that ok? will this break something?
that's ok
no it shouldn't break anything.
however if you're going to delete some of the messages. you have to
renumb
El mié, 23-05-2012 a las 11:42 +0200, Danny Dias escribió:
> Can i delete like this:
> rm
> -rf /var/spool/asterisk/voicemail/voicemailcontextcustomer/300/INBOX/*.*
You can make that without problems
> Is that ok? will this break something?
Yes, that's ok
regards,
--
Ing CIP. Alejand
Looks like Swift() (whatever that is) is not returning ?
On 24 May 2012 23:07, Justin Killen wrote:
> ** ** **
>
> Here is the output from the cli:
>
> ** **
>
> dozer*CLI> core show channels
>
> Channel Location State Application(Data)
>
> DAHDI/5-1
Why don't you use AMI? There's are phpami project if you google.
Sent from my iPhone
On May 25, 2012, at 1:51 AM, Kamlesh Kumar wrote:
> Hi,
>
> I'm using AMI to get the extension status but always get -1 i.e. extension
> not found.
>
> #!/usr/bin/php -q
> include_once ("phpagi-2.14/phpa
- Original Message -
> From: "Jayesh Labade"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Thursday, May 24, 2012 4:10:29 PM
> Subject: [asterisk-users] Asterisk MixMonitor starts recording 44
> bytes file
> Hello All,
> I have installaed asterisk 10.4 in m
Here is the output from the cli:
dozer*CLI> core show channels
Channel Location State Application(Data)
DAHDI/5-1s@DB_LOOKUP:24 Up Swift(""Schedule for employee
1 active channel
1 active call
1528 calls processed
dozer*CLI> core show channel dahdi/
My question is:
Is it really possible to have the asterisk configuration in the database server
instead of having it in conf files? HOW? I am asking this because what I
noticed in AsteriskNow and in A2Billing and Vicidial or Goautodial that
whatever I do configuration in the GUI, then the conf
Thanks for all for the help and kindly reply.
One last point that will help me alot:
I am thinking to have 4 Servers running Asterisk and 2 Servers to be for
database. The load to be distributed on the 4 Asterisk Servers with ability to
be redundant (using any redundancy technique). The 4 Aster
- Original Message -
> From: "Jayesh Labade"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Thursday, May 24, 2012 3:10:29 PM
> Subject: [asterisk-users] Asterisk MixMonitor starts recording 44 bytes file
>
> Hello All,
>
>
> I have installaed asterisk 10.4
Hello All,
I have installaed asterisk 10.4 in my machine. Now suddenly MixMonitor
application starts generating 44 Bytes of Recording file.
Is this new tye of Bug? Help me..
Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com
--
__
Hi, I'm using AMI to get the extension status but always get -1 i.e. extension
not found. #!/usr/bin/php -q
request['agi_extension'];$as->connect("localhost", "user",
"passwd");$status = $as->ExtensionState($exten,'context',1);
$status1 = $status['Status'];
$agi->verbose("Extension status is
Hello Steve, it's working fine, thanks for your suupport. thanks,Kamlesh
> Date: Tue, 22 May 2012 10:36:20 -0700
> From: asterisk@sedwards.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] use of Read cmd with AGI
>
> Un-top-posting...
>
> > From: alejandro.belt...@s
About a month ago, we switched our PRIs from being run through a Nortel
Meridan system to an Asterisk based PSTN gateway using a TE210P card.
Since the cut over I have been getting reports of DTMF tones being heard
by my internal users when on calls to/from the PSTN.
I have confirmed via loggi
I had considered this, however, I was trying not to buy another DID. It may
end up being the best solution.
On May 24, 2012 12:26 PM, "A J Stiles"
wrote:
> On Thursday 24 May 2012, Cody Harris wrote:
> > I'm trying to implement a fax/voice switch. I have faxdetect=both in my
> > sip.conf, and wh
On Thursday 24 May 2012, Cody Harris wrote:
> I'm trying to implement a fax/voice switch. I have faxdetect=both in my
> sip.conf, and when I use sip, it works well. However, from what I can
> tell, there's no such option for IAX2 connections.
>
> Any ideas on what I can do here, or am I out of l
Sorry I hit send by mistake (touchscreens, sigh)
I've had good success with faxing over voip, I'm not expecting it to be
perfect, and my provider (voip.Ms) is planning on t.38, but I'm looking for
an interm solution. Audio faxing has worked every attempt both sending
receiving (5 and 5).
Should I
I'm running on 1.8 as of now
On May 24, 2012 11:00 AM, "Kevin P. Fleming" wrote:
> On 05/24/2012 09:44 AM, Tim Nelson wrote:
>
>> BUT, even if fax is detected on an IAX2 channel, the only reason would be
>> to change dialplan logic accordingly correct? There is no T.38 equivalent
>> within IAX2,
Thanks Kevin,
updtl debug is what I am looking for, I guess.
Arstan
Sent from my iPhone
On May 24, 2012, at 11:25 PM, "Kevin P. Fleming" wrote:
> On 05/24/2012 10:19 AM, Arstan Jusupov wrote:
>> I am sending and receiving fax.
>>
>> I have an issue where sending and receiving is intermittent.
On 05/24/2012 10:19 AM, Arstan Jusupov wrote:
I am sending and receiving fax.
I have an issue where sending and receiving is intermittent. Provider is
claiming that It doesn't always receives t.38.
This is very confusing. In your diagram, you show the connection to the
provider being an E1.
I am sending and receiving fax.
I have an issue where sending and receiving is intermittent. Provider is
claiming that It doesn't always receives t.38.
So I thought if I could see if Asterisk is sending and receiving t.38 as it
should be.
Oh yeah, I am using ATA with t.38 support which is con
On 05/24/2012 09:54 AM, Arstan wrote:
Dear list,
I have a project where I have:
Asterisk 10 <-->AudioCodes <--> E1<--> Provider
AudioCodes supports T.38 and passes the faxes through E1 to the
provider. From what I read, Asterisk 10 has the most stable(full) T.38
among other releases.
Asterisk
On 05/24/2012 09:44 AM, Tim Nelson wrote:
BUT, even if fax is detected on an IAX2 channel, the only reason would be to
change dialplan logic accordingly correct? There is no T.38 equivalent within
IAX2, which means the OP will be handling faxes over a clear VoIP channel. The
information here i
Dear list,
I have a project where I have:
Asterisk 10 <-->AudioCodes <--> E1<--> Provider
AudioCodes supports T.38 and passes the faxes through E1 to the provider.
>From what I read, Asterisk 10 has the most stable(full) T.38 among other
releases.
My Question: Can I somehow see in the logs if T.
AsteriskNOW is a GUI on top of Asterisk; it does not change the ability
of the system to handle call load.
I thought the AsteriskNOW GUI was now a FreePBX clone. If so, every
call now uses a perl script to make the call. This is considerably more
overhead than a dial-plan written in native
- Original Message -
> On 05/23/2012 08:41 PM, Cody Harris wrote:
> > Hello All,
> > I use IAX2 as the incoming connection from my DID provider. For
> > whatever reason, this works best for me, SIP connections lag very
> > frequently and only have about a 50% success rate for incoming
> >
On 05/23/2012 08:41 PM, Cody Harris wrote:
Hello All,
I use IAX2 as the incoming connection from my DID provider. For
whatever reason, this works best for me, SIP connections lag very
frequently and only have about a 50% success rate for incoming calls
(they get dropped mysteriously).
I'm tryin
If I were troubleshooting this, the next thing I would do is verify
connectivity on the relevant ports – more plainly, make sure that there's
not a firewall rule with unintended consequences somewhere between your
asterisk and your ISP. Otherwise, as Alejandro suggests – check with
Vitelity support
On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N
wrote:
> yes I did that, even then i am not able to make outbound and inbound as
> well.
>
>
That's weird. Guess you're gonna have to place a detailed ticket to
them. It sounds like a network problem to me but without any detailed
info it's hard
Thanks for your input.
I failed to mention my setup: Centos 5.8, Asterisk 1.8.11.1, libpri 1.4.12,
DAHDI 2.5.1
I have a rhino r1t4 connected to 2 channel banks (adit 600). Also a digium
B410P for connection to PSTN.
Unfortunately rhino drivers don't compile against DAHDI 2.6.1 so I cannot
test i
Is anybody else experiencing this problem ?
--
Thanks, Phil
- Original Message -
> Hello,
>
> a client attempted to transfer a call today which failed and returned
> the channel back to her. When this happened on the console we saw:
>
> Got OK on REFER Notify message
>
> the version
yes I did that, even then i am not able to make outbound and inbound as
well.
On Thu, May 24, 2012 at 12:42 PM, Alejandro Imass wrote:
> On Thu, May 24, 2012 at 2:01 AM, Gopalakrishnan N
> wrote:
> > Hi Alejandro,
> >
> > I removed the registration and tried as like yours, even inbound calls
>
On Thu, May 24, 2012 at 2:01 AM, Gopalakrishnan N
wrote:
> Hi Alejandro,
>
> I removed the registration and tried as like yours, even inbound calls are
> not landing, anyways let me check with vitelity support.
>
In the Vitel web app you ust set the routing method to the IP of your
pbx, maybe tha
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