Hi Alejandro,
I removed the registration and tried as like yours, even inbound calls are
not landing, anyways let me check with vitelity support.
Hi Stephan,
I am not using any SBC. As i said let me check with their support.
Thanks for all the views comments.
Regards,
On Wed, May 23, 2012
On Thu, May 24, 2012 at 2:01 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com 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
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 a...@p2ee.org wrote:
On Thu, May 24, 2012 at 2:01 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Hi Alejandro,
I removed the registration and tried as
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 that we
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
On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com 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
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
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
- 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
calls
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
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.38
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
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 10
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
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 kpflem...@digium.com 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
I'm running on 1.8 as of now
On May 24, 2012 11:00 AM, Kevin P. Fleming kpflem...@digium.com 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
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
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 luck?
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 asterisk_l...@earthshod.co.uk
wrote:
On Thursday 24 May 2012, Cody Harris wrote:
I'm trying to implement a fax/voice switch. I have faxdetect=both
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
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:
Hi, I'm using AMI to get the extension status but always get -1 i.e. extension
not found. #!/usr/bin/php -q
?phpinclude_once (phpagi-2.14/phpagi.php);
include_once (/phpagi-2.14/phpagi-asmanager.php);
$agi = new AGI();
$as = new AGI_AsteriskManager();
$exten =
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
--
- Original Message -
From: Jayesh Labade jayesh.lab...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, May 24, 2012 3:10:29 PM
Subject: [asterisk-users] Asterisk MixMonitor starts recording 44 bytes file
Hello
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
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
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/5-1
- Original Message -
From: Jayesh Labade jayesh.lab...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, May 24, 2012 4:10:29 PM
Subject: [asterisk-users] Asterisk MixMonitor starts recording 44
bytes file
Hello
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 kamlesh_...@hotmail.com wrote:
Hi,
I'm using AMI to get the extension status but always get -1 i.e. extension
not found.
#!/usr/bin/php -q
?php
Looks like Swift() (whatever that is) is not returning ?
On 24 May 2012 23:07, Justin Killen jkil...@allamericanasphalt.com wrote:
** ** **
Here is the output from the cli:
** **
dozer*CLI core show channels
Channel Location State Application(Data)
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. Alejandro
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
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