Hi.
Thank you.
You mean do each call separately? That works without a glitch, nothing
peculiar.
Thx,
BC
On 09/25/12 23:28, Danny Nicholas wrote:
Do the call both ways again and check(post) the CLI output.
*From:*asterisk-users-boun...@lists.digium.com
On Tuesday 25 September 2012, Matt Hamilton wrote:
Which one (InnoDB or MyISAM) is preferred for CDR as far as write
performance is concerned?
Thanks,
Matt
MyISAM is faster (on Linux anyway); but you'd better have a UPS on the
machine, because it is not very tolerant of unclean
On Mon, Sep 24, 2012 at 02:17:29PM -0700, Steve Edwards wrote:
On Mon, 24 Sep 2012, Asterisk Newb wrote:
Thanks, situated the problem with the following:
exten = _212555.,1,Authenticate(/etc/asterisk/pins||3,j)
exten = _212555.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
Two suggestions:
1)
Dear All,
I want to play music in my AGI while i am searching for a field in DB.
Actually during some processes in AGI i need to play music .
Thanks in advanced.
Regards,
Mehdi
--
_
-- Bandwidth and Colocation Provided by
Hi,
I was trying to register a VoIP trunk in Asterisk , where its keep on
sending Register message to the server, where I am not getting any response
from server.
But whereas if i register in Xlite softphone the account is getting
registered.
I suspect it could be network related issue, but
+1 - if your query is going to take long enough that you need to play music,
you need to optimize the process somewhere. FWIW, if you do play music, you
will need to fork the process as the music process is not asynchronous.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
The Asterisk server and softphone are hitting the firewall from two
different points. Start there.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan
N
Sent: Wednesday, September 26, 2012 7:45 AM
To: Asterisk Users
there is no firewall, its just the router gave by the service provider. May
be the SIP port issue?
Regards.
On Wed, Sep 26, 2012 at 6:17 PM, Danny Nicholas da...@debsinc.com wrote:
The Asterisk server and softphone are hitting the firewall from two
different points. Start there.
** **
Another possibility - you registered from the softphone first and the
provider took the IP address from your PC and locked out the IP address of
your Asterisk server.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan
N
Hi,
How are you connected to server ? How have you configured your asterisk
server to register to other side ? What about any NAT involved in your
scenario ?Turn on sip debug and share your registrations.
BR
Sammy
On Sep 26, 2012 5:54 PM, Danny Nicholas da...@debsinc.com wrote:
Another
But even then all the IP go via router, so when it goes to service provider
it will go as the same IP address, since its coming from the same network.
Because the softphone and asterisk machine are local network which is
commonly connected to a router.
Regards.
On Wed, Sep 26, 2012 at 6:31 PM,
Hi team,
I had setup an asterisk with freepbx and I want to forward the calls to
mobile when nocone is picking up calls in the ringroup. I have already
added custom ext and given string as Local/mobno/from -internal. But now
reciever is geting pilot number only I need to get the callers number in
You have to set the caller ID before dial because it's a new call:
[default]
Exten = s,1,answer
Exten = s,n,Set(CALLERID(num)=${EXTEN})
Exten = s,n,dial.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darin Iv
Sent: Wednesday,
If I remember correctly, INNODB offers row level locking while MyISAM
does not.
On 09/26/2012 05:18 AM, Thorsten Göllner wrote:
Am 26.09.2012 10:45, schrieb A J Stiles:
On Tuesday 25 September 2012, Matt Hamilton wrote:
Which one (InnoDB or MyISAM) is preferred for CDR as far as write
Hello,
I'm having issues connecting throu PRI with the following error Requested
transfer capability: 0x00 - SPEECH
Below are the logs:
== Using SIP RTP CoS mark 5
-- Executing [97052660@voipphones:1] Set(SIP/4856-0003,
CALLERID(num)=x) in new stack
-- Executing
On 12-09-26 10:35 AM, motty.cruz wrote:
Hello,
I'm having issues connecting throu PRI with the following error Requested
transfer capability: 0x00 - SPEECH
Below are the logs:
== Using SIP RTP CoS mark 5
-- Executing [97052660@voipphones:1] Set(SIP/4856-0003,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent: Wednesday, September 26, 2012 7:52 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8.15.0, Requested
You are set up as a USA PRI, but not dialing a USA TN.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz
Sent: Wednesday, September 26, 2012 11:13 AM
To: 'Asterisk Users Mailing List - Non-Commercial
On 12-09-26 11:12 AM, motty.cruz wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent: Wednesday, September 26, 2012 7:52 AM
To: asterisk-users@lists.digium.com
Subject: Re:
Hi all,
does someone knows what happen with voipuser.org web site and services?
Registration failed since more than 24 hours and no access to the web
site :-(
Regards
--
Daniel
--
_
-- Bandwidth and Colocation Provided by
You need to modify your dialplan to change 9xxx to 1aaaxxx. I think
most U.S. SIP providers want a 10 digit number.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent: Wednesday,
Hi All,
Someone can tell me if asterisk support the SIP History-Info?
If it supports, how can enable it?
I searched on Google, but I could not find anything...
Thanks for all
Best Regards
MC
http://www.connectmi.it/?utm_source=email-signatureutm_medium=emailutm_campaign=Email%2Bsignature
That may depend on the flavor of Asterisk you are using and whether you are
using flat or realtime log files.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Colombo
Sent: Wednesday, September 26, 2012 10:33 AM
To:
Hi,
Thanks for reply
What do you mean with Using flat or Realtime log files?
I need this line in the SIP Invite :
History-Info:
sip:+3906330xx...@enter.it;user=phone;cause=302;privacy=history;index=1
History-Info: sip:+3906330X@enterSIP/2.0 100 Trying
how can I provide the data that you
Versions 1.8 and 11 (probably 10 as well) let you query SIP information. 1.2
and 1.4 (1.6 also I think) do not. If you are in a small environment, you
can turn on SIP debug and put that in a separate log (would eat up the disk
in a few days in most real environments).
From:
Hello.
I have asterisk 1.8.18 which connects to ISDN PRI. All phones are sip,
Aastra 6757i. Everything works as expected.
We also have a FAX machine. We need to be able to use that FAX machine to
send or receive faxes. We are planning to have a dedicated did for faxes.
Before, FAX machine was
Am 26.09.2012 17:53, schrieb Mark Robinson:
I have asterisk 1.8.18 which connects to ISDN PRI. All phones are sip,
Aastra 6757i. Everything works as expected.
We also have a FAX machine. We need to be able to use that FAX machine
to send or receive faxes. We are planning to have a dedicated did
On Wednesday 26 September 2012, Darin Iv wrote:
Hi team,
I had setup an asterisk with freepbx and I want to forward the calls to
mobile when nocone is picking up calls in the ringroup. I have already
added custom ext and given string as Local/mobno/from -internal. But now
reciever is geting
Hi,
On my invite trace I don't have history-info.
Could you explain me how do I put history-info on SIP INVITE?
-- Executing [+39@trunk-squire-incoming:1]
Dial(SIP/trunk-squire-outcoming-0045, SIP/) in new stack
== Using SIP RTP CoS mark 5
Audio
Marco Colombo wrote:
Hi,
Hola,
On my invite trace I don’t have history-info.
Could you explain me how do I put “history-info” on SIP INVITE?
You can't. That specific RFC (4244) is not implemented within chan_sip.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan
Is there a way to have Asterisk respond appropriately when receiving a DTMF
Flash event via SIP? I'm finding some WiFi SIP phones, specifically the
Quickphones QA-342 want to handle transfers/3-way calls by sending a DTMF Flash
event instead of handling it properly like every other damn VoIP
On 09/26/2012 05:53 PM, Mark Robinson wrote:
Hello.
I have asterisk 1.8.18 which connects to ISDN PRI. All phones are sip,
Aastra 6757i. Everything works as expected.
We also have a FAX machine. We need to be able to use that FAX machine
to send or receive faxes. We are planning to have a
On Wed, 26 Sep 2012, Mehdi Rahimi wrote:
I want to play music in my AGI while i am searching for a field in DB.
Actually during some processes in AGI i need to play music .
Many moons ago, I had a client that wanted 'instantaneous' response to a
credit card authorization request. (The
On 26/09/12 05:35 AM, Mehdi Rahimi wrote:
I want to play music in my AGI while i am searching for a field in DB.
Actually during some processes in AGI i need to play music .
Probably Local channels to the rescue here.
--
Leif Madsen
http://www.oreilly.com/catalog/asterisk
--
Asterisk 1.8.10.1~dfsg-1ubuntu1
Trying to build a simple announcement of the queue status. QUEUEHOLDTIME
is always zero. What am I doing wrong?
queues.conf
[general]
autofill=yes
shared_lastcall=yes
[StandardQueue](!)
musicclass=default
strategy=rrmemory
joinempty=no
leavewhenempty=yes
hi
Thanks for replay. now asterisk accepts calls. But 32 second later, calls
drop.
Error code:
-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
On Wed, Sep 26, 2012 at 2:47 PM, Carlos Rojas crt.ro...@gmail.com wrote:
Hello
Yes, there is, in sip.conf you
Has anyone had experience using a SIP trunk provided by Paetec over MPLS?
With or without FreePBX
Regards,
Jared Baxley
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
Hello,
I have set the default_linemode=e1 even then it does not work. The pri show
spans show as down.
I added the below line in /etc/modeprobe.d/dahdi.conf
options wct4xxp default_linemode=e1
I also removed dahdi and asterisk complete and reinstalled the entire
package but with now success.
38 matches
Mail list logo
