On Mon, 4 Jan 2010, Neeraj Chand wrote:
I currently run small scale mysql queries from the dialplan
[snip]
This currently takes about 4 seconds to complete.
If I run two simultaneous queries, this goes up to about 9 seconds for
both queries to complete.
4 seconds for a query is a bit
Is there any fix or workaround for the DNS problem (old standing bug that
when the box starts and domain names do not resolve quickly enough from
DNS then asterisk stops using the outgoing trunks.
I read on the list before that it is considered a huge and unacceptable
load for asterisk servers
Hello list !
I have some configuration issues with queues, but I'm sure they are
minor and for someone who has already configured queues it could be
trivial.
This is my queue configuration :
[VC_support_queue]
musicclass = default
strategy = ringall
timeout = 20
retry = 5
wrapuptime=15
On 30 Dec 2009, at 19:43, vijay.go...@alliance-infotech.com wrote:
Hi Sir,
We have integrated Skype with Asterisk (skype user id:- rexesbposolutions).
Each call which is coming to skype account is getting transfered to Asterisk
Queue. It has following two cases:
case 1: When we call
Looking at the source in the rpms from the asterisk package site
appears that oslec is not built and enabled for the kmod rpms.
Anyone know an existing repo or have direction on how to enable
this to built for those rpms?
Thanks,
jlc
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Hi,
So I'm using Asterisk Realtime Queues and Queue members on 1.4.28.
I've noticed if there are no people in the queue when a call enters,
even after a queue member enters, the call is never rang to him.
From the debug, it seems that Asterisk is only grabbing the queue
member list upon
Hi,
I'm trying to get ZapRAS working but not getting very far..
Asterisk CLI shows:
WARNING[3355]: app_zapras.c:173 run_ras: wait4 returned -1: No child processes
and /var/log/messages shows:
using the plugin option requires root privilege
Can anyone shed any light on this and any fix?
hadi motamedi wrote:
Sorry . I didn't get the point clearly . In the SIP Invite message , it
says my audio endpoint is IP x.x.x.x port x, and I can use codecs
A,B,C. The remote endpoint responds with a 200 OK, saying my audio
stream is at IP y.y.y.y port y, and I choose codec B. Can you
Will Payne wrote:
I'm trying to get ZapRAS working but not getting very far..
Asterisk CLI shows:
WARNING[3355]: app_zapras.c:173 run_ras: wait4 returned -1: No child processes
and /var/log/messages shows:
using the plugin option requires root privilege
Can anyone shed any light on
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Neeraj Chand
Sent: Monday, January 04, 2010 1:17 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MYSQL queries from dial plan
[mysql
On 4 Jan 2010, at 08:34, Remco Barendse wrote:
Is there any fix or workaround for the DNS problem (old standing bug
that
when the box starts and domain names do not resolve quickly enough
from
DNS then asterisk stops using the outgoing trunks.
I read on the list before that it is
I have recently moved our asterisk server from our LAN to a Debian
Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our
network. Our phones are behind a natted firewall. An ITSP provides a
PSTN to SIP termination for incoming calls
Public ITSP --Asterisk server--Natted
From reading the documentation that came with dahdi-tools I gathered that the
second span should be span=2,2... designating it as a backup timing source
in case your primary span=1,1... should die. I am not sure that you can
designate two primary timing sources and have seamless failover in a
On 4 Jan 2010, at 13:48, Kevin P. Fleming wrote:
ZapRAS forks off to pppd to handle the PPP session, it does not
implement PPP itself. You will have to be running Asterisk as root for
this to work, or provide a wrapper for pppd that ZapRAS can execute with
the suid bit set so that pppd runs
Will Payne wrote:
I'm looking to periodically nudge Asterisk into making an ISDN
connection, setting up PPP and then (possibly by then starting an AGI
script) grabbing a file by FTP over the PPP link.
If I'm overcomplicating it or going about it completely the wrong way, a
point in the
On 4 Jan 2010, at 16:28, Kevin P. Fleming wrote:
Will Payne wrote:
I'm looking to periodically nudge Asterisk into making an ISDN
connection, setting up PPP and then (possibly by then starting an AGI
script) grabbing a file by FTP over the PPP link.
If I'm overcomplicating it or going
Hello folks.
I'm looking into having a web page displaying asterisk callers.
We are a call centre, and having operators answering calls at home or
whatever, they would need to have a real time application to display how
manny callers are queuing, for how long etc.
At first, I thought of phpagi.
Put the commonly used domain names + appropriate ips into /etc/hosts?
John
2010/1/4 Steve Howes steve-li...@geekinter.net:
On 4 Jan 2010, at 08:34, Remco Barendse wrote:
Is there any fix or workaround for the DNS problem (old standing bug
that
when the box starts and domain names do not
On 4 Jan 2010, at 16:46, Tiago Geada wrote:
Hello folks.
I'm looking into having a web page displaying asterisk callers.
We are a call centre, and having operators answering calls at home or
whatever, they would need to have a real time application to display how
manny callers are
2010/1/4 Will Payne w...@teambadger.co.uk
On 4 Jan 2010, at 16:46, Tiago Geada wrote:
Hello folks.
I'm looking into having a web page displaying asterisk callers.
We are a call centre, and having operators answering calls at home or
whatever, they would need to have a real time
I have posted my problem on the link below, but didn't get any answer. I am
hoping someone here can help me with this issue. Here's my problem:
I am using H323 to talk between Asterisk and Avaya IP Office 500. For
some strange reason, when we are talking on a VoIP call, we get
disconnected
How to register/configure Sip accounts to register per gateway?
All the accounts I have are registered individually but with mix (FXO/FXS)
AudioCodes MP-114 this does not work, I have to have FXO port registered per
gateway (not individually).
--
Joseph
Hello,
I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time
On Sun, 3 Jan 2010, Olle E. Johansson wrote:
No, Asterisk only supports one port.
You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple
addresses and ports and forward to Asterisk on the same or
1 jan 2010 kl. 20.04 skrev Shariq Khan:
I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time
On Sun, 3 Jan 2010, Steve Edwards wrote:
You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple
addresses and ports and forward to Asterisk on the same or different
Could you explain this one a bit more...
You run openSER on the same box as asterisk, and have multiple such boxes,
with the purpose of failover? But if a box goes down with openser on it,
then there is no forwarding. (And most phones can only reg with peer). If
you move openSER to another
I am implementing one dialer type of application.
In which i am first dialing one source number and sending it to
conference and then starting dialing the different destination numbers.
i have used meetme application of asterisk for this as i dont want to
disconnect the main source number.
Un-top-posting...
On Sun, 3 Jan 2010, Steve Edwards wrote:
You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple
addresses and ports and forward to Asterisk on the same or different
boxes.
I like to configure systems with OpenSER running on each box,
forwarding calls to
Robert Broyles wrote:
Hi,
So I'm using Asterisk Realtime Queues and Queue members on 1.4.28.
I've noticed if there are no people in the queue when a call enters,
even after a queue member enters, the call is never rang to him.
From the debug, it seems that Asterisk is only grabbing the
Thank you Doug and Nguyen. I have had your recommendation but I still can
not get SIP inbound for my broadvoice line to work. My configuration and SIP
debugs are attached and just to recap I have done the following:
My outgoing works great. I can dial from one SIP extension to another
internally
Is there a way to not compile in lpc10 support using the ./configure
command?
./configure --disable-lpc10 or something like that?
if that is not available whats the easiest way to remove lpc10 support
with out doing the make menuselect. I want to do it automatically at
install not have to enter
Hi guys,
Am having a strange SIP problem in my call centre. The call centre has about 70
SIP agents (some of the are using SIP hard phones, other SIP softphones), and
occasionally most of the SIP peers (hardphones and softphones) become
UNREACHABLE and then after few second again REACHABLE.
Hello,
1 jan 2010 kl. 20.04 skrev Shariq Khan:
I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time
On Sun, 3 Jan 2010, Steve Edwards wrote:
You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple
addresses and ports and forward to Asterisk on the same or
On 29/12/09 10:22 AM, Leif Neland wrote:
I want some cheap ip-phones with auto-answer, to work as paging system
at dinnertime.
Options, please.
Use some of the Chinese PA1688 or AR1688 phones - support auto answer,
IAX/SIP etc.
Prices around $45
--
Cheers,
Matt Riddell
Managing Director
Has anyone found an ITSP that will relay T.38 fax to an asterisk 1.6.x
instance AND do it reliably? If so, I can think of a number of locations
with copper loops that could be scrapped. I'm actually quite surprised at
what an underwhelming number of ITSP's that say they support T.38 (zero so
On Sun, 3 Jan 2010, Steve Edwards wrote:
You can configure OpenSER/Kamailio/OpenSIPS to listen to multiple
addresses and ports and forward to Asterisk on the same or
different boxes.
On Mon, 4 Jan 2010, Vikram Ragukumar wrote:
Would it be more efficient to use libnetfilter_queue() to
Hi,
This is a naive question, but is there a way in my AGI script to
simultaneously play audio and listen for DTMF or voice responses?
I've heard VOIP hackers call this inbargeability; it's the ability
to barge in to a playing audio clip.
I'm planning to use Lumenvox for the DTMF and voice
Have you tried something like qualify=10 ?
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Sure, as long as you use whatever is equivalent to the Background()
dial plan app, or Background() itself.
On 01/04/2010 08:41 PM, Quinn Weaver wrote:
Hi,
This is a naive question, but is there a way in my AGI script to
simultaneously play audio and listen for DTMF or voice responses?
I've
On Mon, 4 Jan 2010, Quinn Weaver wrote:
This is a naive question, but is there a way in my AGI script to
simultaneously play audio and listen for DTMF or voice responses?
t2:vtpv:18:04:59 show agi stream file
Usage: STREAM FILE filename escape digits [sample offset]
Send the given
On Jan 4, 2010, at 3:06 PM, Steve Edwards wrote:
On Mon, 4 Jan 2010, Neeraj Chand wrote:
I currently run small scale mysql queries from the dialplan
[snip]
This currently takes about 4 seconds to complete.
If I run two simultaneous queries, this goes up to about 9 seconds for
both
On Mon, Jan 4, 2010 at 1:46 PM, Kevin P. Fleming kpflem...@digium.comwrote:
hadi motamedi wrote:
Sorry . I didn't get the point clearly . In the SIP Invite message , it
says my audio endpoint is IP x.x.x.x port x, and I can use codecs
A,B,C. The remote endpoint responds with a 200 OK,
Dear All
Further to my previous inquiry regarding Asterisk sending dialed digits in
one-by-one digit format when we had ISDN PRI link with the PSTN switch , you
told me that we are expected to enable overlap dialing . At now , we have
the same configuration but sip connection to the external sip
Hi ,
Can any one tell me that how to automatically dial a list of numbers
from database .I have seen a methodology in the post but am not clear vth
that.
Thanks
Pinky
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