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
autofill=
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
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 enteri
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? Googli
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 yo
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
> -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 i
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 firewall-
>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 failove
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
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 th
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
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. I
Put the commonly used domain names + appropriate ips into /etc/hosts?
John
2010/1/4 Steve Howes :
>
> 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 eno
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
>
> 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 d
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 afte
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
>> 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 diffe
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 indep
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.
Bot
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,
>>> fo
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 t
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 w
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. Som
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 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 li
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 re
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?
>
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 [sample offset]
Send the given file, allowing play
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 second
On Mon, Jan 4, 2010 at 1:46 PM, Kevin P. Fleming wrote:
> 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 aud
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 se
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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