Hello,
I've some questions concerning ldap.
1. I think ldap attributes doesn't have a certain order, at least, all
ldap editors that I use can't define an order. But it seems that the
attribute order is relevant for entries like AstAccountDisallowedCodec
and AstAccountAllowedCodec.
2. Is
Hello,
We see some strange behavior with phone calls, we use Asterisk 1.8.3.3.
SIP clients (all behind NAT at different locations, so not a single NAT
solution is used):
- x-lite
- linksys pap2t
- polycom kirk (multiple type numbers)
- polycom (multiple type numbers, hardware phones)
Our
On Thursday 26 May 2011, virendra bhati wrote:
How to make outgoing calls from DID and what is theway to get incoming
calls from DID.
First of all, get your dialplan and zaptel configuration working to the extent
as you can make SIP to SIP calls between extensions, and you can make
outgoing
On Fri, 2011-05-27 at 10:31 +0200, Mark Scholten wrote:
Hello,
We see some strange behavior with phone calls, we use Asterisk 1.8.3.3.
SIP clients (all behind NAT at different locations, so not a single NAT
solution is used):
- x-lite
- linksys pap2t
- polycom kirk (multiple type
On 26/05/11 23:18, Satish Patel wrote:
Thanks,
I went through this example before. I was confuse and wondering how
should I add third queue in this picture?
From the example:
*CLI database put queue_agent 0001/available_queues support^sales
support^sales is a list of queues. Put
i have installed asterisk and i have 3 sip 104 ,105 and 106
Now I can make the calls with theses sip without issue
I want to configure the outbound calls for these sips like that:
104 permission to call any number, but for 105 and 106 I want to specify
some numbers to call
Any help
you need to make dial plan .
On Fri, May 27, 2011 at 2:59 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
i have installed asterisk and i have 3 sip 104 ,105 and 106
Now I can make the calls with theses sip without issue
I want to configure the outbound calls for these sips
thanks for reply i have a dial plan but can you please give me an exemple
regrads
2011/5/27 mahesh katta maheshka...@flexydial.com
you need to make dial plan .
On Fri, May 27, 2011 at 2:59 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
i have installed asterisk and i have
Route 104 into one dialplan context and 105 106 into a more restrictive one.
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
On May 27, 2011, at 5:29 AM, salaheddine
On Friday 27 May 2011, salaheddine elharit wrote:
i have installed asterisk and i have 3 sip 104 ,105 and 106
Now I can make the calls with theses sip without issue
I want to configure the outbound calls for these sips like that:
104 permission to call any number, but for 105 and 106 I want
104 extension should call all outgoing calls, for example you can give one
particular context for104 ,EX: ALL is a context
sip.conf
[104]
username=104
secret=123
nat=yes
canreinvite=yes
context=ALL
extensions.conf
[ALL]
_0X.,1,Dial(zap/go,20,tTo)
On Fri, May 27, 2011 at 3:15 PM, Alex Balashov
Hi
I've spent two days trying to solve this issue but to no prevail and I'm
hoping to get some help.
I've configured Asterisk as a SIP client, running on OpenWRT on an embedded
device with onboard FXS and ATA. Asterisk is connecting to an external SIP
provider on the Internet who in turn
A.J.Stiles given perfect example .
On Fri, May 27, 2011 at 3:26 PM, mahesh katta maheshka...@flexydial.comwrote:
104 extension should call all outgoing calls, for example you can give one
particular context for104 ,EX: ALL is a context
sip.conf
[104]
username=104
secret=123
nat=yes
Dear sir,
Please have you any document of Redfone or links. I need to learn this about
redfone. i am totally confusing in this.
--
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross
hi:
please refer this:
http://support.red-fone.com/index.php?_m=knowledgebase_a=viewarticlekbarticleid=20
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
Date: Fri, 27 May 2011 15:58:52 +0530
From:
That's cool. I will give it a shot and let you guys know.
--
Sent from my iPhone
On May 27, 2011, at 5:18 AM, Paul Hayes p...@provu.co.uk wrote:
On 26/05/11 23:18, Satish Patel wrote:
Thanks,
I went through this example before. I was confuse and wondering how
should I add third queue in
OK. Im trying to setup voicemail on ODBC.
My objective is to create some relation in voicemail_data and cdr table
based on uniqueid.
--
regards,
Abdul Basit
On Fri, May 27, 2011 at 12:12 AM, vip killa vipki...@gmail.com wrote:
try using voicemail_odbc
On Thu, May 26, 2011 at 2:19 PM,
Hi,
core show function CHANNEL mentions Additional items may be available from
the channel driver providing
the channel; see its documentation for details.
Where can you find DAHDI-related info ?
More specifically I would to know within dialplan, which Dahdi (trunk) group
current dahdi channel
thank you for your response now i can do that without any issue ,i have
just one question
when i verify after this solution all the calls now boot from g1
before i have this
exten = _0612.,1,Set(CALLERID(number)=520460587)
exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
Hi There,
We have single PRI with multiple DID numbers and its working fine in receiving
call. And if you make outbound call it will send main-line CallerID (company
name). Now we want individual caller id for per extensions on outbound calls.
like if i call someone he will get my extension
Add Set(CALLERID(num)=617838${CALLERID(num)}) to your dialplan for outgoing
calls.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
satish patel
Sent: Friday, May 27, 2011 10:42 AM
To: asterisk-users
That is very cool,
Is that means it will overwrite my global callerid setting at dahdi-channels?
root@sfpbx1:/home/satish# cat /etc/asterisk/dahdi-channels.conf | grep callerid
callerid=6178387100
-S
From: ewiel...@nyigc.com
To: asterisk-users@lists.digium.com
Date: Fri, 27 May 2011
I'm looking for recommendations for standalond PRI to SIP converters. (Needs
to be outside the asterisk box - so a PCIe card won't do)
I've used redfone but this project doesn't need the redundancy features...
Thanks!
--
_
--
Yes, but only for that call. You should not generally set the callerid= for
PRI channels.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
satish patel
Sent: Friday, May 27, 2011 11:12 AM
To:
On 05/27/2011 05:10 PM, Michelle Dupuis wrote:
I'm looking for recommendations for standalond PRI to SIP converters. (Needs
to be outside the asterisk box - so a PCIe card won't do)
I've used redfone but this project doesn't need the redundancy features...
Have a look at Patton or
On 27/05/11 16:10, Michelle Dupuis wrote:
I'm looking for recommendations for standalond PRI to SIP converters. (Needs
to be outside the asterisk box - so a PCIe card won't do)
I've used redfone but this project doesn't need the redundancy features...
Thanks!
A 2nd Asterisk box with a PCIe
Op 27-05-11 17:10, Michelle Dupuis schreef:
I'm looking for recommendations for standalond PRI to SIP converters. (Needs
to be outside the asterisk box - so a PCIe card won't do)
I've used redfone but this project doesn't need the redundancy features...
Thanks!
--
Is there a way to disable all SIP registration and block any requests? The
reason I'm asking is this particular Asterisk server will just be
originating calls. I've noticed sip attacks where the attacker attempts to
register a user 100x per second causing CPU to rise significantly.
--
On Friday 27 May 2011, satish patel wrote:
Hi There,
We have single PRI with multiple DID numbers and its working fine in
receiving call. And if you make outbound call it will send main-line
CallerID (company name). Now we want individual caller id for per
extensions on outbound calls. like
On Fri, 27 May 2011, Patrick Lists wrote:
On 05/27/2011 05:10 PM, Michelle Dupuis wrote:
I'm looking for recommendations for standalond PRI to SIP converters.
(Needs to be outside the asterisk box - so a PCIe card won't do)
I've used redfone but this project doesn't need the redundancy
Hi There,
We have very old asterisk 1.2 running in production and it has following
setting in /etc/zaptel.conf. I have read on web about span and they told
span= span num ,timing source,line build out
(LBO),framing,coding[,yellow]
Just wondering why it has timing source 0 ? 0=master,
From: Patrick Lists
Sent: Fri 5/27/2011 11:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] standalone PRI-to-SIP converter
On 05/27/2011 05:10 PM, Michelle Dupuis wrote:
I'm looking for recommendations for standalond PRI to SIP converters.
Block inbound udp port 5060 using your firewall?
Thanks,
--Warren Selby, dCAP
On May 27, 2011, at 10:45 AM, vip killa vipki...@gmail.com wrote:
Is there a way to disable all SIP registration and block any requests? The
reason I'm asking is this particular Asterisk server will just be
On Fri, 27 May 2011, vip killa wrote:
Is there a way to disable all SIP registration and block any requests?
The reason I'm asking is this particular Asterisk server will just be
originating calls. I've noticed sip attacks where the attacker attempts
to register a user 100x per second causing
Hello All,
I'm installing asterisk 1.6 on FreeBSD from ports; I'm not sure what options
should install; can anybody points to good howto on FreeBSD, I defenetely
appreciate! There are a log info for Linux but very little for FreeBSD.
Thanks,
-motty
--
Hi Guys,
in next week i plan to upgrade my Asterisk Server. To buy the optimal Hardware
i have a question.
What is better more cores (eg. 2x quadcore) or more CPU speed for a server that
handle a lot of of Meetme Concerences with hundreds of concurrent G711 alaw
Channels (no transcoding) ?
This is working great! Thanks a lot paul.
One more question before we have Agent/ configured in queueMetrics so i
need to change them in queueMetrics with SIP/ right ?
Date: Fri, 27 May 2011 10:18:39 +0100
From: p...@provu.co.uk
To: asterisk-users@lists.digium.com
Subject: Re:
On Fri, May 27, 2011 at 05:30:02PM +, dan...@danielknoll.de wrote:
What is better more cores (eg. 2x quadcore) or more CPU speed for a server
that handle a lot of of Meetme Concerences with hundreds of concurrent G711
alaw Channels (no transcoding) ?
in my opinion, more cores are
From: dan...@danielknoll.de
Sent: Friday, May 27, 2011 1:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] More Cores or more CPU Speed
Hi Guys,
in next week i plan to upgrade my Asterisk Server. To buy the optimal
Hardware i have a
Oh! wait i got following error when i trying to Unpause my queue. do you have
any idea ?
holler*CLI
== Using SIP RTP CoS mark 5
-- Executing [*99@from-sip:1] Verbose(SIP/7102-000e, 2,UnPausing
member in all queues) in new stack
== UnPausing member in all queues
-- Executing
In this book example there is a printing issue at Unpaused section. it should
be like following
same = n,GotoIf($[${UPQMSTATUS} = UNPAUSED]?agent_unpaused,1:agent_not_found,1)
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 27 May 2011 18:41:18 +
Subject: Re:
On 27/05/11 03:18 PM, satish patel wrote:
In this book example there is a printing issue at Unpaused section. it
should be like following
same = n,GotoIf($[${UPQMSTATUS} =
UNPAUSED]?agent_unpaused,1:agent_not_found,1)
Please file stuff like this as errata at
This has been submitted.
-S
Date: Fri, 27 May 2011 16:05:28 -0400
From: leif.mad...@asteriskdocs.org
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] [SOLVED] Asterisk 1..8 multiple queue
On 27/05/11 03:18 PM, satish patel wrote:
In this book example there is a
Hi
The timing source is the clock of the system. When a equipment is 0, the
other should be 1. The correct is: 0=slave, 1=master. The default for
private systems is slave.
Att,
Rafael Saraiva
2011/5/27 satish patel satish...@hotmail.com
Hi There,
We have very old asterisk 1.2 running in
You mean say
0=Slave (Use PSTN clock)
1=Master(generate Internal clock)
So best option is 0 for all span if you connected on PSTN right ?
Date: Fri, 27 May 2011 17:27:43 -0300
From: rafaels...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DAHDI span timeing
On Fri, May 27, 2011 at 08:57:15PM +, satish patel wrote:
You mean say
0=Slave (Use PSTN clock)
1=Master(generate Internal clock)
So best option is 0 for all span if you connected on PSTN right ?
Not really. Looking in system.conf.sample in dahdi-tools [1]
Choose 1 to make the
Tell me in one word. We have 2 PRI line connected with sangoma card what option
would be good for me?
0 or 1 ?
-S
Date: Fri, 27 May 2011 16:11:03 -0500
From: sruff...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DAHDI span timeing source
On Fri, May 27,
On Fri, May 27, 2011 at 09:20:46PM +, satish patel wrote:
Tell me in one word. We have 2 PRI line connected with sangoma card what
option would be good for me?
0 or 1 ?
Look at the two last sentences of the first paragraph I quoted below. I
believe that is your answer...and it's not 0
Got it but still confused. As per your example I should go with
Port 1
Span=1,1,0
Port 2
Span=2,2,0
Correct me if I'm wrong.
--
Sent from my iPhone
On May 27, 2011, at 5:32 PM, Shaun Ruffell sruff...@digium.com wrote:
On Fri, May 27, 2011 at 09:20:46PM +, satish patel wrote:
Tell me
On 5/27/11 2:20 PM, satish patel wrote:
Tell me in one word. We have 2 PRI line connected with sangoma card what option
would be good for me?
0 or 1 ?
that would depends on what's the other end of the 2 PRI connected to.
Date: Fri, 27 May 2011 16:11:03 -0500
From: sruff...@digium.com
On Fri, May 27, 2011 at 05:40:30PM -0400, Satish Patel wrote:
Got it but still confused. As per your example I should go with
Port 1
Span=1,1,0
Port 2
Span=2,2,0
Correct me if I'm wrong.
Yes. That looks correct based on my understanding of your situation.
--
Shaun Ruffell
Digium, Inc.
It's connected to teclo ATT PSTN for outside calling.
So definitly they are master and we are slave but I'm confused about
0 is master or slave? Because few people saying 1 is master and 0 is
slave ? I didn't find any clear document every one trying to explain
science but none of clear.
Thanks also let me clear one thing this pri is PSTN connected to ATT
techo.
So they are master.
--
Sent from my iPhone
On May 27, 2011, at 5:51 PM, Shaun Ruffell sruff...@digium.com wrote:
On Fri, May 27, 2011 at 05:40:30PM -0400, Satish Patel wrote:
Got it but still confused. As per your
I guess you are wrong here correct one is 0=master 1=slave
If you connect to PSTN the you should user span=1,1,0
Check out http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html
--
Sent from my iPhone
On May 27, 2011, at 4:27 PM, Rafael dos Santos Saraiva rafaels...@gmail.com
wrote:
Really, You're right. This option define the priority of the interface as
regenerator of clock:
priority 0 = its own clock
priority 1 = the clock of the telco
2011/5/27 Satish Patel satish...@hotmail.com
I guess you are wrong here correct one is 0=master 1=slave
If you connect to PSTN the
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