Hello
I need to add sip extensions from my UI so without going through sip.conf so
i created table
CREATE TABLE `sipfriends` (
`name` varchar(40) NOT NULL default '',
`username` varchar(40) default '',
`secret` varchar(40) NOT NULL default '',
`context` varchar(40) NOT NULL default '',
hi, all
thanks for reply,
but actually i have configured sip to realtime and i got this message
"SIP Seeding peer from *astdb*: 'sip_ext' at sip_...@asterisk_ip:5060 for
60"
so i have to know that my sip ext is stored in astdb or not.
any other suggetion ?
Regards,
On Wed, Jan 27, 2010 at 4:37
Thanks for the reply jamie :-)
Does ordinary EPBXs in US have those ports or do you need special EPBXs?
--Siju
On Wed, Jan 27, 2010 at 8:32 PM, Jamie A. Stapleton
wrote:
> In this case, a SIP provider would not be required.
>
> Obviously, you will need ports on your EPBX to connect the Digium c
Just wondering if there are any Linux-based hard phones out there -- if
so, it'd be neat to see if I couldn't take advantage of the underlying OS.
Thanks,
-Ken
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On Wed, Jan 27, 2010 at 7:09 AM, Karl Fife wrote:
>>> On Mon, Jan 25, 2010 at 12:07:55PM -0600, Karl Fife wrote:
>>> > From: "cb" Sent: Sunday, January 24, 2010 12:42
>>> > I use the Snom 3
2010/1/28 Carlos Chavez :
> On Wed, 2010-01-27 at 10:27 +0800, Zhang Shukun wrote:
>> hi,all
>>
>> i have just upgrade from 1.4.28 to 1.6.2.1. all works fine now except
>> realtime queue.
>>
>> it seems queue_table works fine, but queue_member_queue not work, the
>> two tables works fine when in 1.
Hi Guys,
I have tested and isntalled Asterisk 1.6.2 with FreePBX from Digium repos
based on this url:
http://www.asterisk.org/downloads/yum
BUT that doesn't seem to work with Fedora instance which I am running on
Amazon Ec2. Apparently Asterisk 1.4 is natively included in Fedora
repository but n
Alec Davis wrote:
> Did you get this resolved? And how if you did.
> We've been have the same random PRI lockup issue for years now.
>
Really?
We have a 1.4.x box hooked up directly to our Definity G3R via a PRI and
a TN464F. I have yet to experience any PRI issues (That I'm aware of)
Doug
On Wednesday 27 January 2010 15:18:41 thorsten.stoffre...@gmx.de wrote:
> Hi,
>
> im a student and we are devloping a training sytem for
> radio operators (for ships, police, ...) at our university.
> So far we are using a simple own protocol for speech and data
> transmission, works well at a Lan.
On Wednesday 27 January 2010 12:55:18 David Gibbons wrote:
>
> many people around think mysql is not a good option for database, they
> think mysql
>
> is only suit for small business. but i want to have a try. i need to
> convince them to use this.
>
>
> This statement is absolute BS. Give me so
Hi Kevin
Kevin P. Fleming a écrit :
> [...]
> This conversation brings to mind two possible ways we could improve
> Asterisk to help users from falling into this trap:
>
> 1) When a sip.conf entry is defined as 'type=friend' *and* has a
> specific host IP address (not dynamic), we could just ignor
Definity what? G3? I did that once, a real pain but doable. I don't
remember the settings but if I had a terminal in front of me, I am
sure I could get it work.
Thanks,
Steve T
On Wed, Jan 27, 2010 at 5:42 PM, C F wrote:
> We didn't fix it yet. For the moment the Definity is not connected
> d
On Thu, Jan 28, 2010 at 12:03 AM, Danny Nicholas wrote:
> I wonder how this would work for you?
>
> - exten => 1000,1,ForkCDR
>
> - exten => 1000,2,Queue(blah)
>
> - exten => 1000,3,Hangup
>
>
>
> This should do 2 CDR’s for each queued call. CDR 1 would be the DAHDI to
> Queue time, CDR 2 queue
I wonder how this would work for you?
- exten => 1000,1,ForkCDR
- exten => 1000,2,Queue(blah)
- exten => 1000,3,Hangup
This should do 2 CDRs for each queued call. CDR 1 would be the DAHDI to
Queue time, CDR 2 queue to hangup.
_
From: asterisk-users-boun...@lists.digium.com
You'd need RTP ports open for asterisk then.
Transfers and parking can be done at the SIP level, asterisk doesn't
have to be in the RTP path, as it can reinvite itself into the
callpath as necessary.
On Wed, Jan 27, 2010 at 5:23 AM, Vincent wrote:
> Hello
>
> I think I finally understood the iss
Can you link the howto or other documentation you are following to set this up?
What version of asterisk?
Did you edit extconfig.conf?
Heres a howto for 1.4.x
http://hostseries.com/asterisk-realtime-installation-guide/
On Wed, Jan 27, 2010 at 8:39 AM, ahmed magdy wrote:
> Hello
>
> I need to add
We didn't fix it yet. For the moment the Definity is not connected
directly to Asterisk, we route all communications between Asterisk and
the Definity over the PSTN.
The plan is to play around with all protocol settings to figure out which one
is the most stable, from what I understand - however I
On Wed, Jan 27, 2010 at 8:22 PM, Danny Nicholas wrote:
> This stands to be corrected, but as I understand it, the queue command on
> it’s own would not generate a second CDR any more than a transfer to an
> extension. The way I understand the queue/agent/call relationship is this:
>
>1. age
Hi,
im a student and we are devloping a training sytem for
radio operators (for ships, police, ...) at our university.
So far we are using a simple own protocol for speech and data
transmission, works well at a Lan. Now we are looking for a way to
connect the devices over the internet.
I did so
Having looked at the outputs into PMS they are very simple stop start records.
Line by line text that can easily be recreated. They have about 4-5 fields,
origin number, destination, time of call, duration, or similar things
Usually they go out via a serial port or TCP port expecting a terminal
This stands to be corrected, but as I understand it, the queue command on
its own would not generate a second CDR any more than a transfer to an
extension. The way I understand the queue/agent/call relationship is this:
1. agent(s) login to queue this may or may not create a CDR entry
2.
2010/1/27 Håkon Nessjøen
> On Wed, Jan 27, 2010 at 3:37 PM, Danny Nicholas wrote:
>
>> Just a shot in the dark – what is the endbeforehexten value in cdr.conf?
>>
> It was not defined.
>
> And the result was the same either it was set to yes or no. The cdr closing
> when user B hangs up, gets t
many people around think mysql is not a good option for database, they
think mysql
is only suit for small business. but i want to have a try. i need to
convince them to use this.
This statement is absolute BS. Give me some factual, backed statements by
trained database professionals who don't
> 2010/1/27 Steve Edwards :
>> So don't use ODBC, use Pro*C...
On Wed, 27 Jan 2010, Zhang Shukun wrote:
> you said" Personally, I'd vote for an AGI using whatever C API your DB
> provides" what do you think about phpagi and cagi, if i choose the agi
> method. while phpagi seems used more popul
On Wed, Jan 27, 2010 at 6:10 PM, Kevin P. Fleming wrote:
> 1) When a sip.conf entry is defined as 'type=friend' *and* has a
> specific host IP address (not dynamic), we could just ignore the 'user'
> part and create only the 'peer' part. This would result in incoming
> calls being matched by IP ad
Did you get this resolved? And how if you did.
We've been have the same random PRI lockup issue for years now.
I've opened a mantis bug https://issues.asterisk.org/view.php?id=16713 and
hopefully we can get this issue resolved.
Alec
-Original Message-
From: asterisk-users-boun...@lists.
On Wed, 2010-01-27 at 10:27 +0800, Zhang Shukun wrote:
> hi,all
>
> i have just upgrade from 1.4.28 to 1.6.2.1. all works fine now except
> realtime queue.
>
> it seems queue_table works fine, but queue_member_queue not work, the
> two tables works fine when in 1.4.28.
>
> is that something chan
Administrator TOOTAI wrote:
> Olle E. Johansson a écrit :
>> 27 jan 2010 kl. 11.47 skrev Administrator TOOTAI:
>>
>>
>>> Hi,
>>>
>>> we had an attack on a server and we don't understand how it was
>>> possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL,
>>> network 188.161.128
On Wed, 27 Jan 2010, Mark Wiater wrote:
> the mitel 3300 sends SMDR on TCP 1752. It spews software and hardware
> logs in the same manner, different ports.
This particular model (need to get the model number) has a serial
connection. I'm all for putting a serial sniffer between them (if the
Hello
I need to add sip extensions from my UI so without going through sip.conf so
i created table
CREATE TABLE `sipfriends` (
`name` varchar(40) NOT NULL default '',
`username` varchar(40) default '',
`secret` varchar(40) NOT NULL default '',
`context` varchar(40) NOT NULL default '',
the mitel 3300 sends SMDR on TCP 1752. It spews software and hardware logs in
the same manner, different ports.
On 1/27/2010 11:00 AM, Steve Howes said:
> On 27 Jan 2010, at 15:48, Jeff LaCoursiere wrote:
>> Sounds good to me, but without the spec I'm stuck in a catch 22!
>
> tcpdump? (assumin
On 27 Jan 2010, at 15:48, Jeff LaCoursiere wrote:
> Sounds good to me, but without the spec I'm stuck in a catch 22!
tcpdump? (assuming IP). Bet its fairly simple plain text or something.
Steve
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-- Bandwidth and Colocation
Hi,
A potential client (hotel) has a Property Management System that talks the
"Mitel" protocol to their current Mitel PBX in order to receive CDRs
(which end up being rated by the PMS system and charged back to guests).
Does anyone know of any (free or otherwise) docs on this protocol, or
be
Hello Danny,
what do you mean by 'all the CDR fields' ?
The Destination-field shows '1' or '2'. The dstchannel shows the correct
SIP-channel. But this is not the same as the 'real' destination namely
the SIP-account of my SIP-phone.
Jonas.
On Wed, 2010-01-27 at 08:22 -0600, Danny Nicholas wrote
This is WAY OT but I had no idea what fnal.gov was, so I checked it out:
http://computing.fnal.gov/xms/Services/Getting_Services/Web_at_Fermilab/Professional_Home_Pages_at_Fermilab
And I quote "...professional information about themself..."
About themself? Really? Really?
That is all.
Cheers
Da
Hi,
I have very limited experience with BRIs, and now I have a project which
requires to hook up an asterisk server to a client's Simen Hicom PBX with 32
BRI ports. In this regard I am looking for the right ISDN-BRI cards which I
can install in an Asterisk server. I need two types of cards, one wh
On Wednesday 27 January 2010 01:48:47 Zhang Shukun wrote:
> how does the system recognize them. i mean queue_name is not an
> configure option in agent.conf
The name between the square brackets in queues.conf is the queue_name.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitte
On Wednesday 27 January 2010 01:34:57 Zhang Shukun wrote:
> 2010/1/23 Steve Edwards :
> > On Fri, 22 Jan 2010, Zhang Shukun wrote:
> >> as you know, we can use MYSQL command to visit mysql database
> >>
> >> but if i use other database like Oracke,sybase,etc, Could i use MYSQL
> >> command ?
> >
>
>> On Mon, Jan 25, 2010 at 12:07:55PM -0600, Karl Fife wrote:
>> > From: "cb" Sent: Sunday, January 24, 2010 12:42
>> > I use the Snom 370 all day long at work. I have never had a problem
>> > adjusting the volume. I change it multiple times a day as I keep my
>> > handset on one volume and my hea
In this case, a SIP provider would not be required.
Obviously, you will need ports on your EPBX to connect the Digium card to.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Siju George
Sent: Wednesday, Janua
Olle E. Johansson a écrit :
> 27 jan 2010 kl. 11.47 skrev Administrator TOOTAI:
>
>
>> Hi,
>>
>> we had an attack on a server and we don't understand how it was
>> possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL,
>> network 188.161.128.0/18
>>
>> Hacked account had followi
On Wed, Jan 27, 2010 at 3:37 PM, Danny Nicholas wrote:
> Just a shot in the dark – what is the endbeforehexten value in cdr.conf?
>
It was not defined.
And the result was the same either it was set to yes or no. The cdr closing
when user B hangs up, gets the full duration of call A at that poin
wins mallow a écrit :
> On Wed, 2010-01-27 at 11:47 +0100, Administrator TOOTAI wrote:
>
>> [...]
>>
> Check your sip.conf
> allowguest=no
>
>
Guest are allowed and going to a different context. Logs are showing
that calls are going out to the from-111 context, so its this account
whic
Just a shot in the dark what is the endbeforehexten value in cdr.conf?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Håkon Nessjøen
Sent: Wednesday, January 27, 2010 7:59 AM
To: Asterisk Users Mailing List
Subject: [aste
forkCDR might be helpful; also, you might want to check all of the CDR
fields.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Wednesday, January 27, 2010 3:25 AM
To: Asterisk Mailing
Subject: [asterisk-
Astdb is a "built-in" Berkley database that Asterisk uses via a specific
command set. It is (IMO) simpler to use than MYSQL, POSTGRES or whatever
other flavor of database you might use (odbc, etc). It does not
(necessarily) store realtime values; it's more of a simple push/pull single
key databas
Hi,
I'm having problems with CDR's and Queues in Asterisk 1.6.1.
Heres three examples:
Normal call:
User A calls in to asterisk, gets a PlayFile, and hangs up. This gives 1
CDR as expected.
Call to a Queue and then a playfile afterwards:
User A calls into asterisk, goes into a queue, asteri
Hello
I think I finally understood the issue/solution, but I'd like to make
sure I'm correct:
- In Diana Cionoiu's famous article on Freshmeat
(http://freshmeat.net/articles/nat-traversal-for-the-sip-protocol),
regardless of whether SIP end-points use a public IP or are behind a
NAT, RTP packets
Hi,
At my previous company we ran 1.4.x.x (underneath DiVitas.com software) and our
Polycom IP 550 would use DND without a problem, but the IP 331 (on exactly the
same server) didn't work with DND. So it may be a model-specific problem rather
than your Asterisk config.
Stuart
_
27 jan 2010 kl. 11.47 skrev Administrator TOOTAI:
> Hi,
>
> we had an attack on a server and we don't understand how it was
> possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL,
> network 188.161.128.0/18
>
> Hacked account had following setup:
>
> [111]
> type=friend
> use
On Wed, 2010-01-27 at 11:47 +0100, Administrator TOOTAI wrote:
> Hi,
>
> we had an attack on a server and we don't understand how it was
> possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL,
> network 188.161.128.0/18
>
> Hacked account had following setup:
>
> [111]
> type=f
Thanks a lot guys. Exactly what I needed.
Best regards,
Örn
On Tue, Jan 26, 2010 at 8:48 PM, Olle E. Johansson wrote:
>
> 26 jan 2010 kl. 16.48 skrev Örn Arnarson:
>
> > Hi guys,
> >
> > I am wondering (and have been unable to find out thus far) whether
> Asterisk sets some special channel vari
Hi, all
What is the use of astdb?
Is it used to store realtime values like sip etc.
Regards,
Bhrugu Mehta
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To UNSUBSCRIBE or
On Tue, 2010-01-26 at 13:17 -0600, Kevin P. Fleming wrote:
> Jeff Brower wrote:
>
> > How do you know for sure fax detection is turned off? It sounds to me like
> > your changes to the dahdi config file are
> > being ignored. Maybe put something in there that should cause an error or
> > somet
I'm using Asterisk 1.4.27.
In queues.conf I do not find this option. I have added it, reloaded
Asterisk, but still the destination is '1' or '2'.
Does it make a difference of my queue members are just SIP-accounts in
stead of agents ?
member => SIP/VCsupport,1,Jonas
member => Agent/VCjoeri,2,Joe
Hi,
we had an attack on a server and we don't understand how it was
possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL,
network 188.161.128.0/18
Hacked account had following setup:
[111]
type=friend
username=111
context=from-111
host=11.22.33.44
dtmfmode=auto
qualify=yes
nat=
from queue.conf
; UpdateCDR behavior.
;This option is implemented to mimic chan_agents behavior of populating
;CDR dstchannel field of a call with an agent name, which you can set
;at the login time with AddQueueMember membername parameter.
;
; updatecdr = no
I've never used it. YMMV
Hi,
If I get a Dignum Card and fit it into my computer do I still need an
SIP provider to connect through my EPBX to a Public Telephone System?
Thanks
--Siju
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-- Bandwidth and Colocation Provided by http://www.api-digital.
Hello list,
I'm using an IVR where the caller chooses between 1. sales 2. support.
When choosing 1 the caller is directed to the sales-queue when choosing
2 the caller is directed to the support-queue.
Then the caller is directed to a free agent.
I notice in the CDR-rapports that the destination
2010/1/27 Steve Edwards :
> Un-mid-posting...
>
>>> On Fri, 22 Jan 2010, Zhang Shukun wrote:
>>>
as you know, we can use MYSQL command to visit mysql database but if i
use other database like Oracke,sybase,etc, Could i use MYSQL command ?
>>>
>> 2010/1/23 Steve Edwards :
>
>>> ODBC will d
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