Hi,
I installed the XRMS Open source CRM(xrms-2006-07-25-v1.99.2-.tar.gz) in the
PC. Installed the CTI plugin inside as per the readme file inside
/plugins/cti/README.txt(I am attaching that as well for the reference). I
could not able to view the later development inside the application. Can
2009/8/6 Alex Samad a...@samad.com.au
On Fri, Jul 24, 2009 at 08:28:48AM -0500, Danny Nicholas wrote:
Here's how I think your dialplan should look:
exten = 101,1,Ringing
exten = 101,2,Answer()
exten = 101,3,Dial(SIP/quentin,10)
exten = 101,n,VoiceMail(1...@default,u)
exten =
- what's the difference between a subscribe request et a register
request ?
A subscription in the SIP protocol is saying Hey, I'd like to be
notified when something happens. This is most often used when a phone
wants to subscribe to the state of another extension, or to the status
of a
2009/8/6 Alex Balashov abalas...@evaristesys.com
Sure it is. Just get a media gateway that does T.38 - and does it
relatively well.
Wich the Pattons do quite well afaik.
Chris
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Hi
I'm having an issue with just one of the phones (snom300) attached to
our asterisk server (1.4.17 using RealTime)
Sometimes (not consistently), any outbound call cust off at 20 seconds
exactly and I see the following in my asterisk console
[Aug 6 10:37:24] WARNING[1679]: chan_sip.c:1946
harry R schrieb:
But as someone else wrote before, you can do a dialplan like this
exten = 101,1,Ringing
exten = 101,n,Answer()
exten = 101,n,Dial(SIP/quentin,10)
exten = 101,n,Goto(101-${DIALSTATUS},1)
exten = 101-NOANSWER,1,VoiceMail(1...@default,u)
exten =
Hello to all
I have a queue where often my agents get stuck and cannot logoff.
This is very bad, because agents cannot login again, and in Queuemetrics
reports the agents appear to be online.
How can I create a timeout to my agents and for the queue to kick them?
Thanks
Regards
Joao Pereira
--
Hello
I have recently configured TDM400P with four FXO ports.
My next requirement is to configure for E1 line. which contain 30
phone lines and 2 for signalling information.
The problem is I dont want to go for E1 line directly .Is it
possible to get simulation for E1 line ... so that i can
Hi,
Would it be sane to run ntop on the same box as Asterisk or better to
mirror a LAN port etc?
http://www.ntop.org/OpenSourceVoipMonitoring.pdf
Thanks.
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AstriCon 2009 -
exten = 101,1,Ringing
exten = 101,n,Answer()
exten = 101,n,Dial(SIP/quentin,10)
exten = 101,n,Goto(101-${DIALSTATUS},1)
exten = 101-NOANSWER,1,VoiceMail(1...@default,u)
exten = 101-NOANSWER,n,Playback(vm-goodbye)
exten = 101-NOANSWER,n,Hangup()
exten = 101-BUSY,1,Playback(busy)
Moises Silva moises.si...@gmail.com writes:
Just for the record, Sangoma Media Gateway does exactly that, leave all
your PSTN interfaces (BRI, SS7, PRI) in another box and communicates with
Asterisk through the Woomera protocol.
What is the advantage of doing Woomera rather than a complete
Greetings list,
Wondering if anyone can shed some light on an apparent change in CDRs between
1.2 and 1.4.
One of our clients runs a virtual PA service and has a few hundred DDIs - one
for each client. Their * box is set up with short codes they prefix their calls
with to set the correct
http://kb.digium.com/entry/95/
On Thu, Aug 6, 2009 at 6:02 AM, ABBAS
SHAKEELshakeel.abbas@gmail.com wrote:
Hello
I have recently configured TDM400P with four FXO ports.
My next requirement is to configure for E1 line. which contain 30
phone lines and 2 for signalling information.
The
Dear all,
Picked up some more BT usb adapters and got a flood of error messages as
follows:
hci_scodata_packet: *hci0 SCO packet for unknown connection handle 92*
Anyone has any idea how to deal with this?
Sasa Bobek
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Hello,
I'm running Asterisk 1.6.1.0 compiled from scratch under debian Lenny and
I can't delete message from VoiceMailMain using option 7
Default folder is /var/spool/asterisk/voicemail and it's owned by
asterisk:asterisk with 777 permissions
Apparently VoicemailMain delete the message and
Hello. I think i've seen this problem, it was generated by a missing ACK
on 200 OK. If that is the case try modifying session timer parameters in
sip.conf so a missing ACK will not lead to call termination.
Peter
Ishfaq Malik wrote:
Hi
I'm having an issue with just one of the phones
thanks alot David that was really helping
how can i test for calls etc ie to generate and make calls stuff like that
On Thu, Aug 6, 2009 at 3:30 PM, David Backebergdbackeb...@gmail.com wrote:
http://kb.digium.com/entry/95/
On Thu, Aug 6, 2009 at 6:02 AM, ABBAS
Cheers for that.
We solved the problem this time by updating the routers firmware to the
latest version but if it happens again I'll look into what you're saying.
Can I just clarify that you mean the RTP Timers though? It's a
production server and doing sip reloads is massively frowned upon.
On Thu, 6 Aug 2009, ABBAS SHAKEEL wrote:
thanks alot David that was really helping
how can i test for calls etc ie to generate and make calls stuff like that
I think what Dvid didn't say was that you need a 2nd Asterisk box with E1
card, as well as that cable
Gordon
Lolz exactly right
thats what i was wondering with TDM400P can i do that
i think i need TE420 for this .
I will get that after two or three days then i will ask later on how to test
thanks Alot
On Thu, Aug 6, 2009 at 5:14 PM, Gordon
Hendersongordon+aster...@drogon.net wrote:
On Thu, 6
Hello
I have configured TDM400P with asterisk .
The problem is that when i make a call to server. and while going on
it dont detects call hang up.
ie i called the Asterisk server and it start playing files that i
indicated to do so in extensions.conf
i suddenly put down the phone. now the server
On 6 Aug 2009, at 11:33, Sébastien Cramatte wrote:
Apparently VoicemailMain delete the message and inmediatly undelete
it !
This the same issue as in this post :
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg198336.html
Note that I'm using the spanish voiceset.
==
Assuming you are connected to a regular phone line, the hang up signal from
the phone line would be a break or reversal of polarity of the DC signal on
the phone line. (We connect to PRIs, so our signaling is on a data channel.
I assume you don't. )
The first question you need to answer is Are
I was having the same problem with about half of my POTS lines.
I switched the polarity on the connections for those lines and the problem
disappeared.
-Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
I am not finding anything relating to this on Google, so I thought I'd
pose the question here...
I am running Asterisk 1.4 on a CentOS 5 Linux box. I needed to use a
custom built PHP5.2.10 install to interconnect with our Firebird SQL
database, which I've done. But I noticed that the default
We have added Asterisk to a line of 'mission critical' servers at our
business, and being in the web application development business one of
the core things we do is to monitor web server availability.
I'd like to add Asterisk to the servers that our monitoring systems are
handling, and also
On 6 Aug 2009, at 15:21, Myles Wakeham wrote:
#!/usr/bin/php -q
which I would assume I simply need to change to:
#!/usr/local/bin/php -q
for my build, but is this enough?
You could always test it..
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On Thu, 6 Aug 2009, Myles Wakeham wrote:
I am running Asterisk 1.4 on a CentOS 5 Linux box. I needed to use a
custom built PHP5.2.10 install to interconnect with our Firebird SQL
database, which I've done. But I noticed that the default install path
for PHP5 on this box appears to be
How about a shell script on the monitoring server:
#!/bin/sh
trunk=`ssh aster...@astbox asterisk -r -x 'sip show registry' | grep USERNAME`
state=`echo $trunk | awk '{print $4}'
if state is 'Registered', yay!
else, UHOH!
EOF
Based on that ssh/shell script framework (you'd obviously need host
Hello Everybody,
I have a genuine problem in Asterisk setup.
I have three inbound trunks in my asterisk box, everything is
working fine but the only problem is when any user make an out-
going call through his/her extension it goes with same number labeled
on this.
Can we set each of
Are these trunks or PRI/ISDN circuits, or phone lines?
If either of the first two, the callerID sent with the call should be their
ID, which should be the appropriate number of digits your area telco
expects. Depending on your agreement with them, they may be supplying the
number, rather than
I've been Googling all morning and searching voip-info.org but not quite
finding what I'm looking for.
I've read that you can modify the billing/account information on a CDR via
AGI but I can't find an example or a how to.
I'd like to then assign specific accounts in the CDRs. Possible?
Thanks,
On 6 Aug 2009, at 16:32, kumarshantanu wrote:
Hello Everybody,
Hi.
I have a genuine problem in Asterisk setup.
Ok.
I have three inbound trunks in my asterisk box, everything is
What kind of trunks.
working fine but the only problem is when any user make an out-
going call through
Greetings again List.
I'm facing a strange case with one of the productive Asterisk servers..
i have 3 providers sending traffic to the call center where agents pickup the
calls.
calls come into the server Queue Agents
Last October .. an undersea cable got disconnected placing Egypt and the
David writes:
How about a shell script on the monitoring server:
#!/bin/sh
trunk=`ssh aster...@astbox asterisk -r -x 'sip show registry' | grep
USERNAME`
state=`echo $trunk | awk '{print $4}'
if state is 'Registered', yay!
else, UHOH!
EOF
Based on that ssh/shell script framework
Steve writes:
#!/usr/bin/php -q
which I would assume I simply need to change to:
#!/usr/local/bin/php -q
This should work. Did you try it?
Yes, its working fine. The only problem is that when I went to
'uninstall' the standard PHP installation that came with CentOS 5 on
this
Hi, how can I make an inbound call to be coded by the agent and be stored in
the database for a later report??
I mean, I want the agent to dial after the call is finished, a code that
means Age information, another code that means where to get more info, etc.
I really really appreciate your help.
You want a dialplan or AGI that works in the h (hangup) context. What
you really will probably want is a callback to accept the digits.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rodrigo Cruz
Sent: Thursday,
Hi everyone.
We have an asterisk server in our main office and phones at each
remote site. The remote offices are connected via a MPLS which, to my
knowledge has no natting going on.
The problem I have is that any call from a remote phone to a remote
phone (even on the same remote lan) results
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
J. G. wrote:
I've been Googling all morning and searching voip-info.org
http://voip-info.org but not quite finding what I'm looking for.
I've read that you can modify the billing/account information on a CDR
via AGI but I can't find an example or
At 5:24 PM on 06 Aug 2009, ABBAS SHAKEEL wrote:
Hello
I have configured TDM400P with asterisk .
The problem is that when i make a call to server. and while going on
it dont detects call hang up.
ie i called the Asterisk server and it start playing files that i
indicated to do so in
Gah - I've been trying to find the proper search syntax all day.. I Googled
asterisk CDR Function and it's the first thing that comes up...
Sometimes I wonder whether or not my brain works right..
Thanks Barry!
On Thu, Aug 6, 2009 at 3:03 PM, Barry L. Kline blkl...@attglobal.netwrote:
Hi, has anybody some python code algorithm to parse an extension pattern?
I have a number and need to know if match with some pattern.
Thansk!
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AstriCon 2009 - October 13 - 15
Danny Nicholas escribió:
You want a dialplan or AGI that works in the h (hangup) context.
What you really will probably want is a callback to accept the digits.
*From:* asterisk-users-boun...@lists.digium.com
On Thu, Aug 6, 2009 at 6:19 AM, Benny Amorsen
benny+use...@amorsen.dkbenny%2buse...@amorsen.dk
wrote:
Moises Silva moises.si...@gmail.com writes:
Just for the record, Sangoma Media Gateway does exactly that, leave all
your PSTN interfaces (BRI, SS7, PRI) in another box and communicates
Perhaps it's only basic in certain parts of the world... I know I've never
experienced a voicemail system with such a feature...
I'm not saying having the option would be bad... but... I'd prefer voicemail
to get some more common, more requested of me, features first and that's
personally where
At 3:32 PM on 06 Aug 2009, kumarshantanu wrote:
I have a genuine problem in Asterisk setup.
I have three inbound trunks in my asterisk box, everything is
working fine but the only problem is when any user make an out-
going call through his/her extension it goes with same number labeled
on
Myles Wakeham schrieb:
Steve writes:
#!/usr/bin/php -q
which I would assume I simply need to change to:
#!/usr/local/bin/php -q
This should work. Did you try it?
Yes, its working fine.
if I can rely
on the #! setting in the file, that's good enough for me.
It's
Anyone have any firsthand experience implementing OpenSBC
(opensourcesip.org)? Have a possible consulting gig referral.
Cory J. Andrews
Director New Market Initiatives
Sayers Media Group
VoIP Supply, LLC
454 Sonwil Drive
Buffalo, NY 14225
716-250-3402 OFFICE
716-630-1548 FAX
Hi,
This sounds udp RTP problem.
Might be you have some firewall rules that block this kind of traffic ?
As soon I remember, Asterisk by default use random port between 1
and 2 for rtp traffic (you can adjust this in rtp.conf).
- Sebastien
Jonathan Moore escribió:
Hi everyone.
We
Hello!
What are the nat_sip modules you mention?
When I set up a linux router some time ago and configured sip.conf
with net=yes, everything went smoothly just like any other router.
Elliot
On Mon, Aug 3, 2009 at 8:45 PM, Gordon
Hendersongordon+aster...@drogon.net wrote:
On Mon, 3 Aug 2009,
On Thursday 06 August 2009 05:27:57 Chris Bagnall wrote:
Greetings list,
Wondering if anyone can shed some light on an apparent change in CDRs
between 1.2 and 1.4.
One of our clients runs a virtual PA service and has a few hundred DDIs -
one for each client. Their * box is set up with short
On Thu, Aug 6, 2009 at 7:30 AM, ABBAS
SHAKEELshakeel.abbas@gmail.com wrote:
thanks alot David that was really helping
how can i test for calls etc ie to generate and make calls stuff like that
You have four ports. You can use a T1/E1 crossover to send calls from
one port to a different
On Thu, Aug 6, 2009 at 5:17 PM, SŽébastien
Cramattescrama...@zensoluciones.com wrote:
Hi,
This sounds udp RTP problem.
Might be you have some firewall rules that block this kind of traffic ?
As soon I remember, Asterisk by default use random port between 1
and 2 for rtp traffic (you
On 7/08/09 2:28 AM, Myles Wakeham wrote:
We have added Asterisk to a line of 'mission critical' servers at our
business, and being in the web application development business one of
the core things we do is to monitor web server availability.
I'd like to add Asterisk to the servers that our
The subject of tomorrow's VoIP Users Conference will be mobile VoIP.
If you have any interest, please join us. I myself am tesing a bunch
of iPod applications to use with all the usual suspects: OnSIP,
Sipgate, Gizmo, Skype, your asterisk box, etc.
Details for joining the call are are at
Hi,
I've tried two SIP clients so far and both have unusable outgoing
audio quality. Skype app sounds fine, and recording the same mic
sounds fine, so I can only assume there is an issue with the clients
themselves.
Both clients allow you to register and make calls via SIP with any
abitrary
Thanks Alot C. Chad Wallace
it worked.
On Fri, Aug 7, 2009 at 12:09 AM, C. Chad
Wallacecwall...@lodgingcompany.com wrote:
At 5:24 PM on 06 Aug 2009, ABBAS SHAKEEL wrote:
Hello
I have configured TDM400P with asterisk .
The problem is that when i make a call to server. and while going on
Which generation of the handset are you using? They differ in their
processing power and that may account for at least some of it.
For example, I had a first-generation EDGE iPhone and tried SIP clients
over wifi and couldn't get anything useful out of them either. I'm yet
to attempt it
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