If you can somehow waive the same username requirement, the solution
is quite simple:
exten = xxx,n,Dial(SIP/user1SIP/user2SIP/user3...SIP/usern,xxx)
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Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax:
Hi All;
I configured some queues, and I configured the dialed numbers for login and
logout for the agents.
Two agents are logged in, the first two calls are received at the agents and
they answered and hangup. Again, the two agents are idle and ready to receive
calls. The third and call goes
Bilal ,
if you can do a core show queue QUEUENUMBER and paste the output here at the
moment of this problem it will be helpful to see what is the status of your
agents at that moment for the queue.
does it help if the agents logout then back in instead of disconnecting the
call and calling
On Sun, 18 Sep 2011 22:28:32 +0200, Gilles codecompl...@free.fr
wrote:
For some reason, even through I'm using a headset, there's a lot of
echo and after a few seconds, it sounds like it enters a very fast
loop before the echo stops somewhat. IOW, unusable sound.
Problem solved: Tried XLite 4.1
I'm trying to set-up a Cisco 6945 with SCCP firmware under AsterisK 1.6
with sccp-b 3.0.4. Does anyone have a working example they can share:
the TFTP root directory files and the sccp.conf file?
Thanks.
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Hi List,
I have one query, I am using Go autodial in this using auto dialing.
autodial can do only whenever customer pick the call that call will go to
agents.
but problem autodial dialing the database in that I am not getting NC data
means, not reachable,switch off ,outofservice data. how can I
Hello,
currently I run a php script in cron which polls for information on SIP
peers using the /usr/sbin/asteirsk -rx method.
I notice that after some time the Asterisk interface freezes, SIP Peer
registrations become unreachable and sip reload or any other command on
the CLI does not
I am running asterisk 1.4.41.2 and dahdi 2.4.1 (64 bit centos)
I only have one small issue.
I initiate a call over AMI, call is answered and I run my AGI.
sometimes when I make calls out to cell phones I ask to press 1 to confirm
the user hears the message and press 1 but I never get the 1 back
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Monday, September 19, 2011 9:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] /usr/sbin/asterisk -rx and AMI
Hello,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, September 19, 2011 9:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] question on DTMF
I am
No, I have no suggestions. What you are describing has nothing whatsoever to
do with registration. Registration only applies to calls from Asterisk to the
phone. It has nothing to do with calls from the phone to Asterisk.
-Original Message-
From:
Depending on the cell phone you are calling, the DTMF length may need to be
set to LONG (I know this applies to Verizon phones).
Danny
I am not familiar with this setting - where is it exactly.
I looked in my sip.conf and did not see anything.
Thanks-
Jerry
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, September 19, 2011 10:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] question on DTMF
On Sun, 18 Sep 2011, Catalin S. wrote:
Is about outgoing calls from multiple devices with the same username at
aprox same time. The overwritten is for incomming calls. I want to
prevent using the same account in multiple devices at same time. The
solution with IP will not apply because users
Recently, we were notified that the mechanism included in our Asterisk
source code releases to download and build support for the iLBC codec
had stopped working correctly; a little investigation revealed that this
occurred because of some changes on the ilbcfreeware.org website. These
changes
I know over time SIP OPTIONS message handling has changed and I've seen
some write ups that seem to indicate that an s extension in the default
context is needed now to get them to work.
It's probably my error in any case.
So, what am I doing wrong or what do I need to do to get the sip ping
Every request needs a From tag.
--
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/
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-- Bandwidth
On 09/19/2011 09:33 AM, Alex Balashov wrote:
Every request needs a From tag.
Uh... OK. Isn't this a From tag:
From: sip:p...@xx.xx.xx.xx
Line three of what I send?
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-- Bandwidth and Colocation Provided by
no, you need a tag i.e From: sip:p...@xx.xx.xx.xx;tag=xxx, where xx
is a unique identifier
see the definition of SIP Dialog
Dialog: A dialog is a peer-to-peer SIP relationship between two
UAs that persists for some time. A dialog is established by
SIP messages, such
On 09/19/2011 01:11 PM, Bruce Ferrell wrote:
On 09/19/2011 09:33 AM, Alex Balashov wrote:
Every request needs a From tag.
Uh... OK. Isn't this a From tag:
From: sip:p...@xx.xx.xx.xx
Line three of what I send?
No, that's a From URI.
A From tag is a header parameter that is appended to
On 09/19/2011 01:16 PM, Alex Vishnev wrote:
no, you need a tag i.e From: sip:p...@xx.xx.xx.xx;tag=xxx, where
xx is a unique identifier
see the definition of SIP Dialog
Dialog: A dialog is a peer-to-peer SIP relationship between two
UAs that persists for some time. A dialog
Thank you Alex. That was useful information. The sip_ping.pl program
I was using used to work without the tag. It seems asterisk now demands it.
Bruce Ferrell
On 09/19/2011 10:18 AM, Alex Balashov wrote:
On 09/19/2011 01:16 PM, Alex Vishnev wrote:
no, you need a tag i.e From:
Hello,
Looking for Asterisk-FreePBX in a Flash Technical Support in Tampa
Please contact me by phone: 813-842-6941
Thank You,
[cid:image001.gif@01CC76D3.D2584900]
Secure2ware Inc.
813-425-5900
Keith A. Ware, ext. 211
ke...@secure2ware.commailto:ke...@secure2ware.com
Hi List,
I have a system that connects into Asterisk 1.4.41 using CISCO
CCM 7. Everything works great except when a call is transferred to the
operator. The call goes to the operator via a native bridge and is
completed, then a phantom process starts and tries to launch a new call
Hello All,
Thank you all for responding so quickly.
I just hired someone for Technical Support.
Just wanted to inform everyone.
Thank you all and good luck to everyone!
Keith
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-- Bandwidth and Colocation Provided by
This is going to sound ridiculous, but there appears to be a ghost DID
in our system. We are going to get the number ported to us, but it has
not happened yet. From a phone outside of our voip system, the call
still goes through. When calling the did from a phone within our system,
there is
Is about outgoing calls from multiple devices with the same username at
aprox same time. The overwritten is for incomming calls. I want to prevent
using the same account in multiple devices at same time. The solution with
IP will not apply because users may be behind nat or will change
Please check our voice sms and fax broadcasting / smart autodialler / smart
predictive dialler based on asterisk named ictbroadcast , it provide real
time report of busy, answered, congestion , failed, no answer call
statistics of running campaign
HTTP://www.ictinnovations.com/ictbroadcast
Hi Danny,
If you explain some more about this phantom process !! I've never seen
asterisks doing this before. This initiation of a new call is always
dependent upon arrival of an INVITE. I doubt its CCM that is doing some
re-INVITES or sort of keepalive for this call and thus a phantom call is
Without your dialplan there isnt much that can be done to help.
Can you please post your relevant dialplans?
Whats voip1 and voip2?
When you say outside the voip system call goes thru, to where?
Who has the number currently?
Any sip debug you care sharing?
On Mon, Sep 19, 2011 at 6:51 PM, Aaron
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