Displaying 20 results from an estimated 400 matches similar to: "call queuing not detecting caller hang up when call originates from voip provider"
2008 Nov 27
0
trunk peer not registering after migrating installation
I have an odd problem. I have just installed asterisk on an ubuntu
box, and migrated the previous configuration of asterisk (on another
ubuntu box) to this new server (scp -pr xxx at oldserver:/etc/asterisk/*
/etc/asterisk/)
Asterisk worked fine on the old server, but on this server my SIP
trunk peer does not login after initial server startup. "sip show
peers" shows my phones
2010 Jan 04
2
caller getting cut off intermittently
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-->extension (192.168.1.x)
Everything works fine (incoming/outgoing audio etc.) except
2010 Jan 30
1
forward call back up same trunk to external cell phone problem
Hi
If I have an incoming call coming down a SIP trunk to a particular
internal SIP extension- I can answer the extension fine, all works
well
However, if I change extension.conf from dialling the internal
extension to forward the call to an external cell phone (up the same
trunk as the incoming leg of the call) I cannot get any audio and get
the following error message on the console:
[Jan 30
2006 Oct 31
0
SIP with Qualify and NAT
Hi guys,
I'm having a really strange problem, which I'm pretty sure has only
appeared since my last upgrade (1.2.12.1) .
It's about NAT and Qualify. I'm using Asterisk to register with some
external SIP providers. However, they're always marked as UNREACHABLE,
when they weren't before!
A typical debug looks like this:
hera*CLI> sip reload
Reloading
2005 Jan 13
1
Registration of SIP
Hi,
I am getting this problem when trying to register with Voipfone.co.uk. It
used to work, and I haven't changed anything that I know of.
Jan 13 10:22:37 WARNING[21645]: acl.c:213 ast_get_ip_or_srv: Unable to
lookup 'voipfone.co.uk.voipfone.co.uk'
Why does the domain name appear twice? I don't know when it stopped working.
In SIP.CONF
[sip_proxy-out]
type=peer
2005 Sep 27
0
asterisk@home inbound call problem to SIP trunk. (voipfone UK)
Hi all,
I have recently installed Asterisk@home and outbound calling is
working great. However I am strugglings with inbound calls. I have
setup a trunk for my provider, voipfone and in the inbound area on AMP
I have the following :-
user context name = 3011XXXX
context=from-pstn
dtmfmode=rfc2283
fromdomain=voipfone.co.uk
host=voipfone.co.uk
insecure=very
secret=XXXXXX
type=user
user=3011XXXX
2006 Nov 29
3
Siemens Gigaset C450 IP vs S450 IP
I've just ordered a Siemens Gigaset C450 IP cordless
IP/DECT phone, given that it's supported by asterisk
http://www.voipuser.org/review_41.html
However, I see that a slightly better Gigaset S450 IP
is available for only a slight price premium.
Are there any user experiences with the S450 IP?
--
Eugen* Leitl <a href="http://leitl.org">leitl</a> http://leitl.org
2007 Sep 10
2
Siemans SIP/PSTN phone S450
Hi All,
Just added a Siemens DECT SIP/PSTN S450 phone to login to my A*k server,
and I see "Got SIP response 405 "Method Not Allowed" back from
192.168.3.64" but the phone seems to work ok.
Any ideas where it falls over in the SIP protocol? I've included this
in the debug below.
ubiphone*CLI>
<-- SIP read from 192.168.3.64:5060:
--- (0 headers 0 lines) Nat
2011 Jan 19
1
intermittent problem on 1.4
We're trying to forward an incoming SIP call from voipfone (UK ITSP) that
originated from a UK landline back up a SIP trunk to the same ITSP and on to
another UK landline number.
UK Landline->voipfone->asterisk 1.4->voipfone->UK landline
About 1 in 3 times the call at the final landline is silent and we see "RTP
Read too short" scrolling on the console log.
Where do we
2007 May 25
1
hosts.equiv management?
My first public-facing deployment of Puppet is likely to be on some
parallel computing nodes we have here. One of our fluid dynamics
packages expects to be able to rsh from one master node to other slave
nodes in the same batch queue. I think I''ve made some good class
definitions up to this last point, where code duplication rears its head.
Example of the problem:
class ch405-host
2009 May 07
1
Macro arguments on app_queue
hi list, i have a question about the args of queue:
when we use Queue() app, there are some arguments than can use. help from
CLI:
Queue(queuename[,options[,URL[,announceoverride[,timeout[,AGI[,macro[,gosub[,rule]]]]]]]])
well.. i'm trying to identify who has taken the call on a queue, and, when
agent conected, launch a macro with some args based on who takes the call
i do:
exten =>
2008 Oct 06
1
R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)
2008/10/5 robert.boardman at gmail.com <robert.boardman at gmail.com>
> Kevin P. Fleming wrote:
> > Olivier wrote:
> >
> >
> >> 2. R Hook-flash key is now available to transfer calls.
> >> In s450IP web management server, its defaults settings are :
> >> Application-type: dtmf-relay
> >> Application-signal: 16
> >>
>
2005 Aug 20
0
Re: Asterisk-Users Digest, Vol 13, Issue 131
Hello
I have problem with transfer call if using ACD
When i using ACD with agent and queue setting, i cannot monitor call and transfer call. this's my setting
- i have 2 IAX phone (phone number as 201, 202), agent.conf
agent => 1001,4321,member 1
agent => 1002,4321,member 2
agent => 1003,4321,Tin
then, queues.conf
[MyQueue]
music=default
strategy=ringall
timeout=30
retry=5
2015 Jul 29
2
Queues don't follow dialplan if no members are registered
----- Original Message -----
> From: "John Kiniston" <johnkiniston at gmail.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
> Sent: Tuesday, July 28, 2015 12:12:05 PM
> Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are registered
>
> In your queues.conf do you
2007 Sep 14
6
DECT SIP phones
Hi folks:
I know it's come up a few times before, but I need some more detail.
I'm looking for a SIP DECT (cordless) phone for North American
installations. I've heard only of the Siemens Gigaset S450/C450 phones.
Apparently these aren't sold for use in NAm, even though they're
supposed to be legal (in the United States, anyway).
On top of that, I understand they have some
2017 Apr 19
2
Can't compile Asterisk on Ubuntu 16
Hey;
Thank you very much. I was able to install asterisk from your link. One
other question. How are you starting asterisk? Do you use an init script or
systemd? Do you think that you could share the script you use?
Thanks Again;
John V.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan H
Sent:
2006 Jun 16
3
Queues and hangup caller on Agent hangup
Hi List,
Just one more question that may sounds stupid to some people but I can't
find the solution for now,
I have the following dialplan:
exten => queue,n,Queue(myqueue)
exten => queue,n,NoOp(ENDQUEUE)
I don't understand why the NoOp is never triggered, the incoming call is
always hangup when the agent hangup...
Is this a behaviour we can't get rid off without patching
2015 Jul 28
2
Queues don't follow dialplan if no members are registered
Hello,
I am running Asterisk 11 on CentOS 6.x. I have configured several queues as
follows in extensions.conf:
exten => s,1,Queue(myqueue,rtnC,18)
same => n,Background(user_unavail)
same => n,WaitExten(10)
exten => 1,1,Voicemail(1111 at my-vm,s)
This rings the phones in the queue for 18 seconds. If no queue members answer,
the caller is then prompted to press 1 and leave a
2010 May 12
1
No ringtone when going from queue to dial-command
Hello list,
when I sent an incoming call first to a queue and after the timeout to a
dial-command, while the correspondent's phone rings there is no ringtone
for the caller...
So it goes like this :
1. dial(SIP/account1,20)
2. queue(myqueue,,,,20)
3. dial(SIP/account2)
In step 1 there is a ringtone for the caller.
In step 2 there is musiconhold (class default) for the caller.
In step 3
2009 Feb 18
2
Setting SIP header on agent calls made by a queue
Hello list,
I am trying to set a custom SIP header on all calls that are made by the app
queue because I want to track a certain state at the SIP level.
If I use the following code:
exten => s,n,SIPAddHeader(X-Unique-ID: ${UNIQUEID})
exten => s,n,Queue(myQueue)
this works fine for the FIRST call made from the queue to an agent; but if
that call does not go through, it's not repeated