Just upgraded to 1.8, we use the multi lot parking feature by dialling *4.
We are not getting the parking slot announcement being played to the person
who parks the call, so it's impossible to tell which slot they've gone
into. Could someone check our config?
On Debian Squeeze using packages from
Asterisk 1.6.2.20 on Debian Lenny
I'm finding that if no one answers an attended transfer (timeout set by
atxfernoanswertimeout), then the transferrer is handed back to the original
caller, and a beep is played.
In 1.4 I was able to indicate the timeout and failure by setting xferfailsound
to a
Thanks for help- suggestion fixed the issue
John
On 21 November 2011 11:25, John Taylor j...@vetsurgeon.org.uk wrote:
Thanks AJ- have set it to 5 mins via telnet: srv dhcp leasetime 600. Will
get permission to try new firmware later!
JT
On 21 November 2011 10:45, Arthur Stanfield
One of our clients has a Draytek Vigor 2920- their natted Snom phones
behind it are registered to an Asterisk 1.4 server on an external public IP.
I've set QOS, bandwidth management and turned off the SIP ALG via telnet
but I'm still having some problems with some of the phones losing
Message -
From: John Taylor j...@vetsurgeon.org.uk
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, 21 November, 2011 10:20:14 AM
Subject: [asterisk-users] vigor 2920 problems
One of our clients has a Draytek Vigor 2920- their natted
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
Have recently installed some Snom phones into an office. Phones are
natted and connect to a 1.4 server on a public IP
We can make outgoing calls, but are unable to answer incoming calls.
The phone rings, but the call cannot be picked up. Other phones on
other sites connected to the server are
Why not write the file to /tmp using MixMonitor, then use the command
option to trigger an AGI script that will move the data into your
database then delete the original file?
John
On 24 September 2010 04:23, Govind, Mahesh (NSN - IN/Bangalore)
mahesh.gov...@nsn.com wrote:
The reason is when
Before I reinvent the wheel, I'm looking for a script then when run will
- launch tcpdump (or equivalent) on the server and capture all SIP and
UDP traffic to an IP address
- then, rather than me manually analysing with wireshark, will analyze
the cap file and produce stats on jitter, lag, delta
We're using firmware 7.3.30 on an installation of Snom 300 phones.
Should we stick with it, or do the newer firmwares have better support
for Asterisk?
Thanks
John
--
_
-- Bandwidth and Colocation Provided by
I have an Asterisk server on our LAN that serves our office VOIP
phones with a SIP trunk to voipfone (UK ITSP). All LAN calls are
ulaw/alaw
We use attended transfer extensively. If our trunk is ulaw/alaw they work fine.
If the trunk is ilbc we have problems
1- incoming PSTN call routed via
Hi- can anyone help with this. I'm really stuck as apparently it
should work. Is it a problem with the ITSP, with using the same trunk
for both legs of the call etc?
John
On 30 January 2010 08:57, John Taylor j...@vetsurgeon.org.uk wrote:
Hi
If I have an incoming call coming down a SIP trunk
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
Hi,
I've now set dtmfmode=rfc2833 and that seems to have fixed it
John
2010/1/7 John Taylor j...@vetsurgeon.org.uk:
We're now getting this problem on outgoing calls. I've forced the port
to 100FD but still no joy. Anyone any ideas how to debug this- have
added verbose to logger.conf
Hi all,
I've now set dtmfmode=rfc2833 instead of inband and that seems to have fixed it
John
2010/1/4 John Taylor j...@vetsurgeon.org.uk:
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
We're now getting this problem on outgoing calls. I've forced the port
to 100FD but still no joy. Anyone any ideas how to debug this- have
added verbose to logger.conf
Thanks for any help
John
2010/1/4 John Taylor j...@vetsurgeon.org.uk:
I have recently moved our asterisk server from our LAN
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
Put the commonly used domain names + appropriate ips into /etc/hosts?
John
2010/1/4 Steve Howes steve-li...@geekinter.net:
On 4 Jan 2010, at 08:34, Remco Barendse wrote:
Is there any fix or workaround for the DNS problem (old standing bug
that
when the box starts and domain names do not
,1)
exten = 31592123457,1,Goto(trunk1,s,1)
exten = 31592123458,1,Goto(trunk1,s,1)
exten = 3159212,1,Goto(trunk2,s,1)
exten = 31592123334,1,Goto(trunk2,s,1)
exten = 31592123335,1,Goto(trunk2,s,1)
On 14 dec 2009, at 10:39, Olle E. Johansson wrote:
11 dec 2009 kl. 23.21 skrev John
Thanks - have done that and am now trying a one out. However, I'd
still like to know whether 1 asterisk server can support multiple
trunks/registry entries. Does it cause problems?
Thanks
John
2009/12/3 Tim Nelson tnel...@rockbochs.com:
- John Taylor j...@vetsurgeon.org.uk wrote:
I want
:
On Fri, Dec 11, 2009 at 10:23 AM, John Taylor j...@vetsurgeon.org.uk wrote:
Thanks - have done that and am now trying a one out. However, I'd
still like to know whether 1 asterisk server can support multiple
trunks/registry entries. Does it cause problems?
yes, Asterisk does support multiple
I have multiple trunks to the same ITSP. Incoming calls to any trunk
go to the last incoming label defined in those trunks' contexts in
sip.conf.
My ITSP insists on insecure=very in the trunk context; is this the cause?
John
2009/12/11 Noah Miller noahisaacmil...@gmail.com:
I assume if all the
I want to use an asterisk box to provide a voip service to a number of
separate companies.
I have a VOIP provider who I want to trunk with. As far as I can see
it there are 2 options
1. Have 1 SIP trunk to one account at the provider who gives me
multiple incoming numbers; this is less than
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 [EMAIL PROTECTED]:/etc/asterisk/*
/etc/asterisk/)
Asterisk worked fine on the old server, but on this server my SIP
trunk
Dear all
I've got call queuing working when calls originate from my local site.
After testing I migrated it to calls originating from our voip
provider- it should ring an extension, then queue . All works well
apart from if the caller hangs up when queued: the call hangs around
in the queue as a
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