Am 05.03.2015 um 15:09 schrieb James B. Byrne:
On Thu, March 5, 2015 05:30, Ruben Rögels wrote:
Am 05.03.2015 um 01:09 schrieb James B. Byrne:
I am trying to determine how the transfer button on the Snom-870
works
with Asterisk. Is the ## special code employed as when it is
entered
Am 05.03.2015 um 01:09 schrieb James B. Byrne:
I am trying to determine how the transfer button on the Snom-870 works
with Asterisk. Is the ## special code employed as when it is entered
through the handset or is the blind transfer through the phone
function accomplished in a different
Hi Felix,
you have several things to check:
netstat -a -n --udp --tcp
will show you connections and connection attempts on network layer level.
You have to look for incoming connections to port 5060 and if the call
has been established for connections on your rtp ports. (see rtp.conf).
If
Hi Akhilesh,
I got below error:
configure: *** XML documentation will not be available because the
'libxml2' development package is missing.
configure: *** Please run the 'configure' script with the
'--disable-xmldoc' parameter option
configure: *** or install the 'libxml2' development
just another thought: if you send the message by mail, do you need to
save it?
regards,
Ruben
Am 21.08.2012 18:45, schrieb Danny Nicholas:
Assuming that you are using the standard 100 message limit, just check for
INBOX/MSG0100.txt and send the message.
-Original Message-
From:
Okay, so have a look at mailcmd= option in voicemail.conf
mailbox will mean a e-mail-box in the next lines.
What you need to do is wirting a shell script or what ever to check for
the return code of the smtp session (normally it should be a 450 in case
of full mailbox).
In case of 450 mailbox
to check for the return code of the smtp session?
I've never done :p
Thanks,
Danilo
Il 21/08/12 19:05, Ruben Rögels ha scritto:
Okay, so have a look at mailcmd= option in voicemail.conf
mailbox will mean a e-mail-box in the next lines.
What you need to do is wirting a shell script or what ever
Am 01.08.2012 17:15, schrieb motty.cruz:
Hello,
I have anolog lines coming throug Dahdi to Asterisk Server, one of the
anolog lines is used for fax line. I received fax fine without any problems
using Iaxmodem with Hylafax Server. Outgoint fax is the problem, when
IAXMODEM dial out using Dahdi
Hi Dan,
my wild speculation: It's some kind of timing/synchronisation problem.
Do you use jitter buffer an/or echo cancelation?
Best regards,
Ruben
-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Dan
Hello,
I configured asterisk in sip.conf like that:
=
register = username:sec...@sipgate.de:5060/number
[sipgate-out]
port=5060
type=friend
insecure=invite
nat=yes
username=username
fromuser=username
fromdomain=sipgate.de
secret=secret
host=sipgate.de
qualify=5000
canreinvite=no
=
Hi Olivier,
I suppose you give strace a try.
It's a powerful debugging utility, you should be able to find everything
you are looking for.
best regards,
Ruben
Am 16.01.2012 11:14, schrieb Olivier:
Hi,
I've recently upgraded a system from 1.8 to asterisk 10 and also
updated spandsp while
Am 12.01.2012 18:50, schrieb mahesh katta:
I was search for free license but for this Digium require purchase any
Hardware then they can provide Free License.
But I have no Digium Device , I am using Grand stream FXO Gateway and
Asterisk.1.8.XX .
I was connected like
Am 12.01.2012 12:44, schrieb mahesh katta:
Hi,
Any one give me about FAX in Asterisk.
PSTNFXO GATEWAYASTERISK-1.4.27(OR)ASTERISK-1.8.X.X
whenever some one is Fax to PSTN its convert into pdf format
Help me any links or pdf .. for setup this. ?
Best Regards,
Mahesh
Am 12.01.2012 14:09, schrieb mahesh katta:
WARNING[6982]: pbx.c:1851 pbx_extension_helper: No application
'ReceiveFAX' for extension (macro-faxin, s,
12)
[Jan 12 18:36:00] == Spawn extension (macro-faxin, s, 12) exited
non-zero on 'SIP/gxw-000b'
Am 30.11.2011 21:47, schrieb NaJIm:
Hi All,
How can I find out One way latency from my PBX to my SIP Trunk Provider.
My SIP provider recommends a One way latency of 100ms for good Voice
quality. Ping request to their IP Address gives me a response in approx.
260ms.
Will that be good enough
Hello List,
I'm a little bit confused as I read about IEEE 801.2.q
So, my actual question is: Does a switch stating to support IEEE 801.2q
also supports VLAN trunking?
As I understand the standard, I suppose it does, but I'm not sure.
Can someone clarify this for me, please?
Thank you vermy
On 29.11.2011 11:45, Doug Lytle wrote:
Ruben Rögels wrote:
Does a switch stating to support IEEE 801.2q
also supports VLAN trunking?
I don't know if you miss-typed or not. But, 802.1q is VLAN. If it
was typed incorrectly, then yes. I just recently setup a Linux DHCP
to handle multiple
Am 29.11.2011 14:41, schrieb Douglas Mortensen:
Yes. That's exactly what 802.1q is. Technically 802.1q allows the
network devices to tag each Ethernet frame with a VLAN ID. This way if
you have 3 vlans, they can all be trunked over 1 Ethernet port by
means of tagging the VLAN ID.
-
Doug
Number of wished concurrent calls times 4 = Number of ports you'll
need to setup in rtp.conf ;-)
regards,
Ruben
Am 02.11.2011 16:05, schrieb Jonas Kellens:
Hello,
thank you for your answer...
Current range (rtp.conf) : 11500 - 11650
Current calls : 20 à 25
Is this not sufficient ??
Am 04.10.2011 10:33, schrieb Sebastian Arcus:
Hello list,
I use Asterisk with one sipgate.co.uk trunk. Asterisk connects to
sipgate.co.uk as a sip agent/client (with register = statement in
sip.conf).
If I restrict the number of ports used in rtp.conf (to 1-10005 for
example) - will
So I was hoping I would be able to set the source IP that we use when
talking to the two different SIP friends. I see externip in general
options, but is there nothing equivalent that can be set per user/peer?
Hi,
as far as I know, you cant do this on a per peer basis.
I suppose you run two
Am 08.08.2011 18:37, schrieb J Gao:
Hello, All,
I have a question about using SRV record. One of SIP provider is using
DNS SRV record. If I use IP address of the SIP proxy server I can
successfully register my Asterisk 1.8.5. But If I try to use the domain
name like:
/register =
is it possible to prevent 100% cpu usage by asterisk, with ulimit?
Hi,
it is possible, but not recommended. There is a reason, why the asterisk
process needs 100% of CPU.
What is your scenario?
How many extension?
What type of extensions? (SIP, etc)
How many concurrent calls?
Hardware?
You
on a phone, then the PBX won't answer.
Thanks.
O Venres, 5 de Agosto de 2011 00:04:02 Ruben Rögels escribiu:
Hi,
your concept using Wait() won't work here.
Try it like this:
[incoming]
exten = s,1,Dial(DAHDI/1234,30) ; This will ring the phone 30s
exten = s,n,BackGround(wellcome-message
Hi,
your concept using Wait() won't work here.
Try it like this:
[incoming]
exten = s,1,Dial(DAHDI/1234,30) ; This will ring the phone 30s
exten = s,n,BackGround(wellcome-message)
exten = s,n,Voicemail(1234)
exten = #,1,Hangup()
So, of you answer the call within 30s, you'll get the call on your
...and why do we all mess around with IT stuff and asterisk in special?
Spoiler: because we can...!
;-)
regards,
Ruben
Am 26.07.2011 10:16, schrieb A J Stiles:
On Tuesday 26 Jul 2011, Gilles wrote:
On Tue, 26 Jul 2011 07:28:27 +, Soeren Malchow (MCon)
soeren.malc...@mcon.net wrote:
Am 08.07.2011 08:58, schrieb DHAVAL INDRODIYA:
Hi All,
I am having Problem in detecting DTMF on analog lines. basically are
system is in india and telco provider is BSNL [Bharat sanchar Nigam
LImited].
We have Purchased Analog card From chinaroby.com http://chinaroby.com
which is X1600P
Hi All;
I know that incoming calls for the agent can be recorded, but how I can let
the outbound calls for the agents to be recorded? I can determine the
directory to store the outbound calls of the agents to be other than the
directory to store the incoming calls of the agents?
Hello,
after I solved my problem with the fax processing after receiving,
I got another problem while sending a fax: the header is not set properly.
I use a PHP_Script to upload a PDF file and to generate a call file.
A bash script is looking for existent call files in the web directory
and
Hello,
I have a noticed strange behavior in Asterisk 1.6.18.2 with ReceiveFax
Digium FAX Driver: 1.6.2.0_1.3.0 (optimized for i686_32).
I use a context [capi-in] for icoming ISDN calls:
==
[capi-in]
; Faxe fuer Ruben
exten = 12345,1,Macro(faxin,ruben.roeg...@jumping-frog.org,${EXTEN})
Personally I would use HTTP too.
Simple reason: You are much more flexible with it and a in most
scnearios you have a webserver running anyway.
I build some PHP-Script to provision SNOM VoIP phones for mass
deployment and it works like a charm.
Regards,
Ruben
--
Hi Ruben,
You should be looking at this thread
http://lists.digium.com/pipermail/asterisk-users/2011-June/263995.html
Presently I don't have the time to generate and send logs however soon
after my last post I did perform additional testing.
I am using ReceiveFAX using SPANDSP
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