Hello,
I had posted this mail some time back, Having got no responses I tried
one suggestion I received in another thread and replaced all IAX
trunks with SIP trunks. That has resolved this issue. Asterisk now
does not hit more than 100% CPU and there is no call disturbance. CPU
usage is now is
Hello,
I've created Call Center with Asterisk (1.6.0.5). Call Center's agents
are not Asterisk SIP user's, but other's voip gw SIP 5 class users.
Everything works fine, except when one agent wants transfer call to other
agent. They do it with flash hook. So and two voip gws (Asterisk and other
Hi,
Sorry about posted a protected link, I forgot we'd closed the site to
spammers since we don't use it anymore. The useful content was
re-posted in our list.
---
URI
OK, have done. Issue ID 0015963.
Steve Hindmarch
BT Design
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Tilghman Lesher
Sent: 24 September 2009 15:11
To: Asterisk Users Mailing List - Non-Commercial
Hi!
I have installed Asterisk 1.6.1 on SuSE Linux (from OpenSuSE
repository), configured http.conf and manager.conf according to the
manual.
However, whenever I try to connect to Asterisk manager via web browser
(http://192.168.0.1: , where xxx port defined in asterisk -
http.conf),
Hi!
I have installed Asterisk 1.6.1 on SuSE Linux (from OpenSuSE
repository). As a clients I use XLite on Mac, all on the same LAN.
Server where asterisk is is barely loaded at 5% CPU, have a lot of RAM
and plenty of disk space on LEVEL 5 RAID.
Calls to another SIP server (also asterisk)
On Fri, 2009-09-25 at 13:01 +0300, andreil1 wrote:
Hi!
I have installed Asterisk 1.6.1 on SuSE Linux (from OpenSuSE
repository). As a clients I use XLite on Mac, all on the same LAN.
Server where asterisk is is barely loaded at 5% CPU, have a lot of RAM
and plenty of disk space on
A small clarification - a package I'm referring to is called Asterisk GUI, not
Asterisk Manager. Sorry for mistype.
On Friday 25 September 2009 01:00:53 pm andreil1 wrote:
Hi!
I have installed Asterisk 1.6.1 on SuSE Linux (from OpenSuSE
repository), configured http.conf and manager.conf
# lsusb
Bus 002 Device 003: ID e4e4:1160
Bus 002 Device 001: ID :
Bus 001 Device 003: ID 0403:e6c8 Future Technology Devices
International, Ltd
Bus 001 Device 002: ID 0403:6001 Future Technology Devices
International, Ltd FT232 USB-Serial (UART) IC
Bus 001 Device 001: ID :
Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6
install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1
FXO port and 1 FXS port. I have a POTS line from my phone company attached
to the POTS line.
I have asked for disconnect supervision to be provisioned on
Pls see below output.
I would like to remove the last 3 peers.
How can I do this ?
Thx
Vai
[trixbox ~]# /usr/sbin/asterisk -rx sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Format
Hold Last Message
192.168.1.126(None) MjkzYjNiMmY 00101/4 0x0
Have you looked into minimum message length and/or silence parameters?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Brown
Jr
Sent: Friday, September 25, 2009 10:20 AM
To: Asterisk Users Mailing List -
Hello,
I have one problem and I need to disable dtmf (disable rfc2833, info and
inband) on one (other peers must support dtmf) SIP peer . Is it possible?
Workaround would be use g729 codec with dtmfmode=inband.
Maybe there is better solution?
Thanks for help.
--
Pagarbiai / Best
On Fri, 25 Sep 2009, Giedrius Augys wrote:
Hello,
I have one problem and I need to disable dtmf (disable rfc2833, info and
inband) on one (other peers must support dtmf) SIP peer . Is it possible?
Workaround would be use g729 codec with dtmfmode=inband.
I don't think that would
find the code in dahdi and put printk so you can see in dmesg or
/var/log/messages
if that gets ever detected
also you may try hanguponpolarityswitch=yes in chan_dahdi.conf
Martin
On Fri, Sep 25, 2009 at 10:19 AM, Stephen Brown Jr
stephen.brow...@gmail.com wrote:
Ok so this is officially
Giedrius Augys wrote:
Hello,
I have one problem and I need to disable dtmf (disable rfc2833, info and
inband) on one (other peers must support dtmf) SIP peer . Is it possible?
Workaround would be use g729 codec with dtmfmode=inband.
Maybe there is better solution?
Thanks for help.
I have, but I wanted to see if I could fix this problem before I started
experimenting with that.
On 9/25/09 11:24 AM, Danny Nicholas wrote:
Have you looked into minimum message length and/or silence parameters?
rather you could
disallow=alaw
disallow=ulaw
and set dmtfmode=inband
since only g711 codec is clear enough to detect dtmf reliably
Martin
On Fri, Sep 25, 2009 at 10:30 AM, Giedrius Augys voi...@gmail.com wrote:
Hello,
I have one problem and I need to disable dtmf (disable rfc2833, info
If you're really going to pursue this, I'd buy stock in Zoloft - I've got a
TDM400 and TDP410 and they both drive me nuts on POTS issues.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Brown
Sent: Friday, September
2009/9/25 Martin asteriskl...@callthem.info
rather you could
disallow=alaw
disallow=ulaw
and set dmtfmode=inband
since only g711 codec is clear enough to detect dtmf reliably
Martin
On Fri, Sep 25, 2009 at 10:30 AM, Giedrius Augys voi...@gmail.com wrote:
Hello,
I have one
On Friday 25 September 2009 10:19:39 Stephen Brown Jr wrote:
Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6
install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1
FXO port and 1 FXS port. I have a POTS line from my phone company attached
to the POTS
I've tried the hanguponpolarityswitch parameter as well with no success :(
Any clues where in the DAHDI code I might find reference to disconnect
supervision timing?
On 9/25/09 11:39 AM, Martin wrote:
find the code in dahdi and put printk so you can see in dmesg or
/var/log/messages
if that
Have you been able to find a satisfiable config? My biggest headache is
the useless voicemails being left if the caller hangs up during the
greeting, otherwise it appears to work as intended.
Agree on the Zoloft, this is driving me nuts!
On 9/25/09 11:45 AM, Danny Nicholas wrote:
If
2009/9/25 Martin asteriskl...@callthem.info
rather you could
disallow=alaw
disallow=ulaw
and set dmtfmode=inband
since only g711 codec is clear enough to detect dtmf reliably
Martin
On Fri, Sep 25, 2009 at 10:30 AM, Giedrius Augys voi...@gmail.com wrote:
Hello,
I have one
No joy, but a suggestion. The default voicemail call is Voicemail(xxx,b) or
Voicemail(xxx,u) for busy or unavailable. If you did Voicemail(xxx,s)
(silent) you could playback the voicemail greeting from the dialplan, then
check the line using the ChanisAvail function before launching
RSCL Mumbai schrieb:
Pls see below output.
I would like to remove the last 3 peers.
How can I do this ?
[trixbox ~]# /usr/sbin/asterisk -rx sip show channels
Use grep. (See `man grep`.)
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
On Fri, Sep 25, 2009 at 04:30:19PM +0200, Loic Didelot wrote:
# lsusb
Bus 002 Device 003: ID e4e4:1160
Bus 002 Device 001: ID :
Bus 001 Device 003: ID 0403:e6c8 Future Technology Devices
International, Ltd
Bus 001 Device 002: ID 0403:6001 Future Technology Devices
On Fri, Sep 25, 2009 at 10:27 PM, Philipp Kempgen philipp.kemp...@amooma.de
wrote:
RSCL Mumbai schrieb:
Pls see below output.
I would like to remove the last 3 peers.
How can I do this ?
[trixbox ~]# /usr/sbin/asterisk -rx sip show channels
Use grep. (See `man grep`.)
I may not
On Fri, Sep 25, 2009 at 11:19:39AM -0400, Stephen Brown Jr wrote:
Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6
install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1
FXO port and 1 FXS port. I have a POTS line from my phone company attached
to the
, recordingcheck,20090925-133244,1253899963.12) in new
stack
[Sep 25 13:32:44] -- Launched AGI Script
/var/lib/asterisk/agi-bin/recordingcheck
[Sep 25 13:32:44] recordingcheck,20090925-133244,1253899963.12: Inbound
recording not enabled
[Sep 25 13:32:44] -- DAHDI/1-1AGI Script recordingcheck completed
Thank you,
the mounting part was missing in my setup.
Loïc.
On Fri, 2009-09-25 at 20:01 +0300, Tzafrir Cohen wrote:
On Fri, Sep 25, 2009 at 04:30:19PM +0200, Loic Didelot wrote:
# lsusb
Bus 002 Device 003: ID e4e4:1160
Bus 002 Device 001: ID :
Bus 001 Device 003: ID
Hi,
I've seen this ISDN subaddress feature added to libpri.
Which countries are using it ?
How is this billed ? Do you have to pay an extra to your telco to benefit
from this subaddresses ?
Cheers
___
-- Bandwidth and Colocation Provided by
On Fri, 2009-09-25 at 16:58 -0500, das sandesh wrote:
Hi All,
I have a senario where we have multiple locations and all have the
ability to call using 1NX pattern, so we have created multiple
contexts so the outbound goes fine, but while transfer occurs (after
picking the inbound
Note to those Americans scratching their heads over this: nano-BTS systems
are not so unusual in the Netherlands, unlike the USA.
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
Sent:
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