Hi, sorry for my insistence but I would your aid for my problem.
Thanks.
--
Salvatore.
- Original Message -
From: Sasa s...@shoponweb.it
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, June 03, 2010 9:51 AM
Subject
you setup
everything else that leads to trunk.
Respectfully
Michael D Mosier
Ftoc Certified
On Jun 9, 2010 2:53 AM, Sasa s...@shoponweb.it wrote:
Hi, sorry for my insistence but I would your aid for my problem.
Thanks.
--
Salvatore.
- Original Message -
From: Sasa s
Hi, I have tried with a some change in IVR configuration but the result
isn't changed, I have tried with Enable Directory and Enable Direct Dial
disabled, also I have tried with timeout=1 but nothing is changed !
My IVR configuration is:
trixbox1*CLI dialplan show ivr-2
[ Context 'ivr-2'
: Wednesday, June 02, 2010 5:34 PM
Subject: Re: [asterisk-users] Delay in IVR
On Mon, 2010-05-24 at 14:41 +0100, Kingsley Tart wrote:
On Mon, 2010-05-24 at 15:09 +0200, Sasa wrote:
HI, I have in 'inbound route' a IVR, with press 1 or 2 the destination
call
is always a ring group called '600
HI, I have in 'inbound route' a IVR, with press 1 or 2 the destination call
is always a ring group called '600', my problem is that after press 1 (but
this problem is present also with press 2) before that the inbound call is
transfer to extension pass 10/11 seconds !
In attach log file about
What is the verdict? There was one positive response, but would like to
hear a few more. In addition, what I am looking at is FXO ports to be used
with GSM gateways, so any recommendations for specific cards are welcomed.
From my experience with PRI cards, I am a little biased toward Sangoma.
Dear all,
Picked up some more BT usb adapters and got a flood of error messages as
follows:
hci_scodata_packet: *hci0 SCO packet for unknown connection handle 92*
Anyone has any idea how to deal with this?
Sasa Bobek
___
-- Bandwidth and Colocation
yes, only one device per USB dongle.
On Sat, Jul 18, 2009 at 4:22 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
Hello,
I read on the wiki that chan_mobile supports one device per dongle. Is
this still the case?
From the official website, there is very little info but this line
In general, I found it hard to get chan_mobile working straight out of the
box, and although there is a great effort to make it so, phone manufacturers
are not helping by making command sets and BT implementations different from
device to device, SW version to SW version. Elastix seems to have
successfully? If so, I
will consider the switch. I can't jump to another distribution easily
because I have a working environment that will make really hard the
migration.
On Sat, Jul 18, 2009 at 10:57 AM, Sasa Bobek sasa.bobek...@gmail.comwrote:
In general, I found it hard to get chan_mobile
Truth is you don't need anything more then Asterisk to configure a call
center
On Mon, Jul 13, 2009 at 2:19 PM, ashish chauhan
ashishchauhan07...@gmail.com wrote:
Dear all,
I am new to asterisk.i like to configure call center using
asterisk.please can anyone tell me open source
Just google/bing it. http://voip-info.org/wiki/view/chan_mobile
On Thu, Jul 9, 2009 at 12:56 PM, Olivier oza-4...@myamail.com wrote:
2009/7/2 Carlos Ruiz Diaz carlos.ruizd...@gmail.com
Check chan_mobile. Now is mature enough to be used in a server with low
CPS.
The USB connectivity will be
Could not agree more. I had chan_mobile up and running with an older
version of Trix, but never managed to recreate it with the latest versions.
Other people I talked to even suggested that it was made on purpose. With
elastix the only problem I had was the missing mobile.conf.example, but you
I had loads of issues when trying i on trix, but the same procedure worked
like a charm with elastix.
On Sun, Jul 5, 2009 at 7:09 PM, Razza razz...@gmail.com wrote:
I've been failing to get chan_mobile working, so am looking to the experts
to help :o)
I have followed this guide -
Chan_mobile supports SMS with a limited number of phones
On Fri, Jul 3, 2009 at 4:14 PM, Steve Totaro stot...@totarotechnologies.com
wrote:
Great for Chan_Mobile and GSM modem for SMS in Kannel or if Asterisk
supports SMS over GSM modem.
I know chan_mobile had SMS in the future at one point
Same here.
On Fri, Jun 26, 2009 at 2:40 PM, Razza razz...@gmail.com wrote:
Hi all, does anyone know of an application that will run in Windows (in my
case users PC's) and behave in a similar fasion to chan_mobile? I'd like the
app to register with asterisk, then talk to a (or a number of)
Hi all,
Before I start with analog GSM gateways I wanted to check if maybe someone
actually got a working combination of chan_mobile and bluez. If you do
please share specifics like versions, phone, BT chipset, any other relevant
info.
Thanks,
Sasa Bobek
Subject: Re: [asterisk-users] Cisco 7941G Auth
Hey Sasa,
I have templates of all the files you need here (SEP file, extension
file):
http://dave.vc/wordpress/wp-content/uploads/2008/11/phoneadd.zip
If you need further assistance, let me know.
Thanks
Dave
-Original Message
Hi all,
We have been planing for a long time to set up GSM mobile trunks for
termination, and were planing on going with analog GSM adapters connected to
a VoIP gateway. Should we be concerned with such a set-up as far as voice
quality and other issues are concerned? Any experiences with GSM
The price difference is HUGE. Analog i about 66% cheaper.
On Tue, Jun 23, 2009 at 12:44 PM, Gordon Henderson
gordon+aster...@drogon.net gordon%2baster...@drogon.net wrote:
On Tue, 23 Jun 2009, Sasa Bobek wrote:
Hi all,
We have been planing for a long time to set up GSM mobile trunks
, 23 Jun 2009, Sasa Bobek wrote:
Hi all,
We have been planing for a long time to set up GSM mobile trunks for
termination, and were planing on going with analog GSM adapters
connected to
a VoIP gateway. Should we be concerned with such a set-up as far as
voice
quality and other
of the FXS/FXO port is about 50E.
Sasa
On Tue, Jun 23, 2009 at 1:17 PM, Gordon Henderson
gordon+aster...@drogon.net gordon%2baster...@drogon.net wrote:
On Tue, 23 Jun 2009, Sasa Bobek wrote:
The price difference is HUGE. Analog i about 66% cheaper.
But you then need some sort of analogue
be something like this:
device
loadInformationSIP41.8-0-2SR1S/loadInformation
/device
And you shouldn't need the tlv file.
-Jonathan
On Fri, Jun 19, 2009 at 8:25 AM, Sasa s...@shoponweb.it wrote:
David Gibbons wrote:
I've found that different types of TFTP servers return
Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco
7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem
is that Cisco phone isn't authenticated on Asterisk.
In tftp directory I have:
apps41.1-1-1-15.sbn
cnu41.3-1-1-15.sbn
copstart.sh
.
--
Salvatore.
- Original Message -
From: John Novack jnov...@stromberg-carlson.org
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, June 19, 2009 4:38 PM
Subject: Re: [asterisk-users] Cisco 7941G Auth
Sasa wrote:
Hi
Discussion
Subject: Re: [asterisk-users] Cisco 7941G Auth
Sasa wrote:
Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco
7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my
problem
is that Cisco phone isn't authenticated on Asterisk.
In tftp directory I
Discussion
asterisk-users@lists.digium.com
Sent: Monday, March 09, 2009 6:51 PM
Subject: Re: [asterisk-users] Portech MV3770 Caller-ID
2009/3/9 Sasa s...@shoponweb.it
Hi, I have a problem with Asterisk-1.4.22 (with TB 2.6.2) Portech
MV-370,
my problem is that when arrived an external call I
Hi, I have a problem with Asterisk-1.4.22 (with TB 2.6.2) Portech MV-370,
my problem is that when arrived an external call I don't view (on my
internal phone) the phone number but I have the number extension that is
configured on MV-370.
The MV-370 configuration is:
Mobile to Lan Table :
0 *
Hi, I use Asterisk-1.2.26 (with Trixbox-2.1.12) and Portech MV-370 and my
problem is that when I try to call an external mobile phone via Portech I
have alway busy and in log file:
Called Portech/348xxx -- Got SIP response 486 Busy Here back from
192.168.1.2-- SIP/Portech-086e5ee0 is busy ==
to a version
it can cope with.
If its not asking for any files then usually what I have done is to go
to the lowest SIP version 2 or 3 for changing from the call manager to
SIP and reset the phone to factory defaults and try and get it to start
the change again
Cheers Duncan
Sasa wrote:
Hi Duncan
P0S3-06-2-00.sbn
-rwxr-xr-x 1 root root 486570 2007-04-27 14:51 P0S3-06-2-00.bin
-rwxr-xr-x 1 root root 392214 2007-04-27 14:51 P0S30202.bin
I can't recall if I need all the 08-6 versions
Cheers Duncan
Sasa wrote:
Hi Duncan,
I have tried more times to make the reset phone but is displays
whether its asking for SIP
or SCCP files, and if SIP which version of firmware for the phone
Cheers Duncan
Sasa wrote:
Hi Dave,
I don't view nothing in tftp server because the phone is stopped on start
screen with displayed 'upgrading' and MAC address..I don't understand
what
happened after
/var/log/messages'
should do the trick.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sasa
Sent: Monday, October 13, 2008 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7906g SIP
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sasa
Sent: Thursday, October 09, 2008 10:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7906g SIP
Hi Dave,
I have tried restore to factory default value (as you have recommended
and you should be good to go.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sasa
Sent: Monday, October 13, 2008 9:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7906g SIP
Hi,
I have try
: Thursday, October 09, 2008 1:27 PM
Subject: Re: [asterisk-users] Cisco 7906g SIP
Sasa schrieb:
I need other files other than those obtained with
cmterm-7911_7906-sip.8-0-4sr1.cop ??
cmterm is the callmanager software. You need to get the non-callmanager
SIP-software. Contact your local Cisco
Subject: Re: [asterisk-users] Cisco 7906g SIP
Sasa,
You can actually just rename the .cop file to a .tar.gz file. Cisco
doesn't have (to my knowledge) any non-callmanager SIP software. The SIP
load is just a SIP load, not a SIP load unique to generic SIP or
callmanager.
Dave
: David Gibbons [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, October 09, 2008 2:59 PM
Subject: Re: [asterisk-users] Cisco 7906g SIP
Sasa,
Sometimes I have to do a hard reset of the phone in order to get
Hi, sorry for my insistence but for me is a big problem ! :-( ...someone
have the same problem ?
Thanks in advance.
--
Salvatore.
- Original Message -
From: Sasa [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
the current hardware in the status, if its SIP it will be
something like POS-0806... (I haven't got a phone handy to check) but
there is a reasonable amount of info on voipinfo about the process
Cheers Duncan
Sasa wrote:
Hi, I have a problem with Cisco 7906G and SIP protocol use with Asterisk
Hi, I have a problem with Cisco 7906G and SIP protocol use with Asterisk
1.2.26.
I have uploaded in my tftp server the firmware
'cmterm-7911_7906-sip.8-0-4SR1' that use 'SIP11.8-0-4SR1S.loads' and in
SEPmacaddress.cnf.xml I have:
loadInformationSIP11.8-0-4SR1S/loadInformation
..but in tftp
Hi, I use Asterisk 1.2.17 with BRIstuffed-0.3.0-PRE-1y-e (with Trixbox
2.2.12) and I have three ISDN card with chipset HFC on PCI slot, my problem
is that after a inactivity period one o two isdn card are disconnected:
asterisk1*CLI zap show status
HFC-S PCI A ISDN card 0 [TE] layer 1 AC
HFC-S
Hi, I use:
Trixbox-2.2.4
FreePBX-2.3.1.0
Asterisk-1.2.17
BRIstuffed-0.3.0-PRE-1y-e
Zaptel-1.2.19
..with two ISDN cards, often but occasionally the dial out failed but is
possible to receive external call.
My zapata.conf conf is:
[trunkgroups]
[channels]
language=it
context=from-pstn
Tzafrir Cohen wrote:
New:
loadzone=it
defaultzone=it
span=1,1,3,ccs,ami
bchan=1,2
dchan=3
span=2,1,3,ccs,ami
bchan=4-6
dchan=6
..in zapata.conf I have:
; new part:
switchtype=euroisdn
signalling = bri_net
priindication=outofband
group = 1
channel = 1-2
group = 2
channel = 4-5
Tzafrir Cohen wrote:
You have been quite short on details. For instance: what distribution of
Linux? What version of Zaptel?
Do you have another Zaptel card? It seems you either have two zaphfc
cards or one dual-BRI card. If so, the procedure is slightly more
complicated, as you basically
-users@lists.digium.com
Sent: Thursday, November 29, 2007 1:50 AM
Subject: Re: [asterisk-users] Fw: Remove a TDM Card
On Wed, Nov 28, 2007 at 04:59:22PM +0100, Sasa wrote:
Hi, sorry but perhaps I don't have explained clearly my problem...now I
have
a box voip that must be replace with another
Hi, sorry for my insistence but this is a big problem for me..my steps for
remove card are ok ?
Thanks.
--
Salvatore.
- Original Message -
From: Sasa [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, November 26, 2007 4:25 PM
Subject: [asterisk-users] Remove
the card in. Bigger issue is getting around the
renumbering of channels when you remove hardware at the bottom
--
Salvatore.
- Original Message -
From: Sasa [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, November 26, 2007 4:25 PM
Subject: [asterisk-users
Hi, I would like remove a Digium TDM2400P from Asterisk (version 1.2.13) box
but when I remove card from the PC after reboot Asterisk not started
correctly.
On box now with TDM Card I have:
[EMAIL PROTECTED]:~# lsmod
Module Size Used by
zaphfc
Hi, also I have called Cisco suport to ask how to use SIP protocol on Cisco
7941G (and my Astersik), the their answer is the following:
..SIP Firmware for the 7941G phone only works with Call Manager 5.x. You
must have CCM 5.x to use this firmware, is needeful to buy a CCM
license for use
Hi, on 7941G is needful the Call Manager license, the firmware for SIP use
is available (with login) on 7912 and 7940.
Thanks.
--
Salvatore.
- Original Message -
From: Patrick [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi, sorry for my intrusion... I have the same problem with Cisco 7941G, can
I do buy the the Smartnet registration also for 7941G or this license is
available only for 7940G ?
Thanks.
--
Salvatore.
- Original Message -
From: Cory Andrews [EMAIL PROTECTED]
To: Asterisk Users
Hi, I have a asterisk/voip newbie and I am sorry if my quetion is banal.
I used in my private LAN, Express Talk on Windows XP and Asterisk latest
version on Fedora Core 4 , with this configuration in Express Talk
Lines menu:
Setting for Line: Default Line Settings
Full 'friendly' Display Name:
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