I'm trying to receive a t.38 fax from a Metaswitch 7.3. I have full
control over the metaswitch, but it is in production.
I have the Metaswitch hooked to Asterisk 1.8.7.0 (middleman named s3).
Then I have the target Asterisk 1.8.7.0 with res_fax_digum 1.8.4_1.3.0
(named pbx1) registered to
2011/10/5 Adam Moffett adamli...@plexicomm.net
**
someone have been installed Asterisk (Trixbox) on VirtualBox which is
installed on a Linux host (Ubuntu server 10.04 specifically).
I want to know if it is convenient or not, and the reaseons if i should
on shouldn't do it.
I just
hi
i have used virtualbox on fedora and installed elastix (like trixbox) .
there isnt any problem .
have fun
On Tue, Oct 4, 2011 at 10:10 PM, Esteban Cacavelos
estebancacave...@gmail.com wrote:
someone have been installed Asterisk (Trixbox) on VirtualBox which is
installed on a Linux host
Dear all;
I have three agents and I need the calls to be always send for agent1 and if he
is busy then to be sent for agent2 and if he is busy then to be sent for agent3
and if all busy then to stay in the waiting until one of those three agents is
available. How?
Do I have to set the strategy
Hi,
On some systems, I can see sometimes that Asterisk crashed before being
automatically restarted (by a watch guard script).
To find the root cause of this, I'm wondering if I should let asterisk run
for ever with backtrace options DEBUG_THREADS and DONT_OPTIMIZE turned on.
Which negative side
On 10/11/2011 04:24 AM, bilal ghayyad wrote:
Dear all;
I have three agents and I need the calls to be always send for agent1
and if he is busy then to be sent for agent2 and if he is busy then to
be sent for agent3 and if all busy then to stay in the waiting until
one of those three agents
Am 06.10.2011 18:25, schrieb Olivier:
Hi,
I'm looking for an old-style receptionnist SIP phone with the
requirements bellow.
I've found one theorically matching these (Yealink plus 3 LCD
expansion module).
Would you recommend an other one ?
My requirements are :
- each LCD expansion module
On a BT line, how do I determine whether the number on an incoming call has
been deliberately withheld (by dialling 141) or is merely unavailable (e.g.
because it originated from overseas or passed through some ancient switching
equipment) ?
In the first case, I want the caller to be
Hi,
We can't read the messages in our mailbox always getting
-- SIP/tootaiAUDIO-0001 Playing
'/var/spool/asterisk/voicemail/default/100/Old/msg0002.slin' (language 'fr')
[Oct 11 13:24:50] WARNING[26778]: app_voicemail.c:7802 play_message:
Playback of message
Hello,
I have a strange situation with my asterisk 1.8.7.0 version. I compiled as
usual everything seems to be ok but from time to time when i look on my
console i get the following error message:
[Oct 11 14:44:52] WARNING[17646]: chan_sip.c:2754 _sip_tcp_helper_thread:
Failure to write to
Thanks to all for the responses. Boss calls overseas a lot and has an
unlimited data plan, so this coupled with the rates that we get for
our VoIP calls it is much cheaper than what Verizon charges.
JohnM
On 10/11/2011 1:29 AM, Jeremy Kister wrote:
On 10/10/2011 10:08 PM, Andres wrote:
I
On 10/10/2011 10:31 PM, linux guy wrote:
On Mon, Oct 10, 2011 at 8:08 PM, Andresand...@telesip.net wrote:
I would recommend Acrobits. Not free but only a few bucks. It works fine
with ATT 3G.
This begs the question... which is more expensive (and where)...
making a regular cell
Hi guys,
This may be a fairly simple answer but here's my problem
I have 2 Asterisk servers on 2 sites connected via an IAX trunk
If user 1001 on site a wants to call user 1001 on site b they will dial
51001
This is taken care of by a 5| dialing rule on the outbound route.
The problem I'm
It seems to me as though this is happening:
Asterisk A
Exten = 5,1,Dial(IAX/${EXTEN:1})
So the call goes to Asterisk B as
Exten = 5,1,Dial(IAX/1001)
So you need to change the IAX dial out command on Asterisk A to not truncate
the 5 and set up 5 on Asterisk B's inbound context so
On Tue, Oct 11, 2011 at 02:53:26PM +0100, Jonathan Archer wrote:
How can I get the 5 to stay where it is so that lookups work correctly?
is it part of the outbound CID?
My trunking (prefix 9 to get trunk access from either side of the link)
includes things like:
exten =
On Tue, Oct 11, 2011 at 08:26:22AM -0400, john Millican wrote:
Thanks to all for the responses. Boss calls overseas a lot and has an
unlimited data plan, so this coupled with the rates that we get for
our VoIP calls it is much cheaper than what Verizon charges.
My own experience with my 3G
I also installed tirxbox on virtualbox without problems.
I guess i have to open a new discussion because the problem is about
PCIPASSTHROUGH feature on virtualbox?. I cannot get to work this feature on
my virtualbox. There are a few discussions about pcipassthrough but based on
KVM or Xen. I
2011/10/11 A. M. Hoffmeister ans...@hoffmeister-online.de
Am 06.10.2011 18:25, schrieb Olivier:
Hi,
I'm looking for an old-style receptionnist SIP phone with the requirements
bellow.
I've found one theorically matching these (Yealink plus 3 LCD expansion
module).
Would you recommend an
On Tue, Oct 11, 2011 at 12:58 AM, Asterisk Man theasterisk...@gmail.comwrote:
snip
Event: QueueMember
Queue: 1
Name: 3
Location: SIP/
Membership: dynamic
Penalty: 2
CallsTaken: 0
LastCall: 0
Status: 5
Paused: 0
I would first troubleshoot why this Queue Member is showing up as
On 10/11/2011 09:57 AM, Olivier wrote:
It would be great if list-based subscriptions could be added to Asterisk
features.
They can be, if someone wrote the code to handle them.
I'm reading this list daily and strangely, I didn't see many people
mentionning or asking for this feature or for
On Fri, Aug 19, 2011 at 6:39 AM, Jim Boykin boykin...@gmail.com wrote:
convert mp3 to sln, this itself will give you quiet a big capacity boost.
How does sln boost capacity?
On Wed, Aug 17, 2011 at 12:21 PM, Morten M. Hansen m...@bellcom.dk wrote:
On 2011-08-16 21:14, Warren Selby wrote:
On 10/11/2011 01:50 AM, Jeremy Kister wrote:
I'm trying to receive a t.38 fax from a Metaswitch 7.3. I have full
control over the metaswitch, but it is in production.
I have the Metaswitch hooked to Asterisk 1.8.7.0 (middleman named s3).
Then I have the target Asterisk 1.8.7.0 with
On Tue, Oct 11, 2011 at 9:56 AM, Esteban Cacavelos
estebancacave...@gmail.com wrote:
I guess i have to open a new discussion because the problem is about
PCIPASSTHROUGH feature on virtualbox?. I cannot get to work this feature on
my virtualbox. There are a few discussions about pcipassthrough
Hi,
I'm facing a strange problem.
My setup is:
Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob
cellphone
When Alice calls Asterisk which forwards the incoming call to Bob, sometimes
Bob sees Alice's number, sometimes he sees a default CallerID (which happens
to match the dialed
AJ,
Banging my head other a similar problem here in US.
What I know so far the callerid function produces the following bitmap flag:
1. CID Private Name
2. CID Private Number
3. CID Unknown Name
4. CID Unknown Number
5. CID Message Waiting
6. CID No Message Waiting
For example, Flag=3
Hi,
Has someone successfully deflected calls using a Digium BRI board enabled
asterisk ?
How would you describe your experience ?
Which Asterisk, Libpri and Dahdi versions are required for this ?
Regards
--
_
-- Bandwidth and
Yes!, i installed the extension pack.
I attached this device to the VM.
*03:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface.*
But, when i run the VM i got this error.
VBoxManage: error: Cannot assign non-shared host interrupt handler:
VERR_RESOURCE_BUSY
On Tue, Oct 11, 2011 at 11:06 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
I'm facing a strange problem.
My setup is:
Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob
cellphone
When Alice calls Asterisk which forwards the incoming call to Bob,
sometimes Bob sees Alice's
And now they've gone back and reactivated that protocol change, breaking
chan_gtalk again. Applying the small patch from
https://issues.asterisk.org/jira/browse/ASTERISK-18301 got things
operational again. Let's see how long this one goes before they go back to
the old protocol version.
--
On Tuesday 11 October 2011, Olivier wrote:
Hi,
I'm facing a strange problem.
My setup is:
Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob
cellphone
When Alice calls Asterisk which forwards the incoming call to Bob,
sometimes Bob sees Alice's number, sometimes he sees a
Then there's also the point where it makes more sense to drop a GSM card into
your Asterisk box and get a cheap unlimited mobile to mobile plan for a SIM and
use that to transit your calls to VoIP.
Although that won't help the original asker, though, since he mentioned
Verizon.
On Oct 11,
Dear Tariq;
About elastix.org, this can be use with Asterisk or it is coming as a complete
IP Telephony, Call Center, IVR and Reporting?
Because, I do not need to install another IP Telephony on the server which
already has asterisk which is an IP Telephony, this will cause a problem in the
If the asterisk box starts up a MeetMe conference
with the 't' flag for talk only mode does asterisk send some
kind of SIP command to the devices joining the conference
to say dont send me audio back as I'll ignore it anyway as
I am the only one doing the talking.
Does that happen?
I'd like to
El 11/10/11 11:58, bilal ghayyad escribió:
Dear Tariq;
About elastix.org, this can be use with Asterisk or it is coming as a complete
IP Telephony, Call Center, IVR and Reporting?
Because, I do not need to install another IP Telephony on the server which
already has asterisk which is an IP
2011/10/11 Warren Selby wcse...@selbytech.com
On Tue, Oct 11, 2011 at 11:06 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
I'm facing a strange problem.
My setup is:
Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob
cellphone
When Alice calls Asterisk which forwards the incoming
2011/10/11 A J Stiles asterisk_l...@earthshod.co.uk
On Tuesday 11 October 2011, Olivier wrote:
Hi,
I'm facing a strange problem.
My setup is:
Alice cellphone --GSMISDN-- Asterisk -- ISDN GSM-- Bob
cellphone
When Alice calls Asterisk which forwards the incoming call to
Dear Dale;
Penalty is priority, or what exactly?
Also, how I can set the penalty of the member?
Regards
Bilal
--
Dear all;
I have three agents and I need the calls to be always
send for agent1
and if he is busy then to be sent for
On 10/11/2011 11:48 AM, Kevin P. Fleming wrote:
Well, as a starting point, I'd suggest disabling directmedia
(canreinvite) on s3. It should be possible for directmedia to be enabled
for RTP and not interfere with UDPTL, but there could still be lingering
problems there.
yep, you hit the nail
Hello,
i am having an issue with the DENY permit thingy in the Extensions.conf
whenever i use the permit deny , all the calls coming from another sip-trunk
to my asterisk ,start to fail doesn't use the Extensions dial plans that i
created
my context contain the following:
[context1]
.
Permit deny in your example applies only to incoming calls to Asterisk from the
device which authenticates as context1. A very illogical name for a SIP
peer/user/friend, but I've seen stranger things.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I do not know if order is important but I always deny all then permit what I
want to permit.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Oct 11, 2011, at 1:15 PM, hussein korbani wrote:
Hello,
i am having an issue with the DENY permit thingy in the
According to this
http://www.voip-info.org/wiki/view/Asterisk+sip+permit-deny-mask you are
only allowing traffic In from 1.2.3.4.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hussein
korbani
Sent: Tuesday, October 11, 2011 3:15 PM
On 10/11/2011 12:11 PM, Jerry Geis wrote:
If the asterisk box starts up a MeetMe conference
with the 't' flag for talk only mode does asterisk send some
kind of SIP command to the devices joining the conference
to say dont send me audio back as I'll ignore it anyway as
I am the only one doing
On 10/11/2011 02:04 PM, Jeremy Kister wrote:
On 10/11/2011 11:48 AM, Kevin P. Fleming wrote:
Well, as a starting point, I'd suggest disabling directmedia
(canreinvite) on s3. It should be possible for directmedia to be enabled
for RTP and not interfere with UDPTL, but there could still be
On Tue, Oct 11, 2011 at 3:15 PM, hussein korbani h.korb...@gmail.comwrote:
Hello,
i am having an issue with the DENY permit thingy in the Extensions.conf
whenever i use the permit deny , all the calls coming from another
sip-trunk to my asterisk ,start to fail doesn't use the Extensions
Hi,
It was already answered...
You must always deny before allow.
Regards,
Carlos
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Quoting A J Stiles asterisk_l...@earthshod.co.uk:
On a BT line, how do I determine whether the number on an incoming call has
been deliberately withheld (by dialling 141) or is merely
unavailable (e.g.
because it originated from overseas or passed through some ancient switching
equipment)
2011/10/12 Kevin P. Fleming kpflem...@digium.com:
then Asterisk *could* stop sending audio towards the device connected to that
channel.
then Asterisk *could* send it a message telling it to not bother sending any
audio.
I think in any case Asterisk must not halt any audio data stream. It
My analog card, uses a PCI slot and a 12V power connector, is configured
with 2 FXO and 2 FXS modules. I can ring handsets connected to the FXS
ports but I can't dial out from them. Is extensions.conf where I need
to make changes?
[root@robin asterisk]# cat chan_dahdi.conf
[trunkgroups]
Thanks Warren,
I have been using X-lite for member and the system from where it is running
was down at that time.
My concern was, if 'Queuestatus' shows two members as logged in for the
Queue then why not 'Queuesummary'?
Any other pointer?
--AM
On Tue, Oct 11, 2011 at 8:34 PM, Warren Selby
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