Hi guys,
I've been trying to get my ISDN-10 line up for the past few days, but
its been going up and down. I am using OpenVox D110P card on
asterisk version 1.4.21. It seems to me like a cable problem. I tried
using Ethernet straight cable (12, 45, 36, 78) and also a straight
cable where
Try a T1 crossover cable:
http://www.voip-info.org/wiki/view/crossover+T1+cable
On Tue, Mar 31, 2009 at 12:37 AM, Steven J. Douglas stev...@moij.bizwrote:
Hi guys,
I've been trying to get my ISDN-10 line up for the past few days, but
its been going up and down. I am using OpenVox
Hi Brandon,
When using the current straight cable, it sometimes worked i.e. I can
make calls from the PSTN into the asterisk. Do you still think that I
should try a crossover cable? Thanks.
Regards,
Steve.
Brandon B. wrote:
Try a T1 crossover cable:
I'm using this:
asterisk-1.4.18 on Debian 4.0R6
asterisk-addons-1.4.7
libpri-1.4.7
zaptel-1.4.12.1
regarding the dialplan:
exten = _5.,1,Noop(llamada SIP en 'sip_sercom' a ${EXTEN});
exten = _5.,n,Wait(1);
exten = _5.,n,Dial(SIP/${EXTEN},${TIMEOUTDIAL},Tto);
exten = _5.,n,Hangup();
exten =
I'm running asterisk on Ubuntu 8.10. I have two 'register' lines in
iax.conf for registering with two remote servers. However only the
first one registers at system startup. I always have to issue an 'iax2
reload' command before * registers with the second remote host.
This only happens at
Maybe your network is not ready when asterisk first fires up?
-steve
Yahya Mohammad wrote:
I'm running asterisk on Ubuntu 8.10. I have two 'register' lines in
iax.conf for registering with two remote servers. However only the
first one registers at system startup. I always have to issue an
2009/3/31 Steven J. Douglas stev...@moij.biz:
Yahya Mohammad wrote:
I'm running asterisk on Ubuntu 8.10. I have two 'register' lines in
iax.conf for registering with two remote servers. However only the
first one registers at system startup. I always have to issue an 'iax2
reload' command
Hi,
It there any way of getting queue data from within a dialplan in order
to change call routing based on what is already happening? Something
like the following would be ideal:
exten = X.,n,Set(WAITING=${QUEUE(qname|waiting)})
exten = X.,n,Set(TALKING=${QUEUE(qname|talking)})
Can anyone
The CLI shows zap is necessary for conference recording. Can I enable
conference recording if using ztdummy or dahdi, how? ango
-- Executing [...@owt_meetme:4] MeetMe(SIP/3601-c80b4520,
5599|rcixMP) in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe
Yeah but doesnt help for extensions that have or require call-limit=1.
--- On Tue, 31/3/09, carl Lougher c_loug...@yahoo.co.uk wrote:
From: carl Lougher c_loug...@yahoo.co.uk
Subject: Re: [asterisk-users] Call-limit=1 breaks attended transfer
To: Asterisk Users Mailing List - Non-Commercial
I have found that you get good results by setting a per-device
GROUP_COUNT(), which prevents dialling if it is non-zero, and setting
call-limit to 999.
In Asterisk 1.0.x there were separate in- and out-bound call limits,
but IIRC this was pretty broken, and was removed.
See
Hi all
Somebody know with IAX support pickup call feature in the last 1.4 .X
asterisk release ?
With SIP I use features.conf and works fine, but no way to make works
with IAX.
Thanks
___
-- Bandwidth and Colocation Provided by
Kevin
That's great to hear. I'm using 1.4.21.2, but I can't see where or how
it's configurable.
I have researched the musiconhold / musicclass options in sip.conf as
well as the various documented classes and modes within
musiconhold.conf but I can't see how I tell it to just relay the media
Richard Brady wrote:
I have researched the musiconhold / musicclass options in sip.conf as
well as the various documented classes and modes within
musiconhold.conf but I can't see how I tell it to just relay the media
stream straight on.
Well, first, I was mistaken, and support for this
Ok, this is where it gets interesting. Consider the case of a PBX
which has its own MOH source and is talking via Asterisk to another
PBX.
If that PBX wants to put the call on hold while sending its own MOH,
you would probably argue that it should not send a re-INIVTE at all,
but should simply
If there is such a thing as a complete Reset, the mfg. is mum on it. The NA is
supposed to be the generic open version. However if a provider chose to lock
it up, they can make it extremely hard to get into. Vonage routinely does this
to their PAP2s, (not NA). You can Google the topic and
Christian Victor wrote:
2009/3/30 Peer Oliver Schmidt po...@theinternet.de
mailto:po...@theinternet.de
The Horst-Box Professional has a lot of problems in the ADSL area
(like stopping transfers after a dozen or so megabytes for example),
and I have had lots of needs to
Hi,
i wanna set my Asterisk to send over iax2 the dialtones as inband, that means
that in the indications.conf of Asterisk1 is set to country=us and in
Asterisk2 is set to country=at. The two Asterisk are connected together over
iax2.
So when a user from Asterisk2 calls 160 (160 is a phone on
Hello
Considering how cheap PCI modems are compared to even the cheapest
PCI hardware from Digium, OpenVox, Sangoma, etc I was wondering
why Zaptel can't be used with those to connect an Asterisk server to
a POTS line for low-level use? It just seems overkill for SOHO users
who only get a
Fred wrote:
Hello
Considering how cheap PCI modems are compared to even the cheapest
PCI hardware from Digium, OpenVox, Sangoma, etc I was wondering
why Zaptel can't be used with those to connect an Asterisk server to
a POTS line for low-level use? It just seems overkill for SOHO
On Tue, Mar 31, 2009 at 03:09:39PM +0200, Fred wrote:
Hello
Considering how cheap PCI modems are compared to even the cheapest
PCI hardware from Digium, OpenVox, Sangoma, etc I was wondering
why Zaptel can't be used with those to connect an Asterisk server to
a POTS line for
Hi,
I've a problem: I can't configure DAHDI with ech canceller OSLEC.
I have Asterisk 1.4.24 and DAHDI 2.1.0.2. I compiled also OSLEC.
But when in /etc/dahdi/systems.conf I insert value echocanceller=oslec,1-4,
command dahdi_cfg - give me an error about oslec.
Someone can help me?
On Tue, Mar 31, 2009 at 05:02:36PM +0200, Marco Sambo wrote:
Hi,
I've a problem: I can't configure DAHDI with ech canceller OSLEC.
I have Asterisk 1.4.24 and DAHDI 2.1.0.2. I compiled also OSLEC.
But when in /etc/dahdi/systems.conf I insert value echocanceller=oslec,1-4,
command dahdi_cfg
On Tuesday 31 March 2009 08:09:39 Fred wrote:
Considering how cheap PCI modems are compared to even the cheapest
PCI hardware from Digium, OpenVox, Sangoma, etc I was wondering
why Zaptel can't be used with those to connect an Asterisk server to
a POTS line for low-level use? It just seems
On the landing page of the Polycom web site there's a We're listening
nanosurvey, asking what is the one thing Polycom can do to improve their
products. The link points here:
http://polycom.zuberance.com/survey.htm
I wrote a sentence about tweaking the user interface on the IP Soundpoint
Hi All,
I just experienced a weird issue and though I'd share.
I have a pretty standard business PBX setup for a business customer,
local extensions, Linksys phones, call comes in and rings local
extension
exten = 101,1,Dial(SIP/101,20,tr)
the physical phone has call forward enabled to the users
I am about to connect to a new provider who requires 20ms payload sizes in
g729a. Is this configurable on asterisk? Is 20ms the default?
Cheers,
j
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
I believe 20 is the standard.
30 is where it might be tricky.
Tom
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Tuesday, March 31, 2009 1:37 PM
To: asterisk-users@lists.digium.com
People should use .020 ms sample rates for RTP as it's the standard. 0.030
was i think the old SPA implementations which caused MR, Roboto kind of
grabling.
You should find a way to patch your sip core i assume, but dev's could tell
you where.
We offer 0.020 , Telcos offer 0.020 , Hardware
Hi guys!!
This is something that have always bother me, hope you can help me... :)
I've 8 server connected using IAX / DUNDi, it works just fine.
However, sometimes when some of our links goes down the server takes
forever to appear back as OK at DUNDi's list and people can't call the
other Box.
I'm having a strange issue, and not really sure where to even begin to
troubleshoot it. First let me explain that I have all agents setup
locally ( local/1...@agents/n)
A call will come in and ring to the agent. When the agent answers the
call, they just hear a dial tone. Agent hangs up.
Sorry if this is a top-post, I'm using MS Outlook.
Does DSP work when all is well? What about dundi flush or dundi show
trans?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Durante
Sent: Tuesday,
Has anyone here ever had the occasion to setup a system that would
dynamically alter it's codec preferences based on trafffic? That is,
presuming that the system is on a limited bandwidth connection is would
start to prefer a compressed codec as the call volume increased?
Perhaps shifting from
Maybe something like that could be done by using set groups and counting
the number of calls, and at a specified threshold (i.e. 6 simultaneous
calls) you`d specify g729 for new calls.
Shifting an ongoing call might be impossible though.
Mike
-Original Message-
From:
Hi all,
After * starts the command queue show would not show any of the realtime
queues, but just the ones that are in the queues.conf file. In this state de
AMI would not send any QueueMemberStatus for that queues until a call is
received by that realtime queue.
Anyone knows any whay to
Gabriel Ortiz Lour wrote:
Hi all,
After * starts the command queue show would not show any of the
realtime queues, but just the ones that are in the queues.conf file. In
this state de AMI would not send any QueueMemberStatus for that queues
until a call is received by that realtime
Hi Danny,
On Tue, Mar 31, 2009 at 3:32 PM, Danny Nicholas da...@debsinc.com wrote:
Sorry if this is a top-post, I'm using MS Outlook.
Does DSP work when all is well? What about dundi flush or dundi show
trans?
Yes, when everything is OK all the calls goes just fine! Perfectly actually...
I
Mark Michelson escribió:
Gabriel Ortiz Lour wrote:
Hi all,
After * starts the command queue show would not show any of the
realtime queues, but just the ones that are in the queues.conf file. In
this state de AMI would not send any QueueMemberStatus for that queues
until a call is
Hi ,
On Tue, Mar 31, 2009 at 4:41 PM, Tiago Durante tiagodura...@gmail.com wrote:
Hi Danny,
On Tue, Mar 31, 2009 at 3:32 PM, Danny Nicholas da...@debsinc.com wrote:
Sorry if this is a top-post, I'm using MS Outlook.
Does DSP work when all is well? What about dundi flush or dundi show
Karl Fife wrote:
On the landing page of the Polycom web site there's a We're
listening nanosurvey, asking what is the one thing Polycom can do to
improve their products. The link points here:
http://polycom.zuberance.com/survey.htm
I wrote a sentence about tweaking the user interface on
On Tue, 31 Mar 2009 16:12:45 -0400, Mike wrote:
Maybe something like that could be done by using set groups and counting
the number of calls, and at a specified threshold (i.e. 6 simultaneous
calls) you`d specify g729 for new calls.
Shifting an ongoing call might be impossible though.
Mike
Hi,
I posted this on the Asterisk forum months back with no real answer() so i'll
try here :o)
Details:
There is 3 asterisk boxes called X, Y and Z.. all boxes peer with each other
via IAX2 and have dialplans setup... etc etc
There will be asterisk based clients connecting via IAX2, and for
Mhmm. Thaht's strange!
modinfo oslec
--
modinfo: could not find module oslec
and
modinfo dahdi_echocan_oslec
--
filename: /lib/modules/2.6.26-1-486/dahdi/dahdi_echocan_oslec.ko
license:GPL
author: Tzafrir Cohen tzafrir.co...@xorcom.com
description:DAHDI OSLEC
On Tue, 31 Mar 2009, Michael Graves wrote:
Has anyone here ever had the occasion to setup a system that would
dynamically alter it's codec preferences based on trafffic? That is,
presuming that the system is on a limited bandwidth connection is would
start to prefer a compressed codec as
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