- Original Message -
> From: "John Novack SCII_U"
> To: "Asterisk Users Mailing List, Non-Commercial Discussion"
> , "Andrew Martin"
>
> Sent: Monday, October 8, 2018 4:29:41 PM
> Subject: Re: [asterisk-users] Dropped calls when all
Hello,
I am running Asterisk 11.17 with DAHDI 2.9.0 and an OpenVox A800P with 8x
analog
POTS lines coming into my Asterisk server from the phone company. Internally, I
have about 180 SIP clients defined in sip.conf. What appears to be happening is
that if existing calls are consuming all 8
.
-- oscar wilde
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digital_signature_gpg.asc
Description: Digital GPG signature (see http://martin-krafft.net/gpg/sig-policy/999bbcc4/current
And then use Dial(Sip/Provider1/callednumber)
No registration should be neccessary in this case unless you want to receive
calls as well. (you will need to change type to friend too in this case...)
Martin
>
> Therefore the username alone will not be unique, but the combination of
> usernam
ell number to
receive/send calls in the office if possible.
This document describes some details, I just dont want to reinvent the wheel
if somebody did already...
https://www2.eecs.berkeley.edu/Pubs/TechRpts/2013/EECS-2013-18.p
e your asterisk can
get correct public IP - "externip=" ...
Martin
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Check out the new Asterisk community forum at: https://community.aster
s possible that a newer Asterisk will work with the v4
kernel, but in any case I'd be interested in finding out the root of
the problem at hand.
Any hints appreciated. Thank you!
>>> sip.conf <<<
[general]
nat=auto_force_rport,auto_comedia
[mtvic-main]
md5secret=xxx
context=mt
SIP phone, e.g.
[0020fe8200de]
; abbreviated
md5secret=abcdabcdabcdabcadbcdabcadbcdabcd
context=in-martin
setvar=DEFAULT_ORIGIN=11
When I make a call with this phone, the dialplan has access to
${DEFAULT_ORIGIN}.
However, when a call comes in through the sipgate trunk and gets
rou
also sprach martin f krafft <madd...@madduck.net> [2015-09-02 14:16 +0200]:
> However, when a call comes in through the sipgate trunk and gets
> routed to the in-trunk-sipgate context, the ${FOO} variable is not
> set and thus not available from the dialplan.
Thanks to [TK]-Fend
for additional debug information?
Thanks,
Andrew Martin
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- Original Message -
From: John Kiniston johnkinis...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, July 29, 2015 11:53:13 AM
Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are
- Original Message -
From: John Kiniston johnkinis...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, July 28, 2015 12:12:05 PM
Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are
Hello,
I am running Asterisk 11 on CentOS 6.x. I have configured several queues as
follows in extensions.conf:
exten = s,1,Queue(myqueue,rtnC,18)
same = n,Background(user_unavail)
same = n,WaitExten(10)
exten = 1,1,Voicemail(@my-vm,s)
This rings the phones in the queue for 18 seconds. If no
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more and more.
-- sir walter scott
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.
-- g. niruta
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seconds
Andrew Martin wrote:
snip
Joshua,
As a mitigation for this problem, could I increase the timerb option in
sip.conf
to a large value, say 1 hour (instead of the default 32 seconds)? What
other
consequences would there be from this change?
I don't know if chan_sip
32 seconds
Andrew Martin wrote:
Since some packet loss is a possibility, I assume the protocol has
mechanisms
for dealing with it. What should be happening differently in the
communication
when packet loss occurs? Should the phone just be re-sending the OK,
instead of
printing 0
seconds
Andrew Martin wrote:
- Original Message -
snip
Most noteworthy is that the phone seems to send the OK for cseq 103, but it
seems that the asterisk server never received this OK, which is why it kept
re-transmitting the INVITE (103). Is this OK supposed to go
- Original Message -
From: Steve Davies davies...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 13, 2015 11:39:29 AM
Subject: Re: [asterisk-users] Retransmission Timeout results in dropped
calls after 32
- Original Message -
From: Andrew Martin amar...@xes-inc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, May 11, 2015 4:18:58 PM
Subject: Re: [asterisk-users] Retransmission Timeout results in dropped
calls after
- Original Message -
From: Joshua Colp jc...@digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, May 11, 2015 1:24:53 PM
Subject: Re: [asterisk-users] Retransmission Timeout results in dropped
calls after 32
seconds
Andrew Martin wrote:
- Original Message -
snip
By doing a number of test calls today, I have managed to reproduce this
while
sip debugging was on, so I have that information available now as well:
http://pastebin.com/ZJqzdvY3
This was a call from 113 to 146 via
- Original Message -
From: Andrew Martin amar...@xes-inc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, May 11, 2015 1:35:07 PM
Subject: Re: [asterisk-users] Retransmission Timeout results in dropped
calls after
- Original Message -
From: Andrew Martin amar...@xes-inc.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, May 8, 2015 5:12:28 PM
Subject: [asterisk-users] Retransmission Timeout results in dropped calls
after 32 seconds
timeout problem?
Here is my sip.conf:
general]
directmedia=no
directrtpsetup=no
dtmfmode=rfc2833
context=internal
allowsubscribe=no
qualify=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
localnet=10.10.32.0/255.255.248.0
[123]
secret=11
host=dynamic
type=friend
Thanks!
Andrew Martin
James,
The WaitExten()s just provide a pause between the two Queue() calls to
let the first group of phones finish ringing. In this example I am ringing
the same group (queue_level_1) twice, however in a real-world scenario I
would ring queue_level_1 and then ring queue_level_2 which each have a
- Original Message -
From: Guenther Boelter gboel...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Tuesday, May 5, 2015 1:05:44 AM
Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
Looking into it further, in my case it does not appear to be a
- Original Message -
From: Administrator TOOTAI ad...@tootai.net
To: asterisk-users@lists.digium.com
Sent: Friday, May 1, 2015 6:42:38 AM
Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
Le 01/05/2015 00:05, Andrew Martin a écrit :
- Original
to debug these problems? Since it is intermittent, I am not
always able to reproduce (sometimes the calls work just fine).
Thanks,
Andrew Martin
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- Original Message -
From: Administrator TOOTAI ad...@tootai.net
To: asterisk-users@lists.digium.com
Sent: Thursday, April 30, 2015 4:43:33 PM
Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk
Hello,
when I am in the Asterisk CLI, I can exit with 'exit' or 'quit'. Ctrl+d
has no effect. Is there any way to bind Ctrl+d to exit/quit ?
Also, when I am in asterisk CLI, I can use command history and readline
functions such as CTRL+r to search. But not all functions are available.
For
to this are dropped.
Sometimes they only allow some ports, eg. 80, 443, or protocols (http...)
Martin
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(bash, mysql, ..)
$ cat /etc/inputrc
\e[5~: history-search-forward
\e[6~: history-search-backward
is there a way to make it work in asterisk ?
thanks,
Martin
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allowed. (h.264 in my
case) Not only there is no trnscoding available but also video codec
negotiation doesn't (or didn't...) work properly.
Martin
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New
is established (and both parties' phones
suggest that), the traffic flows only via Asterisk (directmedia
= update,nonat), so the problem is likely to be found there, no?
Before I shower you with debug logs and traces, I am wondering if
this sounds familiar to anyone…?
Thanks,
--
martin | http
to 46.244.255.146:8058 (type 00, seq 026000, ts
3578986600, len 000160)
-- SIP/lehel-sipgate-3573 answered SIP/lehel-martin-3572
-- Remotely bridging SIP/lehel-martin-3572 and
SIP/lehel-sipgate-3573
Sent RTP P2P packet to 46.244.255.146:8058 (type 08, len 000160)
Sent RTP
Hello,
Im trying to modify the 'From' field in my sip headers in order to include
extra info (user=tel) as it follows:
From : sip:005114824403@200.91.0.146;user =tel
However asterisk is still doing this header:
sip:111@1.1.1.1;tag=as167b4b82
Is there a way to accomplish this?
? Fax detection doesn't work reliably over
compressed codecs (g729 etc...), in my case didn't work at all...
try to add:
directmedia=no
disallow=all
allow=ulaw
allow=alaw
to your peer definition.
Martin
---
Tato zpráva neobsahuje viry ani jiný škodlivý kód - avast! Antivirus je aktivní.
http
, SIP only works over TCP.
Where do u download the SIP firmware usually for your Cisco IP Phones?
Search for cmterm-7941_7961-sip.8-3-1.zip
I also have some other files here but I don't remember what was the reason for
them :-(
Martin
Your kindly help is highly appreciated.
Regards
Bilal
Hi,
I am new to Asterisk. I'm using it behind a kamailio sip-router to provide
voicemail boxes to sip-users.
I followed these instruction:
http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+Kamailio+1.5.x
to set everything up, using ARA with a MySQL DB.
After a few
when compressed codec (g729, gsm...)
is in use? I've read its unreliable but does not work at all for me.
(Asterisk 1.8.13 installed from Debian repository)
Thanks
Martin
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Am 21.01.2013 14:21, schrieb Olivier:
Hello,
I've seen some desktop SIP phones (Snom, Yealink) intregrate a VPN
(OpenVPN ?) client.
Has someone experience to share about that particular feature ?
Is this experience rather successful ?
My underlying question is can one supervise and
Am 13.01.2013 03:17, schrieb Adolphus Enaboifo:
Hi List Members ,
its been about one months since I built my first Asterisk server.
What I want to know is: are there ways to make Asterisk take recorded
reminders.
This is the scenario I have in mind.
1 You place a call to a specific extension
Am 11.01.2013 02:42, schrieb Christopher Harrington:
Wow, that seems wildly bandwidth inefficient. Is it possible to do
multicast VoIP?
Snom phones[*] do support multicast streaming. You can setup an
IP port combination that the phone will accept audio at; once
stream data starts arriving,
?
--
martin | http://madduck.net/ | http://two.sentenc.es/
if one cannot enjoy reading a book over and over again,
there is no use in reading it at all.
-- oscar wilde
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the settings.
These are — I think — my base requirements. What would you add?
--
martin | http://madduck.net/ | http://two.sentenc.es/
quick!! act as if nothing has happened!
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:
http://madduck.net/blog/2012.10.19:configuration-management/
--
martin | http://madduck.net/ | http://two.sentenc.es/
gentoo: the performance placebo.
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only have my own three use-cases to refer to, and
I would probably impose my own paradigms…
Does anyone already have something done into that domain?
--
martin | http://madduck.net/ | http://two.sentenc.es/
she is absolutely inadmissible into society. many a woman has a past,
but I am told
and dial a third party, and when
I come back to the conference room, I bring along the third party.
Put differently: I don't really want my correspondents to have to do
anything, just wait and listen to MOH.
--
martin | http://madduck.net/ | http://two.sentenc.es/
nullum magnum ingenium sine
to have clear separations (unless you cannot clearly
separate), but I am not convinced that this decreases complexity.
--
martin | http://madduck.net/ | http://two.sentenc.es/
toleranz heißt, die fehler der anderen entschuldigen.
takt heißt, sie nicht bemerken
also sprach Administrator TOOTAI ad...@tootai.net [2012.11.08.1018 +0100]:
For a 3 way conference, all those days phones are able to do this.
Yeah, except I want Asterisk to handle that, not my phone (which
might lose reception or run out of battery etc.).
--
martin | http://madduck.net
(Linux) to manage setups?
Cheers,
--
martin | http://madduck.net/ | http://two.sentenc.es/
not the truth in whose possession any man is, or thinks he is, but
the honest effort he has made to find out the truth, is what
constitutes the worth of man
that Asterisk knows what to do.
Thanks,
--
martin | http://madduck.net/ | http://two.sentenc.es/
the human brain is like an enormous fish --
it is flat and slimy
and has gills through which it can see.
-- monty python
spamtraps
do recall a university using Asterisk that
provided 10 logins for everyone, i.e. if my username was 12345, then
12345[0-9] would all be valid SIP login names using the same
password. Any idea how this was done? 10 stanzas? ;)
--
martin | http://madduck.net/ | http://two.sentenc.es/
brevity
for converting
channels into conferences, but I could not get any of them working.
Does anyone have a working example they would be willing to share?
Thanks,
--
martin | http://madduck.net/ | http://two.sentenc.es/
picture yourself in a boat on a river
with tangerine trees and marmelade skies
sort of automated tool (chef / puppet). This
will help you get on the right path.
The new machine for the 6th site is up and running (provisioning
(not image-based) took less than half an hour). What now? ;)
--
martin | http://madduck.net/ | http://two.sentenc.es/
science without religion
to keep scouting
for existing solutions. I prefer not to cook my own solutions but to
adopt and contribute to existing (free) solutions.
--
martin | http://madduck.net/ | http://two.sentenc.es/
whatever you do will be insignificant,
but it is very important that you do
Hi list,
I'm trying to setup an asterisk 1.8.0 with MFC/R2, but I'm having these
error messages loading chan_dahdi:
module load chan_dahdi.so
ERROR[9241]: chan_dahdi.c:11848 mkintf: Signalling requested on channel
1 is MFC/R2 but line is in ISDN PRI signalling
ERROR[9241]: chan_dahdi.c:16180
On Wed, 2010-11-10 at 17:10 +0200, Tzafrir Cohen wrote:
On Wed, Nov 10, 2010 at 11:42:18AM -0300, Martin Spinassi wrote:
Hi list,
I'm trying to setup an asterisk 1.8.0 with MFC/R2, but I'm having these
error messages loading chan_dahdi:
module load chan_dahdi.so
ERROR[9241
On Wed, 2010-11-10 at 10:51 -0500, Miguel Molina wrote:
El 10/11/10 10:31, Martin Spinassi escribió:
On Wed, 2010-11-10 at 17:10 +0200, Tzafrir Cohen wrote:
On Wed, Nov 10, 2010 at 11:42:18AM -0300, Martin Spinassi wrote:
Hi list,
I'm trying to setup an asterisk 1.8.0 with MFC/R2
Hi Olivier,
I remember having had a similar discussion a few years ago. I will paste
my postings from around May 2007 further down.
First, I did not try sending SMS over VOIP to the phone, just over Voip
to an ATA and then over analogue line (or ISDN) to the phone. So I have
no idea wether the
algorithm
to your country's busy signal frequencies ... Just record the busy
signal with ztmonitor and send to someone for code patch...
regards
Martin
On Wed, Jul 28, 2010 at 4:54 PM, bruce bruce bruceb...@gmail.com wrote:
Hmmwhat about call waiting?
You mean, when a call comes in on that specific
Well for the best test you can call in on that line and fire Echo()
app and then you'll see if the lines
hangup by themselves ... is you use fxsks/fxs_ks signaling and it's
supported by your lines
then it's that that makes remote hangup possible
regards
Martin
On Fri, Jul 30, 2010 at 9:12 AM
t38pt_usertpsource is meant for such a case...?
[asterisk 1.6]-LAN-[NAT gateway]inet-[NAT gateway]-LAN-[ATA]-[FAX]
Has anybody some positive experience with this?
Any idea why NAT messes up the port numbers?
Martin L
your password
if you were nice enough to set it within a known statistical easy guess
2) either use complicated passwords and sip accounts other than
100-199 1000- or install the fail2ban
Martin
On Fri, Jun 11, 2010 at 4:55 PM, sean darcy seandar...@gmail.com wrote:
This is a small 12 line
if you know IP then ban with iptables
iptables -A INPUT -s IP -j REJECT
Martin
On Fri, Jun 11, 2010 at 8:41 PM, Martin asteriskl...@callthem.info wrote:
When will you people learn ... you set the secret=
and it's one of the many frequent passwords most people sets out of
being lazy
lol when then if he knows the IP of his provider plus a few phones he
can just allow these ... and problem solved forever
Martin
On Fri, Jun 11, 2010 at 9:02 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Fri, 11 Jun 2010, Martin wrote:
if you know IP then ban with iptables
iptables
On 8.5.2010 00:40, Jeff Brower wrote:
Martin-
checkout new open source voipmonitor.org SIP packet sniffer. I've
developed it for my telco company and I've decided to share it.
Testing and contributions are welcome!
VoIPmonitor is open source live network packet sniffer which analyze
Hello,
I've choosen only MOS-LQE because it is calculated only on network
parameters, which is loss, burstinnes and delay (which is converted to
loss by jitterbuffer simulator). It does not takes into account voice
(payload). There is no effective objective methods (today) which
predicts MOS.
On Sat, May 8, 2010 at 2:34 PM, mosbah abdelkader
mosbah.abdelka...@gmail.com wrote:
Thank you Martin,
So the MOS-LQE does not inform bout payload itself but predicts the MOS
based on networks metrics
yes exactly. LQE is Listen Quality Emodel (E-model is parametric model
which takes
Martin
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asterisk-users mailing
Hi,
checkout new open source voipmonitor.org SIP packet sniffer. I've
developed it for my telco company and I've decided to share it.
Testing and contributions are welcome!
VoIPmonitor is open source live network packet sniffer which analyze
SIP and RTP protocol. It can run as daemon or analyzes
it on their end.
Fail2ban can send you a Whois info about every blocked IP. Im just not sure if
any kind of reporting will help :-(
Zeeshan A Zakaria
Martin L
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Are you sure writing to the right list???
Martin
- Original Message -
From: Sarfaraz Chougule
To: sarfaraz.choug...@gmail.com
Sent: Monday, April 05, 2010 4:54 PM
Subject: [asterisk-users] Please sign Petition - Stop Child Labour
Hello Friends,
Kind request to you all
Dear Colleagues,
I installed a Wellgate 3804A and overnight lines on all this with frying,
putting other lines Wellgate 3804A is well, so I guess it's a problem the first
team which is already out of warranty, anyone know how can I fix this? or where
to send it in or capital Buenos Aires to
dahdi config I need for this dedicated type of
T1?
Thanks
Martin
:-)
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H323 seemed to be enabled by default, so I just disabled the H.323
module as we do not use it.
Rob
How did you disable it? I dont see any module containing h323 in its name.
(ast. 1.9)
Martin
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Randall
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you think the bots don't know this password ???
Martin
On Tue, Jan 12, 2010 at 11:43 AM, Juan C. Villa juan...@villafam.com wrote:
Hey guys,
I've been running asterisk on my server for some time now (currently
running Asterisk 1.6.2.0). I am having security issues with my SIP
accounts
Martin
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/DAHDI chunk size and that directly affects system load.
Second question - the document explains how to change the chunk size for
Sangoma hardware. Is there a general way to do that for DAHDI?
Thanks is advance!
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Rd, Bldg 1
Rochester, NY 14624
Office
reports BUSY after some time. I couldnt test it too much because of crappy
Verizon's DSL line in that location (doesn't work well even after 6
months...:-( )
Martin
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asterisk-users
registry entries...
if it didn't ... it would be just a crippled sip endpoint
lets say more ... Asterisk can do whatever you want it to do (within
reason and technical boundaries);
just code it in or request a feature
Martin
Thanks
John
2009/12/3 Tim Nelson tnel...@rockbochs.com:
- John
had to deal
with patch before. I usually just take the release version of asterisk and
install it as is.
P.S. I would like to keep the version 1.4.21 because it's the last version that
I know of that use Zaptel by default instead of DAHDI.
Thanks
Martin
not as easy like opensbc
Martin
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AsteriskNow use CentOS 5 and it comes preinstalled with dahdi and
asterisk with the freepbx GUI interface and it seems to be missing all
the dev packages
Martin
On 2009-11-17, at 02:19, Olivier wrote:
2009/11/17 Martin Roy m...@mac.com
I was previously using an old computer running
question should I use hpec
or oslec with my TDM400 card? I also tried to recompile dahdi to use
oslec (before I found that Digium had hpec) but then I get an error
message that the source of my kernel cannot be found so I can never
actually compile a new version of dahdi.
Thanks
Martin
incoming call handling peer is at the very end.
Pretty wacky.
I am hopefully back on the road though with working caller ID as well.
Marty
On Nov 14, 2009, at 11:10 AM, Martin Joseph wrote:
Ok I am replying to myself, because I still don't have this figured
out,, but I think I have more
Ok I am replying to myself, because I still don't have this figured
out,, but I think I have more info.
On Nov 5, 2009, at 8:57 PM, Martin Joseph wrote:
Hello again Asterisk people.
I am running Asterisk 1.42 on an old PowerPC ibook. I have had this
deployed for several years now
Grandstream HT503. For me works just fine. 1xFXO 1xFXS port. Each port has its
own sip account.
Martin
- Original Message -
From: jonas kellens
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, November 12, 2009 5:38 AM
Subject: Re: [asterisk-users
If you doesn't need transcoding, you doesn't need any licenses...
Martin
- Original Message -
From: Vinícius Fontes vinic...@canall.com.br
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, November 06, 2009 11:43 AM
Subject: Re
On Nov 6, 2009, at 5:14 AM, John A. Sullivan III wrote:
On Thu, 2009-11-05 at 20:57 -0800, Martin Joseph wrote:
Hello again Asterisk people.
I am running Asterisk 1.42 on an old PowerPC ibook. I have had this
deployed for several years now, with pretty good results.
Recently I added
Hello again Asterisk people.
I am running Asterisk 1.42 on an old PowerPC ibook. I have had this
deployed for several years now, with pretty good results.
Recently I added a callerid service to my landline (qwest).
I am using the audiocodes MP114 2fxo/2fxs gateway, which is an
outstanding
/chan_dahdi.conf
Martin
On Mon, Nov 2, 2009 at 10:28 AM, Vieri rentor...@yahoo.com wrote:
--- On Sat, 10/31/09, Martin asteriskl...@callthem.info wrote:
On Sat, Oct 31, 2009 at 5:27 AM,
Tzafrir Cohen tzafrir.co...@xorcom.com
wrote:
I'm not sure if handling of overlap hasn't changed
since
...
just do pri debug span 1
Martin
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overlapdial=yes in zapata.conf/chan_dahdi.conf
google it out
Martin
On Fri, Oct 30, 2009 at 6:54 AM, Vieri rentor...@yahoo.com wrote:
Hi,
I have a PRI euroisdn link between an Alcatel PBX and Asterisk.
I'm having some trouble with overlap dialing.
Suppose I dial '874053' from an Alcatel
...
at least that's how it was designed to work
Martin
On Fri, Oct 30, 2009 at 8:25 AM, Vieri rentor...@yahoo.com wrote:
I forgot to mention that I already have overlapdial=yes in zapata.conf.
Besides, overlapdial=yes is only for dialing out from Asterisk. Anyway,
that option is set.
Any other
so you're either testing it wrong or it's been broken since that
worked fine years ago
you may try adding the . after then extension ... I don't remember
maybe it's needed
eg:
exten = 1004000.,...
but better yet
exten = 100400XX,...
Martin
On Fri, Oct 30, 2009 at 10:08 AM, Vieri rentor
no, I meant this
s,1,Set(H=us)
s,n,Dial(,,g)
s,n,Set(H=them)
h,1,Noop(${H} hanged up)
That might or may not work ... since I didn't actually check it
Martin
On Mon, Oct 26, 2009 at 9:05 AM, Danny Nicholas da...@debsinc.com wrote:
So this *should* work??
[outgoing]
- exten = s,1,Dial(DAHDI
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