Hello Sam,
Do you have any recommendation to overcome these NAT issues?
On 8/14/15, Sam Basan sba...@bluebe.net wrote:
Hi,
It's looks like you are having NAT problem.
Packets from the provider fail reaching your box.
נשלח מטלפון נייד
בתאריך 14 באוג' 2015 15:56, Daniel - Asterisk earohua
Hello friends:
I am facing cutoffs randomly when negotiating calls.
The PBX dials the destination, the provider (softswitch) receives the
request *[1]* and sudenly the PBX hangs up the call* [2]* while the
provider is still dialing it, as a consequence the remote peer receives a
ghost call.
Hello friends,
I have been experienced suddenly stops for my Asterisk server, I do not why
is it happening. Asterisk's debug messages only tell me I have lacked g729
codec for translation to one peer minutes before the crashes occur
[2014-05-27 09:48:30] WARNING[15384][C-017c] channel.c:
Dear list:
When I call an specific number on the PSTN, the provider who holds the
destination number give back an specified sound just after admitting their
incoming calls. Is there a way to allow Asterisk to compare sounds received
to decide what is the Telco answering the call?
I'm planning to
Hello Friends:
I've just installed Asterisk 11 on my Linux (debian) server but it is not
starting up when trying with asterisk -vvc and service asterisk
start. Starting process just stop and shows: Illegal instruction as
final output.
Looking at logs I fouind at
, on Compiler Flags menu
you have to deselect BUILD NATIVE parameter. Then make, make install,
make samples, make config
Regards
El 25/11/2013 11:49, Leandro Dardini escribió:
On which kind of processor are you trying to run asterisk? Is it a real or
emulated CPU?
Leandro
2013/11/25 Daniel
2013/11/25 Daniel - Asterisk earohua...@gmail.com
Hello Friends:
I've just installed Asterisk 11 on my Linux (debian) server but it is not
starting up when trying with asterisk -vvc and service asterisk
start. Starting process just stop and shows: Illegal instruction as
final
Hello everyone, I'd changed the server and mutt started working, but I'll
test your advices and wil let you lnow ass soon as I can.
Thank you!
Elder
On Mon, Jun 24, 2013 at 7:38 AM, Larry Moore lmo...@omninet.net.au wrote:
On 22/06/2013 2:17 PM, Steve Edwards wrote:
On Sat, 22 Jun 2013,
(or equivalent log file)? If
so, mutt basically works and the messages should give some clues.
(2) What happens if you call mutt without any attachments?
I am using mutt in exactly the same way and it works.
jg
Am 19.06.2013 21:50, schrieb Daniel - Asterisk:
Hi Andre:
I added echo to provide
Hello everyone,
I'm trying to send a received fax with mutt, when I try it from the Linux
shel it works, but when trying with Asterisk's System command it doesn't.
Successful Linux command:
echo | mutt -s New fax earohua...@gmail.com -a /tmp/faxes/20130619.tif
Unsuccessful Asterisk Command:
...@gmail.comwrote:
Probably Asterisk does not know where mutt is, specify it's path in your
System command.
On 2013-06-19, at 2:03 PM, Daniel - Asterisk earohua...@gmail.com wrote:
Hello everyone,
I'm trying to send a received fax with mutt, when I try it from the Linux
shel it works
, ) in
new stack
Elder D. Arohuanca
Lima - Peru
On Wed, Jun 19, 2013 at 1:38 PM, Andre Courchesne voipfor...@gmail.comwrote:
Why echo | ?
Alsy are you sire of the content of ${FAXDEST} and ${tempfax}.
Add some NoOp before.
On 2013-06-19, at 2:29 PM, Daniel - Asterisk earohua...@gmail.com
Hey Philipp, I will try soon the new version and let you know.
Currently my users are pointing to a PBX in my local-private network with
no problems.
When I use wireshark I see my internal peers trying to send the ACK packets
4 or 5 times until hangup, at the same time the PBX are requesting
Hello everyone,
I've suffering cut offs after 6 or 7 seconds a call is answered, incoming
calls are working fine, but outgoing ones show the gollowing messages when
are being dropped:
[2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt:
Retransmission timeout reached on transmission
Users (softphones) are behind a NAT, Asterisk has its own public ip address
On Wed, May 15, 2013 at 1:30 PM, Asghar Mohammad asghar...@gmail.comwrote:
asterisk is behind nat?
On Wed, May 15, 2013 at 8:18 PM, Daniel - Asterisk
earohua...@gmail.comwrote:
Hello everyone,
I've suffering
There was 2-way audio and suddenly, the calls when down.
On Wed, May 15, 2013 at 1:30 PM, Gertjan Baarda gertjan.baa...@gmail.comwrote:
When the call is snswered, is there 2-way audio? Seems a natting issue.
On Wednesday, May 15, 2013, Daniel - Asterisk wrote:
Hello everyone,
I've
Mohammad asghar...@gmail.comwrote:
please show us peer configuration.
On Wed, May 15, 2013 at 9:46 PM, Daniel - Asterisk
earohua...@gmail.comwrote:
Users (softphones) are behind a NAT, Asterisk has its own public ip
address
On Wed, May 15, 2013 at 1:30 PM, Asghar Mohammad asghar
.
if someone need to create a new interface i can help.
On Wed, Apr 10, 2013 at 11:22 PM, Daniel - Asterisk
earohua...@gmail.comwrote:
Hello Brynjolfur Thorvardsson,
Can I take a look at you CDR reporting tool?
I'm planning on using it on Postgresql but MySQL could be used too.
Thank you
Hello all,
I need the bootrom.ld file to set up some Polycoms I have
Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A
I've publiched on my FTP files downloaded from
http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html
(3.2.3
: 316-688-8208
From:Daniel - Asterisk earohua...@gmail.com
To:Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date:04/12/2013 12:42 PM
Subject:[asterisk-users] Polycom Soundpoint IP 330 provisioning
Sent
This can be useful too: Application, main: Label=BOOT, Version=3.2.3.0021
29-Mar-07 16:05
It is the log the phones are sending to my FTP
Thanks!
Elder
On Fri, Apr 12, 2013 at 12:50 PM, Daniel - Asterisk earohua...@gmail.comwrote:
Hello Kevin,
Could you please tell me where I can found
and it helps to have the proper
background before getting started.
-Dave
On 04/12/2013 01:50 PM, Daniel - Asterisk wrote:
Hello Kevin,
Could you please tell me where I can found the 'application' my phones
are looking for?
I've already downloaded spip_ssip_vvx_3_2_3_release_**sig combined
Hello Brynjolfur Thorvardsson,
Can I take a look at you CDR reporting tool?
I'm planning on using it on Postgresql but MySQL could be used too.
Thank you!
Elder D. Arohuanca
dCAP
Lima - Peru
On Fri, Feb 10, 2012 at 11:55 AM, asterisk jobs asteriskcod...@gmail.comwrote:
No, that doesn't do
and it seems to be an abandoned project. If you know
about a tool or product to download CDR reports and update SIP realtime
tables please let me know.
Regards,
Elder D. Arohuanca
dCAP
Lima - Peru
On Wed, Mar 20, 2013 at 11:58 AM, Daniel - Asterisk earohua...@gmail.comwrote:
Hello everyone,
I wonder
Thanks Jon!, I think it'd be time to go back to the old known databases, I
was motivated for light performance of SQLite
Elder
On Tue, Apr 9, 2013 at 1:21 PM, jon pounder j...@inline.net wrote:
On 04/09/2013 01:44 PM, Daniel - Asterisk wrote:
sqlite is not really a multiuser dbms, so
Hello everyone,
I wonder if there's a product that I can install on my debian-based server
to extract CDRs (it'd be better if Excel's downloads are available), also
it would be desirable if I can access additional table to update rows (e.g.
sip for realtime)
Please let me know what you know.
/25/2013 11:48 AM, Daniel - Asterisk wrote:
Hello Mahendra,
I've just installed Asterisk from source on my Raspberry Pi model B,
this is what I did:
sudo apt-get install build-essential
sudo apt-get install libncurses5-dev
sudo apt-get install libssl-dev
sudo apt-get install libxml2
Hello Mahendra,
I've just installed Asterisk from source on my Raspberry Pi model B, this
is what I did:
sudo apt-get install build-essential
sudo apt-get install libncurses5-dev
sudo apt-get install libssl-dev
sudo apt-get install libxml2-dev
cd /usr/src/
sudo wget
, 2013 at 2:14 PM, Daniel - Asterisk earohua...@gmail.comwrote:
I did follow instructions in debian without problems, this issue arise
when trying with Centos 5.8 and 5.9.
On Debian 6.0.6 i wrote:
./ast_tls_cert -C 10.200.x.y -O Company -d /etc/asterisk/keys/
and I got ca.cert which
6, 2013 at 10:59 PM, Daniel - Asterisk
earohua...@gmail.comwrote:
Hi List,
I'm trying to set my Asterisk 1.8.20.1 with TLS on CentOS 5.9, it was
easy and straightforward with Debian 6.0.6, but when I introduce this
command on CentOS:
#./ast_tls_cert -C 10.200.108.17 -O MyCompany -d /etc
at 12:23 AM, Daniel - Asterisk
earohua...@gmail.comwrote:
Hello Kepin,
I don's know if there's a difference, I changed order with the same
result. Did you find a different script with CentOS?
Elder
On Wed, Feb 6, 2013 at 6:16 PM, kepin sinatra insanlaks...@gmail.comwrote:
hi daniel
Hi List,
I'm trying to set my Asterisk 1.8.20.1 with TLS on CentOS 5.9, it was easy
and straightforward with Debian 6.0.6, but when I introduce this command on
CentOS:
#./ast_tls_cert -C 10.200.108.17 -O MyCompany -d /etc/asterisk/keys/
I got this error message:
hostname: Unknown host
Same
Thank you Carlos,
What does mean 'por-out'?
I'm expecting 1 min/month in out.
Elder
On Thu, Nov 29, 2012 at 5:50 PM, Carlos Alvarez car...@televolve.comwrote:
On Thu, Nov 29, 2012 at 3:22 PM, Daniel - Asterisk
earohua...@gmail.comwrote:
Hello List,
Since I'm looking for a new
Hello List,
Since I'm looking for a new VoIP provider for US origination/termination, I
will very appreciate if you can chare your experience with Flowroute,
Vitelity and Voip.ms
Thanks in advance!
Elder D. Arohuanca
dCAP 1497
Lima - Peru
--
g729 no
no yes45
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jan 12, 2012, at 3:00 PM, Kevin P. Fleming wrote:
On 01/12/2012 11:57 AM, Daniel - Asterisk wrote:
The simplest
On Wed, Jan 11, 2012 at 6:10 PM, Daniel - Asterisk earohua...@gmail.comwrote:
Hi folks,
I'm having problems when I try to record my calls using MixMonitor or
Monitor. Calls are working well and audio quality is good.
But I just can't get recorded audio in one leg with both applications
Hi folks,
I'm having problems when I try to record my calls using MixMonitor or
Monitor. Calls are working well and audio quality is good.
But I just can't get recorded audio in one leg with both applications. It
happens with internal calls too. As it seems, the problem is my g729
licensing
Hi,
I had some problems with sip peers losing connection suddenly without real
network issues. In my case, it was useful to refresh the table used for
real time configuration, we made an script for Postgres like this
PGUSER=user PGPASSWORD=password vacuumdb --full --table 'sip_buddies'
Hello all,
I recently found this when looking an IAX trunk:
context=*
Does it have a special meaning or is it the same like 'default'?
Thanks,
Elder
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
, Shaun Ruffell wrote:
On Mon, Aug 15, 2011 at 04:56:04PM -0500, Daniel - Asterisk wrote:
Hi guys,
Did you get some explanation? I'm suffering the exact issue. It could be
I
need some additional dependencies than before?
Elder, I'm assuming that you too have no pri show channels? If so
Could you please share a little sample showing how to get connected to AMI
with php?
Thanks a lot!
Elder
On Mon, Aug 1, 2011 at 3:40 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 08/01/2011 03:35 PM, Paul Belanger wrote:
On 11-08-01 04:24 PM, Daniel - Asterisk wrote:
?php
function
Hi guys,
Did you get some explanation? I'm suffering the exact issue. It could be I
need some additional dependencies than before?
Regards,
Elder
On Mon, Jul 25, 2011 at 7:59 AM, Soeren Malchow (MCon)
soeren.malc...@mcon.net wrote:
Dear Shaun,
First, thanks for you answer
The installed
:04PM -0500, Daniel - Asterisk wrote:
Hi guys,
Did you get some explanation? I'm suffering the exact issue. It could be
I
need some additional dependencies than before?
Elder, I'm assuming that you too have no pri show channels? If so are
you
running from packages or source, and if from
Hi guys, I hope you could help me.
I am trying to get connected through AMI but something is not working. Both
php code and manager.conf were working well in asterisk 1.4
1. Sometimes it gets connected and sometimes it doesn't:
== Connect attempt from '192.168.25.241' unable to authenticate
On the CLI write: sip show channels
If there are lots of bye channels you have the same problem than me.
I've tried waiting with the call generator -sipp- and channels
finished when there are a few. But they're not ending faster enough
when I send lots of concurrent calls.
Elder
2011/7/5, A E
I'm trying to get working SIPp with media but something is wrong (it's
working well without media), please help:
This is the command I send at SIPp server:
./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err
This is the result I see:
Last Error: Aborting call on
the
endpoints.
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
On Jul 4, 2011, at 3:29 PM, Daniel - Asterisk earohua...@gmail.com wrote:
I'm trying to get
://tinyurl.com/3hx5652
On Thu, May 12, 2011 at 11:52 AM, Daniel - Asterisk
earohua...@gmail.comwrote:
Hello Everyone,
I wonder if someone could share a manual about using SIPp for Asterisk's
testing.
I'll be gratefull
Regards,
Elder Arohuanca
Lima - Peru
On Tue, Sep 30, 2008 at 12
Hello Everyone,
I wonder if someone could share a manual about using SIPp for Asterisk's
testing.
I'll be gratefull
Regards,
Elder Arohuanca
Lima - Peru
On Tue, Sep 30, 2008 at 12:59 PM, zac wolfe zac.wo...@gmail.com wrote:
Sipp looks pretty good! I don't know how I missed this one. This
-0500, Daniel - Asterisk wrote:
Hi Danny,
Could you please let me know what function do I use to get if the
queue is full?
Elder
On Mon, Feb 7, 2011 at 10:42 AM, Danny Nicholas da...@debsinc.com
wrote
Dear list,
I want to avoid sending calls to a queue when it is full. From the fact that
'maxlen' must be at least 1 (I wish it could be zero but it isn't) I'd like
to know if there's a way to do it. Setting the Queue() timeout to a little
value is not the most suitable option.
I'm using asterisk
Hello guys,
I have this problem when a call is received in my PBX:
(Caller) -- (Redirecting Service) -- (E1 PRI) -- (Asterisk PBX) --
(Internal Phone)
Reception works fine, but when conversation finishes and the agent at
internal phone hangs up, the call at caller's side is still alive for
many
Hello list,
I'm sending calls to a queue in the attended way, that is, *1.* the original
call is put on hold, *2.* a second line is open to call the queue,
*3.*after an agent is connected the original call is transfered to its
final
destination.
1. Zap/1-1 -- SIP/agentA-tag1
2.
number and the way you prefer.
*Step 7:*
Choosing Dialing Mode: Protocol Management - FXO Settings, I select One
Stage.
Hope it helps.
Elder Daniel
On Wed, Dec 2, 2009 at 2:08 PM, Daniel - Asterisk earohua...@gmail.comwrote:
I've set at Protocol Management FXO Settings Dialing Mode == One
It was a pending draft I forgot to send.. sorry.
On Fri, Jan 29, 2010 at 1:23 PM, Matt Collins mcoll...@ccdservice.netwrote:
Damn, where were you 6 months ago? ;)
Daniel - Asterisk wrote:
Just if it is helps someone, based on information at the blog:
http://allabouthobby.blogspot.com
Hi list,
I'm trying to get ready the MP-104 FXO to use qith my box, but when I send
calls I hear only dial tone and after a few seconds I get busy signal.
I very appreciate your advices.
Command line results and SIPconfigurations follows:
*CLI*
-- Executing [7991696...@total:1]
interface drop-down
control).
Let us know if this helps.
JDB
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Daniel - Asterisk
*Sent:* Wednesday, December 02, 2009 12:33 PM
*To:* Asterisk Users Mailing List - Non-Commercial
Hi Cristina,
You can find meanings in queuelog.txt (or queuelog.tex in * 1.6), it's
attached.
Daniel
On Fri, Sep 11, 2009 at 11:14 AM, Maria Cristina Bayno falls_m...@yahoo.com
wrote:
Hello Team,
Can you help me on this? I have attached here the queue logs of my
asterisk. I've searching a
I've changed it and is working now, I though the second parameter was the
name of the databse accordingly to
/usr/src/asterisk/configs/extconfig.conf.sample
Thank you,
Daniel
On Mon, Sep 7, 2009 at 10:26 PM, Tilghman Lesher tles...@digium.com wrote:
On Monday 07 September 2009 17:16:12 Daniel
Hi list,
I hope someone could help me. I've started using Asterisk 1.6.0.14 to get
queue logs in real time with odbc (our databases are all PostgreSQL) but
it's not working. However, cdr odbc is working well. When asterisk starts
next message appears:
WARNING[4217] config.c: Realtime mapping for
Good morning,
I'm having suddenly cut-offs and I don`t know why. It's been hapenning since
I enabled cdr_odbc/func_odbc in my system.
We use func_odbc to register some queue member's events (login, pause, etc.)
at an external DB ('remoto' connector) and to uptade local tables at a local
DB
Dear all,
I wanna know what can I do to get the PBX's clock from
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Dear all,
I wanna know what can I do to get the PBX's clock from an external AMI
server, especially with Asterisk-Java Library.
Thanks by your answers.
Elder Arohuanca Lagos
t. +51 1 994149553
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:00 AM, Steve Howes st...@geekinter.net wrote:
On 28 Apr 2009, at 16:49, Daniel - Asterisk wrote:
Dear all,
I wanna know what can I do to get the PBX's clock from
You sir, are made of fail.
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Try two.
On Sun, Feb 22, 2009 at 9:11 PM, Daniel - Asterisk earohua...@gmail.comwrote:
Hi,
I've just installed DAHDI at two PBXs as follows:
*PBX-1PBX-2*
FXO - FXS
When I try to send calls from PBX-1 to PBX-2 I just receive the message:
Starting simple switch
Hi,
I've just installed DAHDI at two PBXs as follows:
*PBX-1PBX-2*
FXO - FXS
When I try to send calls from PBX-1 to PBX-2 I just receive the message:
Starting simple switch on 'DAHDI/1-1
It seems like if PBX-1 couldn't send DTMFs, but when I set immediate=yes at
Hi everyone,
Currectly I'm having some troubles to get correct status of my calls throug
ISDN lines, when outbound calls don't get its destination I always receive
NO ANSWER as ${DIALSTATUS} despite the fact I know the target number
doesn't exists or is busy at that time.
Maybe there is
It was a lack of free space in disk, because a big load of recorded calls
and logs.
Daniel
On Thu, Oct 23, 2008 at 12:40 PM, Daniel - Asterisk [EMAIL PROTECTED]wrote:
I'm restarting my system without solution and I've extended my call limit
to 10 calls (asterisk.conf) to avoid call
Hi,
I'm using queue configuration as follows:
- queues from* queues.conf*
- queue_members from *external Database thru ODBC*, using* Local channels
* as interface
- sip extensions from *external Database thru ODBC*
When a call is sent from queue to an interface (local channel), it
I've restarted the service and zombie channels were killed.
Daniel
On Wed, Oct 15, 2008 at 3:29 PM, Steve Murphy [EMAIL PROTECTED] wrote:
On Tue, 2008-10-14 at 17:24 -0500, Daniel - Asterisk wrote:
Hello everyone,
I'm getting DIALSTATUS=CHANUNAVAIL when a call is trying to get one
Suddenly my system crash whem I see core show channels are increasing until
reaches its limit at asterisk.conf
It seems channels (Local, Zap, SIP) are not being closed.
The problem persists and I don't know what to do
Please help me!
___
-- Bandwidth
My version is 1.4.21.1
On Thu, Oct 23, 2008 at 11:38 AM, Daniel - Asterisk [EMAIL PROTECTED]wrote:
Suddenly my system crash whem I see core show channels are increasing until
reaches its limit at asterisk.conf
It seems channels (Local, Zap, SIP) are not being closed.
The problem persists
- Asterisk [EMAIL PROTECTED]wrote:
My version is 1.4.21.1
On Thu, Oct 23, 2008 at 11:38 AM, Daniel - Asterisk [EMAIL PROTECTED]wrote:
Suddenly my system crash whem I see core show channels are increasing
until reaches its limit at asterisk.conf
It seems channels (Local, Zap, SIP
Hello everyone,
I'm getting DIALSTATUS=CHANUNAVAIL when a call is trying to get one of my
queue interfaces, despite the fact it is free at that time, can you give
help?
1. I see many sip channels from that extension:
[EMAIL PROTECTED] asterisk -rx *sip show channels* |grep 648
Peer
Yes it is, every counter is set to zero:
asterisk -rx module reload app_queue.so
Regards,
Daniel Arohuanca
t.+51 1 994149553
Peru
On Thu, Oct 2, 2008 at 12:05 PM, Atis Lezdins [EMAIL PROTECTED] wrote:
On Thu, Oct 2, 2008 at 7:32 PM, voip crazy [EMAIL PROTECTED] wrote:
When the asterisk a
I've upgraded my version to 1.4.21.1 since last week and things seem to be
fine.
thanks,
Daniel
On Wed, Jul 23, 2008 at 11:27 AM, Chento Arohuanca [EMAIL PROTECTED]wrote:
I´ll be upgrading my box this weekend and let you know the consequences.
I´m new at the community and it would be good
Hi everyone,
I really need your help. Just now my queue member status are not being
refreshed correctly, when a call is answered the status is set as UNKNOWN
instead of IN USE. After the call is hanged up the state persists as
UNKNOWN.
I have tried using module reload app_queue.so but the only
This problem was fixed when I upgraded my box to version 1.4.21.1
Thanks everyone,
Daniel
On Mon, Jun 30, 2008 at 2:01 PM, Chento Arohuanca [EMAIL PROTECTED]wrote:
I forgot it!, I'm using Asterisk 1.4.19.1 version.
On Mon, Jun 30, 2008 at 1:47 PM, Chento Arohuanca [EMAIL PROTECTED]
wrote:
Hi friends,
Where can I get some information to understand messages like the following
ones?
*NOTICE[6455] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary
D-channel of span 1*
*NOTICE[6455] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel
of span 1*
* ERROR[6455] chan_zap.c:
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