Hi is a list could be off topic ;) , but someone has installed the latest
version of app_swift on centos 6 for asterisk 1.8
I'm trying to make with this manual, but have had no success
http://www.cepstral.com/en/support/telephony/faq?os=linuxsection=getting-started
gcc -I/opt/swift/include
Hi list, I need some help to improve my cdr, now in my company are
asking me how
to know which of my phone numbers are most used when receiving calls from
the PSTN and incoming the IVR
was thinking about using userfield field, and I'm trying to do, I have at
the moment 4 channel DAHDI
; DAHDI
thnk , works great ..
2013/11/14 Michael Gilleran mgille...@realtyim.com
*Try this. The warning and notice error’s are basically telling you
whats wrong*
[in]
exten = s,1,Set(CDR(userfield)=23X6)
same = s,n,Goto(in2,s,1)
Mike
*From:* troxlinux [mailto:xserverli
thnk Matthew , would be great if someone could add this to the h323 channel
, if I could, I would, but the saddest thing is that I will have to be
seeing these warning on my console
2013/10/23 Matthew Jordan mjor...@digium.com
On Wed, Oct 23, 2013 at 7:12 PM, Vladimir Mikhelson
:
ast_log(LOG_WARNING, Don't know how to indicate condition
%d on %s\n,
condition, callToken);
Ultimately the developer will take of the issue properly.
Thank you,
Vladimir
On 10/24/2013 10:04 AM, troxlinux wrote:
thnk Matthew , would be great if someone could add this to the h323
Thnk Matt,I'm looking for your mail, but can not find
2013/10/24 Matthew Jordan mjor...@digium.com
On Thu, Oct 24, 2013 at 11:41 AM, troxlinux xserverli...@gmail.comwrote:
Hi, make the change in the file chan_h323.c and comment these lines
// default:
// ast_log
Hi, I recently changed my version of asterisk to 11.XX, and I see a waning
with h323, with version 1.8 did not have these warning
I have connected one avaya ip office 500 h323 with asterisk and the 1.8
version did not have these messages
Oct 23 17:20:35] WARNING[7593][C-00aa]:
thnk Richard Mudgett for your quick response , but I have a question I am
using the asterisk 11 with ooh323 by default, I can update it?
2013/10/23 Richard Mudgett rmudg...@digium.com
On Wed, Oct 23, 2013 at 6:22 PM, troxlinux xserverli...@gmail.com wrote:
Hi, I recently changed my
Hi list, I'm trying to put my voicemail on asterisk realtime with 11.XX,
generate tables in a couple of files in the folder realtime / mysql ,
voicemail_messages.sql and voicemail.sql
the connection with mysql and odbc works well
isql asterisk useradmin xxx
2013/10/14 Warren Selby wcse...@selbytech.com
On Mon, Oct 14, 2013 at 11:13 AM, troxlinux xserverli...@gmail.comwrote:
Hi list, I'm trying to put my voicemail on asterisk realtime with 11.XX,
generate tables in a couple of files in the folder realtime / mysql ,
voicemail_messages.sql
at 12:19 PM, troxlinux xserverli...@gmail.comwrote:
res_config_mysql
[general]
dbhost = 127.0.0.1
dbname = asterisk
dbuser = root
dbpass = x
dbport = 3306
dbsock = /var/lib/mysql/mysql.sock
extconfig.conf
voicemail= mysql,asterisk,voicemail_messages
First issue I see - you've got
ok , thnk Warren , your help has been invaluable.
2013/10/14 Warren Selby wcse...@selbytech.com
On Mon, Oct 14, 2013 at 2:15 PM, troxlinux xserverli...@gmail.com wrote:
thnk Warren , I only see one warning message
[Oct 14 13:12:14] WARNING[2736][C-]: app_voicemail.c:3768
Hi list, I'm trying to attach a Avaya with Asterisk, call the extension 3241
to 1042 belonging to avaya, but only sounds rings and when I pick up the
phone keeps ringing
08-14-13 07:26:17 AM-856ms Line = 18, Channel = 1, SIP Message = Response,
Direction = From Switch, From = 3241@172.16.8.40,
Hi list, I try to install asterisk on vps server , but fails when I want to
install dahdi
[root@shark dahdi-linux-2.6.3-rc1]# make
make -C drivers/dahdi/firmware firmware-loaders
make[1]: Entering directory
`/usr/src/dahdi-linux-2.6.3-rc1/drivers/dahdi/firmware'
make[1]: Leaving directory
...@digium.com
On Tue, Jun 04, 2013 at 12:09:24PM -0600, troxlinux wrote:
You do not appear to have the sources for the
2.6.32-358.6.2.el6.x86_64 kernel installed. make: ***
[modules] Error 1
What is the output of:
ls -lat /lib/modules/`uname -r`/build
and
ls -lat /lib/modules/`uname -r`/build
in $(KSRC_SEARCH_PATH); do if [ -d $$dir ]; then
echo $$dir; break; fi; done)
endif
endif
2013/6/4 Russ Meyerriecks rmeyerrie...@digium.com
On Tue, Jun 04, 2013 at 12:50:41PM -0600, troxlinux wrote:
ls -lat /lib/modules/`uname -r`/build
lrwxrwxrwx 1 root root 50 May 23 13:06
/lib/modules/2.6.32
2013/6/4 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Jun 04, 2013 at 01:33:56PM -0600, troxlinux wrote:
thnk Russ , I have seen the Makefile, but I see many KSRC, where exactly
would
put
KVERS:=$(shell uname -r)
endif
ifndef KSRC
ifneq (,$(wildcard /lib/modules/$(KVERS)/build
if it is true I have not any hardware but I need help to solve it and I
think it could serve other future
2013/6/4 James Cloos cl...@jhcloos.com
t == troxlinux xserverli...@gmail.com writes:
t I try to install asterisk on vps server , but fails when I want to
t install dahdi
ls -lat /usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64
ls: cannot access /usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64: No such
file or directory
2013/6/4 Russ Meyerriecks rmeyerrie...@digium.com
On Tue, Jun 04, 2013 at 01:59:05PM -0600, troxlinux wrote:
Hi here again, I've tried both
excelente works fine ...
thnk russ
2013/6/4 Russ Meyerriecks rmeyerrie...@digium.com
On Tue, Jun 04, 2013 at 02:53:07PM -0600, troxlinux wrote:
ls -lat /usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64
ls: cannot access /usr/src/linux-headers-2.6.32-358.6.2.el6.x86_64: No
such
file
2011/7/27 Vladimir Mikhelson v...@mikhelson.com:
Do you have any network devices or VPN tunnels in between the Asterisk
and Avaya?
Hi , the server does not have connections vpn I have and it in the
same LAN that avaya
The reason I am asking it looks like a potential networking issue.
ok,
2009/10/5 CunningPike cunningp...@gmail.com:
I can add a recommendation for iSymphony - cheaper than dirt, easy to
configure, and the users like it.
CP
Hi , but this is free?
regardss
--
rickygm
http://gnuforever.homelinux.com
___
--
hi list , is having problems when sending a fax with hylafax and a
card sangoma A200D, when he sends it arrives to the destination but it
paginates appears in white
this is my log
Aug 20 16:11:08 voz FaxSend[6715]: MODEM Supports 40 ms, 20 ms/scanline
Aug 20 16:11:08 voz FaxSend[6715]: MODEM
2009/4/12 Martin asteriskl...@callthem.info:
1) your asterisk box talks to OpenSIPS
yes , he talk with opensips
2) in that case OpenSIPS should traverse NAT
no , my users are of opensips , asterisk is set mode comedia
3) you should not do nat=yes for that device since Asterisk talks to
señores alguien le ha presentado este problema al acceder al voicemail
o al hacer una llamada a la pstn
1940 Playing 'vm-received' (language 'es')
-- SIP/111-08d91940 Playing 'digits/yesterday' (language 'es')
-- SIP/111-08d91940 Playing 'digits/at' (language 'es')
--
uff , no me fije que envié un mensaje en español a la lista de ingles ...
I send sip log
---
Retransmitting #2 (NAT) to 192.168.10.3:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.10.3;branch=z9hG4bKf4e2.2a85e302.0;received=192.168.10.3
Via: SIP/2.0/UDP
2009/4/12 Alex Balashov abalas...@evaristesys.com:
Bastante para que no este perdido en la calle.
jajaja , yo a la verdad me asuste el verte escribir español ya que
solo en ingles te leo ... en esta lista y en la kamailio ...
saludoss
rickygm
http://gnuforever.homelinux.com
when I make a call to the pstn it shows me this error:
aximum retries exceeded on transmission
9d4a24f8-b6737...@192.168.10.19 for seqno 102 (Critical Response) --
See doc/sip-retransmit.txt.
[Apr 11 20:35:34] WARNING[3169]: chan_sip.c:1998 retrans_pkt: Hanging
up call
Bayardo you need to ask questions but exact, nobody has a glass ball
to help you
regardss
rickygm
2009/1/26 Bayardo Sanchez bayardo.sanc...@gmail.com
i have a problem need help
== Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on
'SIP/8022-b7225740'
-- Got SIP
I have dahdi-linux-2.1.0.3 in centos 5.2 and the last version oslec svn
I have installed oslec and loaded, but it doesn't work me with dahdi
modinfo oslec
filename: /lib/modules/2.6.18-92.1.22.el5/kernel/net/ipv4/oslec.ko
description:Open Source Line Echo Canceller Zaptel Wrapper
Hi list, I install dahdi-linux successfully with the module of oslec
for the echo, but when I specify it in the system.conf the echo
canceller oslec it shows me errors:
DAHDI_ATTACH_ECHOCAN failed on channel 4: Invalid argument (22)
I see that the echo cancellers is supported: mg2, kb1, sec2,
Hi list, I have for a year I have an account to call with broadvoice from
about 3 days beginning a not registered problem of, asterisk shows to a
message of error with the DNS, and my dns this working fine
WARNING[5770]: chan_sip.c:7595 transmit_register: Probably a DNS error for
registration to
my service was very well until I have not had behind, for some days I made
any change and my dns it works perfectly..
I check my account and my parameters in broadvoice and look that they
changed the out proxy for my account
if I change in the sip the parameter host = proxy-nyc.broadvoice.com
Hi amit not you if you can create a group of extensions to spy, but for
example if your extensions are of 3 digits your you can create something
like that...
exten = _*5XXX,1,ChanSpy(SIP/${EXTEN:2},bq)
it configures the hint inside the dialplan to be able to see the state of
the extensions
best
Hi list, recently install asterisk 1.4.21 in a centos 5, and after
having installer the zaptel 1.4.10.1 and libpri 1.4.4 I don't see in
the directory module any codec, and neither app.
almost install all the asterisk options
this worries to me !
alone I see these packages inside the directory
excellent contribution to the asterisk community andy congratulations Nicolas
rickygm ...
2008/5/16 Nicolás Gudiño [EMAIL PROTECTED]:
Hello,
I have finally released the queue stats package to the public.. please go to:
http://www.asternic.org/stats
To get it or see the online demo.
--
works very well , features.conf
2008/5/1 Jose P. Espinal [EMAIL PROTECTED]:
Hello List,
Does anyone here have call pickup (with *8 ) working ok on Asterisk
version 1.4.19.1 ?
Thanks in advice,
--
___
-- Bandwidth and Colocation Provided
hI list, I have some problems with a TDM01B , when I am talking on the
phone with another person it cuts himself the call, this alone I am
presented when I make calls to the pstn, with internal extensions I
don't have problems
I show them the log in the CLI
-- Nobody picked up in 68000 ms
my dialplan this ok , I have removed the lines of polarity of my
zapata.conf, and I can speak without the call cuts herself, but this
affects my billing , It should show that this ringing
;busydetect=yes
;answeronpolarityswitch=yes
;hanguponpolarityswitch=yes
It seems to be a problem of the
somebody knows some application web that allows me to call to my
internal extensions of my asterisk, example click to call.
I was proving the click to call of this example but it doesn't work
http://www.voipjots.com/2006/02/click-to-call-with-your-asteriskhome.html
greeting
rickygm
where I can find information?, I continue looking for in google and I
don't find a lot
greetings
2008/3/11, monim benayad [EMAIL PROTECTED]:
We use : Kannel + AGI
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
03.01.2008, 20:19 -0600 schrieb troxlinux:
hello list, happy new year to all, also to digium for their great work
with asterisk .
I want to make an automatic call marking an extension from my dial
plan , an example that when marking the extension 100, tell me it
records their message
hello list, happy new year to all, also to digium for their great work
with asterisk .
I want to make an automatic call marking an extension from my dial
plan , an example that when marking the extension 100, tell me it
records their message, mark the hour of their automatic call and at
the end
my final ivr is this, he works me very well
exten = 110,1,GotoIfTime(08:00-18:00|mon-fri|*|*?110,in)
exten =110,n,Dial(SIP/111,86,Tt)
exten =110,n,Dial(SIP/112,86,Tt)
exten =110,n,Hangup()
exten = 110,n(in),Set(TIMEOUT(digit)=2)
exten = 110,1,Answer()
exten = 110,2,Background(introm)
exten =
hello list, I am trying to arm an ivr for schedule of office and
outside of office
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
[in]
include =scheduleofservice|08:00-18:00|mon-fri|*|*
include =outsideofschedule|18:00-23:59|*|*|*
include
the schedule of my server this configured with -6:00, and this correct
one with the normal hour of my country, I made the change but I don't
work me
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
[in]
include =scheduleofservice|08:00-18:00|mon-fri|*|*
include
Verbosity is at least 20
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/sipurafxo-b77038e8, ) in
new stack
-- Executing [EMAIL PROTECTED]:5] BackGround(SIP/sipurafxo-b77038e8,
intron) in new stack
-- SIP/sipurafxo-b77038e8 Playing 'introm' (language 'es')
== Begin MixMonitor Recording
god blesses them and happy new year to the whole list..
rickygm
2007/12/26, Kerry S [EMAIL PROTECTED]:
Thank you.
We new what you meant.
And a Merry Christmas and a Happy New Year to you too.
On Dec 26, 2007 7:33 PM, Josué Conti [EMAIL PROTECTED] wrote:
Yep, excuse me I typed quickly.
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