I want to use JabberSend in my dialplan, but I saw that my Asterisk does
not support Jabber.
Also I have nowhere a module res_jabber.so...
So I thought I'd rebuild my Asterisk. In menuselect I saw that
res_jabber was dependent of 'iksemel' and 'gnutls'.
In my yum repositories I can find a gnutls.
6-23 at 13:13 -0400, Steve Totaro wrote:
>
>
>
> On Tue, Jun 23, 2009 at 9:36 AM, jonas kellens
> wrote:
>
> Do you understand what is happening ?
>
>
> -- Executing [0473775...@intern:2]
> Dial("SIP/twinkle-08de049
Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/twinkle-0a0567f8' status is
'CONGESTION'
Really destroying SIP dialog
'340811e66bc43ba36fb5d507066fc...@192.168.2.2' Method: INVITE
Really destroying SIP dialog 'xfdsxekzwoxc...@localhos
Do you understand what is happening ?
-- Executing [0473775...@intern:2] Dial("SIP/twinkle-08de0490",
"SIP/3starsnet/0473775006") in new stack
-- Called 3starsnet/0473775006
-- SIP/3starsnet-08d70ea8 is making progress passing it to
SIP/twinkle-08de0490
-- Got SIP response 500 "Servi
-0400, Steve Totaro wrote:
>
>
>
> On Tue, Jun 23, 2009 at 8:09 AM, jonas kellens
> wrote:
>
> -- Executing [0473775...@intern:1]
> NoOp("SIP/twinkle-088e6ea8", "conversation to GSM") in new
> stack
> -- E
-- Executing [0473775...@intern:1] NoOp("SIP/twinkle-088e6ea8",
"conversation to GSM") in new stack
-- Executing [0473775...@intern:2] Dial("SIP/twinkle-088e6ea8",
"SIP/3starsnet/0473775006") in new stack
-- Called 3starsnet/0473775006
-- Got SIP response 482 "Loop Detected" back fr
write to in the table, most notably
> accountcode and userfield. There is more info here.
>
> http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr
>
> I'm not sure about defining additional columns and writing to them
> through the dialplan but I don't think
exten=s,4,Playback(/record/deneme.gsm)
should be
exten=s,4,Playback(/record/deneme)
so without a format.
On Tue, 2009-06-23 at 11:31 +0300, Oguzhan Kayhan wrote:
> Hello,
> I am trying to create a simple IVR for testing..
> What i did is to create a voice file from asterisk-gui.
> And i saw
Thanks for your reply. I saw that info also on voip-info.org.
I was wondering if I could define other columns, like those used for
billing (as defined in my sip.conf).
Jonas.
On Tue, 2009-06-23 at 09:22 +0100, Ishfaq Malik wrote:
> Hi
>
> The calldate column is the date and time of the call,
clidchannel
dstchannel lastapp lastdatastart answer end duration
billsec disposition amaflags
Why does it want to write to a column calldate ?? Where is this
defined ??
Thanks for the help !
Jonas.
On Fri, 2009-06-19 at 14:13 -0500, Miguel Molina wro
On Thu, 2009-06-18 at 11:52 -0500, Tilghman Lesher wrote:
>
> In modules.conf: noload => cdr_csv.so
>
Are there other modules I need to load or unload ??
asterisk*CLI> module show like cdr
Module Description
Use Count
cdr_addon_mysql.so MySQL CDR Backend
There are some things that are not that clear to me :
When I want to write CDR-info to an external MySQL-DB
- do I need to install the asterisk-addons prior to installing Asterisk
or after having installed Asterisk ??
- How do I tell Asterisk not to write CDR-info to the Master.csv file
but into
On Fri, 2009-06-12 at 23:58 +1000, Alex Samad wrote:
>
> what is the best way forward to recompile with hardware echo canceller
> support.
>
No need to do anything special during compilation. For hardware echo
cancellation just put the option "echocancel=yes" in chan_dahdi.conf
__
How about this :
if you add the option 'g' in your Dial()-command, then when the caller
hangs up Asterisk will continue to execute the commands hat follow.
You could then read the ${DIALSTATUS}-variable (which will be 'CANCEL')
and execute a command based on this.
GoToIf($["${DIALSTATUS}"="ANSWER
On Sat, 2009-05-30 at 23:15 -0300, Daniel Bareiro wrote:
>
> I was testing calling from my cell phone to an analog telephone and if the
> other person hangs before I do it, I see that in the my cell phone the call
> even continues persisting so that if the person of the other endpoint take the
>
On my TDM410P pci-card I have an hardware echo cancellation module
(Digium VPMADT032 EC Modul).
I have set 'echocancel=yes' in my chan_dahdi.conf to activate this
hardware module.
Do I now have 2 echo cancellers that are activated ? A software echo
canceller and a hardware echo canceller ??
Form
I have posted a similar problem earlier on this mailing list with my
Asterisk-system + TDM410 + Grandstream telephones.
But there has not yet been a response to this.
My client is also experiencing a 'simplex' conversation. There seems
that audio can only flow 1 one way at the same time.
What I h
Hey list !
I'm getting the feedback of a customer that a conversation is like half
duplex : when he talks, the other end of the call is no longer heard.
What could be the cause of these drop-outs ?
A call that is coming in from the PSTN is routed through an IVR-system
to the correct internal SIP
I thought that /var/log/maillog was for sendmail ?? I'm not using
sendmail...
My /var/log/maillog is empty :
[r...@asterisk ~]# cat /var/log/maillog
[r...@asterisk ~]#
How about the system()-application ?? Why is that also not working for
me ??
On Fri, 2009-05-22 at 16:25 +0100, Geraint Lee
David,
what is your SMTP-client then ?
Did you change the mailcommand 'mailcmd' in voicemail.conf ?? Or is it
still /usr/sbin/sendmail ??
I use version 1.4.24.
Thanks for your reply.
Greetingz,
Jonas.
On Fri, 2009-05-22 at 10:59 -0400, David wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Has
:33:09 jonas kellens wrote:
> > My /root/.msmtprc-file has the following :
> > # Set default values for all following accounts.
> > defaults
> > logfile ~/.msmtp.log
>
> > There is NO entry in the logfile of msmtp (/root/.msmtp.log). No error,
> > no success.
digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
european=Europe/Copenhagen|'vm-received' a d b 'digit
Hey there list !
I'm receiving negative feedback when people try to pickup another
ringing phone by pressing *8 on there own Grandstream device.
These are my setting that should make pickup possible :
all my sip-clients (Grandstream) have this in their config (sip.conf) :
callgroup=1
pickupgrou
Gordon,
have you not defined a context [BLF_group] in your extensions.conf ??
And a subscribecontext in sip.conf ?
The Grandstream documentation does mention this.
Have you configured the speed dial buttons (to the right of your grandstream)
or the phone line buttons (to the left of the display
To feed your curiosity... I'm about to implement it.
I have several GXP2020 and GXP1200 Grandstream telephones. I'm reading
documentation to know how to start and what to expect.
I'm hoping that implementing BLF on these Grandstreams in combination with
Asterisk is easier then configuring sla.co
Check out the Grandstream GXP-serie also...
http://www.grandstream.com/gxp2020.html
You can program the line buttons to support BLF (red, red blinking, green)
>- Oorspronkelijk bericht -
>Van
: Olivier [mailto:oza-4...@myamail.com]
>Verzonden
: dinsdag
, mei
19, 2009 08:21 AM
>Aan
: 'A
Today I get the remark that a call got disconnected after 10 minutes.
This what my VERBOSE-logfile tells me :
[May 18 15:36:30] VERBOSE[3940] logger.c: -- Executing
[00493516...@intern:1] NoOp("SIP/51-b76023b8", "Gesprek naar GSM-nummer
via Telenet") in new stack
[May 18 15:36:30] VERBOSE[3940
0600, Brandon B. wrote:
> mutt will not deliver a email message, so you are using the wrong
> command. The email message with attachment is created by Asterisk and
> needs msmtp to deliver the message.
>
> On Sun, May 10, 2009 at 9:10 AM, jonas kellens
> wrote:
> >
I call the firm from my portable at home (zoiper softphone). I have
internal extension 60, and I call the internal SIP-client 10 at the firm
via an IAX-connection over internet.
My colleague at phone 10 answers my call. I ask him to transfer me with
my colleague at extension 50. He then presses "t
I have changed the features.conf file, yes.
And I put this in my extensions.conf :
include => parkedcalls
Is it better to put "exten => 90,1,park()" into my dialplan ?
Greetingz,
Jonas.
On Thu, 2009-05-14 at 16:08 -0500, Danny Nicholas wrote:
> Did you change 700 to 90 in features.conf? I’d
I have the same problem with Asterisk 1.4.24 and a Grandstream GXP2020
SIP-phone.
I want to park a call by pressing the 'TRANSFER' and then 90. My parking
lots are from 91 till 95.
The call is parked at extension 91, but the parking lot '91' is not
announced by Asterisk...
I have tried to park the
When I call my Asterisk-server from my cell phone on one of the
PSTN-numbers that terminate in a FXO-module on my TDM410P Digium card,
and in the dialplan the end of a context is reached and Asterisk needs
to execute the Hangup()-command, I notice the following :
- Asterisk tells me that the conve
Dave,
can you help me with my configuration of mutt (MUA) + msmtp (MTA) ?
I have included the following in my voicemail.conf :
mailcmd=/usr/sbin/mutt
But how will Asterisk know how to use Mutt to attach its
voicemail-message (.wav-file) ???
I use Mutt together with msmtp to send me weekly the
MailboxNumber => password,name,e-mail,pager,options
50 => 4569,Jonas Kellens,jonas.kell...@thecomputerstore.be,,tz=belgie|
attach=yes
But I do not receive an e-mail after having left a voicemail message on
the voicemailbox 50.
What mail-server does Asterisk uses to send his mail ???
Sendma
Gavin,
My Asterisk-server has 2 interfaces :
- eth0 = LAN-interface (for SIP-phones to register)
- eth1 = WAN-interface (for IAX-trunking to IAX-provider)
Asterisk is behind NAT (has internal IP-address 192.168.3.248 for WAN_if)
SETUP :
m0n0wall 192.168.3.250 -- 192.168.3.248 (WAN)-Asterisk-(L
Thanks for the feedback !
I know the IP-address of my Asterisk-server.
The WAN-interface of my Asterisk-box is set manually (ifcfg-eth1).
I have port 4569 forwarded on my NAT/firewall.
Strangely I have the same 'notice' when being attached directly to the internet
(so no firewall in between).
An
I have connected my Asterisk-box directly to my internetconnection. I
have disabled my firewall.
Still I am unable to register with my IAX-provider. Can someone please
point me out why I am unable to register my Asterisk to another
Asterisk-box ?
A RegReq is send to the other Asterisk-box but no r
According to my IAX-provider, an account has been created for me on
their Asterisk-server...
But the Asterisk CLI tells me this :
asterisk*CLI> iax2 reload
== Parsing '/etc/asterisk/iax.conf': Found
[Apr 30 20:51:30] NOTICE[6391]: chan_iax2.c:10124 set_config: Ignoring
bindport on reload
[Apr 3
I have Asterisk 1.4.24 en a Digium TDM410 with EC and with 3 FXO
modules.
When I plug one PSTN-line into a FXO-port I am able to receive calls on
this line and I can also make calls from an internal SIP-phone to the
external PSTN-network.
Still I am bothered about something that appears on the CL
text=default
port=5060
bindaddr=192.168.4.248
srvlookup=yes
disallow=all
allow=alaw
allow=gsm
allow=ulaw
language=be
[BT201]
type=friend
context=intern
host=dynamic
username=BT201
secret=testpaswoord
canreinvite=no
callerid=Jonas Kellens <52>
qualify=yes
[GXP1200]
type=friend
context=inter
part of extensions.conf:
exten => 11,1,Answer()
exten => 11,n,NoOp(CallerID : ${CALLERID(all)})
exten => 11,n,Playback(/tmp/welkom-tcs.alaw)
exten => 11,n,GoToIfTime(09:00-17:59|mon-fri|*|*?open,s,1)
; wordt doorgerouteerd naar context open, maar indien gesloten :
exten => 11,n,NoOp(Oproep tijdens
t (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
3 channels to configure.
[r...@asterisk asterisk]# /usr/sbin/dahdi_hardware
pci::04:05.0 wctdm24xxp+ d161:8005 Wildcard TDM410P
[r...@
Even if Zaptel is compiled, you can also compile Dahdi because Asterisk
will choose the DAHDI-module... it seems.
So I left Zaptel... and compiled Dahdi (everything went well, I followed
the steps) en then Asterisk again (with dahdi support!!).
Yet another episode in this nightmare :
[r...@aster
you ???
Can I edit one of these files to make chan_dahdi.conf interact with
zaptel.conf (zaptel kernel module) in stead of the dahdi-linux kernel
modules ??
Greetingz,
Jonas.
On Mon, 2009-04-20 at 15:57 +0300, Tzafrir Cohen wrote:
> On Sun, Apr 19, 2009 at 05:17:38PM +0200, jonas kellens w
Jonas.
Forwarded Message
> From: jonas kellens
> To: asterisk-users@lists.digium.com
> Subject: Zaptel to Dahdi
> Date: Sun, 19 Apr 2009 17:17:39 +0200
>
> VoIP-wiki.org states :
>
> /etc/zaptel.conf Becomes /etc/dahdi/system.conf
> /etc/asterisk/
VoIP-wiki.org states :
Digium resources http://www.asterisk.org/zaptel-to-dahdi
/etc/zaptel.conf Becomes /etc/dahdi/system.conf
/etc/asterisk/zapata.conf Becomes /etc/asterisk/chan_dahdi.conf
Now, what do I have installed on my system :
/etc/zaptel.conf and /etc/asterisk/chan_dahdi.conf
Will
How do I know that de hardware echo canceller module on my Digium
TDM403E is recognized by Asterisk ?
After having configured /etc/zaptel.conf :
[r...@asterisk etc]# /sbin/ztcfg -vv
Zaptel Version: 1.4.12.1
Echo Canceller: MG2
Configuration
==
Channel map:
Channel 01: FX
How come the mask is 255.255.255.255 ??
asterisk*CLI> iax2 show peers
Name/UsernameHost Mask Port
Status
jonaskellens/jo 192.168.4.169 (D) 255.255.255.255 4569
Unmonitored
1 iax2 peers [0 online, 0 offline, 1 unmonitored]
Greetingz,
Jonas.
O
[r...@asterisk asterisk]# cat iax.conf
[general]
autokill=yes
bindport=4569
bindaddr=0.0.0.0
[jonaskellens]
type=friend
host=dynamic
;auth=md5
username=jonaskellens
password=zoiper
callerid="Jonas Kellens" <100>
context=intern
disallow=all
allow=gsm
allow=speex
allow=alaw
as
14:38:01.229941 IP 192.168.4.240.sip > 192.168.4.248.sip: SIP, length:
889
14:38:01.230127 IP 192.168.4.248.sip > 192.168.4.240.sip: SIP, length:
515
14:38:01.251558 IP 192.168.4.240.sip > 192.168.4.248.sip: SIP, length:
497
14:38:01.271714 IP 192.168.4.240.sip > 192.168.4.248.sip: SIP, length:
106
86 register_verify: No
registration for peer 'jonaskellens' (from 192.168.4.169)
...
My iax.conf-file :
[r...@asterisk asterisk]# cat iax.conf
[general]
autokill=yes
bindport=4569
bindaddr=0.0.0.0
[Jonas]
type=friend
host=dynamic
;auth=md5
username=jonaskellens
password=zoiper
callerid=&
I have 2 SIP-clients defined in my sip.conf :
[GXP1200]
type=friend
context=intern
host=dynamic
username=GXP1200
secret=testpaswoord
canreinvite=yes
[BT201]
type=friend
context=intern
host=dynamic
username=BT201
secret=testpaswoord
canreinvite=yes
When I make a call from one to another this is d
For an Asterisk-environment with no more then 10 SIP-phones, I would
open 10 x 4 = 40 UDP ports for RTP/RTCP-traffic ( 4/call). Can you
confirm ?!
rtp.conf :
rtpstart=30500
rtpend=30550
Ok, there's 50 here... a round number right ?!
All SIP-communication stays on the LAN. There's a NIC connected
There is something wrong with my IPtables !!!
When i do :
service iptables stop
I see my phones register on the CLI !!
I can place a call and the phone rings !! I see a whole lot of
SIP-requests on the CLI with SDP-message in body !! That's good news...
What is wrong with my IPtables-rule I've
I will summarize everything again and try to answer all the questions
asked while I was away.
First I stop Asterisk :
[r...@asterisk asterisk]# /usr/sbin/asterisk -r
Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WAR
On Mon, 2009-04-13 at 13:21 -0700, Steve Edwards wrote:
> On Mon, 13 Apr 2009, jonas kellens wrote:
>
> > 1) IP-phones get there IP from a DHCP
>
> The source of the address is not the issue.
>
> > I still see no register-message on the CLI. This really should happen
&
>
> On Mon, Apr 13, 2009 at 06:18:58PM +0200, jonas kellens wrote:
>
> > I pick up the phone of the BT201 and dial 211... nothing happens.
> > I pick up the phone of the GXP1200 and dial 210... nothing happens.
> >
> > I would love to have your feedback on this.
Hey there again !
I've changed some things now :
1) IP-phones get there IP from a DHCP
2) sip-accounts simplified :
[r...@asterisk asterisk]# cat sip.conf
[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
[210]
type=friend
context=intern
host=dynamic
[
ere.
I followed the book "Asterisk, the future of telephony"...
Thanks for your reply !
Greetingz,
Jonas.
On Mon, 2009-04-13 at 14:04 -0400, Barry L. Kline wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> jonas kellens wrote:
> > I pick up the phone
These are the settings on my BT201 (GXP1200 is the same interface) :
Account Name:(e.g., MyCompany)
SIP Server:(e.g., sip.mycompany.com, or IP address)
Outbound Proxy:(e.g., proxy.myprovider.com, or IP address)
SIP User ID:(the user part of an SIP address)
--> I put here the s
I pick up the phone, and dial 211 on the BT201. This is the Asterisk
CLI :
Connected to Asterisk 1.4.24 currently running on asterisk (pid = 3895)
Verbosity is at least 5
asterisk*CLI>
Nothing is displayed... it stays that way...
Jonas.
> On Mon, 2009-04-13 at 11:59 -0500, James A. Shigley w
Danny,
this is from the Asterisk CLI :
asterisk*CLI> dialplan reload
Dialplan reloaded.
== Parsing '/etc/asterisk/extensions.conf': Found
-- Registered extension context 'default'
-- Including context 'intern' in context 'default'
-- Registered extension context 'intern'
-- Adde
James,
when I run Asterisk -vr and I enter 210 on one phone to call the
other, nothing is displayed on the CommandLine...
I know this is not right, just don't know what is wrong. I really need
someone to guide me a bit...
[r...@asterisk asterisk]# /usr/sbin/asterisk -vr
Asterisk 1.4.24,
Tony Plack,
this is the result form Asterisk CLI :
[r...@asterisk asterisk]# /usr/sbin/asterisk -vr
Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.
This is free soft
Mon, 2009-04-13 at 12:28 -0400, Michael van der Stoop wrote:
>
> jonas kellens wrote:
> > Hi there,
> >
> > this is the first time that I'm building an Asterisk-server.
> >
> > I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
> >
Hi there,
this is the first time that I'm building an Asterisk-server.
I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
Zaptel is for later, when configuring the POTS-line. Now first internal
communication with SIP.
Thought it would go easier...
I have 2 Grandstream IP-pho
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