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
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
-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
european=Europe/Copenhagen|'vm-received' a d b 'digits/at' HM
belgie=Europe/Brussels|'vm-received' Q 'digits/at' R
[Voicemail-context]
60 = 4569,Jonas Kellens,jonas.kell...@telenet.be
In my extensions.conf I has the following :
exten = 2000,1
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.
Is Asterisk running as root or as the asterisk user
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-
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
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
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
:
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
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
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]
. 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 jonas.kell...@telenet.be
wrote:
Dave,
can you help me with my
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 put
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
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
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
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
-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 ???
Sendmail is not active on my CentOS-box. I have
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).
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
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
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
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
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=intern
host=dynamic
username=GXP1200
secret=testpaswoord
canreinvite
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
: 03)
3 channels to configure.
[r...@asterisk asterisk]# /usr/sbin/dahdi_hardware
pci::04:05.0 wctdm24xxp+ d161:8005 Wildcard TDM410P
[r...@asterisk asterisk]# /usr/sbin/dahdi_tool
shows me an 'OK' under Alarm for my Wildcard TDM410
Jonas Kellens
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 :
.
Forwarded Message
From: jonas kellens jonas.kell...@telenet.be
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/zapata.conf Becomes /etc
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 wrote:
VoIP-wiki.org
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:
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
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
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=Jonas Kellens 100
context=intern
disallow=all
allow=gsm
allow=speex
allow=alaw
On the CLI :
Verbosity is at least 20
asterisk*CLI
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:
1060
[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
asterisk*CLI iax2 reload
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.
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
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 marks...@digium.com
Asterisk comes
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
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
.
On 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.
Zaptel is for later, when configuring the POTS-line. Now
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 marks...@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for
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,
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'
--
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
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
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, and dial 211 on the BT201. This is the Asterisk CLI :
/Connected
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
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. Where could this problem be
situated ?
Your basic mistake
, 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
now, as they are defined host=dynamic !
I suspect you have not [correctly] configured the phones to register
801 - 850 of 850 matches
Mail list logo
