The only solution I see to have a PICKUPMARK-variable created on an
incoming channel, and have the same PICKUPMARK on another created
channel (the one that does the pickup) is to work with a database like
MySQL.
I see no other way to separate multiple incoming channels (with their
own
Hello
Out of curiosity, are those weaknesses still there in Asterisk 1.6, or
have they been fixed?
How does FreeSWITCH compare to Asterisk?
http://www.freeswitch.org/node/117
Thank you.
--
_
-- Bandwidth and Colocation
Hi guys,
Thanx a lot to all of you.
My call is now forwarded to sip form PSTN, but again a new problem is
coming.
When i pick up the call from my softphone it says the can not access speaker
or microphone. But i have my headphone plugged in and in working stage.
on softphone:
Fri 18:08:17
hi,all
for a long time, i cant understand the difference between
${CDR(calldate)} and ${CDR(start)} , ${CDR(answer)}
i know ${CDR(start)} mean when a call is start. and ${CDR(answer)}
means when a call was pick up.
but what's ${CDR(calldate)} mean?
Could you help me ?
Thansk a lot!
--
Hi,
We have free pbx installed on asterisk 1.4.25.1. Mysql is installed and
asterisk is connecting to it. CDR modules are all loaded as well.
For some reason, it is not creating master.csv and no cdrs are generated.
Can anyone help please.
---
Kind Regards,
Deepika
Hello dear list.
I am currently working on a Automatic attendant, and the core things work, but
I think the loop function isn't working as expected.
I am testing this environment: a sip internal call from 301 to 501.
The setup here is when 301 calls 501, and 301 doesn't enter an extension, it
On Fri, Jun 18, 2010 at 10:51:40AM +0200, Aksel Celasun wrote:
Hello dear list.
I am currently working on a Automatic attendant, and the core things work,
but I think the loop function isn't working as expected.
I am testing this environment: a sip internal call from 301 to 501.
The
Extensions.conf
[mainmenu]
exten = 501,1,Answer
exten = 501,n,Wait(2)
exten = 501,n,Playback(velkommen_abacus)
exten = 501,n,Set(Loop=0)
exten = 501,n,While($[${Loop} 3])
exten = 501,n,Background(tast123vent_)
exten = 501,n,WaitExten(5)
exten = 501,n,Set(Loop=$[${Loop}+1])
exten =
Cdr status shows:
CDR logging: enabled
CDR mode: simple
CDR output unanswered calls: no
It is not showing 'CDR registered backend'
Thanks,
Deepika
From: Deepika Nijhawan [mailto:deepika.nijha...@oxygen8.com]
Sent: 18 June 2010 09:37
To: 'asterisk-users@lists.digium.com'
Aksel Celasun wrote:
This should be:
exten = 501,n(LoopEnd),EndWhile
I don't understand, i do have the same thing you wrote above.
The difference between yours and his is that you had a n,(LoopEnd) and
it should be n(LoopEnd)
Doug
--
Ben Franklin quote:
Those who would give up
On 18 June 2010 10:38, Deepika Nijhawan deepika.nijha...@oxygen8.comwrote:
Cdr status shows:
CDR logging: enabled
CDR mode: simple
CDR output unanswered calls: no
It is not showing ‘CDR registered backend’
Thanks,
Deepika
Have you compiled asterisk-addons and selected to
Ah, I missed the comma, thank you, and thank you Tzafrir Cohen!
Best regards
Aksel
-Opprinnelig melding-
Fra: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Doug Lytle
Sendt: 18. juni 2010 11:41
Til: Asterisk Users Mailing List -
Hello,
I have a problem in Asterisk 1.4 each day I need to restart *asterisk
service asterisk* restart in order to unblock the calls
My question how can I do in order to check the issue, and if there is any
tool or log?
Thanks and regards.
--
This week our guest at 12 noon EDT (http://vuc.me/next for your local
time) is Acme Packet, maker of Session Border Controllers. We look
forward to learning more about these.
Join in G722 wideband by calling sip:200...@login.zipdx.com starting
just before 12 Noon EDT or Skype:vuc.me or see
Hi, all
Would any of you be able to suggest physical SIP phones that support inbuilt
VPN capabilities; akin to the Snom 370/870 ?
--
Thanks, Phil
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
I have a Cisco SPA525G that seems to support it, but I've never needed that.
I would assume most Cisco SPA phones would support that too.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of --[ UxBoD ]--
Hello again dear list.
Could you please help with this?
Thank you for all support, you are great, and i am now at a late stage in the
setup and tweaking this server,
So I hope you can help me again.
I Can't make include the context nighttime. Just to demonstrate if it works, I
have a
I do a cron to execute /usr/sbin/asterisk -rx restart when convenient
each day at 4:45 AM. This doesn't really solve any problems, just does
housekeeping to keep a clean environment, since some installs/os'es lend
themselves to memory leaks.
_
From:
Minor edit on the include = nighttime|12:30-8:00|mon-fri|*|*
Correct now.
Fra: Aksel Celasun
Sendt: 18. juni 2010 14:30
Til: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Emne: Error trying to add context: Context 'internal' tries to include
nonexistent context
Hi, I tell you I've made some calls from a land-phone to my IVR in
order to avoid the possible poor quality of cell phone's DTMF, and
when I called extension 1003 I was connected to extension 1000
againthe same error.
My IVR says dial 1 to connect to operator or dial the extension in
case you
I would definitely change the prompt from 1 to 0. It is not an advisable
practice to have an IVR selection that can be misinterpreted like this.
Assuming that all of your extensions are in 1000-1999, 2 for the operator
would be just as good; the important thing is that you don't have a single
I am using asterisk 1.4.32 and wish to connect using SIP to a nortel
1000 switch
with the ability to have 90 calls at a one time outgoing or incoming.
the nortel reseller is asking me what to do. I dont know nortel at all.
I thought I just needed a SIP trunk and IP address of the their server
Any ideas, please?
Anahi Ludueña
From: a_ludu...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Thu, 17 Jun 2010 19:54:30 +
Subject: Re: [asterisk-users] Music on Hold problema
I have wav files in the /var/lib/asterisk/mohmp3...
Anahi Ludueña
From:
Post the /var/lib/asterisk/mohmp3 listing and musiconhold.conf
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña
Sent: Friday, June 18, 2010 9:18 AM
To: asterisk-users@lists.digium.com
Subject: Re:
On Fri, Jun 18, 2010 at 7:44 AM, Aksel Celasun ak...@abacus-it.no wrote:
Minor edit on the include = nighttime|12:30-8:00|mon-fri|*|*
Correct now.
This isn't how you do time based checks in asterisk. Lookup the application
GotoIfTime.
--
Thanks,
--Warren Selby
http://www.selbytech.com
--
Hi,
As Danny said, asterisk is looking for slin or ulaw files. Are your wav
files in any of these formats? Did you just copied them from somewhere
without changing their format? Also note they should be 8KHz mono 16 bit
files. You can do this in a simple utility like Windows Recorder.
Zeeshan A
what do you mean unblock the calls exactly?
-- Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308
Date: Fri, 18 Jun 2010 11:12:55 +0100
From: salah.elharit...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk issue
Hello,
Just a note to anyone in the Kansas City area that I've relaunched the
KCAUG website/group at http://kcaug.org. Please drop by and join the
group. :)
--
Kyle Sexton
--
_
-- Bandwidth and Colocation Provided by
The list of /var/lib/asterisk/mohmp3 is:
-rw-rw 4 asterisk asterisk 184 Oct 19 2009
LICENSE-asterisk-moh-freeplay-wav
-rw-rw-r-- 4 asterisk asterisk 882748 Oct 19 2009 QuajiroPromo.sln
-rw-rw-r-- 4 asterisk asterisk 834682 Oct 19 2009 TristeAlegriaPromo.sln
-rw-rw 4 asterisk
do you have any tool in order to check what happened in asterisk during the
hangs of calls
2010/6/18 salaheddine elharit salah.elharit...@gmail.com
Hello,
I have a problem in Asterisk 1.4 each day I need to restart *asterisk
service asterisk* restart in order to unblock the calls
My
On Friday 18 June 2010 03:21:32 Zhang Shukun wrote:
hi,all
for a long time, i cant understand the difference between
${CDR(calldate)} and ${CDR(start)} , ${CDR(answer)}
i know ${CDR(start)} mean when a call is start. and ${CDR(answer)}
means when a call was pick up.
but what's
On Friday 18 June 2010 09:49:39 Warren Selby wrote:
On Fri, Jun 18, 2010 at 7:44 AM, Aksel Celasun ak...@abacus-it.no wrote:
Minor edit on the include = nighttime|12:30-8:00|mon-fri|*|*
Correct now.
This isn't how you do time based checks in asterisk. Lookup the
application GotoIfTime.
We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180
- has a PRI connection to a T-1. Another server is the router to the
internet. All phones in the office and the workstations are on the network.
Most of the internal phones are aastra 9133i's. Here the network config
from
Just a guess - the phones are down because they can't get to the DCHP
server. If you can't ping 10.10.10.44 you'll never reach 10.10.10.180.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent:
- sean darcy seandar...@gmail.com wrote:
We have a 10.10.0.0 internal network. The asterisk server -
10.10.10.180
- has a PRI connection to a T-1. Another server is the router to the
internet. All phones in the office and the workstations are on the
network.
Most of the internal
On Fri, 18 Jun 2010, sean darcy wrote:
We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180
- has a PRI connection to a T-1. Another server is the router to the
internet. All phones in the office and the workstations are on the network.
Most of the internal phones are
On 06/18/2010 12:53 PM, Danny Nicholas wrote:
Just a guess - the phones are down because they can't get to the DCHP
server. If you can't ping 10.10.10.44 you'll never reach 10.10.10.180.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On 06/18/2010 12:57 PM, Tim Nelson wrote:
- sean darcyseandar...@gmail.com wrote:
We have a 10.10.0.0 internal network. The asterisk server -
10.10.10.180
- has a PRI connection to a T-1. Another server is the router to the
internet. All phones in the office and the workstations are on
On 06/18/2010 01:19 PM, Gordon Henderson wrote:
On Fri, 18 Jun 2010, sean darcy wrote:
We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180
- has a PRI connection to a T-1. Another server is the router to the
internet. All phones in the office and the workstations are on
Based on my somewhat similar experience a few times, when this happens, try
to resolve an IP, to make sure there is a valid DNS server accessible to
Asterisk. If not, either make asterisk a DNS as well, or remove any domain
name entries from /etc/resolv file and replace them with the IP addresses
thanks for your response
how can i create and execute this cron
2010/6/18 Danny Nicholas da...@debsinc.com
I do a cron to execute “/usr/sbin/asterisk –rx “restart when convenient”
“ each day at 4:45 AM. This doesn’t really “solve” any problems, just does
“housekeeping” to keep a clean
On 06/18/2010 01:42 PM, Zeeshan Zakaria wrote:
Based on my somewhat similar experience a few times, when this happens,
try to resolve an IP, to make sure there is a valid DNS server
accessible to Asterisk. If not, either make asterisk a DNS as well, or
remove any domain name entries from
Crontab -e will open your crontab for editing (if you are root)
Add this line
45 4 * * * /usr/sbin/asterisk -rx restart when convenient
And exit the editor
This will restart your asterisk at 4:45 am every day unless a call is active
at that time. If a call is active, asterisk will restart
Nice and colorful tutorial for cronjobs.
http://www.linuxconfig.org/Linux_Cron_Guide
-Bruce
On Fri, Jun 18, 2010 at 1:55 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
thanks for your response
how can i create and execute this cron
2010/6/18 Danny Nicholas da...@debsinc.com
Un-top-posting...
On Fri, 18 Jun 2010, salaheddine elharit wrote:
I have a problem in Asterisk 1.4 each day I need to restart asterisk
service asterisk restart in order to unblock the calls
2010/6/18 Danny Nicholas da...@debsinc.com
I do a cron to execute “/usr/sbin/asterisk –rx “restart
On Fri, 18 Jun 2010, bruce bruce wrote:
Nice and colorful tutorial for cronjobs.
http://www.linuxconfig.org/Linux_Cron_Guide
Colorful, but missing valuable content like: setting environment
variables, especially MAILTO and PATH; and time specification nicknames
like @daily.
man 5 crontab
Did you check /etc/resolv? Does it point to any DNS by domain name?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-18 2:04 PM, sean darcy seandar...@gmail.com wrote:
On 06/18/2010 01:42 PM, Zeeshan Zakaria wrote:
Based on my somewhat similar experience a few times...
www.ilovetovoip.com
thank you so much for your help and support :)
2010/6/18 Danny Nicholas da...@debsinc.com
Crontab –e will open your crontab for editing (if you are root)
Add this line
45 4 * * * /usr/sbin/asterisk –rx “restart when convenient”
And exit the editor
This will restart your asterisk at
Abandoning all hope of un-top-posting...
On Fri, 18 Jun 2010, sean darcy wrote:
(Sean has a problem and several posters suspect it is DNS related.)
On Fri, 18 Jun 2010, Zeeshan Zakaria wrote:
Did you check /etc/resolv? Does it point to any DNS by domain name?
If you mean /etc/resolv.conf and
All:
I am using the standard voicemail in asterisk. Everything works well,
except, if a users wants to record their own personal greeting, it
doesn't playback.
I can see the soundfile being created. I suspect it is a setting in the
voicemail.conf, or an option I am over-looking on the
Hi again
Thank you Warren, GotoIfTime was the deal!
And easy to use!
Gr8.
Best regards.
Aksel
Fra: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Warren Selby
Sendt: 18. juni 2010 16:50
Til: Asterisk Users Mailing List - Non-Commercial
Thank you for the info.
As I wrote to Warren GotoIfTime was easy to use and seemed more flexible,
Got it working now! Perfect!
Only one thing left now, and my system is pretty much ready for live testing,
Surely easy for the user list, so it will come in another mail soon, after I
have done
On 06/18/2010 03:09 PM, Steve Edwards wrote:
Abandoning all hope of un-top-posting...
On Fri, 18 Jun 2010, sean darcy wrote:
(Sean has a problem and several posters suspect it is DNS related.)
On Fri, 18 Jun 2010, Zeeshan Zakaria wrote:
Did you check /etc/resolv? Does it point to any DNS
The Asterisk Development Team has announced the release of Asterisk 1.4.33.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.33 resolves several issues reported by the
community, and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 1.6.2.9.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.9 resolves several issues reported by the
community, and would have not been
On Fri, 2010-06-18 at 13:59 -0400, sean darcy wrote:
If the internet server is down, there can't be a valid DNS server
accessible to Asterisk. The asterisk server is a caching name server,
but obviously won't be able to resolve addresses not in its cache.
Asterisk clearly doesn't need
Hello everyone.
Successfully patched to the new version, but when trying to compile, I
get this :
/usr/src/asterisk/asterisk/include/asterisk/options.h:102:56: error:
operator '' has no right operand
Dahdi is fresh from the SVN trunk. Am I missing something ?
Thanks !
Hoggins!
On 06/18/2010 05:08 PM, Hans Witvliet wrote:
On Fri, 2010-06-18 at 13:59 -0400, sean darcy wrote:
If the internet server is down, there can't be a valid DNS server
accessible to Asterisk. The asterisk server is a caching name server,
but obviously won't be able to resolve addresses not in
On Fri, 18 Jun 2010, sean darcy wrote:
I'm running named as a caching nameserver. /etc/resolv.conf point to
localhost. But, obviously, it only responds from the cache, since the
root servers are unavailable.
If the root servers are not available, what is available to cache?
tcpdump is an
But why can't my phones call. The outgoing lines are PRI/DAHDI T1. No
sip. No iax. Why does the asterisk machine have to resolve any address?
The internal phones can't even call each other, even though they have
hard ip addresses.
Same for doing DHCP for handing out addresses to your
On 06/18/2010 06:19 PM, Cary Fitch wrote:
But why can't my phones call. The outgoing lines are PRI/DAHDI T1. No
sip. No iax. Why does the asterisk machine have to resolve any address?
The internal phones can't even call each other, even though they have
hard ip addresses.
Same for doing
Hi!
But why can't my phones call. The outgoing lines are PRI/DAHDI T1. No sip.
No iax. Why does the asterisk machine have to resolve any address?
Probably because you have one or more register = statements in your
sip.conf and Asterisk is trying badly - but without success - to register
On 6/18/2010 7:26 PM, sean darcy wrote:
On 06/18/2010 06:19 PM, Cary Fitch wrote:
But why can't my phones call. The outgoing lines are PRI/DAHDI T1. No
sip. No iax. Why does the asterisk machine have to resolve any address?
The internal phones can't even call each other, even though
CENTOS 5.5
dahdi 1.4.3.0.1
uname -r
2.6.18-194.3.1.el5PAE
[r...@localhost dahdi-linux-2.3.0.1]# service dahdi start
Loading DAHDI hardware modules:
FATAL: Module dahdi not found.
wct4xxp: FATAL: Module wct4xxp not found.
[FAILED]
It appears as though the 489 Bad Event response to the NAT keep alive
event responds to the local address, instead of responding to the
NATted address.
This causes Linksys phones to go amber (no registration) after a short
amount of time after placing calls.
Turning the Linksys NAT keep alive off
Thank you.
another quesion is i want to get ${CDR(answer)} and ${CDR(end)} in the
hangup section. i can get ${CDR(answer)} sucessfully but get
${CDR(end)} is null.
i know i can set any variable i want into CDR table if i want .
but i want to know without any setting . which variabes will set
Dear Asterisk friends,
Please help me to clarify my doubt. After monitor SIP and RTP traffic with
Wireshark. I found that both SIP and RTP traffic between 2 sip clients must be
passed through Asterisk.
Is it possible that 2 sip clients connect with each other directly for RTP
session
On 06/19/10 15:19, Kamonwat Sookkara wrote:
Dear Asterisk friends,
Please help me to clarify my doubt. After monitor SIP and RTP
traffic with Wireshark. I found that both SIP and RTP traffic between
2 sip clients must be passed through Asterisk.
Is it possible that 2 sip clients
James Lamanna schrieb:
It appears as though the 489 Bad Event response to the NAT keep alive
event responds to the local address, instead of responding to the
NATted address.
This causes Linksys phones to go amber (no registration) after a short
amount of time after placing calls.
Turning
69 matches
Mail list logo
