unknown unknown
GNU/Linux
Thanks inadvance,
---
Paul Belanger (mailto:[EMAIL PROTECTED])
Technical Support Specialist
Cisco Certified Network Associate
Pronexus Inc. - A Powerful Voice in Communication Solutions
---
Tel: 613.271.8989 ext. 516
Derrick,
Thanks for the ideas. I have since removed any USB/Firewire/un-needed
hardware from loading in the MOBO BIOS and recompiled the kernel to boot.
However I still seem to have the same problem.
Here is some more information.
# lsmod
Module Size Used byNot tainted
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello all,
I was looking for some information about using Asterisk to convert an
incoming H.323 call to and outgoing SIP call. Is this possible?
PB
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.5 (MingW32)
Comment: Using GnuPG with Thunderbird -
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
I have been successful in getting Digest authentication to work with my
Mitel 5055 IP Phones, however I'm wondering if Asterisk still supports
Simple authentication? I know it has been depreciated in the RFC, but I
have some phones with don't
: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Belanger
|Sent: Wednesday, January 12, 2005 1:06 PM
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Using asterisk to convert H.323 to SIP?
|
| Hello all,
|
| I was looking for some information about using Asterisk to convert
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Enable debugging to see the reason:
CLI sip debug
quote
A UAS rejecting an offer contained in an INVITE SHOULD return a 488 (Not
Acceptable Here) response. Such a response SHOULD include a Warning
header field value explaining why the offer was
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Ya, it has been a little slow for me today too.
PB
Matthew Boehm wrote:
| Hey guys, I sent an email to the list at 2:57PM central. I just now see it
| on the list, and its 3:23PM.
|
| Anyone else experience this? I am sending this email at 3:24PM
---
Paul Belanger (mailto:[EMAIL PROTECTED])
Technical Support Specialist
Cisco Certified Network Associate
Pronexus Inc. - A Powerful Voice in Communication Solutions
---
Tel: 613.271.8989 ext. 516
Fax: 613.271.8388
http://support.pronexus.com
G'day all,
I'm trying to come up with a quick, easy solution to have a static
inbound number in my dialplan, rotate calling 2 numbers. Example:
1st call into asterisk
exten = 1234,1,Dial(sip/,10)
exten = 1234,n,Dial(sip/,10)
2nd call into asterisk
exten = 1234,1,Dial(sip/,10)
I do link the idea of have a queue answer the calls and route to the
extensions, but will have to figure out a way to do this with have the
SIP extensions logging into the queues.
On Thu, May 8, 2008 at 1:53 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
An option to rotate between numbers is to
I noticed safe_asterisk is nolonger used from the init.d script (on
ubuntu) for asterisk-1.6.0-beta9. I'm curious if there is another
init.d script out there, or even the best way to call safe_asterisk.
Or is safe_asterisk nolonger the script of choice for starting,
restart asterisk.
One of the
Morning list,
Was curious if it is possible to decrease the time asterisk takes to
answer an incoming call to a zaptel interface.
Example:
[Jun 11 09:33:06] VERBOSE[4489] logger.c: -- Starting simple
switch on 'Zap/2-1'
[Jun 11 09:33:10] NOTICE[4489] chan_zap.c: Got event 18 (Ring Begin)...
number!
;
;immediate=yes
---
On Wed, Jun 11, 2008 at 9:38 AM, Paul Belanger [EMAIL PROTECTED] wrote:
Morning list,
Was curious if it is possible to decrease the time asterisk takes to
answer an incoming call to a zaptel interface.
Example:
[Jun 11 09:33:06] VERBOSE[4489] logger.c
Thanks Steve,
Forgot about callerID. We are not using callerID on the lines and
have disabled it. Asterisk now answers right away.
Thanks again,
PB
Do you actually have callerID on your line? That takes about two
seconds. Try removing it and see how much faster Asterisk answers.
That
List,
Anybody have success with Digium's G729 codec and asterisk 1.6.0?
Reading
http://www.russellbryant.net/blog/index.php/2008/03/05/codec_g729-v34-builds-now-available/
is seems they are build for 1.6 and trunk. But all I could find / use
is 1.4 builds from
List,
Having a little trouble with the following. Let me prefix by saying I
have blind transfers working from the following setup.
Inbound call [from-zap] (SIP/sv0071iv) answers.
Zaptel - Asterisk - SIP extension
SIP extension then blind transfers [from-sip]
---
SIP extension - Asterisk -
,SendDTMF(${EXTEN})
exten = _5XXX,n,Hangup()
Thanks again,
PB
On Tue, Jun 17, 2008 at 11:15 AM, Paul Belanger [EMAIL PROTECTED] wrote:
List,
Having a little trouble with the following. Let me prefix by saying I
have blind transfers working from the following setup.
Inbound call [from-zap
List,
Could anybody speak to the status of development in 1.6 branch? I
know support for SIP over TCP is pretty new / experimental but it
seems active development of it has slowed or stopped in recent months.
Is that a correct statement? Is SIP over TCP more a community project
or something
List,
What is the best way to restart asterisk after it crashes? Before we
used safe_asterisk, but looks like it has been removed from init.d
script on ubuntu (debian).
Suggestions?
Thanks,
PB
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Anybody else get theses warning?
[Jun 26 10:08:55] WARNING[3158]: chan_zap.c:4747 zt_handle_event:
Ring/Off-hook in strange state 6 on channel 1
PB
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AstriCon 2008 - September 22
List,
Anybody have a script around that will do this? We have to run
valgrind and asterisk to help troubleshoot a bug in the tracker.
Since we do not know how to reproduce the error, we'd like to run them
from an init.d script (simalar to safe_asterisk), email on crash, and
restart asterisk.
List,
We're working on an upcoming job that may require us to access a web
service (WS). I'm curious to hear peoples thoughts on the best way to
do this with asterisk. We'll be submitting a single number to the WS
and it will return a success or error.
One solution would be to write a simple
Can anybody confirm if this is the correct power adapter outputs:
12V DC 400mA
You adapter will have to outputs listed on it.
Thanks,
PB
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AstriCon 2008 - September 22 - 25
Hello,
I think there maybe an issue with my refer transfers. See below or attached:
No. TimeSourceDestination Protocol Info
1 0.00192.168.1.2 192.168.1.5 SIP/SDP
Request: INVITE sip:[EMAIL PROTECTED], with session
Evening all,
Just got my first PRI got event: HDLC Abort (6) on Primary D-channel of
span 1 error message. Our production box has been up for ~2 month. We
are Asterisk 1.0.9 with Slackware 10.1. Now I have search the lists
from this message and hear all the problem. Everything from asterisk
See inline comments:
Peter Svensson wrote:
What span is your clock source? A TE405P card can only operate in one
clock domain at a time. I.e. the same clock will be used on all of them.
Not correct, I actual have span 1 connected to my telco and span 2
connected to a Norstar PBX. See
Ryan Williams wrote:
I understand how CID works and how you must set CID when dialing out on
a PRI and how the phone company sets the name.
I was wondering how this works in regards to inbound calls. I have a pri
and I get the number that the caller is coming from but I do not get the
name.
Are your problems with incoming calls to your PRI or outgoing calls?
Are the calls being dropped or just not hitting your asterisk box?
PB
JOAO CARLOS MOURA wrote:
Hi All
Help.
We are using a T1 with Paetec Telecom in the Miami area, with a Digium card
into our Asterisk
software, and in
Oh you but I did, was not impressed. So, I sent them a friendly email
(hehe) asking WTF? What burns my ass, is they used a reply address of
[EMAIL PROTECTED]
PB
Jay Milk wrote:
Got an email this morning with the subject Welcome to Gizmo Project.
I didn't sign up with those yokels. Anyone else
See comments inline.
David Stude wrote:
Hi all,
I'm currently gearing up for a possible PBX replacement project using
Asterisk, and I'm just breaching the iceberg of information that's
available. I typically like to have something thick with pages in front
of me. Mahler's book was the first
Olle,
Awesome! Now that everybody know your aiming for September 1 for
Asterisk 1.2, I'm sure will make it.
Come' on Asterisk community, step up to the plate!
PB
Olle E. Johansson wrote:
Dear Asterisk Community,
Asterisk 1.0 was released at Astricon 2004, in September last year. It's
How about posting the output from the console? version of asterisk,
zaptel, etc.
Also, have you checked out http://www.voip-info.org/wiki-Asterisk+cmd+Dial
quote
Return codes
If all the called channels are busy, Dial will exit with a return code
of 0 and will continue in the current context
check in modules.conf:
load=res_indications.so
is it there?
Bernie Courtney wrote:
indications.conf reads as follows
[general]
country=us
[us]
description = United States / North America
ringcadance = 2000,4000
dial = 350+440
busy = 480+620/500,0/500
ring = 440+480/2000,0/4000
congestion =
Hello,
See comments inline
Alvaro Parres wrote:
Hi list:
I have a client that needs to connect a Asterisk PBX with a TE110P
of Digium and one Nortel Option 11.
Actually the Nortel Option 11 have a AMI E1 card. With it the have
problems of clock sync.
Is the Nortel the CPE or
with the PRI it's going to be easy all the work ??
Only one question the Nortel guys here, say that they need one more
clock to have a PRI card, is this correct
On 8/5/05, Paul Belanger [EMAIL PROTECTED] wrote:
Hello,
See comments inline
Alvaro Parres wrote:
Hi list:
I have
In you sip.conf what if you change:
register = 7771::[EMAIL PROTECTED]/7771
to
register = 7771:[EMAIL PROTECTED]/7771
PB
Jenna Cole wrote:
im using iptel.org SER proxy.
the proxy is working without authentication.
the problem is that the Asterisk is not sending a
REGISTER sip message.
Today the front page of http://www.voip-info.org/ was taken out by a
spammer. It also seem the history page for http://www.voip-info.org/
was also nuked. I've restored the best I could using google cache, but
still missing some information.
Who is an admin on http://www.voip-info.org/ and
Can you see the INVITE if you put up a trace on your gateway
(209.XXX.XXX.113)? Asterisk is not getting anything back that is why it
retransmits 5 times.
PB
OMS wrote:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
From: 512538XXX sip:[EMAIL
Where are your calls coming from? Are you connected to the Telco or PBX?
PB
Panitaxx wrote:
Hi,
thanks for your response. here is the log of one call:
Enabled debugging on span 1
Asterisk*CLI
Protocol Discriminator: Q.931 (8) len=33
Call Ref: len= 2 (reference 72/0x48) (Originator)
What type of client (Analog, SIP, IAX, etc??). Also, is
res_indications.so loaded?
PB
Stephen J. Wilcox wrote:
Hello,
can anyone help with my problem below, searching doesnt show any results..
thanks
Steve
On Wed, 3 Aug 2005, Stephen J. Wilcox wrote:
Hi,
I'm seeing a problem where if
lspci -v what output do you get? Also, what OS are you using?
Jeff Borders wrote:
I think I have a bad FXS module on my TDM400P.
Channel map:
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
2 channels configured.
ZT_CHANCONFIG
Can somebody who has a SoundPoint 501 please confirm the power adapter input /
output settings:
Input: 120V AC 60HZ 20W
Output: 24V DC 500mA
PB
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Thanks for all the replies! Looks like I was shipped the wrong
powersupply. I figured as much, cause when I first plugged it in it
took a while to boot, and started to smell something burning. :(
Time to RMA it back and get them to ship me the proper parts.
PB
Paul Belanger wrote:
Can
See comments inline!
Damon Estep wrote:
I have officially engaged in a pissing contest with the local Telco over
PRI calling name delivery.
Welcome to my world, I deal with theses guys daily! Errgiant arn't
they. We have a saying around work 'The telco is always wrong!'.
The telco
Cause No. 34 - No circuit available (circuit/channel congestion)
This cause indicates that there is no appropriate circuit/channel
presently available to handle the call.
http://www.telos-systems.com/?/techtalk/cause.htm
Might want to talk with your telco
BTW: Don't cross-post!
Matt
Hello list,
From time to time, I get the following warning in my message log.
Jun 23 15:56:40 WARNING[559]: PRI: XXX Missing handling for mandatory IE 12
(cs0, Connected
Number) XXX
Should I be concerned? To my knowledge I have not had an problems because of
it, but if
somebody can give me
Hello,
Quick Diagram:
Telco-PRI - Asterisk - Norstar PRI - Norstar PBX
(DMS100) (TE405P) (DMS100)
|
|
V
Cisco 7960G
(SIP)
I'm trying to change the Origination Number on my outgoing PRI, and running
into a weird
#root service asterisk start
Starting asterisk: [ OK ]
# ps aux
does asterisk show up as a process?
PB
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http://www.unlimitel.ca
not sure if they offer DID for Detroit
Technical Support wrote:
Can anyone recommend a good IAX provider offering numbers in Toronto and
Detroit?
___
open /etc/asterisk/modules.conf and add the following:
load app_meetme.so
save and close file; reload asterisk
Fabio Montemaggiore wrote:
I have install Flash Operator Panel but Asterisk show
this message:
WARNING[3564]: pbx.c:1650 pbx_extension_helper: No
application 'Meetme' for extension
On Sat, Jun 19, 2010 at 5:21 AM, michel freiha mich...@gmail.com wrote:
Waiting your reply
Reply: Do not cross-post to #asterisk-dev
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On Mon, Jun 21, 2010 at 12:25 PM, RSCL Mumbai rscl.mum...@gmail.com wrote:
What is the simplest way to achieve this ??
Use the transfer button on your phone?
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together, but each conference will be
hosted on there respective server.
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On Tue, Jun 22, 2010 at 5:47 PM, dan...@danielknoll.de wrote:
Can i join 2 dahdi (meetme) channels from different servers?
No
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How is anybody able to help when you XXX the relevant information?
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and dtmf options to res/res_agi.c
https://issues.asterisk.org/view.php?id=15531
[patch] MGCP Business Phone Packages patch
https://issues.asterisk.org/view.php?id=15159
[patch] chan_mgcp new feature: digitmaps definitions
https://issues.asterisk.org/view.php?id=16173
--
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Polybeacon
On Thu, Jun 24, 2010 at 2:56 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Its possible but not easy. Search for n-way conferencing on voip-info.org,
it has all the details on how to do it.
Or you could post the direct link:
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
--
Paul
On Fri, Jun 25, 2010 at 7:25 AM, Eyal Goltzman egoltz...@gmail.com wrote:
How can I trace\debug my dialplan?
*CLI dialplan show 1...@context
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[applicationmap]
zapflash = *0,callee,flash,()
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On Sat, Jun 26, 2010 at 8:25 AM, Gilles codecompl...@free.fr wrote:
Is there an up-to-date list of Asterisk appliances, ideally broken
down by price (ie. not just entreprise stuff, but also SOHO)?
Might get better results on asterisk-biz, and posting your budget price range.
--
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.
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New to Asterisk? Join
and using the settings below. You could easy
estimate the max amount of disk space one mailbox would use.
voicemail.conf
[general]
format=gsm
; Maximum number of messages per folder.
maxmsg=10
; Maximum length of a voicemail message in seconds
maxsecs=180
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On Mon, Jun 28, 2010 at 5:15 AM, John Taylor j...@vetsurgeon.org.uk wrote:
Any idea what may be happening?
acknowledged
https://issues.asterisk.org/view.php?id=16287
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?file_id=17192type=bug' -O -
| patch -p0 ??
This patch was merged in Asterisk 1.4.17, so you are already running it.
Does this mean I have a patched asterisk ? (I ask this because some
applications require a non-patched asterisk version)
Yes.
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Jabber
to hem
what is the billing software can i use to calculate the the calls and
manage the rate
Time to do some reading: http://astbook.asteriskdocs.org/
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On Tue, Jun 29, 2010 at 3:23 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
is there anyway to resolve it out, Means if SIP wants to send each call to
192.168.1.30 , but without entry in /etc/hosts.
Setup a DNS server.
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On Mon, Jun 28, 2010 at 10:04 PM, Zhang Shukun bit...@gmail.com wrote:
i want to know what is the best OS for install Asterisk 1.6.2.9,
which should work properly on working system.
Ubuntu 10.04 Server ?
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On Tue, Jun 29, 2010 at 12:51 PM, Kenny Watson
kwat...@geniusgroupltd.com wrote:
Is it simply a case of converting the prompts into other codecs and asterisk
will pick these up?
Yes, install both g729 and ulaw/alaw prompts to avoid trans-coding altogether.
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On Tue, Jun 29, 2010 at 10:06 AM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
Any idea what could be causing this?
Yes, network delay, packet loss, the Internet. Implement QoS and
bandwidth monitoring.
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the same results on my cell phone so I don't lose any
calls in the process.
*CLI core show application FollowMe
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On Thu, Jul 1, 2010 at 7:49 AM, Deepesh D deep.d2...@gmail.com wrote:
So that both extensions 101 and 102 rings simultaneously.
Yes, or use a local channel to dial multiple extensions.
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On Fri, Jul 2, 2010 at 1:22 PM, unsero...@aol.com wrote:
Where can i get the header files Asterisk 1.6.1.20 for Debian Lenny?
$ apt-get install asterisk-dev
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On Fri, Jul 2, 2010 at 1:38 PM, unsero...@aol.com wrote:
This are the header files for 1.4, not for 1.6.
Then how did you install asterisk 1.6?
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On Fri, Jul 2, 2010 at 1:51 PM, unsero...@aol.com wrote:
but i can't find header-files or dev-files there.
include folder
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On Mon, Jul 5, 2010 at 7:29 PM, AMARDEEP SINGH yahhod...@gmail.com wrote:
Please share experience if anyone have successfully configured Dialogic
JCT-T1 card with asterisk?
Your not going to find much; there is no channel driver for Dialogic.
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.
The more testers the better. If you have any problems or questions,
jump on #asterisk-testing on Freenode.
[1]
https://issues.asterisk.org/search.php?project_id=7status_id=55sticky_issues=onsortby=last_updateddir=DESChide_status_id=90
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On Tue, Jul 6, 2010 at 1:09 AM, C.Savinovich
c.savinov...@itntelecom.com wrote:
I am writing to you privately because I am an asterisk consultant and if you
need any help I can help you for a fee.
Unfortunately your email is not private, now that it is on a public list.
--
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access to the machine.
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New to Asterisk? Join us
trying to reach this client ?
Dial(SIP/Context)
This is all documented in sip.conf, otherwise the book
(http://astbook.asteriskdocs.org/).
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is not setup properly for outbound, you have no
credentials defined.
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.
If you can dialout without using AGI, then capture a 2nd debug log,
and post it. We can then compare why one works and the other does
not.
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On Thu, Jul 8, 2010 at 6:29 AM, Adil Zaaraoui adilzeaara...@yahoo.fr wrote:
But it does not work.
Any suggestion
Without posting a debug log it makes it hard to troubleshoot.
http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
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On Thu, Jul 8, 2010 at 2:51 AM, Manmohan Singh Jandu
manmoha...@gmail.com wrote:
crashes giving segmentation fault.
Read doc/backtrace.txt on how to capture and generate a backtrace.
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On Fri, Jul 9, 2010 at 7:38 AM, Adil Zaaraoui adilzeaara...@yahoo.fr wrote:
ok it works i had a problem with a syntax:
i had to wrire:
exten =_!X.,n(external),Dial(SIP/011212664800...@pstn2,,S(20))
Correct,
Dial(SIP/lo...@pstn2/011212664800450,,S(20))
Is not valid syntax
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On Fri, Jul 9, 2010 at 12:57 PM, Mike Ely mike...@amyskitchen.net wrote:
Has anyone figured out how to detect the actual cellphone answer rather than
the bogus one sent by the cell carrier?
*CLI core show application AMD
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Paul Belanger | dCAP
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Jabber: paul.belan
On Fri, Jul 9, 2010 at 12:42 PM, Jerry Geis ge...@pagestation.com wrote:
and show status gives me condition RED of course.
Physical problem, check cables / telco
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode
On Sun, Jul 11, 2010 at 5:40 AM, Peter Childs pchi...@bcs.org wrote:
1. Is there a function I'm missing to do this say..
is_queue_member(queuename,channel)
*CLI core show function QUEUE_MEMBER
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC
.
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
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New to Asterisk? Join
is terribly out of date. Always best to look in your CLI.
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
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check your license use by using:
*CLI g729 show licenses
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
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On Tue, Jul 13, 2010 at 7:41 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
I have no licenses and I want to avoid transcoding all together.
For terminating a call into Asterisk, you need g729 licenses. It is
that simple.
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan
not support trans-coding of video.
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
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modify your
logger.conf, reload logger, enable debugs, reproduce, disable debug
logs, edit logger.conf then reload.
It would be great to do all that from the CLI.
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
of make install show?
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
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New
On Thu, Jul 22, 2010 at 6:53 AM, Chandrakant Solanki
solanki.chandrak...@gmail.com wrote:
Is anything missing in above configuration or something goes wrong.?
kamailio != asterisk
Wrong list.
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC
On Fri, Jul 23, 2010 at 1:16 AM, bruce bruce bruceb...@gmail.com wrote:
Any help is appreciated.
Are you explicitly calling Hangup() within your dial-plans?
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
up the channel.
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
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New
?
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
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New to Asterisk
/app_voicemail.so: undefined symbol:
ast_smdi_mwi_message_destroy
This look to be a build problem with 1.8. We would need to see a copy
of your config.log and output from 'make install'. It is possible
your are loading old modules from 1.6 into 1.8. Check the timestamps
on these modules.
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Paul
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