[root@go ~]#
dahdi_hardware
pci::04:02.0 wcb4xxp+ d161:b410 Digium Wildcard B410P
This was comming and even i enterd that file last.
then also its not connecting
On Wed, Apr 6, 2011 at 9:04 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Wed, Apr 06, 2011 at 07:15:00PM
I am facing one way audio problem in sip trunking between asterisk and
avaya.
+-+ ++
| avaya sip |---| P1 |
+-+ ++
|
|
|
Hi,
Un-top-posting
On Thu, Apr 07, 2011 at 12:53:18PM +0530, mahesh katta wrote:
On Wed, Apr 6, 2011 at 9:04 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Wed, Apr 06, 2011 at 07:15:00PM +0530, mahesh katta wrote:
Sir,
i am using goautodial server , bri card is showing ok
Hello List,
I have found that asterisk supports only forwards in-dialog MESSAGE method.
That is, if the MESSAGE method is sent within an active call.
But according our requirement we need to send MESSAGE method to the other leg
without being in a call (general stateless proxy forward). Is it
On Wed, 6 Apr 2011 09:46:12 +0100 (BST), Gordon Henderson
gordon+aster...@drogon.net wrote:
Have a look at these:
Thanks much Gordon. I'll study the scripts you mentionned. It looks
like iptables is good enough and I won't have to install a second tool
to watch the logs and reconfigure iptables
Sir,
my files are in fistmail that is my configuration.
and till its disconnecting the line
On Thu, Apr 7, 2011 at 2:35 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
Hi,
Un-top-posting
On Thu, Apr 07, 2011 at 12:53:18PM +0530, mahesh katta wrote:
On Wed, Apr 6, 2011 at 9:04 PM,
On Tue, 5 Apr 2011 17:38:15 -0400, Paul Dugas
p...@dugasenterprises.com wrote:
First, this appears to be working for me though I'm not 100% sure of
that and cannot guarantee it will for you in any way, shape or form.
With the lawyering out of the way...
Thanks a lot, Paul.
--
Hi all,
Does anyone compiled asterisk using NKD build in android. Please
give some suggestions.
Thanks
Nikhil
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2011/4/7 Deka, Rajib IN MAA SL rajib.d...@siemens.com
Hello List,
I have found that asterisk supports only forwards in-dialog MESSAGE method.
That is, if the MESSAGE method is sent within an active call.
But according our requirement we need to send MESSAGE method to the other
leg
On Thu, Apr 07, 2011 at 04:48:13PM +0530, mahesh katta wrote:
Sir,
my files are in fistmail that is my configuration.
and till its disconnecting the line
/me gives up. Anybody else wants to take a shot here?
--
Tzafrir Cohen
icq#16849755
On Thu, Apr 07, 2011 at 04:58:33PM +0530, Nikhil wrote:
Hi all,
Does anyone compiled asterisk using NKD build in android. Please
give some suggestions.
Have you tried? What errors do you get?
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
any buddy is there for this solution.
On Thu, Apr 7, 2011 at 5:21 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Thu, Apr 07, 2011 at 04:48:13PM +0530, mahesh katta wrote:
Sir,
my files are in fistmail that is my configuration.
and till its disconnecting the line
/me gives up.
Jonas Kellens jonas.kellens at telenet.be writes:
On 03/16/2011 08:39 PM, Jonas Kellens wrote:
Found the answer to my own question : fromuser in the peer definition
Kind regards,
Jonas.
--
_
Can you extend a
Oh! Boy,
Is it ture 1.8.3 is unstable? We are planning to put this in
production. Please suggest me what should I do?
--
Sent from my iPhone
On Apr 6, 2011, at 8:54 PM, Edwin Lam edwin@officegeneral.com
wrote:
On 4/6/11 3:02 PM, Bryant Zimmerman wrote:
Thanks for your response. I
On Apr 6, 2011, at 8:54 PM, Edwin Lam edwin@officegeneral.com
wrote:
On 4/6/11 3:02 PM, Bryant Zimmerman wrote:
Thanks for your response. I have added the patch for 18818 per
Michel Verbrask's
recomendation. It appers that it has made quite a difference. I
don't have an PRI
Sir,
I am using B410p card which BRI. and Mediatrix4400 is bri line provider in
dubai.
below configuration is my bri card configuration. and when try to connect
the call its going disconnect on cli getting
[Apr 6 09:36:37] WARNING[6433]: channel.c:3443 ast_request: No channel type
registered
Holy cow!!
Can I just build 1.8.2 over existing 1.8.3 ?
When we are going to fix all this thing???
--
Sent from my iPhone
On Apr 7, 2011, at 8:37 AM, Bryant Zimmerman brya...@zktech.com
wrote:
On Apr 6, 2011, at 8:54 PM, Edwin Lam edwin@officegeneral.com
wrote:
On 4/6/11 3:02 PM,
On Thu, 2011-04-07 at 08:37 -0400, Bryant Zimmerman wrote:
On Apr 6, 2011, at 8:54 PM, Edwin Lam edwin@officegeneral.com
wrote:
On 4/6/11 3:02 PM, Bryant Zimmerman wrote:
Thanks for your response. I have added the patch for 18818 per
Michel Verbrask's
recomendation. It appers
We don't have realtime configuration everything is in plain text file.
Is 1.8.3 has realtime issue or general issue?
--
Sent from my iPhone
On Apr 7, 2011, at 8:51 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
On Thu, 2011-04-07 at 08:37 -0400, Bryant Zimmerman wrote:
On Apr 6, 2011, at 8:54
On Apr 7, 2011, at 8:51 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
On Thu, 2011-04-07 at 08:37 -0400, Bryant Zimmerman wrote:
On Apr 6, 2011, at 8:54 PM, Edwin Lam edwin@officegeneral.com
wrote:
On 4/6/11 3:02 PM, Bryant Zimmerman wrote:
Thanks for your response. I have added the
On 11-04-07 08:20 AM, Satish Patel wrote:
Is it ture 1.8.3 is unstable? We are planning to put this in production.
Please suggest me what should I do?
This is a loaded question, since it really depends on what you plan to
do. What does your migration plan look like? What sort of testing have
Right now I'm testing 1.8.3 in devlopment and respose it pretty good
without realtime. (I didn't set realtime).
I ran stress test with sipp and pass 5000 call with RTP and no issue
at all. I got hogging at system resource but no issue at asterisk.
Look like I might go with 1.8.3 and later
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Message: 2
Date: Thu, 07 Apr 2011 12:51:48 +0200
From: Gilles codecompl...@free.fr
Subject: Re: [asterisk-users] Iptables
Good morning ...
I'm using Asterisk 1.4.40 AgentCallbackLogin in a Call
Center. What is happening isthat when the Call Center has more than
15 simultaneous calls the login application isextremely slow to fall
into the low priority, ie, the agent can log in, but takes about 1minute
to drop in
Best I can tell, multi-tenant parking also hasn't worked in any of the 1.8.x
releases.
Chris
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Chris Owen - Garden City (620) 275-1900 - Lottery (noun):
President - Wichita (316) 858-3000 -A
On 7 Apr 2011, at 14:32, Deka, Rajib IN MAA SL wrote:
Is the following is the link for getting the source,
http://svn.asterisk.org/svn/asterisk/trunk/
Please try not to reply to the entire digest..
S
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- Original Message -
On 11-04-07 08:20 AM, Satish Patel wrote:
Is it ture 1.8.3 is unstable? We are planning to put this in
production.
Please suggest me what should I do?
This is a loaded question, since it really depends on what you plan
to
do. What does your migration plan
Is the following trunk has development version of out-of-call messaging
capability, also what is the version of asterisk,
http://svn.asterisk.org/svn/asterisk/trunk/
Regards,
Rajib
--
Message: 10
Date: Thu, 7 Apr 2011 14:42:35 +0100
From: Steven Howes
On 11-04-07 09:59 AM, Deka, Rajib IN MAA SL wrote:
Is the following trunk has development version of out-of-call messaging
capability, also what is the version of asterisk,
http://svn.asterisk.org/svn/asterisk/trunk/
I don't believe the branches has been merged into trunk, you can use
Hi,
I know it sounds weird, and this is part of the reason I have not
reported that sooner.As I upgraded from 1.6.2.x to 1.8.x several
months ago I am experiencing this problem. If a call is initiated from
a DAHDI extension after no DAHDI extensions were used for some time,
arbitrary DTMF
2011/4/7 Bryant Zimmerman brya...@zktech.com
For me 1.8.3.2 has been the worst build that I have tried to use as far a
stability in a very long time.
Hi,
If my memory serves me right, first usable 1.4 version was 1.4.21 or
something.
Time will tell if things are improving and hopefully next
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, April 07, 2011 10:27 AM
To: brya...@zktech.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Asterisk 1.8.3
2011/4/7 Danny Nicholas da...@debsinc.com
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
*Sent:* Thursday, April 07, 2011 10:27 AM
*To:* brya...@zktech.com; Asterisk Users Mailing
On 11-04-07 11:26 AM, Olivier wrote:
Is the asterisk testing framework easy enough to work with so that we could
feed new tests into it and help devs to identify such regressions before GA
release ?
+1
There is a learning curve to creating tests for the testsuite[1], but
nothing too drastic.
Any ideas on why callers who call into my customer's SIP trunk are not hearing
a ringback tone? I had this on one other asterisk system, and wound up needing
to set progressinband=yes in the SIP trunk config.
I have set this on the current system restarted asterisk, but to no avail.
I am
On 11-04-07 09:45 AM, --[ UxBoD ]-- wrote:
And don't forget that call pickup crashes Asterisk from what would appear
release 1.8.1 upwards! We have had to back level to that latest 1.6 branch.
https://issues.asterisk.org/view.php?id=18654
I ran into this issue as well on 1.8.3.2, but I
Is it possible to start MOH when calling to DAHDI Channel that has ISDN E1
connected with it. When the called party press hold on his phone then
asterisk start MOH??
--
Regards,
Shariq Khan
0333-3501125
--
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-- Bandwidth and
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shariq Khan
Sent: Thursday, April 07, 2011 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MOH on DAHDI PRI Channels
Is it
_
[Shariq Khan]
Is it possible to start MOH when calling to DAHDI Channel that has ISDN E1
connected with it. When the called party press hold on his phone then
asterisk start MOH??
[Danny Nicholas]
Question #1
Dial(DAHDI/1/5551212,20,m) will play moh until the other
Danny,
Thanks for the support, but i need to hold the customer and play MOH after
answering the call. As you know that the signalling codes of SIP and ISDN
are almost same, that's why i was thinking that MOH can work on DAHDI as
well.
--
Regards,
Shariq Khan
0333-3501125
On Thu, Apr 7, 2011 at
On 7 April 2011 17:02, Douglas Mortensen d...@impalanetworks.com wrote:
Any ideas on why callers who call into my customer's SIP trunk are not
hearing a ringback tone? I had this on one other asterisk system, and wound
up needing to set progressinband=yes in the SIP trunk config.
I have set
asterisk-users-boun...@lists.digium.com wrote:
On 03/30/2011 01:32 PM, SebA wrote:
So, I've compiled and installed libpri-1.4.11.5,
dahdi-linux-complete-2.4.1+2.4.1 and asterisk-1.6.2.17.1, but
chan_dahdi is not getting built. If I do a make menuselect in
asterisk I see it listed with XXX,
asterisk-users-boun...@lists.digium.com wrote:
On 03/30/2011 01:32 PM, SebA wrote:
So, I've compiled and installed libpri-1.4.11.5,
dahdi-linux-complete-2.4.1+2.4.1 and asterisk-1.6.2.17.1, but
chan_dahdi is not getting built. If I do a make menuselect in
asterisk I see it listed with XXX,
I presume you mean contrib/scripts/install_prereq but I'm not sure how to
use it or whether it is applicable to this situation. I had a look over the
source code and it seems to be heavily dependent on what distribution you
are running. For Debian, quite a lot are listed, but for Redhat it is
On Wednesday 06 April 2011 10:09:07 satish patel wrote:
I used following hint dialplan and i ran show hints but its showing only
one extension what about other 200 phones status ?
exten = _7[456]XX,hint,SIP/${EXTEN}
exten = _7[456]XX,1,Macro(stdexten,${EXTEN},sip/${EXTEN})
shirley*CLI
On Wednesday 06 April 2011 09:49:17 Jon Farmer wrote:
Hi
I have a call into a MeetMe conference that when I do a core show
channel returns
NativeFormats: 0x4 (ulaw)
WriteFormat: 0x1000 (g722)
ReadFormat: 0x1000 (g722)
Can someone explain what the differences between Native, Wite
On Wednesday 06 April 2011 14:53:00 Hans Witvliet wrote:
I'm going to have a go with realtime mysql.
Just wondering, most examples i came across while googling, was with 1.6
systems.
So any drastic changes with 1.8.3, table-layout? other pitfalls?
This isn't a pitfall that comes with the
Very weird mate...I would have replied sooner, but in reality there's a
LOT of troubleshooting to be done and it would require working with your
provider. It sounds like (if you're sending a bye when your calls
disconnect) you never receive an actual 200 OK stating the call is
picked up and so
On 4/7/2011 11:02 AM, Douglas Mortensen wrote:
Any ideas on why callers who call into my customer's SIP trunk are not
hearing a ringback tone? I had this on one other asterisk system, and wound
up needing to set progressinband=yes in the SIP trunk config.
I have set this on the current
Is there a way to login to a voicemail box when someone pushes '#' during
greeting?
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Thursday, April 07, 2011 1:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] asterisk login to voicemail
Is there a
i'm afraid my setup is more complex than that
[inbound]
exten = _X.,1,agi(route.pl)
after some logic using mysql, route.pl then does:
$AGI-exec(VoiceMail, $options);
at that point, I would like the caller to be able to push '#' and be
prompted for Password for that particular mailbox
On
If you add the exten = #,1 line to the end of the inbound context, that
should do it for you. If not, change $AGI-exec(VoiceMail,$options) to go
to a context instead of running Voicemail directly.
_
From: asterisk-users-boun...@lists.digium.com
I'm sorry I'm new to AGI programming but i did this:
$AGI-set_variable(vmbox, $options);
$AGI-set_context(voicemail);
and in extensions.conf i have:
[voicemail]
exten = s,1,VoiceMail(${vmbox},su)
exten = #,n,VoiceMailMain(${EXTEN}@4)
I keep getting 603 declined when i call the
You're on the right track, but # is going to blow away ${EXTEN} so you are
going to have to hard-code that value or use a different variable that
contains what should have been in ${EXTEN}. Also, #,n has to be #,1 (each
part of a context has to have line 1 - not my rule, Asterisk's)
_
Ok, i have this now...
[voicemail]
exten = s,1,VoiceMail(${vmbox},su)
exten = *,1,VoiceMailMain(${callednum})
in AGI i have:
$AGI-set_variable(callednum, $options);
$AGI-set_variable(vmbox, $options);
$AGI-set_context(voicemail);
I'm getting a busy signal and this
As I see it, callednum and vmbox should not be the same. Vmbox is a good
mailbox you're going to reach if the user doesn't hit #, callednum is the
fallback number that you are going to use and should be an established
mailbox (3-4 digits) not a full number (10 digits you have indicated).
Actually the mailbox is 7167435000...
in this case the two variables are the same and the mailbox 7167435000 does
exist
I want someone to call 7167435000 reach mailbox 7167435000 (VoiceMail), if
they push * during greeting, then i went them to prompted for a PIN for that
mailbox
Re-opening this issue.
If i dial number which doesn't existing then i am getting following error. So
is there anyway i can fix my dialplan to check whether that number exist or not
or i can check channel status.
shirley*CLI
== Using SIP RTP CoS mark 5
-- Executing [7623@from-sip:1]
Here's your solution
[vmtest]
exten = s,1,background(vm-Family,3)
exten = s,n,waitexten(3)
exten = s,n,Voicemail(${callnum}@default)
exten = *,1,VoicemailMain(${callnum}@default)
exten = #,1,VoicemailMain(${callnum}@default)
exten = i,1,Voicemail(${callnum}@default)
exten =
Unfortunately, that solution will not work for me... The user must be able
to hit * during the greeting of any voicemail and be prompted for the
Password to that particular mailbox looks like i got a lot of
programming to do to create a work around for this... thanks for your
help...
On Thu,
One more thought - assuming that your users all have greetings recorded, you
could change vm-family to
/var/spoo/asterisk/voicemail/default/${callnum}/busy and keep the same plan.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Indeed, that is what i would do except many users will not have a greeting.
so those without a greeting will not be able to login unless i generate a
canned greeting which i think i will have to do.
On Thu, Apr 7, 2011 at 4:33 PM, Danny Nicholas da...@debsinc.com wrote:
One more thought –
Just a guess but is it possible one of your SIP peers (7623 or 7624)
has an invalid IP address of 0.0.29.200? I wonder what sip show
peers shows.
On Thu, Apr 7, 2011 at 4:03 PM, satish patel satish...@hotmail.com wrote:
Re-opening this issue.
If i dial number which doesn't existing then i
They are on valid IP address range and working properly but when i switched off
that phone and dialing number from other phone i am getting following WARNING!!
So i would like to have some thing like who check CHANNEL first and then say
Phone is not register or If phone is available it will
That should be CUT all caps I think
-Original Message-
From: satish patel satish...@hotmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 7 Apr 2011 20:45:21
To: asterisk-usersasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial
Yes! You are right! Its working. Now issue is we have SIP extension for
local office users and same number has IAX extension for remote
traveling users. How could i use ChanIsAvail with best action ?
I did following
exten = s,1,ChanIsAvail(${ARG2}IAX2/${ARG1},20,t)
exten =
Hi Everyone,
We want to be able to momentarily or temporarily provide CONGESTION or
DE-REGISTER a SIP PEER to asterisk through WEB GUI. I don't want to indulge
into dial-plan and write changes to .conf file every-time. Is there any way
that a SIP PEER can be de-registered for an amount of time or
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Thursday, April 07, 2011 4:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Any way to temporarily disable a registered
Hi,
Now and then our SIP phones ring with asterisk showing as the caller-ID.
Upon picking up the receiver, there is about five seconds of silence and
then the channel is closed (hangup). Can anyone offer some insight? Here's
relevant snippets from my extensions.conf and Master.csv log:
This
I have inbound calls going directly to a ring group. When callers call in, they
(the callers) hear complete silence even though the phones that are part of the
ring group ARE ringing properly. Employees can answer the calls when their
phones ring, and everything works fine.
The problem is
Steve. Thanks for the insight. I won't pretend to know what early-audio is,
but I guess I'm about to find out :-).
Also, I believe that I have a nearly identical setup like this with the exact
same SIP provider w/o any trouble. However, I think that system must be running
asterisk 1.4 or 1.2
On 4/7/2011 4:54 PM, Douglas Mortensen wrote:
I have inbound calls going directly to a ring group. When callers call in,
they (the callers) hear complete silence even though the phones that are part
of the ring group ARE ringing properly. Employees can answer the calls when
their phones
Unfortunately, that solution will not work for me... The user must be able to
hit * during the greeting of any voicemail and be prompted for the Password
to that particular mailbox looks like i got a lot of programming to do to
create a work around for this... thanks for your help...
We were getting a lot of those. We installed IPTables with blocking of
everything outside of North America and they all but vanished.
No direct evidence, but a pretty good empirical guess that they were related
to hackers trying to get paths to the US.
CF
-Original Message-
From:
We experience exact same thing on DAHDI with Sangoma USB FXO device on short
circuited lines. Phantom calls are actually due to a short in the lines that
happen occasionally.
-Bruce
On Thu, Apr 7, 2011 at 7:16 PM, Warren Selby wcse...@selbytech.com wrote:
On Thu, Apr 7, 2011 at 4:53 PM, Brian
First, I'm pretty sure avaya peer needs to friend. Try adding the below to
sip.conf and do a reload.
[general]
externip = the.wan.ext.ip
localnet = 192.168.1.0/255.255.255.0
If that doesn't work, add nat=yes to avaya peer=friend
Skyler
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