I am using Elastix. Asterisk is used for PBX application in Elastix. I want
to make customize MOH. But not able to use MOH. I make simple extension in
asterisk conf file but no success :(
Below are the details of configuration files.
Even default MOH is also not working
*Asterisk Version 1.6
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
From: asterisk-users-boun...@lists.digium.co
On 04/07/11 03:00, Shariq Khan wrote:
I am facing one way audio problem in sip trunking between asterisk and
avaya.
+-+ ++
| avaya sip |---| P1 |
+-+ ++
|
|
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 wrote:
> On Thu, Apr 7, 2011 at 4:53 PM, Brian Henning wrote:
>
>> H
On Thu, Apr 7, 2011 at 4:53 PM, Brian Henning wrote:
> 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
> r
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: aster
> 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 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
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
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 simpl
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 l
_
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
SIP
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
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 => s,n,NoOp(${AVAIL
That should be CUT all caps I think
-Original Message-
From: satish patel
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 7 Apr 2011 20:45:21
To: asterisk-users
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] WARNING chan
They are on valid IP address range and working properly but when i switched off
that phone and dialing number from other phone i am getting following WARNING!!
So i would like to have some thing like who check CHANNEL first and then say
"Phone is not register" or If phone is available it will r
Just a guess but is it possible one of your SIP peers (7623 or 7624)
has an invalid IP address of 0.0.29.200? I wonder what "sip show
peers" shows.
On Thu, Apr 7, 2011 at 4:03 PM, satish patel wrote:
>
> Re-opening this issue.
>
> If i dial number which doesn't existing then i am getting follow
Indeed, that is what i would do except many users will not have a greeting.
so those without a greeting will not be able to login unless i generate a
"canned" greeting which i think i will have to do.
On Thu, Apr 7, 2011 at 4:33 PM, Danny Nicholas wrote:
> One more thought – assuming that your
One more thought - assuming that your users all have greetings recorded, you
could change vm-family to
/var/spoo/asterisk/voicemail/default/${callnum}/busy and keep the same plan.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
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,
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 => t,1,Voice
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]
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 (VoiceMailMain)
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).
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 t
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)
_
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
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
[mailto:asterisk-users-boun...@lists
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
_
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 wa
Is there a way to login to a voicemail box when someone pushes '#' during
greeting?
--
_
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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
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 yo
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 th
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 N
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})
>
> shir
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 only
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 wi
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 wi
On 7 April 2011 17:02, 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 curren
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 9
_
[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 end
_
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 possib
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
--
_
-- Bandwidth and Coloc
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 didn't
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 usin
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.
2011/4/7 Danny Nicholas
>--
>
> *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-Comm
_
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 Bryant Zimmerman
>
> 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 1.10 would be
u
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 di
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
russell
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
Subject: Re:
- 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 migr
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
--
_
-- Bandwidth and Co
Best I can tell, multi-tenant parking also hasn't worked in any of the 1.8.x
releases.
Chris
--
-
Chris Owen - Garden City (620) 275-1900 - Lottery (noun):
President - Wichita (316) 858-3000 -A stu
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 p
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Message: 2
Date: Thu, 07 Apr 2011 12:51:48 +0200
From: Gilles
Subject: Re: [asterisk-users]
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 u
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
On Apr 7, 2011, at 8:51 AM, Ishfaq Malik wrote:
> On Thu, 2011-04-07 at 08:37 -0400, Bryant Zimmerman wrote:
>>
>> On Apr 6, 2011, at 8:54 PM, Edwin Lam
>> wrote:
>>
>>> On 4/6/11 3:02 PM, Bryant Zimmerman wrote:
Thanks for your response. I have added the patch for 18818 per
Mic
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 wrote:
On Thu, 2011-04-07 at 08:37 -0400, Bryant Zimmerman wrote:
On Apr 6, 2011, at 8:54 PM, Edwin Lam
wrot
On Thu, 2011-04-07 at 08:37 -0400, Bryant Zimmerman wrote:
>
> On Apr 6, 2011, at 8:54 PM, Edwin Lam
> 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 h
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"
wrote:
On Apr 6, 2011, at 8:54 PM, Edwin Lam
wrote:
> On 4/6/11 3:02 PM, Bryant Zimmerman wrote:
>>
>> Thanks for yo
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 for
On Apr 6, 2011, at 8:54 PM, Edwin Lam
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
>> connections as all of
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
wrote:
On 4/6/11 3:02 PM, Bryant Zimmerman wrote:
Thanks for your response. I have added the patch for 18
Jonas Kellens 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 littl
any buddy is there for this solution.
On Thu, Apr 7, 2011 at 5:21 PM, Tzafrir Cohen wrote:
> 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
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
+9
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 jabber:tzafrir.co.
2011/4/7 Deka, Rajib IN MAA SL
> 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 i
Hi all,
Does anyone compiled asterisk using NKD build in android. Please
give some suggestions.
Thanks
Nikhil
--
_
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New to Asterisk? Join us for a live in
On Tue, 5 Apr 2011 17:38:15 -0400, Paul Dugas
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.
--
_
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 wrote:
> 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 >w
On Wed, 6 Apr 2011 09:46:12 +0100 (BST), Gordon Henderson
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 on the fly.
--
__
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 po
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 wrote:
>
> > On Wed, Apr 06, 2011 at 07:15:00PM +0530, mahesh katta wrote:
> > > Sir,
> > >
> > > i am using goautodial server , bri card is showing ok but when i t
I am facing one way audio problem in sip trunking between asterisk and
avaya.
+-+ ++
| avaya sip |---| P1 |
+-+ ++
|
|
|
+---
[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 wrote:
> On Wed, Apr 06, 2011 at 07:15:00PM +0530, mahesh katta wrote:
>
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