Thank you Paul.
I have downloaded the code.
How out-of-call messaging can be configured in the Dialplan?
Regards,
Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com
Mob: +91-9176780669 |
Wow, thanks, that worked...
in case anyone is interested this is what i did
[voicemail]
exten = a,1,VoiceMailMain(${MAILBOXID}@${MAILBOXCONTEXT},p)
in AGI...
$AGI-set_variable(MAILBOXID, $options);
$AGI-set_variable(MAILBOXCONTEXT,4);
$AGI-set_context(voicemail);
$AGI-exec(VoiceMail,
SIP/8.224.32.2-AGI Rx SET VARIABLE MAILBOXID 7167435000
SIP/8.224.32.2-AGI Tx 200 result=1
SIP/8.224.32.2-AGI Rx SET VARIABLE MAILBOXCONTEXT 4
SIP/8.224.32.2-AGI Tx 200 result=1
SIP/8.224.32.2-AGI Rx SET CONTEXT voicemail
And the mailbox 7167435000
Hi list,
I have a user, referenced by his IMSI (IMSI208300618462231), who is
assigned to extension 2111 in /etc/asterisk/extensions.conf and
sip.conf (see below).
From time to time, registration of this user fails (see below), but I
do not know why. Anybody has a clue what could be wrong ? Is
Hey all!
I'm trying to figure out a way to manipulate a call's caller ID based off of
the caller's caller ID. Basically, I've got a situation where anything that
goes through an Nortel Opt11's IVR comes out with the caller ID 400 (the
Opt11's IVR's ext). When the call goes out the trunk that
Add exten = 6000,n,StartMusicOnHold() to the end of your current dialplan and
try again.
Thanks,
--Warren Selby, dCAP
On Apr 8, 2011, at 1:51 AM, virendra bhati virbh...@gmail.com wrote:
I am using Elastix. Asterisk is used for PBX application in Elastix. I want
to make customize MOH. But
Why are you using agi for this ? They are inbuild features of asterisk.
Or may be I am missing something
--
Sent from my iPhone
On Apr 8, 2011, at 8:26 AM, vip killa vipki...@gmail.com wrote:
Wow, thanks, that worked...
in case anyone is interested this is what i did
[voicemail]
exten =
Almost,
If you use Asterisk version 1.6 or higher use
Exten = ExecIf($[${CALLERID(number)} = 400]?Set(CALLERID(num)=)
Or
Exten = ExecIf($[${CALLERID(number)} = 400]?Set(CALLERID(all)=)
Michel Verbraak
**
http://www.intercommit.nl/
On 08-04-11 15:56, Louis Carreiro wrote:
Hey all!
Where this revers IP comes from ?
== Using SIP RTP CoS mark 5
-- Executing [7623@from-sip:1] Macro(SIP/7527-006b,
stdexten,7623,SIP/7623) in new stack
-- Executing [s@macro-stdexten:1] ChanIsAvail(SIP/7527-006b,
SIP/7623IAX2/7623,20,t) in new stack
-- Hungup
Hi Everyone,
Looking to create a flash status update page from Asterisk events using PHP
MING but I can't seem to find much documentation other than those on PHP
site. Has anyone ever tried or has came across PHP Ming usage for Asterisk?
Any hints of much appreciated.
***I am aware of FOP using
On 11-04-08 10:48 AM, satish patel wrote:
Where this revers IP comes from ?
== Using SIP RTP CoS mark 5
-- Executing [7623@from-sip:1] Macro(SIP/7527-006b,
stdexten,7623,SIP/7623) in new stack
-- Executing [s@macro-stdexten:1] ChanIsAvail(SIP/7527-006b,
Perfect! That was it! Thanks!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michel
Verbraak
Sent: Friday, April 08, 2011 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
No I am not using any realtime config. its text file..
shirley*CLI core show settings
PBX Core settings
-
Version: 1.8.3.2
Build Options: LOADABLE_MODULES
Maximum calls: 250 (Current 0)
Maximum open file handles: Not set
From: Chris Owen ow...@hubris.net
Sent: Thursday, April 07, 2011 9:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8.3
Best I can tell, multi-tenant parking also
On 4/8/2011 4:57 AM, Naomi Rosenberg wrote:
Hi,
I would have thought that when spawning a channel using the Originate()
dialplan command, variables prefixed with two underscores would be preserved.
However this does not work in the following case.
Dialplan code:
[intern]
exten =
@Paul - many time i am gettting following SIP error when channel isn't
available. I want to get rid on this revers thing. I tried all version
1.8.1,1.8.2,1.8.3 but not fix :(
[Apr 8 11:52:22] WARNING[13920]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2f40580 (len 793) to 0.0.29.200:5060
Thanks. That's as I thought (feared). Dial is not an option in this case but I
have come up with a workaround involving using a reference number as the
extension and then doing a database call. Not pretty but it works!
Naomi
- Original Message -
From: Sherwood McGowan
On Fri, Apr 08, 2011 at 09:11:20AM +, salaheddine elharit wrote:
i have a question related to CRC, yesterday i had an issue in our span when
i verify with zttool i found recovering instead ok
i verify the zaptel.conf and i found
# Autogenerated by /usr/sbin/zapconf on Thu Apr 7
Another option is to pass the information in the extension. At times I have an
extension like
_[s][o][m][e]-[e][x][a][m][p][l][e].
And call it like some-example:info1:info2 and use cut to extract the info1 and
info2 values. Not real pretty but as this is computer generated calls it gets
the
On 4/8/2011 10:57 AM, Naomi Rosenberg wrote:
Thanks. That's as I thought (feared). Dial is not an option in this case but
I have come up with a workaround involving using a reference number as the
extension and then doing a database call. Not pretty but it works!
Naomi
I'm not sure why
On 4/8/2011 11:05 AM, Jim Dickenson wrote:
Another option is to pass the information in the extension. At times I have
an extension like
_[s][o][m][e]-[e][x][a][m][p][l][e].
And call it like some-example:info1:info2 and use cut to extract the info1
and info2 values. Not real pretty but as
Look at this sip debug its saying something related Retransmitting #1 (no NAT)
to 0.0.29.200:5060:
-- Executing [7624@from-sip:1] Macro(SIP/7527-00c2,
stdexten,7624,SIP/7624) in new stack
-- Executing [s@macro-stdexten:1] Dial(SIP/7527-00c2,
can you explain how this can be done simpler?
On Fri, Apr 8, 2011 at 10:10 AM, Satish Patel satish...@hotmail.com wrote:
Why are you using agi for this ? They are inbuild features of asterisk.
Or may be I am missing something
--
Sent from my iPhone
On Apr 8, 2011, at 8:26 AM, vip killa
I have this for same function.
[voice-mail]
;VM for external users calling from PSTN prompt for mailbox number and pin
exten = 8000,1,Answer()
exten = 8000,n,Wait(1)
exten = 8000,n,VoicemailMain(@default)
exten = 8000,n,Hangup()
;VM for internal users only pin
exten = 8500,1,Answer()
exten
On 11-04-08 11:55 AM, satish patel wrote:
@Paul - many time i am gettting following SIP error when channel isn't
available. I want to get rid on this revers thing. I tried all version
1.8.1,1.8.2,1.8.3 but not fix :(
Best you can do is collect a full debug[1] log and see when the issue is
Dan et al;
This looks like a perfect solution.
However, I have one issue. If I initiate the macro manually (put it in
the proper context/dialplan) it works. I see the *.wav file being
created and growing in the /var/spool/asterisk/monitor directory.
If I try to implement it adding the
On 7 April 2011 23:04, Douglas Mortensen d...@impalanetworks.com wrote:
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
On 11-04-08 12:56 PM, Paul Belanger wrote:
On 11-04-08 11:55 AM, satish patel wrote:
@Paul - many time i am gettting following SIP error when channel isn't
available. I want to get rid on this revers thing. I tried all version
1.8.1,1.8.2,1.8.3 but not fix :(
Best you can do is collect a
I can try but i have same issue with chan_sip channel also. and next we have
scheduled to put this box 1.8.3.2 in production :(
-S
Date: Fri, 8 Apr 2011 13:16:30 -0400
From: pabelan...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] IAX2/0.0.29.199
On
I have opened case here: https://issues.asterisk.org/view.php?id=19087
Date: Fri, 8 Apr 2011 13:16:30 -0400
From: pabelan...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] IAX2/0.0.29.199
On 11-04-08 12:56 PM, Paul Belanger wrote:
On 11-04-08 11:55 AM,
I have just compiled asterisk 1.6.x and its working without any issue no error
related revers lookup etc.. Look like there is some glitch in asterisk 1.8 :(
-S
Date: Fri, 8 Apr 2011 13:16:30 -0400
From: pabelan...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re:
I tried to compile your version and got bunch of error on make and it failed
to compile.
root@satish-desktop:/home/satish/issue18183# make
[CC] chan_iax2.c - chan_iax2.o
chan_iax2.c: In function âsocket_processâ:
chan_iax2.c:11533: error: invalid storage class for function
Is there a way for asterisk's voicemail to send an email (including
voicemail attachment) to multiple email addresses?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Friday, April 08, 2011 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] send voicemail to multiple emails
Is there
On 4/8/2011 1:13 PM, vip killa wrote:
Is there a way for asterisk's voicemail to send an email (including
voicemail attachment) to multiple email addresses?
--
_
-- Bandwidth and Colocation Provided by
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Sherwood McGowan
Sent: Friday, April 08, 2011 1:16 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] send voicemail to multiple emails
That does not sound easy... besides these email addresses would be taken
from a MySQL database.
The easiest way would be to set up an alias in your MTA configuration.
That way, you could configure the mailbox for the alias email address
and copies would be sent to all addresses in the alias
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Friday, April 08, 2011 1:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] send voicemail to multiple emails
That
On 4/8/2011 1:18 PM, vip killa wrote:
That does not sound easy... besides these email addresses would be
taken from a MySQL database.
The easiest way would be to set up an alias in your MTA configuration.
That way, you could configure the mailbox for the alias email address
On 4/8/2011 1:20 PM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Sherwood McGowan
Sent: Friday, April 08, 2011 1:16 PM
To: asterisk-users@lists.digium.com
Subject: Re:
Hi Everyone,
I am testing Asterisk 1.8 AMI events. The voipinfo page on AMI events is
specific to 1.6. I am wondering if the developers cared to write about the
new events that are spit out in Asterisk 1.8 somewhere on the web?
I checked the tar ball for asterisk 1.8 and documentation doesn't
On 11-04-08 02:56 PM, Bruce B wrote:
Hi Everyone,
I am testing Asterisk 1.8 AMI events. The voipinfo page on AMI events is
specific to 1.6. I am wondering if the developers cared to write about the
new events that are spit out in Asterisk 1.8 somewhere on the web?
It doesn't exist. The only
Dan et al;
This looks like a perfect solution.
However, I have one issue. If I initiate the macro manually (put it in
the proper context/dialplan) it works. I see the *.wav file being
created and growing in the /var/spool/asterisk/monitor directory.
If I try to implement it adding the
On Fri, Apr 8, 2011 at 3:35 PM, Silver Thorne szilvertho...@gmail.comwrote:
Dan et al;
This looks like a perfect solution.
snip
It pretty much is. I've used it in similar situations. I was just about to
respond to your original post, but I see you reposted here, so I'll respond
here.
On Fri, Apr 8, 2011 at 1:18 PM, vip killa vipki...@gmail.com wrote:
That does not sound easy... besides these email addresses would be taken
from a MySQL database.
It's actually what you're going to end up doing, whether you do it on the
MTA level or your code it into your script that you
Hi Selby,
First of all thanks for reply, I added that line at the end of dialplan but
still the same.
Actually *asterisk default MOH is also not playing*. after 12 sec I get this
message...
* -- Stopped music on hold on SIP/1001-000f*
Strange when I did the same thing with asterisk 1.4.39
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