Re: [asterisk-users] How to secure our Asterisk server from hacker's ?
Hi List, Yes you are right but I want to cross check to outside world to. How they will support me in such case... :) On Thu, Jun 16, 2011 at 11:23 AM, Alex Balashov abalas...@evaristesys.comwrote: I thought the idea was that Asterisk Engineers already know the answers to such questions? On 06/16/2011 01:52 AM, virendra bhati wrote: Hi List, I want to secure my server from the hacker's. What is the case by which I can protest it. I have done security of Dialplan, Sip,IAX base security. For linux we are working on Iptables. What else is left so that I will do it too... -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] #include filename
Hi, I am using asterisk1.2 In this, my dialplan is going large , so i need to configure this small pieces for this, i did in my extensions.conf when I dial the 123 its not going , means that file is not reading. is there any parameters to add any where ? please tell me this #include is not working ... extensions.conf [general] [global] trunk=zap/g0 #include exten-internal.conf [default] exten = _X.,1,Answer exten = _X.,2,Dial(Zap/g0/999898999,,to) exten = _X.,3,Hangup /etc/asterisk/exten-internal.conf exten = 123,1,Answer exten = 123,2,Dial(SIP/5024,,t) exten = 123,2,Hangup Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inbound call not dialing exten
Hi all,
I have 100 DID's which is 4578900 to 4578999 , and i have 5001 to 5099
extensions. when incomming call come to this DID no. (4578901) that time
5001 extestinsion should ring.
below my dial plan is not getting any result , inthat has any mistakes.
please help.
exten = _45789XX,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _45789XX,1,Set(Dest=2{EXTEN:-2})
exten =
_45789XX,2,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten = _45789XX,3,Dial(SIP/${Dest},,tTo)
exten = _45789XX,4,Hangup
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
--
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Re: [asterisk-users] How to secure our Asterisk server from hacker's ?
Fail2ban From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Thursday, June 16, 2011 11:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to secure our Asterisk server from hacker's ? Hi List, Yes you are right but I want to cross check to outside world to. How they will support me in such case... :) On Thu, Jun 16, 2011 at 11:23 AM, Alex Balashov abalas...@evaristesys.com wrote: I thought the idea was that Asterisk Engineers already know the answers to such questions? On 06/16/2011 01:52 AM, virendra bhati wrote: Hi List, I want to secure my server from the hacker's. What is the case by which I can protest it. I have done security of Dialplan, Sip,IAX base security. For linux we are working on Iptables. What else is left so that I will do it too... -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] #include filename
On Thu, Jun 16, 2011 at 12:24:15PM +0530, mahesh katta wrote: Hi, I am using asterisk1.2 In this, my dialplan is going large , so i need to configure this small pieces for this, i did in my extensions.conf when I dial the 123 its not going , means that file is not reading. is there any parameters to add any where ? please tell me this #include is not working ... extensions.conf [general] [global] trunk=zap/g0 #include exten-internal.conf #include means that the content of the file is added verbatim here. Which means that you added those 'exten' definitions in the [global] section. But that is a special section from which definitions of extensions are not read. Thus the content of the f ile is practically ignored (or generate a parse error. Not really sure). [default] Place the '#include' line at this line or somewhere below. exten = _X.,1,Answer exten = _X.,2,Dial(Zap/g0/999898999,,to) exten = _X.,3,Hangup /etc/asterisk/exten-internal.conf exten = 123,1,Answer exten = 123,2,Dial(SIP/5024,,t) exten = 123,2,Hangup That said: Asterisk 1.2 is an obsolete version. I honestly hope you don't use it for any new installation. If you have a a new installation, please use 1.8 . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inbound call not dialing exten
On Thu, Jun 16, 2011 at 12:48:39PM +0530, mahesh katta wrote:
Hi all,
I have 100 DID's which is 4578900 to 4578999 , and i have 5001 to 5099
extensions. when incomming call come to this DID no. (4578901) that time
5001 extestinsion should ring.
below my dial plan is not getting any result , inthat has any mistakes.
please help.
exten = _45789XX,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _45789XX,1,Set(Dest=2{EXTEN:-2})
exten =
_45789XX,2,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten = _45789XX,3,Dial(SIP/${Dest},,tTo)
exten = _45789XX,4,Hangup
Best Regards,
There is one obvious problem. But you missed an obvious debugging step
to debug it:
In the asterisk CLI:
core set verbose 3
Now, look at the trace. What do you see at the time of the call?
Compare that to the dialplan you wrote.
And once again: do not use Asterisk 1.2 on a new installation.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Re: [asterisk-users] DIGIUM PRI CARDS REQUIRE
Hi, I hope this is not rude of my part. I normally avoid answering mails that relate in such way commercially to hardware. This list is non-commercial If you want to ask questions of commercial nature, please use Asterisk-biz: http://lists.digium.com/mailman/listinfo/asterisk-biz Please follow up on this thread in privat email and not on-list. (For the record: this mail was sent after on-list Mahesh Katta's reply). Regards, Oh, and: you should avoid using Asterisk 1.2 on a new installation. Please use Asterisk 1.8 ;-) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to secure our Asterisk server from hacker's ?
On Thu, Jun 16, 2011 at 12:26:13PM +0500, Faisal Hanif wrote: Fail2ban Fail2Ban protects you from one (or two) specific types of attacks: someone trying to establish many connections to your PBX (e.g.: SIP register, or SIP invite) in order to try to guess a username/password combination the hard way (brute force). Or maybe just to consume your resources. But this is far from the only tthing to do to secure the PBX. I suggest that you do some research, and come back to us with more specific questions. For starters: http://www.catb.org/~esr/faqs/smart-questions.html Oh, and: did I mention you should not use Asterisk 1.2 for a new install, but rather use 1.8? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Web based call back
On Thu, Jun 16, 2011 at 11:26:21AM +0800, asterisk asterisk wrote:
Hi,
I am looking for a simple solution to do this.
I wish to have the user to enter their preferred method of connection i.e.
for the cheapest solution to their desktop phone or mobile phone, then plan
callfile based on the number that user provided and dial to the user.
Any suggestions?
But doing so *is* simple. See a simple example attached. It relies on an
assumption that the origination IP address authenticates a user and also
the user's location (specifically: the phone).
You would probably need your own schema for that. But the actual dialing
is very simple.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
#!/usr/bin/perl -w -T
# A web-based dialer for Asterisk.
# Authenticates and sets originating channel by IP address.
# Written by Tzafrir Cohen tzafrir.co...@xorcom.com
# Copyright (C) 2011, Xorcom
# This program is free software; you can redistribute and/or
# modify it under the same terms as Perl itself.
# Required manager.conf settings:
# [dialer-web]
# secret=top-secret
# deny=0.0.0.0/0.0.0.0
# permit=127.0.0.1/255.255.255.0
# read=
# write=call
### Apache rewrite rules:
# RewriteEngine on
# RewriteRule ^/dial$ /cgi-bin/dial.cgi [R]
# RewriteRule ^/dial/(.*) /cgi-bin/dial.cgi?dial_number=$1 [R]
use strict;
use CGI qw/:standard/;
use Asterisk::Manager;
$ENV{PATH} = /bin:/usr/bin;
my $Style = EOT;
!--
body { background-color: #eee; }
#main { background-color: white; }
h1 { color: blue; text-align: center }
.error {
background-color: red;
}
--
EOT
my %CallerMap = (
'172.16.5.128' = 'DAHDI/33',
'172.16.5.181' = 'DAHDI/42',
'172.16.5.122' = 'SIP/279',
'172.16.5.139' = 'DAHDI/38',
'172.16.5.121' = 'SIP/235',
'172.16.5.39' = 'SIP/235',
);
my $Title = Asterisk Dialer;
my $admin = Tzafrir;
my @Phones = sort values(%CallerMap);
my $device;
if (exists $CallerMap{remote_host()}){
$device = $CallerMap{remote_host()};
}
sub dial($$) {
my $device = shift;
my $number = shift;
my $astman = new Asterisk::Manager;
$astman-user('dialer-web');
$astman-secret('top-secret');
$astman-host('localhost');
$astman-connect ||
die Could not connect to Asterisk manager interface: $!;
my %response = $astman-sendcommand(
Action = 'Originate',
Channel = $device,
Context = 'from-internal',
Exten = $number,
Priority = 1,
);
if ($response{Response} eq 'Success') {
return 1;
}
# Print response for debugging
print p({-class='error'}, Originate from $device to $number failed);
for (keys %response) {
print p([$_ = $response{$_}]);
}
return 0;
}
sub footer() {
my $base_url = url(-base=1)./dial;
print
hr,
p(The Xorcom Dialer at , a({-href=$base_url},$base_url),
. Also try ,
a({-href=$base_url/*65},$base_url./.em(number)),
),
end_html;
}
print
header,
start_html(-title=$Title, -style={-code=$Style}),
h1($Title),
;
my $number = param('dial_number');
if (defined $number) {
# Basic sanitation: Do allow '#', '*', numbers and digits.
# Nothing else.
$number =~ s/[^[:alnum:]\*#]//g;
my $rc = dial($device, $number);
if ($rc) {
print p(Dialed from [$device] to [$number]);
} else {
print p({-classerror},
Error dialing from [$device] to [$number]);
}
}
if(! defined $device) {
my $host = remote_host();
print
p(You are connected from [$host], and no phone is ,
assigned to it. Please contact $admin.),
;
footer();
exit 0;
}
print
div({-id='main'},
p(Call from [$device] to:),
start_form,
p(textfield('dial_number'), submit(Dial)) ,
end_form,
),
;
footer();
--
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Re: [asterisk-users] Inbound call not dialing exten
On Thursday 16 Jun 2011, mahesh katta wrote:
Hi all,
I have 100 DID's which is 4578900 to 4578999 , and i have 5001 to 5099
extensions. when incomming call come to this DID no. (4578901) that time
5001 extestinsion should ring.
below my dial plan is not getting any result , inthat has any mistakes.
please help.
exten = _45789XX,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _45789XX,1,Set(Dest=2{EXTEN:-2})
exten =
_45789XX,2,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM
}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0)) exten =
_45789XX,3,Dial(SIP/${Dest},,tTo)
exten = _45789XX,4,Hangup
Firstly, you've got two 1 steps in that. Unless you are doing something
complicated with GOTOs (and if you are, then there's probably a better way
of doing it), use 1 for the first step and n (next) for all subsequent
steps.
If nothing else, it means you can add extra NoOp() statements to put debugging
information on the console, and later comment out or remove them without
forced renumbering (which brings its own opportunities to introduce errors).
Secondly, you're setting ${Dest} to 2 followed by the last 2 digits of the
dialled number. But what you really want is 50 followed by the last 2
digits of the dialled number. So it should be
exten = _45789XX,n,Set(Dest=50{EXTEN:-2})
--
AJS
Answers come *after* questions.
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[asterisk-users] Channel variables not available during xfer?
Hi all. I've got a Grandstream GXP2xxx that is working fine in gerneral. However, when I try to transfer a call, my dialplan doesn't have access to any of the channel's variables. It seems that this is a known issue in 1.6.x. Has it been fixed in 1.8.x? Is 1.8 stable enough for production? Should I move to 1.8, yet? So far, the only work-around I've come up with is a separate context for EACH sip account, with the variables hard-wired FUGLY! -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sending SMS on Friday at 12 Noon EDT
Hi, This Friday Chris Matthieu of SMSified.com will explain how to send SMS from your apps. As usual, there will be talk about Asterisk, questions, answers, and comments about telephony, networks, VoIP and even some OT. All are welcome to join the weekly average of 35-60 callers live. If you can't, join see http://vuc.me for the recorded versions. If you are working on something that might be of interest to the VoIP USers Conference, please get in touch and we can book you as a guest. Friday at conference time (http://vuc.me/next for the time in your zone) please join us via mp3 stream, Gtalk, PSTN, SIP, Skype or web widget. SIP:200...@login.zipdx.com (g722, g711) Skype:vuc.me Gtalk:voipusersconfere...@gmail.com (use call computer) The widgets and mp3 URL will appear during the call on http://vuc.me Hear you there, :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inbound call not dialing exten
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
On Thu, Jun 16, 2011 at 1:40 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
On Thursday 16 Jun 2011, mahesh katta wrote:
Hi all,
I have 100 DID's which is 4578900 to 4578999 , and i have 5001 to 5099
extensions. when incomming call come to this DID no. (4578901) that time
5001 extestinsion should ring.
below my dial plan is not getting any result , inthat has any mistakes.
please help.
exten = _45789XX,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _45789XX,1,Set(Dest=2{EXTEN:-2})
exten =
_45789XX,2,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM
}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0)) exten =
_45789XX,3,Dial(SIP/${Dest},,tTo)
exten = _45789XX,4,Hangup
Firstly, you've got two 1 steps in that. Unless you are doing something
complicated with GOTOs (and if you are, then there's probably a better way
of doing it), use 1 for the first step and n (next) for all
subsequent
steps.
If nothing else, it means you can add extra NoOp() statements to put
debugging
information on the console, and later comment out or remove them without
forced renumbering (which brings its own opportunities to introduce
errors).
Secondly, you're setting ${Dest} to 2 followed by the last 2 digits of
the
dialled number. But what you really want is 50 followed by the last 2
digits of the dialled number. So it should be
exten = _45789XX,n,Set(Dest=50{EXTEN:-2})
sir when I add this its not ringing on sip exten
log will come like
Accepting call from '0559566768' to '4578924' on channel 0/3, span
1
-- Executing AGI(Zap/3-1, agi://127.0.0.1:4577/call_log) in new
stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning
0
-- Executing Set(Zap/3-1, Dest=50{EXTEN:-2}) in new
stack
-- Executing MixMonitor(Zap/3-1,
/var/spool/asterisk/astrec/20110616-122003-0559566768-4578924-1308212403.2225.gsm|av(0)V(0))
in new stack
-- Executing Dial(Zap/3-1, SIP/50{EXTEN:-2}||tTo) in new
stack
== Begin MixMonitor Recording
Zap/3-1
== Parsing '/etc/asterisk/manager.conf':
Found
== Manager 'sendcron' logged on from
127.0.0.1
== Manager 'sendcron' logged off from
127.0.0.1
Jun 16 12:20:07 WARNING[7409]: chan_sip.c:2018 create_addr: No such host:
50{EXTEN
Jun 16 12:20:07 NOTICE[7409]: app_dial.c:1076 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to
destination)
== Everyone is busy/congested at this time
(1:0/0/1)
-- Executing Hangup(Zap/3-1, ) in new
stack
== Spawn extension (default, 4578924, 5) exited non-zero on
'Zap/3-1'
-- Executing DeadAGI(Zap/3-1, agi://
127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-3-CHANUNAVAIL--)
in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] Inbound call not dialing exten
On Thursday 16 Jun 2011, mahesh katta wrote:
-- Executing Set(Zap/3-1, Dest=50{EXTEN:-2}) in new
stack
-- Executing MixMonitor(Zap/3-1,
/var/spool/asterisk/astrec/20110616-122003-0559566768-4578924-1308212403.2
225.gsm|av(0)V(0)) in new stack
-- Executing Dial(Zap/3-1, SIP/50{EXTEN:-2}||tTo) in new
stack
Oops, my bad; I missed out a punctuation mark.
Before I tell you the answer, though, have a good look at the console
diagnostic messages and see if you can spot for yourself what it's doing
wrong. What is it trying to dial, and what *should* it be trying to dial?
And didn't it feel good, knowing you fixed it yourself?
-- SPOILER FOLLOWS --
There should be a $ sign in the Set() step:
Set(Dest=50${EXTEN:-2})
so it will use the rightmost 2 characters of ${EXTEN}.
--
AJS
Answers come *after* questions.
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[asterisk-users] sipp application/dtmf-relay not work properly in Asterisk!
hi, everyone i want to use sipp to auto answer the ivr， to simulate the keypad send digital sequence， so i try to send DTMF by application/dtmf-relay, but i have got this error message in the asterisk CLI, Could you help me? Thanks! [Jun 16 17:11:34] WARNING[26321]: rtp.c:3207 ast_rtp_senddigit_begin: Don't know how to represent ' the whole CLI message as follows: --- SIP read from UDP:211.150.88.154:5067 --- INFO sip:01025475845@211.150.88.155:5060 SIP/2.0 Via: SIP/2.0/UDP 211.150.88.154:5067;branch=z9hG4bK-3-1-7;rport From: 1000 sip:1000@211.150.88.154:5067;tag=1 To: 01025475845 sip:01025475845@211.150.88.155:5060 Call-Id: 1-3@211.150.88.154 CSeq: 2 INFO Contact: sip:1000@211.150.88.154:5067 Event: dtmf Content-Type: application/dtmf-relay Content-Length:31 Signal= 11037845 Duration= 100 - --- (10 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: --- Transmitting (NAT) to 211.150.88.154:5067 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 211.150.88.154:5067 ;branch=z9hG4bK-3-1-7;received=211.150.88.154;rport=5067 From: 1000 sip:1000@211.150.88.154:5067;tag=1 To: 01025475845 sip:01025475845@211.150.88.155:5060;tag=as7af6d579 Call-ID: 1-3@211.150.88.154 CSeq: 2 INFO Server: Asterisk PBX 1.6.2.13 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 [Jun 16 17:11:34] WARNING[26321]: rtp.c:3207 ast_rtp_senddigit_begin: Don't know how to represent ' -- Appreciate your kindly advise and help. Thanks Regards Sucan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS on Friday at 12 Noon EDT
Hi, If I am right then will you discuss about the sending sms with asterisk into that conference ? On Thu, Jun 16, 2011 at 1:41 PM, randulo rand...@randulo.com wrote: Hi, This Friday Chris Matthieu of SMSified.com will explain how to send SMS from your apps. As usual, there will be talk about Asterisk, questions, answers, and comments about telephony, networks, VoIP and even some OT. All are welcome to join the weekly average of 35-60 callers live. If you can't, join see http://vuc.me for the recorded versions. If you are working on something that might be of interest to the VoIP USers Conference, please get in touch and we can book you as a guest. Friday at conference time (http://vuc.me/next for the time in your zone) please join us via mp3 stream, Gtalk, PSTN, SIP, Skype or web widget. SIP:200...@login.zipdx.com (g722, g711) Skype:vuc.me Gtalk:voipusersconfere...@gmail.com (use call computer) The widgets and mp3 URL will appear during the call on http://vuc.me Hear you there, :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco IP Phones 7942G (skinny): TFTP and required files
Dears; I am sure that you have experience with Cisco IP Phones. I need to be sure if someone used Cisco 7942G in skinny firmware with Asterisk 1.8 and how it was (if fine or it has a problem). Are the below the only 3 needed files to be placed in the tftpboot directory: CTLSEPB8BEBF22AB62.tlv (which is empty file, just we place it with its name). SEPB8BEBF22AB62.cnf.xml XMLDefault.cnf.xml So, do I have to add any other file? One more thing: in the above mentioned files, do I have to determine the firmware that the Phone should take it and I have to place this firmware in the tftpboot directory? Note: I am using tftp-server (as my OS if fedora). Is there any permission need to be given for the files in the /var/lib/tftpboot/? Or no need as the phones are going to download them and not upload new files? Looking forward for a help PLZ. Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones 7942G (skinny): TFTP and required files
I've no experience with that phone model or protocol. But if you run a tftp trace you'll see what files the phone is looking for. Check my old thread on pbxinaflash forums for details. i -- Original Message -- Received: 04:59 AM COT, 06/16/2011 From: bilal ghayyad bilmar...@yahoo.com To: ianworthing...@usa.net, rswago...@gmail.com, s...@open-t.co.uk, cass...@cassius.org, wcse...@selbytech.com, asterisk-users@lists.digium.com Subject: Cisco IP Phones 7942G (skinny): TFTP and required files Dears; I am sure that you have experience with Cisco IP Phones. I need to be sure if someone used Cisco 7942G in skinny firmware with Asterisk 1.8 and how it was (if fine or it has a problem). Are the below the only 3 needed files to be placed in the tftpboot directory: CTLSEPB8BEBF22AB62.tlv (which is empty file, just we place it with its name). SEPB8BEBF22AB62.cnf.xml XMLDefault.cnf.xml So, do I have to add any other file? One more thing: in the above mentioned files, do I have to determine the firmware that the Phone should take it and I have to place this firmware in the tftpboot directory? Note: I am using tftp-server (as my OS if fedora). Is there any permission need to be given for the files in the /var/lib/tftpboot/? Or no need as the phones are going to download them and not upload new files? Looking forward for a help PLZ. Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS on Friday at 12 Noon EDT
On Thu, Jun 16, 2011 at 11:52 AM, virendra bhati virbh...@gmail.com wrote: If I am right then will you discuss about the sending sms with asterisk into that conference ? We can if someone wants to, that's how the VUC works. :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Have your suggestions on Hardware configuration for Asterisk.
Hi all, I will really appreciate if you can spend some time to share your experience or point me in right direction. I have been told to prepare a single box Asterisk system (No Distributed architecture) for following features. -Asterisk 1.8 -300 SIP extensions (sip.conf) -8 port PRI card (E1) with h/w based echo cancellation -All calls to be recorded -IVR -ACD (Real time Queue) -240 concurrent calls in the worst case (DAHDI to SIP and/or SIP 2 DAHDI). -No transcoding -Remote Database -Recordings and logs are to be cleaned up/moved every other week. I have gone through some documents for Asterisk dimensioning ( http://www.voip-info.org/wiki/view/Asterisk+dimensioning , http://www.voip-info.org/wiki/view/Asterisk+hardware+recommendations). Seeing the factsheet I plan to go for Quad core, 3+Ghz, 8GB RAM. I don't want you people to do a homework for me, but it would be a great help if you can have some suggestions based on your past experience before I procure the System and test it. Suggestions on specific brands for server are also welcome. I am very well aware of the consequences of 'single point of failure', but have no choice for the moment. TYIA, [SATISH] -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MixMonitor
hello list,
i have asterisk 1.4 with IAX and sip i have configured the MixMonitor in
order to record the conversation
i can record all the calls inbound and outbound without problem.
but when i receive an inbound call from customer in IAX(1000) and i want to
transfer the call to other phone SIP(223)
the conversation between customer and IAX is recorded but the conversation
between customer and sip is not recorded
extensions.conf
exten = 223,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,Dial(SIP/${EXTEN},,KkTt)
exten = 223,n,Hangup();
any help please
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Re: [asterisk-users] change destination on digit
Check the option of 'd' in Dial().
d: Allow the calling user to dial a 1 digit extension while waiting for a
call to be answered. Exit to that extension if it exists in the current
context, or the context defined in the ${EXITCONTEXT} variable,if it exists.
[SATISH]
On Wed, Jun 15, 2011 at 7:03 PM, vip killa vipki...@gmail.com wrote:
Is there an easy way to setup diaplan so when someone pushes a digit such
as * during a call, they will be transferred to another destination.
For example, a caller is hearing ringing while calling a UA, but instead of
waiting for the UA to pick up, they can push * and go directly to that UA's
voicemail.
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Re: [asterisk-users] Google Voice receiving call problem
Hey Elliot;
Would you mind posting your dialplan for your Google Voice config? I am
having a hell of a time getting it to do *anything*.
Perhaps I am just fat-fingering.
Would you mind? Thanks in advance.
Glen
On 6/13/2011 19:02, Elliot Murdock wrote:
Hello,
I am using 1.8.4.2 and while outgoing seems to work, incoming still
does not route calls in to the appropriate context.
Please advise.
Thank you,
Elliot
On Sat, Apr 16, 2011 at 4:24 PM, William Stillwell
will...@stillwellsoft.com wrote:
You must have 1.8+ its already been posted the 1.6 didn’t get a backport fix
in the jabber protocol.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
Dardini
Sent: Saturday, April 16, 2011 3:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Google Voice receiving call problem
Hello,
I have a Google Voice phone number and want to connect it to my asterisk box
to have calls handled to my SIP account.
When I call the number I receive the correct INCOMING request on Jabber
portion of asterisk, but the call is not connected to the gtalk part.
JABBER: asterisk INCOMING:iq
from=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
to=ldard...@gmail.com/asterisk438D86E0
id=jingle:10.176.108.16-15899749:1:457BCF36 type=setses:session
type=initiate id=SIP784359174@10.177.37.1
initiator=+17174695...@voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4
xmlns:ses=http://www.google.com/session;pho:description
xmlns:pho=http://www.google.com/session/phone;pho:payload-type id=0
name=PCMU clockrate=8000/pho:payload-type id=101
name=telephone-event//pho:descriptiontransport
behind-symmetric-nat=false can-receive-from-symmetric-nat=false
xmlns=http://www.google.com/transport/raw-udp/transport
xmlns=http://www.google.com/transport/p2p//ses:session/iq
No other messages are logged. Where is my mistake?
I am using asterisk 1.6.2.7 (Ubuntu packaging) and following there are the
relevant files.
Thank you
Leandro
### jabber.conf
[general]
autoregister=yes
[asterisk]
type=client
serverhost=talk.google.com
username=ldard...@gmail.com
secret=**
priority=1
port=5222
usetls=yes
usesasl=yes
buddy=ldard...@gmail.com
status=available
### gtalk.conf
[general]
context=default
bindaddr=0.0.0.0
allowguest=yes
[guest]
disallow=all
allow=ulaw
context=google-in
[ldardini]
username=ldard...@gmail.com
disallow=all
allow=ulaw
context=google-in
connection=asterisk
extension.ael
context google-in {
s = {
NoOp( Call from Gtalk );
Dial(SIP/@,60,r);
};
}
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Re: [asterisk-users] MixMonitor
On 16/06/11 07:36 AM, salaheddine elharit wrote: hello list, i have asterisk 1.4 with IAX and sip i have configured the MixMonitor in order to record the conversation but when i receive an inbound call from customer in IAX(1000) and i want to transfer the call to other phone SIP(223) the conversation between customer and IAX is recorded but the conversation between customer and sip is not recorded Is the call coming from IAX(1000) or going to IAX(1000)? Note that when you transfer calls around and are using MixMonitor() (or any recording) that you have to think of the recording as being associated with the incoming channel, and the recording should essentially follow it around. So if you have a call coming in like this: ITSP -- Asterisk -- Dialplan -- Mixmonitor -- Dial(SIP/1000) Then the MixMonitor() is associated with the channel created when the call came in from the ITSP. If that channel is then transferred, the recording should follow it around. Can you elaborate a bit more on the call flow and show the console output? -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor
thanks for your response
the call is going to IAX(1000), i have i DID Number when the customer call
this number 0520XX the call is goint to agent
IAX. in my dialplan i have
exten = 223,1,MixMonitor(blah.wav)
exten = 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,Dial(SIP/223)
and in extensions.conf i have
exten = 223,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,Dial(SIP/${EXTEN},,KkTt)
exten = 223,n,Hangup();
thanks and regards
2011/6/16 Leif Madsen leif.mad...@asteriskdocs.org
On 16/06/11 07:36 AM, salaheddine elharit wrote:
hello list,
i have asterisk 1.4 with IAX and sip i have configured the MixMonitor in
order to record the conversation
but when i receive an inbound call from customer in IAX(1000) and i want
to transfer the call to other phone SIP(223)
the conversation between customer and IAX is recorded but the
conversation between customer and sip is not recorded
Is the call coming from IAX(1000) or going to IAX(1000)? Note that when you
transfer calls around and are using MixMonitor() (or any recording) that you
have to think of the recording as being associated with the incoming
channel, and the recording should essentially follow it around.
So if you have a call coming in like this:
ITSP -- Asterisk -- Dialplan -- Mixmonitor -- Dial(SIP/1000)
Then the MixMonitor() is associated with the channel created when the call
came in from the ITSP. If that channel is then transferred, the recording
should follow it around.
Can you elaborate a bit more on the call flow and show the console output?
--
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http://www.oreilly.com/catalog/asterisk
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[asterisk-users] fast AGI memory leaks
Could someone point me in the right direction of how to create a Fast AGI script without memory leaks? I was told i need to clear the result set for mysql queries, Im not sure how to do that. My script is a simple perl script of 70 lines doing database lookups and executing dial and voicemail I'd be happy to post the code, thanks in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIGIUM PRI CARDS REQUIRE
Tzafrir, Whats up with this 1.2 vs 1.8 signature? On Thu, Jun 16, 2011 at 3:38 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: Hi, I hope this is not rude of my part. I normally avoid answering mails that relate in such way commercially to hardware. This list is non-commercial If you want to ask questions of commercial nature, please use Asterisk-biz: http://lists.digium.com/mailman/listinfo/asterisk-biz Please follow up on this thread in privat email and not on-list. (For the record: this mail was sent after on-list Mahesh Katta's reply). Regards, Oh, and: you should avoid using Asterisk 1.2 on a new installation. Please use Asterisk 1.8 ;-) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor
On 16/06/11 09:20 AM, salaheddine elharit wrote:
thanks for your response
the call is going to IAX(1000), i have i DID Number when the customer
call this number 0520XX the call is goint to agent
IAX. in my dialplan i have
exten = 223,1,MixMonitor(blah.wav)
exten = 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,Dial(SIP/223)
and in extensions.conf i have
exten = 223,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,Dial(SIP/${EXTEN},,KkTt)
exten = 223,n,Hangup();
OK, well nothing looks obviously wrong there from what I can tell.
What is your console output doing though when you do the transfer? Are
you using Asterisk transfers? What version of Asterisk are you using?
Leif.
--
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Re: [asterisk-users] PAP2T provisioning via SRV record?
On 06/16/2011 07:58 AM, Mike Diehl wrote: Well, I ran a simple test by trying to configure the second port to use the DNS SRV record, as described below. Here is what I have: (sanitized) == Proxy_2_ diehlnet.com/Proxy_2_ Outbound_Proxy_2_ fqdn/Outbound_Proxy_2_ Display_Name_2_ ua=nausername/Display_Name_2_ User_ID_2_ ua=nausername/User_ID_2_ Password_2_ ua=napassword/Password_2_ Use_Auth_ID_2_ ua=naYes/Use_Auth_ID_2_ Auth_ID_2_ ua=nausername/Auth_ID_2_ Use_DNS_SRV_2_yes/Use_DNS_SRV_2_ == With this configuration, the second port does NOT register. A sniffer trace on the inside interface of my router gives me some clues, though: 23:54:34.906089 IP 10.0.1.87.60198 208.67.222.222.53: 1+ A? diehlnet.com. (30) 23:54:35.102409 IP 208.67.222.222.53 10.0.1.87.60198: 1 1/0/0 A 173.10.242.193 (46) 23:54:35.104484 IP 10.0.1.87.5061 173.10.242.193.5060: UDP, length: 527 23:54:35.104553 IP 173.10.242.193 10.0.1.87: icmp 556: 173.10.242.193 udp port 5060 unreachable It seems that the device is still looking for an A record for diehlnet.com, which does exist. It should be looking for the SRV record. What am I missing? Make sure you also have set: DNS_SRV_Auto_Prefix_2_ ua=naYes/DNS_SRV_Auto_Prefix_2_ From the manual: Enables the phone to automatically prepend the proxy or outbound proxy name with _sip._udp when performing a DNS SRV lookup on that name. Defaults to no. Best regards, Marius -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor
i have asterisk 1.4 and also i have aheeva applicaton also installed in my
server
in the consol this call may be monitored or recorded
best regrads
2011/6/16 Leif Madsen leif.mad...@asteriskdocs.org
On 16/06/11 09:20 AM, salaheddine elharit wrote:
thanks for your response
the call is going to IAX(1000), i have i DID Number when the customer
call this number 0520XX the call is goint to agent
IAX. in my dialplan i have
exten = 223,1,MixMonitor(blah.wav)
exten = 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,Dial(SIP/223)
and in extensions.conf i have
exten = 223,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,Dial(SIP/${EXTEN},,KkTt)
exten = 223,n,Hangup();
OK, well nothing looks obviously wrong there from what I can tell.
What is your console output doing though when you do the transfer? Are you
using Asterisk transfers? What version of Asterisk are you using?
Leif.
--
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http://www.oreilly.com/catalog/asterisk
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Re: [asterisk-users] MixMonitor
Try switching the Set and MixMonitor commands so the AUDIOHOOK_INHERIT will
be in effect when Mixmonitor starts
exten = 223,1,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,MixMonitor(blah.wav)
exten = 223,n,Dial(SIP/223)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Thursday, June 16, 2011 9:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MixMonitor
i have asterisk 1.4 and also i have aheeva applicaton also installed in my
server
in the consol this call may be monitored or recorded
best regrads
2011/6/16 Leif Madsen leif.mad...@asteriskdocs.org
On 16/06/11 09:20 AM, salaheddine elharit wrote:
thanks for your response
the call is going to IAX(1000), i have i DID Number when the customer
call this number 0520XX the call is goint to agent
IAX. in my dialplan i have
exten = 223,1,MixMonitor(blah.wav)
exten = 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,Dial(SIP/223)
and in extensions.conf i have
exten = 223,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,Dial(SIP/${EXTEN},,KkTt)
exten = 223,n,Hangup();
OK, well nothing looks obviously wrong there from what I can tell.
What is your console output doing though when you do the transfer? Are you
using Asterisk transfers? What version of Asterisk are you using?
Leif.
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http://www.oreilly.com/catalog/asterisk
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Re: [asterisk-users] No audio after a reinvite changing codec
On 15/06/2011 8:15 PM, Matteo Campana wrote: HI list, no idea?? :) There not much substance in the information provided for an assessment to be made. I would suggest you capture the network traffic between UAC (g711) Asterisk UAS ensuring the snap length is large enough to capture the whole packet and do the same with traffic between Asterisk UAC Provider then use Wireshark and its telephony feature to analyse VoIP calls, check the flows, you may discover the problem this way! Larry. M. On Mon, Jun 13, 2011 at 6:55 PM, Matteo Campana matteo.camp...@gmail.com mailto:matteo.camp...@gmail.com wrote: Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UACASTERISK UAS | ASTERISK UAC PROVIDER g711 -- g711 | g729 --- g729 rtp rtp After a while, we have the reinvite sent by the SIP provider with g711 in the SDP. So asterisk need to change audio codec from g729 to g711 and correctly we see on debug the following line: Oooh, we need to change our audio formats since our peer supports only g729 and asterisk send back 200 OK to the provider. At this point we have one way rtp audio: UACASTERISK UAS | ASTERISK UAC PROVIDER g711 -- g711 | g711 --- g711 rtp rtp So the problem is that UAC does not hear audio at all. Any idea? (Asterisk version: 1.4.33.1). Thanks in advance, Matteo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No audio after a reinvite changing codec
We experience the same thing. The solution we use is to not change codecs in the middle of a call. I assumed it was an issue with our upstream. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Larry Moore Sent: Thursday, June 16, 2011 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No audio after a reinvite changing codec On 15/06/2011 8:15 PM, Matteo Campana wrote: HI list, no idea?? :) There not much substance in the information provided for an assessment to be made. I would suggest you capture the network traffic between UAC (g711) Asterisk UAS ensuring the snap length is large enough to capture the whole packet and do the same with traffic between Asterisk UAC Provider then use Wireshark and its telephony feature to analyse VoIP calls, check the flows, you may discover the problem this way! Larry. M. On Mon, Jun 13, 2011 at 6:55 PM, Matteo Campana matteo.camp...@gmail.com wrote: Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 -- g711 | g729 --- g729 rtp rtp After a while, we have the reinvite sent by the SIP provider with g711 in the SDP. So asterisk need to change audio codec from g729 to g711 and correctly we see on debug the following line: Oooh, we need to change our audio formats since our peer supports only g729 and asterisk send back 200 OK to the provider. At this point we have one way rtp audio: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 -- g711 | g711 --- g711 rtp rtp So the problem is that UAC does not hear audio at all. Any idea? (Asterisk version: 1.4.33.1). Thanks in advance, Matteo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAP2T provisioning via SRV record?
On 6/14/2011 5:08 AM, Paul Hayes wrote: On 13/06/11 19:44, Mike Diehl wrote: Hi all, I'm trying to provision my PAP2T's to use a SVR lookup to find the Asterisk server. I'm using a provisioning file that contains an element like: Proxy_1_ _sip._udp.example.com/Proxy_1_ However, the PAP doesn't seem to be able to find my server with this hostname. The DNS records are in place because my Polycom and Grandstream servers work just fine. What else do I need to do to get the PAP to work this way? TIA, There's a setting in the Line 1 and Line 2 page called Use DNS SRV which is set to No by default for some reason. Set this to yes and set the proxy to example.com. So something like: Use_DNS_SRV_1_yes/Use_DNS_SRV_1_ Proxy_1_example.com/Proxy_1_ In addition to this, you also need to set the DNS_SRV_Auto_Prefix to 'yes'. DNS_SRV_Auto_Prefix_1_yes/DNS_SRV_Auto_Prefix_1_ cheers, Paul. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Technical Support http://www.neuroredes.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAP2T provisioning via SRV record?
On Thursday 16 June 2011 8:10:38 am Marius Pedersen wrote: On 06/16/2011 07:58 AM, Mike Diehl wrote: Well, I ran a simple test by trying to configure the second port to use the DNS SRV record, as described below. Here is what I have: (sanitized) == Proxy_2_ diehlnet.com/Proxy_2_ Outbound_Proxy_2_ fqdn/Outbound_Proxy_2_ Display_Name_2_ ua=nausername/Display_Name_2_ User_ID_2_ ua=nausername/User_ID_2_ Password_2_ ua=napassword/Password_2_ Use_Auth_ID_2_ ua=naYes/Use_Auth_ID_2_ Auth_ID_2_ ua=nausername/Auth_ID_2_ Use_DNS_SRV_2_yes/Use_DNS_SRV_2_ == With this configuration, the second port does NOT register. A sniffer trace on the inside interface of my router gives me some clues, though: 23:54:34.906089 IP 10.0.1.87.60198 208.67.222.222.53: 1+ A? diehlnet.com. (30) 23:54:35.102409 IP 208.67.222.222.53 10.0.1.87.60198: 1 1/0/0 A 173.10.242.193 (46) 23:54:35.104484 IP 10.0.1.87.5061 173.10.242.193.5060: UDP, length: 527 23:54:35.104553 IP 173.10.242.193 10.0.1.87: icmp 556: 173.10.242.193 udp port 5060 unreachable It seems that the device is still looking for an A record for diehlnet.com, which does exist. It should be looking for the SRV record. What am I missing? Make sure you also have set: DNS_SRV_Auto_Prefix_2_ ua=naYes/DNS_SRV_Auto_Prefix_2_ From the manual: Enables the phone to automatically prepend the proxy or outbound proxy name with _sip._udp when performing a DNS SRV lookup on that name. Defaults to no. That was the final change that made it work. Wish I had the manual you have. Mine didn't say much. Thank you. -- Take care and have fun, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAP2T provisioning via SRV record?
On 06/16/2011 04:49 PM, Mike Diehl wrote: On Thursday 16 June 2011 8:10:38 am Marius Pedersen wrote: On 06/16/2011 07:58 AM, Mike Diehl wrote: Well, I ran a simple test by trying to configure the second port to use the DNS SRV record, as described below. Here is what I have: (sanitized) == Proxy_2_ diehlnet.com/Proxy_2_ Outbound_Proxy_2_ fqdn/Outbound_Proxy_2_ Display_Name_2_ ua=nausername/Display_Name_2_ User_ID_2_ ua=nausername/User_ID_2_ Password_2_ ua=napassword/Password_2_ Use_Auth_ID_2_ ua=naYes/Use_Auth_ID_2_ Auth_ID_2_ ua=nausername/Auth_ID_2_ Use_DNS_SRV_2_yes/Use_DNS_SRV_2_ == With this configuration, the second port does NOT register. A sniffer trace on the inside interface of my router gives me some clues, though: 23:54:34.906089 IP 10.0.1.87.60198 208.67.222.222.53: 1+ A? diehlnet.com. (30) 23:54:35.102409 IP 208.67.222.222.53 10.0.1.87.60198: 1 1/0/0 A 173.10.242.193 (46) 23:54:35.104484 IP 10.0.1.87.5061 173.10.242.193.5060: UDP, length: 527 23:54:35.104553 IP 173.10.242.193 10.0.1.87: icmp 556: 173.10.242.193 udp port 5060 unreachable It seems that the device is still looking for an A record for diehlnet.com, which does exist. It should be looking for the SRV record. What am I missing? Make sure you also have set: DNS_SRV_Auto_Prefix_2_ ua=naYes/DNS_SRV_Auto_Prefix_2_ From the manual: Enables the phone to automatically prepend the proxy or outbound proxy name with _sip._udp when performing a DNS SRV lookup on that name. Defaults to no. That was the final change that made it work. Wish I had the manual you have. Mine didn't say much. If you take a look at the Cisco Small Business SPA300 Series, SPA500 Series, and WIP310 IP Phone Administration Guide you can find information on most of the settings there as the PAP2T and SPA-series configuration options are quite similar. Best regards, Marius -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor
hi Danny
thank you for your response i switched the MixMonitor and i still have the
same result
any help please
2011/6/16 Danny Nicholas da...@debsinc.com
Try switching the Set and MixMonitor commands so the AUDIOHOOK_INHERIT will
be in effect when Mixmonitor starts
exten = 223,1,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,MixMonitor(blah.wav)
exten = 223,n,Dial(SIP/223)
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
elharit
*Sent:* Thursday, June 16, 2011 9:13 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] MixMonitor
i have asterisk 1.4 and also i have aheeva applicaton also installed in my
server
in the consol this call may be monitored or recorded
best regrads
2011/6/16 Leif Madsen leif.mad...@asteriskdocs.org
On 16/06/11 09:20 AM, salaheddine elharit wrote:
thanks for your response
the call is going to IAX(1000), i have i DID Number when the customer
call this number 0520XX the call is goint to agent
IAX. in my dialplan i have
exten = 223,1,MixMonitor(blah.wav)
exten = 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,Dial(SIP/223)
and in extensions.conf i have
exten = 223,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,Dial(SIP/${EXTEN},,KkTt)
exten = 223,n,Hangup();
OK, well nothing looks obviously wrong there from what I can tell.
What is your console output doing though when you do the transfer? Are you
using Asterisk transfers? What version of Asterisk are you using?
Leif.
--
Leif Madsen
http://www.oreilly.com/catalog/asterisk
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Re: [asterisk-users] fast AGI memory leaks
On Thu, 16 Jun 2011, vip killa wrote: Could someone point me in the right direction of how to create a Fast AGI script without memory leaks? I was told i need to clear the result set for mysql queries, Im not sure how to do that. My script is a simple perl script of 70 lines doing database lookups and executing dial and voicemail I'd be happy to post the code, thanks in advance. Read the MySQL documentation? Post on the MySQL mailing list? Google? Failing finding any information on how to free the result, you could always just close the connection. That's pretty effective in a heavy handed sort of way. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones 7942G (skinny): TFTP and required files
Hello, I do not use the skinny firmware. By the way, questions like this are best shared with the asterisk-users group mailing list, so that a large segment of the Asterisk community can benefit from the questions and answers. Cassius Smith -- On 6/16/11 4:59 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; I am sure that you have experience with Cisco IP Phones. I need to be sure if someone used Cisco 7942G in skinny firmware with Asterisk 1.8 and how it was (if fine or it has a problem). Are the below the only 3 needed files to be placed in the tftpboot directory: CTLSEPB8BEBF22AB62.tlv (which is empty file, just we place it with its name). SEPB8BEBF22AB62.cnf.xml XMLDefault.cnf.xml So, do I have to add any other file? One more thing: in the above mentioned files, do I have to determine the firmware that the Phone should take it and I have to place this firmware in the tftpboot directory? Note: I am using tftp-server (as my OS if fedora). Is there any permission need to be given for the files in the /var/lib/tftpboot/? Or no need as the phones are going to download them and not upload new files? Looking forward for a help PLZ. Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor
Since AUDIOHOOK_INHERIT is a backport from 1.8, something may be amiss in
the 1.4 IAX rendition. I assume your install would not be friendly for a
1.8 upgrade?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Thursday, June 16, 2011 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MixMonitor
hi Danny
thank you for your response i switched the MixMonitor and i still have the
same result
any help please
2011/6/16 Danny Nicholas da...@debsinc.com
Try switching the Set and MixMonitor commands so the AUDIOHOOK_INHERIT will
be in effect when Mixmonitor starts
exten = 223,1,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,MixMonitor(blah.wav)
exten = 223,n,Dial(SIP/223)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Thursday, June 16, 2011 9:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MixMonitor
i have asterisk 1.4 and also i have aheeva applicaton also installed in my
server
in the consol this call may be monitored or recorded
best regrads
2011/6/16 Leif Madsen leif.mad...@asteriskdocs.org
On 16/06/11 09:20 AM, salaheddine elharit wrote:
thanks for your response
the call is going to IAX(1000), i have i DID Number when the customer
call this number 0520XX the call is goint to agent
IAX. in my dialplan i have
exten = 223,1,MixMonitor(blah.wav)
exten = 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,Dial(SIP/223)
and in extensions.conf i have
exten = 223,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,Dial(SIP/${EXTEN},,KkTt)
exten = 223,n,Hangup();
OK, well nothing looks obviously wrong there from what I can tell.
What is your console output doing though when you do the transfer? Are you
using Asterisk transfers? What version of Asterisk are you using?
Leif.
--
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http://www.oreilly.com/catalog/asterisk
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Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
The Asterisk version is 1.8.3.2 The Cisco IP Phone is 7942G and it is running now skinny. The used TFTP is tftp-server which is installed in fedora. I placed the following two files (but look like it was not taken from the TFTP, as nothing appeared in the messages), but I am able to to ping from the asterisk box to the vlan that the Phone is connected, so no problem in the reachability: SEPB8BEBF22AB62.cnf.xml xmlDefault.CNF.XML Are the files name correct? Or the Cisco IP Phone 7942G are not working fine with Asterisk or the tftp-server? Cisco has changed the file name format a few times, so you may want to copy xmlDefault.CNF.XML to XMLDefault.cnf.xml The more important steps is how have you configured the phone to locate the TFTP server? Are you using option 150 in DHCP, or manually setting the TFTP server address on the phone. Technically you do not need a TFTP server, since the Skinny phones will try to use the TFTP server address for registration, so you can just set that address to point to your asterisk server. A TFTP server is needed if you want custom ringtones or to manage software updates. For small setups or my home, I skipped setting up the TFTP server until I wanted to update the phone firmware. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] show channels does not show hold status
I have two calls (626 and 542) coming into the same phone (524). SIP/524-05b5!smvoice-sip!!1!Up!AppDial!(Outgoing Line)!_2XX!!3!9!SIP/542-05b4 SIP/542-05b4!smvoice-sip!_2XX!8!Up!Dial!SIP/524|30|!542!!3!9!SIP/524-05b5 SIP/524-05b3!smvoice-sip!!1!Up!AppDial!(Outgoing Line)!_2XX!!3!40!SIP/526-05b2 SIP/526-05b2!smvoice-sip!_2XX!8!Up!Dial!SIP/524|30|!209!!3!40!SIP/524-05b3 One is on hold and one is active. Shouldnt there be something in the core show channels that says if the call is on hold? I need the same for calls coming over DAHDI. What I'm trying to accomplish is to use the AMI to transfer a call, however, I dont want to transfer the call that is on HOLD. I want to transfer the call that they are currently talking to. I have not found a good way to accomplish this. I tried the extensionstate but that was not working in all cases. like DAHDI. Any ideas? Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones 7942G (skinny): TFTP and required files
On 16/06/11 19:12, Cassius Smith wrote: Hello, I do not use the skinny firmware. By the way, questions like this are best shared with the asterisk-users group mailing list, so that a large segment of the Asterisk community can benefit from the questions and answers. Cassius Smith Agreed -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bridged Digital call
Hi All
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged isdn
channel in 1.6 and 1.4 using the following;-
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten = _X.,3,Noop(${CHANNEL})
exten = _X.,4,hangup
exten = _X.,5,(Set(CHANNEL(transfercapability)=DIGITAL)
exten = _X.,6,dial(DAHDI/g1/${EXTEN})
exten = _X.,7,hangup
this still dials and aswers in 1.8 but no frames are passed and the call
times out and drops
I have also tried
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten = _X.,3,Noop(${CHANNEL})
exten = _X.,4,hangup
exten = _X.,5,Noop
exten = _X.,6,dial(DAHDI/g1d/${EXTEN})
exten = _X.,7,hangup
with exactly the same outcome,
I wondered if I'm missing something in 1.8, has anyone got this working?
Regards
Robb
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[asterisk-users] CDRs in 1.8
I'm using ISDN30 for a bridged application in all the old versions of asterisk the time slot number is shown in the channels and dstchannel fields of the cdr I understand this has chaned in 1.8,is there a way of getting the time slot information stored somewhere at the end of the call so this can be interigated? Thanks in advance Regards Robb -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue not sending call to Agent
After a Good Call from a PSTN phone if I do a sip prune realtime peer 9013XX9XX8 (9013XX9XX8 being the phone number of the Agent/Member) then I can call the number again and not get the issue. So this has something to do with the stuff that is put in my peer table after a call. Any ideas? On Tue, Jun 14, 2011 at 5:18 PM, Duane Larson duane.lar...@gmail.comwrote: One more piece to add. I had mentioned before that I could get a call from a PSTN user to work the first time. So here is all the output of a Good call from a PSTN user after I have performed a RELOAD on asterisks CLI http://pastebin.com/9RSvQsmN And when the caller or agent hangs this call up all calls from the PSTN afterward get put in the queue automatically and the agent never gets called. On Tue, Jun 14, 2011 at 4:37 PM, Duane Larson duane.lar...@gmail.comwrote: Ok. Something isn't right. With a user that is local to my SIP user database calls the queue phone number everything works without issue. It is when a remote user (like someone from the PSTN) calls the queue phone number that the caller gets put into the queue and the agent/member doesn't receive the call. I have captured debugs from OpenSIPS and Asterisk and I can't really see any difference. I also executed the commands you told me where I could. Here are the debugs Good call from local SIP user to Queue LocalUser - OpenSIPSProxy - Asterisk (then asterisk calls the agent/member) - OpenSIPSProxy - Agent http://pastebin.com/Fa9y3CXQ Bad call from PSTN Caller to Queue PSTN Gatway - OpenSIPSB2BUA - OpenSIPSProxy - Asterisk (then asterisk doesn't call Agent/Member for some reason) http://pastebin.com/VBA9nGAs Thanks for looking at this. Currently this happens every time. Any call from a local user gets put in queue and agent is called right away, but any call from PSTN user gets put in queue and agent isn't called but the agent shows as Asterisk18*CLI queue show irock.com has 1 calls (max unlimited) in 'ringall' strategy (6s holdtime, 131s talktime), W:0, C:12, A:15, SL:0.0% within 0s Members: SIP/9013XX9XX8 with penalty 10 (dynamic) (Busy) has taken 12 calls (last was 1991 secs ago) Callers: 1. SIP/9013XX9XX8-002d (wait: 0:02, prio: 0) When it is a good call and I do queue show I see this Asterisk18*CLI queue show irock.com has 0 calls (max unlimited) in 'ringall' strategy (5s holdtime, 131s talktime), W:0, C:12, A:16, SL:0.0% within 0s Members: SIP/9013XX9XX8 with penalty 10 (dynamic) (Busy) has taken 12 calls (last was 2079 secs ago) No Callers *How come with the Bad Call the Agent/Member shows up in a queue show as being a Member and a Caller???* On Mon, Jun 13, 2011 at 11:58 PM, Satish Barot satish4aster...@gmail.com wrote: I am not sure but seems like Agent channel not being released from Asterisk. Next time when this happens, try 'core show channels' to check whether Agent channel is released or not. [SATISH] On Mon, Jun 13, 2011 at 9:12 PM, Duane Larson duane.lar...@gmail.comwrote: Yesterday I rebooted the server and it seems to be working again. Not sure what the reboot might have changed. Hopefully it doesn't happen again but I can't be sure. To answer your question I have the sip.conf in my mysql database and in MySQL I have callcounter set to yes. I don't have a column of 'qualify' in my database for the sip users. For my config I am using OpenSIPS as the register and proxy. Asterisk is only used for voicemail and ACD/Hunt groups. On Mon, Jun 13, 2011 at 12:53 AM, Satish Barot satish4aster...@gmail.com wrote: Provide the entry for Agent SIP/9013XX9XX8 along with parameters 'callcounter' and 'qualify' from sip.conf. Also provide CLI outputs of 'core show channels',sip show peers' and 'queue show' when... (1)First caller enters the Queue (2)First caller gets connected with Agent (3)First caller gets disconnected from Agent (4)Second caller enters the Queue You may have sequences changed for step no 3 and 4 in your scenario. [SATISH] -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Queue Log in Mysql
It is possible to log queue in mysql without turning on realtime asterisk? Thanks! []'sf.rique -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDRs in 1.8
I'm using ISDN30 for a bridged application in all the old versions of asterisk the time slot number is shown in the channels and dstchannel fields of the cdr I understand this has chaned in 1.8,is there a way of getting the time slot information stored somewhere at the end of the call so this can be interigated? Check the ChangeLog of your release to see if the fix to add CHANNEL(dahdi_channel) is present. The fix also added a new AMI DAHDIChannel event. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridged Digital call
Hi All
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged isdn
channel in 1.6 and 1.4 using the following;-
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten = _X.,3,Noop(${CHANNEL})
exten = _X.,4,hangup
exten = _X.,5,(Set(CHANNEL(transfercapability)=DIGITAL)
exten = _X.,6,dial(DAHDI/g1/${EXTEN})
exten = _X.,7,hangup
this still dials and aswers in 1.8 but no frames are passed and the
call times out and drops
I have also tried
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten = _X.,3,Noop(${CHANNEL})
exten = _X.,4,hangup
exten = _X.,5,Noop
exten = _X.,6,dial(DAHDI/g1d/${EXTEN})
exten = _X.,7,hangup
with exactly the same outcome,
Both of these methods should work after doing a quick look a the code.
Does the outgoing call SETUP indicate digital capability?
Richard
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Re: [asterisk-users] Goggle voice incoming dialplan
Do anyone know how to receiving incoming call from GV number associated with an non gmail.com account? I have custom domains under google and would like to receiving calls via asterisk. The google chat function is missing in these GV accounts. On Thu, Jun 16, 2011 at 11:30 AM, asterisk asterisk aster...@ck-lee.comwrote: Thanks and will try. On Thu, Jun 16, 2011 at 11:28 AM, Jamie A. Stapleton jstaple...@computer-business.com wrote: exten = accou...@gmail.com,1,Answer() exten = accou...@gmail.com,n,Wait(2) exten = accou...@gmail.com,n,SendDTMF(1) exten = accou...@gmail.com,n,Dial(SIP/device1) exten = accou...@gmail.com,1,Answer() exten = accou...@gmail.com,n,Wait(2) exten = accou...@gmail.com,n,SendDTMF(1) exten = accou...@gmail.com,n,Dial(SIP/device2) From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk Sent: Wednesday, June 15, 2011 11:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Goggle voice incoming dialplan Hi, I am a question to handle incoming goggle voice. I have put several GV accounts into the jabber.conf. How can I direct different accounts to different extensions? Help with example is much appreciate Thanks, CK -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Goggle voice incoming dialplan
Only GV numbers that can terminate to a Google Chat Account can be connected directly to asterisk. Otherwise you will need to get a free SIP Account, and route calls to it. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk Sent: Thursday, June 16, 2011 11:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Goggle voice incoming dialplan Do anyone know how to receiving incoming call from GV number associated with an non gmail.com account? I have custom domains under google and would like to receiving calls via asterisk. The google chat function is missing in these GV accounts. On Thu, Jun 16, 2011 at 11:30 AM, asterisk asterisk aster...@ck-lee.com wrote: Thanks and will try. On Thu, Jun 16, 2011 at 11:28 AM, Jamie A. Stapleton jstaple...@computer-business.com wrote: exten = accou...@gmail.com,1,Answer() exten = accou...@gmail.com,n,Wait(2) exten = accou...@gmail.com,n,SendDTMF(1) exten = accou...@gmail.com,n,Dial(SIP/device1) exten = accou...@gmail.com,1,Answer() exten = accou...@gmail.com,n,Wait(2) exten = accou...@gmail.com,n,SendDTMF(1) exten = accou...@gmail.com,n,Dial(SIP/device2) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk Sent: Wednesday, June 15, 2011 11:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Goggle voice incoming dialplan Hi, I am a question to handle incoming goggle voice. I have put several GV accounts into the jabber.conf. How can I direct different accounts to different extensions? Help with example is much appreciate Thanks, CK -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Goggle voice incoming dialplan
Can this non gmail.com GV number be terminated at some sip accounts so that I can bridge to it via asterisk as client? On Fri, Jun 17, 2011 at 11:48 AM, William Stillwell will...@stillwellsoft.com wrote: Only GV numbers that can terminate to a Google Chat Account can be connected directly to asterisk. ** ** Otherwise you will need to get a free SIP Account, and route calls to it.* *** ** ** ** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *asterisk asterisk *Sent:* Thursday, June 16, 2011 11:39 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Goggle voice incoming dialplan ** ** Do anyone know how to receiving incoming call from GV number associated with an non gmail.com account? I have custom domains under google and would like to receiving calls via asterisk. The google chat function is missing in these GV accounts. On Thu, Jun 16, 2011 at 11:30 AM, asterisk asterisk aster...@ck-lee.com wrote: Thanks and will try. ** ** On Thu, Jun 16, 2011 at 11:28 AM, Jamie A. Stapleton jstaple...@computer-business.com wrote: exten = accou...@gmail.com,1,Answer() exten = accou...@gmail.com,n,Wait(2) exten = accou...@gmail.com,n,SendDTMF(1) exten = accou...@gmail.com,n,Dial(SIP/device1) exten = accou...@gmail.com,1,Answer() exten = accou...@gmail.com,n,Wait(2) exten = accou...@gmail.com,n,SendDTMF(1) exten = accou...@gmail.com,n,Dial(SIP/device2) From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk Sent: Wednesday, June 15, 2011 11:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Goggle voice incoming dialplan Hi, I am a question to handle incoming goggle voice. I have put several GV accounts into the jabber.conf. How can I direct different accounts to different extensions? Help with example is much appreciate Thanks, CK -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** ** ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Web based call back
Thanks. Will need some time to look into. On Thu, Jun 16, 2011 at 3:56 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Thu, Jun 16, 2011 at 11:26:21AM +0800, asterisk asterisk wrote: Hi, I am looking for a simple solution to do this. I wish to have the user to enter their preferred method of connection i.e. for the cheapest solution to their desktop phone or mobile phone, then plan callfile based on the number that user provided and dial to the user. Any suggestions? But doing so *is* simple. See a simple example attached. It relies on an assumption that the origination IP address authenticates a user and also the user's location (specifically: the phone). You would probably need your own schema for that. But the actual dialing is very simple. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Goggle voice incoming dialplan
I believe you can send a GV # to any US Phone number. That is beyond the scope of this list. But in order to directly terminate the GV # into Asterisk (without using SIP) you must be able to terminate the GV # into a Google Chat Account. as what is being done is theoretically making asterisk a GTalk/Chat/Jabber Client. There is currently no other way to directly terminate a GV # into Asterisk.. (there was some sip rumblings a few weeks ago, but it's all been disabled) as Google is trying to tread lightly in the VoIP Realm as to prevent FCC/Communications issues with taxation technical support issues., etc. (this is just a guess on my part) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk Sent: Thursday, June 16, 2011 11:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Goggle voice incoming dialplan Can this non gmail.com GV number be terminated at some sip accounts so that I can bridge to it via asterisk as client? On Fri, Jun 17, 2011 at 11:48 AM, William Stillwell will...@stillwellsoft.com wrote: Only GV numbers that can terminate to a Google Chat Account can be connected directly to asterisk. Otherwise you will need to get a free SIP Account, and route calls to it. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk Sent: Thursday, June 16, 2011 11:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Goggle voice incoming dialplan Do anyone know how to receiving incoming call from GV number associated with an non gmail.com account? I have custom domains under google and would like to receiving calls via asterisk. The google chat function is missing in these GV accounts. On Thu, Jun 16, 2011 at 11:30 AM, asterisk asterisk aster...@ck-lee.com wrote: Thanks and will try. On Thu, Jun 16, 2011 at 11:28 AM, Jamie A. Stapleton jstaple...@computer-business.com wrote: exten = accou...@gmail.com,1,Answer() exten = accou...@gmail.com,n,Wait(2) exten = accou...@gmail.com,n,SendDTMF(1) exten = accou...@gmail.com,n,Dial(SIP/device1) exten = accou...@gmail.com,1,Answer() exten = accou...@gmail.com,n,Wait(2) exten = accou...@gmail.com,n,SendDTMF(1) exten = accou...@gmail.com,n,Dial(SIP/device2) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk asterisk Sent: Wednesday, June 15, 2011 11:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Goggle voice incoming dialplan Hi, I am a question to handle incoming goggle voice. I have put several GV accounts into the jabber.conf. How can I direct different accounts to different extensions? Help with example is much appreciate Thanks, CK -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
