[asterisk-users] PHP AGI Problems
(Accidentally posted this to asterisk-dev, should be here)
fgets is only returning one character... either when run as an AGI or
run as a test on PHP on CLI...
Example, enter , then fgets returns '3'.
Also, GET DATA seems to be returning early and the loop keeps
prompting 'invalid'...
Any suggestions on how to improve my AGI class so it actually works?
Thanks.
[code]
#!/usr/bin/php
?
//playspecific.php
require_once('database.class.php');
require_once('AGI.class.php');
$database = new database();
$AGI = new AGI();
$database-selectdb(switchboard);
while ($error_count 5 !$msg_result) {
$result = $AGI-send_cmd(GET DATA /bswitch/menu/enter-msg-id 3000 4);
$msg_id = $result[result];
if ($msg_id 1000){
echo \ndebug: msgid 1000, invalid ($msg_id)!\n;
//var_dump($msg_id);
$error_count += 1;
$AGI-send_cmd(EXEC PLAYBACK /bswitch/menu/invalid-msg-id);
}
else{
echo \ndebug: ($msg_id) okay valid msgid, lets check sql\n;
$msg_result=check_msg($msg_id);
if(!$msg_result){
$error_count +=1;
}
}
}
$row = mysql_fetch_array($messageresult);
$AGI-send_cmd(EXEC CONTROL STREAM FILE $row[path]);
$AGI-send_cmd(EXEC PLAYBACK beep);
function check_msg($ID){
$msg_result=$database-querydb(**sql query filtered**);
return $msg_result;
}
?
?php
//AGI.class.php
class AGI {
public $agivars;
function _get_agivars(){
$agivars = array();
while (!feof(STDIN)) {
$agivar = trim(fgets(STDIN));
if ($agivar === '') {
break;
}
$agivar = explode(':', $agivar);
$agivars[$agivar[0]] = trim($agivar[1]);
}
return 0;
}
function send_cmd($cmd){
fputs(STDOUT, $cmd . \n);
fflush(STDOUT);
$data = fgets(STDIN, 4096);
return $data;
}
function __construct(){
$this-_get_agivars();
}
}
?
[/code]
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk memory problems
Hi guys, we have the following problem After putting our Asterisk/PHP application on production, there is one big problem, ie memory leak after a period of usage (about 20MB after 2 minutes) There are also two more info that may help: 1) Asterisk consumes 495mb of memory on this production server, while only 42kB on local/development machine 2) agi_ccmain (and other agi scripts) consumes 146mb of memory on production, while only 32kB on local Also on local, we do not experience the memory leak (or maybe since the memory usage itself is small it's hardly noticeable) The only difference that I am aware of is that on production, DAHDI is installed My questions are: - How come Asterisk, and my PHP/AGI scripts consume so much memory on production compared to local machines? - Any idea why there is a memory leak? IAll of the PHP scripts are done executing, so memory should all be released. Is it a bug on Asterisk? We are using Asterisk 1.6.0.6 CentOS 5.2 on production Any help is kindly appreciated Sincerely, -- Ikin Wirawan Chief Executive Officer PT Walden Global Services Integrity, Learning, Sharing, Excellence http://www.wgs.co.id http://www.kiranatama.com - Web 2.0 development http://www.qorser.com - VoIP solutions http://www.hellomedia.co.id - Digital design promotions ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug or feature in 1.6.1 (Was: How to register with TCP transport) ?
I filed https://issues.asterisk.org/view.php?id=15202 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over VPN
run tcpdump, while trying to connect to asterisk to see what ports are requiered. default SIP port is UDP 5060, but as mentioned before all your traffic should go over VPN so port openening shouldn't be a problem On Wed, May 27, 2009 at 8:40 AM, Marco Sambo derwid...@gmail.com wrote: Ok, but if I want to open only SIP port on firewall, which ones? I have the following situation: computer A (softphone) firewall computer B (asterisk) and I dont' want to open any ports, only SIP and voice. 2009/5/26 David Gibbons d...@videon-central.com Assuming you mean the firewall in front of the client, you don’t need to open any ports as long as the VPN client is tunneling all traffic to and from the Asterisk server. I set NAT=yes in the config file for the extensions behind a VPN. -Dave *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Marco Sambo *Sent:* Tuesday, May 26, 2009 11:21 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] SIP over VPN Hi all, I have a question. I have a VPN and I want to use a SIP softphone on my notebook using with asterisk. But I have some problem with firewall and port. Someone knows which ports I should open on my firewall??? I can't connect ... Thanks all. Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mvh, Aurimas Skirgaila ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
On Tue, May 26, 2009 at 07:52:53PM +0300, Tzafrir Cohen wrote: On Tue, May 26, 2009 at 12:44:26PM -0400, Jon Pounder wrote: Wilton Helm wrote: one thing I missed mentioning about fxs devices - the linksys/sipura ones actually allow you to set line characteristics on the slic inside it. you can vary from the 600ohm default, and tweak gains a bit. Some mix of a capacitive line or different resistance may help. never tried myself but there are a ton of things you can play with. Any of those are actually important? For the sake of completeness: try: /sbin/modinfo wctdm or: /sbin/modinfo wctdm24xxp You'll see quite a few parameters, many of which are essentially SLIC (or DAA, for the FXO port) tweaks. I suppose that if it were useful, there were already some demand to make it tweakable (safely. Merely writing an arbitrary value to some register at some point may or may not be wise). Hence my question. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Delay and Zombie Channels Problem
Hi All, I have problems with delay, this delay come when there is a lot of zombie channels, I use phpagi for process the call, but when the active call is less than 10 calls thats working good, this problems (zombie channels and delay) come when more than 10 active calls, i have tried to optimized the php but still same. is there somebody here can help me.. -- Thanks Niko ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Video Conference Software (Open Source)
On Fri, May 15, 2009 at 4:10 AM, Cesar Real realn...@hotmail.com wrote: Hi: I am use ISABEL Software for video conference. Hi Before I try, if this software really free? please explanation. Thank you. http://www.agora-2000.com/pdfs/Isabel-4.10_Introduction.pdf http://videoconferencia.reuna.cl/wiki/index.php/Isabel -- Date: Mon, 27 Apr 2009 09:57:14 +0700 From: joko.pit...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Video Conference Software (Open Source) I am looking for Video Conference Software (Open Source) , But but not for free Trial.. please give reference about it. Thanks -- Get news, entertainment and everything you care about at Live.com. Check it out! http://www.live.com/getstarted.aspx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silly (??) question about chan_dahdi
Hi Martin, thanks for your suggestions, but You define context= for the channels in dahdi.conf and then in extensions.conf you define those numbers in that particular context name eg: dahdi.conf context=incoming channel = 1-15,17-31 in /etc/asterisk there is a chan_dahdi.conf and a dahdi-channels.conf which were automatically created (and a little bit modified by me): chan_dadhi.conf: [trunkgroups] [channels] language=de context=incoming switchtype=euroisdn group = 1 channel = 1,2,4,5,7,8,10,11 overlapdial=yes usecallerid=yes callwaiting=no callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes group=1 callgroup=1 pickupgroup=1 dahdi-channels.conf: ; Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 AMI/CCS RED group=0,11 context=incoming switchtype = euroisdn signalling = bri_cpe channel = 1-2 group = 63 ; Span 2: B4/0/2 B4XXP (PCI) Card 0 Span 2 AMI/CCS RED group=0,12 context=incoming switchtype = euroisdn signalling = bri_cpe channel = 4-5 group = 63 ; Span 3: B4/0/3 B4XXP (PCI) Card 0 Span 3 (MASTER) AMI/CCS group=0,13 context=incoming switchtype = euroisdn signalling = bri_cpe channel = 7-8 group = 63 ; Span 4: B4/0/4 B4XXP (PCI) Card 0 Span 4 AMI/CCS RED group=0,14 context=incoming switchtype = euroisdn signalling = bri_cpe channel = 10-11 group = 63 extensions.conf: - [incoming] exten = 8304479,1,ANSWER() exten = 8304479,2,WAIT(10) exten = 8304479,3,HANGUP() exten = 8304478,1,ANSWER() exten = 8304478,2,WAIT(10) exten = 8304478,3,HANGUP() Asterisk still doesn't pick up calls for these two numbers. I'm a little bit irritated by the fact, that in the cli asterisk mentions ISDN PRI signalling while we are using BRI . BTW, the warning is currently correct, the is an isdn cable only in port 3. == Parsing '/etc/asterisk/chan_dahdi.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found [May 27 09:39:35] WARNING[5145]: chan_dahdi.c:4664 handle_alarms: Detected alarm on channel 1: Red Alarm -- Registered channel 1, ISDN PRI signalling [May 27 09:39:35] WARNING[5145]: chan_dahdi.c:4664 handle_alarms: Detected alarm on channel 2: Red Alarm -- Registered channel 2, ISDN PRI signalling [May 27 09:39:35] WARNING[5145]: chan_dahdi.c:4664 handle_alarms: Detected alarm on channel 4: Red Alarm -- Registered channel 4, ISDN PRI signalling [May 27 09:39:35] WARNING[5145]: chan_dahdi.c:4664 handle_alarms: Detected alarm on channel 5: Red Alarm -- Registered channel 5, ISDN PRI signalling -- Registered channel 7, ISDN PRI signalling -- Registered channel 8, ISDN PRI signalling [May 27 09:39:35] WARNING[5145]: chan_dahdi.c:4664 handle_alarms: Detected alarm on channel 10: Red Alarm -- Registered channel 10, ISDN PRI signalling [May 27 09:39:35] WARNING[5145]: chan_dahdi.c:4664 handle_alarms: Detected alarm on channel 11: Red Alarm -- Registered channel 11, ISDN PRI signalling -- Automatically generated pseudo channel == Registered channel type 'DAHDI' (DAHDI Telephony Driver) The two numbers above are correct, when I use an ISDN phone it rings for both numbers. Thanks for your help, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PHP AGI Problems
2009/5/27 Atlanticnynex atlanticny...@gmail.com
(Accidentally posted this to asterisk-dev, should be here)
fgets is only returning one character... either when run as an AGI or
run as a test on PHP on CLI...
Example, enter , then fgets returns '3'.
It's not... There are problems with the way you are handling the return
data...
Also, GET DATA seems to be returning early and the loop keeps
prompting 'invalid'...
That's happening because you are sending junk to asterisk...
Any suggestions on how to improve my AGI class so it actually works?
Have a look at http://www.voip-info.org/wiki/view/Asterisk+AGI+php...
Thanks.
[code]
snip
while ($error_count 5 !$msg_result) {
$result = $AGI-send_cmd(GET DATA /bswitch/menu/enter-msg-id 3000 4);
$msg_id = $result[result];
Problem here, you're looking to extract the value for key result (which as
it's not quoted could be interpreted as a constant) from the array $result,
but, $result isn't an array
if ($msg_id 1000){
echo \ndebug: msgid 1000, invalid ($msg_id)!\n;
Junk is being sent to asterisk which is responding with errors (echo sends
to STDOUT too), you're only getting one line at a time of response, so,
after the first time in you are not getting the results you'd expect in the
GET DATA, you're getting the rest of the error messages from the junk you
sent...
?php
//AGI.class.php
class AGI {
public $agivars;
snip
function send_cmd($cmd){
fputs(STDOUT, $cmd . \n);
fflush(STDOUT);
$data = fgets(STDIN, 4096);
return $data;
}
So, as said above, you're sending the command here, then taking the next
line from asterisk and dumping it back unprocessed in any way...
When you're sending the GET DATA command, if you enter , what you are
getting back is...
'200 result=\n'
$result actually contains this value, however, when you set $msg_id to
$result[result] what you get is the first character in $msg_id, so, in this
case you get '2'...
As this is lower than 1000 you go into the invalid number part of your code
where you echo a debug message then send the command to play the invalid
sound file... Sending the command to play the sound file results in a 1 line
response being received, your echo command put out two invalid lines, so,
what is actually received as a response this time is...
'510 Invalid or unknown command\n'
The next command you send is the GET DATA command to try and get a new
number, however, at this point there is already data waiting in the buffer,
so as soon as the data is sent, your fgets returns straight away with...
'510 Invalid or unknown command\n'
as a result of the second line of debug output that you sent...
$result contains this string and again, you get the first character in
$msg_id, so now $msg_id equals '5', this again is lower than 1000, so you go
into the invalid number part of your code and drop more junk into asterisk
and continue...
I hope that's useful in explaining the problems you are experiencing... as I
said above, check out the page on voip-info, it should be enough for you to
get this fixed up nicely... One think you'll notice about the examples on
there, debug output is sent to a file, that way you can record information
for debugging without polluting your agi...
d
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silly (??) question about chan_dahdi
On Wed, May 27, 2009 at 09:55:13AM +0200, Stefan-Michael Guenther wrote: Hi Martin, thanks for your suggestions, but You define context= for the channels in dahdi.conf and then in extensions.conf you define those numbers in that particular context name eg: dahdi.conf context=incoming channel = 1-15,17-31 in /etc/asterisk there is a chan_dahdi.conf and a dahdi-channels.conf which were automatically created (and a little bit modified by me): chan_dadhi.conf: [trunkgroups] [channels] language=de context=incoming switchtype=euroisdn group = 1 channel = 1,2,4,5,7,8,10,11 Anything after that line has no effect. overlapdial=yes usecallerid=yes callwaiting=no callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes group=1 callgroup=1 pickupgroup=1 dahdi-channels.conf: ; Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 AMI/CCS RED group=0,11 context=incoming switchtype = euroisdn signalling = bri_cpe channel = 1-2 group = 63 You have no #include of that (which is probably a good thing, as it overlaps with channels you already set in chan_dahdi.conf itself). Hence the value of that file is mostly informative: this is what dahdi_genconf generated :-) [snip] -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI and hangup issue when playing the IVR
Good day , I have configured TDM410P (asterisk 1.6.x) on Cent OS 5, but dahdi take some time to hangup the call when playing the IVR..(it will send the hangup signal after finishing the IVR promt..) is there any specific setting to avoid such incidents ? iam using busycount as 3, signalling=fxs_ks ;toneduration=100 callwaiting=yes threewaycalling=yes callreturn=yes echocancel=128,param1=32,param2=0,param3=14 echocancelwhenbridged=yes echotraining=yes echotraining=800 busycount=3 hanguponpolarityswitch=yes ringtimeout=8000 group=1 context=incoming immediate=yes jitterbuffers=4 jbenable = yes echocancel=yes channel=1-4 ;overlapdial=yes ;pulsedial=yes dtmfmode=rfc2833 ;relaxdtmf=yes ;rxgain=10.0 ;txgain=8.0 any ideas please! Thanks, Tharanga Abyeseela ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Domains
Noone can give me a clue on this ? How Domains are used within Asterisk ? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 26 May 2009 12:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Domains Hi, I'm trying to understand an issue I'm seeing between two Asterisk servers. I think it has to do with Domain definitions. Server A), has extension 5550 defined. Has a sip client 2000 defined, and has guest-invites enabled. Server B), Dials to server A for any 5550 dialled. Has sip client 2000 and 2001 defined. If I register at server B as client 2001, and dial 5550 then the call works, and is placed through to server As logic successfully. But if I call in as client 2000, then the call fails, server A shows no log at all of the call (even a sip set debug ip ip showed nothing - though tcpdump did show the inbound invite). However if I remove the definition of client 2000 from server A, then the call succeeds. So I think that for a defined account server A is wanting to challenge for a password, even though the inbound call is not a local account - hence my trying now to understand if and how Asterisk uses Domains. If I define a serverA.company.com domain on server A, will it ignore the challenge for an INVITE coming from server B ?? Thanks Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstDB wildcards
Hi All,
I need to use partial matches on the CIDNAME family I have stored in
AstDB. For example, an organisation might have several numbers with
the same area code and the same first few digits:
1234 567890
1234 567889
1234 567824
...
I'd like to store these (e.g.) as CIDNAME/12345678* (where * is a
wildcard) so that I can retrieve the organisation name from
extensions.conf with:
Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
Does AstDB support this (I'm using Asterisk 1.4.22.1)?
I know that I can create a function to iterate backwards through the
number until a partial match is met, but I'd rather use built-in
functionality should it exist.
TIA,
--
Geoff
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstDB wildcards
Hi,
On 12:09, Wed 27 May 09, Geoff Lane wrote:
Hi All,
I need to use partial matches on the CIDNAME family I have stored in
AstDB. For example, an organisation might have several numbers with
the same area code and the same first few digits:
1234 567890
1234 567889
1234 567824
...
I'd like to store these (e.g.) as CIDNAME/12345678* (where * is a
wildcard) so that I can retrieve the organisation name from
extensions.conf with:
Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
Does AstDB support this (I'm using Asterisk 1.4.22.1)?
Nope.
I know that I can create a function to iterate backwards through the
number until a partial match is met, but I'd rather use built-in
functionality should it exist.
It's your only option.
--
Michiel van Baak
mich...@vanbaak.eu
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer aficionados are both called users?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over VPN
Hi, but if I want to open only SIP port on firewall, which ones? 5060 and check the /etc/asterisk/rtp.conf. You might want to limit the ports: rtpstart=1 ;rtpend=2 rtpend=10099 -- Chau y hasta luego, Thorolf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] stucked calls in asterisk 1.4
hello, i have a problem with stucked or hanging calls in asterisk 1.4.25 we had this problem before and so we upgradet from 1.2.32 to 1.4.25 but it still exists and as i could see, happens even more. on this server there are 1500 clients registered all with qualify on and we had 2 routing server with 4 E1 pstn connects on each. The connect between this server and the 2 routing server runs over sip. This problem only appears on this server and not on the routing server. Even if i soft hangup a stucked channel i get the sip response 481 call/leg transaction doesnt exists back from the routing server, so one call leg had allready send a bye but this server hasnt closed the call. i think there could be a network problem but also the server itself and the switch where the 3 servers are connected is younger than 3 months and we had the same problem with this system on an older server. sometimes i see that most of the sip peers get unreachable or too lagged so i think that there could be a problem for asterisk to handle that amount of pakets. i´ve just splittet up the traffic on 2 interfaces, so that normal traffic from the clients comes to eth0 and the traffic from and to the routing servers runs over eth1 but the problem still occurs. do you have any ideas what i could do to solve this problem? best regards steve smith ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pressing number 2 in dialplan
Hello! I am having an odd problem in that when the caller dials extension 2 in a dialplan, the system waits 3 to 4 seconds before proceeding. This doesn't happen when any other other extensions are dialed, including an identical dialplan on other another extension! Is this a bug? Later, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pressing number 2 in dialplan
Hello All! I see where I'm going off. I have other extensions that start with 2, so Asterisk is delaying things. Check out http://www.pbxer.com/problem-resolved-the-slow-ringing-extension I think this is important and should be documented somewhere, if it is not already. Elliot On 5/27/09, Elliot Murdock murdo...@gmail.com wrote: Hello! I am having an odd problem in that when the caller dials extension 2 in a dialplan, the system waits 3 to 4 seconds before proceeding. This doesn't happen when any other other extensions are dialed, including an identical dialplan on other another extension! Is this a bug? Later, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pressing number 2 in dialplan
On 14:49, Wed 27 May 09, Elliot Murdock wrote: Hello! I am having an odd problem in that when the caller dials extension 2 in a dialplan, the system waits 3 to 4 seconds before proceeding. This doesn't happen when any other other extensions are dialed, including an identical dialplan on other another extension! Do you have other extensions that start with a two and are longer then 1 digit ? If so, that's the reason. Is this a bug? Later, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No full duplex communication ?
Hey list ! I'm getting the feedback of a customer that a conversation is like half duplex : when he talks, the other end of the call is no longer heard. What could be the cause of these drop-outs ? A call that is coming in from the PSTN is routed through an IVR-system to the correct internal SIP-phone (Grandstream GXP2020). Where do I start searching for this problem ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pressing number 2 in dialplan
Hello Michiel, Yep...that's the reason. I changed those other extensions and everything is fine now. Asterisk is pretty clever, just need to keep up with it. Thanks Elliot On 5/27/09, Michiel van Baak mich...@vanbaak.info wrote: On 14:49, Wed 27 May 09, Elliot Murdock wrote: Hello! I am having an odd problem in that when the caller dials extension 2 in a dialplan, the system waits 3 to 4 seconds before proceeding. This doesn't happen when any other other extensions are dialed, including an identical dialplan on other another extension! Do you have other extensions that start with a two and are longer then 1 digit ? If so, that's the reason. Is this a bug? Later, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstDB wildcards
Maybe splitting key-value like this would help ? database put Prefix1234 56789 database put Prefix1234 56788 database put Prefix1222 56789 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stucked calls in asterisk 1.4
On Wed, May 27, 2009 at 7:32 AM, Stefan Schmidt s...@sil.at wrote: i have a problem with stucked or hanging calls in asterisk 1.4.25 only appears on this server and not on the routing server. Even if i I'm confused. So the server where the calls get stuck has both SIP and DAHDI/Zaptel channel calls? And the SIP side of those calls isn't getting hungup? It is true that if your server doesn't receive a BYE packet it will think that the call is still there. Does your dialplan that attaches to the routing server do a hangup at the end of the call? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI and hangup issue when playing the IVR
On Wed, May 27, 2009 at 4:19 AM, Tharanga thara...@roomsnet.com wrote: I have configured TDM410P (asterisk 1.6.x) on Cent OS 5, but dahdi take some time to hangup the call when playing the IVR..(it will send the Dahdi should hangup the call when you tell it to. Are you sure there isn't silence at the end of your IVR message, or a DTMF timeout, or something else like that that is playing silence and keeping the line open? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400P in PCI-X Slot
Hi, Does anyone know if a TDM400P will work in a PCI-X slot? Work is offloading a few (2+ year old) workstations. If I act fast I can buy one, but it only has 1 PCI slot (and I'll need that for something else.) There are several PCI-X slots available, so this would be the only option for the TDM400P. Thanks, MikeC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silly (??) question about chan_dahdi
On Wed, May 27, 2009 at 3:55 AM, Stefan-Michael Guenther asteris...@in-put.de wrote: extensions.conf: - [incoming] exten = 8304479,1,ANSWER() exten = 8304479,2,WAIT(10) exten = 8304479,3,HANGUP() exten = 8304478,1,ANSWER() exten = 8304478,2,WAIT(10) exten = 8304478,3,HANGUP() Asterisk still doesn't pick up calls for these two numbers. Is that your entire extensions.conf? If so, this is part of your problem. If you do the square-bracket [] style contexts, you need to have jumps to them from the default context. So your plan should say extensions.conf [default] exten = 8304479,1,ANSWER() exten = 8304479,2,WAIT(10) exten = 8304479,3,HANGUP() exten = 8304478,1,ANSWER() exten = 8304478,2,WAIT(10) exten = 8304478,3,HANGUP() And you will need to cli dialplan reload to reload extensions.conf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P in PCI-X Slot
Michael C. Cambria wrote: Does anyone know if a TDM400P will work in a PCI-X slot? It will. Work is offloading a few (2+ year old) workstations. If I act fast I can buy one, but it only has 1 PCI slot (and I'll need that for something else.) There are several PCI-X slots available, so this would be the only option for the TDM400P. PCI-X (not PCI-E) slots are backwards compatible with PCI slots, by definition. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stucked calls in asterisk 1.4
hello David Backeberg schrieb: On Wed, May 27, 2009 at 7:32 AM, Stefan Schmidt s...@sil.at wrote: i have a problem with stucked or hanging calls in asterisk 1.4.25 only appears on this server and not on the routing server. Even if i I'm confused. So the server where the calls get stuck has both SIP and DAHDI/Zaptel channel calls? There are 3 servers. Server A call it PBX there are the sip clients connected Server B call it gateway1 has an Sangoma Card in it Server C call it gateway2 also has an sangoma Card in it A call comes from server B or C to server A and then to a client, gets stucked on Server A when PSTN side hangs up. On server B or C the call is closed. This also happens in other direction, if client dials out over Server a to server b or c to the pstn net. And the SIP side of those calls isn't getting hungup? thats correct. It is true that if your server doesn't receive a BYE packet it will think that the call is still there. Does your dialplan that attaches to the routing server do a hangup at the end of the call? on the routing server the call is closed every time. There we didnt have this problem. On this server (B+C) also terminate calls from a ser proxy and another asterisk server but the call stucks only on server A. best regards steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with T.38 media headers
Hi Guys, Something I have noticed while dealing with T.38 and re-invites in Asterisk 1.4.22. I have a provider who re-invites with the following sdp (message flow PROVIDER_EQPMT - ASTERISK): . v=0. o=SIP_5F9 123456 654322 IN IP4 CONN_IP_PROVIDER. s=-. c=IN IP4 CONN_IP_PROVIDER. t=0 0. m=audio 0 RTP/AVP 0. m=image 26858 udptl t38. a=T38FaxMaxBuffer:288. a=T38FaxRateManagement:transferredTCF. a=T38FaxUdpEC:t38UDPRedundancy. The answer coming from asterisk in this case is: . v=0. o=root 3484 3485 IN IP4 CONN_IP_ASTERISK. s=session. c=IN IP4 CONN_IP_ASTERISK. t=0 0. m=image 4653 udptl t38. a=T38FaxVersion:0. a=T38MaxBitRate:9600. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:200. a=T38FaxMaxDatagram:200. a=T38FaxUdpEC:t38UDPRedundancy. I see a problem here since the number of matched media streams from the offer does not match with the number of matched media streams in reply from asterisk (notice the m=audio 0 RTP/AVP 0 not present in the reply). Please let me know if there are workarounds on this issue, or if this could be a bug on asterisk side. Best regards, Mario Staphorst _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with T.38 media headers
This is not a problem. Asterisk is under no obligation to offer an audio codec in return. mario staphorst wrote: Hi Guys, Something I have noticed while dealing with T.38 and re-invites in Asterisk 1.4.22. I have a provider who re-invites with the following sdp (message flow PROVIDER_EQPMT - ASTERISK): . v=0. o=SIP_5F9 123456 654322 IN IP4 CONN_IP_PROVIDER. s=-. c=IN IP4 CONN_IP_PROVIDER. t=0 0. m=audio 0 RTP/AVP 0. m=image 26858 udptl t38. a=T38FaxMaxBuffer:288. a=T38FaxRateManagement:transferredTCF. a=T38FaxUdpEC:t38UDPRedundancy. The answer coming from asterisk in this case is: . v=0. o=root 3484 3485 IN IP4 CONN_IP_ASTERISK. s=session. c=IN IP4 CONN_IP_ASTERISK. t=0 0. m=image 4653 udptl t38. a=T38FaxVersion:0. a=T38MaxBitRate:9600. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:200. a=T38FaxMaxDatagram:200. a=T38FaxUdpEC:t38UDPRedundancy. I see a problem here since the number of matched media streams from the offer does not match with the number of matched media streams in reply from asterisk (notice the m=audio 0 RTP/AVP 0 not present in the reply). Please let me know if there are workarounds on this issue, or if this could be a bug on asterisk side. Best regards, Mario Staphorst Express yourself instantly with MSN Messenger! MSN Messenger http://clk.atdmt.com/AVE/go/onm00200471ave/direct/01/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stucked calls in asterisk 1.4
On Wed, May 27, 2009 at 9:26 AM, Stefan Schmidt s...@sil.at wrote: Server A call it PBX there are the sip clients connected A call comes from server B or C to server A and then to a client, gets stucked on Server A when PSTN side hangs up. On server B or C the call You may not have properly configured your card for the way to detect hangups. If this is going to a proprietary PBX, you may need to change around the line signaling. Some kinds of line signaling reverse the polarity on the line voltage to signal a hangup. Other lines don't. The cheap trick is to try your settings both ways and when one way works to use that. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk memory problems
On Wed, May 27, 2009 at 2:33 AM, Ikin Wirawan i...@qorser.com wrote: There are also two more info that may help: 1) Asterisk consumes 495mb of memory on this production server, while only 42kB on local/development machine 2) agi_ccmain (and other agi scripts) consumes 146mb of memory on production, while only 32kB on local This is the second or more questions in a few days that claims memory leak with no proof. Please google 'memory leak' before throwing around the term. You do not have any evidence of that. As for why it takes more memory in production than in your test, an obvious guess is because there are more simultaneous instances of your process in your production environment than in your test environment. Here's how you compare apples to apples... Put your test server into production, and then compare the memory usage. You have way too many variables to make any conclusions about anything. My questions are: - How come Asterisk, and my PHP/AGI scripts consume so much memory on production compared to local machines? I don't know, and neither do you. Switch the systems and try again. Do you have the same version, same compiled options, and same number of simultaneous PHP engines running in your test? The obvious suggestion is you are running more instances of your AGIs in prod than in test. Ergo, more memory usage. - Any idea why there is a memory leak? - You don't have any evidence of a memory leak. Linux caches programs in ram in the event they get run again. Linux the kernel and other smart programs cache/buffer previously used items in memory, in the (usually likely) event that you want them again, and they won't have to be read from hard drive in the future. That's not called a memory leak. That's called evidence-based performance enhancements. Is it a bug on Asterisk? Doubtful. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call in progress tones
On Tue, May 26, 2009 at 8:46 PM, Mikel Lindsaar raasd...@gmail.com wrote: Does anyone know of a way to have tones played during the call progress stage of the call? Any ideas? Sorry, you had the same ideas I had. Are you getting a lot of complaints from users that certain international calls take a long time to setup? You could detect what was dialed and route accordingly, and have a caveat play before the dial to those particular calls instead. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stucked calls in asterisk 1.4
David Backeberg schrieb: On Wed, May 27, 2009 at 9:26 AM, Stefan Schmidt s...@sil.at wrote: Server A call it PBX there are the sip clients connected A call comes from server B or C to server A and then to a client, gets stucked on Server A when PSTN side hangs up. On server B or C the call You may not have properly configured your card for the way to detect hangups. If this is going to a proprietary PBX, you may need to change around the line signaling. Some kinds of line signaling reverse the polarity on the line voltage to signal a hangup. Other lines don't. The cheap trick is to try your settings both ways and when one way works to use that. its not a proprietary pbx its just a self developed asterisk and the server where the card is recognize the hangup, but the bye from the server (b or c) to the pbx dont work. as i said the routing server also handles calls from an ser proxy and another asterisk server where iax accounts terminates and this problem is only on the pbx server. Maybe it is a network problem but the quality of the rtp streams is ok but i think that there are too much sip pakets for the system. there are around 1600 sip users registerd, 300 - 400 sip channels (register, options, notifys and invite) and 600 - 700 subscriptions so there is much sip traffic. this server also does rtp handling and have 50 to 100 calls (active ones) and in peek time there is around 10mbit of traffic with 5 to 6 kpps. maybe a problem with udp buffer size?? thanks for your help so far david! best regards steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.6.0.9: Now Unable to create ... 'DAHDI'
Still trying to upgrade to 1.6.0.9 for 1.4. It worked - it worked all day yesterday, but this morning: -- Executing [646xxxy...@longdistance:1] Answer(SIP/172-08276a60, ) in new stack .. -- Executing [646xxx...@longdistance:6] Dial(SIP/172-08276a60, DAHDI/g2/1646xxx) in new stack May 27 09:56:57] WARNING[16589]: app_dial.c:1468 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) dahdi seems up. I restarted. Rebooted. Now I've reverted to 1.4. CLI dahdi show channels Chan Extension Context Language MOH Interpret BlockedState pseudofrom-pstn en default In Service 1from-pstn en default In Service 2from-pstn en default In Service 3from-pstn en default In Service 4from-pstn en default In Service 5from-pstn en default In Service 6from-pstn en default In Service 7from-pstn en default In Service 8from-pstn en default In Service dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TE120P Card 0 OK 1 0 0 ESF B8ZS YEL 0 db (CSU)/0-133 feet (DSX-1) CLI dahdi show version DAHDI Version: 2.1.0.4 Echo Canceller: MG2 What should I try next? Any help appreciated. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with T.38 media headers
Hi, I think this is not completely right, The scenario is: Carrier == Asterisk 1.4 == T.38 ATA box. What happends is that the header disappears within the Asterisk server and is not reaching the ATA.I think the SDP headers should be passed through in all circumstances, even if Asterisk 1.4 is only doing T.38 passthrough? Regards, Mario Date: Wed, 27 May 2009 09:44:56 -0400 From: abalas...@evaristesys.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problem with T.38 media headers This is not a problem. Asterisk is under no obligation to offer an audio codec in return. mario staphorst wrote: Hi Guys, Something I have noticed while dealing with T.38 and re-invites in Asterisk 1.4.22. I have a provider who re-invites with the following sdp (message flow PROVIDER_EQPMT - ASTERISK): . v=0. o=SIP_5F9 123456 654322 IN IP4 CONN_IP_PROVIDER. s=-. c=IN IP4 CONN_IP_PROVIDER. t=0 0. m=audio 0 RTP/AVP 0. m=image 26858 udptl t38. a=T38FaxMaxBuffer:288. a=T38FaxRateManagement:transferredTCF. a=T38FaxUdpEC:t38UDPRedundancy. The answer coming from asterisk in this case is: . v=0. o=root 3484 3485 IN IP4 CONN_IP_ASTERISK. s=session. c=IN IP4 CONN_IP_ASTERISK. t=0 0. m=image 4653 udptl t38. a=T38FaxVersion:0. a=T38MaxBitRate:9600. a=T38FaxRateManagement:transferredTCF. a=T38FaxMaxBuffer:200. a=T38FaxMaxDatagram:200. a=T38FaxUdpEC:t38UDPRedundancy. I see a problem here since the number of matched media streams from the offer does not match with the number of matched media streams in reply from asterisk (notice the m=audio 0 RTP/AVP 0 not present in the reply). Please let me know if there are workarounds on this issue, or if this could be a bug on asterisk side. Best regards, Mario Staphorst Express yourself instantly with MSN Messenger! MSN Messenger http://clk.atdmt.com/AVE/go/onm00200471ave/direct/01/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ What can you do with the new Windows Live? Find out http://www.microsoft.com/windows/windowslive/default.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stucked calls in asterisk 1.4
On Wed, May 27, 2009 at 10:30 AM, Stefan Schmidt s...@sil.at wrote: as i said the routing server also handles calls from an ser proxy and another asterisk server where iax accounts terminates and this problem is only on the pbx server. Maybe it is a network problem but the quality of the rtp streams is ok but i think that there are too much sip pakets for the system. there are around 1600 sip users registerd, 300 - 400 sip channels (register, options, notifys and invite) and 600 - 700 subscriptions so there is much sip traffic. this server also does rtp handling and have 50 to 100 calls (active ones) and in peek time there is around 10mbit of traffic with 5 to 6 kpps. maybe a problem with udp buffer size?? Now that I better understand your problem, I'm out of ideas. You are correct that if a BYE sip packet gets lost, a) it won't get retransmitted if it's UDP b) the side that's waiting for the hangup will think the call is still active I've seen this in my system where the network switch went down while calls were active. The system where the call was happening caught the hangup, but the trunk system never got the bye as the network was down. My only questions are: Are you using a quality network switch, or maybe you can use a cross-over cable to eliminate collisions with other systems? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.0.9: Now Unable to create ... 'DAHDI'
On Wed, May 27, 2009 at 10:46:39AM -0400, sean darcy wrote: Still trying to upgrade to 1.6.0.9 for 1.4. It worked - it worked all day yesterday, but this morning: -- Executing [646xxxy...@longdistance:1] Answer(SIP/172-08276a60, ) in new stack .. -- Executing [646xxx...@longdistance:6] Dial(SIP/172-08276a60, DAHDI/g2/1646xxx) in new stack May 27 09:56:57] WARNING[16589]: app_dial.c:1468 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) Is that correct? ast_verb(2, Everyone is busy/congested at this time (%d:%d/%d/%d)\n, numlines, num.busy, num.congestion, num.nochan); We have a total of 1 channels, of which: - 0 are busy - 0 are congested - 1 don't exist (failed to be generated?) dahdi seems up. I restarted. Rebooted. Now I've reverted to 1.4. CLI dahdi show channels Chan Extension Context Language MOH Interpret BlockedState pseudofrom-pstn en default In Service 1from-pstn en default In Service 2from-pstn en default In Service 3from-pstn en default In Service 4from-pstn en default In Service 5from-pstn en default In Service 6from-pstn en default In Service 7from-pstn en default In Service 8from-pstn en default In Service dahdi show status Description Alarms IRQbpviol CRC4 Fra Codi Options LBO Wildcard TE120P Card 0 OK 1 0 0 ESF B8ZS YEL 0 db (CSU)/0-133 feet (DSX-1) Isn't the span in alarm? Do the channels in group=2 have 'InAlarm: Yes' in 'dahdi show channel NN'? -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with T.38 media headers
mario staphorst wrote: Carrier == Asterisk 1.4 == T.38 ATA box. What happends is that the header disappears within the Asterisk server and is not reaching the ATA. I think the SDP headers should be passed through in all circumstances, even if Asterisk 1.4 is only doing T.38 passthrough? Asterisk is not a proxy; SIP signaling is never 'passed through'; the two legs of a call are completely separate and Asterisk bridges them together when necessary. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400P in PCI-X Slot
Just check the version of the card (5v vs 3v) - I don't think PCI X is compatible with the older 5v cards. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Wednesday, May 27, 2009 9:20 AM To: Asterisk Users List Subject: Re: [asterisk-users] TDM400P in PCI-X Slot Michael C. Cambria wrote: Does anyone know if a TDM400P will work in a PCI-X slot? It will. Work is offloading a few (2+ year old) workstations. If I act fast I can buy one, but it only has 1 PCI slot (and I'll need that for something else.) There are several PCI-X slots available, so this would be the only option for the TDM400P. PCI-X (not PCI-E) slots are backwards compatible with PCI slots, by definition. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.0.9: Now Unable to create ... 'DAHDI'
On Wed, 2009-05-27 at 10:46 -0400, sean darcy wrote: -- Executing [646xxx...@longdistance:6] Dial(SIP/172-08276a60, DAHDI/g2/1646xxx) in new stack It appears you're attempting to dial DAHDI/g2/1646xxxyyy instead of DAHDI/g2/1646xxx... Did you mean to put those extra quotes in there? -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silly (??) question about chan_dahdi
Hi, I have set context=default both in /etc/asterisk/dahdi-channels.conf and /etc/asterisk/chan_dahdi.conf, and created the necessary context with extens for both numbers I have replaced the ISDN cable, the LED on the card is green, but still asterisk doesn't react to call to 8304478 and 8304479, while the ISDN phone does, No, they are not both connected at the same time. Stefan David Backeberg schrieb: On Wed, May 27, 2009 at 3:55 AM, Stefan-Michael Guenther asteris...@in-put.de wrote: extensions.conf: - [incoming] exten = 8304479,1,ANSWER() exten = 8304479,2,WAIT(10) exten = 8304479,3,HANGUP() exten = 8304478,1,ANSWER() exten = 8304478,2,WAIT(10) exten = 8304478,3,HANGUP() Asterisk still doesn't pick up calls for these two numbers. Is that your entire extensions.conf? If so, this is part of your problem. If you do the square-bracket [] style contexts, you need to have jumps to them from the default context. So your plan should say extensions.conf [default] exten = 8304479,1,ANSWER() exten = 8304479,2,WAIT(10) exten = 8304479,3,HANGUP() exten = 8304478,1,ANSWER() exten = 8304478,2,WAIT(10) exten = 8304478,3,HANGUP() And you will need to cli dialplan reload to reload extensions.conf -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with T.38 media headers
Hi Kevin, Thank you for your reply.I understand that Asterisk is not a SIP proxy, but shouldnt this header be passed on in order to provide proper T.38 passthrough support in this case?As far as i can see is this header really needed to make the T.38 connection successfull, when i setup the call directly to the ATA the reinvite is going fine. Do you have any idea how we can fix this issue? Best regards, Mario Date: Wed, 27 May 2009 10:13:27 -0500 From: kpflem...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] problem with T.38 media headers mario staphorst wrote: Carrier == Asterisk 1.4 == T.38 ATA box. What happends is that the header disappears within the Asterisk server and is not reaching the ATA. I think the SDP headers should be passed through in all circumstances, even if Asterisk 1.4 is only doing T.38 passthrough? Asterisk is not a proxy; SIP signaling is never 'passed through'; the two legs of a call are completely separate and Asterisk bridges them together when necessary. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ What can you do with the new Windows Live? Find out http://www.microsoft.com/windows/windowslive/default.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silly (??) question about chan_dahdi
On Wed, May 27, 2009 at 12:19 PM, Stefan-Michael Guenther asteris...@in-put.de wrote: I have replaced the ISDN cable, the LED on the card is green, but still asterisk doesn't react to call to 8304478 and 8304479, while the ISDN phone does, No, they are not both connected at the same time. Please paste in your entire extensions.conf I'm not sure you understood what I told you on my last reply. Please also paste in the output of: cli dialplan show ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silly (??) question about chan_dahdi
You should also do core set verbose 10 so you can see how the dialplan executes on these calls. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Wednesday, May 27, 2009 11:23 AM To: asteris...@in-put.de; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Silly (??) question about chan_dahdi On Wed, May 27, 2009 at 12:19 PM, Stefan-Michael Guenther asteris...@in-put.de wrote: I have replaced the ISDN cable, the LED on the card is green, but still asterisk doesn't react to call to 8304478 and 8304479, while the ISDN phone does, No, they are not both connected at the same time. Please paste in your entire extensions.conf I'm not sure you understood what I told you on my last reply. Please also paste in the output of: cli dialplan show ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.0.9: Now Unable to create ... 'DAHDI'
Jared Smith wrote: On Wed, 2009-05-27 at 10:46 -0400, sean darcy wrote: -- Executing [646xxx...@longdistance:6] Dial(SIP/172-08276a60, DAHDI/g2/1646xxx) in new stack It appears you're attempting to dial DAHDI/g2/1646xxxyyy instead of DAHDI/g2/1646xxx... Did you mean to put those extra quotes in there? Of course not! They were off in a definition in extensions.ael( which seems to override the definitions in extensions.conf, BTW) TRUNK =DAHDI/g2 . Commented out the extensions.ael TRUNK definition, and we're good to go. Thanks for the great eyes. I owe you a beer, at the least. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-addon 1.6.1 problem
On Tuesday 26 May 2009 21:16:29 Rilawich Ango wrote: On Tue, May 26, 2009 at 10:33 PM, Tilghman Lesher wrote: On Tuesday 26 May 2009 02:52:18 Rilawich Ango wrote: I download asterisk-addon 1.6.1 but the VoIP phone failed to register to the system with the message below. [May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317 realtime_mysql: MySQL RealTime: Invalid database specified: asterisk [May 26 15:45:11] WARNING[29665]: res_config_mysql.c:317 realtime_mysql: MySQL RealTime: Invalid database specified: asterisk sip I use the same configuration file (res_mysql.conf extconfig.conf) in 1.6.0 but failed. Any big change in 1.6.1? Please read UPGRADE.txt in the asterisk-addons directory. I follow it to set [readhost.asterisk] and [writehost.asterisk] and extconfig.conf sippeers = mysql,readhost.asterisk/writehost.asterisk,sipfriends. However the error message still existed. Can you give me an example of res_mysql.conf and extconfig.conf? Using the default res_mysql.conf, you'd specify in extconfig.conf: sippeers = mysql,general,sipfriends since [general] is the default context name in res_mysql.conf. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Trunk groups
Hey all,
I have 2 GSM to Voip gateways and probably we will grow up to 4 more
gateways. I already created a macro to make failover happen between
gateways, but can imagine that everytime I add a new gateway I will need to
modify the macro. The initial intention of this macro was to failover
between different techonolgies.
So I was hoping to create a Sip Trunk group using the same idea as
truckgroup under dahdi but for sip trunks.
Is that possible?, have you ever done this before?
My Idea is:
sip_trunk1 = SIP/gateway1
sip_trunk2 = SIP/gateway2
sip_trunk3 = SIP/gateway3
gsm_trunkgoup = sip_trunk1 ; sip_trunk2 ; sip_trunk3
[user]
exten = _0.,1,wait()
exten = _0.,n,Dial(gsm_trunkgoup/${ exten:1},30)
exten = _0.,n,Hangup
Thanks,
--
--
*Mariano Lecuona*
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] setting CDR values on failed calls
Hi All, I am relatively new to Asterisk. I have CDR enabled and successfully writing to MS SQL server. In my cdr table I am setting the userfield value with a line in my dialplan. If a call is placed to an invalid number (e.g. 12125551212), I see a cdr record created, however, my userfield value never gets set since the call never made it into the context of my dialplan. I am using AMI with the Originate command to invoke the call. How can I set this value before the call is actually passed to my voip provider (of whom quickly responds with Got SIP response 500 'Service Unavailable' back from myVoipIPaddress) ? Thanks in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stucked calls in asterisk 1.4
David Backeberg schrieb: Now that I better understand your problem, I'm out of ideas. thats the point where i stand ;) You are correct that if a BYE sip packet gets lost, a) it won't get retransmitted if it's UDP b) the side that's waiting for the hangup will think the call is still active I've seen this in my system where the network switch went down while calls were active. The system where the call was happening caught the hangup, but the trunk system never got the bye as the network was down. My only questions are: Are you using a quality network switch, or maybe you can use a cross-over cable to eliminate collisions with other systems? all server are in one rack in our datacenter and are connected to an HP Procurve 2650 switch, which has been setup around 3 months ago, cause of the old switch died silent in the night. all server had two interfaces and i have allready tried to route the traffic between the pbx and the routing server over the second interface, where database requests normally run. But this didnt solved the problem too. i will try to increase the UDP buffer size in the linux kernel, maybe this will take some affect. best regards steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto-congesting call due to slow response
Hello, I'm running several asterisks in a carrier environment. The asterisks do mainly gateway business between E1 cards and IAX with some routing logic. On one key server I see issues of Auto-congesting call due to slow response coming every number of calls. The IAX peer is in the same subnet, the servers are not really loaded. Versions in use are 1.2.2 and 1.4.23-rc3, with rsa key authentication in use any ideas? kind regards -- Alexander Topolanek ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Playtones Volume
I've researched my brains out on this, and can't find any answer. Is there a way to adjust the level of the tones generated through the Playtones command? I'm thinking that I may have been approaching this incorrectly by targeting indications.conf since the tones are being called via the Playtones application. My sense is that it's not possible due to the lack of response from the way I was approaching the problem initially. Thanks again Lee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Domains
It might be worth clarifying what the question is, i'm pretty lost. Cheers Geraint 2009/5/27 Adrian Marsh adrian.ma...@ubiquisys.com Noone can give me a clue on this ? How Domains are used within Asterisk ? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Adrian Marsh *Sent:* 26 May 2009 12:14 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Domains Hi, I’m trying to understand an issue I’m seeing between two Asterisk servers. I think it has to do with Domain definitions. Server A), has extension 5550 defined. Has a sip client 2000 defined, and has guest-invites enabled. Server B), Dials to server A for any 5550 dialled. Has sip client 2000 and 2001 defined. If I register at server B as client 2001, and dial 5550 then the call works, and is placed through to server As logic successfully. But if I call in as client 2000, then the call fails, server A shows no log at all of the call (even a sip set debug ip ip showed nothing – though tcpdump did show the inbound invite). However if I remove the definition of client 2000 from server A, then the call succeeds. So I think that for a defined account server A is wanting to challenge for a password, even though the inbound call is not a local account – hence my trying now to understand if and how Asterisk uses Domains. If I define a serverA.company.com domain on server A, will it ignore the challenge for an INVITE coming from server B ?? Thanks Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silly (??) question about chan_dahdi
Hi, David Backeberg schrieb: Please paste in your entire extensions.conf I'm not sure you understood what I told you on my last reply. here it is: [general] static = yes writeprotect = no priorityjumping=yes [globals] [default] exten = 8304479,1,ANSWER() exten = 8304479,2,WAIT(10) exten = 8304479,3,HANGUP() exten = 8304478,1,ANSWER() exten = 8304478,2,WAIT(10) exten = 8304478,3,HANGUP() Re: exten = 83086921,1,ANSWER() exten = 83086921,2,WAIT(10) exten = 83086921,3,HANGUP() exten = 83086920,1,ANSWER() exten = 83086920,2,WAIT(10) exten = 83086920,3,HANGUP() Please also paste in the output of: cli dialplan show *CLI dialplan show [ Context 'default' created by 'pbx_config' ] '8304478' = 1. ANSWER() [pbx_config] 2. WAIT(10) [pbx_config] 3. HANGUP() [pbx_config] '8304479' = 1. ANSWER() [pbx_config] 2. WAIT(10) [pbx_config] 3. HANGUP() [pbx_config] '83086920' = 1. ANSWER() [pbx_config] 2. WAIT(10) [pbx_config] 3. HANGUP() [pbx_config] '83086921' = 1. ANSWER() [pbx_config] 2. WAIT(10) [pbx_config] 3. HANGUP() [pbx_config] [ Context 'parkedcalls' created by 'features' ] '700' = 1. Park() [features] [ Context 'app_queue_gosub_virtual_context' created by 'app_queue' ] 's' =1. NoOp() [app_queue] [ Context 'app_dial_gosub_virtual_context' created by 'app_dial' ] 's' =1. NoOp() [app_dial] -= 7 extensions (15 priorities) in 4 contexts. =- Thanks for your help support, Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] stucked calls in asterisk 1.4
On Wed, May 27, 2009 at 1:49 PM, Stefan Schmidt s...@sil.at wrote: all server are in one rack in our datacenter and are connected to an HP Procurve 2650 switch, which has been setup around 3 months ago, cause of the old switch died silent in the night. all server had two interfaces and i have allready tried to route the traffic between the pbx and the routing server over the second interface, where database requests normally run. But this didnt solved the problem too. i will try to increase the UDP buffer size in the linux kernel, maybe this will take some affect. I will say that asterisk-1.6 is supposed to have a better SIP stack than 1.4. Perhaps the difference in performance will help you. Specifically, check out: http://svn.digium.com/svn/asterisk/branches/1.6.0/CHANGES http://svn.digium.com/svn/asterisk/branches/1.6.1/CHANGES I'd recommend you go to at least 1.6.1.* series ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playtones Volume
On Wed, 2009-05-27 at 13:51 -0400, Lee Spenadel wrote: I’ve researched my brains out on this, and can’t find any answer. Is there a way to adjust the level of the tones generated through the Playtones command? The only thing I can think of is to use the VOLUME dialplan function before calling PlayTones() to decrease the volume on the Tx side, and then possibly restore it after calling StopPlayTones(). I haven't tested it to see if it works. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RLT Transfers and NI2
Hello Users, quick question for all you fine PRI users out there using RLT transfers. Scenario: 1) Inbound call received on channel one 2) Call is transferred to an agent on channel 2 3) RLT is enabled on this PRI, thus channels are released on the Asterisk box So far, all is fine, but our PRIs have 4 enabled RLT channels, thus after the 5th transfer, while the previous calls are still active, we get a busy (congestion), when we in fact have many PRI channels left. Again, this is a normal opration for RLT. My question to you is: Can we attempt a Bridged transfer instead of a RLT upon reception of a Congestion RLT result? Specifics: Asterisk 1.4.24.1 libpri 1.4.9 Allstream DS3 muxed out to 28 PRIs, b8zs, esf, NI2 signalling Hope this makes sense to one of you. Nic. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silly (??) question about chan_dahdi
Maybe this is an american thing, but why are your working lines 8 digits and your non-working 7 digits? Pardon if this was addressed earlier in the thread. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Wednesday, May 27, 2009 2:53 PM To: asteris...@in-put.de; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Silly (??) question about chan_dahdi On Wed, May 27, 2009 at 2:23 PM, Stefan-Michael Guenther asteris...@in-put.de wrote: *CLI dialplan show [ Context 'default' created by 'pbx_config' ] '8304478' = 1. ANSWER() [pbx_config] 2. WAIT(10) [pbx_config] 3. HANGUP() [pbx_config] '8304479' = 1. ANSWER() [pbx_config] 2. WAIT(10) [pbx_config] 3. HANGUP() [pbx_config] '83086920' = 1. ANSWER() [pbx_config] 2. WAIT(10) [pbx_config] 3. HANGUP() [pbx_config] '83086921' = 1. ANSWER() [pbx_config] 2. WAIT(10) [pbx_config] 3. HANGUP() [pbx_config] Your extensions look good. At this point I'm guessing there's something silly going on with your physical lines. Do you have an ordinary telephone for your country that you can plug in and confirm the lines work properly there. I think at this point you need help from somebody who knows line signaling for your country. Either your phone company or Digium tech support if you have a digium card. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silly (??) question about chan_dahdi
On Wed, May 27, 2009 at 2:23 PM, Stefan-Michael Guenther asteris...@in-put.de wrote: *CLI dialplan show [ Context 'default' created by 'pbx_config' ] '8304478' = 1. ANSWER() [pbx_config] 2. WAIT(10) [pbx_config] 3. HANGUP() [pbx_config] '8304479' = 1. ANSWER() [pbx_config] 2. WAIT(10) [pbx_config] 3. HANGUP() [pbx_config] '83086920' = 1. ANSWER() [pbx_config] 2. WAIT(10) [pbx_config] 3. HANGUP() [pbx_config] '83086921' = 1. ANSWER() [pbx_config] 2. WAIT(10) [pbx_config] 3. HANGUP() [pbx_config] Your extensions look good. At this point I'm guessing there's something silly going on with your physical lines. Do you have an ordinary telephone for your country that you can plug in and confirm the lines work properly there. I think at this point you need help from somebody who knows line signaling for your country. Either your phone company or Digium tech support if you have a digium card. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silly (??) question about chan_dahdi
asterisk -rvvv CLI pri debug span 3 place a call and catch the stuff off the screen you can also try pri intense debug span 3 if there's nothing showing up it's possible your number comes with 0 or so ... so you might want to do exten = _X.,1,BLAH instead of specifying the exact numberes Martin On Wed, May 27, 2009 at 11:19 AM, Stefan-Michael Guenther asteris...@in-put.de wrote: Hi, I have set context=default both in /etc/asterisk/dahdi-channels.conf and /etc/asterisk/chan_dahdi.conf, and created the necessary context with extens for both numbers I have replaced the ISDN cable, the LED on the card is green, but still asterisk doesn't react to call to 8304478 and 8304479, while the ISDN phone does, No, they are not both connected at the same time. Stefan David Backeberg schrieb: On Wed, May 27, 2009 at 3:55 AM, Stefan-Michael Guenther asteris...@in-put.de wrote: extensions.conf: - [incoming] exten = 8304479,1,ANSWER() exten = 8304479,2,WAIT(10) exten = 8304479,3,HANGUP() exten = 8304478,1,ANSWER() exten = 8304478,2,WAIT(10) exten = 8304478,3,HANGUP() Asterisk still doesn't pick up calls for these two numbers. Is that your entire extensions.conf? If so, this is part of your problem. If you do the square-bracket [] style contexts, you need to have jumps to them from the default context. So your plan should say extensions.conf [default] exten = 8304479,1,ANSWER() exten = 8304479,2,WAIT(10) exten = 8304479,3,HANGUP() exten = 8304478,1,ANSWER() exten = 8304478,2,WAIT(10) exten = 8304478,3,HANGUP() And you will need to cli dialplan reload to reload extensions.conf -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem with T.38 media headers
mario staphorst wrote: Thank you for your reply. I understand that Asterisk is not a SIP proxy, but shouldnt this header be passed on in order to provide proper T.38 passthrough support in this case? As far as i can see is this header really needed to make the T.38 connection successfull, when i setup the call directly to the ATA the reinvite is going fine. T.38 negotiation has already been improved in later releases than what you are using, so I'd suggest upgrading to 1.4.25 (or 1.4.26-rc1) before continuing, as it is possible that your issue has already been fixed. However, there are still areas we've identified where our T.38 negotiation needs some additional work, and we'll be trying to address those shortly. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Domains
I read through your question a couple of times. Basically you have server A which has extension 2000 and 5550. Server B has extension 2000 and 2001. You configure a (soft)phone as extension 2001 and dial 5550 which succeeds but you dial 2000 and the call fails.Have you tried turning up the debug verbosity in the console and watching the call flow on Server B? I don't know what would prompt Server B to try passing the call to Server A but that should become apparent in the debug information.If the 'domain' you are referring too his the FQDN then that has nothing to do with the price of bread as far as I can tell. Noone can give me a clue on this ? How Domains are used within Asterisk ?From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 26 May 2009 12:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Domains Hi, I’m trying to understand an issue I’m seeing between two Asterisk servers. I think it has to do with Domain definitions. Server A), has extension 5550 defined. Has a sip client 2000 defined, and has guest-invites enabled. Server B), Dials to server A for any 5550 dialled. Has sip client 2000 and 2001 defined. If I register at server B as client 2001, and dial 5550 then the call works, and is placed through to server As logic successfully. But if I call in as client 2000, then the call fails, server A shows no log at all of the call (even a sip set debug ip ip showed nothing – though tcpdump did show the inbound invite). However if I remove the definition of client 2000 from server A, then the call succeeds. So I think that for a defined account server A is wanting to challenge for a password, even though the inbound call is not a local account – hence my trying now to understand if and how Asterisk uses Domains. If I define a serverA.company.com domain on server A, will it ignore the challenge for an INVITE coming from server B ?? Thanks Adrian___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Trunk groups
AFAIK, unfortunatelly it's not the same as with ZAP channels where you can
group multiple lines together.
I ended up using slightly modified superdial macro:
http://www.voip-info.org/wiki/view/Superdial+macro.
if you add new gateway it's not necesarry to edit the macro, just add new
line in dialing context.
[out_via_superdial]
exten =
s,1,Macro(superdial,IAX2/voip1/${tfnumber}voip,1,yourname,8005551234,voipjet)
exten =
s,2,Macro(superdial,IAX2/alpeh-com/${tfnumber}voip,1,yourname,8005551234,aleph)
...
exten =
s,9,Macro(superdial,IAX2/orange/${tfnumber}voip,1,yourname,8005551234,orange)
On 5/27/09, Mariano Lecuona mlecu...@gmail.com wrote:
Hey all,
I have 2 GSM to Voip gateways and probably we will grow up to 4 more
gateways. I already created a macro to make failover happen between
gateways, but can imagine that everytime I add a new gateway I will need to
modify the macro. The initial intention of this macro was to failover
between different techonolgies.
So I was hoping to create a Sip Trunk group using the same idea as
truckgroup under dahdi but for sip trunks.
Is that possible?, have you ever done this before?
My Idea is:
sip_trunk1 = SIP/gateway1
sip_trunk2 = SIP/gateway2
sip_trunk3 = SIP/gateway3
gsm_trunkgoup = sip_trunk1 ; sip_trunk2 ; sip_trunk3
[user]
exten = _0.,1,wait()
exten = _0.,n,Dial(gsm_trunkgoup/${ exten:1},30)
exten = _0.,n,Hangup
Thanks,
--
--
*Mariano Lecuona*
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Mvh,
Aurimas Skirgaila
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Trunk groups
I've improved this since this revision, but now a days I don't use limited
systems. But my code has been used in places that need 100 concurrent
outgoing lines.
[macro-which-line]
exten = s,1,set(TRIES=0)
exten = s,n(nextone),set(TRIES=$[${TRIES} + 1]) ; increment TRIES by 1
exten = s,n,set(DIALSTRING=${TRY${TRIES}}) ; assign TRYn to DIALSTRING
exten = s,n,gotoif($[${DIALSTRING} = ]?donehere) ; see if we've run out
of things to try
exten = s,n,ChanIsAvail(${DIALSTRING}) ; it will be up or down, no need for
this to be exclusive
exten = s,n,gotoif($[${AVAILSTATUS} = 0]?:nextone)
exten = s,n,gotoif($[${GROUP_COUNT(${DIALSTRING})} = 2]?nextone) ; have we
used up the allowed calls on this channel
exten = s,n,set(GROUP()=${DIALSTRING}) ; Lock the line! Yay...
exten = s,n,Dial(${DIALSTRING}/1${ARG1}) ; dial the phone
exten = s,n,GotoIf($[${DIALSTATUS} = BUSY]?donehere) ; Don't keep
dialing
exten = s,n,NoOp(Moving to the next one...);
exten = s,n,goto(nextone) ; TEMP
exten = s,n(donehere),MacroExit() ; we only get here if everything failed
Then in GLOBALS you just set things like:
TRY0=SIP/trunk1
TRY1=SIP/trunk2
TRY3=SIP/other1
The above code is limited to 2 lines per channel. The code I used
originally (not sure where I found it anymore, might have been this mailing
list or might have been Voip-Info) support defining how many channels you
wanted to use for each provider (ie, provider1 has 2 lines free, but
provider2 has 5 lines). The original code didn't hold up though since if
multiple lines were being dialed at the exact same instance they would both
return the same availability before dialing the line. So in this one, I try
to lock the line early and if I get some other kind of error I move on to
the next group because I might have failed due to another race condition.
Anyways, tons of problems when you're limited on channels. Mine is the best
and one of a very few I've ever seen. SuperDial, I feel, is a silly idea.
It's exactly the same as a regular Dial string. No clue why you'd use it
over Dial. And the reason Dial doesn't work is because if the Dial'ed line
hangs up it returns back to the orginal Dial Plan. Doesn't help at all.
You hang up on the person, the person goes to the next line in the dial
plan, and you get called again. You hang up, they call you back again.
Soulds like a good way to use up air time.
Nicholas Blasgen
Partner / Network Operations
Refractive Dialer LLC
415.692-5277 (w)
408.497.9796 (c)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto-congesting call due to slow response
I'd look at the packet delay. Log some of the packets, see how long until you get a response from the remote host. If the delay is really long, then that's the issue (which by the response I assume that's exactly what's happening). Lower the load on the system and see if the delay improves. Or you can increase the timeout if you really wanted. channels/chan_iax2.c With debugging on, it seems that this data is available. But it's the same timeout as a Peer would be. And trust me, that timeout is huge. So that makes me think of another issue. I know with my VoIP provider, they told me not to trust the PEER POKE responses because I kept seeing my provider connect, disconnect, connect, disconnect. They had me turn off the qualification. (This is all SIP so I'm not sure how it translates to IAX). Might not want to waste the packets to send data to a server that is always available. If you don't get any help, you can try opening it as a bug on Digium's Bug Tracker but I assume the issue isn't a bug but just an overloaded system with a slow response time. Nicholas Blasgen Partner / Network Operations Refractive Dialer LLC 415.692-5277 (w) 408.497.9796 (c) Please update your contact records with my new work number. On Wed, May 27, 2009 at 10:52 AM, Alexander Topolanek at...@ocv.org wrote: Hello, I'm running several asterisks in a carrier environment. The asterisks do mainly gateway business between E1 cards and IAX with some routing logic. On one key server I see issues of Auto-congesting call due to slow response coming every number of calls. The IAX peer is in the same subnet, the servers are not really loaded. Versions in use are 1.2.2 and 1.4.23-rc3, with rsa key authentication in use any ideas? kind regards -- Alexander Topolanek ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting CDR values on failed calls
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate ActionId and Account can be set. Nicholas Blasgen Partner / Network Operations Refractive Dialer LLC 415.692-5277 (w) 408.497.9796 (c) Please update your contact records with my new work number. On Wed, May 27, 2009 at 10:24 AM, John Regal jre...@gmail.com wrote: Hi All, I am relatively new to Asterisk… I have CDR enabled and successfully writing to MS SQL server. In my cdr table I am setting the userfield value with a line in my dialplan. If a call is placed to an invalid number (e.g. 12125551212), I see a cdr record created, however, my userfield value never gets set since the call never made it into the context of my dialplan. I am using AMI with the Originate command to invoke the call. How can I set this value *before* the call is actually passed to my voip provider (of whom quickly responds with “Got SIP response 500 ‘Service Unavailable’ back from *myVoipIPaddress*”) ? Thanks in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
