Re: [asterisk-users] Using x-lite -Call failed 404 not found
Actually I registered two users in my X-lite. Both the users registered in different asterisk servers. While calling, first you have to right click on the x-lite and the click on the required server. Then make call. It will work. -Original Message- From: [EMAIL PROTECTED] on behalf of Vincent Sent: Mon 1/28/2008 12:40 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Using x-lite -Call failed 404 not found On Mon, 28 Jan 2008 12:01:43 +0530, [EMAIL PROTECTED] wrote: I have installed asterisk.When I start asterisk it starts normally and shows the status running. My partner also installed asterisk. I registered 1 user of her server and 1 user of my server in X-lite. I am able to call or receive call from the users registered in her server but not in my server. Its giving error call failed 404 not found I got the solution. Which was? ___ -- 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 Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com winmail.dat___ -- 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] Maybe a little OT---USB Handset
Hans Witvliet [EMAIL PROTECTED] writes: On Sun, 2008-01-27 at 20:19 +0100, Benny Amorsen wrote: You can't have a USB handset without a soft phone. You can get some which automatically run the phone software when plugged in, but that only works in Windows. How about a udev rule? Sorry, I didn't mean to imply that it was impossible to achieve the same in Linux. Just that the existing commercial solutions are targeting Windows. /Benny ___ -- 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] Maybe a little OT---USB Handset
Tzafrir Cohen [EMAIL PROTECTED] writes: Asterisk (chan_alsa, chan_oss, chan_console) is a soft phone. True, but deploying Asterisk to user PC's is not a particularly attractive option. /Benny ___ -- 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] mwi with sip
Hi, I am trying to utilize MWI with sip channel. when my client sens a SUBSCRIBE to asterisk I get info that user not found: - [Jan 28 11:49:02] --- (19 headers 0 lines) --- [Jan 28 11:49:02] Creating new subscription [Jan 28 11:49:02] Sending to 192.168.129.38 : 7060 (no NAT) [Jan 28 11:49:02] Found peer 'hellboy' [Jan 28 11:49:02] Looking for hellboy in routing-sip (domain ms.sip.rd.touk.pl) [Jan 28 11:49:02] --- Transmitting (no NAT) to 192.168.129.38:7060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.129.38:7060;branch=z9hG4bKadcf.1752dae4.0;received= 192.168.129.38 Via: SIP/2.0/UDP 192.168.0.165:7360;rport=7360;branch=z9hG4bKdxcekurc From: hellboy sip:[EMAIL PROTECTED];tag=qrrlr To: hellboy sip:[EMAIL PROTECTED];tag=as70810877 Call-ID: [EMAIL PROTECTED] CSeq: 968 SUBSCRIBE User-Agent: TouK S.K.A Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 does user have to be registered in asterisk I am using asterisk as media server but my users are registered at other sip proxy. Please point me what do I miss? Best regards tomasz ___ -- 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] Intercepting DTMF to initiate Voice Drop
On 1/25/08, Raj Jain [EMAIL PROTECTED] wrote: Hi, I'm trying to implement a Voice Drop service within Asterisk dial-plan. The service is supposed to work as following: 1. A initiates a call to B 2. The call is answered by B's answering machine 3. A hears the answering machine's greeting and the recording beep 4. A speaks a few words into the recording to personalize the message 5. A presses some DTMF keys (say, '##') to initiate Voice Drop 6. PBX intercepts DTMF and starts playing a prerecorded announcement to B 7. A is released from the call as soon as the Voice Drop is initiated 8. PBX releases the call to B at the end of the announcement To acheive this I need to intercept DTMF in the middle of a call and initiate an action based on that. I couldn't find an option in the Dial() application to break out of it on receipt of a particular DTMF sequence. Does the Dial() application support such a capability? I've tried the 'G' option in the Dial() application but that splits the call as soon as it is answered, whereas, I need to split the call after it is established based on a DTMF stimulus. Are there any other ways of accomplishing this goal? Any thoughts, ideas? Thank you, You should take a look at this: http://www.voip-info.org/wiki-Asterisk+config+features.conf See the applicationmap section. It should allow you to execute something upon keypress. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- 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] Monitoring audio on channel?
Hi all, I'm using asterisk to provide a simple service for official time (hour + minutes + seconds). The system and application (asterisk + zap detection + custom application) is monitored by Nagios with some scripts I have created using examples from voip-info.org. But I still need to monitor with Nagios if the audio is inserted in the voice channel successfully. What I'm looking for is detection of possible hardware problems with the digium card TE210P. Is there a way to do that? Best regards. Yahoo! Encuentros. Ahora encontrar pareja es mucho más fácil, probá el nuevo Yahoo! Encuentros http://yahoo.cupidovirtual.com/servlet/NewRegistration___ -- 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] unable to hear voice with asterisk 1.4.15
On Monday 28 January 2008 14:00:27 Rahul Yadav wrote: Hi all i am getting a serious problem.I am using asterisk 1.4.15 and dialing outbound through sip. The problem is that whenever i dial a number the other person can hear my voice but i dont hear anything. Have you tried: http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions Boyko ___ -- 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] unable to hear voice with asterisk 1.4.15
Hi all i am getting a serious problem.I am using asterisk 1.4.15 and dialing outbound through sip. The problem is that whenever i dial a number the other person can hear my voice but i dont hear anything. help me Thanks Rahul ___ -- 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] Peak number of calls?
I use mrtg,
I call this from MRTG: `/usr/local/groundwork/nagios/libexec/asterisk-mrtg.pl
-1 Zap -2 SIP`
Here is my asterisk-mrtg.pl: (note, I have tweaked this, but I do not know
where I had gotten my original reference)
---
#!/usr/bin/perl -w
use strict;
use IO::Socket;
use Getopt::Long;
$|=1;
my $host = 172.16.200.5;
my $username = changeduser;
my $password = changespass;
my (
$version, $response, $message, $line, $chan1, $chan2,
$verbose, $help, $command,
$warning, $critical,
%warnval, %critval,
%channels,
$sock,
$key,
$s,
$i,
);
my $stop = 0;
my $port = 5038;
my $exitcode = 0;
my $cause = ;
sub warning {
$s = shift;
$s =~ s/[\r\n]//g;
print WARNING: $s\n if ($verbose);
exit(1);
}
sub error {
$s = shift;
$s =~ s/[\r\n]//g;
print ERROR: $s\n if ($verbose);
exit(2);
}
sub unknown {
$s = shift;
$s =~ s/[\r\n]//g;
print UNKNOWN: $s\n if ($verbose);
exit(3);
}
sub syntax {
$s = shift;
unless ($s =~ m/Help:/) {
$s = Error: (.$s.) or $s = 'Unknown';
}
print $s\n unless ($help);
print Syntax: $0 -h host -u username -p password [-cwv]\n;
print * --username -u Username\n;
print * --password -p Password\n;
print * --host -h Host\n;
print --port -P n Port (if not using $port)\n;
print --chan1 -1 xxx Display channel xxx as 1.\n;
print --chan2 -2 xxx Display channel xxx as 2.\n;
print --verbose -vVerbose\n;
print --help -H This help\n;
exit(3);
}
Getopt::Long::Configure('bundling');
GetOptions
(p=s= \$password, password=s = \$password,
u=s= \$username, username=s = \$username,
h=s= \$host, host=s = \$host,
P=s= \$port, port=s = \$port,
H = \$help, help = \$help,
v = \$verbose, verbose= \$verbose,
chan1=s= \$chan1,1=s= \$chan1,
chan2=s= \$chan2,2=s= \$chan2);
syntax(Help:) if ($help);
syntax(Missing username) unless (defined($username));
syntax(Missing password) unless (defined($password));
syntax(Missing host) unless (defined($host));
syntax(Missing channels) if (!defined($chan1) or !defined($chan2));
if (defined($warning)) {
foreach $s (split(/,/, $warning)) {
syntax(Warning value given, $s, is invalid)
unless ($s =~ /^(\w+)=(\d+)$/);
$warnval{$1} = $2;
print Clear to give WARNING after $2 connections on $1\n if ($verbose);
}
}
if (defined($critical)) {
foreach $s (split(/,/, $critical)) {
syntax(Critical value given, $s, is invalid)
unless ($s =~ /^(\w+)=(\d+)$/);
$critval{$1} = $2;
print Clear to give CRITICAL after $2 connections on $1\n if ($verbose);
}
}
unless ($sock = IO::Socket::INET-new(PeerAddr = $host, PeerPort = $port,
Proto = 'tcp')) {
print(Could not connect to asterisk server .$host.:.$port.\n) if
($verbose);
exit(2);
}
$version = $sock;
print $version if ($verbose);
print $sock Action: Login\r\nUsername: $username\r\nSecret:
$password\r\nEvents: off\r\n\r\n;
print Action: Login\r\nUsername: $username\r\nSecret: $password\r\n\r\n if
($verbose);
$response = $sock;
$message = $sock;
$s = $sock;
print $response.$message if ($verbose);
print $s if ($verbose);
exit(1) unless ($response =~ m/^Response:\s+(.*)$/i);
exit(1) unless ($1 =~ m/Success/i);
print $sock Action: Status\r\n\r\n;
print Action: Status\r\n\r\n if ($verbose);
$response = $sock;
$message = $sock;
print $response.$message if ($verbose);
unknown(Unknown answer $response (wanted Response: something)) unless
($response =~ m/^Response:\s+(.*)$/i);
unknown($response didn't say Success) unless ($1 =~ m/Success/i);
unknown(Unknown answer $response (wanted Message: something)) unless
($message =~ m/^Message:\s+(.*)$/i);
unknown(didn't understand message $message) unless ($1 =~ m/Channel status
will follow/i);
$stop=0;
while (($stop == 0) ($line = $sock)) {
print $line if ($verbose);
if ($line =~ m/Channel:\s+(\w+)\//) {
$channels{$1}++;
print Found $1 channel\n if ($verbose);
}
if ($line =~ m/Event:\s*StatusComplete/i) {
$stop++;
}
}
# Log out
print $sock Action: Logoff\r\n\r\n;
undef($s);
for ($i=0;$i2;$i++) {
if (defined($channels{$chan1})) {
print $channels{$chan1} . \n;
} else {
print 0\n;
}
if (defined($channels{$chan2})) {
print $channels{$chan2} . \n;
} else {
print 0\n;
}
}
---
my $username should be a user in the asterisk management file.
my $password should be the password for that account.
This gives me this output:
`Daily' Graph (5 Minute Average)
Max
Average
Current
Zap channels
5 Zap channels in use
1 Zap channels in use
0 Zap channels in use
SIP channels
6 SIP channels in use
1 SIP channels in use
0 SIP channels in use
--
--
Steven
http://www.connectech.org/
Gordon Henderson [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Is there any way to find-out the peak number of calls that an asterisk
system has
[asterisk-users] Dial agent channel - busy
Hi, when I'm trying to call the following extension exten = 6002,1,Verbose(1|Extension 6002) exten = 6002,n,Dial(Agent/6002) exten = 6002,n,Hangup() the call is terminated and I get the following warning from asterisk: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Agent' (cause 17 - User busy) When calling the agent with Dial(SIP/6002) no problem occurs. What could be wrong? Some additional information about the configuration: The asterisk version is 1.4.10 - In users.conf I defined a user 6002: [6002] fullname = Test Agent email = [EMAIL PROTECTED] secret = 1234 zapchan = 1 hasvoicemail = yes vmsecret = 1234 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = international host=dynamic - In agents.conf I added the agent agent = 6002,1234,Test Agent - and in queues.conf I added a queue testQueue2: [testQueue2] music=default strategy=ringall timeout=15 retry=5 wrapuptime=0 maxlen = 0 announce-frequency = 0 announce-holdtime = no member = Agent/6002 servicelevel = 60 - Thanks a lot, Thomas -- Thomas Kenner ___ -- 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] Compatibility Issues with dell poweredge195and TE110P card
zttest: --- Results after 44 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.988350 Dell 2950 Wildcard TE410P (2nd Gen) (rev 02) (with echo cancel daughter board) 1 PRI configured to Telco 1 PRI configured to old Panasonic DBS 576 being used just as a mux for our fax machines. -- -- Steven http://www.connectech.org/ broadband Voice [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] How did this workout? I am considering getting the Dell PowerEdge 2950. On 11/5/07, Steven [EMAIL PROTECTED] wrote: 2950s work fine. I have had the parity error for over a year with no noticable problems. It is working fine. I did have to make some IRQ changes to clean up the system. I did these on my Dell 1750 test machine, but have made the same changes on my production machine. The changes basically redue the IRQ load from other cards, like the RAID card, which will reduce the bus's capacity for processing all of the TDM IRQs. It also allocates just one CPU full time for all of the TDM IRQs. The changes are below: ref: FYI on zttool output on SMP system --- Results after 56 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.999564 Only 2 were 99.987793, the 54 others were all 100.00. I got this by making the changes below on my dual proc Dell 1750. setpci -v -s 01:08.1 LATENCY_TIMER=8 setpci -v -s 00:0f.1 LATENCY_TIMER=8 setpci -v -s 01:04.0 LATENCY_TIMER=8 setpci -v -s 01:02.0 LATENCY_TIMER=8 setpci -v -s 00:0f.2 LATENCY_TIMER=8 setpci -v -s 01:04.0 LATENCY_TIMER=8 (these are USB, SCSI HW RAID driver, Ethernet, Video, etc. I did not alter ZAP cards, nor any bridges or buses) echo 1 /proc/irq/17/smp_affinity (Ethernet) echo 1 /proc/irq/18/smp_affinity (SCSI HW RAID Driver) echo 2 /proc/irq/20/smp_affinity (TDM) echo 2 /proc/irq/24/smp_affinity (TE411P) I also turned of the startup of irqbalance. The setpci changes did the most work concerning reaching 100% in zttest. Irqbalance was causing the the processor handling the interrupts of the zap cards to change very often. This would impose a delay during the change and cause the zttest numbers to drop/be inconsistent. Because I turned irqbalance off, the irqs are processed round robin style, which is also not good. Therefore, I hard coded the processor affinity for the zap cards to one proc and all other high load irqs to the other proc. If you have more than 2 procs, you can spread them out even more. If you do not turn off irqbalance, the affinity changes will be overwritten by it. I made these changes on a live system without issue. I set these changes in /etc/rc.d/rc.local to reset them after reboots. -- -- Steven http://www.glimasoutheast.org Brian Hutchinson [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] You guys are scaring me! I'm building a 2950 with SAS RAID 5 on the new PERC and it will have 2 TE420P's. I hope it works or my bacon will fry. On 10/25/07, Joseph Begumisa [EMAIL PROTECTED] wrote: Has anyone had any compatibility issues with a TE110P card installed on a Dell Poweredge 1950?I noted the following error on the LCD display of the Dell Poweredge 1950: E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0. Yes, I have had this problem with a dell PE1650, 1850, SC1400, and PE650. I have a TE410P that does it. It may not be wise, but I just ignore the orange blinking LCD display (or light, depending on the model). I did try reseating the card, and it works for a few weeks, and then goes back to the same old thing. Yes, that happened too. Digium has graciously offered to send me a TE120P with the Digium VoiceBus technology which I will test out on the Dell 1950. Will post my findings thereafter. Joseph. ___ --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 ___ --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 ___ -- 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] Dial agent channel - busy
On 1/28/08, Thomas Kenner [EMAIL PROTECTED] wrote: Hi, when I'm trying to call the following extension exten = 6002,1,Verbose(1|Extension 6002) exten = 6002,n,Dial(Agent/6002) exten = 6002,n,Hangup() the call is terminated and I get the following warning from asterisk: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Agent' (cause 17 - User busy) When calling the agent with Dial(SIP/6002) no problem occurs. What could be wrong? I never got this working, not sure why (wiki states that it should work). However Agent channel is considered obsolete - because of locking problems. You should consider using Local channels with GROUP_COUNT, and if you're using call queues, you would want to use this backported patch from 1.6. http://lists.digium.com/pipermail/asterisk-dev/2008-January/031545.html Regards, Atis Some additional information about the configuration: The asterisk version is 1.4.10 - In users.conf I defined a user 6002: [6002] fullname = Test Agent email = [EMAIL PROTECTED] secret = 1234 zapchan = 1 hasvoicemail = yes vmsecret = 1234 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = international host=dynamic - In agents.conf I added the agent agent = 6002,1234,Test Agent - and in queues.conf I added a queue testQueue2: [testQueue2] music=default strategy=ringall timeout=15 retry=5 wrapuptime=0 maxlen = 0 announce-frequency = 0 announce-holdtime = no member = Agent/6002 servicelevel = 60 - Thanks a lot, Thomas -- Thomas Kenner ___ -- 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 -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- 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 and MRTG, a little help please...
JR, That script runs fine. You should be able to run it first manually, if so please copy and paste the error. Thanks, Alejandro, After reading the sparse info and attempting to get this running, I'm unsuccessful and could use some guidance. I already have a MRTG server up and running serving hundreds of router interface graphs. I would like to add SIP/IAX channel graphs for all our asterisk servers. I'm running asterisk 1.2 and MRTG 2.4.17. I tried the script from http://karlsbakk.net/asterisk/ but get errors that MRTG does not recognize the first few lines, so I think I'm running into MRTG version compatibility issues. Can anyone send me the MRTG scripts that may work with this setup? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] Asterisk and MRTG, a little help please...
After reading the sparse info and attempting to get this running, I'm unsuccessful and could use some guidance. I already have a MRTG server up and running serving hundreds of router interface graphs. I would like to add SIP/IAX channel graphs for all our asterisk servers. I'm running asterisk 1.2 and MRTG 2.4.17. I tried the script from http://karlsbakk.net/asterisk/ but get errors that MRTG does not recognize the first few lines, so I think I'm running into MRTG version compatibility issues. Can anyone send me the MRTG scripts that may work with this setup? I have the exact same script from that web page and it runs fine on Asterisk 1.2 and MRTG 2.9.29 (default RH ES3.0 install). I also have other servers on RH9 and MRTG 2.10.13 running without issue as well. What errors are you getting? Unknown option: h Unknown option: 1 Unknown option: 2 ERROR: Line 3 (use strict;) in CFG file (10.10.14.102.cfg) does not make sense I named the script file the IP address of the server.cfg instead of asterisk-mrtg. I call the script from the command line: # env LANG=C /usr/bin/mrtg 10.10.14.102.cfg -h 10.10.14.102 -1 SIP -2 IAX2 JR -- JR Richardson Engineering for the Masses ___ -- 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] zaptel and oslec
On Mon, Jan 28, 2008 at 04:37:25PM -0200, Pablo Allietti wrote: hi all i have a te110p installed in my system with a lot of Echo.. i decide to install the oslec echo supressor but when y try to add the module i have this problem. [EMAIL PROTECTED] zaptel-1.4.7.1]# insmod wct1xxp.ko insmod: error inserting 'wct1xxp.ko': -1 Unknown symbol in module [EMAIL PROTECTED] zaptel-1.4.7.1]# That's because there's a module missing. Why not just use modprobe? If you actually want to figure out what module it was: dmesg| tail (and it is probably either oslec or zaptel) Or, if you used insmod because you want modules from your working directory, then please help me improve http://svn.digium.com/svn/zaptel/branches/1.4/zap_auto -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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] zaptel and oslec
hi all i have a te110p installed in my system with a lot of Echo.. i decide to install the oslec echo supressor but when y try to add the module i have this problem. [EMAIL PROTECTED] zaptel-1.4.7.1]# insmod wct1xxp.ko insmod: error inserting 'wct1xxp.ko': -1 Unknown symbol in module [EMAIL PROTECTED] zaptel-1.4.7.1]# some advice? thanks. -- .- ___ -- 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] Peak number of calls?
On Thu, 24 Jan 2008, Steven wrote: I use mrtg, I call this from MRTG: `/usr/local/groundwork/nagios/libexec/asterisk-mrtg.pl -1 Zap -2 SIP` Here is my asterisk-mrtg.pl: (note, I have tweaked this, but I do not know where I had gotten my original reference) Hi Steven, The original location of this is: http://karlsbakk.net/asterisk/scripts/asterisk-mrtg The down-side is that mrtg only samples every 5 minutes, and as there is no running total, just an instantaneous count in gauge mode, it has the opportunity to miss calls made when it's not sampling. So this is OK for busy sites where a few calls either way won't be noticed, but I'm dealing with relatively low call volumes of 1-2 a minute, with very occasional bursts to (maybe, I don't know yet!) 4-6 simultaneous calls. (I have a customer who has 4 ISDN2e ports; 8 channels) and they want to know if they can lose one port and save some money) I'm working on something slightly better that will poll it more often, but still provide an mrtg interface. I'll post details here, but it's not that high a priority right now. Cheers, Gordon `Daily' Graph (5 Minute Average) Max Average Current Zap channels 5 Zap channels in use 1 Zap channels in use 0 Zap channels in use SIP channels 6 SIP channels in use 1 SIP channels in use 0 SIP channels in use -- -- Steven http://www.connectech.org/ Gordon Henderson [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Is there any way to find-out the peak number of calls that an asterisk system has had? Not the total number of calls, but the maximum number of simultaneous calls. I know I can porobably go through the CDR logs and look for calls which have overlapped in time, but I'm wondering if there's some counter somewhere I could access... (I'm looking for evidence for an ISDN client who wants to know if he's spent too much on the number of ISDN lines he has installed!) Cheers, Gordon ___ -- 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] Shut down one Zap line
On Monday 28 January 2008 14:10, Steve Totaro wrote: On Jan 28, 2008 1:33 PM, Ron Joffe [EMAIL PROTECTED] wrote: I have a system with 2 TE220B (2xPRI). I am looking for a method to shutdown one of the 4 zap lines. Something like a ifconfig eth3 down command. It should be the equivalent of physically unplugging one PRI circuit. With or without reloading or restarting Asterisk? Without reloading or restarting Asterisk. I think if you are actually trying to bring down the PRI without stopping changing configs and starting Asterisk, there is no way currently. That is what I am attempting. If you just want to take them out of use but not down, you could write a shell script to change your dialplan and put the PRIs in different groups. Lastly, outside of Asterisk you could use a CSU/DSU that supports telnet or even an iBoot or other web controlled power outlet device to turn it on or off. iBoot works well but I prefer this unit http://www.controlbyweb.com/webswitch/index.html because it is priced better and has two outlets (plus you can mount it easily on a wall board). I have been using both products for various things and they work flawlessly. Steve, I think you might have given me an idea there. Thanks, Ron ___ -- 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] Shut down one Zap line
I have a system with 2 TE220B (2xPRI). I am looking for a method to shutdown one of the 4 zap lines. Something like a ifconfig eth3 down command. It should be the equivalent of physically unplugging one PRI circuit. ___ -- 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 and MRTG, a little help please...
JR Richardson wrote:
6. Re: That script runs fine. You should be able to run it first
manually, if so
please copy and paste the error.
mrtg:/var/www/mrtg# env LANG=C /usr/bin/mrtg
2008-01-28 11:16:01: WARNING: Could not get any data from external
command '/var/mrtg/10.10.14.102.cfg -h 10.10.14.102 -1 SIP -2 IAX2'
Maybe the external command did not even start. (Illegal seek)
2008-01-28 11:16:01: WARNING: Problem with External get
'/var/mrtg/10.10.14.102.cfg -h 10.10.14.102 -1 SIP -2 IAX2':
Expected a Number for 'in' but nothing'
2008-01-28 11:16:01: WARNING: Problem with External get
'/var/mrtg/10.10.14.102.cfg -h 10.10.14.102 -1 SIP -2 IAX2':
Expected a Number for 'out' but nothing'
2008-01-28 11:16:01: ERROR: Target[asterisklab2][_IN_] '
$target-[0]{$mode} ' did not eval into defined data
2008-01-28 11:16:01: ERROR: Target[asterisklab2][_OUT_] '
$target-[0]{$mode} ' did not eval into defined data
ntcp-mrtg:/var/www/mrtg#
I can see the script log into the manager interface on the asterisk
server at 10.10.14.102, and there are active SIP channels during
script execution.
Any ideas?
You need to take a step back and first test the script without using
MRTG. Execute it like this:
# /opt/bin/asterisk-mrtg -h localhost -u XXX -p -1 SIP -2 Zap
10
10
10
10
You should get 4 lines of numbers. That respresents your SIP and Zap
channels. Once you get past this step go back and plug it into your
MRTG config.
Andres
http://www.neuroredes.com
Thanks.
JR
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Re: [asterisk-users] Asterisk and MRTG, a little help please...
6. Re: That script runs fine. You should be able to run it first
manually, if so
please copy and paste the error.
mrtg:/var/www/mrtg# env LANG=C /usr/bin/mrtg
2008-01-28 11:16:01: WARNING: Could not get any data from external
command '/var/mrtg/10.10.14.102.cfg -h 10.10.14.102 -1 SIP -2 IAX2'
Maybe the external command did not even start. (Illegal seek)
2008-01-28 11:16:01: WARNING: Problem with External get
'/var/mrtg/10.10.14.102.cfg -h 10.10.14.102 -1 SIP -2 IAX2':
Expected a Number for 'in' but nothing'
2008-01-28 11:16:01: WARNING: Problem with External get
'/var/mrtg/10.10.14.102.cfg -h 10.10.14.102 -1 SIP -2 IAX2':
Expected a Number for 'out' but nothing'
2008-01-28 11:16:01: ERROR: Target[asterisklab2][_IN_] '
$target-[0]{$mode} ' did not eval into defined data
2008-01-28 11:16:01: ERROR: Target[asterisklab2][_OUT_] '
$target-[0]{$mode} ' did not eval into defined data
ntcp-mrtg:/var/www/mrtg#
I can see the script log into the manager interface on the asterisk
server at 10.10.14.102, and there are active SIP channels during
script execution.
Any ideas?
Thanks.
JR
--
JR Richardson
Engineering for the Masses
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Re: [asterisk-users] zaptel and oslec
this is the output of dmesg [EMAIL PROTECTED] zaptel-1.4.7.1]# dmesg |tail wct1xxp: disagrees about version of symbol zt_transmit wct1xxp: Unknown symbol zt_transmit wct1xxp: disagrees about version of symbol zt_rbsbits wct1xxp: Unknown symbol zt_rbsbits wct1xxp: disagrees about version of symbol zt_unregister wct1xxp: Unknown symbol zt_unregister wct1xxp: disagrees about version of symbol zt_register wct1xxp: Unknown symbol zt_register wct1xxp: disagrees about version of symbol zt_alarm_notify wct1xxp: Unknown symbol zt_alarm_notify [EMAIL PROTECTED] zaptel-1.4.7.1]# Tzafrir Cohen wrote: On Mon, Jan 28, 2008 at 04:37:25PM -0200, Pablo Allietti wrote: hi all i have a te110p installed in my system with a lot of Echo.. i decide to install the oslec echo supressor but when y try to add the module i have this problem. [EMAIL PROTECTED] zaptel-1.4.7.1]# insmod wct1xxp.ko insmod: error inserting 'wct1xxp.ko': -1 Unknown symbol in module [EMAIL PROTECTED] zaptel-1.4.7.1]# That's because there's a module missing. Why not just use modprobe? If you actually want to figure out what module it was: dmesg| tail (and it is probably either oslec or zaptel) Or, if you used insmod because you want modules from your working directory, then please help me improve http://svn.digium.com/svn/zaptel/branches/1.4/zap_auto -- .- Pablo Allietti E-mail: [EMAIL PROTECTED] | LACNIC Phone : +598 2 604 | http://LACNIC.NET ___ -- 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] zaptel and oslec
this is the output of dmesg [EMAIL PROTECTED] zaptel-1.4.7.1]# dmesg |tail wct1xxp: disagrees about version of symbol zt_transmit wct1xxp: Unknown symbol zt_transmit wct1xxp: disagrees about version of symbol zt_rbsbits wct1xxp: Unknown symbol zt_rbsbits wct1xxp: disagrees about version of symbol zt_unregister wct1xxp: Unknown symbol zt_unregister wct1xxp: disagrees about version of symbol zt_register wct1xxp: Unknown symbol zt_register wct1xxp: disagrees about version of symbol zt_alarm_notify wct1xxp: Unknown symbol zt_alarm_notify [EMAIL PROTECTED] zaptel-1.4.7.1]# Tzafrir Cohen wrote: On Mon, Jan 28, 2008 at 04:37:25PM -0200, Pablo Allietti wrote: hi all i have a te110p installed in my system with a lot of Echo.. i decide to install the oslec echo supressor but when y try to add the module i have this problem. [EMAIL PROTECTED] zaptel-1.4.7.1]# insmod wct1xxp.ko insmod: error inserting 'wct1xxp.ko': -1 Unknown symbol in module [EMAIL PROTECTED] zaptel-1.4.7.1]# That's because there's a module missing. Why not just use modprobe? If you actually want to figure out what module it was: dmesg| tail (and it is probably either oslec or zaptel) Or, if you used insmod because you want modules from your working directory, then please help me improve http://svn.digium.com/svn/zaptel/branches/1.4/zap_auto -- .- Pablo Allietti E-mail: [EMAIL PROTECTED] | LACNIC Phone : +598 2 604 | http://LACNIC.NET ___ -- 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 and MRTG, a little help please...
Unknown option: h Unknown option: 1 Unknown option: 2 ERROR: Line 3 (use strict;) in CFG file (10.10.14.102.cfg) does not make sense I named the script file the IP address of the server.cfg instead of asterisk-mrtg. I call the script from the command line: # env LANG=C /usr/bin/mrtg 10.10.14.102.cfg -h 10.10.14.102 -1 SIP -2 IAX2 The problem is that is not how its supposed to work. The file is just a script that you need to run from withing a regular mrtg.cfg file. So first you build a regular mrtg.cfg file and test it out with something you are used to like Ethernet Traffic or whatever. Then you add something like this at the bottom: Title[servername]: Server title PageTop[servername]: h1servername.domain.com/h1 Target[servername]: `/usr/local/bin/asterisk-mrtg -h servername.domain.com -1 SIP -2 IAX2` Options[servername]: gauge,integer MaxBytes[servername]: 90 YLegend[servername]: Active channels ...You see the Target Line...Thats where the perl script from http://karlsbakk.net/asterisk/ goes. It is not an MRTG config file as you have tried to use it. Andres http://www.neuroredes.com JR ___ -- 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] Shut down one Zap line
On Jan 28, 2008 1:33 PM, Ron Joffe [EMAIL PROTECTED] wrote: I have a system with 2 TE220B (2xPRI). I am looking for a method to shutdown one of the 4 zap lines. Something like a ifconfig eth3 down command. It should be the equivalent of physically unplugging one PRI circuit. With or without reloading or restarting Asterisk? I think if you are actually trying to bring down the PRI without stopping changing configs and starting Asterisk, there is no way currently. If you just want to take them out of use but not down, you could write a shell script to change your dialplan and put the PRIs in different groups. Lastly, outside of Asterisk you could use a CSU/DSU that supports telnet or even an iBoot or other web controlled power outlet device to turn it on or off. iBoot works well but I prefer this unit http://www.controlbyweb.com/webswitch/index.html because it is priced better and has two outlets (plus you can mount it easily on a wall board). I have been using both products for various things and they work flawlessly. Thanks, Steve Totaro ___ -- 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] Shut down one Zap line
I have a system with 2 TE220B (2xPRI). I am looking for a method to shutdown one of the 4 zap lines. Something like a ifconfig eth3 down command. It should be the equivalent of physically unplugging one PRI circuit. ___ -- 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] Loosing user's registration with asterisk as no-root
Hello list, hope some one could help me find the answer. Asterisk 1.4.16.2 installd as no-root user The main issue is that every now and then, cd * box seems to loose the user's registrations, there is nothing in the console, absolutely no messages, only when another friend trys to dial an extension I can see this on messages logs [Jan 26 10:35:46] WARNING[23015] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) And in the asterisk console the call is inmidiatly redirected to user's voicemail When this happens, I issue a reload command and everything gets back to normal. This is wired, before I was using the same version installed as root, and this behaviour wasn't present. Any adias? Thanks in advace! ___ -- 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] ISDN Internal Bus?
Hello there - I have an AVM C4 card, which has 4 controllers. I am able to dial from outside to SIP phones - but not to ISDN phones. (an from SIP to the outside world) I want to have Asterisk betwen the PSTN and the internal ISDN phones. So, controller 1 of the c4 card is connected to the NTBA - and I plugged a phone directly into controller 2. Please do not say Duu - but I thought that was how it worked. this seems to be wrong. Is the C4 card unable to power the internal bus? would I need to make a (cheap) internal bus if this is not the way to do it? Álainn * The cheese-mites asked how the cheese got there, And warmly debated the matter; The orthodox said that it came from the air, And the heretics said from the platter. Anon. ___ -- 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 DTMF Troubleshoot
How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
messages related to DTMF... or if I just do a global SIP debug for
that matter I am using RFC DTMF but it's not being passed to the
PSTN and I need to debug this further. I've tried to increase the
verbosity and the debug ('set debug n') and that didn't help either. I
assume this is because even RFC2833 sends the DTMF as RTP which
wouldn't show up anyways but how to troubleshoot DTMF issues?
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Re: [asterisk-users] zaptel and oslec
On Mon, Jan 28, 2008 at 05:07:54PM -0200, Pablo Allietti wrote: this is the output of dmesg [EMAIL PROTECTED] zaptel-1.4.7.1]# dmesg |tail wct1xxp: disagrees about version of symbol zt_transmit wct1xxp: Unknown symbol zt_transmit wct1xxp: disagrees about version of symbol zt_rbsbits wct1xxp: Unknown symbol zt_rbsbits wct1xxp: disagrees about version of symbol zt_unregister wct1xxp: Unknown symbol zt_unregister wct1xxp: disagrees about version of symbol zt_register wct1xxp: Unknown symbol zt_register wct1xxp: disagrees about version of symbol zt_alarm_notify wct1xxp: Unknown symbol zt_alarm_notify [EMAIL PROTECTED] zaptel-1.4.7.1]# module zaptel not loaded. Again, the simplest thing is: # If you haven't done so already: #make install # and then: modprobe wct1xxp -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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] SIP DTMF Troubleshoot
On Mon, 2008-01-28 at 18:05 -0500, Andrew Joakimsen wrote:
How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
messages related to DTMF... or if I just do a global SIP debug for
that matter I am using RFC DTMF but it's not being passed to the
PSTN and I need to debug this further. I've tried to increase the
verbosity and the debug ('set debug n') and that didn't help either. I
assume this is because even RFC2833 sends the DTMF as RTP which
wouldn't show up anyways but how to troubleshoot DTMF issues?
I'd first turn on rtp debug and see if that helps. If that's not
enough information, I'd go into logger.conf and add dtmf to the logger
and messages lines (and any others you care about), and then do a
logger reload from the Asterisk CLI.
--
Jared Smith
Community Relations Manager
Digium, Inc.
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Re: [asterisk-users] Asterisk and IPv6
On Thu, 2007-06-28 at 22:37 -0500, Russell Bryant wrote: Bent Bagger wrote: When will these additions make their way into the Asterisk mainstream It has not yet been merged into the main development tree, but I'm sure it will be before Asterisk 1.6 is released. Any progress on IPv6 ? Still completely seperate code, or is it already being merged into the tree... Perhaps i overlooked it, but i couldn't find any reference in: http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/CHANGES?view=co HtH, Hans ___ -- 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] IAX Calls - One Way Audio
Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for the whole organization, answering calls that come in for both locations. We have a problem where some calls (seemingly randomly) appear to get one way audio. This only happens for inbound calls off the PSTN, if they follow this pattern (which is a fair number of calls): Call comes in from PSTN to site A, gets put into a queue to be answered by receptionist as site B. Receptionist answers the call, and then puts the call on hold to perform an attended transfer to an extension at site A. (The call from the receptionist to the extension is OK). When the receptionist hits the 'transfer' button to actually transfer the call, the original caller cannot hear anything. The internal extension can hear the caller OK. This problem does not occur on every call. Since the issue has risen its head, I have enabled core, sip and iax debugging, but I am of yet unable to get the issue to occur on its own, to have a good look at the log files. FYI, I have disabled the asterisk Dial Commands in FreePBX, to solve another issue (where call audio bounces between the servers for a call that is transferred between sites and back again). Both servers are asterisk version 1.2.23, freepbx version 2.3.1.0. I have posted the contents of the iax.conf file below (which is identical on both servers). If there is any further information I can provide, please let me know and I can get this information. [general] disallow=all allow=g729 mailboxdetail=yes jitterbuffer=no ;maxjitterbuffer=500 ;jittershrinkrate=1 bandwidth=low tos=lowdelay trunk=yes notransfer=yes #include iax_general_custom.conf #include iax_registrations_custom.conf #include iax_registrations.conf #include iax_custom.conf #include iax_additional.conf Any suggestions are very welcome. Regards, Daniel ___ -- 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 DTMF Troubleshoot
Too much info then too little info.
Basically the issue is the provider this happens even when we send
them the calls in IAX because they talk SIP to the same gateway.
I just need to prove it to these people. Anyone have any DTMF issues
between Asterisk and a Quintum gateway?
On Jan 28, 2008 6:47 PM, Jared Smith [EMAIL PROTECTED] wrote:
On Mon, 2008-01-28 at 18:05 -0500, Andrew Joakimsen wrote:
How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
messages related to DTMF... or if I just do a global SIP debug for
that matter I am using RFC DTMF but it's not being passed to the
PSTN and I need to debug this further. I've tried to increase the
verbosity and the debug ('set debug n') and that didn't help either. I
assume this is because even RFC2833 sends the DTMF as RTP which
wouldn't show up anyways but how to troubleshoot DTMF issues?
I'd first turn on rtp debug and see if that helps. If that's not
enough information, I'd go into logger.conf and add dtmf to the logger
and messages lines (and any others you care about), and then do a
logger reload from the Asterisk CLI.
--
Jared Smith
Community Relations Manager
Digium, Inc.
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[asterisk-users] MFC/R2
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Followed the instructions at http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 I dead end at patching the channels Makefile. There have been some changes since these instructions were made. I added the chan_unicall.c to the channels folder but asterisk doesnt pick it up added chan_unicall.o to the Makefile and asterisk pukes anyone have instructions for building in to 1.4.8+ Thanks - -- James Finstrom Rhino Equipment Corp. Tel: 1-800-785-7073 ext. 6344 FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ext 6344 FWD: 633686 ext 6344 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHnm1edloC7YyaIOoRAje2AJwPlRqj4rl+guvJVm2O25Et6TYkpgCePDYy w66jmc+wLO6zF6G5Tjz5hcc= =yR6F -END PGP SIGNATURE- ___ -- 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 DTMF Troubleshoot
I think your best bet is to do a packet capture and look for RTP packets
with an RTP Event payload (rtpevent display filter).
On Mon, 28 Jan 2008, Andrew Joakimsen wrote:
How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
messages related to DTMF... or if I just do a global SIP debug for
that matter I am using RFC DTMF but it's not being passed to the
PSTN and I need to debug this further. I've tried to increase the
verbosity and the debug ('set debug n') and that didn't help either. I
assume this is because even RFC2833 sends the DTMF as RTP which
wouldn't show up anyways but how to troubleshoot DTMF issues?
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--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671
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Re: [asterisk-users] MFC/R2
On Mon, 2008-01-28 at 17:03 -0700, James Finstrom wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Followed the instructions at http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 I dead end at patching the channels Makefile. There have been some changes since these instructions were made. I added the chan_unicall.c to the channels folder but asterisk doesnt pick it up added chan_unicall.o to the Makefile and asterisk pukes anyone have instructions for building in to 1.4.8+ Thanks The only change that needs to be made to the asterisk/channels/Makefile is: chan_h323.so: chan_h323.o h323/libchanh323.a $(ECHO_PREFIX) echo[LD] $^ - $@ $(CMD_PREFIX) $(CXX) $(PTHREAD_CFLAGS) $(ASTLDFLAGS) $(SOLINK) -o $@ $ h323/libchanh323.a $(CHANH323LIB) -L$(PWLIBDIR)/lib $(PTLIB) -L $(OPENH323DIR)/lib $(H323LIB) -L/usr/lib -lcrypto -lssl -lexpat endif -- chan_unicall.o: chan_unicall.c $(ECHO_PREFIX) echo [LD] $^ - $@ $(CMD_PREFIX) $(CC) -I../include -c $(CFLAGS) -D_GNU_SOURCE -DAST_MODULE=\$*\ -o chan_unicall.o chan_unicall.c chan_unicall.so: chan_unicall.o $(CMD_PREFIX) $(CC) $(SOLINK) -o $@ $ -lunicall -lxml2 -lsupertone -lspandsp -ltiff $(ZAPLIB) - chan_misdn.o: ASTCFLAGS+=-Imisdn Just insert what is between the dashes (the other lines are there for reference so you know where to insert the rest. That is it. Asterisk should now compile chan_unicall. I would recommend that you go to http://www.moythreads.com/astunicall/ in order to get the newest support for Unicall on Asterisk 1.4.x - -- James Finstrom Rhino Equipment Corp. Tel: 1-800-785-7073 ext. 6344 FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ext 6344 FWD: 633686 ext 6344 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHnm1edloC7YyaIOoRAje2AJwPlRqj4rl+guvJVm2O25Et6TYkpgCePDYy w66jmc+wLO6zF6G5Tjz5hcc= =yR6F -END PGP SIGNATURE- ___ -- 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 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- 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] IAX Calls - One Way Audio
Does turning off the notransfer help? I would imagine that dropping the second server out of the equation might be useful, and save some bandwidth. PaulH On Tue, 2008-01-29 at 10:38 +1100, Daniel Cole wrote: Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for the whole organization, answering calls that come in for both locations. We have a problem where some calls (seemingly randomly) appear to get one way audio. This only happens for inbound calls off the PSTN, if they follow this pattern (which is a fair number of calls): Call comes in from PSTN to site A, gets put into a queue to be answered by receptionist as site B. Receptionist answers the call, and then puts the call on hold to perform an attended transfer to an extension at site A. (The call from the receptionist to the extension is OK). When the receptionist hits the 'transfer' button to actually transfer the call, the original caller cannot hear anything. The internal extension can hear the caller OK. This problem does not occur on every call. Since the issue has risen its head, I have enabled core, sip and iax debugging, but I am of yet unable to get the issue to occur on its own, to have a good look at the log files. FYI, I have disabled the asterisk Dial Commands in FreePBX, to solve another issue (where call audio bounces between the servers for a call that is transferred between sites and back again). Both servers are asterisk version 1.2.23, freepbx version 2.3.1.0. I have posted the contents of the iax.conf file below (which is identical on both servers). If there is any further information I can provide, please let me know and I can get this information. [general] disallow=all allow=g729 mailboxdetail=yes jitterbuffer=no ;maxjitterbuffer=500 ;jittershrinkrate=1 bandwidth=low tos=lowdelay trunk=yes notransfer=yes #include iax_general_custom.conf #include iax_registrations_custom.conf #include iax_registrations.conf #include iax_custom.conf #include iax_additional.conf Any suggestions are very welcome. Regards, Daniel ___ -- 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] Asterisk and IPv6
Hans Witvliet wrote: Any progress on IPv6 ? Still completely seperate code, or is it already being merged into the tree... Perhaps i overlooked it, but i couldn't find any reference in: http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/CHANGES?view=co There has been progress. It is not yet merged into the main tree, though. I would expect it to go in within the first few releases of 1.6 ... -- Russell Bryant Senior Software Engineer Open Source Team Lead 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] Asterisk 1.6.0-beta2 and 1.4.18-rc1 Now Available
The Asterisk development team has released versions 1.6.0-beta2 and and 1.4.18-rc1. The new beta for 1.6 is available for download from http://downloads.digium.com/. The release candidate for 1.4.18 is only available via svn. It is available for anyone that would like to help test 1.4.18 over the next couple of days before it gets officially released. To download the 1.4.18 release candidate: $ svn co http://svn.digium.com/svn/asterisk/tags/1.4.18 1.4.18-rc1 To make a tarball out of the previous checkout, do: $ svn export 1.4.18-rc1 asterisk-1.4.18-rc1 $ rm -rf 1.4.18-rc1 $ tar -czvf asterisk-1.4.18-rc1.tar.gz asterisk-1.4.18-rc1/ Please report any issues to http://bugs.digium.com/. Thank you for your support! ___ -- 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] GSM SIM Cards and Digium, or GSM SIM Adaptor
Try cyber-telecom.net May be get a X100P with a CT-G1000 or G2000 Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Tuesday, January 15, 2008 3:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] GSM SIM Cards and Digium, or GSM SIM Adaptor On Mon, 14 Jan 2008, Steve Totaro wrote: On Jan 14, 2008 8:43 AM, Gordon Henderson [EMAIL PROTECTED] wrote: On Mon, 14 Jan 2008, bilal ghayyad wrote: Hi List; Is there an Digium cards support GSM SIM cards so we can fix an SIM card to be used for calls within mobiles as it is less rate? Or I have to use an FXS to SIM adaptor? If yes, then anyone advise a models and prices? A few solutions - If you want to go down the analogue port route, then this: http://www.discountphonesystems.co.uk/acatalog/Analogue_GSM_Gateway.html works well. If you want to stay in the SIP/digital domain, then there are these too: http://www.discountphonesystems.co.uk/acatalog/IP-GSM-Gateway.html There are PCI cards which take 1-4 SIM cards, but I've not heard much about them in the news recently, and last time I enquired, their price was somewhat substantial... (However the lack of additional wiring, mains plugs, etc. required may well be an advantage in their favour) I'd presonally avoid any sort of bluetooth solution in anything resembling a commercial environment. Give me a bit of wire, anyday! Cheers, Gordon http://lists.digium.com/mailman/listinfo/asterisk-users Gordon, I am not sure if it has been included in chan_mobile yet but I remember talk about adding direct USB connectivity to the code. Anyways, GSM is wireless so I'd assume you would avoid this whole scenario. Actually - er - a-ha - indeed... However the GSM part if it all is sort of unavoidable !!! But I'd rather have a wire going from an asterisk box to a phone device than use an additional wireless connection... I'm not sure I'd get away with installing a box in a customers permises, then plug in a USB/Bluetooth dongle, and then just leave a mobile-phone cable-tied to the rack... At home/lab, sure. It's fun and neat, but not in the office where even a USB cable from the server to the phone might be frowned upon. (The phone would get nicked!) Cheers, Gordon ___ -- 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] Dial agent channel - busy
What does 'show agents' give you? 'show queues' would be useful too. PaulH On Mon, 2008-01-28 at 13:36 +0100, Thomas Kenner wrote: Hi, when I'm trying to call the following extension exten = 6002,1,Verbose(1|Extension 6002) exten = 6002,n,Dial(Agent/6002) exten = 6002,n,Hangup() the call is terminated and I get the following warning from asterisk: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Agent' (cause 17 - User busy) When calling the agent with Dial(SIP/6002) no problem occurs. What could be wrong? Some additional information about the configuration: The asterisk version is 1.4.10 - In users.conf I defined a user 6002: [6002] fullname = Test Agent email = [EMAIL PROTECTED] secret = 1234 zapchan = 1 hasvoicemail = yes vmsecret = 1234 hassip = yes hasiax = no hash323 = no hasmanager = no callwaiting = no context = international host=dynamic - In agents.conf I added the agent agent = 6002,1234,Test Agent - and in queues.conf I added a queue testQueue2: [testQueue2] music=default strategy=ringall timeout=15 retry=5 wrapuptime=0 maxlen = 0 announce-frequency = 0 announce-holdtime = no member = Agent/6002 servicelevel = 60 - Thanks a lot, Thomas ___ -- 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 GSM
Try cyber-telecom.net May be get a X100P with a CT-G1000 or G2000 Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Sunday, January 20, 2008 11:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP GSM I'd like to add a device to my Asterisk server to leverage my cellular account. Does anyone on-list have experience with hardware gateways vs using cah_bluetooth and an old cell phone? I'm considering something like http://www.mobigater.com/index.php?p=5 Thanks, Michael -- Michael Graves mgravesatmstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- 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] GSM SIM Cards and Digium, or GSM SIM Adaptor
It is more economical to get a hardware GSM Gateway from places like cyber-telecom.net and then plug it in a X100P Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad Sent: Monday, January 14, 2008 8:43 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] GSM SIM Cards and Digium, or GSM SIM Adaptor Hi List; Is there an Digium cards support GSM SIM cards so we can fix an SIM card to be used for calls within mobiles as it is less rate? Or I have to use an FXS to SIM adaptor? If yes, then anyone advise a models and prices? Regards Bilal Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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 GSM
On Jan 20, 2008 4:40 PM, Michael Graves [EMAIL PROTECTED] wrote: I'd like to add a device to my Asterisk server to leverage my cellular account. Does anyone on-list have experience with hardware gateways vs using cah_bluetooth and an old cell phone? We use the Junghanns.NET duoGSM PCI card with the bristuff driver. Did not have any problems with it yet, works as expected. Regards, Geert ___ -- 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 GSM
On Tue, Jan 29, 2008 at 09:12:11AM +0800, Sam Tam wrote: Try cyber-telecom.net May be get a X100P with a CT-G1000 or G2000 a) this should be on the biz list b) why don't you post from your cyber-telecom.net address? c) it must be the end of the sales cycle and trying to get a bit more revenue in? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- 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 GSM
With that sort of set up, If for example i get a 8 channel GSM gateway and the X100P can i make more than 1 concurrent call though the gateway with the X100P or does it only support 1 call at a time? What im looking to do is get a multi channel GSM gateway, and have the ability to make more than 1 call at once through it. Thanks -Kev Sam Tam wrote: Try cyber-telecom.net May be get a X100P with a CT-G1000 or G2000 Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Sunday, January 20, 2008 11:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP GSM I'd like to add a device to my Asterisk server to leverage my cellular account. Does anyone on-list have experience with hardware gateways vs using cah_bluetooth and an old cell phone? I'm considering something like http://www.mobigater.com/index.php?p=5 Thanks, Michael -- Michael Graves mgravesatmstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- 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 -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, and is believed to be clean. ___ -- 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 GSM
Not to be out of topic but gmail is really good at managing mailing list that why I use gmail. Also it can filter out spam and my cyber-telecom website is not very good at doing that.. I am trying to offer a solution to a person who want a solution. And if that is a bit too much then next time may be I should keep my month shut.. But hey I though a mailing list is trying to get other users helping each other. There is no way I cannot see my post being non constructive .. Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: Tuesday, January 29, 2008 9:44 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP GSM On Tue, Jan 29, 2008 at 09:12:11AM +0800, Sam Tam wrote: Try cyber-telecom.net May be get a X100P with a CT-G1000 or G2000 a) this should be on the biz list b) why don't you post from your cyber-telecom.net address? c) it must be the end of the sales cycle and trying to get a bit more revenue in? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- 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] Asterisk and MRTG, a little help please...
You need to take a step back and first test the script without using MRTG. Execute it like this: # /opt/bin/asterisk-mrtg -h localhost -u XXX -p -1 SIP -2 Zap 10 10 10 10 You should get 4 lines of numbers. That respresents your SIP and Zap channels. Once you get past this step go back and plug it into your MRTG config. I run the script without using mrtg but I don't get any reply: mrtg:# /var/mrtg/asterisk-mrtg -h [ipaddress] -u [user] -p [pass] -1 SIP -2 IAX2 mrtg:# I can see the script log into the manager on the asterisk server, but doesn't return any figures. i haven't modified anything in the script. I'm wondering if the script is OK and maybe Asterisk manager is not setup correctly or asterisk is not returning proper value. Here is what I have configured in manager.conf: [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 [user] secret = pass deny=0.0.0.0/0.0.0.0 permit=[subnet of mrtg server] read = system,call,log,verbose,command,agent,user Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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 GSM
Hello Kev There are 2 solutions for this actually there are more but I don't want to make things too complicate this time. 1. get a 8 ports fxo card for asterisk. There are ebay and many other on the asterisk-biz list that sell them or me of course then get a 8 ports gsm gateway and plug those in. Then they are setup just like a normal phone line Try to look at the voip wiki for x100p config 2. get a 8 ports fxo voip gateway and 8 ports gsm gateway then put those together. They end up working the same way. Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kev S Sent: Tuesday, January 29, 2008 9:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP GSM With that sort of set up, If for example i get a 8 channel GSM gateway and the X100P can i make more than 1 concurrent call though the gateway with the X100P or does it only support 1 call at a time? What im looking to do is get a multi channel GSM gateway, and have the ability to make more than 1 call at once through it. Thanks -Kev Sam Tam wrote: Try cyber-telecom.net May be get a X100P with a CT-G1000 or G2000 Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Sunday, January 20, 2008 11:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP GSM I'd like to add a device to my Asterisk server to leverage my cellular account. Does anyone on-list have experience with hardware gateways vs using cah_bluetooth and an old cell phone? I'm considering something like http://www.mobigater.com/index.php?p=5 Thanks, Michael -- Michael Graves mgravesatmstvp.com blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- 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 -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, and is believed to be clean. ___ -- 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] Asterisk and MRTG, a little help please...WORKING
On 1/28/08, JR Richardson [EMAIL PROTECTED] wrote: You need to take a step back and first test the script without using MRTG. Execute it like this: # /opt/bin/asterisk-mrtg -h localhost -u XXX -p -1 SIP -2 Zap 10 10 10 10 You should get 4 lines of numbers. That respresents your SIP and Zap channels. Once you get past this step go back and plug it into your MRTG config. I run the script without using mrtg but I don't get any reply: mrtg:# /var/mrtg/asterisk-mrtg -h [ipaddress] -u [user] -p [pass] -1 SIP -2 IAX2 mrtg:# I can see the script log into the manager on the asterisk server, but doesn't return any figures. i haven't modified anything in the script. I'm wondering if the script is OK and maybe Asterisk manager is not setup correctly or asterisk is not returning proper value. Here is what I have configured in manager.conf: [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 [user] secret = pass deny=0.0.0.0/0.0.0.0 permit=[subnet of mrtg server] read = system,call,log,verbose,command,agent,user I added write = system,call,log,verbose,command,agent,user to manager.conf and things started working. I did not realize write was needed for the script to poll and get a response. no more errors and I get the proper channel count. Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] IAX Calls - One Way Audio
Why not give the receptionist a two line phone? Register one line on server 1 and the other on server 2. Then the bounce back and forth goes away saving bandwidth. Lyle Daniel Cole wrote: Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for the whole organization, answering calls that come in for both locations. We have a problem where some calls (seemingly randomly) appear to get one way audio. This only happens for inbound calls off the PSTN, if they follow this pattern (which is a fair number of calls): Call comes in from PSTN to site A, gets put into a queue to be answered by receptionist as site B. Receptionist answers the call, and then puts the call on hold to perform an attended transfer to an extension at site A. (The call from the receptionist to the extension is OK). When the receptionist hits the 'transfer' button to actually transfer the call, the original caller cannot hear anything. The internal extension can hear the caller OK. This problem does not occur on every call. Since the issue has risen its head, I have enabled core, sip and iax debugging, but I am of yet unable to get the issue to occur on its own, to have a good look at the log files. FYI, I have disabled the asterisk Dial Commands in FreePBX, to solve another issue (where call audio bounces between the servers for a call that is transferred between sites and back again). Both servers are asterisk version 1.2.23, freepbx version 2.3.1.0. I have posted the contents of the iax.conf file below (which is identical on both servers). If there is any further information I can provide, please let me know and I can get this information. [general] disallow=all allow=g729 mailboxdetail=yes jitterbuffer=no ;maxjitterbuffer=500 ;jittershrinkrate=1 bandwidth=low tos=lowdelay trunk=yes notransfer=yes #include iax_general_custom.conf #include iax_registrations_custom.conf #include iax_registrations.conf #include iax_custom.conf #include iax_additional.conf Any suggestions are very welcome. Regards, Daniel ___ -- 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
[asterisk-users] Dialogic card
Hi list, Anyone knows where I can get information about configuring a Dialogic card to run with Asterisk?? The model I have is D/120JCT-LS. Somebody told me that I had to buy the driver, but I don't know if this is true and if so, who, how and how much... ___ -- 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 GSM
On Tue, Jan 29, 2008 at 10:44:36AM +0800, Sam Tam wrote: Not to be out of topic but gmail is really good at managing mailing list that why I use gmail. Also it can filter out spam and my cyber-telecom website is not very good at doing that.. I am trying to offer a solution to a person who want a solution. And if that is a bit too much then next time may be I should keep my month shut.. But hey I though a mailing list is trying to get other users helping each other. Right. Only your advice is not exactly impartial, as the mail may suggest. Even when we want to be impartial, we aren't really. And the readers should be well aware of that. There is no way I cannot see my post being non constructive .. Sam If your email address cannot convey that information, you could use your signature to provide that information. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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] Asterisk's DANGEROUS Transfer CDR's
Hi All, PLEASE READ if you depend on Asterisk CDR's and support transfers. Apologies for the shout but I'm desperate to get others to agree Asterisk has a big problem with the CDR's that are generated for transfers. I can understand why not too many people are interested as transfers are complicated and messy. However for those of us having to support transfers and depending on Asterisk CDR's for our billing we are in a sticky predicament! For anyone using Asterisk in a provider environment unaware of any problem I urge you to do a simple blind transfer on your system and check your CDR's. Most Asterisk based providers I tested are blocking transfers but I did find some other providers out there missing billable call legs! My goal is to try and get acknowledgement that there is a serious problem here that warrants a re-think about how Asterisk CDR's are generated. In an effort to succinctly encapsulate the problem I've produced the call and CDR flows below. Hopefully they make sense but if not I'm more than happy to elaborate and share my test results (the flows below won't be legibile without a mono spaced font, copy and pasting into notepad will make them readable). Blind Transfer (1.2 and 1.4): Time CallsCDRs | Dest | Dur(s) | |---|| T0 -| Alice -- * -- Bob | || | | || Tt -| Carol -- * -- Bob -| Bob | Tt | | | || Te -| End -| Carol | Te | Attended Transfer (1.2): Time CallsCDRs | Dest | Dur(s) | |---|| T0 -| Alice -- * -- Bob | || | | || T1 -| Alice -- * -- Carol | || | | || Tt -| Carol -- * -- Bob -| Bob | Tt | | | Carol | Tt - T1| | | s | Tt | | | || Te -| End -| s | Te | Attended Transfer (1.4): Time CallsCDRs | Dest | Dur(s) | |---|| T0 -| Alice -- * -- Bob | || | | || T1 -| Alice -- * -- Carol | || | | || Tt -| Carol -- * -- Bob -| || | | || Te -| End -| Bob | Te | | Bob | Te - T1| To put it another way here are some examples of how Asterisk systems and transfers can be exploited. 1. Place a call to a mobile you plan on having a lengthy call to. As soon as the call is establised blind transfer it to a low or free cost destination. You will only be billed for the mobile call up to the time it takes you to do the transfer the remainder of the call will be billed at the low cost or free destination. 2. With Asterisk 1.4 place a call to two billable destinations and then transfer them together. You'll only be billed for each destination up until the time it takes you to transfer. 3. With Asterisk 1.2 place a call to a low cost or free destination. Then place a call to an expensive destination and do an attended transfer. You'll only be billed for the expensive destination up unitl the time it takes to do the transfer. I have opened a bug on the issue but I suspect without input from others having the same problem it will fade away. http://bugs.digium.com/view.php?id=11849 From my point of view the design solution to this problem would be as simple as changing the CDR generation from one CDR per bridge to generating a CDR for each end of a bridge. When the end of a bridge changes or the bridge is hungup a CDR(s) would be generated. The implementation would undoubtedly be a lot more difficult but if the design could be agreed upon at least those of us in between a rock and a hard place on this could decide to sponsor development, offer a bounty etc. Regards, Greyman. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail ___ -- 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] MFC/R2
Hi, Do you know http://www.moythreads.com/astunicall/ ??? Luis A P Barbosa 2008/1/28, James Finstrom [EMAIL PROTECTED]: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Followed the instructions at http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 I dead end at patching the channels Makefile. There have been some changes since these instructions were made. I added the chan_unicall.c to the channels folder but asterisk doesnt pick it up added chan_unicall.o to the Makefile and asterisk pukes anyone have instructions for building in to 1.4.8+ Thanks - -- James Finstrom Rhino Equipment Corp. Tel: 1-800-785-7073 ext. 6344 FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ext 6344 FWD: 633686 ext 6344 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHnm1edloC7YyaIOoRAje2AJwPlRqj4rl+guvJVm2O25Et6TYkpgCePDYy w66jmc+wLO6zF6G5Tjz5hcc= =yR6F -END PGP SIGNATURE- ___ -- 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
[asterisk-users] Asterisk mem leak behavior?
So here is my setup. Hardware: Intel P3 1.2 Ghz 1 GB RAM 36 GB Drives Mirrored Software: CentOS 5 2.6.18 Kernel Asterisk 1.4.14 Zaptel 1.4.7 (redfone) LIbpri 1.4.2 I'm using TDMoE with my PRI using a product called fonebridge from a company called redfone. They require that I use their own build of zaptel and I am trying to figure out if the problem is with them or something else. The TDMoE traffic is running over a dedicated interface on the asterisk server. Nothing but TDMoE traffic goes over this interface, it does not even have an IP assigned to it. After a about 6-7 days asterisk will stop passing calls through the PRI. It will continue to accept calls from extension to extension or to voicemail, etc. But nothing over zap channels. From the CLI if I check the status of the channels it shows all is well. If I use zttool it shows all OKs. And the lights on the fonebridge itself are green across the board as well. If I issue the command from the CLI to restart asterisk restart now it drops me to another CLI prompt as if it ran the command, but it didn't asterisk continues to run. It takes me dropping out of the CLI and issueing an init command to restart /etc/init.d/asterisk restart. Once asterisk restarts all is well and things keep moving, but 5 or so days later the same thing happens. What's the best way to trouble shoot this? Any and all comments are welcome. Let me know if you need more details. - Mark ___ -- 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] IAX Calls - One Way Audio
Thanks Paul and Lyle for the suggestions. I would like to keep the phones configuration to one line for now, and see if I can solve the problem rather then just work around it. I have changed he notransfer option, will see what happens over the next few days. Thanks again for the suggestions, any further input is very much welcome. Many Thanks, Daniel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Tuesday, 29 January 2008 11:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Calls - One Way Audio Does turning off the notransfer help? I would imagine that dropping the second server out of the equation might be useful, and save some bandwidth. PaulH On Tue, 2008-01-29 at 10:38 +1100, Daniel Cole wrote: Hello List, I am currently having a bit of a strange issue with a pair of asterisk servers that we recently set up. For a bit of background, this particular business has two sites in two different towns, about 10 minutes apart. They have 3 analogue PSTN lines connected to the asterisk servers at each location, via a Sangoma A200 (with HEC). They are trying to have just the one receptionist for the whole organization, answering calls that come in for both locations. We have a problem where some calls (seemingly randomly) appear to get one way audio. This only happens for inbound calls off the PSTN, if they follow this pattern (which is a fair number of calls): Call comes in from PSTN to site A, gets put into a queue to be answered by receptionist as site B. Receptionist answers the call, and then puts the call on hold to perform an attended transfer to an extension at site A. (The call from the receptionist to the extension is OK). When the receptionist hits the 'transfer' button to actually transfer the call, the original caller cannot hear anything. The internal extension can hear the caller OK. This problem does not occur on every call. Since the issue has risen its head, I have enabled core, sip and iax debugging, but I am of yet unable to get the issue to occur on its own, to have a good look at the log files. FYI, I have disabled the asterisk Dial Commands in FreePBX, to solve another issue (where call audio bounces between the servers for a call that is transferred between sites and back again). Both servers are asterisk version 1.2.23, freepbx version 2.3.1.0. I have posted the contents of the iax.conf file below (which is identical on both servers). If there is any further information I can provide, please let me know and I can get this information. [general] disallow=all allow=g729 mailboxdetail=yes jitterbuffer=no ;maxjitterbuffer=500 ;jittershrinkrate=1 bandwidth=low tos=lowdelay trunk=yes notransfer=yes #include iax_general_custom.conf #include iax_registrations_custom.conf #include iax_registrations.conf #include iax_custom.conf #include iax_additional.conf Any suggestions are very welcome. Regards, Daniel ___ -- 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 ___ -- 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 mem leak behavior?
What if you issue restart now and then ctrl-c or ctrl-d out of Asterisk? IMO TMDoE support is very legacy I don't think its really been maintained since the 1.0 builds. On Jan 29, 2008 1:24 AM, Mark Greene [EMAIL PROTECTED] wrote: So here is my setup. Hardware: Intel P3 1.2 Ghz 1 GB RAM 36 GB Drives Mirrored Software: CentOS 5 2.6.18 Kernel Asterisk 1.4.14 Zaptel 1.4.7 (redfone) LIbpri 1.4.2 I'm using TDMoE with my PRI using a product called fonebridge from a company called redfone. They require that I use their own build of zaptel and I am trying to figure out if the problem is with them or something else. The TDMoE traffic is running over a dedicated interface on the asterisk server. Nothing but TDMoE traffic goes over this interface, it does not even have an IP assigned to it. After a about 6-7 days asterisk will stop passing calls through the PRI. It will continue to accept calls from extension to extension or to voicemail, etc. But nothing over zap channels. From the CLI if I check the status of the channels it shows all is well. If I use zttool it shows all OKs. And the lights on the fonebridge itself are green across the board as well. If I issue the command from the CLI to restart asterisk restart now it drops me to another CLI prompt as if it ran the command, but it didn't asterisk continues to run. It takes me dropping out of the CLI and issueing an init command to restart /etc/init.d/asterisk restart. Once asterisk restarts all is well and things keep moving, but 5 or so days later the same thing happens. What's the best way to trouble shoot this? Any and all comments are welcome. Let me know if you need more details. - Mark ___ -- 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
[asterisk-users] Installation of gatekeeper-H323plus
Hi, I am trying to install Gatekeeper. I have installed pwlib and trying to install h323 plus. I have set the path as PWLIBDIR=/root/pwlib export PWLIBDIR OPENH323DIR=/root/open323 export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH I also configured /etc/ld.so.conf include /root/pwlib/lib include /root/h323plus/lib Then I tried running ldconfig. But it gave AVC access denied Then I gave # ./configure #make opt Its giving error. /usr/bin/ld: cannot find -lpt_linux_x86_r collect2: ld returned 1 exit status make[1]: *** [/root/h323plus/lib/libh323_linux_x86_r.so.1.19-beta7] Error 1 make[1]: Leaving directory `/root/h323plus/src' make: *** [opt] Error 2 Please help me out. Thanking you, Preeta Pandey Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- 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] Compatibility Issues with dell poweredge 195and TE110P card
broadband Voice ha scritto: Irqbalance was causing the the processor handling the interrupts of the zap cards to change very often. This would impose a delay during the change and cause the zttest numbers to drop/be inconsistent. Irqbalance is a good idea BUT some kernel (and some driver) is not so happy to the irq affinity change. The right idea is to disable IRQBALANCE, to assign at boot che irq to one physical CPU (remember HT and multicore CPU share the same IO). And also, some Linux kernel is so bad that freeze with irqbalance running. :-) Bye. begin:vcard fn:Massimo Nuvoli n:Nuvoli;Massimo org:Progetto Archivio SRL adr:;;Via Giustetto 75;Abbadia Alpina Pinerolo;TO;10060;Italia email;internet:[EMAIL PROTECTED] title:Amministratore Delegato tel;work:0121303544 tel;fax:0121040601 x-mozilla-html:FALSE url:www.progettoarchivio.com version:2.1 end:vcard signature.asc Description: OpenPGP digital signature ___ -- 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] Compatibility Issues with dell poweredge195and TE110P card
Steven ha scritto: zttest: --- Results after 44 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.988350 Dell 2950 Wildcard TE410P (2nd Gen) (rev 02) (with echo cancel daughter board) 1 PRI configured to Telco 1 PRI configured to old Panasonic DBS 576 being used just as a mux for our fax machines. I had a lot of problems with the similar 2850, the issue i found is the irq of the PCI slot shared with one of the eth onboard. Every slot share his irq with a device onboard... After a lot of test i bougt a Sangoma card, all problems solved. Now the system is 5 PRI (1 4PRI and 1 1PRI) all working ok (more than 400.000 channel/year). The only Digium board is a TDM400 that, i think, works for some strange miracle... 1) Dell is a strange Company, there is not support for my Debian 4.0 distro (even with gold warranty), so no help from us. 2) Digium tell if you use strange hardware and/or shared irq. 3) Sangoma The card works with shared IRQ. All is true, but only one is a solution. :-) Bye. P.s. You can change the server but the sharing of the irq is very common. begin:vcard fn:Massimo Nuvoli n:Nuvoli;Massimo org:Progetto Archivio SRL adr:;;Via Giustetto 75;Abbadia Alpina Pinerolo;TO;10060;Italia email;internet:[EMAIL PROTECTED] title:Amministratore Delegato tel;work:0121303544 tel;fax:0121040601 x-mozilla-html:FALSE url:www.progettoarchivio.com version:2.1 end:vcard signature.asc Description: OpenPGP digital signature ___ -- 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] Quad core is not a good idea! (was: Asterisk on Dell PowerEdge 2950)
broadband Voice ha scritto: Does anyone have any compatibilty issues with Dell *PowerEdge^TM 2950 III 2-Socket, Quad-Core 2U*? I plan on using this with the Digium T1 cards. Thanks. Consider 2 socket dual core CPU with more mhz, Asterisk is more IO than computation. The quad core CPU is lower in MHZ and all the CPUs share the same IO. The architecture design of the XEON board is wrong for really IO intensive apps, consider stepping up to a AMD arch. Also the 2950 is a 2U server that use IRQ sharing (WHY?), wrong with boards like the T1 Digium.. Bye. begin:vcard fn:Massimo Nuvoli n:Nuvoli;Massimo org:Progetto Archivio SRL adr:;;Via Giustetto 75;Abbadia Alpina Pinerolo;TO;10060;Italia email;internet:[EMAIL PROTECTED] title:Amministratore Delegato tel;work:0121303544 tel;fax:0121040601 x-mozilla-html:FALSE url:www.progettoarchivio.com version:2.1 end:vcard signature.asc Description: OpenPGP digital signature ___ -- 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
