Re: [asterisk-users] Asterisk Crash in asterisk 10.0.0.
According to https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions Asterisk 10 was EOLd on 2013-12-15 and has been on security fix only for a year before that. If you find the bug and figure out how to fix it, the fix will never be released because Asterisk 10 is EOLd.Take a look at the Asterisk 11 changelog, numerous fixes for seg faults. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arun Ram Sent: Friday, February 14, 2014 12:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Crash in asterisk 10.0.0. Hi Eric Wieling, Thanks for your reply. what is the reason for that crash?? . when i read the core dump i found something like signal 11. what it means because of signal 11 asterisk crashed . Before upgrading i need to submit a report to my team for that i need a valid reason for that crash. .: NOTES INFORMATION :. ### found note section at offset: 0x4294 ### --- note 0 at offset 0x4294 --- padding: 4 bytes note name size: 0x5 bytes note description size: 0x90 bytes note name: CORE note type: PRSTATUS [1] signal number: 11 extra code: 0 errno: 0 current signal: 11 set of pending signals: 0 set of held signals:0 pid:5136 ppid: 5136 pgrp: 2770 sid:2101 user time: 0.32994 sec system time:0.26995 sec cumulative user time: 0.0 sec cumulative system time: 0.0 sec bool pr_fpvalid:1 On Fri, Feb 14, 2014 at 10:57 AM, Eric Wieling ewiel...@nyigc.com wrote: Upgrade to 11. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N Sent: Friday, February 14, 2014 12:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Crash in asterisk 10.0.0. Enable debugging module and backtrace and re-compile so that you will bactrace of the crash logs. Regards On 14 Feb 2014 10:29, Arun Ram arunram@gmail.com wrote: Hi guys, I need a desperate help from you regarding this asterisk crash issue. On Thu, Feb 13, 2014 at 5:48 PM, Arun Ram arunram@gmail.com wrote: Hi, I am facing asterisk crash issue in my Asterisk 10.0.0. safe asterisk generated a core dump in /tmp path . I viewed the core dump using viewcore in linux. can anyone tell the reason for the crash . waiting eagerly for an answer from asterisk support guys. please the find the core dump attachment too .. Below is the information in core dump -- Thanks Regards Arunram.c The Power of someone has the power to do something.. anything !! -- Thanks Regards Arunram.c The Power of someone has the power to do something.. anything !! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks Regards Arunram.c The Power of someone has the power to do something.. anything !! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk?
Re: [asterisk-users] Want Queues to ignore mobile operators voice mails and continue ringing...?
Hi Daniel Le 14/02/2014 07:33, Daniel van den Berg a écrit : Hi All, Lets say I want to setup a queue that will handle inbound calls to dynamically added agents that are all mobile numbers. Now when I do this setup it works, it loads the agents dynamically and if the mobile phone is on and have reception it works. But when the phone is for arguments sake off or dont have reception it goes to voice mail for that mobile phone. I don't want this to happen...:) I would like for the queue to continue ringing until there is a time out specified which then takes the caller out of the queue and to voice mail which I then intend to mail somewhere. I guess my question is can this be done in Asterisk? Can I force clients in this queue not to leave a voice message on the mobile phone but rather the Asterisk system? Because when the mobile phone which is an agent in the queue goes to voice mail it answers the call and then plays the voice mail message. My initial thoughts are to maybe ask the mobile operator to switch off the voice mail functionality on those mobile phones and rather give a busy or engaged tone, but I would rather want to do this in Asterisk. Any help or advise on this matter will be greatly appreciated. Use ChanIsAvail command before adding agents dynamically -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Want Queues to ignore mobile operators voice mails and continue ringing...?
On 14/02/14 06:33, Daniel van den Berg wrote: Hi All, Lets say I want to setup a queue that will handle inbound calls to dynamically added agents that are all mobile numbers. Now when I do this setup it works, it loads the agents dynamically and if the mobile phone is on and have reception it works. But when the phone is for arguments sake off or dont have reception it goes to voice mail for that mobile phone. I don't want this to happen...:) I would like for the queue to continue ringing until there is a time out specified which then takes the caller out of the queue and to voice mail which I then intend to mail somewhere. I guess my question is can this be done in Asterisk? Can I force clients in this queue not to leave a voice message on the mobile phone but rather the Asterisk system? Because when the mobile phone which is an agent in the queue goes to voice mail it answers the call and then plays the voice mail message. My initial thoughts are to maybe ask the mobile operator to switch off the voice mail functionality on those mobile phones and rather give a busy or engaged tone, but I would rather want to do this in Asterisk. Any help or advise on this matter will be greatly appreciated. Thanks! Daniel van den Berg SureTel - South Africa I would suggest using the 'M' option on the Dial command to run a macro. The macro can just wait fir a key to be pressed and until it is pressed the Dial is still effectively ringing. So if it does go to voicemail then the call wont get put through. You need to make sure you have a suitable value set to abandon the agent call if its ringing too long. The callee may also find they are left multiple voicemail messages. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Want Queues to ignore mobile operators voice mails and continue ringing...?
On 14/2/14 9:21 am, Gareth Blades wrote: I would suggest using the 'M' option on the Dial command to run a macro. The macro can just wait fir a key to be pressed and until it is pressed the Dial is still effectively ringing. So if it does go to voicemail then the call wont get put through. You need to make sure you have a suitable value set to abandon the agent call if its ringing too long. The callee may also find they are left multiple voicemail messages. This is the approach we've used in the past: force the recipient to hit a button to accept the call, something which their mobile voicemail will never be able to do. The alternative - and it only really applies if you have control of the mobiles in question - is to disable the mobile network's voicemail service entirely, and manage diverts from the handset. That way you can then recreate your own 'mobile voicemail' service on your asterisk platform with all the normal asterisk VM benefits such as email delivery, etc. You can then of course detect when those mobiles 'divert' to voicemail (since it's now on your system), and kick them out of the queue at that point. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Want Queues to ignore mobile operators voice mails and continue ringing...?
Hi all, How does one detect the 'divert' to voicemail? Say we have PRI lines and as wel as SIP Trunks to connect to mobile phones. How can asterisk know if the call is being diverted?? On 14 February 2014 10:11, Chris Bagnall aster...@lists.minotaur.cc wrote: On 14/2/14 9:21 am, Gareth Blades wrote: I would suggest using the 'M' option on the Dial command to run a macro. The macro can just wait fir a key to be pressed and until it is pressed the Dial is still effectively ringing. So if it does go to voicemail then the call wont get put through. You need to make sure you have a suitable value set to abandon the agent call if its ringing too long. The callee may also find they are left multiple voicemail messages. This is the approach we've used in the past: force the recipient to hit a button to accept the call, something which their mobile voicemail will never be able to do. The alternative - and it only really applies if you have control of the mobiles in question - is to disable the mobile network's voicemail service entirely, and manage diverts from the handset. That way you can then recreate your own 'mobile voicemail' service on your asterisk platform with all the normal asterisk VM benefits such as email delivery, etc. You can then of course detect when those mobiles 'divert' to voicemail (since it's now on your system), and kick them out of the queue at that point. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Want Queues to ignore mobile operators voice mails and continue ringing...?
On 14/2/14 10:54 am, Tiago Geada wrote: How does one detect the 'divert' to voicemail? If you're using the mobile network's voicemail service, you can't as a general rule; you've no reliable way of knowing whether that call was answered by the user or their voicemail service. However, if you're providing the mobile voicemail service yourself from your asterisk platform, then you can detect the *incoming* call from the mobile device in question to their mailbox and act accordingly. As I said in my earlier reply though, it depends on you having end-to-end control of the mobile devices in question and your mobile operator will allow their voicemail service to be completely disabled. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Not Starting after YUM Update - Solved
On 13 Feb 2014, at 09:55, Aldo Bergamini aabe...@gmail.com wrote: Hi, I did compile the latest DAHDI and LibPRI, with no success… So I thought about updating the Asterisk package to the last known 1.6.2 release. Now it's crashing at some different point. This is the the strace result: Hi all, thanks for the hints. I did solve the problem… After reinstalling both DAHDI and LibPRI I first did recompile Asterisk, using the 1.6.2 release tarball. This proved to still make fuss, so that I redid the installation, this time with the latest 1.6.0 code. Aside complaining about some incompatible modules, the PBX went up well. I did even reload the binaries for G729, Skype and the fax stuff. They do not activate themselves (as an aside from the update nightmare, I had to replace an ethernet card), but this is probably due to host identification fingerprinting… Does anybody know if it is still possible to receive a 'move to new host' authorisation from Digium for the now unsupported Skype bridge? Thanks and best regards, Aldo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP OPTIONS storm?
- Original Message - SIP options message is due to check the peer registration is keepalive. As per my understanding it might be because of network flap may be wireshark trace can give you any clue. Regards Correct. I understand the role and function of the OPTIONS requests. The issue is why was Asterisk sending out 65Mbps worth of them to one peer? I did get a capture of the traffic, but nothing appears to explain *why* the traffic was there to begin with. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Want Queues to ignore mobile operators voice mails and continue ringing...?
On Friday 14 Feb 2014, Tiago Geada wrote: Hi all, How does one detect the 'divert' to voicemail? Say we have PRI lines and as wel as SIP Trunks to connect to mobile phones. How can asterisk know if the call is being diverted?? It can't. But you know (from the STD code) whether the call is being made to a mobile or land line; and you can have a good guess how long the mobile telco's voicemail timeout is. So as long as your Dial() to the mobile phone times out sooner than the mobile network rings out for before deciding that nobody is going to answer, *your* voicemail will win. This will break if somebody reduces their voicemail timeout from the default; but hardly anybody ever changes the default settings in practice. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP OPTIONS storm?
On 13 Feb 2014, at 18:10, Tim Nelson tnel...@rockbochs.com wrote: I recently experienced an odd situation. I have an Asterisk 11.5.0 system (Box A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At some point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box A. I do have qualify=yes for the peer on both sides, and the qualifyfreq is not set (aka default of 60secs). Just because Box B was receiving 65MBps doesn’t mean box A was sending them. I suspect it’s probably the same one repeated, due to some kind of network problem. Do you have a pcap so you can look for the ID in the packets to see if they are the same? Would be good if you can prove A sent them too (traffic stats from SNMP monitoring or something). S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Want Queues to ignore mobile operators voice mails and continue ringing...?
On Friday 14 Feb 2014, Tiago Geada wrote: Hi all, How does one detect the 'divert' to voicemail? Say we have PRI lines and as wel as SIP Trunks to connect to mobile phones. How can asterisk know if the call is being diverted?? It can't. But you know (from the STD code) whether the call is being made to a mobile or land line; and you can have a good guess how long the mobile telco's voicemail timeout is. So as long as your Dial() to the mobile phone times out sooner than the mobile network rings out for before deciding that nobody is going to answer, *your* voicemail will win. This will break if somebody reduces their voicemail timeout from the default; but hardly anybody ever changes the default settings in practice. ...or if the moble is turned off -Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ConfBridge on asterisk 11
I believe I am running an AGI (to put users in a conf) before the confbridge is built. So the users are not really get in the conf... exten X,1,run agi to put users in conf exten X,n,ConfBridge() How do I have in the dial plan ConfBridge() and someplace run an AGI that brings the users I want into that Conf. I cannot delay in the AGI and wait for the conf because the conf is not built until I return from the AGI... Any thoughts? Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge on asterisk 11
On Friday 14 Feb 2014, Jerry Geis wrote:
I believe I am running an AGI (to put users in a conf) before the
confbridge is built. So the users are not really get in the conf...
exten X,1,run agi to put users in conf
exten X,n,ConfBridge()
How do I have in the dial plan ConfBridge() and someplace
run an AGI that brings the users I want into that Conf.
I cannot delay in the AGI and wait for the conf because
the conf is not built until I return from the AGI...
Any thoughts?
Thanks,
Jerry
Make your AGI script fork itself; have the child process detach, and the
parent process exit. Then, the AGI call will return quickly to the dialplan;
and meanwhile, the script can continue in the background at its own leisure.
Example code (Perl) follows:
#!/usr/bin/perl -w
use strict;
use Asterisk::AGI;
my $child_pid;
my $AGI = new Asterisk::AGI;
my %params = $AGI-ReadParse();
$SIG{CHLD} = IGNORE;
if ($child_pid = fork) {
# This is executed in the parent process
exit;
}
elsif (defined $child_pid) {
# This is executed in the child process
close STDIN;
close STDOUT;
close STDERR;
# Now we are detached
#
# This is where we do the funky stuff
#
exit;
}
else {
# Oh, s#!t
die Could not fork: $!;
};
# We should never, ever get here
exit;
--
AJS
Answers come *after* questions.
--
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[asterisk-users] Dialer software for Asterisk...
I have a customer with a more or less unique need. Right now we are using Wombat as a dialer software so they can contact clients for QA purposes. Everything is working very well and their contact center productivity is way up from the old manual dialing method. The only thing we are having a problem with is that they have up to 5 phone numbers to contact a single customer. Obviously we cannot load all numbers into the dialer because we do not want to contact the same customer 5 times. Does anyone know of a dialer for Asterisk that can take several phone numbers for the same contact and if any of those answers it will not try the other numbers? Most of the dialers I have looked at cannot relate information for different numbers so there is no way to tell if you have already contacted a specific customer with a different number. I really do not want to develop a new dialer software (well, while the dialer is not that difficult the interfaces, reports and backends are a pain to maintain). Anyone know of a commercial or open source software that can handle this kind of dialing? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialer software for Asterisk...
Have a look at vicidial it has alternate number dialing capability. Mituo On Saturday, February 15, 2014, Carlos Chavez cur...@telecomabmex.com wrote: I have a customer with a more or less unique need. Right now we are using Wombat as a dialer software so they can contact clients for QA purposes. Everything is working very well and their contact center productivity is way up from the old manual dialing method. The only thing we are having a problem with is that they have up to 5 phone numbers to contact a single customer. Obviously we cannot load all numbers into the dialer because we do not want to contact the same customer 5 times. Does anyone know of a dialer for Asterisk that can take several phone numbers for the same contact and if any of those answers it will not try the other numbers? Most of the dialers I have looked at cannot relate information for different numbers so there is no way to tell if you have already contacted a specific customer with a different number. I really do not want to develop a new dialer software (well, while the dialer is not that difficult the interfaces, reports and backends are a pain to maintain). Anyone know of a commercial or open source software that can handle this kind of dialing? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967196 Cell: +91-9820332422 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Changing gateway address
Hello, I inherited an Asterix phone system. I am well versed in Windows based platforms but have zero experience in Linux and Asterix, no make matters worse I have no documentation on this system. I had to change the entire networks gateway address for various reasons but now the Asterix system will not send messages via email. I think it is because of the gateway change. How do I change the gateway address? Is this product something I could contract out to have remote support? Thanks, Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing gateway address
On Fri, Feb 14, 2014 at 5:40 PM, Dave Swangler ctit...@live.com wrote: Hello, I inherited an Asterix phone system. I am well versed in Windows based platforms but have zero experience in Linux and Asterix, no make matters worse I have no documentation on this system. I had to change the entire networks gateway address for various reasons but now the Asterix system will not send messages via email. I think it is because of the gateway change. How do I change the gateway address? Is this product something I could contract out to have remote support? Thanks, What you describe is more of a Linux support issue then specific to Asterisk. Depending on your OS, will dictate how to change your gateway. check /etc/network/inferfaces if you are ubuntu / debian. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
