Re: [asterisk-users] DAHDI Error
This sounds like you have it set for T1 somehow? Have you upgraded anything lately? Other than that, a Trend tester will show the problem(s) to you. BTW - E1's are 32 channel (not 31). It's 30B+2D. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup Sent: 13 May 2011 16:57 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DAHDI Error I can dial 1-24 channels but not after that. There are 8 E1s. Box was working fine and carrying traffic on all E1s before. Just recently i noticed this problem has occurred. On 13 May 2011 16:30, Rafael Visser visser.raf...@gmail.com wrote: I didn't understand very well.. So you cant dial on the first 24 channels? Did you take care on the jumper of the card?. There is something related to E1 (31 channels) or T1 (24 channels). And check the system.conf either. rv 2011/5/13 deeps backup backup.de...@gmail.com I have checked destination numbers are correct as otherwise calls to those numbers are connecting fine. I opened verbose logs and digged into it more. I found out can't dial any channels from DAHDI/24 on first E1. Before that channel calls are going through fine. I tried test calls to second E1 and can't dial on it either. When I check channel or E1 status it is showing fine. Checked chan_dahdi and system conf files and see all channels are configured fine. Could you please help? On 13 May 2011 15:07, deeps backup backup.de...@gmail.com wrote: On 13 May 2011 14:06, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup Sent: Friday, May 13, 2011 9:02 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DAHDI Error Hi, Sometimes calls on Asterisk fail to connect to DAHDI channels and giving below error: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) There are 8 E1 connected on server and only 15-20 simultaneous calls. All channels and E1 are showing in service without any alarms. Could anyone please let me know why this is happening? The message is likely coming from the telco or from the destination number. It is a common issue. I usually put something in my dialplan to retry all calls that receive an unexpected hangup cause to work around the telco seemingly randomly sending back odd hangup causes. You should not retry ALL calls, only ones with unexpected hangup causes. -- _ -- 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 I have checked destination numbers are correct as otherwise calls to those numbers are connecting fine. I opened verbose logs and digged into it more. I found out can't dial any channels from DAHDI/24 on first E1. Before that channel calls are going through fine. I tried test calls to second E1 and can't dial on it either. When I check channel or E1 status it is showing fine. Checked chan_dahdi and system conf files and see all channels are configured fine. Could you please help? -- _ -- 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 If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has
Re: [asterisk-users] Backport of DEVICE_STATE to 1.4
https://issues.asterisk.org/view.php?id=15818 That's where I get it from. If it contains errors, then why not report it there? Cheers -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 13 May 2011 15:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Backport of DEVICE_STATE to 1.4 Hi, Here http://www.voip-info.org/wiki/view/Asterisk+func+device_State you can find a link to download a backported for Asterisk 1.4 version of DEVICE_STATE function. (Elsewhere, you can find reference to another backported function DEVSTATE which seems to behave the same as DEVICE_STATE). As I would like to prepare as much as possible, my dialplan to 1.6 and beyond, I would prefer to use DEVICE_STATE if possible. Anyway, a quick inside this fucn_devstate.c file shows that some (all ?) Log or Error messages are still refering to DEVSTATE. My question is which is the best source to get DEVICE_STATE function for Asterisk 1.4 ? Regards If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085. -- _ -- 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] DAHDI Error
On Mon, May 16, 2011 at 09:18:33AM +0100, Andrew Thomas wrote: BTW - E1's are 32 channel (not 31). It's 30B+2D. Technically, yes. But channel (time slot) 0 never makes it to DAHDI. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.8.4 keeps quitting console by itself
Hi! I've noticed 1.8.4 keeps quitting console by itself. Is this a bug or feature? :) Nick -- _ -- 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] 1.8.4 quitting console
actually i just noticed that it quits console because asterisk restarts itself after: [2011-05-16 13:48:45] ERROR[11106] tcptls.c: Unable to connect SIP socket to 192.168.1.108:5060: Connection timed out -- _ -- 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] Backport of DEVICE_STATE to 1.4
2011/5/16 Andrew Thomas a...@datavox.co.uk
https://issues.asterisk.org/view.php?id=15818
That's where I get it from.
If it contains errors, then why not report it there?
Cheers
As this bug is considered fixed, I think you can't add any comment
anymore.
Unfortunately, you can still see lines mentionning DEVSTATE function like :
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, DEVSTATE function called with no custom
device
name!\n);
return -1;
}
I opened issue 19300 for that.
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Re: [asterisk-users] OT - Which Android handset with Wifi-only ?
2011/5/15 Jonathan Thurman jonat...@thurmantech.com On Sun, May 15, 2011 at 10:16 AM, sean darcy seandar...@gmail.com wrote: anyone actually used this on Android to connect to an asterisk server? Yes. I purchased it a while ago from the Marketplace, and had some issues with sound quality as my specific phone (Motorola Atrix) isn't officially supported yet. However, the support people at CounterPath have been extremely responsive, and the latest version works much better. I have not tested the G.729 codec. It's a good app, but I would buy it from CouterPath directly next time as their refund policy is longer than 15 minutes and they list the supported devices. Hopefully they will add video support soon. -Jonathan -- _ -- 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 I took a closer look at counterpath web site to discover an approved handsets list. Too bad all of them, at the moment, seem to be rather high end dual-mode (WiFi-2G/3G) handsets Cheers -- _ -- 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] Backport of DEVICE_STATE to 1.4
On 11-05-16 07:29 AM, Olivier wrote:
As this bug is considered fixed, I think you can't add any comment
anymore.
Unfortunately, you can still see lines mentionning DEVSTATE function like :
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, DEVSTATE function called with no custom
device
name!\n);
return -1;
}
I opened issue 19300 for that.
Sorry, but backported code is not supported on the issue tracker. You'll need to
use a version of Asterisk that natively supports the DEVICE_STATE() function and
which has maintenance support status (i.e. Asterisk 1.8).
Thanks,
Leif.
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[asterisk-users] AMI perl daemon
Would anybody know how to run a perl script as a daemon that would stay connected to asterisk via AMI? Right now, my AMI script connects to the manager interface, originates a call, disconnects. The script will be run maybe 20+ per minute. It would make more sense to me to have the script run as a daemon and have a persistent connection to asterisk's AMI. Thank you in advance for your input. -- _ -- 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] AMI perl daemon
On 05/16/2011 08:14 AM, vip killa wrote: Would anybody know how to run a perl script as a daemon that would stay connected to asterisk via AMI? Right now, my AMI script connects to the manager interface, originates a call, disconnects. The script will be run maybe 20+ per minute. It would make more sense to me to have the script run as a daemon and have a persistent connection to asterisk's AMI. Thank you in advance for your input. Well, you would just write the Perl script in such a way as to not close the connection :-), but continue reading from the socket, ideally in an asynchronous manner. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI perl daemon
If the script were called each time an extension were dialed in a dialplan for example, wouldn't each new instance of the script need to re-connect to AMI, run command, disconnect? On Mon, May 16, 2011 at 8:16 AM, Alex Balashov abalas...@evaristesys.comwrote: On 05/16/2011 08:14 AM, vip killa wrote: Would anybody know how to run a perl script as a daemon that would stay connected to asterisk via AMI? Right now, my AMI script connects to the manager interface, originates a call, disconnects. The script will be run maybe 20+ per minute. It would make more sense to me to have the script run as a daemon and have a persistent connection to asterisk's AMI. Thank you in advance for your input. Well, you would just write the Perl script in such a way as to not close the connection :-), but continue reading from the socket, ideally in an asynchronous manner. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] AMI perl daemon
On 05/16/2011 08:19 AM, vip killa wrote: If the script were called each time an extension were dialed in a dialplan for example, wouldn't each new instance of the script need to re-connect to AMI, run command, disconnect? Well, yes, if you invoke a new instance of the script each time, that is what would happen. The desired approach is to have some means of communicating with the running daemon to indicate to it that it should originate a call, perhaps via a control socket/API. If your invocation is in the dial plan, the simplest thing to do would be to build a FastAGI server in Perl. This CPAN module can save some work: http://search.cpan.org/~jaywhy/Asterisk-FastAGI-0.02/lib/Asterisk/FastAGI.pm Then have that process either maintain a persistent AMI connection, or open a new one each time if you don't feel like/don't know how to implement the asynchronous approach. When you want to initiate a dial, just call: exten = ...,x,AGI(agi://some.server.ip/your_script) Of course, you could also use call files if the script is executing on the same Asterisk server as the one on which the dials take place. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI perl daemon
Thank you, that makes sense but actually I would be invoking the script using the externnotify in voicemail.conf, similar to externnotify = /var/lib/asterisk/scripts/notify.pl I assume externnotify cannot call the FastAGI server...correct? On Mon, May 16, 2011 at 8:23 AM, Alex Balashov abalas...@evaristesys.comwrote: On 05/16/2011 08:19 AM, vip killa wrote: If the script were called each time an extension were dialed in a dialplan for example, wouldn't each new instance of the script need to re-connect to AMI, run command, disconnect? Well, yes, if you invoke a new instance of the script each time, that is what would happen. The desired approach is to have some means of communicating with the running daemon to indicate to it that it should originate a call, perhaps via a control socket/API. If your invocation is in the dial plan, the simplest thing to do would be to build a FastAGI server in Perl. This CPAN module can save some work: http://search.cpan.org/~jaywhy/Asterisk-FastAGI-0.02/lib/Asterisk/FastAGI.pm Then have that process either maintain a persistent AMI connection, or open a new one each time if you don't feel like/don't know how to implement the asynchronous approach. When you want to initiate a dial, just call: exten = ...,x,AGI(agi://some.server.ip/your_script) Of course, you could also use call files if the script is executing on the same Asterisk server as the one on which the dials take place. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] AMI perl daemon
On 05/16/2011 08:33 AM, vip killa wrote: Thank you, that makes sense but actually I would be invoking the script using the externnotify in voicemail.conf, similar to externnotify = /var/lib/asterisk/scripts/notify.pl http://notify.pl I assume externnotify cannot call the FastAGI server...correct? That is correct. But you can call a script that notifies the daemon through a FIFO or UNIX domain socket, if local, or network socket if remote. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-cpu utilization 60 %
check your running process, if you have more than one asterisk in your top re install your asterisk. On Sun, May 15, 2011 at 7:07 PM, Satish Patel satish...@hotmail.com wrote: Check this out http://www.moythreads.com/wordpress/2009/05/06/why-does-asterisk-consume-100-cpu/ -- Sent from my iPhone On May 15, 2011, at 4:08 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, May 15, 2011 at 08:24:08AM +0200, Leandro Dardini wrote: 2011/5/15 RSCL Mumbai rscl.mum...@gmail.com On Sat, May 14, 2011 at 11:43 AM, Leandro Dardini ldard...@gmail.com wrote: Check if someone is brute forcing your asterisk accounts. It used to happen to me before I install fail2ban. You can easily check the full log of asterisk or with just a tcpdump -i any -n port 5060 or port 4569. Thx for the tcpdump command. Checked, all looks good. Packets coming from trusted domains only. What should be the next step ? Thx Sans Have you tried to restart asterisk? As last chance, install strace and check what is asterisk doing. Get the pid (PID) of the running asterisk and run: strace -p PID -f -F /tmp/strace.log Not exactly. Asterisk is multi-threaded. strae traces a specific thread. To see the most active thread, press 'H' (shift-h) in top. Wait for the display to refresh at least twice (on the first time it won't make sense) and now check to see which is the top thread. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Pezhman Lali -- _ -- 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] AMD tweaking
Hi, long time ago, I came up with an optimal configuration set for my environment - good detection and little false positives. Unfortunately some people are always being detected as Answering Machines. I'm not up to re-adjust my precious balance of initial_silence/max_words/... , so I'm thinking about to check if the pickup time is equal to the pickup time when the same phone number was previously detected as AM - if the pickup time is different from the last time, - it's HUMAN, else proceed standard AMD(). has anyone done this before,so I wouldn't be reinventing bicycle? -- Mvh, Aurimas Skirgaila -- _ -- 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] Different box for SIP and RTP
Hello, Is there way I can use two Asterisk box, one to maintain SIP packets and other for RTP traffic? Thanks, Mohammad -- _ -- 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] AMD tweaking
You would have to make the tolerance of variance fairly high. There are many reasons why pickup time by a mechanical device such as an answering machine or a fax machine may vary quite significantly. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On May 16, 2011, at 8:56 AM, Aurimas Skirgaila a.skirga...@gmail.com wrote: Hi, long time ago, I came up with an optimal configuration set for my environment - good detection and little false positives. Unfortunately some people are always being detected as Answering Machines. I'm not up to re-adjust my precious balance of initial_silence/max_words/... , so I'm thinking about to check if the pickup time is equal to the pickup time when the same phone number was previously detected as AM - if the pickup time is different from the last time, - it's HUMAN, else proceed standard AMD(). has anyone done this before,so I wouldn't be reinventing bicycle? -- Mvh, Aurimas Skirgaila -- _ -- 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] Different box for SIP and RTP
On 05/16/2011 09:00 AM, Mohammad Khan wrote: Is there way I can use two Asterisk box, one to maintain SIP packets and other for RTP traffic? No, the signaling and bearer plane are integrated in Asterisk. But you can use reinvites to hand off RTP processing to third-party endpoints and bypass Asterisk, in qualifying call scenarios and network topologies. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-cpu utilization 60 %
On Mon, May 16, 2011 at 6:19 PM, Pezhman Lali l...@lopl.net wrote: check your running process, if you have more than one asterisk in your top re install your asterisk. On Sun, May 15, 2011 at 7:07 PM, Satish Patel satish...@hotmail.comwrote: Check this out http://www.moythreads.com/wordpress/2009/05/06/why-does-asterisk-consume-100-cpu/ Moving forward with the suggestion provided on the above link, I have the activity dump of all asterisk processes when the load was 22%. Need help in understanding the output. What should I look for which would indicate undue CPU utilization. Thank you every one for your continued support. Thread 45 (Thread 0x4175d940 (LWP 4129)): #0 0x00307500aee9 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 #1 0x004df6a3 in ?? () #2 0x004eb76e in ?? () #3 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #4 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 44 (Thread 0x417d9940 (LWP 4130)): #0 0x0030744cb186 in poll () from /lib64/libc.so.6 #1 0x0042d181 in ?? () #2 0x004eb76e in ?? () #3 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #4 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 43 (Thread 0x41855940 (LWP 4131)): #0 0x00307500aee9 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 #1 0x00490ea3 in ?? () #2 0x004eb76e in ?? () #3 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #4 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 42 (Thread 0x41ef4940 (LWP 4132)): #0 0x00307449a3f1 in nanosleep () from /lib64/libc.so.6 #1 0x004de5df in ?? () #2 0x004eb76e in ?? () #3 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #4 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 41 (Thread 0x413ed940 (LWP 4133)): #0 0x0030744cb186 in poll () from /lib64/libc.so.6 #1 0x004ea175 in ast_wait_for_input () #2 0x004e0a61 in ast_tcptls_server_root () #3 0x004eb76e in ?? () #4 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #5 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 40 (Thread 0x4148e940 (LWP 4134)): #0 0x00307500aee9 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 #1 0x00460fc9 in ?? () #2 0x004eb76e in ?? () #3 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #4 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 39 (Thread 0x41ce8940 (LWP 4135)): #0 0x00307500aee9 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 #1 0x004df6a3 in ?? () #2 0x004eb76e in ?? () #3 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #4 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 38 (Thread 0x4150a940 (LWP 4136)): #0 0x0030744cd212 in select () from /lib64/libc.so.6 #1 0x00473315 in ?? () #2 0x004eb76e in ?? () #3 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #4 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 37 (Thread 0x418d1940 (LWP 4137)): #0 0x00307500aee9 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0 #1 0x2aaab9d46e03 in ast_unregister_file_version () from /usr/lib64/asterisk/modules/res_timing_pthread.so #2 0x004eb76e in ?? () #3 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #4 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 36 (Thread 0x4194d940 (LWP 4138)): #0 0x0030744cb186 in poll () from /lib64/libc.so.6 #1 0x0048a4d0 in ast_io_wait () #2 0x2aaac5360d3b in ?? () from /usr/lib64/asterisk/modules/pbx_dundi.so #3 0x004eb76e in ?? () #4 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #5 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 35 (Thread 0x419c9940 (LWP 4139)): #0 0x00307449a3f1 in nanosleep () from /lib64/libc.so.6 #1 0x00307449a214 in sleep () from /lib64/libc.so.6 #2 0x2aaac5360ba4 in ?? () from /usr/lib64/asterisk/modules/pbx_dundi.so #3 0x004eb76e in ?? () #4 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #5 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 34 (Thread 0x41d64940 (LWP 4140)): #0 0x00307449a3f1 in nanosleep () from /lib64/libc.so.6 #1 0x00307449a214 in sleep () from /lib64/libc.so.6 #2 0x2aaac5357272 in ast_unregister_file_version () from /usr/lib64/asterisk/modules/pbx_dundi.so #3 0x004eb76e in ?? () #4 0x00307500673d in start_thread () from /lib64/libpthread.so.0 #5 0x0030744d3f6d in clone () from /lib64/libc.so.6 Thread 33 (Thread 0x41256940 (LWP 4141)): #0 0x00307449a3f1 in nanosleep () from /lib64/libc.so.6 #1 0x2aaac1eff28e in ast_unregister_file_version () from /usr/lib64/asterisk/modules/pbx_spool.so #2 0x004eb76e in
Re: [asterisk-users] Different box for SIP and RTP
Can't that third-party be an asterisk box? After hand off RTP processing, does the first box (who, hand off) still in charge of SIP packets? On Mon, May 16, 2011 at 9:13 AM, Alex Balashov abalas...@evaristesys.comwrote: On 05/16/2011 09:00 AM, Mohammad Khan wrote: Is there way I can use two Asterisk box, one to maintain SIP packets and other for RTP traffic? No, the signaling and bearer plane are integrated in Asterisk. But you can use reinvites to hand off RTP processing to third-party endpoints and bypass Asterisk, in qualifying call scenarios and network topologies. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] question on digium repo
I an running centos 5. I added this to the digium.repo file in /etc/yum.repos.d directory. [digium-current] name=CentOS-$releasever - Digium - Current baseurl=http://packages.digium.com/centos/$releasever/current/$basearch/ enabled=1 gpgcheck=0 #gpgkey=http://packages.digium.com/RPM-GPG-KEY-Digium Then I did yum install asterisk14 addons | 951 B 00:00 base | 2.1 kB 00:00 base/primary_db | 2.2 MB 00:03 digium-current | 1.1 kB 00:00 digium-current/primary | 33 kB 00:00 digium-current 260/260 extras | 2.1 kB 00:00 extras/primary_db | 244 kB 00:00 updates | 1.9 kB 00:00 updates/primary_db | 544 kB 00:01 Setting up Install Process No package asterisk14 available. What did I miss? jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma A400 background noise after a while
Dear Mr.Shokei Salam as I heard, some HP servers are very sensitive, about the os, you must install the os only from the included cd and smart drive. best On Mon, May 16, 2011 at 7:23 AM, Moises Silva moises.si...@gmail.comwrote: On Wed, May 4, 2011 at 1:01 PM, M Shokuie sena...@gmail.com wrote: Dear folks, We have recently installed A400D card with 12 FXO modules, the serer is HP DL180 G6, cards works fine but after a while all the calls get an awful noise, you can not get what each side says. The noise cleares as soon as we restart wanrouter but not asterisk (i mean asterisk restart does not solve). We previsouly confronted this situation with PRI cards but not analogs, wanpipe version is 3.5.18 and zaptel 1.4.12 also tested with recent DAHDI with out any help. ifconfig doesnt show any overruns or errors. Once earlier we had the same problem and come to the conclusion to change the mainboard but this time i got mad as i couldnt change a 3000$ HP server that easy. Is there a way i could get if there is any problem of interrupts, when i check interrupts i could not see any shared interrupts for Snagoma card. Anyhelp would be highly appreciated. -- Hello M Shokuie, This kind of troubleshooting is better addressed by Sangoma technical support staff. You can send an email to techd...@sangoma.com and you will be taken care of. Regards, Moises Silva Senior Software Engineer, Software Development Manager Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Pezhman Lali -- _ -- 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] Different box for SIP and RTP
On 11-05-16 09:13 AM, Alex Balashov wrote: On 05/16/2011 09:00 AM, Mohammad Khan wrote: Is there way I can use two Asterisk box, one to maintain SIP packets and other for RTP traffic? No, the signaling and bearer plane are integrated in Asterisk. But you can use reinvites to hand off RTP processing to third-party endpoints and bypass Asterisk, in qualifying call scenarios and network topologies. You could try directrtpsetup=yes which is similar to directmedia, except the audio is redirected in the initial INVITEs rather than reinviting the media a few RTP packets in. Leif. -- _ -- 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] AMD tweaking
Thank you, Alex yes, I expect the pickup time to vary within 1 second (it's just a guess). If I have to tolerate higher bias, so I would start doubting about the efficiency of this method. On Mon, May 16, 2011 at 4:00 PM, Alex Balashov abalas...@evaristesys.comwrote: You would have to make the tolerance of variance fairly high. There are many reasons why pickup time by a mechanical device such as an answering machine or a fax machine may vary quite significantly. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On May 16, 2011, at 8:56 AM, Aurimas Skirgaila a.skirga...@gmail.com wrote: Hi, long time ago, I came up with an optimal configuration set for my environment - good detection and little false positives. Unfortunately some people are always being detected as Answering Machines. I'm not up to re-adjust my precious balance of initial_silence/max_words/... , so I'm thinking about to check if the pickup time is equal to the pickup time when the same phone number was previously detected as AM - if the pickup time is different from the last time, - it's HUMAN, else proceed standard AMD(). has anyone done this before,so I wouldn't be reinventing bicycle? -- Mvh, Aurimas Skirgaila -- _ -- 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 -- Mvh, Aurimas Skirgaila -- _ -- 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] Backport of DEVICE_STATE to 1.4
Ah! Forgot about that.
Looks like your on your own Olivier.
Sorry
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif
Madsen
Sent: 16 May 2011 13:12
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Backport of DEVICE_STATE to 1.4
On 11-05-16 07:29 AM, Olivier wrote:
As this bug is considered fixed, I think you can't add any comment
anymore. Unfortunately, you can still see lines mentionning DEVSTATE
function like :
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, DEVSTATE function called with no
custom device
name!\n);
return -1;
}
I opened issue 19300 for that.
Sorry, but backported code is not supported on the issue tracker. You'll
need to use a version of Asterisk that natively supports the
DEVICE_STATE() function and which has maintenance support status (i.e.
Asterisk 1.8).
Thanks,
Leif.
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If you are not the intended recipient, employee or agent responsible
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any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
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Re: [asterisk-users] Asterisk-cpu utilization 60 %
On Sun, May 15, 2011 at 4:08 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: Not exactly. Asterisk is multi-threaded. strae traces a specific thread. To see the most active thread, press 'H' (shift-h) in top. Wait for the display to refresh at least twice (on the first time it won't make sense) and now check to see which is the top thread. strace -f -ff ASTERISK_PID traces all threads on my system. -- _ -- 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] res_timing_timerfd.so Vs res_timing_dahdi.so
Thanks Leif, I had changed it to res_timing_dahdi and since last few days it seem good. -S Date: Sun, 15 May 2011 15:48:03 -0400 From: leif.mad...@asteriskdocs.org To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so On 11-05-13 11:39 AM, isr...@gmail.com wrote: I haven't tried with timerfd but with timer pthread 1.8 is very unstable I think I have seen a post to the list from kevin fleming that the same is for timerfd that there is a nasty bug which they haven't found the reason for yet My experience is that you should pretty much always use res_timing_dahdi unless you're on a platform on which you can't install DAHDI. You don't need any hardware to use timing from DAHDI because timing is generated by the kernel. My order of preference for stability is: * res_timing_dahdi * res_timing_timerfd * res_timing pthread The timerfd and pthread modules are relatively new, and sometimes people run into stability problems while using them. If you can use res_timing_dahdi I recommend you do so. Leif. -- _ -- 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] Asterisk-cpu utilization 60 %
On Mon, May 16, 2011 at 10:01:36AM -0400, Mark Deneen wrote: strace -f -ff ASTERISK_PID traces all threads on my system. But do you really want that? Asterisk has many threads generating quite a lot of noise (threads periodically polling something). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-cpu utilization 60 %
On Mon, May 16, 2011 at 05:19:20PM +0430, Pezhman Lali wrote: check your running process, if you have more than one asterisk in your top re install your asterisk. Reinstall? Care to explain why? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Backport of DEVICE_STATE to 1.4
2011/5/16 Andrew Thomas a...@datavox.co.uk
Ah! Forgot about that.
Looks like your on your own Olivier.
Not yet as I found this one :
http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/func_devstate.c
In this one, even logs are up to date (ie references to DEVICE_STATE) but I
don't know if there other differences with other backported functions (from
issue 15818, for instance) are meangingful.
Maybe this link could be added to issue 19300 for further reference.
Now it seems I don't have to change my dialplan (for DEVSTATE/DEVICE_STATE)
when moving from 1.4 to 1.6 or 1.8.
Thanks for helping anyway.
Sorry
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif
Madsen
Sent: 16 May 2011 13:12
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Backport of DEVICE_STATE to 1.4
On 11-05-16 07:29 AM, Olivier wrote:
As this bug is considered fixed, I think you can't add any comment
anymore. Unfortunately, you can still see lines mentionning DEVSTATE
function like :
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, DEVSTATE function called with no
custom device
name!\n);
return -1;
}
I opened issue 19300 for that.
Sorry, but backported code is not supported on the issue tracker. You'll
need to use a version of Asterisk that natively supports the
DEVICE_STATE() function and which has maintenance support status (i.e.
Asterisk 1.8).
Thanks,
Leif.
-- _
-- 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
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The contents
of this message, and any attachments, are the property of DataVox,
and are intended for the confidential use of the named recipient only.
If you are not the intended recipient, employee or agent responsible
for delivery of this message to the intended recipient, take note that
any dissemination, distribution or copying of this communication and
its attachments is strictly prohibited, and may be subject to civil or
criminal action for which you may be liable.
Every effort has been made to ensure that this e-mail or any attachments
are free from viruses. While the company has taken every reasonable
precaution to minimise this risk, neither company, nor the sender can
accept liability for any damage which you sustain as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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--
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Re: [asterisk-users] Asterisk-cpu utilization 60 %
Sorry fro hijacking thread. I have following process running on my asterisk eating around 2 or 3% CPU constantly. I knew events0/1 is CPU queue but why only single queue is busy ? I have kernel running preemtive with 1000Hz satish@campbx1:~$ ps aux | grep events root 9 1.7 0.0 0 0 ?SMay08 201:35 [events/0] root10 0.0 0.0 0 0 ?SMay08 1:19 [events/1] Date: Mon, 16 May 2011 17:37:16 +0300 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk-cpu utilization 60 % On Mon, May 16, 2011 at 05:19:20PM +0430, Pezhman Lali wrote: check your running process, if you have more than one asterisk in your top re install your asterisk. Reinstall? Care to explain why? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on digium repo
On 05/16/2011 08:36 AM, Jerry Geis wrote: I an running centos 5. I added this to the digium.repo file in /etc/yum.repos.d directory. [digium-current] name=CentOS-$releasever - Digium - Current baseurl=http://packages.digium.com/centos/$releasever/current/$basearch/ enabled=1 gpgcheck=0 #gpgkey=http://packages.digium.com/RPM-GPG-KEY-Digium Then I did yum install asterisk14 addons | 951 B 00:00 base | 2.1 kB 00:00 base/primary_db | 2.2 MB 00:03 digium-current | 1.1 kB 00:00 digium-current/primary | 33 kB 00:00 digium-current 260/260 extras | 2.1 kB 00:00 extras/primary_db | 244 kB 00:00 updates | 1.9 kB 00:00 updates/primary_db | 544 kB 00:01 Setting up Install Process No package asterisk14 available. What did I miss? jerry You missed the Asterisk repo. Replace all instances of digium.com with asterisk.org (and then Digium with Asterisk). packages.digium.com is Digium modules, such as FaxForAsterisk, whereas packages.asterisk.org is Asterisk, DAHDI, libpri, etc. -- _ -- 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-cpu utilization 60 %
http://www.moythreads.com/wordpress/2009/05/06/why-does-asterisk-consume-100-cpu/ Moving forward with the suggestion provided on the above link, I have the activity dump of all asterisk processes when the load was 22%. Need help in understanding the output. What should I look for which would indicate undue CPU utilization. Any finding in my *asterisk.stack.txt ? *Thank you.* * -- _ -- 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-cpu utilization 60 %
On Mon, May 16, 2011 at 10:33 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Mon, May 16, 2011 at 10:01:36AM -0400, Mark Deneen wrote: strace -f -ff ASTERISK_PID traces all threads on my system. But do you really want that? Asterisk has many threads generating quite a lot of noise (threads periodically polling something). Probably not. I was merely referring to the statement that strace only traces a particular thread. I would do top -H and then strace the asterisk threads with high CPU numbers. -M -- _ -- 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-cpu utilization 60 %
First grab LWP thread ID which is eating more CPU ps -LlFm -p `pidof asterisk` Now look into your asterisk.stack.txt and search particular LWP thread ID see following example Thread 10 (Thread 0x41d8f940 (LWP 3406)): #0 0x0033ce2ca436 in poll () from /lib64/libc.so.6 #1 0x004933c0 in ast_io_wait () #2 0x2aaabd9510cd in network_thread () #3 0x004f8b2c in dummy_start () #4 0x0033cee06367 in start_thread () from /lib64/libpthread.so.0 #5 0x0033ce2d2f7d in clone () from /lib64/libc.so.6 Now you have piece of cake. whatever the issue is you can find in above few lines.. -S Date: Mon, 16 May 2011 20:38:34 +0530 From: rscl.mum...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk-cpu utilization 60 % http://www.moythreads.com/wordpress/2009/05/06/why-does-asterisk-consume-100-cpu/ Moving forward with the suggestion provided on the above link, I have the activity dump of all asterisk processes when the load was 22%. Need help in understanding the output. What should I look for which would indicate undue CPU utilization. Any finding in my asterisk.stack.txt ? Thank you. -- _ -- 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
[asterisk-users] AMI check if connection is alive
I'm using a perl daemon i wrote to connect to AMI and perform actions. The daemon connects to asterisk via AMI at start up. Is there anyway to check if the AMI connection is still alive, for example every 2 seconds. if the connection is not alive, re-connect to AMI? Also, does AMI timeout after a certain amount of time of not sending commands? -- _ -- 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] AMI perl daemon
Alex is pointing you in the right direction. You should want a single daemon running that then gets notified by the voicemail script, either through a FIFO, a socket, or by dropping a file in a watched directory. If you are going to write a daemon, I would suggest looking at : http://search.cpan.org/~greenbean/Asterisk-AMI-v0.2.5/ It has integration with event loops and should work well for what you are doing. It also has some features for detecting disconnects and timeouts. On Mon, May 16, 2011 at 5:42 AM, Alex Balashov abalas...@evaristesys.com wrote: On 05/16/2011 08:33 AM, vip killa wrote: Thank you, that makes sense but actually I would be invoking the script using the externnotify in voicemail.conf, similar to externnotify = /var/lib/asterisk/scripts/notify.pl http://notify.pl I assume externnotify cannot call the FastAGI server...correct? That is correct. But you can call a script that notifies the daemon through a FIFO or UNIX domain socket, if local, or network socket if remote. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Missing Config Files under /etc/asterisk
Hi I have followed https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-YUM%28CentOS%2FRedHat%29, to my surprise there is only one config file by the name zapata.conf under /etc/asterisk/ There are no other config files. Any thing i am missing ? Please suggest/guide. Regards, Kaushal -- _ -- 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] Step by step guide
Hi, Are there step by step guide to configure Digium Card in Asterisk ? I have done it using Sangoma Card. Please suggest/guide. Regards, Kaushal -- _ -- 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 1.4.41 - Warning and Notice about contact info and stale nonce
Of course it's 1.4.41. And the result is that devices doesn't register anymore. Thanks for any hint. Le 14/05/2011 17:37, Administrator TOOTAI a écrit : Hi list, We have devices since more then 4 years which where running well with Asterisk. But with latest version (1.38 or more) we face problem with those devices when they try to register. We got [2011-05-14 17:18:06] WARNING[28559]: chan_sip.c:9950 register_verify: Failed to parse contact info --- Transmitting (NAT) to XXX.XXX.XXX.XXX:5062 --- SIP/2.0 400 Bad Request Followed by [2011-05-14 17:19:06] NOTICE[28559]: chan_sip.c:9502 check_auth: Correct auth, but based on stale nonce received from 'sip:7...@yyy.yyy.yyy.yyy;user=phone;tag=63d2ba80bffb016f' Checking logs we found Contact: * in headers before the failed parse contact info. We checked in source chan_sip and saw the parse info reject with Error 400 after the auth is correct comment. We modified in sip.conf the type=peer in type=friend, same result. Could someone explain us what happends here? Thanks -- 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] Step by step guide
I have Digium Card - Two (2) span digital T1/E1/J1/PRI PCI-Express x1 card On Mon, May 16, 2011 at 9:35 PM, Kaushal Shriyan kaushalshri...@gmail.com wrote: Hi, Are there step by step guide to configure Digium Card in Asterisk ? I have done it using Sangoma Card. Please suggest/guide. Regards, Kaushal -- _ -- 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] Missing Config Files under /etc/asterisk
Any thing i am missing ? Please suggest/guide. Hello Kaushal, try this: yum install asterisk18-configs* (You could do a 'yum list asterisk18*' to see what packages you might want/need) Regards, -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- 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] Step by step guide
On Mon, May 16, 2011 at 09:36:10PM +0530, Kaushal Shriyan wrote: I have Digium Card - Two (2) span digital T1/E1/J1/PRI PCI-Express x1 card Is there something you were looking for that is not in the user's manual [1]? [1] http://docs.digium.com/TE220/te200series-user-manual.pdf -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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 1.4.41 - Warning and Notice about contact info and stale nonce
Administrator TOOTAI wrote: Of course it's 1.4.41. And the result is that devices doesn't register anymore. Thanks for any hint. If you are installing from source, check out if some modules did not load properly due to undefined symbols. # asterisk -gvvc | tee output.txt CLI stop gracefully Then review that output.txt file. -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- 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] Light indicator managed by Asterisk
Hello,
this light indicator thing is working just great by following the same
guide as BLF (with hints).
There is just 1 thing bothering me : it is a call that is being made to
an extension, which Asterisk immediately hangs up. This makes the
IP-phone go beep beep beep beep, a normal ringtone when the other end
(Asterisk) has terminated the call.
But is there a way to give a signal to the phone that the line has not
been disconnected so it does not make this annoying beep beep beep
beep sound ? Perhaps a stupid question...
This is my dialplan :
exten = ,1,NoOp(devstate)
exten = ,n,Answer()
exten =
,n,GoToIf($[${DEVICE_STATE(Custom:light)}=BUSY]?unbusy:busy)
exten = ,n(busy),Set(DEVICE_STATE(Custom:light)=BUSY)
exten = ,n,Hangup()
exten = ,n(unbusy),Set(DEVICE_STATE(Custom:light)=NOT_INUSE)
exten = ,n,Hangup()
After the Hangup(), the IP-phone goes beep beep beep beep indicating
the call has ended. I should be glad with this ringtone signal, but not
in this case.
Kind regards,
Jonas.
On 05/12/2011 07:34 PM, Eric Wieling wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Jonas Kellens
Sent: Thursday, May 12, 2011 1:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Light indicator managed by Asterisk
On 05/12/2011 07:12 PM, Carlos Chavez wrote:
On Thu, 2011-05-12 at 18:50 +0200, Jonas Kellens wrote:
Hello,
is there some way to make Asterisk light up a certain light on an
IP-phone ?
Like MWI, the message waiting indicator can light up if there is
voicemail.
Could this light, or even other lights (like BLF-buttons)
be used to
give a visual notification to the user ?
For example : if a certain value is set in the Mysql-DB
and Asterisk
reads out this value, can Asterisk react upon it inside
the dialplan
to make a light lit up ?
2nd example : if a certain extension is called, can we
perform inside
the dialplan an action that makes a light lit up on a Snom
or Yealink
IP-phone ?
I don't know if all this is at all possible, but it doesn't harm
asking I guess...
If BLF works, then maybe more things are possible in the same way.
Just thinking outside the box here.
BLF lights can be manipulated with Hints and the
DEVSTATE function to
set custom device states. This way you can have a BLF
light react to
any event you want.
This means that extensions/hints need to be defined to be able to
control a BLF-light that monitors this extension ?
I agree that this gives some control over a light/button on
an IP-phone.
The MWI can be manipulated in several ways. Last week
someone asked
this question and got several answers.
You don't perhaps have a link to the discussion ? I don't
really follow
this list constantly so I've certainly missed out on this subject.
pbx*CLI core show application minivmmwi
-= Info about application 'MinivmMWI' =-
[Synopsis]
Send Message Waiting Notification to subscriber(s) of mailbox.
[Description]
This application is part of the Mini-Voicemail system, configured in min
ivm.conf.
MinivmMWI is used to send message waiting indication to any devices whose
channels have subscribed to the mailbox passed in the first parameter.
[Syntax]
MinivmMWI(username@domain,urgent,new,old)
[Arguments]
username
Voicemail username
domain
Voicemail domain
urgent
Number of urgent messages in mailbox.
new
Number of new messages in mailbox.
old
Number of old messages in mailbox.
[See Also]
Not available
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Re: [asterisk-users] Missing Config Files under /etc/asterisk
Thanks Jose it worked like a charm :) On Mon, May 16, 2011 at 9:52 PM, Jose P. Espinal j...@slackware-es.com wrote: Any thing i am missing ? Please suggest/guide. Hello Kaushal, try this: yum install asterisk18-configs* (You could do a 'yum list asterisk18*' to see what packages you might want/need) Regards, -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- 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] outbound calls via google voice not answered by toll free numbers with ivrs
Apologize for following up to my own question, but wanted to mention that some toll free numbers with ivrs work fine. Only run into issues with certain numbers like the test number in my previous email. Any ideas? On Fri, May 13, 2011 at 10:26 AM, Gaurav P gaurav.lists+asterisk-us...@gmail.com wrote: Hi All, I'm using Asterisk 1.8.2 with outbound calls using Google Voice. I've been having issues calling several toll free numbers where the call 'is ringing' but never transitions to 'answered'. These are toll free numbers which are typically answered by an ivrs where you enter eg. a conference bridge number. I searched google and the closest reported issues I found are - https://issues.asterisk.org/view.php?id=18319 (on 1.6.x) and https://issues.asterisk.org/view.php?id=5266 (where the ibm support number listed does not work for my setup either) The number in the second ticket can be used as a test case - 800-426-7378- and I'm hoping someone has run into this before. I have already tried both 'auto' and 'rfc2833' settings for dtmfmode and can provide any additional details about my setup. Thanks in advance! -Gaurav -- _ -- 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] Need help defining a stackexchange (i.e. stackoverflow) for telephony
It's nearly there now, just need a few more votes in order for it to trigger the next phase. Please take a moment to vote if you're interested: http://area51.stackexchange.com/proposals/12932/telephony/ On Mon, 9 May 2011, Simon P. Ditner wrote: For those of that are fans of stackoverflow.com, and stackexchange.com, there's an effort to define a telephony stackexchange site. It's still in the definition phase. What it needs to move forwards is more votes on on/off topic questions, and perhaps some better questions to vote for or against. If you're interested in helping out, or following the progress, visit: http://area51.stackexchange.com/proposals/12932/telephony/ Cheers, spd -- _ -- 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] AMI perl daemon
i was able to create a daemon that queries a database every 2 seconds for outbound calls. the daemon originates a call to a destination determined by the database. what i've noticed is, after the originate, the script never does anything else. it seems i have to use Async or the AMI will disconnect, so i tried using OriginateHack=1 but still no dice... any ideas? On Mon, May 16, 2011 at 11:37 AM, Ryan Bullock rrb3...@gmail.com wrote: Alex is pointing you in the right direction. You should want a single daemon running that then gets notified by the voicemail script, either through a FIFO, a socket, or by dropping a file in a watched directory. If you are going to write a daemon, I would suggest looking at : http://search.cpan.org/~greenbean/Asterisk-AMI-v0.2.5/ It has integration with event loops and should work well for what you are doing. It also has some features for detecting disconnects and timeouts. On Mon, May 16, 2011 at 5:42 AM, Alex Balashov abalas...@evaristesys.com wrote: On 05/16/2011 08:33 AM, vip killa wrote: Thank you, that makes sense but actually I would be invoking the script using the externnotify in voicemail.conf, similar to externnotify = /var/lib/asterisk/scripts/notify.pl http://notify.pl I assume externnotify cannot call the FastAGI server...correct? That is correct. But you can call a script that notifies the daemon through a FIFO or UNIX domain socket, if local, or network socket if remote. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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
[asterisk-users] dahdi command not available
Hi All, I have just latest branch of asterisk 1.8 and i didn't found dahdi command in CLI everything seem fine. am i missing something ? campbx2*CLI dahdi tab tab No such command 'dahdi' (type 'core show help dahdi' for other possible commands) campbx2*CLI root@campbx1:/etc/wanpipe# wanrouter hwprobe --- | Wanpipe Hardware Probe Info | --- 1 . AFT-A102-SH : SLOT=2 : BUS=7 : IRQ=3 : CPU=A : PORT=1 : HWEC=64 : V=37 2 . AFT-A102-SH : SLOT=2 : BUS=7 : IRQ=3 : CPU=A : PORT=2 : HWEC=64 : V=37 Card Cnt: A101-2=1 root@campbx2:/etc/asterisk# lsmod Module Size Used by dahdi_echocan_mg2 5662 23 wanec 381336 0 af_wanpipe 34483 0 wanpipe 813623 1 wanrouter 52003 6 wanec,af_wanpipe,wanpipe sdladrv 221273 4 wanec,af_wanpipe,wanpipe,wanrouter dahdi 210313 2 dahdi_echocan_mg2,wanpipe crc_ccitt 1675 1 dahdi fbcon 39612 71 tileblit2487 1 fbcon font8053 1 fbcon bitblit 5875 1 fbcon softcursor 1565 1 bitblit -- _ -- 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] AMI perl daemon
A normal Originate over the AMI will block all other actions until it completes. So to do other commands while the Originate is still going you have to call Originate with the Async option. I would suggest using an Originate with the 'Async' option and OriginateHack=1. If that is still not working I would have to see your code. Unfortunately I am not on irc today. On Mon, May 16, 2011 at 11:16 AM, vip killa vipki...@gmail.com wrote: i was able to create a daemon that queries a database every 2 seconds for outbound calls. the daemon originates a call to a destination determined by the database. what i've noticed is, after the originate, the script never does anything else. it seems i have to use Async or the AMI will disconnect, so i tried using OriginateHack=1 but still no dice... any ideas? On Mon, May 16, 2011 at 11:37 AM, Ryan Bullock rrb3...@gmail.com wrote: Alex is pointing you in the right direction. You should want a single daemon running that then gets notified by the voicemail script, either through a FIFO, a socket, or by dropping a file in a watched directory. If you are going to write a daemon, I would suggest looking at : http://search.cpan.org/~greenbean/Asterisk-AMI-v0.2.5/ It has integration with event loops and should work well for what you are doing. It also has some features for detecting disconnects and timeouts. On Mon, May 16, 2011 at 5:42 AM, Alex Balashov abalas...@evaristesys.com wrote: On 05/16/2011 08:33 AM, vip killa wrote: Thank you, that makes sense but actually I would be invoking the script using the externnotify in voicemail.conf, similar to externnotify = /var/lib/asterisk/scripts/notify.pl http://notify.pl I assume externnotify cannot call the FastAGI server...correct? That is correct. But you can call a script that notifies the daemon through a FIFO or UNIX domain socket, if local, or network socket if remote. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi command not available
Run Service dahdi start -Original Message- From: satish patel satish...@hotmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Mon, 16 May 2011 18:41:01 To: asterisk-usersasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi command not available -- _ -- 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] AMI perl daemon
http://pastebin.com/W5h9AMrQ anything else you need to see? On Mon, May 16, 2011 at 2:45 PM, Ryan Bullock rrb3...@gmail.com wrote: A normal Originate over the AMI will block all other actions until it completes. So to do other commands while the Originate is still going you have to call Originate with the Async option. I would suggest using an Originate with the 'Async' option and OriginateHack=1. If that is still not working I would have to see your code. Unfortunately I am not on irc today. On Mon, May 16, 2011 at 11:16 AM, vip killa vipki...@gmail.com wrote: i was able to create a daemon that queries a database every 2 seconds for outbound calls. the daemon originates a call to a destination determined by the database. what i've noticed is, after the originate, the script never does anything else. it seems i have to use Async or the AMI will disconnect, so i tried using OriginateHack=1 but still no dice... any ideas? On Mon, May 16, 2011 at 11:37 AM, Ryan Bullock rrb3...@gmail.com wrote: Alex is pointing you in the right direction. You should want a single daemon running that then gets notified by the voicemail script, either through a FIFO, a socket, or by dropping a file in a watched directory. If you are going to write a daemon, I would suggest looking at : http://search.cpan.org/~greenbean/Asterisk-AMI-v0.2.5/ It has integration with event loops and should work well for what you are doing. It also has some features for detecting disconnects and timeouts. On Mon, May 16, 2011 at 5:42 AM, Alex Balashov abalas...@evaristesys.com wrote: On 05/16/2011 08:33 AM, vip killa wrote: Thank you, that makes sense but actually I would be invoking the script using the externnotify in voicemail.conf, similar to externnotify = /var/lib/asterisk/scripts/notify.pl http://notify.pl I assume externnotify cannot call the FastAGI server...correct? That is correct. But you can call a script that notifies the daemon through a FIFO or UNIX domain socket, if local, or network socket if remote. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] AMI perl daemon
Calling -action() is going to wait for the originate to finish (even if you use 'Async'). I think the default timeout for Originate is 60 seconds or so before it fails. I would recommend reading up on: http://search.cpan.org/~mlehmann/AnyEvent-5.34/ http://search.cpan.org/~mlehmann/AnyEvent-DBI-2.1/ You could us a timer to periodically poll your database and do non-blocking originates (with async) with callbacks to catch the response, update the log, and do the delete. On Mon, May 16, 2011 at 11:49 AM, vip killa vipki...@gmail.com wrote: http://pastebin.com/W5h9AMrQ anything else you need to see? On Mon, May 16, 2011 at 2:45 PM, Ryan Bullock rrb3...@gmail.com wrote: A normal Originate over the AMI will block all other actions until it completes. So to do other commands while the Originate is still going you have to call Originate with the Async option. I would suggest using an Originate with the 'Async' option and OriginateHack=1. If that is still not working I would have to see your code. Unfortunately I am not on irc today. On Mon, May 16, 2011 at 11:16 AM, vip killa vipki...@gmail.com wrote: i was able to create a daemon that queries a database every 2 seconds for outbound calls. the daemon originates a call to a destination determined by the database. what i've noticed is, after the originate, the script never does anything else. it seems i have to use Async or the AMI will disconnect, so i tried using OriginateHack=1 but still no dice... any ideas? On Mon, May 16, 2011 at 11:37 AM, Ryan Bullock rrb3...@gmail.com wrote: Alex is pointing you in the right direction. You should want a single daemon running that then gets notified by the voicemail script, either through a FIFO, a socket, or by dropping a file in a watched directory. If you are going to write a daemon, I would suggest looking at : http://search.cpan.org/~greenbean/Asterisk-AMI-v0.2.5/ It has integration with event loops and should work well for what you are doing. It also has some features for detecting disconnects and timeouts. On Mon, May 16, 2011 at 5:42 AM, Alex Balashov abalas...@evaristesys.com wrote: On 05/16/2011 08:33 AM, vip killa wrote: Thank you, that makes sense but actually I would be invoking the script using the externnotify in voicemail.conf, similar to externnotify = /var/lib/asterisk/scripts/notify.pl http://notify.pl I assume externnotify cannot call the FastAGI server...correct? That is correct. But you can call a script that notifies the daemon through a FIFO or UNIX domain socket, if local, or network socket if remote. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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
Re: [asterisk-users] AMI perl daemon
On 05/16/2011 03:19 PM, Ryan Bullock wrote: You could us a timer to periodically poll your database and do non-blocking originates (with async) with callbacks to catch the response, update the log, and do the delete. http://en.wikipedia.org/wiki/Anti-pattern http://en.wikipedia.org/wiki/Database-as-IPC *ducks* :-) -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI perl daemon
could you suggest a better method where the perl-daemon stays persistently connected to asterisk's AMI ? On Mon, May 16, 2011 at 3:21 PM, Alex Balashov abalas...@evaristesys.comwrote: On 05/16/2011 03:19 PM, Ryan Bullock wrote: You could us a timer to periodically poll your database and do non-blocking originates (with async) with callbacks to catch the response, update the log, and do the delete. http://en.wikipedia.org/wiki/Anti-pattern http://en.wikipedia.org/wiki/Database-as-IPC *ducks* :-) -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] AMI perl daemon
On 05/16/2011 03:35 PM, vip killa wrote: could you suggest a better method where the perl-daemon stays persistently connected to asterisk's AMI ? It is not the AMI connection that is under discussion. The AMI connection will be over a TCP socket regardless, because that is the nature of the interface. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI perl daemon
yes, my problem is i would like a persistent connection to AMI because there will maybe be 20+ originates per second. On Mon, May 16, 2011 at 3:37 PM, Alex Balashov abalas...@evaristesys.comwrote: On 05/16/2011 03:35 PM, vip killa wrote: could you suggest a better method where the perl-daemon stays persistently connected to asterisk's AMI ? It is not the AMI connection that is under discussion. The AMI connection will be over a TCP socket regardless, because that is the nature of the interface. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] AMI perl daemon
On 05/16/2011 03:38 PM, vip killa wrote: yes, my problem is i would like a persistent connection to AMI because there will maybe be 20+ originates per second. What was wrong with Ryan's original Async suggestion to address that? In other words, what is currently the problem? I looked at your code and it seems to me that your while() loop is going to exit when all rows are retrieved from your 'active' table as part of the current SELECT transaction. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI perl daemon
you are incorrect, the while loop never exits...and i already have async=1 On Mon, May 16, 2011 at 3:41 PM, Alex Balashov abalas...@evaristesys.comwrote: On 05/16/2011 03:38 PM, vip killa wrote: yes, my problem is i would like a persistent connection to AMI because there will maybe be 20+ originates per second. What was wrong with Ryan's original Async suggestion to address that? In other words, what is currently the problem? I looked at your code and it seems to me that your while() loop is going to exit when all rows are retrieved from your 'active' table as part of the current SELECT transaction. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] AMI perl daemon
On 05/16/2011 03:44 PM, vip killa wrote: you are incorrect, the while loop never exits...and i already have async=1 I meant this while() loop: while(my @row = $res-fetchrow()) In other words, it is possible for additional rows to be added to the table after the SELECT query has been made and prior to the conclusion of this loop, which will incur a sleep(2) penalty. That's not terribly important, though. So, just to make sure I understand, you're calling the $astman-action with the Originate parameters and it's sitting there and blocking until the Originate completes with a final disposition, despite the presence of the Async option? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI perl daemon
yes, it's originating the call and never responding. On Mon, May 16, 2011 at 3:47 PM, Alex Balashov abalas...@evaristesys.comwrote: On 05/16/2011 03:44 PM, vip killa wrote: you are incorrect, the while loop never exits...and i already have async=1 I meant this while() loop: while(my @row = $res-fetchrow()) In other words, it is possible for additional rows to be added to the table after the SELECT query has been made and prior to the conclusion of this loop, which will incur a sleep(2) penalty. That's not terribly important, though. So, just to make sure I understand, you're calling the $astman-action with the Originate parameters and it's sitting there and blocking until the Originate completes with a final disposition, despite the presence of the Async option? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] AMI perl daemon
On 05/16/2011 03:48 PM, vip killa wrote:
yes, it's originating the call and never responding.
This sounds to me like a possible problem with the Asterisk::AMI module,
although I am unsure what the problem is, since I am not familiar with
its internal architecture and have never used it.
To debug, try initiating the AMI connection and issuing the Originate
statement raw:
use IO::Socket;
...
my $mgr_sock = IO::Socket::INET-new(
'PeerAddr' = '127.0.0.1',
'PeerPort' = 5038,
'Type' = SOCK_STREAM,
'Protocol' = 'TCP',
'Timeout' = 5);
print $mgr_sock
Action: login\r\n .
Username: XX\r\n .
Secret: XX\r\n .
\r\n;
sleep(1);
...
print $mgr_sock
Action: Originate\r\n .
Channel: Local/$row[2]\@outbound\r\n .
Context: page\r\n .
CallerID: $row[1]\r\n .
Exten: $row[1]\r\n .
Priority: 1\r\n .
Async: 1\r\n .
\r\n;
while(defined($mgr_sock) $_ = $mgr_sock) {
print;
}
sleep(1);
close $mgr_sock;
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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Re: [asterisk-users] AMI perl daemon
Umm thank you...apparently AMI::Asterisk sucks because that code did
everything i needed in one try. thanks again!
On Mon, May 16, 2011 at 3:58 PM, Alex Balashov abalas...@evaristesys.comwrote:
On 05/16/2011 03:48 PM, vip killa wrote:
yes, it's originating the call and never responding.
This sounds to me like a possible problem with the Asterisk::AMI module,
although I am unsure what the problem is, since I am not familiar with its
internal architecture and have never used it.
To debug, try initiating the AMI connection and issuing the Originate
statement raw:
use IO::Socket;
...
my $mgr_sock = IO::Socket::INET-new(
'PeerAddr' = '127.0.0.1',
'PeerPort' = 5038,
'Type' = SOCK_STREAM,
'Protocol' = 'TCP',
'Timeout' = 5);
print $mgr_sock
Action: login\r\n .
Username: XX\r\n .
Secret: XX\r\n .
\r\n;
sleep(1);
...
print $mgr_sock
Action: Originate\r\n .
Channel: Local/$row[2]\@outbound\r\n .
Context: page\r\n .
CallerID: $row[1]\r\n .
Exten: $row[1]\r\n .
Priority: 1\r\n .
Async: 1\r\n .
\r\n;
while(defined($mgr_sock) $_ = $mgr_sock) {
print;
}
sleep(1);
close $mgr_sock;
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
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Re: [asterisk-users] AMI perl daemon
question... how reliable is what you wrote? as long as the daemon is running
will the AMI stay connected?
On Mon, May 16, 2011 at 4:08 PM, vip killa vipki...@gmail.com wrote:
Umm thank you...apparently AMI::Asterisk sucks because that code did
everything i needed in one try. thanks again!
On Mon, May 16, 2011 at 3:58 PM, Alex Balashov
abalas...@evaristesys.comwrote:
On 05/16/2011 03:48 PM, vip killa wrote:
yes, it's originating the call and never responding.
This sounds to me like a possible problem with the Asterisk::AMI module,
although I am unsure what the problem is, since I am not familiar with its
internal architecture and have never used it.
To debug, try initiating the AMI connection and issuing the Originate
statement raw:
use IO::Socket;
...
my $mgr_sock = IO::Socket::INET-new(
'PeerAddr' = '127.0.0.1',
'PeerPort' = 5038,
'Type' = SOCK_STREAM,
'Protocol' = 'TCP',
'Timeout' = 5);
print $mgr_sock
Action: login\r\n .
Username: XX\r\n .
Secret: XX\r\n .
\r\n;
sleep(1);
...
print $mgr_sock
Action: Originate\r\n .
Channel: Local/$row[2]\@outbound\r\n .
Context: page\r\n .
CallerID: $row[1]\r\n .
Exten: $row[1]\r\n .
Priority: 1\r\n .
Async: 1\r\n .
\r\n;
while(defined($mgr_sock) $_ = $mgr_sock) {
print;
}
sleep(1);
close $mgr_sock;
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
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Re: [asterisk-users] AMI perl daemon
On 05/16/2011 04:08 PM, vip killa wrote: Umm thank you...apparently AMI::Asterisk sucks because that code did everything i needed in one try. thanks again! Awesome! Happy to help. A more sophisticated and high-performance version of this--required for scaling out to multiple Asterisk servers or higher dialing pace--would involve a daemon that maintains multiple AMI connections, uses non-blocking sockets and synchronous I/O multiplexing (i.e. IO::Select) to mux I/O from those connections as well as some sort of IPC control socket/interface which can be hit with originate requests, instead of using a database table for that purpose. An HTTP API would be a good way to do that from the dial plan. The package HTTP::Server::Simple::CGI (available from CPAN) is a good way to add a minimalistic web server thread. Then you can issue CURL calls from the dial plan to it, and it can go pick an Asterisk AMI connection over which to issue the Originate, and possibly even return a success or failure via HTTP if the connection is held open. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI perl daemon
On 05/16/2011 04:10 PM, vip killa wrote: question... how reliable is what you wrote? as long as the daemon is running will the AMI stay connected? I don't know, it was kind of off-the-cuff. I would probably throw a while() loop around it to reconnect if the connection is lost. But I see no reason why it should disconnect unless the Asterisk AMI service has some sort of inactivity timeout. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI perl daemon
forgive me for i am very new to asterisk and perl. but how could you detect if you were disconnected from AMI? On Mon, May 16, 2011 at 4:14 PM, Alex Balashov abalas...@evaristesys.comwrote: On 05/16/2011 04:10 PM, vip killa wrote: question... how reliable is what you wrote? as long as the daemon is running will the AMI stay connected? I don't know, it was kind of off-the-cuff. I would probably throw a while() loop around it to reconnect if the connection is lost. But I see no reason why it should disconnect unless the Asterisk AMI service has some sort of inactivity timeout. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] AMI perl daemon
On 05/16/2011 04:17 PM, vip killa wrote:
forgive me for i am very new to asterisk and perl. but how could you
detect if you were disconnected from AMI?
if(defined($mgr_sock)) would evaluate to false. That's all you need to
do with the plain vanilla blocking I/O you're using now.
Down the road, if non-blocking I/O is set, there are other strategies.
The traditional way was an ioctl() FIONREAD that returned 0 for the
bytes value, though the ioctl() call did not fail:
require 'sys/ioctl.ph';
...
my $bytes_waiting = pack(L, 0);
ioctl($mgr_sock, FIONREAD(), $bytes_wating);
$bytes_waiting = unpack(L, $bytes_waiting);
if($bytes_waiting == 0) {
# Far-end disconnected.
close($mgr_sock);
return;
}
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] Voicemail Configuration
Thanks, that's given me some ideas. I don't think I can totally roll my
own, since I also make use of the MWI features of voicemail. Another
thread pointed out the existence of minivm, which I hadn't realized was
available. I just need to find the time to play around with some of the
proposed options.
John
On 5/14/2011 10:46 PM, virendra bhati wrote:
this will help you..
; DIY VOICEMAIL
*[ck987_vm_record]*
;start recording after the beep. Press # when done.
exten = 1,1,Playback(/home/ck987/asterisk_sounds/vm-record-start)
;build this call's recorded message file nameuniqueID_phone number
;every call is assigned a unique id.
exten = 1,n,Set(record_file=${UNIQUEID}_${CALLERID(num)})
; records into my vm_msg folder.
;Ends if # is hit, silence for 2 secs, or recording lasts for 60 seconds
exten = 1,n,Record(/home/ck987/asterisk_sounds/vm_msg/${record_file}.wav,2,60)
; 2 to review message, 3 to re record, or hang up
exten = 1,n,Background(/home/ck987/asterisk_sounds/vm-record-end)
exten = 1,n,WaitExten(5)
exten = 1,n,Playback(/home/ck987/asterisk_sounds/bye)
exten = 1,n,Hangup()
exten = 2,1,Playback(/home/ck987/asterisk_sounds/vm_msg/${record_file})
; press 1 to re-record, or hangup if satisfied
exten = 2,n,Background(/home/ck987/asterisk_sounds/vm-record-again)
exten = 2,n,WaitExten(5)
;no response, hang up on person.
exten = 2,n,Playback(/home/ck987/asterisk_sounds/bye)
exten = 2,n,Hangup()
;go back to the record option
exten = 3,1,Goto(ck987_vm_record,1,1)
;--DIY VOICEMAIL ADMIN--
*[ck987_vm_admin]*
; pass is 9988, jump to n+101 if authentication fails, expect 4 digits
exten = 1,1,Authenticate(9988,j,4)
; get number of voicemail messages
; SHELL function returns the output from a system command
;ls -1 lists visible files in a list.wc -l will count how many lines
there are. 1 line per file!
exten = 1,n,Set(num_messages=${SHELL(ls -1
/home/ck987/asterisk_sounds/vm_msg/ | wc -l)})
;you have...
exten = 1,n,Playback(/home/ck987/asterisk_sounds/you-have)
exten = 1,n,SayDigits(${num_messages})
;...messages!
exten = 1,n,Playback(/home/ck987/asterisk_sounds/messages)
;get file names.sed command trims off any .wav exten.tr command trims off
whitespace and line feeds.
exten = 1,n,Set(file_names=${SHELL(ls -m /home/ck987/asterisk_sounds/vm_msg/
| sed's/.wav//g' | tr -d' \n')})
;start message counter
exten = 1,n,Set(msg_counter=1)
; 1 to repeat message, 3 to go to the next message, 7 to go to the previous
message
exten = 1,n,Playback(/home/ck987/asterisk_sounds/msg-options)
exten = 1,n,Goto(ck987_vm_play_message,1,1) ; playback loop
;try again if password is wrong
exten = 1,102,Goto(ck987_vm_admin,1,1)
*[ck987_vm_play_message]*
exten = 1,1,Background(beep)
exten = 1,n,Set(current_message=${CUT(file_names,\,,${msg_counter})})
exten = 1,n,NoOp(${file_names} ${current_message} ${msg_counter})
exten = 1,n,Background(/home/ck987/asterisk_sounds/vm_msg/${current_message})
exten = 1,n,Goto(3,1)
; next message: add 1 and go back to the top of the loop
exten = 3,1,Set(msg_counter=$[${msg_counter} + 1])
exten = 3,n,Set(msg_counter=${IF($[${msg_counter}
${num_messages}]?${num_messages}:${msg_counter})})
exten = 3,n,Goto(1,1)
; previous message: subtract 1 and go back to the top of the loop
exten = 7,1,Set(msg_counter=$[${msg_counter} - 1])
;make sure number never goes below 1
exten = 7,n,Set(msg_counter=${IF($[${msg_counter} 1]?1:${msg_counter})})
exten = 7,n,Playback(/home/ck987/asterisk_sounds/previous_message)
exten = 7,n,Goto(1,1)
;delete message
exten = *,1,System(rm /home/ck987/asterisk_sounds/vm_msg/${current_message})
;number of files has changed. reload number of files and file names
;ls -1 lists visible files in a list.wc -l will count how many lines
there are. 1 line per file!
exten = *,n,Set(num_messages=${SHELL(ls -1
/home/ck987/asterisk_sounds/vm_msg/ | wc -l)})
;get file names.sed command trims off any .wav exten.tr command trims off
whitespace and line feeds.
exten = *,n,Set(file_names=${SHELL(ls -m /home/ck987/asterisk_sounds/vm_msg/
| sed's/.wav//g' | tr -d' \n')})
exten = *,n,Goto(1,1)
it's the actual voicemail example which is use by asterisk it self
On Tue, May 10, 2011 at 2:53 AM, John Marvin jm-aster...@themarvins.org
mailto:jm-aster...@themarvins.org wrote:
On 5/9/2011 3:08 PM, Roger Burton West wrote:
You could use Monitor to record the whole call, then use an AGI
to do
something with it on hangup if the other conditions haven't been
satisfied...?
I understand how to do the first part, and I at least understand
that I could do something fancy with the AGI capability. But what I
don't know is how I can take the recording and insert it into a
voicemail box such that it can be retrieved through the normal
VoiceMailMain mechanism.
Re: [asterisk-users] Asterisk 1.4.41 - Warning and Notice about contact info and stale nonce
Le 16/05/2011 18:27, Jose P. Espinal a écrit : Administrator TOOTAI wrote: Of course it's 1.4.41. And the result is that devices doesn't register anymore. Thanks for any hint. If you are installing from source, check out if some modules did not load properly due to undefined symbols. # asterisk -gvvc | tee output.txt CLI stop gracefully Then review that output.txt file. Don't think that the problem is here: the devices are working well with previous version of asterisk on the same server. Also, other devices from other manufacturer are still working ok. Question is why auth is OK but registration failed? On 1.4.40 we juste had to change the device local port (eg from 5061 to 5062) and registration was OK. On 1.4.41 this trick is no more working. And stale nonce should have an end of life in our mind, but doesn't. Thanks for your tip. -- 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
[asterisk-users] Reporting Tool: To show who is login, queue, ... etc
Hi All; It look like there are some free (open source) tools that are used for Asterisk reporting special for call center (to see number of agents logged in, number of calls now, .. etc), and to be used as dashboard. Can someone direct me for something really is suitable and stable? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reporting Tool: To show who is login, queue, ... etc
On 05/16/2011 05:54 PM, bilal ghayyad wrote: It look like there are some free (open source) tools that are used for Asterisk reporting special for call center (to see number of agents logged in, number of calls now, .. etc), and to be used as dashboard. Can someone direct me for something really is suitable and stable? The best package for this is commercial, but quite inexpensive: QueueMetrics. It's quite worth it. http://www.queuemetrics.com/ -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not
Seriously guys. Why would anyone other than the two of you need to read this. It's a personal conversation. We all know who you both are and your achievements etc. The longer the conversation goes on the more off topic it becomes :-) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] 1.8.4 Core Dump after installing from source
On 13/05/11 4:38 PM, Jose P. Espinal wrote: Hello, After installing Asterisk from source in Slackware 13.1, I get the following error: Error loading module 'res_config_odbc.so': /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: ast_odbc_clear_cache Then a core dump. Do a backtrace on the core dump. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] iptables for Asterisk - Any good guides out there?
On 15/05/11 12:40 PM, Steve Edwards wrote: Adding a couple of lines to root's crontab like: # Min hour DOM month DOW command # -- # */5 * * * * /etc/init.d/iptables stop make it easy to enable an 'iptables failsafe' (by un-commenting the last line) while you're fiddling about. What a great idea! I've never thought of doing that! -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] AMI check if connection is alive
On 17/05/11 3:25 AM, vip killa wrote: I'm using a perl daemon i wrote to connect to AMI and perform actions. The daemon connects to asterisk via AMI at start up. Is there anyway to check if the AMI connection is still alive, for example every 2 seconds. if the connection is not alive, re-connect to AMI? Also, does AMI timeout after a certain amount of time of not sending commands? Send an Action: Ping\r\n\r\n command. You should receive a response. Run a timer on it and if you don't get a response reconnect. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] 3. Re: ITSP Multi IPs (Alex Balashov) Asterisk-users Digest, Vol 82, Issue 33
Alex,
Thank you so much for your response. I've been so consumed with other
business that I only just now getting back to this issue. We have
implemented your suggestion which is perfect. Thank you again.
I've never asked a question of the community before and I'm extremely happy
with the rapid response I received.
Somewhat related to this initial problem I have an additional
problem.
In extensions.conf we have identified one ITSP for sending outbound calls
to. From the problems we've been having with our IPSPs being unavailable at
different times we need to be able to send calls to ITSPs that are available
in a cascading fashion.
I would like to know how to configure extensions.conf so that calls will
attempt in the cascading fashion sending calls to our our ITSPs until one
excepts the call.
I was thinking of implementing something like this, but wonder if I should
add some sort of timer or delay? I would like to call to the excepted by
the first available ITSP in some order.
; send all outgoing calls directly to the ITSP
exten = _1NXXNXX,n,Dial(SIP/${EXTEN}@ITSP-one)
exten = _1NXXNXX,n,Dial(SIP/${EXTEN}@ITSP-two)
exten = _1NXXNXX,n,Dial(SIP/${EXTEN}@ITSP-three)
Thank you for taking a look,
Claude
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Re: Contents of asterisk-users digest...
Today's Topics:
1. Re: QueueCallerAbandon is not triggering after1.8.3.3...
(Louis Carreiro)
2. ITSP Multi IPs (Claude Hayn)
3. Re: ITSP Multi IPs (Alex Balashov)
4. 1.8 and prematuremedia problem (d tbsky)
5. Re: 1.8 and prematuremedia problem (Satish Patel)
6. Re: 1.8 and prematuremedia problem (d tbsky)
7. Re: Background music during a call (Rizwan Hisham)
8. Re: OT - Which Android handset with Wifi-only ? (Olivier)
9. Re: OUTBOUND CALLER ID (mahesh katta)
10. Re: OUTBOUND CALLER ID (DHAVAL INDRODIYA)
11. Re: OUTBOUND CALLER ID (mahesh katta)
12. Re: 40sec between dial execution and sending SIP request
(Pezhman Lali)
--
Message: 1
Date: Mon, 9 May 2011 20:54:39 -0400
From: Louis Carreiro carreir...@gmail.com
Subject: Re: [asterisk-users] QueueCallerAbandon is not triggering
after 1.8.3.3...
To: Asterisk Users asterisk-users@lists.digium.com
Message-ID: BANLkTi=h24l6mu-fxn0cc-fbpqgxfrj...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
Has anyone else noticed this?
v/r,
Me
On Fri, May 6, 2011 at 12:11 PM, Louis Carreiro carreir...@gmail.comwrote:
Has anyone else noticed that QueueCallerAbandon is not showing up in
the AMI after the 1.8.3.3? Am I missing something? I'm getting what
seems like everything else but QueueCallerAbandon.
v/r,
Me
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Message: 2
Date: Mon, 9 May 2011 21:12:53 -0400
From: Claude Hayn chayn...@gmail.com
Subject: [asterisk-users] ITSP Multi IPs
To: asterisk-users@lists.digium.com
Message-ID: 00c601cc0eaf$642a4820$2c7ed860$@gmail.com
Content-Type: text/plain; charset=us-ascii
Hi,
I'm hoping someone has a suggestion for us.
We have an ITSP that sends inbound traffic to us. Unannounced to us last
week they started alternately sending traffic from two IP addresses, instead
of the one we knew about. Some calls would pass, and others would be dumped
as unauthenticated.
I added the 2nd IP to the sip.conf file to allow for this, and everything
was fine until this morning.
This morning the first IP started being rejected even though it was listed
in the sip.conf file. As soon as I commented out the .45 IP address that I
added last week traffic flowed again.
We need to authorize traffic from both IP addresses.
Any suggestions?
Sip.conf:
[incoming-trunk]
type=peer
insecure=port,invite
context=default-incoming
host=XX.XXX.XXX.40
canreinvite=nonat
qualify=yes
disallow=all
allow=ulaw
nat=yes
[incoming-trunk]
type=peer
insecure=port,invite
context=default-incoming
host=XX.XXX.XXX.45
canreinvite=nonat
qualify=yes
disallow=all
Re: [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so
On Mon, May 16, 2011 at 10:27 AM, satish patel satish...@hotmail.comwrote: Thanks Leif, I had changed it to res_timing_dahdi and since last few days it seem good. -S Date: Sun, 15 May 2011 15:48:03 -0400 From: leif.mad...@asteriskdocs.org To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] res_timing_timerfd.so Vs res_timing_dahdi.so On 11-05-13 11:39 AM, isr...@gmail.com wrote: I haven't tried with timerfd but with timer pthread 1.8 is very unstable I think I have seen a post to the list from kevin fleming that the same is for timerfd that there is a nasty bug which they haven't found the reason for yet My experience is that you should pretty much always use res_timing_dahdi unless you're on a platform on which you can't install DAHDI. You don't need any hardware to use timing from DAHDI because timing is generated by the kernel. My order of preference for stability is: * res_timing_dahdi * res_timing_timerfd * res_timing pthread The timerfd and pthread modules are relatively new, and sometimes people run into stability problems while using them. If you can use res_timing_dahdi I recommend you do so. Leif. following this advice, is there a quick and minimal way to install/use res_timing_dahdi without having to build/compile/install the whole dahdi package and all the other modules associated with it? back in the zaptel days, I used to be able to modify the Makefile and compile JUST the ztdummy module to provide timing for meetme. Haven't touched * for a while esp. Zaptel/Dahdi, so not sure how it works anymore. I'm assuming to get res_timing_dahdi, I need dahdi_dummy installed at the very least? Do I need the kernel source packages like in the old days to compile DAHDI against the Kernel etc? Thx so much -- _ -- 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] iptables for Asterisk - Any good guides out there?
On 15/05/11 12:40 PM, Steve Edwards wrote: Adding a couple of lines to root's crontab like: # Min hour DOM month DOW command # -- # */5 * * * * /etc/init.d/iptables stop make it easy to enable an 'iptables failsafe' (by un-commenting the last line) while you're fiddling about. On Tue, 17 May 2011, Matt Riddell wrote: What a great idea! I've never thought of doing that! I wish I could take credit for it :) I had a similar 'gee, how obvious' epiphany after having locked myself out of way too many hosts. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iptables for Asterisk - Any good guides out there?
On 17/05/11 1:36 PM, Steve Edwards wrote: On 15/05/11 12:40 PM, Steve Edwards wrote: Adding a couple of lines to root's crontab like: # Min hour DOM month DOW command # -- # */5 * * * * /etc/init.d/iptables stop make it easy to enable an 'iptables failsafe' (by un-commenting the last line) while you're fiddling about. On Tue, 17 May 2011, Matt Riddell wrote: What a great idea! I've never thought of doing that! I wish I could take credit for it :) I had a similar 'gee, how obvious' epiphany after having locked myself out of way too many hosts. Yeah exactly - hence my excitement at the idea :-) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] res_timing_timerfd.so Vs res_timing_dahdi.so
On Mon, May 16, 2011 at 09:26:48PM -0400, A E [Gmail] wrote: following this advice, is there a quick and minimal way to install/use res_timing_dahdi without having to build/compile/install the whole dahdi package and all the other modules associated with it? back in the zaptel days, I used to be able to modify the Makefile and compile JUST the ztdummy module to provide timing for meetme. Haven't touched * for a while esp. Zaptel/Dahdi, so not sure how it works anymore. In the dahdi-linux package you can edit drivers/dahdi/Kbuild and comment out every module except for dahdi.ko. So looking in that file you will see something like: obj-$(DAHDI_BUILD_ALL)$(CONFIG_DAHDI) += dahdi.o #obj-$(DAHDI_BUILD_ALL)$(CONFIG_DAHDI_DUMMY)+= dahdi_dummy.o obj-$(DAHDI_BUILD_ALL)$(CONFIG_DAHDI_DYNAMIC) += dahdi_dynamic.o obj-$(DAHDI_BUILD_ALL)$(CONFIG_DAHDI_DYNAMIC_LOC) += dahdi_dynamic_loc.o Here dahdi_dummy is commented out. Just comment out all the other modules (lines that start with obj-) and leave only dahdi.o. dahdi.ko now automatically monitors the spans and if there isn't one providing timing, it will use the built in timing source which functions very similarly to dahdi dummy of the past. I'm assuming to get res_timing_dahdi, I need dahdi_dummy installed at the very least? Since dahdi-linux 2.3.0, all you need is dahdi.ko. There is no more dahdi_dummy module required unless you specifically install it. Do I need the kernel source packages like in the old days to compile DAHDI against the Kernel etc? You will still need the kernel sources to compile dahdi.ko against. Also when you install dahdi-tools, you will want to comment out all the lines in /etc/dahdi/modules so that the init script does not try to load any of the board drivers. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] When someone helps you, at least let them know if the problem is resolved or not
I like puppies On Mon, May 16, 2011 at 8:05 PM, Matt Riddell li...@venturevoip.com wrote: Seriously guys. Why would anyone other than the two of you need to read this. It's a personal conversation. We all know who you both are and your achievements etc. The longer the conversation goes on the more off topic it becomes :-) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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 -- Sherwood McGowan Telecommunications and VOIP Consultant -- _ -- 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
