Re: [asterisk-users] How to generate core dump?
Ken D'Ambrosio wrote: Asterisk segfaulted on me the other day; how do I tell it to generate a core file so -- if it happens again -- I can attempt to debug? I looked in the obvious places in make menuconfig and didn't see anything appropriate. There should be a file called something like backtrace.txt that gives you instructions on how to compile with debugging options, generate a coredump when Asteriss crashes, as well as information on how to generate a backtrace to determine where the problem is located. This file would be in the doc/ directory in your Asterisk source. -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call file concurrency
Set the ctime of the spool file in the future and Asterisk will not process the file until that time. Danny Nicholas wrote: Here is a link to a better, but possibly dangerous answer. http://www.netadmintools.com/art295.html Since a typical linux box probably allows about 250K files to be simultaneously open, and you need about 2K for system and * overhead, by cutting the max number of files down to about 3K, you would limit the number of calls to about 1K, assuming that each open call is one file handle. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, February 27, 2009 11:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] call file concurrency Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? On Fri, 27 Feb 2009, Danny Nicholas top posted: -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd Read App Issues
Robert Broyles wrote: So I'm using the READ() application within an IVR, and having a strange issue, and wondering if anyone else has had this problem. When calling from an outside line, and entering the digits during the read() part of my dialplan, it's accepting some of the digits twice, though it's only keyed in once. When testing the dialplan internally, it accepts only the digits that I key in. Anyone else experienced this? Yes. Most of the time it is either because I put relaxdtmf=yes in zapata.conf or because my rxgain is too low on that port. -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd Read App Issues
Robert Broyles wrote: Okay. I'm using this all over SIP Trunking with Vitelity. Any other suggestions? Sorry for wasting your time. -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE121 on Asterisk
E-1s are 30 channels with D-Channel on 16. to make it work as E1, if i write a new span like span=1,1,0,ccs,hdb3,crc4 i got the following error when i type dahdi_cfg dahdi_cfg - DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): MG2 Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 08) Channel 09: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 11) Channel 12: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 12) Channel 13: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 13) Channel 14: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 14) Channel 15: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 15) Channel 16: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 16) Channel 17: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 17) Channel 18: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 18) Channel 19: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 19) Channel 20: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 20) Channel 21: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 21) Channel 22: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 22) Channel 23: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 23) Channel 24: D-channel (Default) (Slaves: 24) 24 channels to configure. DAHDI_SPANCONFIG failed on span 1: Invalid argument (22) How can i set a working E1. Any examples will welcome.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gosub behavior change =1.6.0.5 to 1.6.0.6
Why not expand the usage of the i extension? If not in 1.6.0, then some later 1.6. Call it a feature enhancement. Tilghman Lesher wrote: On Wednesday 25 February 2009 11:19:08 sean darcy wrote: Tilghman Lesher wrote: On Wednesday 25 February 2009 09:51:23 Klaus Darilion wrote: Tilghman Lesher schrieb: On Tuesday 24 February 2009 16:07:52 Klaus Darilion wrote: Barry L. Kline wrote: that is supposed to gosub into the incoming extension at priority 1. Versions before 1.6.0.6 would drop into the incoming,i,1 priority if the requested extension wasn't present in the incoming context. Really strange that Goto and Gosub behave different. If Goto behaves that way, that's a bug. As stated in a prior email, the i extension should only be implicitly invoked when waiting for a new extension and the typed extension does not match anything. FYI: If you take a look at the history of http://www.voip-info.org/wiki/index.php?page=Asterisk+i+extension you will find out that the old behavior is there since at least Nov. 2005, and probably used since then. voip-info.org is best known for being often wrong. I think the point being made was that a lot of people thought this was a feature, not a bug. I assume you're asserting the the dev's did not expect this behaviour, even if a large group of users did. That's OK. But there's still the question about why this behaviour is so bad/inconsistent/something that it should be changed. Simply labeling it a bug is just a conclusion. Why is it a bug??? It's a bug, because the i extension has a very limited intended usage, and any additional cases where the i extension is implicitly invoked is therefore a bug. This thread has convinced me not to change Goto in 1.6.0, but I absolutely defend fixing this bug in Gosub, given that I'm the designer of it, and it was never supposed to fail into the i extension. -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI problem using mono (.Net)
Douglas Mortensen wrote: I have a software developer creating a .Net / mono program to use as an AGI script. We are having problems getting it to stream files. From what we can tell, it is talking to asterisk correctly when called from the dial plan. Its stderr output goes to the asterisk console. But asterisk doesn't give any indication that it receives the STREAM FILE command. Asterisk simply quickly executes the program and moves to the next step of the dial plan, as though it didn't receive any commands from the program. We know it is running, and outputting its results, because we have called it from within a bash script, and in doing so, I set the script to output stdout to a txt file for testing (like this /var/log/asterisk/querylog). When we do this, the file does end up with the first line showing STREAM FILE filename. We're at a bit of a loss as to what's going on. We have checked filenames and are pretty sure that there are no typos and that the files are there. Further, I have a perl agi script using asterisk::agi that also does a STREAM FILE which runs without any problem. In our dial plan, my perl script runs, gets data from the user via the keypad, puts it in a channel variable, then exits, and his AGI script is immediately called as the next step of the dial plan receiving the channel variable as an argument. STDERR only goes to the Asterisk console if you are running 1.4 or later and enable agi debug in the CLI. I seem to recall something about AGIs not working correctly (streamfile or DTMF read) if your AGI script does not process the input Asterisk sends it on STDIN when Asterisk starts the AGI. I don't know if it applies here, but it's worth looking at. -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM codec is a good choice ???
Tiago Durante wrote: On Tue, Feb 24, 2009 at 10:41 AM, Olivier oza-4...@myamail.com wrote: 2009/2/24 Alejandro Cabrera Obed aco1...@gmail.com Thanks for your comment about codecsI tell you I can't use G.711 because I use a WAN link, and this is a wide band codec. Is GSM codec totally free (avoid to pay for any license) ??? yes ! Is there a WAN between your hardphones and Asterisk ? GSM is a great and free codec. However if your phones doesn't have it, you'll have to buy the g729 licenses... How much would new phones cost? If they cost more than $10 each then you will spend more money by buying new phones rather than buying a G729 license. Remember G729 is licensed on a per simultaneous channel basis. i.e. if you buy 10 licenses you can have up to 10 G729 channels in use at any one time. You do not need one license per phone. -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] what is the correct character to separate application parameters: , or |
Versions before 0.65 I don't know about From 0.65 to 1.4.x you can use either , or | In 1.6 you must use , (| was removed) Klaus Darilion wrote: Hi! I see lots of examples using , but core show application displays | So what is the correct character to use to separate parameters for application, functions and macros? -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail and ADSI
Jeff LaCoursiere wrote: Anyone using this feature of asterisk's voicemail? I'd never heard of ADSI, and saw it as I was perusing the voicemail source this morning. Is it some kind of visual way of managing voicemail on your phone's display, or does it require a terminal of some kind? It requires an ADSI analog phone. It's not nearly as cool as it sounds like. IIRC, ADSI uses 2400bps FSK bursts to do what it does. It does not help that most ADSI phones seem to be locked by the telco or the vendor. -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel telephone cards and asterisk in another pc
Ignacio wrote: I have some zaptel cards, and I would like to install them in some user's computers. Is there any way to connect those cards with asterisk server (which is in another computer)? All manuals I have read explain how to connect asterisk and zaptel cards in the same computers, but not on different ones. You can install Asterisk on the PCs with the cards in them and then use SIP or IAX2 to transport those calls to your main Asterisk server. -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call file FXO channel problem
Ray Chen wrote: I have problem of using call file to make auto outbound dial through FXO channel. I put Channel: DAHDI/1/xx (xx is the destination PSTN number to dial). For some reason asterisk did not dial the number but the control came to the context that I defined in the call file as if the peer had answed the call. It works if I change the channel from DAHDI to a SIP channel like SIP/4567 or I dial DAHDI/1/xx from a SIP channel. I am using asterisk1.4.23.1. Is it a bug in this release? Analog FXO ports are considered answered as soon as dialing is finished. The telco does not provide a signal to the calling device to indicate the far end answered the phone. This does not apply to PRI or FXS. Virtually all SIP service providers use PRIs. If the service provider used analog you would also experience this when dialing using SIP. -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
