Re: [asterisk-users] Questions on converting to ConfBridge
On 02/10/12 06:07 PM, Richard Kenner wrote: I'm looking at what would be involved in converting from MeetMe to ConfBridge and there seems to be a lot of missing administrative things, but I hope I'm just missing it. There also doesn't seem to be a way to lock conferences or mute or kick out users from the dialplan. What am I missing? You're missing the custom DTMF based menus in confbridge.conf, which allows you to set menus separately for admins and users of the conference bridge. This menu allows you to control kicking, muting, etc of users within the conference bridge. No need to manipulate from the dialplan anymore. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] Case-sensitivity of Dialplan variables.
On 02/10/12 09:02 PM, Vladimir Mikhelson wrote: snip First you need to consider compatibility with currently supported packages which include auto-generated dial plans like AsteriskNow, PIAF, etc. If you plan to break their functionality you need to at least coordinate your move with the maintainers. Then you may want to consider backwards compatibility with packages still widely used but not actively supported any more like Trixbox. Maybe not the best example as their WEB site says, This is the current stable release based on Asterisk 1.6. snip I'm not sure that's really the case. This change would be trunk only, and thus the first time it would show up would be Asterisk 12. Because anyone migrating between major versions should already be looking at CHANGES and UPGRADE.txt, this is just another situation where that would be the case. Deployments already based on a released major version would not be affected. +1 to case-sensitivity. It's the right way!(tm) -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] Case-sensitivity of Dialplan variables.
On 03/10/12 11:49 AM, Raj Mathur (राज माथुर) wrote: In short, my vote goes for case-sensitivity with a grace period for switching over. I disagree. Migrating between major versions should never be something like installing Asterisk 12 over an existing Asterisk 11 (or earlier) system. It should always be a migration between physical (or virtual) boxes. The time to verify your dialplan works in a major release is during the testing phase, not during the omg I installed over my production system! phase. If someone needs to upgrade to a major version, changes as documented in the UPGRADE.txt and CHANGES file would need to be performed anyways, so testing in a staging environment should catch the issues prior to production deployment. Besides, if it was an option, people would just ignore making the changes until the version where the option was no longer available, and we're basically in the same boat as just changing it in the next major version. Consistency for the win!(tm) -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] Case-sensitivity of Dialplan variables.
On 03/10/12 03:01 PM, Michael L. Young wrote: We are probably a year away from seeing a release for the version of Asterisk where this change would occur. We are two years away from an LTS version of Asterisk. So, I think there would be plenty of time for evaluation and testing to be performed by those affected. Especially, as in the case of what Raj mentioned at the beginning of his prior email, not too many people may even be affected by this change just like he won't be. Well said. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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 module app_konference
On 03/10/12 03:50 PM, pankaj pandey wrote: I am looking for a complete conferencing solution over asterisk (meetme is not fulfill my needs) . I googled a lot and see a lot of stuff on appkonference. Is anybody using this module? Please suggest me and give me some feedback on it. Perhaps you could give a better idea as to what your needs are, and why MeetMe() doesn't fulfill them? Perhaps ConfBridge() in Asterisk 10 or later would fulfill those needs? -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?
On 28/09/12 06:50 AM, Markus wrote: Hi list! ConfBridge dtmf_passthrough=no doesn't seem to have any effect. DTMF gets transmitted throughout the conference. I've tried Asterisk 10.7.1 from the official RPMs and 10.8.0 compiled from source. I've confirmed that it's disabled via the CLI confbridge show profile user profilename. It's an all-SIP scenario with RFC2833 as the DTMF protocol. Is this a known bug? Searching the issue tracker (hint, hint) does not return any dtmf_passthrough issues other than this one[0], which doesn't look to be related. Is another channel connected to the conference receiving the DTMF? Is that what you're intending? Because from my understand that is the intention, and not simply to limit the DTMF from being in the conference in the first place. At least that is almost how it reads in your message. [0] https://issues.asterisk.org/jira/browse/ASTERISK-20150 -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] 100 outgoing calls - 10 at a time - /var/spool/asterisk/outgoing/
On 27/09/12 09:01 PM, Patrick Archibald wrote: Is there a way to move 100 .call files in to /var/spool/asterisk/outgoing/ at once and have Asterisk call at maximum 10 at a time? snip I can certainly write a program to limit the number of simultaneous outgoing calls but before I do that I thought I would ask if there is another solution. Generally the preferred method when you're doing this programatically anyways is to use an external script through the Asterisk Manager Interface to generate your calls. Luckily, Russell Bryant has recently create an amioriginate.py[0] script which he's using as an example in the upcoming Asterisk: The Definitive Guide 4e book. [0] https://github.com/russellb/amiutils -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] RealTime table fields ordering
On 28/09/12 04:33 AM, Vieri wrote: So it seems that the contrib directory and the asterisk.org wiki are inconsistent and incomplete. Of course I understand that these are 'contributed' files but they should be proof-read by the Digium devs before packing them up into the official source tarball. snip That of course also implies contributions to review the files prior to release (which have release candidates). That directory contains data that was at one point contributed, and should really be reviewed by the community with any changes required submitted back upstream. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] PLAYIN MUSIC WHILE SEARCHING MYSQL
On 27/09/12 02:13 PM, Mehdi Rahimi wrote: On Wed, Sep 26, 2012 at 11:31 PM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 26/09/12 05:35 AM, Mehdi Rahimi wrote: I want to play music in my AGI while i am searching for a field in DB. Actually during some processes in AGI i need to play music . Probably Local channels to the rescue here. Dear Leif Madsen, Please explain more top posting fix resolved :) It's been quite some time since I did this, so I can't give you a specific example (that's left as an exercise to the reader), and I may be misremembering, but essentially I had 2 Local channels that I called via the Dial() application. One path played MusicOnHold() and the other would perform some fancy stuff (I think it was an API call via CURL() that would attempt to return a valid agent; a sort of dialplan based queuing system that used an external API interface that managed the availability of the agents). Anyways, the one Local channel would play MusicOnHold(), then when the API returns data to CURL(), the dialplan would continue and pull the caller out of the MusicOnHold() application, and then send them to the dialplan section to call the agent. The same principles could be applied here. I think it was a combination of MusicOnHold(), Local channels, and the Bridge() application. Sometimes you just have to be really clever with Asterisk to make it do what you want :) -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] MySQL InnoDB or MyISAM for CDR
On 27/09/12 11:45 AM, Matt Hamilton wrote: Date: Thu, 27 Sep 2012 10:23:35 +0200 From: lenz.lo...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] MySQL InnoDB or MyISAM for CDR I'd go for MyISAM and would set up a remote replica if data integrity is important. If you have like 1000 calls of (say) 30 seconds avg length, and you create 10 events per call, you would expect an event every three seconds. This is about 300 inserts per second. Say 600 at peaks. This should be feasible with server-grade hardware without much difficulty. Also as you always INSERT it behaves as a log file (no seeking, no locking) if the table is optimized. l. We decided to go with MyISAM since it supports concurrent inserts (as you suggested). Data integrity (a slight loss of call records) is something we can live by. Right now we use DRBD for replication, but I guess with MyISAM it doesn't make much sense if the db crashes. We are looking into other options as well. This may or may not be relevant, but you can also check out MySQL/Galera[0] for clustering solutions. Not sure if that gets you closer or further from your goal though :) It uses a modified InnoDB to allow a multi-master MySQL cluster. I used a chef cookbook to deploy it[1]. [0] http://www.codership.com/content/using-galera-cluster [1] http://support.severalnines.com/entries/21453521-opscode-s-chef-mysql-galera-and-clustercontrol -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?
On 28/09/12 07:36 AM, Markus wrote: Am 28.09.2012 13:24, schrieb Leif Madsen: Is another channel connected to the conference receiving the DTMF? Is that what you're intending? Because from my understand that is the intention, and not simply to limit the DTMF from being in the conference in the first place. At least that is almost how it reads in your message. All I'm trying to achieve is that the rest of the conference doesn't hear the beep's when a user presses a key. Users press keys to adjust the volume of the conference, for example. And these key presses get transmitted to all the other users in the conference, which can become quite annoying when there is a larger amount of users. Are you refering to my previous mails about adjusting volume of background music/speech in the conference? This is unrelated - in my test scenario I just set up a simple ConfBridge with no features at all, then dialed in via PSTN (arrives as SIP) from two different phones, and on each phone I can hear the key presses of the other party. OH! I just tested with a SIP softphone (X-Lite), and DTMF does not get passed to the other users! In X-Lite I can hear the DTMF keypresses of the users connected via PSTN (incoming via SIP), but when I hit a key in X-Lite I can't hear that on the PSTN phones. Hmmm ... I am not referring to your previous posts, but your test and results seem to indicate what I had somewhat thought. When you're using X-Lite, you're likely using RFC2833 for the DTMF method, which is out of band, and gets absorbed by Asterisk by it not playing the DTMF into the conference. This is how it should work (and likely does for most scenarios/phones). It sounds like maybe either a configuration or implementation issue on the carrier side though. Are you using inband DTMF there? Asterisk should really be absorbing that too, but sometimes it can't get it all. If you switch to an out of band method like info or rfc2833, does that help? Do you hear the DTMF on a normal call outside of ConfBridge() with the same carrier? I suspect this isn't a ConfBridge() problem, but a general DTMF one. Nice idea on the dtmf_passthrough setting, but it's not really the solution to your problem here. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] 100 outgoing calls - 10 at a time - /var/spool/asterisk/outgoing/
On 28/09/12 08:11 AM, Aldo Bergamini wrote: I am happy to hear that a new release of The Book is in the works! That's good! I'd hate to be working on something no one wanted :) I will have a look at Russell's script as soon as I am back at my work chair: there is however something I am very curious about: it is how you ask Asterisk, over AMI, to launch an external command, script, etc. I was (falsely) assuming that you need a channel to launch a script upon.. To be able to trigger 'commands' over AMI, before any channel exists, opens immense possibilities!! Oh heck ya. You can start up an Asterisk instance and just start doing things with it via your programs. That's the immense power of AMI; it's essentially the Asterisk API. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] User expected behavior of musiconhold and AGI's stream file
On 28/09/12 08:23 AM, Joshua Colp wrote: I am your friendly neighborhood developer here with a question that may impact some of you. You're friendly? :) Right now there is a small discussion occurring on the Asterisk development mailing list about the expected behavior of music on hold and AGI's stream file. Presently if you start music on hold and then call stream file the music on hold will be *stopped* but not *restarted*. Do you think this behavior is correct? I guess part of the question is; can you trigger it to be re-enabled after the stream file? The proposition on the mailing list is to add yet another knob to allow you to control whether it is restarted or not upon completion of the stream file and to change the default behaviour for Asterisk 12 to have it restart music on hold. I look forward to your responses so you can help with the ultimate decision for this discussion. My question about being able to re-enable it poses an interesting one. Since this is a programmatic method of controlling things, do you really want to automatically do something that wasn't explicitly defined? As someone who might interface via a program, I'm thinking I would prefer things to continue operating as they do now. If my program already accounted for this, then I've already triggered MOH to restart after the file. Another question might be; is there a way to determine if MOH was playing prior to my call to stream file so I can reset the previous state? My gut tells me that if this has been like this for a long time, and is how it worked originally, that how it works now be left as the default, and if you want to add an option that allows you to turn it on, that be the best approach here. Changing this can only make it a backwards compatibility issue. Someone who has run into this and needs it to act differently will seek out the new option after reading about it in the CHANGES file. In an ideal scenario, a system upgrade should require the least amount of knob turning. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] User expected behavior of musiconhold and AGI's stream file
On 28/09/12 08:45 AM, Joshua Colp wrote: Leif Madsen wrote: I guess part of the question is; can you trigger it to be re-enabled after the stream file? Sure you can! You can use set music to start it going again as the next command. And that makes sense. I kind of knew the answer already, but used it as a leader to the rest of the discussion :) My question about being able to re-enable it poses an interesting one. Since this is a programmatic method of controlling things, do you really want to automatically do something that wasn't explicitly defined? Personally I'm in the camp of no. Stopping music on hold right now is done to ensure that stream file can do what you ask it to do. Which makes sense. No one wants to play a file over MOH :) As someone who might interface via a program, I'm thinking I would prefer things to continue operating as they do now. If my program already accounted for this, then I've already triggered MOH to restart after the file. Another question might be; is there a way to determine if MOH was playing prior to my call to stream file so I can reset the previous state? There is currently no way to get MOH state but as Asterisk will not arbitrarily start MOH on channels in this situation you can certainly store this information yourself as you would be the one initiating it. OK, so we're on the same page here then. If you were the one initiating it, and you call stream file, then you know it's going to stop the MOH, and you can check your own programmatic state to determine if you should start MOH again. Starting it automatically again might not be the method you want. If you don't want it, now you have to explicitly stop it, which could cause a blip of music to be played after every file. This is certainly a bug which would have to be worked around, and seems like a lot more work than it is worth. My gut tells me that if this has been like this for a long time, and is how it worked originally, that how it works now be left as the default, and if you want to add an option that allows you to turn it on, that be the best approach here. Changing this can only make it a backwards compatibility issue. Someone who has run into this and needs it to act differently will seek out the new option after reading about it in the CHANGES file. Agreed but what I'm having a hard time grasping is the benefit of having this be a configuration option you enable. You *have* to be aware of it to enable it which is the same as being aware of it when writing your AGI. That makes sense to me. I was thinking the same thing, but wasn't sure if Asterisk would have started MOH due to some hold situation or something I hadn't thought of. If the initiation of the MOH was done by the program, then it makes perfect sense to me that it should start it again as long as it's documented that stream file will stop it (if already playing). I think I saw a commit from you today that satisfies that part of it. Based on this discussion, my stance seems to be adding the option just seems silly. A sane method of restarting the MOH already exists, and control should be in the AGI, not in Asterisk. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] PLAYIN MUSIC WHILE SEARCHING MYSQL
On 26/09/12 05:35 AM, Mehdi Rahimi wrote: I want to play music in my AGI while i am searching for a field in DB. Actually during some processes in AGI i need to play music . Probably Local channels to the rescue here. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] confbridge video support
On 25/09/12 10:18 AM, Bryant Zimmerman wrote: Where does video support for confbridge stand? I need to be able to take in multiple video callers and have the active speaking caller displayed to all participants. Are we there yet in Asterisk 10 or 11? Yes, both Asterisk 10 and 11 support this in ConfBridge(). Look at video_mode=follow_talker -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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 community services - Old Mantis instance to be shutdown on Aug 28th, 2012
On 27/08/12 10:08 AM, Asterisk Development Team wrote: As a part of other infrastructure changes we are making to the community services, we will finally shut down Mantis for good. Huzzuh! Does this mean http://issues.asterisk.org will now go directly to JIRA? Leif. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] SLA - Shared Line Appearance - Polycom
On 10/05/12 10:45 AM, eherr wrote: Has anyone been able to get SLA ( Shared Line Appearance ) to work with Polycom phones. I want to be able to emulate a key system but I cannot figure it out. Everything I tried so far is just not working together. Yes, I have managed to get everything working recently with Asterisk 1.8 and Polycom devices (and Aastra, Panasonic, Snom, Digium, etc...) For the Polycom's in particular, look for the attendant/ section in the Polycom admin manual. You will want to setup your resourceList options for the SLA configuration so it monitors correctly (which are really just BLF type keys on the Polycom that monitor a hint on Asterisk). The hints on Asterisk are them controlled via the SLAtrunk() and SLAstation() applications in Asterisk. I'll be updating the SLA section in the next version of Asterisk: The Definitive Guide (4th edition) to elaborate on SLA based on my testing and implementations over the last couple of months. So here is what you need as a recap: * registration line on Polycom * attendant configuration for BLF keys * hints in Asterisk that the attendant keys can monitor * sla.conf configuration for trunks and stations * usage of SLAtrunk() and SLAstation() applications to use the SLA lines, which also changes the device status (that is monitored by the device) -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] Playing Multiple File ( simultaneously ) on Channel
On 27/04/12 11:57 AM, shayne.al...@gmail.com wrote: Yep, But I think that is can be done, independent to Queue app. If we have an application which call MusicOnHold inside! maybe we can control simultaneously playing back of files... if we have such ability then it can be replaced by what exist inside the Queue. or maybe an application which written from scratch... can be help full. I guess you could get crazy and start injecting audio onto the channel through the use of ChanSpy() in whisper mode. It could probably be triggered through AMI to execute a Local channel which then connects to the channel you want to play the audio back onto, and then whisper that audio over top of whatever playback is happening. Check the Audio Manipulation section of the Asterisk Cookbook for some simple examples here: http://ofps.oreilly.com/titles/9781449303822/c03-AudioManipulation_id302347.html I also talked a bit about injecting audio onto a channel at AstriCon 2011 in my Cooking With Asterisk talk. It's the last recipe I talk about in this video: http://www.astricon.net/videos/Cooking-with-Asterisk.html -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] tcp version of toronto - osaka doesn't work
On 02/01/12 04:39 PM, sean darcy wrote: OK, the book is out of date. Do Not put the name of the local device/user in the register statement. Does it work with UDP? If so, then that is a different behaviour. The book only tested with UDP, not TCP, so if it works with UDP, then it was working as expected. Also, please be sure to file errata so that we can look at it for the next printing or version of the book (depending on what the issue actually is). Errata can be filed at http://oreilly.com/catalog/9780596517342/errata/ -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] Best non polycom SIP conference room phone
On 05/01/12 05:24 PM, Kevin P. Fleming wrote: snip Although in my personal opinion, it's really hard to beat the IP5000. That has been my experience as well. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] PITCH_SHIFT()
On 20/12/11 01:15 AM, John Jolly wrote: In Leif Madsen's AstriCon 2010 talk titled 5 Things You Didn't Know Asterisk Could Do https://docs.google.com/viewer?a=vq=cache:hCDfIk4pvngJ:leifmadsen.com/sites/default/files/AstriCon%25202010%2520-%25205%2520Things%2520You%2520Didn't%2520Know%2520Asterisk%2520Could%2520Do.pdf+asterisk+dialplan+func+PITCH+SHIFThl=engl=uspid=blsrcid=ADGEEShzSRqJl26lEybK-TvxHL4hKQrd-mBpAapRV6eyI8ST0E5AosCEqp2bm_h5eORZFwwEZDqzEKpT9Fg244nkCgX4BDEGL6bik4Non5_fgm62fzrBxyIXjm1hnqJx2-yGyVlbdXKdsig=AHIEtbQ2NyYajUzeJshmWKAgZEi0RprNjQpli=1 he mentions that the PITCH_SHIFT() function is designed to be used dynamically and can change the pitch of a channel on the fly using features.conf. Can someone provide me with any information of how this would be accomplished for dynamic use? I'm familiar with the dialplan syntax use examples such as: exten = 1,1,Set(PITCH_SHIFT(tx)=highest); raises pitch an octave exten = 1,1,Set(PITCH_SHIFT(rx)=higher) ; raises pitch more exten = 1,1,Set(PITCH_SHIFT(both)=high) ; raises pitch and so forth, but don't understand how these functions would be called dynamically from features.conf. You'd just create the application_map as documented in features.conf and then apply the PITCH_SHIFT() function to whichever channel you want. Untested, but should look something like: pitch_up_them = 3*,peer/both,Set(PITCH_SHIFT(tx)=high) -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] Preparing to store vm in database
On 09/12/11 01:10 AM, Mike Diehl wrote:
Hi all,
I'm getting ready to start storing all of my voicemail in a mysql database.
I've already got RT sip and RT voicemailboxes working.
I understand that vm storage only works via odbc. I've read all of the
documentation I could find, including:
http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage
===
But there was this comment, at the bottom of the page:
I tried adding ODBC message storage to a 1.2.5 system already using MySQL for
RealTime... Not a good idea, but using ODBC for both Realtime and msg storage
seems good so far.
Is this still true with Asterisk 1.6.2.9? Or do I need to migrate all of my
RT configuration to ODBC?
Interfacing with a database through ODBC in Asterisk is inherently more
stable. The res_odbc interface has received significant amounts of
testing and development over the years, and is definitely the most
stable. One tip I'd offer is to use the latest ODBC drivers from
unixodbc.org and not via the packages on your system, which are usually
quite old (the ones shipped with CentOS 5 are from 2006 if I remember
correctly).
The MySQL and PostgreSQL drivers receive quite a bit less attention.
I've had good luck with res_odbc over the years as I've been deploying
it since Asterisk 1.4. The res_odbc drivers have received lots of
attention from its author, Tilghman Lesher.
===
I also read this comment:
Make sure you load the .WAV file and not the .wav or .gsm or it won't work!
-- is that true? I couldnt get .WAV to work. But when I used a 16bit 8000Hz
file with a .wav extension, it worked fine.
Will I be ok if I just load the .wav file?
If it works for you in development and testing, then it should be fine.
I haven't deployed voicemail audio into the database for a few years,
but I think I used 'wav' as well.
===
Also, I see that I'll have to pre-load all of the voicemail messages and
greeting files into the database, like so:
INSERT INTO voicemail (msgnum,dir,mailboxuser,mailboxcontext,recording) VALUES
(-1,'/var/spool/asterisk/voicemail/CONTEXT/USER/busy','CONTEXT','USER',LOAD_FILE('/var/spool/asterisk/voicemail/CONTEXT/USER/busy.WAV'));
I assume this has to be done for every CONTEXT and every USER, and those
values need to be substituted into both the fields, and the directory path
value.
That makes sense. If you're going to start using voicemail in the
database, then you'll need some data there ;)
===
Finally, is there an AGI command that will play a .wav file from the database?
There is not. Some work was done a few years ago to allow playing audio
from the database like you're thinking, but it was never complete, and
has never become a priority issue for any community developer to
complete. You could of course create an AGI() script that pulled the
audio out of the database, caching the audio for a period of time, then
just played it like you would any other file, and cleaning up the file
after.
--
Leif Madsen
http://www.oreilly.com/catalog/asterisk
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Re: [asterisk-users] Mixing asterisk.conf, asterisk.ael and asterisk realtime
On 09/12/11 08:27 AM, Ishfaq Malik wrote: Hi I'm already mixing asterisk.conf and asterisk realtime architecture (Macros go in .conf) but is it possible to have a Macro in .ael, another in .conf and have them both be callable from the realtime database? asterisk.ael just gets compiled into Asterisk dialplan (like extensions.conf) anyways, so you can. You just have to follow the same rules about there not being duplicate macro names, as the AEL and dialplan logic is going to be combined together in memory. Leif. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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 10.0.0-rc1 Now Available
On 11-11-10 12:12 PM, Danny Nicholas wrote: Yeah! My boss will be much happier having a system that doesn't have the -tail on it. I hear this kind of statement every once in a while, which makes absolutely no sense to me. If you're blindly running a version of any software in production (regardless as to it being tagged a -beta, -rc, -magic_candy, etc) without prior testing, then you're pretty much at the same risk regardless. I could take a random snapshot from a branch and name it something without a tailing hyphen+name, and it'd be pretty much the exact same thing without prior testing in your environment. 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] Frequent Asterisk Restarts
On 11-11-10 01:15 PM, Eric Wieling wrote: The Asterisk source tree has a document with instructions on getting a backtrace from the segfaults so you can report it on the issue tracker. Most up to date documentation is on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Debugging -- _ -- 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] Log for voicemail to email?
On 20/09/11 06:53 PM, Kevin Oravits wrote: I am having a problem with one of my sites where they are not receiving the voicemail to email. I’ve done a lot of troubleshooting and can’t find the issue. It would be helpful if there was a log I could look at so that I could see perhaps where the email is being rejected. Does anyone know of a log that runs on Asterisk that would have this history? Well Asterisk isn't sending the email, the email service on the server is doing that. You'll need to enable the logging for email delivery on the email service itself. (It's possible you haven't installed an MTA or have it disabled. Or perhaps the other ends are rejecting due a missing MX record, or some other email configuration issue.) -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] Fixing an old bug related to extension s - feedback wanted
On 20/09/11 09:34 AM, Danny Nicholas wrote: Just my .02 - fix Originate since the Original Asterisk book, page 125 paragraph 1 says s = start. If s is not really start, I'm going to scrap my 3+ years of dialplan writing and change all of my simple dialplans to read exten= start,1,blah instead of exten = s,1,blah. To me exten= s,1,blah is more intuitive and less vulnerable than exten = _X.,1,blah. The 's' extension does stand for 'start' but I don't think we've ever implied it was a catch-all extension. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] Fixing an old bug related to extension s - feedback wanted
On 20/09/11 03:37 PM, Ira wrote: At 07:09 AM 9/20/2011, you wrote: Using start makes your dialplans much easier to read :-) and makes them more secure as no app will end up there by accident, which may happen in your current systems. When I went and read version 3 it seemed to indicate that start has no actual meaning and I could just as well call it cow or fish. Am I reading it correctly or does the word start actually have a special meaning? No, that extension 'start' (literal) has no special meaning. You absolutely could call it cow, fish, pig, or farmer_john. How you get there is by implicitly calling it. exten = s... on the other hand has always had special meaning as Olle has pointed out, and typically has meant start (for analog lines). Outside of that you shouldn't really be using the 's' extension as your default extensions. The 's' extension has never been a catch-all extension. Olle has found a situation where the 's' extension is being used as a fallback, which is not right, and is suggesting we make Asterisk consistent in it's usage of 's'. I agree with his proposal. But because this functionality (bug) has been around for quite some time, he is asking the community for feedback on who may have inadvertently used the functionality in their dialplans. Apologies for anyone who may have read some documentation that appeared to imply that the 's' extension was a catch-all. In the first and second editions of Asterisk: The Future of Telephony we were mostly using analog lines, and thus the usage of the extension 's' was fairly prominent. There are many other single letter extensions that have extra meaning, such as 'i', 't', etc..., but we never intended to imply that 's' was a catch-all extension. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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 Vs 1.6 Vs 1.8
On 12/09/11 09:48 PM, Joseph wrote: Was NAT problem fixed in 1.8.7 ? I'm using 1.4 but I've tried 1.6 and 1.8.5 and in both cases connection is not working with my provider with SIP + NAT. The connection is showing up as registered but the call is not coming IN (congestion). Can you define NAT problem? I'm unaware of any issues with Asterisk (or end points) behind NAT. It is mostly likely a configuration issue rather than a bug. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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 server: Console or GUI OS ? Init level 3 or init level 5 ?
On 12/09/11 02:21 PM, linux guy wrote: I'm about to start building my asterisk server and I can't seem to find anything that discusses the pros and cons of installing the OS (Fedora 15) as console only or GUI, ie install KDE as well. http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html#Installing_id291070 -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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 on Android?
On 08/09/11 02:19 PM, Cobra 2 wrote: I've chrooted debian onto a Motorola Droid running Cyanogenmod 7 and I've gotten asterisk to run on that just fine. I think the question is, can you answer your incoming calls with the Asterisk running on the device? -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] Beggining asterisk
On 04/09/11 02:51 PM, Tamer Higazi wrote: the 3rd edition is available, but that book covers every thing to run the asterisk PBX. You can read the 3rd edition online at http://ofps.oreilly.com/titles/9780596517342/ HTH! Leif. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] Set(CHANNEL(musicclass)=
On 02/09/11 11:27 PM, Joseph wrote: In asterisk 1.4 I had: exten = s,n,Answer() exten = s,n,SetMusicOnHold(default) But in 1.6 1.8 I think don't need to use: SetMusicOnHold(default) (beside it is deprecated) as it is default. In 1.6 and UP I think it is: Set(CHANNEL(musicclass)= Can somebody explain what do they mean by CHANNEL? CHANNEL() is a dialplan function. You're setting parameters for the current channel by using that function. So instead of using a dialplan application like you were before, you use the CHANNEL() function. exten = s,1,NoOp() same = n,Set(CHANNEL(musicclass)=default) I could use just: exten = s,n,MusicOnHold() There is a lot of documentation on www.voip-info.org but sometimes it is not clear which asterisk version it applies to :-/ (Another good reason to be reading the documentation on https://wiki.asterisk.org/wiki instead :)) -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] trying to build 1.8.6.0 on CentOS 6, problems with ptlib
On 06/09/11 05:14 PM, Kevin P. Fleming wrote: I was trying ./configure --disable-chan_ooh323 and that was not making a difference. It won't, for two reasons: Asterisk modules can't be selected/deselected via the configure script (menuselect is used for that), and chan_ooh323 doesn't use pwlib/openh323, chan_h323 does. However you could select/deselect modules using menuselect if you wanted to automate the process. It's documented over here: http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html#Installing_id293439 (Just thought I'd pass that along as I thought it was pretty neat when I learned about it :)) -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] dialplan pattern help
On 11-07-23 10:30 AM, Armand Fumal wrote:
Hi all,
I need help for make a pattern for a special case that i can't find the
solution.
In my case I want to match these in one pattern:
This is the same ext that can come in 4 cases
exten = _42704701,1,Macro(dialfax,${EXTEN:-8}) ; case with
42704701
exten = _X42704701,1,Macro(dialfax,${EXTEN:-8}) ; case with
042704701
exten = _42704701,1,Macro(dialfax,${EXTEN:-8}) ; case with +3242704701
exten = _XXX42704701,1,Macro(dialfax,${EXTEN:-8}); case with
3242704701
I have try _.42704701 but the parser stop to check after the point .:-(
So did you have any suggestion ?
Ya you can't match anything after the '.' in pattern matching.
I'm not sure the pattern matcher is really capable of doing what you want here.
The only way to do it really is to match less restrictively and perform a check
using dialplan applications/functions, and then if nothing is found, to fall
through.
Perhaps something like:
exten = _XXX,1,NoOp()
same = n,ExecIf($[${EXTEN:-8} = 42704701]?Macro(dialfax,${EXTEN:-8}))
same = n,Verbose(2,Did not match -- falling through)
same = n,Playback(invalid)
same = n,Hangup()
I'm pretty sure that's the only way you can do it in a single line (the ExecIf()
application).
Leif Madsen.
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Re: [asterisk-users] Next Asterisk 1.8 Release
On 17/06/11 03:24 PM, --[ UxBoD ]-- wrote: Hi, When is the next release planned for as very keen to get it into Production but require the call pickup fix. All changes are always available in the 1.8 branch which goes through automated tested continuously. You're welcome to start testing changes in your development systems at any period of time. 1.8.5 release candidates should be available later this week though. Leif. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] MixMonitor
On 16/06/11 07:36 AM, salaheddine elharit wrote: hello list, i have asterisk 1.4 with IAX and sip i have configured the MixMonitor in order to record the conversation but when i receive an inbound call from customer in IAX(1000) and i want to transfer the call to other phone SIP(223) the conversation between customer and IAX is recorded but the conversation between customer and sip is not recorded Is the call coming from IAX(1000) or going to IAX(1000)? Note that when you transfer calls around and are using MixMonitor() (or any recording) that you have to think of the recording as being associated with the incoming channel, and the recording should essentially follow it around. So if you have a call coming in like this: ITSP -- Asterisk -- Dialplan -- Mixmonitor -- Dial(SIP/1000) Then the MixMonitor() is associated with the channel created when the call came in from the ITSP. If that channel is then transferred, the recording should follow it around. Can you elaborate a bit more on the call flow and show the console output? -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] MixMonitor
On 16/06/11 09:20 AM, salaheddine elharit wrote:
thanks for your response
the call is going to IAX(1000), i have i DID Number when the customer
call this number 0520XX the call is goint to agent
IAX. in my dialplan i have
exten = 223,1,MixMonitor(blah.wav)
exten = 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,Dial(SIP/223)
and in extensions.conf i have
exten = 223,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten = 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 223,n,Dial(SIP/${EXTEN},,KkTt)
exten = 223,n,Hangup();
OK, well nothing looks obviously wrong there from what I can tell.
What is your console output doing though when you do the transfer? Are
you using Asterisk transfers? What version of Asterisk are you using?
Leif.
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Re: [asterisk-users] Asterisk 1.8.4.2 Now Available (Security Release)
On 02/06/11 03:35 PM, satish patel wrote: Is this available in current SVN ? Changes are always checked into SVN first and then made available in a tag. Leif. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] [SOLVED] Asterisk 1..8 multiple queue
On 27/05/11 03:18 PM, satish patel wrote:
In this book example there is a printing issue at Unpaused section. it
should be like following
same = n,GotoIf($[${UPQMSTATUS} =
UNPAUSED]?agent_unpaused,1:agent_not_found,1)
Please file stuff like this as errata at
http://oreilly.com/catalog/9780596517342 (left hand side). That way we
can get it fixed up in subversion.
Thanks!
Leif.
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Re: [asterisk-users] Asterisk 1..8 multiple queue
On 26/05/11 04:20 PM, satish patel wrote: Actually right now i have very big AddQueueMember dialplan for every individual queue for login/logout/pause/unpause etc.. ( we have 3 queue) Let me explain my example We have 3 queues ( sales, support, tech) Sales - A,B,C,D,E agents Support - A,B,C,D,E agents tech - A,Z agents Before it was quite simple just specify member in queue but with AddQueueMember its now that case. Before it was just single queue login allowed you to enter in all queue. but in AddQueueMember they have very complex agent login thing. Could you give me example or tell me how i use AddQueueMember in my current setup which i explain you. (multiple queue login and restrict agent for other queue) The solution to your problem is to write some dialplan. I even helped you along by writing some documentation :) http://ofps.oreilly.com/titles/9780596517342/asterisk-ACD.html#ACD_id288626 -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- 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] MagicJack quality
On 11-05-24 01:21 PM, Steve Edwards wrote: If it take the OP (of this thread) 3 years to reply, what does that say about their product support? Par for the course? :) 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] AstManProxy
On 11-05-20 09:37 AM, Ishfaq Malik wrote: Do many people use this? Is it reliable and safe? It may still work, but that code is quite old, and I'm not even sure it's necessary any more. 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] SIP Diversion RDNIS - how to get reason parameter?
On 11-05-20 10:39 AM, Benoit Panizzon wrote: After reading a bit more in chan_sip.c (I'm not a coder) I fear the result is just put in a temporary variable __SIPDIVERSIONREASON but not in a variable useable in the dialplan. You could double check by using DumpChan() to see what channel variables are available for you throughout the dialplan flow. Also check the CHANNEL() and SIP*() functions to see if there is anything there that may be of use. 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] 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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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] 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
Re: [asterisk-users] concurrent call tracking
On 11-05-11 06:36 PM, Skyler wrote: Thanks Dovid, if you don't mind sharing the code and the dial plan side I'd like to take a look at it for sure. The dial plan example Leif replied with is pretty much what I was thinking, just didn't have a clue how to go about it. ;) You could also look into using LOCK() and UNLOCK() dialplan applications to make sure each insert happens sequentially. 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] Asterisk 1.8.4 Now Available
On 11-05-11 09:31 PM, Jose P. Espinal wrote: Download links on the website have not been updated (asterisk.org) Oops sorry! I will fix that right.. now! 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] Asterisk 1.8.4 Now Available
On 11-05-11 10:46 AM, Paul Belanger wrote: On 11-05-11 10:29 AM, Jeremy Kister wrote: I'm a bit confused about this release (and previous releases on the 1.8 track) so please bare with me. I viewed the ChangeLog, but I don't see any of the 'sample issues' listed. why is that ? I would expect to see the 'sample issues' listed after 1.8.4-rc3. Also, is there a reason/procedural error that patches such as: https://issues.asterisk.org/view.php?id=18382 https://issues.asterisk.org/view.php?id=18742 didnt make it into this 1.8.4 release ? Correct, they will appear in 1.8.5-rc1 forward. When -rc1 is created, it will be tagged from the HEAD of branches/1.8.. If 1.8.5-rc2 is create, it is because of an issue / bug was found in 1.8.5-rc1, and will include that fix only. If a new issue is reported after 1.8.5-rc1 and fixed in branches/1.8, it will not be added into 1.8.5 release, but will wait until 1.8.6-rc1. More information about this is documented at http://blogs.asterisk.org/2010/09/02/the-monthly-asterisk-release-cycle/ Basically, it doesn't make sense to do release candidate RC1+x from the head of branches/1.8, because then there'd never be a solid base to test from. When we do an RC1, it is pulled directly from the branch, which gets all changes since the previous RC1. Once and RC1 is created, if something is deemed to trigger a new RC (a regression is a good example), then the RC1 is copied to RC2, and the specific changes are merged into that RC. No further changes other than what was merged in are added (i.e. not all changes in the branch are part of RC2). If additional changes need to be made, then RC2 is copied to RC3, and specific fixes are merged to that RC. This continues until the full release is made, which is an exact copy of the latest RC. So if we had an RC3, then RC3 is copied, without changes, to the release version. So in this case, tags/1.8.4-rc3 as copied to tags/1.8.4, and the only changes were made to the .version file and ChangeLog. Then the standard release process is followed to turn that tag into a .tar.gz and get it onto the downloads site. Any changes made after 1.8.4-rc1 (for example) would then become available in 1.8.5-rc1, because only RC1s contain all changes from the branch directly. HTH, Leif Madsen. -- _ -- 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] concurrent call tracking
On 11-05-11 12:57 PM, Skyler wrote:
I would like to track/store concurrent call usage per user by
day/week/month and get server totals by day/week/month. Google comes up with
mostly info regarding concurrent call limits, though my goal is to calculate
actual concurrent channel usage and add it into reporting. I'm using * 1.6.2
+ mysql - realtime (no gui). Any suggestions / open-source / AGI on where to
start looking into implementing something like this?
Just use SNMP to get the channel usage. If you don't want to use SNMP, then just
use something like GROUP(), GROUP_COUNT() and func_odbc to write channel usage
to the database. Something like
[Outgoing]
exten = _NXXNXX,1,NoOp()
same = n,GoSub(subTotalCallCounter,start,1(outgoing))
[subTotalCallCounter]
exten = start,1,NoOp()
same = n,Set(GROUP(totalcalls)=${ARG1})
same = n,Set(ODBC_TOTAL_CALLS(${ARG1})=${GROUP_COUNT(${ARG1}@totalcalls)})
same = n,Return()
[Incoming]
exten = 4165551212,1,NoOp()
same = n,GoSub(subTotalCallCounter,start,1(incoming))
[LocalSets]
exten = _1XX,1,NoOp()
same = n,GoSub(subTotalCallCounter,start,1(internal))
func_odbc
-
[TOTAL_CALLS]
dsn=myDatabase
writesql=INSERT INTO totalCalls ('type','callcount') VALUES
('${VAL1}','${ARG1}')
Something like that. Totally untested and only written in this email :)
Leif.
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Re: [asterisk-users] CLI - displaying all channel variables
On 11-05-11 12:29 PM, Steve Edwards wrote:
On Wed, 11 May 2011, Eric Wieling wrote:
Generally you should insert a Noop in the dialplan to examine variables.
Noop(EXTEN is ${EXTEN}) for example.
The 'verbose()' application would be an example of 'better practices.'
It's function is obvious rather than just a convenient side-effect.
It has additional functionality in that you can specify the 'verbosity' level
needed.
Agreed. I tend to use NoOp() for an actual No Operation, such as using it on the
first line of an extension:
exten = something_awesome,1,NoOp()
same = n,Verbose(2,Incoming call from ${CALLERID(all)})
same = n,Dial(SIP/someone_awesome)
same = n,Hangup
That way if you want to place things ahead of any line, you can do that without
impunity. Even using Verbose() on the first line can cause problems if you want
to move the Verbose() around and place something before it -- now you have to do
some copy/pasting, and extra work that could be avoided :)
Leif.
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Re: [asterisk-users] question on ways to activate voicemail light on polycom
On 11-05-06 02:56 PM, Watkins, Bradley wrote: Yes, use the MinivmMWI application. That's how I've done it in the past as well. 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] Using Asterisk + other FOSS projects to facilitate a call-in Internet Radio web-frontend
On 11-04-30 03:10 PM, Alec Taylor wrote: Good Evening, I'm setting up an Internet Radio website with call-in functionality, and need to know the kinds of FOSS tools I should install to get the job done. Here's an example of what I'm looking for: http://i56.tinypic.com/aafz4k.png Call protocol: [Producer calls in] [Host calls in] [Guest calls in]-[Screened by Producer, if accepted, conferenced into host] On the website they'll need to be able to call in (mic input grabbed), and listen in (without calling in). I've been suggested many things, including Skype, IceCAST and [currently the most promising] Asterisk+Red5+Red5Phone. Are there any better ways of doing this, and if not, how do I setup asterisk for the above task? Thanks for all suggestions, While it isn't a free option, using chan_skype (license purchased from Digium) may be the easiest solution here. A lot of people know what Skype is, and may already have it installed, and instead of requiring your users to install a softphone, they could just click the Call Me link you provide. Alternatively, you could get a DID and allow people to call in the old fashioned way. You'd have to pay for the interconnectivity, but if you use a SIP provider the costs should not be unreasonable (1-2 cents per minute). You could always look into using Google Talk as well :) 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] Discussion: Are we ready to leave 1.4 behind?
On 11-04-29 02:59 AM, Olle E. Johansson wrote: 29 apr 2011 kl. 01.49 skrev Leif Madsen: Well the issue is that we currently have over 900 open issues in the Asterisk project alone, and with only one primary bug marshal (myself) sometimes things accidentally get closed if it looks like a configuration issue. What's the reason that we only have one bug marshal? We used to ask people to become bug marshals to help, but the last I heard you and Russell did not want community marshals. What went wrong with that? Wasn't it any help? Let me clarify, as it was not at all my intention to imply I was the *only* bug marshal. Poor wording on my part. There are certainly lots of people that help manage the bug tracker, and I'm thankful for everyone who responds to issues, asking for the appropriate information from reporters, and reviewing logs pointing out potential issues which help developers. It's just I'm the main one handling work flow, making sure the tracker doesn't get to the point it was when I started working on it every day (the majority of issues were sitting in 'New' for many weeks). Sorry if it was implied that I'm the only one working on the bug tracker, because that is obviously not the case. I am grateful for any help people can provide, and they are welcome to ask me what they can do to help. I don't remember a discussion where I would persuade people from not helping :) I've tried to make the process for moving issues forward as transparent as possible. Just search Google with site:lists.digium.com leif madsen bug marshal for a few posts about work flow. Additional information is here: https://wiki.asterisk.org/wiki/display/AST/Policies+and+Procedures Information for reporters is here: https://wiki.asterisk.org/wiki/display/AST/Debugging 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
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
On 11-04-28 12:04 PM, Administrator TOOTAI wrote: Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes for few weeks/monthes till 1.8 reaches the level that the community accept to switch to 1.8 What is the guide here? What is the level that the community accepts? Unfortunately that is a statement that is impossible to measure quantitatively. The answer will always be, We're not ready! Having to focus on issues on both the 1.4 and 1.8 branches simultaneously distracts from the goal of making 1.8 stable (which in my several deployments recently, it seems to be). I've also seen very few issues being committed to 1.4 for quite some time, which seems to tell me 1.4 is stable for most deployments. It's not like there has been a flurry of activity around 1.4 and all of a sudden it's being cut off. In my estimation the number of commits to 1.4 going from a few to none is not a significant direction change. Asterisk 1.4 isn't going away. The code base won't stop working on your system -- it will continue happily plugging away as it always has. The code will continue to be available for deployments. With focus being directed to 1.8, the issues that may be blocking you from having a successful migration to, or deployment of, Asterisk 1.8 will get fixed that much sooner. If the community won't, or can't, step up to maintain a community based branch which has very few changes being made to it, then I'm not sure it is fair to expect Digium to do that. 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] Discussion: Are we ready to leave 1.4 behind?
On 11-04-28 04:33 PM, Administrator TOOTAI wrote: Le 28/04/2011 21:47, Leif Madsen a écrit : On 11-04-28 12:04 PM, Administrator TOOTAI wrote: Ok, so why not stay with asterisk 1.4 security *and* bug/regression fixes for few weeks/monthes till 1.8 reaches the level that the community accept to switch to 1.8 What is the guide here? What is the level that the community accepts? Unfortunately that is a statement that is impossible to measure quantitatively. The answer will always be, We're not ready! Don't think so, analyze the answers to this discussion -thanks Ole ;-)-: till 1.8 is not at the feature level and stability of 1.4, people like me will not move to 1.8 Measure is easy :-) But that's what I don't get. No one is *forcing* you to move to 1.8 *right now*. The code base for 1.4 isn't going anywhere. Anyone is able to keep deploying 1.4 (or 1.2, or 1.0, or 0.9 for that matter) to their hearts content. 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] Discussion: Are we ready to leave 1.4 behind?
On 11-04-28 07:02 PM, Ira wrote: At 03:48 PM 4/28/2011, you wrote: OK, maybe not, but if I thought it was a bug and you discover it was a bug and fix it, than who was it who decided it wasn't a bug 15 minutes after I put it in the bug tracker and why did that person have that much power? Look, I know things take time to fix and test, I have no problem with that and I know users report things that aren't bugs as bugs. I develop software and my users do all those annoying things too, but I can't slap them down like that if I expect them to continue being customers. And I know the people who do this are volunteers, but my software is free, so I'm a volunteer too. Well the issue is that we currently have over 900 open issues in the Asterisk project alone, and with only one primary bug marshal (myself) sometimes things accidentally get closed if it looks like a configuration issue. If anyone ever opens an issue they they feel is a bug and the issue is closed, then the best forum is the #asterisk-bugs IRC channel. This allows you to speak with the bug marshals and to work through some additional information that might be required to help determine that something is truly an issue. 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] Discussion: Are we ready to leave 1.4 behind?
On 11-04-28 07:09 PM, Alec Davis wrote: Making an assumption here, I'm sure I cleared the remaining resequencing issues up in 1.4 SVN and 1.6.2 SVN. https://issues.asterisk.org/view.php?id=19032 The issues I uncovered and fixed were when a new voicemail is left, while a mailbox is open for review and the user deletes a message. Can anyone who has this issue currently please test the 1.4 branch? Feedback would be extremely helpful in determining if anything further needs to be done here. If so, then please open a new issue and report here. 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] core show channels consise in asterisk 1.8.3
On 11-04-18 02:47 PM, Jerry Geis wrote: When I do core show channels concise over the AMI interface how do I specify that I want to see the actual channel number like DAHDI/4/xxx where 4 is the actual channel. RIght now I am seeing DAHDI/i1/x where i1 is the span. I could have sworn I saw this issue already reported, but I can't seem to find it. Can you test with the latest 1.8 branch to see if it has already been resolved? I tried for a few minutes to find the issue on Mantis as I'm almost positive that I've seen it filed, but I can't find it. 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] Asterisk 1.8.3
On 11-04-07 09:45 AM, --[ UxBoD ]-- wrote: And don't forget that call pickup crashes Asterisk from what would appear release 1.8.1 upwards! We have had to back level to that latest 1.6 branch. https://issues.asterisk.org/view.php?id=18654 I ran into this issue as well on 1.8.3.2, but I didn't try a newer version, and someone else reported on the issue they don't have that problem with 1.8.4-rc2. Could someone who has this issue on 1.8.3.2 or earlier re-test with the latest 1.8 branch to determine if this is still an issue? 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] SIP Provider Recommendation in US
On 11-03-03 11:22 AM, Brent A. Torrenga wrote: I am becoming frustrated with our current VOIP provider. Does anyone have any suggestions for a provider that supports asterisk well and provides solid service? Voip-info.org has a husge list of providers, but it is impossible to tell the fly-by-night operations from the reputable providers. I've had good luck with bandwidth.com for a couple of customers running call centers. 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] Using voice modem as poor man's FXO in Asterisk 1.8
On 11-02-27 09:12 PM, Stuart Longland wrote: I've tried researching this, and so far, have struggled to find any contemporary information on the issue, so I do apologise if asking this irritates people who have answered this before. I have managed to set up Asterisk 1.8 on the web server here. I have two softphones (Ekiga) able to communicate with it. So far so good. I'm now curious to see if I can link it with the PSTN phone line here. There are several very good answers in this thread, and I suggest reading them. However, if hardware costs are the issue, then my recommendation is always to look at a SIP connection from an ITSP as your connection to the PSTN. The costs are nearly trivial (at least in Canada here you can have a DID for inbound calls for something around $5 a month, with termination costs in the range of 1c/min -- in other commonwealth countries I presume the costs are similar?). My bill rarely rises above $20 a month, and I use my phone a lot. (Business, personal, and 3 DID numbers are included in that cost.) I highly suggest you spend your time and money elsewhere, rather than chasing the dragon that seems to be winmodem FXO connectivity. If you absolutely must have hardware, then I suggest you start with used ATA (analog telephony adapters) that can be found on eBay, kijiji, craigslist, or any other assorted websites. 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] Recieve_Fax caused crash 1.8.2.3
On 11-02-24 08:56 PM, Andrew Latham wrote: And I go back to triple check and compare revision numbers... You are 100% correct, the revision numbers in our local repository are wrong, someone pushed the 1.8.3 RC3 into our local 1.8.2 branch. I apologize and will work to better control my trust of other engineers as this is twice in one week I have looked like an ass. International bandwidth limits change how you work and as a business force the mirroring of as many sources as possible. No worries, you had my heart going there pretty good for a moment! 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] Recieve_Fax caused crash 1.8.2.3
On 11-02-24 04:08 PM, Andrew Latham wrote: There are many updates in 1.8.2.4 that may fix your issue. If you are running any version of 1.8 it should be a quick update. I wouldn't say many. There is one fix in 1.8.2.4 over 1.8.2.3. From the ChangeLog: * Asterisk 1.8.2.4 Released. * AST-2011-002: Multiple array overflow and crash vulnerabilities in UDPTL code The release announcement for AST-2011-002 is here: http://downloads.asterisk.org/pub/security/AST-2011-002.pdf 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] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1
On 11-02-23 10:31 AM, Jose P. Espinal wrote: - Added a new configuration option remotesecret for authentication to remote services. For backwards compatibility, secret still has the same function as before, but now you can configure both a remote secret and a local secret for mutual authentication. - I thought that 'remotesecret' is used to authenticate myself when placing a call to the remote network, as I used to do with 'secret' parameter. I may be mistaken, because I don't use remotesecret, but I think the purpose of that was to allow different authentication depending on the direction. My guess is remotesecret is used to authenticate the remote end when a call is placed into Asterisk, and secret is used when you're placing a call to the remote server. Or it's possible the feature has a bug and an issue should probably be opened on the issue tracker ;) 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] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1
On 11-02-23 10:31 AM, Jose P. Espinal wrote: Hello List, I have a little issue with calls placed to a provider declared on sip.conf, because of a not clear (*for me*) behavior of 'remotesecret' parameter. Actually I was wrong! See here. It is being resolved. https://reviewboard.asterisk.org/r/1107/ 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] Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
On 11-02-22 10:16 AM, Ishfaq Malik wrote: Has this issue been fixed in this release of 1.8 (or even in the previous 1.8.2.3)? https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 No. The ChangeLog would give you the information you're looking for. http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3-rc3 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] Hide the plain text password
On 11-02-14 05:10 PM, Kevin P. Fleming wrote: On 02/14/2011 04:08 PM, Jian Gao wrote: I am building a server for a client. I want them to try out the new Google Voice feature using my GV account. But I don't want expose my GV's password. There is no method to obscure a Google Voice password in the config file. chan_sip supports obscured passwords using 'md5secret', but all other protocols that Asterisk supports need the password in plaintext to be able to perform the authentication process required by that protocol. You could use the #exec method to execute a script, where the configuration can be generated in any method you want on module load. In that way, you can extrapolate the information outside of Asterisk and secure it using any method you want. 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] Hide the plain text password
On 11-02-14 05:08 PM, Jian Gao wrote: I am building a server for a client. I want them to try out the new Google Voice feature using my GV account. But I don't want expose my GV's password. Actually in this case, your best bet is just going to be to create a separate account where you don't care about exposing the password to the user. 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] [modules.conf] Modules still loaded after noload
On 11-02-13 09:52 AM, Gilles wrote: I'm using Asterisk 1.4.20, and can't have Asterisk not load modules I don't need: Does someone know why Asterisk still loads modules even with the above lines in modules.conf? It looks like you're loading Asterisk, which loads all the modules, then modifying modules.conf and just doing a reload at that point. Try either restarting Asterisk to see if the modules still load (it shouldn't). Before doing the reload, I'd do a module unload chan_speex.so then do your reload and see if that works. I'm not sure reload actually looks at modules.conf at that point. It probably just reloads all the modules you have in memory, rather than unloading everything, then parsing modules.conf and loading everything in there back into memory (which I think is what you're expecting). 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] Asterisk Performance
On 11-02-01 05:22 PM, Juan David Diaz wrote: I would like to handle about 250 simultaneous (calls agents only) calls with PRI or a SIP trunk with the following configuration Dell R710 Dual Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz or Single Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz Memory 12GB, 1333MHz RAID 1 - 1 Tb X 2 Is that possible?? While it's certainly hard to accurately determine (without testing) what a system can handle, that certainly seems like an adequate machine to handle that kind of load, especially if you're not transcoding, doing much call recording, etc... (While that doesn't mean it can't handle all that, conferencing, recording calls (disk I/O), and transcoding are the heaviest uses of resources in my experience.) 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] Recommended Windows client to display CID?
On 11-01-26 08:52 AM, Gilles wrote: Hello I'd like to display CID information on users' monitor running Windows. You could use any XMPP client and send a message to it using JabberSend() from the dialplan. We document using it at http://ofps.oreilly.com. 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] Asterisk 1.8.2.3 Now Available
On 11-01-26 04:07 PM, Kevin P. Fleming wrote: On 01/26/2011 03:06 PM, Warren Selby wrote: Just curious, but why is this 1.8.2.3 and not just 1.8.3? I thought the new versioning methods made updates into 1.8.x releases and security updates into 1.8.x.y releases? Security fixes and regression fixes can cause sub-point releases. A version bump from 1.8.2 to 1.8.3 would mean all changes since 1.8.2-rc1 was created would be included. A bump from 1.8.2 - 1.8.2.1 - 1.8.2.2 - etc... includes minor changes based on the base 1.8.2 version with very select fixes. 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] Info on using LDAP with Asterisk?
On 11-01-23 02:56 PM, Jeff B wrote: There does not seem to be very much info out there about using LDAP to create asterisk configurations. Does anyone have some information that they would suggest I start with? We've tried to document some of it here: http://ofps.oreilly.com/titles/9780596517342/ch18.html#ExternalServices_id291590 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
Re: [asterisk-users] MOH and parking
On 11-01-21 08:52 AM, Andrew Thomas wrote: I know that the 'fix' has just been applied (https://issues.asterisk.org/view.php?id=18262) - but why does it stop the moh only to start it again? This, also, seems to cause a CDR problem (see below). After speaking with Shaun and Russell, this is likely related to some other part of code, and the fix that went in shouldn't have caused this issue. It's possible fixing this may have caused some other part of the code that was broken to be more prevalent though. Could you open an issue on bug tracker? 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
Re: [asterisk-users] Call sip:u...@domain.com?
On 10-12-17 06:48 AM, Gilles wrote:
On Fri, 17 Dec 2010 10:16:00 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Then create a prefix for SIP calls
exten=_9.,1,Dial(SIP/${EXTEN:1})
and you dial 9u...@domain.com from XLite
Remember that calling sip URL is not as easy with a phone. Imagine you have an
ATA with DECT or POTS
phone connected on it: how to send alpha characters or @ ?
Thanks Daniel. I added that line above, told Asterisk to reload the
dialplan, and typed the following in XLite:
9*031...@ekiga.net
This is to perform an echo test
http://wiki.ekiga.org/index.php/Fun_Numbers
You have to tell it the host to request the extension from. All you're doing is
dialing SIP/*031600, which with that format, is going to try and call [*031600]
as defined in sip.conf.
You're missing the host that you want to call. The format needs to be
SIP/*031600@some_hostname
What you're trying to do is essentially what FreeNum was designed for:
http://www.freenum.org
We discuss it in this chapter here:
http://ofps.oreilly.com/titles/9780596517342/ch12.html
Leif.
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Re: [asterisk-users] HA: what is missing to keep ongoing calls during failover ?
On 10-12-17 06:17 AM, Olivier wrote: Hi, What is currently missing in Asterisk ecosystem to get 2 servers active-active redundancy such as when server 1 is failing (in some circumstances), its ongoing calls (or most of them) are kept alive and handed over to server 2 ? I remember that a couple of years ago, Avaya claimed it could achieve this with its high end servers. Could it be possible with Asterisk ? Will SCF change this ? That's exactly what the demo at AstriCon showed SCF could do. 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] Which version to use: 1.4 or 1.6 or 1.8
On 10-12-15 09:46 AM, bilal ghayyad wrote: Hi All; I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8? For example, when to decide that I have to go for 1.6 or I have to go for 1.8? It depends on your required usage (features available in version) and your required support level. See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions for information on the support level and time for each of the branches. 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] Asterisk 1.8 Release Schedule
On 10-11-23 07:31 AM, --[ UxBoD ]-- wrote: I have read the wiki entry but unsure when we would likely see a 1.8.0.1 beta or release candidate ? It will be Asterisk 1.8.1-rc1 and that is now available (as of a few minutes ago) http://www.asterisk.org/node/51466 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] wideband recording in Asterisk 1.8
On 10-11-23 08:24 AM, Henry Dogger wrote: I have an aastra 6739i which supports the g722 codec. Which format setting do I need to be able to record in wideband? Tried: wav, gsm, pcm. Nothing seems to give me the result I desire. Shouldn't you try g722 as the format? 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] Volume on meetme recording
On 10-11-15 08:30 AM, Richard Kenner wrote: It's kind of low for me. How does one control that volume? You could use the VOLUME() function prior to joining the conference for channels that are quiet. 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] Best way to connect to a MySQL Database
On 10-11-15 06:04 PM, Matt Darnell wrote: Is this command the best way to access a MySQL database - MYSQL(Connect connid dhhost dbuser dbpass dbname) ? I thought I heard that using ODBC was a bit more stable. Anyone have any experience? Use func_odbc along with res_odbc. I've taken dialplans for customers who were having issues with MYSQL() and had about 9 lines of dialplan compressed down to 1 line of dialplan for the call, with much greater stability. Some information about that here: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-12.html And here: http://ofps.oreilly.com/titles/9780596517342/ch15.html (The second link is to the 3rd edition of the Asterisk book, currently being written, so this is a preview of the text in rough draft form.) 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] Extension Exists
On 10-10-25 04:21 PM, Dan Journo wrote: Hi, When a VOIP user dials an external number, the calls are routed through our SIP provider. Is there a simple way to check whether the DDI exists locally before dialling out to the sip provider? Something like GotoIfExists(5551...@incoming_calls) Well this is really an implementation question. If your data was in a database you could use func_odbc to check if the DID was local. You can check with VALID_EXTEN() to see if a particular extension exists locally. That's check the databse, so if you have a context that contains a list of your local DIDs you can check with that function. If the DIDs are available as a list on a webpage you can use func_curl. Using the DB_EXISTS() function could be used if storing in the Asterisk database. Those are some options. 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] Asterisk 1.8.0 Release Candidate 5 Now Available
On 10-10-18 11:01 PM, Barry Miller wrote: On Mon, Oct 18, 2010 at 07:58:07PM -0400, Asterisk Development Team wrote: On 10-10-18 07:54 PM, Asterisk Development Team wrote: For a full list of changes in the current release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc4 Apologies, this link should be: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc5 -- The Asterisk Development Team Is it worth mentioning somewhere (ChangeLog? This list?) that all the asterisk-core-sounds tarballs were updated today? It would remind someone [me!] who's trying to upgrade from an earlier rc to rc5 a chance to do a 'make sounds' before stopping asterisk for the install. My test system is on a slow link, and waiting for the tarball downloads in the middle of installing is frustrating. If you deselect the sounds from menuselect then you don't have to wait for them to download, and you can update them at your convenience later. 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] dahdi vmware query
On 10-10-19 10:46 AM, Danny Nicholas wrote: Greeting list, I hope this isn’t a “lazy” question. I have been running TDM400P and TDM410P cards in Dell PowerEdge Servers for a few years now. We are moving from physical servers to VMWARE servers. What opportunities should I expect moving these cards into the new machines? Or should I leave the existing machines intact and use IAX to get to the DAHDI lines from the VMWARE servers? Only the host system will be able to see the cards, so regardless you will need to use IAX or SIP to access the Asterisk instance that is hosting the hardware. 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] Queue Agent Getting Additional Calls When on the Phone
On 10-10-15 04:10 AM, Сикорский Сергей wrote: 15.10.2010 9:40, Warren Selby пишет: I think this means you need to set a call-limit for each sip peer Is there any alternative for obsolete call-limit option in 1.6/1.8? The correct answer is to use ringinuse=no in queues.conf and callcounter=yes in sip.conf. 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] Routers that do not show external IPs...
On 10-10-14 12:18 PM, Carlos Chavez wrote: I have a customer that has a Trendnet TEW-435BRM router which has the bad habit of rewriting all external connections so the Asterisk server only sees the IP address of the router itself. Up to today this has not been a problem since all extensions are on the local network but now they want to have a couple external IP phones (SIP). I opened up the ports on the router and my phone can register. The problem is that I have no audio because Asterisk thinks that the phone is on the internal network and does not use the NAT and externip settings. How do you deal with this kind of router so you can have external phones? Typically that is an option you can turn off. It is meant to help with SIP translations and such through the router, but as you're finding out, they typically just get in the way. Check through the web interface/configuration and see if there is anything about VoIP or SIP support in the router, and disable it. 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] Module reload
On 10-10-04 09:44 AM, Flavio Miranda wrote: Hi all, Every time I reload my asterisk it fall down and the following message appear on log: parse error: No category context for line 7 of /etc/asterisk/chan_dahdi.conf If I comment that line, it change to other line. There are some thing wrong with my dahdi? Sounds like you're missing some sort of [header] in the file, such as a [general] or something. Perhaps if you showed the configuration file it would be easier to determine what the actual problem is. 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] Module reload
On 10-10-04 10:59 AM, Flavio Miranda wrote: Asterisk:/var/log/asterisk# pico /etc/asterisk/chan_dahdi.conf ; DAHDI telephony ;language=en ;echocancel=yes echocancelwhenbridged=yes ss7type = itu ss7_called_nai=dynamic ss7_calling_nai=dynamic ;General options usecallerid = yes hidecallerid = no callwaiting = yes threewaycalling = yes transfer = yes echocancel = yes rxgain = 0.0 txgain = 0.0 ;FXO Modules group = 1 echocancel = yes signalling = fxs_ks context = local channel = 1 Ya, looks like you're missing [channels] 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] 1.4, 1.6, 1.8 versions and the H323 gatekeeper functionality
On 10-09-26 01:00 PM, bilal ghayyad wrote: First of all, I am looking to have the H323 Gatekeeper service available at Asterisk, and really I do not know if 1.4 or 1.6 or 1.8 started implementing H323 gatekeeper functionality or not? Until 1.4.26.2 version, there is no h323 gatekeeper functionality. So, any implementation for this feature has been done in the other versions? From what I'm aware of, no additional work has been done on the H323 modules available for Asterisk that implements any sort of gatekeeper functionality. 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] Asterisk 1.8.0 Release Candidate 2 Now Available
On 10-09-26 02:55 PM, Ira wrote: At 10:37 PM 9/24/2010, you wrote: You probably need to install libssl-dev then rerun ./configure. At least I did (Debian Lenny). Seems chan_sip needs res_crypto which needs libssl. Thanks, I tried to figure out what I needed but I failed. That was it, though on CentOS it seems to be openssl-devel. FYI, this is no longer an issue as of today. I opened an issue per the Asterisk development team, and Tilghman fixed the issue. https://issues.asterisk.org/view.php?id=18062 The next release candidate will allow chan_sip to use, but not require, the OpenSSL development libraries. 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
Re: [asterisk-users] rtp problem with 1.8.0-rdc1
On 10-09-23 05:40 PM, cov...@ccs.covici.com wrote: Hi. I am having a very strange problem --aren't they all -- with the release candidate. I have softphone which talks to asterisk from behind nat -- the asterisk is on a public ip -- and when I hit mute on the softphone, all rtp traffic ceases! Now, a version which does work is r281875, this does not happen in that vrsion, but right after that this strange thing starts and is not fixed in the current one. Any assistance here would be appreciated. We're probably going to need some sort of debugging information such as a console trace and SIP (I assume chan_sip) debug. More information here: doc/HOWTO_collect_debug_information.txt 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] Asterisk 1.6.2.13 - have asterisk reply from same IP address
On 10-09-23 07:01 PM, Mike wrote: Hi, I have a server with multiple IP address, Asterisk binding with all of them. I'd like Asterisk to reply to a SIP peer from the same IP address as the peer used to register to Asterisk (as opposed to using the main IP address all the time regardless of how the peer communicated with Asterisk). Is this possible? I know it wasn't with 1.4, but I was told 1.6 had something like this (something to do with not breaking SIP over TCP) This has been a requested feature for quite a while now, but I don't think Asterisk does this yet. Some time in the past Jared Smith had a patch that would do this which was considered hacky but did seem to work in his particular situation. If this has been implemented, I am not aware of it. 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] http://www.asterisk.org/downloads naming schema
On 10-09-22 11:45 AM, Klaus Darilion wrote: Hi! Since some time the download of the newest Asterisk does not contains the version number anymore, but is just called asterisk-1.4-current.tar.gz This gives me a tarball where I do not know the version without looking into the tarball. Thus, IMO it would be very useful to switch back to old schema war the download contained the version number. I don't understand really. The downloads.asterisk.org site contains the current version in the pub/telephony/asterisk/ directory, and there is a symlink to the current version which is named asterisk-1.4-current. On the Downloads page on asterisk.org we have the link setup to asterisk-1.4-current (and 1.6.2-current, etc.) but that again is just the symlink to the currently available version. The Downloads page is also updated with text in the table with the currently available version, such as 1.4.36 or 1.6.2.13, etc, so I'm not sure what you're asking to be changed. We've been doing it like this for quite some time (in the timeframe of years, to my knowledge). 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
Re: [asterisk-users] Bug with Realtime?
On 10-09-16 09:43 AM, Dan Journo wrote: That's not a bug. Only when the phone registers or performs some sort of action (such as placing a call, etc...) does Asterisk query the database. If your phones have a short re-registration time this becomes less of a problem. How do you explain that as soon as I issue a reload command, the realtime phones stop receiving calls? To test your theory, I rebooted the phone so that it had a fresh registration, I made and receives calls successfully, then issued a 'reload', then trying to dial in again, and the phone didnt ring. After a few seconds, the CLI says:- [2010-09-16 14:39:29] NOTICE[24611]: chan_sip.c:17200 sip_poke_noanswer: Peer 'kesher_201' is now UNREACHABLE! Last qualify: 58 How can this not be a bug? The phone works fine for hours, and then as soon as I issue a reload command, its UNREACHABLE. ps. The phone can still make calls after the reload. It just stops receiving calls after a reload. That sounds like a qualify issue in that the phone does not respond to a NOTIFY message. Check the SIP debug and see what is going on. Alternatively you could turn off the qualify option with qualify=no. 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] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11
On 10-09-15 05:25 AM, Jonas Kellens wrote:
I think I've found it :
Asterisk always reboots on this part :
[Sep 15 11:16:32] -- Goto (azura,pbx,1)
[Sep 15 11:16:32] -- Executing [...@azura:1]
NoOp(SIP/INTERTELin-, 3252480333 = pbx formule) in new stack
[Sep 15 11:16:32] -- Executing [...@azura:2]
Set(SIP/INTERTELin-, CDR(accountcode)=AZURAin) in new stack
[Sep 15 11:16:32] -- Executing [...@azura:3]
Set(SIP/INTERTELin-, BRON=473555006 473555006) in new stack
[Sep 15 11:16:32] -- Executing [...@azura:4]
Goto(SIP/INTERTELin-, vakantie) in new stack
[Sep 15 11:16:32] -- Goto (azura,pbx,5)
[Sep 15 11:16:32] -- Executing [...@azura:5]
Macro(SIP/INTERTELin-, vakantie,58) in new stack
[Sep 15 11:16:32] -- Executing [...@macro-vakantie:1]
MYSQL(SIP/INTERTELin-, Connect connid localhost username
passwd AsteriskHosted) in new stack
[Sep 15 11:16:32] -- Executing [...@macro-vakantie:2]
MYSQL(SIP/INTERTELin-, Query resultid 1 SELECT ast1 , ast2 ,
na , naID FROM vakantiedata where ID=58) in new stack
vps2301*CLI
Disconnected from Asterisk server
[Sep 15 11:16:32] Executing last minute cleanups
Dialplan :
[macro-vakantie]
exten = s,1,MYSQL(Connect connid localhost username passwd AsteriskHosted)
exten = s,n,MYSQL(Query resultid ${connid} SELECT ast1 , ast2 , na ,
naID FROM vakantiedata where ID=${ARG1})
exten = s,n,MYSQL(Fetch fetchid ${resultid} AST1 AST2 NA naID )
exten = s,n,NoOp(vakantie-ast1 = ${AST1} vakantie-ast2 = ${AST2} na =
${NA} naID = ${naID})
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
exten = s,n,NoOp(fetchid = ${fetchid})
exten = s,n,GoToIf($[${fetchid}==0]?exit)
exten = s,n,NoOp()
exten = s,n,GoToIfTime(${AST1}?opvakantie)
exten = s,n,GoToIfTime(${AST2}?opvakantie)
exten = s,n(exit),NoOp()
exten = s,n,Set(vakantieresult=continue)
exten = s,n,MacroExit
exten = s,n(opvakantie),NoOp(op vakantie !)
exten = s,n,GoToIf($[${NA}=hangup]?hangup:route)
Do you guys see why Asterisk has problems with this part of the dialplan ?!
I've seen problems with MYSQL() application crashing on customers boxes before.
It is not that well supported, and would greatly recommend you move to
func_odbc
usage for dialplan-database integration.
Not only will it simplify your dialplan, but likely will resolve your crashing
issues as well. I've done this for at least 3 customers who were using MYSQL()
and all crashing issues stopped and their dialplans ended up becoming
significantly more readable.
Leif.
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Re: [asterisk-users] Asterisk 1.6.2.12 Download
On 10-09-15 12:13 PM, Paul Belanger wrote: On Wed, Sep 15, 2010 at 11:54 AM, Ryan Wagonerrswago...@gmail.com wrote: Anybody else notice that the 1.6.2.12 download has a version and changelog for 1.6.2.12-rc1? I can confirm, asterisk-dev notified. Odd, not sure how this happened, but I'll be rebuilding a new release here shortly. Sorry for the noise. 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] Bug with Realtime?
On 10-09-15 03:41 PM, Dan Journo wrote: I think ive found a bug but need someone to double check. Whenever I issue a reload in Asterisk, any realtime extensions stop receiving calls. I have to reboot the sip phones in order to get them to re-register. Can anyone see if they have a similar problem? Asterisk 1.4.32 Mysql realtime. That's not a bug. Only when the phone registers or performs some sort of action (such as placing a call, etc...) does Asterisk query the database. If your phones have a short re-registration time this becomes less of a problem. 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
