Re: [asterisk-users] 100 outgoing calls - 10 at a time - /var/spool/asterisk/outgoing/
On Friday 28 September 2012, Patrick Archibald wrote: Hi, 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? Yes: Move them in batches of 10. Could be as simple as last if ++$n_files 9; if the script is in Perl. You know how many calls you can deal with at once; it's up to you to stay within your own limits. Asterisk just tries its damnedest to do whatever it's been told, without imposing any sort of judgement as to whether it's sane or wholesome. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Disconnect calls : known reasons
Hello, are there any known reasons why Asterisk would disconnect random calls ? My server uses 1,5 GB out of 8 GB RAM My server uses up to 35% CPU at peak There are about 40 concurrent calls. I have 300 RTP-ports available. I just see the call ending, as if one of the connected parties hung up but that is not the case ! So what could be a bottleneck ? Any known reasons for random hangup ? Kind regards, Jonas. -- _ -- 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] Disconnect calls : known reasons
Le 28/09/2012 10:22, Jonas Kellens a écrit : Hello, Hi are there any known reasons why Asterisk would disconnect random calls ? My server uses 1,5 GB out of 8 GB RAM My server uses up to 35% CPU at peak There are about 40 concurrent calls. I have 300 RTP-ports available. I just see the call ending, as if one of the connected parties hung up but that is not the case ! So what could be a bottleneck ? Any known reasons for random hangup ? Check that those calls are not sended to phones which are already online, doesn't accept further calls, and your dialplan don't manage this situation -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RealTime table fields ordering
Hi, According to http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip: Quote: If you place ipaddr before host (in the case of dynamic), you will never load the public IP address of your sip device, as it will be overwritten when host is encountered. UnQuote. From the latest Asterisk source tarball, the 'contrib' directory contains several realtime MySQL table definitions. The sippeers table has column 'ipaddr' before column 'host'. Also, 'permit' comes before 'deny'. Same for allow/disallow. Shouldn't the correct RealTime column/field order be: deny, permit and disallow, allow and host, ipaddr? As a side note, the iaxfriends RealTime MySQL table definition in the 'contrib' directory lacks the deny/permit fields which are quite important. However, the iaxfriends table does have the 'ipaddr' field after the 'host' field and the 'allow' field after 'disallow'. Furthermore, the asterisk.org wiki at: https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure shows the same disorder in deny/permit, allow/disallow and host/ipaddr (MySQL example for RealTime). 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. Or am I wrong about my observations concerning field order and field omissions? Thanks, Vieri -- _ -- 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] Disconnect calls : known reasons
On 28-09-12 10:31, Administrator TOOTAI wrote: Le 28/09/2012 10:22, Jonas Kellens a écrit : Hello, Hi are there any known reasons why Asterisk would disconnect random calls ? My server uses 1,5 GB out of 8 GB RAM My server uses up to 35% CPU at peak There are about 40 concurrent calls. I have 300 RTP-ports available. I just see the call ending, as if one of the connected parties hung up but that is not the case ! So what could be a bottleneck ? Any known reasons for random hangup ? Check that those calls are not sended to phones which are already online, doesn't accept further calls, and your dialplan don't manage this situation Maybe I need to explain a bit further : the call is send to the IP-phone and answered. The call lasts for about 1 à 2 minutes and is then disconnected. Sometimes this happens after about 1 minute, sometimes after about 2 minutes, sometimes it does not happen at all for a whole day or week ! If I say random, I really mean random. I'd love to believe that there is some kind of bottleneck or level that is reached when this happens, but nothing points to that direction. Kind regards, Jonas. -- _ -- 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 Fri, 2012-09-28 at 01:33 -0700, Vieri wrote: Hi, According to http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip: [snip] 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. Or am I wrong about my observations concerning field order and field omissions? Thanks, Vieri how about the line: `ipaddr` varchar(15) DEFAULT NULL, Wonder how they try to squeeze an IPv6 address in it... should be: `ipaddr` varchar(50) DEFAULT NULL, hans -- _ -- 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] Disconnect calls : known reasons
Le 28/09/2012 10:40, Jonas Kellens a écrit : [...] are there any known reasons why Asterisk would disconnect random calls ? My server uses 1,5 GB out of 8 GB RAM My server uses up to 35% CPU at peak There are about 40 concurrent calls. I have 300 RTP-ports available. I just see the call ending, as if one of the connected parties hung up but that is not the case ! So what could be a bottleneck ? Any known reasons for random hangup ? Check that those calls are not sended to phones which are already online, doesn't accept further calls, and your dialplan don't manage this situation Maybe I need to explain a bit further : the call is send to the IP-phone and answered. The call lasts for about 1 à 2 minutes and is then disconnected. Sometimes this happens after about 1 minute, sometimes after about 2 minutes, sometimes it does not happen at all for a whole day or week ! If I say random, I really mean random. I'd love to believe that there is some kind of bottleneck or level that is reached when this happens, but nothing points to that direction. We had this problem with some PSTN call termination providers, sometimes only against some destination. I don't know if your incoming calls are 100% VOIP, I would start to see with providers. You may also check hangupcause and dialstatus variables. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disconnect calls : known reasons
On 28-09-12 10:57, Administrator TOOTAI wrote: Le 28/09/2012 10:40, Jonas Kellens a écrit : [...] are there any known reasons why Asterisk would disconnect random calls ? My server uses 1,5 GB out of 8 GB RAM My server uses up to 35% CPU at peak There are about 40 concurrent calls. I have 300 RTP-ports available. I just see the call ending, as if one of the connected parties hung up but that is not the case ! So what could be a bottleneck ? Any known reasons for random hangup ? Check that those calls are not sended to phones which are already online, doesn't accept further calls, and your dialplan don't manage this situation Maybe I need to explain a bit further : the call is send to the IP-phone and answered. The call lasts for about 1 à 2 minutes and is then disconnected. Sometimes this happens after about 1 minute, sometimes after about 2 minutes, sometimes it does not happen at all for a whole day or week ! If I say random, I really mean random. I'd love to believe that there is some kind of bottleneck or level that is reached when this happens, but nothing points to that direction. We had this problem with some PSTN call termination providers, sometimes only against some destination. I don't know if your incoming calls are 100% VOIP, I would start to see with providers. You may also check hangupcause and dialstatus variables. Pure SIP. Hangupcause 16 Dialstatus Answer It has nothing to do with the provider-side. Kind regards. Jonas. -- _ -- 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 Fri, 9/28/12, Hans Witvliet aster...@a-domani.nl wrote: how about the line: `ipaddr` varchar(15) DEFAULT NULL, Wonder how they try to squeeze an IPv6 address in it... should be: `ipaddr` varchar(50) DEFAULT NULL, I think `ipaddr` varchar(45) DEFAULT NULL, should be enough. Vieri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?
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? Thank you! Markus -- _ -- 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] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?
Hi Leif, Am 28.09.2012 13:24, schrieb Leif Madsen: 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. thanks for your reply. Right, doesn't look 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. 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 ... Thank you! Markus -- _ -- 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 Sep 2012, at 13:27, Leif Madsen leif.mad...@asteriskdocs.org wrote: 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 Hi Leif, I am happy to hear that a new release of The Book is in the works! 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!! TIA, Aldo Sent from my iPhone -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] User expected behavior of musiconhold and AGI's stream file
Hi everyone, I am your friendly neighborhood developer here with a question that may impact some of you. 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? Do you depend on this behavior knowingly? Do you depend on this behavior without knowing it? 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 behavior 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. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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] 100 outgoing calls - 10 at a time - /var/spool/asterisk/outgoing/
On 28 Sep 2012, at 14:24, Leif Madsen leif.mad...@asteriskdocs.org wrote: That's good! I'd hate to be working on something no one wanted :) ;-))) 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. Yes, I know... I have been writing an Objective-C library to control Asterisk over AMI, but I did not see the event that you need to send to just trigger a script, say, to read the log files and get the last calls, or anything similar. I can originate calls with an AMI Originate command and some dialplan glue, but I am missing anything like an Exec AMI command. -- Leif Madsen http://www.oreilly.com/catalog/asterisk TIA, Aldo Sent from my iPhone -- _ -- 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
Leif Madsen wrote: 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? :) 3 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? Sure you can! You can use set music to start it going again as the next command. 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? 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. 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. 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. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP DTMF Flash Event
Tim Nelson wrote: Is there a way to have Asterisk respond appropriately when receiving a DTMF Flash event via SIP? I'm finding some WiFi SIP phones, specifically the Quickphones QA-342 want to handle transfers/3-way calls by sending a DTMF Flash event instead of handling it properly like every other damn VoIP phone on the planet... Asterisk sees the Flash event (via the logs), but does not act upon it. Thoughts? Hola, The functionality you are talking about (server side SIP transfers and conferences) are not really implemented for use like this. All of the pieces exist within Asterisk to achieve the expected end result but as this has only ever come up maybe twice noone has ever taken the time to do it. Sorry! Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?
Hi again Leif, Am 28.09.2012 13:42, schrieb Leif Madsen: 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. Thanks for the suggestions. (RFC2833 was already set on the SIP peer in question) For further debugging I've just done the following tests with these results: DID provider 1, incoming via SIP (Germany): PSTN to DID to X-Lite, RFC2833: hear DTMF, not logged on console PSTN to DID to X-Lite, inband: hear DTMF, not logged on console PSTN to DID to X-Lite, INFO: hear DTMF, not logged on console PSTN to DID to ConfBridge, RFC2833: hear DTMF, LOGGED on console PSTN to DID to ConfBridge, inband: hear DTMF, not logged on console PSTN to DID to ConfBridge, INFO: hear DTMF, not logged on console DID provider 2, incoming via SIP (UK and Netherlands): PSTN to DID to X-Lite, RFC2833: hear DTMF, LOGGED on console PSTN to DID to X-Lite, inband: hear DTMF, LOGGED on console PSTN to DID to X-Lite, INFO: hear DTMF, not logged on console PSTN to DID to ConfBridge, RFC2833: hear DTMF, LOGGED on console PSTN to DID to ConfBridge, inband: hear DTMF, LOGGED on console PSTN to DID to ConfBridge, INFO: hear DTMF, not logged on console DID provider 3, incoming via SIP (Nigeria and Kazakhstan): PSTN to DID to ConfBridge, inband: hear DTMF, LOGGED on console DID provider 4, incoming via SIP (US): PSTN to DID to ConfBridge, RFC2833: hear DTMF, LOGGED on console X-Lite 3.0 to ConfBridge: X-Lite peer is set to dtmfmode=rfc2833 in Asterisk. X-Lite force inband YES, RTP 2833 YES: don't hear DTMF, LOGGED on console X-Lite force inband NO, RTP 2833 YES: don't hear DTMF, LOGGED on console X-Lite force inband YES, RTP 2833 NO: hear DTMF, not logged on console X-Lite force inband NO, RTP 2833 NO: don't hear DTMF, LOGGED on console The last result is kinda strange. Hm. PSTN means that I've tested two times, from a regular landline and from a mobile. Always calling to the providers DID which ends up in Asterisk via SIP. In the case of ConfBridge there were always 2 participants in the conference so that I could check if I hear the DTMF on the other end. not logged on console means that I can hear the DTMF tones in X-Lite/ConfBridge but Asterisk doesn't seem to recognize them (which is fine as not all providers support all DTMF variants). My resume is: DTMF is just fine, ConfBridge dtmf_passthrough is not working at all. Agree? :) Thank you, Markus -- _ -- 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?
Markus wrote: Snipped long results list PSTN means that I've tested two times, from a regular landline and from a mobile. Always calling to the providers DID which ends up in Asterisk via SIP. In the case of ConfBridge there were always 2 participants in the conference so that I could check if I hear the DTMF on the other end. I think your results are sort of skewed. In the case of SIP - SIP if a local bridge occurs things will optimize and you most likely won't see DTMF related messages. They get passed through as packets and not fully interpreted. not logged on console means that I can hear the DTMF tones in X-Lite/ConfBridge but Asterisk doesn't seem to recognize them (which is fine as not all providers support all DTMF variants). What log message are you using to determine this? My resume is: DTMF is just fine, ConfBridge dtmf_passthrough is not working at all. Agree? :) I've looked at the code for dtmf_passthrough, it's dead simple and should be working fine PROVIDED your DTMF is not going through as audio. My suggestion is to take a step back further. Just send incoming calls to the Read application and have it store the received DTMF in a variable. Next step have it output what was received. If that works for all cases then Asterisk is recognizing DTMF fine. This does *not* mean that the tone will be muted fully as my previous email mentioned. You can further test if all cases check out by sending calls to Record and playing back the audio to yourself. If you hear tones and Asterisk also recognized the DTMF then it's not fully muted, or the hardware in question is sending *both* inband and out of band, which is not supported. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Hold problem
Hello everybody, i have a problem with asterisk 1.8 and Call Hold My problem is that Asterisk don't send re-invite when i pick up the call from hold. I already insert canreinvite=no in all my sip channels, set dtmfmode=info in sip.conf and my Dial() command don't insert option like t, T, h, H, w, W or L (with multiple arguments). I already follow this discussion : http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial I run debug with asterisk, and i see that the re-invite are made by asterisk, but in the TO fields is present the local ip address and not the next hop ip. This is the log : http://pastebin.com/ARUC0j4t The asterisk IP : 87.248.56.101 The next hop IP : 87.248.56.100 Is it a bug? i'm already search on google, but i dont find anything. Let me know, if you need more information. Thanks for all Best Regards Marco -- _ -- 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?
Hi Joshua, Am 28.09.2012 15:56, schrieb Joshua Colp: I think your results are sort of skewed. In the case of SIP - SIP if a local bridge occurs things will optimize and you most likely won't see DTMF related messages. They get passed through as packets and not fully interpreted. ah, ok! That explains why nothing was logged when testing PSTN to DID to X-Lite, RFC2833, and something was logged when repeating the call to ConfBridge. not logged on console means that I can hear the DTMF tones in X-Lite/ConfBridge but Asterisk doesn't seem to recognize them (which is fine as not all providers support all DTMF variants). What log message are you using to determine this? Just by watching the console for DTMF after I press a key (logger.conf: console = dtmf). And confirmed by the fact that the DTMF keypress didn't have any effect (such as adjusting the volume, for example). My resume is: DTMF is just fine, ConfBridge dtmf_passthrough is not working at all. Agree? :) I've looked at the code for dtmf_passthrough, it's dead simple and should be working fine PROVIDED your DTMF is not going through as audio. You're right, I was wrong. It is working, but in my tests only in the X-Lite scenario. My suggestion is to take a step back further. Just send incoming calls to the Read application and have it store the received DTMF in a variable. Next step have it output what was received. Ok, good idea, here are the results of Read() and SayDigits(): DID provider 1, RFC2833: input 123, output 123 DID provider 1, inband: input 123, output User entered nothing. DID provider 1, INFO: input 123, output User entered nothing. DID provider 2, RFC2833: input 123, output 123 DID provider 2, inband: input 123, output User entered nothing. DID provider 2, INFO: input 123, ouput User entered nothing. DID provider 3, RFC2833 1st test: input 123, output 1233 DID provider 3, RFC2833 2nd test: input 123, output 12333 DID provider 3, RFC2833 3rd test: input 123, output 123 DID provider 3, inband: input 123, output 123 DID provider 3, INFO: input 123, output User entered nothing. (RFC2833 seems a bit flakey with that provider, thats why I use inband with them.) DID provider 4, RFC2833: input 123, output 123 DID provider 4, inband: input 123, output User entered nothing., but something strange happened here. Console shows: channel.c:4143 __ast_read: DTMF begin '1' received on SIP channel.c:4147 __ast_read: DTMF begin ignored '1' on SIP channel.c:4143 __ast_read: DTMF begin '2' received on SIP channel.c:4147 __ast_read: DTMF begin ignored '2' on SIP channel.c:4143 __ast_read: DTMF begin '3' received on SIP channel.c:4147 __ast_read: DTMF begin ignored '3' on SIP channel.c:4143 __ast_read: DTMF begin '#' received on SIP channel.c:4147 __ast_read: DTMF begin ignored '#' on SIP DID provider 4, INFO: input 123, output User entered nothing., and again the same on the console: channel.c:4143 __ast_read: DTMF begin '1' received on SIP channel.c:4147 __ast_read: DTMF begin ignored '1' on SIP channel.c:4143 __ast_read: DTMF begin '2' received on SIP channel.c:4147 __ast_read: DTMF begin ignored '2' on SIP channel.c:4143 __ast_read: DTMF begin '3' received on SIP channel.c:4147 __ast_read: DTMF begin ignored '3' on SIP channel.c:4143 __ast_read: DTMF begin '#' received on SIP channel.c:4147 __ast_read: DTMF begin ignored '#' on SIP (Asterisk sees the DTMF but doesn't like it?) If that works for all cases then Asterisk is recognizing DTMF fine. This does *not* mean that the tone will be muted fully as my previous email mentioned. I don't see any previous eMail from you on the list and there is nothing in the archives either. Could you re-send it, please? Maybe the info that I'm missing is inside that mail. :) You can further test if all cases check out by sending calls to Record and playing back the audio to yourself. If you hear tones and Asterisk also recognized the DTMF then it's not fully muted, or the hardware in question is sending *both* inband and out of band, which is not supported. Ok, here are the results of Record() and Playback(): DID provider 1, RFC2833: input 123, output hear DTMF (123) DID provider 1, inband: input 123, output hear DTMF (123) DID provider 1, INFO: input 123, output hear DTMF (123) DID provider 2, RFC2833: input 123, output hear DTMF (123) DID provider 2, inband: input 123, output hear DTMF (123) DID provider 2, INFO: input 123, output hear DTMF (123) DID provider 3, RFC2833: input 123, output hear DTMF (123) DID provider 3, inband: input 123, output hear DTMF (123) DID provider 3, INFO: input 123, output hear DTMF (123) DID provider 4, RFC2833: input 123, output hear DTMF (123) DID provider 4, inband: input 123, output hear DTMF (just 1 digit) (Console showed: DTMF begin passthrough) DID provider 4, INFO: input 123, output hear DTMF (just 1 digit) (Console showed: DTMF begin passthrough) X-Lite, RFC2833 in sip.conf: input 123, output hear NOTHING X-Lite, inband in
Re: [asterisk-users] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?
Markus wrote: Hi Joshua, Hola, My suggestion is to take a step back further. Just send incoming calls to the Read application and have it store the received DTMF in a variable. Next step have it output what was received. Ok, good idea, here are the results of Read() and SayDigits(): snipped results to make this email manageable How are you changing the DTMF for each provider? If you are merely changing it using dtmfmode in sip.conf this may or may not change how the provider side sends it. In the case of setting it to rfc2833 it causes RFC2833 to be negotiated in the SDP. Some equipment MAY change to using inband if it has not been negotiated. If that works for all cases then Asterisk is recognizing DTMF fine. This does *not* mean that the tone will be muted fully as my previous email mentioned. I don't see any previous eMail from you on the list and there is nothing in the archives either. Could you re-send it, please? Maybe the info that I'm missing is inside that mail. :) I think I wrote the email in my head, oops. Essentially when doing conversion of inband DTMF to out of band DTMF it is possible for some parts of tones to get through unmuted. You have to strike a balance between detecting the DTMF early enough, not detecting other stuff as DTMF, and muting it. Some implementations may let some leak through. The only way to completely overcome that is to buffer enough audio and delay the stream. You can further test if all cases check out by sending calls to Record and playing back the audio to yourself. If you hear tones and Asterisk also recognized the DTMF then it's not fully muted, or the hardware in question is sending *both* inband and out of band, which is not supported. Ok, here are the results of Record() and Playback(): snipped results again, can be viewed in mailing list archives if anyone is curious If I understand right, all my four DID providers are broken? If the provider is doing the conversion their equipment should mute the inband DTMF as best it can and you should not hear it. How much of the tone are you hearing? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 100 outgoing calls - 10 at a time - /var/spool/asterisk/outgoing/
Thanks everyone for the guidance on my outgoing call question. And I'm looking forward to Asterisk: The Definitive Guide 4e book too! Thanks, PLA -- _ -- 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?
Am 28.09.2012 17:33, schrieb Joshua Colp: Ok, good idea, here are the results of Read() and SayDigits(): snipped results to make this email manageable How are you changing the DTMF for each provider? If you are merely changing it using dtmfmode in sip.conf this may or may not change how the provider side sends it. In the case of setting it to rfc2833 it causes RFC2833 to be negotiated in the SDP. Some equipment MAY change to using inband if it has not been negotiated. Yes, only via dtmfmode in sip.conf. I have no control over the provider side, of course. :) Or are there other options? Essentially when doing conversion of inband DTMF to out of band DTMF it is possible for some parts of tones to get through unmuted. You have to strike a balance between detecting the DTMF early enough, not detecting other stuff as DTMF, and muting it. Some implementations may let some leak through. The only way to completely overcome that is to buffer enough audio and delay the stream. Ah! So because I'm calling from PSTN which is naturally inband (I guess?) my DID provider (or rather some provider in front of him) converts to out of band (RFC2833), but the conversion cannot be fully accurate? If I understand right, all my four DID providers are broken? If the provider is doing the conversion their equipment should mute the inband DTMF as best it can and you should not hear it. How much of the tone are you hearing? About the tone itself: it's always different for each DTMF method/provider I would say. Sometimes it's just a really short glub (no idea how to describe that better), and sometimes it's a full-length beeep. But in any case except for provider 4 inband + INFO I'm hearing all of it (123), so it's anything from glub-glub-glub to beeep-beeep-beeep. :-) Right now, after what you wrote, I'm left with the feeling that RFC2833 is pretty much, well, hmm... useless in a PSTN-VoIP scenario? It works fine when using X-Lite, I suppose because there is no conversion involved and therefore the tones can get completely muted. I'm sad now. Is there nothing I can do to remove DTMF tones from my conferences? :-( Thank you! Markus -- _ -- 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?
Markus wrote: Am 28.09.2012 17:33, schrieb Joshua Colp: Ok, good idea, here are the results of Read() and SayDigits(): snipped results to make this email manageable How are you changing the DTMF for each provider? If you are merely changing it using dtmfmode in sip.conf this may or may not change how the provider side sends it. In the case of setting it to rfc2833 it causes RFC2833 to be negotiated in the SDP. Some equipment MAY change to using inband if it has not been negotiated. Yes, only via dtmfmode in sip.conf. I have no control over the provider side, of course. :) Or are there other options? There isn't. It's up to them. Essentially when doing conversion of inband DTMF to out of band DTMF it is possible for some parts of tones to get through unmuted. You have to strike a balance between detecting the DTMF early enough, not detecting other stuff as DTMF, and muting it. Some implementations may let some leak through. The only way to completely overcome that is to buffer enough audio and delay the stream. Ah! So because I'm calling from PSTN which is naturally inband (I guess?) my DID provider (or rather some provider in front of him) converts to out of band (RFC2833), but the conversion cannot be fully accurate? Correct. The equipment doing the conversion may or may not fully remove the tone. If I understand right, all my four DID providers are broken? If the provider is doing the conversion their equipment should mute the inband DTMF as best it can and you should not hear it. How much of the tone are you hearing? About the tone itself: it's always different for each DTMF method/provider I would say. Sometimes it's just a really short glub (no idea how to describe that better), and sometimes it's a full-length beeep. But in any case except for provider 4 inband + INFO I'm hearing all of it (123), so it's anything from glub-glub-glub to beeep-beeep-beeep. :-) Right now, after what you wrote, I'm left with the feeling that RFC2833 is pretty much, well, hmm... useless in a PSTN-VoIP scenario? It works fine when using X-Lite, I suppose because there is no conversion involved and therefore the tones can get completely muted. It's not completely useless. Some equipment completely removes the tone and life is good. It's also perfectly fine for IVRs. Where it can fall apart is in this exact situation where you have someone on the other side who shouldn't hear the DTMF. That's not often. As for the glub I would say that is a partial squelching (muting). It was able to mute most of the tone but not completely. Unfortunately there's nothing I can suggest to help improve the situation, sorry. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call me now outbound calls in a queue
I want to put a call me now button on the web site that will place the request into an asterisk call queue and then when an agent picks up the call in the queue, place the outbound call to the customer. The following AMI command works, but it calls the customer first, before an agent is necessarily available. Action: Originate Channel: SIP/voipms/customer_number_here Context: external Async: true Application: Queue Data: sales Callerid: Company 8005551212 How can I get an available agent before the customer call is placed? -- _ -- 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: 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 . On Fri, 28 Sep 2012, Leif Madsen wrote: 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 :) A multi-threaded AGI or externalivr() sound easier to me if it meets your needs. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call me now outbound calls in a queue
On 9/28/2012 12:42 PM, Mitch Claborn wrote: I want to put a call me now button on the web site that will place the request into an asterisk call queue and then when an agent picks up the call in the queue, place the outbound call to the customer. The following AMI command works, but it calls the customer first, before an agent is necessarily available. Action: Originate Channel: SIP/voipms/customer_number_here Context: external Async: true Application: Queue Data: sales Callerid: Company 8005551212 How can I get an available agent before the customer call is placed? Use AMI to do a queue status. In the status reply, you'll get a list of all the agents and their status (Not in use, in use, busy, unavailable, or ringing). -- _ -- 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 me now outbound calls in a queue
That approach only works if there are any agents that are not busy on a call - I could pick one, take them out of the queue then connect the call. If all agents are busy, I need to be able to insert the request into the queue so that it gets processed in sequence with the inbound calls. Mitch On 09/28/2012 01:42 PM, James Sharp wrote: On 9/28/2012 12:42 PM, Mitch Claborn wrote: I want to put a call me now button on the web site that will place the request into an asterisk call queue and then when an agent picks up the call in the queue, place the outbound call to the customer. The following AMI command works, but it calls the customer first, before an agent is necessarily available. Action: Originate Channel: SIP/voipms/customer_number_here Context: external Async: true Application: Queue Data: sales Callerid: Company 8005551212 How can I get an available agent before the customer call is placed? Use AMI to do a queue status. In the status reply, you'll get a list of all the agents and their status (Not in use, in use, busy, unavailable, or ringing). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 'Training mode'
I was asked today if we could somehow have a trainee on the phone with a supervisor conferenced in, but somehow have it so anything the supervisor says is only heard by the trainee and not the customer. Is there a feature like that? -- _ -- 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] 'Training mode'
On Fri, Sep 28, 2012 at 5:27 PM, Adam Moffett adamli...@plexicomm.net wrote: I was asked today if we could somehow have a trainee on the phone with a supervisor conferenced in, but somehow have it so anything the supervisor says is only heard by the trainee and not the customer. Is there a feature like that? -- Yup, pretty standard stuff https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_ChanSpy -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strategy for custom data in the CDR
Looking for ideas and comments on my strategy for getting a bit of
custom data into the CDR. It seems to work OK, but I'm open to better
and/or more robust ways to do it.
Problem: get the customerid of the caller from our application into the CDR
Approach:
Before the Queue() command, save the caller's channel name in a varible
_MMCALLERCHANNEL
In the Queue() connect macro, send that channel name along with other
info to the application
The application determines the customerid and uses AMI Setvar to set
variable MMCUSTOMERID on the given channel with the customerid. (Can't
use AGI in-line because the customer will be determined by an actual
human sitting at a terminal during the call and may take a while.)
In the hangup extension, set CDR(customerid) to value of MMCUSTOMERID
Here is the code that is working
[queues]
exten =sales,1,Verbose(2,${CALLERID(all)} entering the sales queue)
same =n,Set(_MMCALLERCHANNEL=${CHANNEL(name)})
same =n,Queue(sales,t,QueueConnected)
same =n,Hangup()
exten =h,1,NoOp(When a sales queue call is hung up)
same =n,Set(CDR(customerid)=${MMCUSTOMERID})
[macro-QueueConnected]
exten =s,1,NoOp()
same =n,Monitor(wav,,mb)
same
=n,Set(OPERATORID=${ODBC_OPERATORID_FROM_ADDRESS(${MEMBERINTERFACE})})
; userfield is mapped to operatorid in cdr_adaptive_odbc because
setting operatorid directly doesn't work here
same =n,Set(CDR(userfield)=${OPERATORID})
; passes the call to the application
same =n,System(${MMAGI}/queue_action.sh
action:connect,queue:sales,member_interface:\\${MEMBERINTERFACE}\\,member_name:\\${MEMBERNAME}\\,cidnum:\\${CONNECTEDLINE(num)}\\,cidname:\\${CONNECTEDLINE(name)}\\,uniqueid:\\${MMLINKEDID}\\,channel:\\${MMCALLERCHANNEL}\\)
--
Mitch
--
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