Unfortunately the tests produce the same results. Next we will move on to explain how to handle situations where a call is parked but is not retrieved before the value specified as the parkingtime option elapses. PDF. The pages in this section will describe what the elements of dialplan are and how to use them in your configuration. scheduled tasks” crashing means your CDR records (queue) are being written as the call ends, and if you had many thousands of entries being written to disk it crashes asterisk (each ring to one phone is an entry, so it goes up fast – for example 10 busy phones, with a between-ring delay of 1 The following examples demonstrate an AudioSocket connection to a server at … When I began experiencing this issue I used MoH as an attempt to narrow down the problem to the simplest dialplan possible. Abdul Salam. I've seen many weird errors in Asterisk before, that didn't harm the actual function of the pbx. Home » Asterisk Users » ERROR During High Volume MoH Dialplan. Use included samples (templates) to create dialplan in minutes. I have it connected to my bell system (installation is in a school) so that we can do overhead paging. Content-Type: text/plain; We want to restart the system by making a call. 0 modules loaded, # grep enable= /etc/asterisk/cdr.conf enable=no. Using the distro and Asterisk 13, you just need to install the ws_node package “npm install -g wscat”. I am using SIPP to test. object used in the code. Since, these error proceeded that I thought that they may be the key to preventing the queue from maxing out. [UPDATED: 29 Mar 2014] - IMPORTANT: THE PATCH IS NO LONGER NEEDED IN ASTERISK 11.5 The following guide was taken off various sources as initial references such as Digium’s Wiki and sipML5’s how to for Asterisk found here. See Section 7 for more information. People are often tempted to implement all sorts of fancy functionality in the emergency services portions of their dialplans, but if a bug in one of your fancy features causes an emergency call to fail, lives could be at risk. Licensing. Privilege Escalations with Dialplan Functions. The dialplan is essentially a scripting language specific to Asterisk and one of the primary ways of instructing Asterisk on how to behave. Download Full PDF Package. Arguments. The Asterisk dialplan is found in the extensions.conf file in the configuration directory, typically /etc/asterisk. [Sep 1 20:36:45] ERROR[10081][C-00007fe5]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x20380b0 (. +1 for horizontal scaling as the best solution in this situation. I set no optimize and better backtrace through “make menuselect” and the output is now, [Aug 28 21:41:16] ERROR[17171][C-0000392d]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x21962b0 (0), #0: [0x61923f] main/utils.c:2475 __ast_assert_failed() (0x6191bb+84), #1: [0x45ffc9] main/astobj2.c:543 __ao2_ref() (0x45fc3d+38C), #2: [0x5320ce] main/frame.c:345 ast_frdup() (0x531e4c+282), #3: [0x531a99] main/frame.c:196 ast_frisolate() (0x531a76+23), #4: [0x60be51] main/translate.c:459 ast_trans_frameout() (0x60bd6e+E3), #5: [0x60be75] main/translate.c:464 default_frameout(), #6: [0x60c46a] main/translate.c:579 ast_translate() (0x60c192+2D8), #7: [0x4c0bf1] main/channel.c:5290 ast_write() (0x4bfb3e+10B3), #8: [0x7fdef8345486] res/res_musiconhold.c:455 moh_files_generator(), #9: [0x4ba212] main/channel.c:3014 generator_force(), #10: [0x4bc23d] main/channel.c:3872 __ast_read(), #11: [0x4be29b] main/channel.c:4399 ast_read() (0x4be27e+1D), #12: [0x4b6312] main/channel.c:1568 ast_safe_sleep_conditional() (0x4b6229+E9), #13: [0x4b64c9] main/channel.c:1613 ast_safe_sleep() (0x4b64a1+28), #14: [0x7fdef8346caa] res/res_musiconhold.c:834 play_moh_exec(), #15: [0x5970a3] main/pbx_app.c:491 pbx_exec() (0x596f87+11C), #16: [0x582edf] main/pbx.c:2923 pbx_extension_helper(), #17: [0x586c30] main/pbx.c:4155 ast_spawn_extension() (0x586bcc+64), #18: [0x5878bb] main/pbx.c:4328 __ast_pbx_run(), #19: [0x589061] main/pbx.c:4651 pbx_thread(), #20: [0x61624e] main/utils.c:1233 dummy_start(). SetAccount - this application sets an account code for billing purposes. That is out of my hands at the moment unless it as well. Printed by Atlassian Confluence 5.6.6, Team Collaboration Software. How you generate this TIFF is important, and may involve many steps. Free PDF. , ——=_NextPart_001_0073_01D32341.E9678B80 It ties everything together, allowing you to route and manipulate calls in a programmatic way. Is that simply a side effect of having so many callers listening to the IVR at the same time? I am using SIPP to test. I copied all my phones extension dial plan and placed it under [local]. Based upon the inline backtrace the ao2 object is likely to be a codec format. Then this time Asterisk actually crashed. Install the FreePBX “Asterisk REST Interface Users” module if necessary. Content-Type: text/plain; charset=”Windows-1252″ This is a simplistic calculation as there are going to be some references that have nothing to do with a call. Download PDF. When set to “yes”, the dialplan will jump to priority +101 on busy, congested, and channel unavailable. However, from Asterisk’s perspective the sending of a fax is fairly straightforward. anyone have any advice on what that could be or because of transcoding? I also commented out all of [internal-office] Reloaded the dial plan and verified that my phones extensions were in fact loaded under [local]. Members are those channels that are active in answering the Queue. The Asterisk dialplan is responsible for routing calls, so it is often referred to as the heart of an Asterisk system. Digium Or Sangoma? Actually, the handling is so limited that if, for some reason, a FastAGI script fails during execution, Asterisk will simply disconnect the call. It sounds like Richard is saying that these refcount logs may not actually be errors and can be ignored in this scenario. On my systems I have MoH and sounds installed in wav, ulaw, alaw, gsm and g729. exten => 1001,n,MusicOnHold(15) exten => 1001,n,Hangup. I did run into a CDR bottleneck as well and have already disabled it, Module Description Use Count Status Support Level But most sip clients and sip servers in the market do not accept RE-INVITE requests. [ 94 ] Although Macro() seems like a general-purpose dialplan subroutine, it has a stack overflow problem that means you should not try to nest Macro() calls more than five levels deep. Premium PDF Package. [Sep 1 20:36:46] WARNING[7761][C-0000770d]: taskprocessor.c:888 taskprocessor_push: The ‘subp:PJSIP/sipp-00000020’ task processor queue reached 500 scheduled tasks. Asterisk- The Definitive Guide, 4th Edition. Basic Handling for Call Parking Timeouts. When I was first approached with this task I mentioned as much. The Asterisk Development Team would like to announce security releases for Asterisk 13, 16, 17 and 18. 01. I The dialplan for handling emergency calls does not need to be complicated. However, the current desire is to work with already existing hardware. This is the task processor that is maxing out. The Asterisk dialplan. * What codecs are you using in this setup? I initially tested with the IVR audio files. A form of scripting language, the dialplan contains instructions that Asterisk follows in response to external triggers. The Asterisk server has to be running in the background for the CLI to start. See Also. You will find it less taxing on the server if you have MoH files and sounds files available in all the possible native formats. Steps 1 and 2 are done entirely within the GUI in advanced settings and Asterisk REST Interface users. The dialplan is written in a special scripting language, and it is extremely powerful. This dial plan application is used for assigning value to a variable. So, after 32 seconds, Asterisk hangs up the call. Jumping in Asterisk v1.2.14: In [general] you can set priorityjumping=yes/no. Each of these lends itself to simplify a different use-case, but they work in exactly the same way. In Asterisk dialplan application we can see that applications like SetCIDName, SetCIDNum, SetLanguage, SetVar are being deprecated in favour of Set ( Set(CALLER(name)=…), Set(CALLER(number)=…), Set(LANGUAGE()=…)). In this case, we’re handling the NOANSWER and BUSY cases, and treating all other result codes as a NOANSWER. Behind the scenes of any VoIP Application for the Asterisk PBX. I was using a MySQL CDR, but I had left the “CSV” type of CDR on. The Asterisk command line interface (CLI) is reached by using the Linux shell command asterisk -r or rasterisk. By default Asterisk sends a RE-INVITE request after a call is established. It … At around 500 calls per second I begin to see the following ERRORs, [Aug 28 17:46:14] ERROR[26150][C-00005594]: frame.c:343 ast_frdup: Excessive refcount 100000 reached on ao2 object 0x26bffc0, [Aug 28 17:46:14] ERROR[26150][C-00005594]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x26bffc0 (0), #0: [0x45d229] /usr/sbin/asterisk(__ao2_ref+0x1a9) [0x45d229], #1: [0x526ce6] /usr/sbin/asterisk(ast_frdup+0x116) [0x526ce6], #2: [0x5fa616] /usr/sbin/asterisk(ast_translate+0x306) [0x5fa616], #3: [0x4bf16b] /usr/sbin/asterisk(ast_write+0x104b) [0x4bf16b], #4: [0x7efeb578230b] /usr/lib/asterisk/modules/res_musiconhold.so(+0x430b) [0x7efeb578230b], #5: [0x4b5b52] /usr/sbin/asterisk() [0x4b5b52], #6: [0x4c259c] /usr/sbin/asterisk() [0x4c259c], #7: [0x4c4a45] /usr/sbin/asterisk() [0x4c4a45], #8: [0x7efeb578478d] /usr/lib/asterisk/modules/res_musiconhold.so(+0x678d) [0x7efeb578478d], #9: [0x58ec79] /usr/sbin/asterisk(pbx_exec+0xb9) [0x58ec79], #10: [0x582e84] /usr/sbin/asterisk() [0x582e84], #11: [0x584e7c] /usr/sbin/asterisk() [0x584e7c], #12: [0x5863fb] /usr/sbin/asterisk() [0x5863fb], #13: [0x60002a] /usr/sbin/asterisk() [0x60002a]. filename. 05. reason - INVALID, ERROR, RESPONSETIMEOUT, ABSOLUTETIMEOUT, or custom value set by the RaiseException() application; context - The context executing when the exception occurred. Here is the situation: I have FreePBX 4.211.64-5 installed and running. –_000_CY4PR2201MB14642220BB9A07CA7AA5EE6BA8960CY4PR2201MB1464_ I’ve tested on asterisk 13.5 and 14.6 with the same results. I do feel like there must be something I’m missing but just can’t to it. The sample file includes many examples of dialplan programming for specific scenarios and environments often common to Asterisk implementations. SetAMAflags - this application sets AMA flags 06. Please ignore the noise, I need to slow down when I read. I have also tested with a separate set of audio files closer to what the actual IVR menu. filename; format - Is the format of the file type to be recorded (wav, gsm, etc). I’m not a fan of 4,000 eggs in one basket. The dialplan is essentially a scripting language specific to Asterisk and one of the primary ways of instructing Asterisk on how to behave. The FRACK itself is benign. The release of Asterisk 18.0.0 resolves several issues reported by the community and would have not been possible without your participation. * What codecs are you using in this setup? ResetCDR - this application resets the CDR 04. Can anyone enlighten me on the meaning and cause of the error? If so would it help to change files I am using are gsm. org/pub/telephony/asterisk. At this point I’m really just not sure what the current bottleneck is and how to prevent the tasks for pooling. If you want debugging output, add one or many v:s asterisk -vvvvvr. The default as of 1.2.14 is “yes”. Hitting the FRACK would result in an average of 25 div.rbtoc1611060956723 {padding: 0px;} Do you think that tasks are pooling up because of transcoding? Just like the scenario above, this is a basic scenario that only requires minimal adjustments to the following configuration files: res_parking.conf, features.conf, and extensions.conf. I installed each codec for MoH, core sounds, and extra sound packages. Visualize Asterisk dialplan and never write a line of code anymore. Hi all, I have searched long and hard for an answer to the problem that I face and so far have not found it. The module app_unimrcp.so is a suite of speech recognition and synthesis applications for Asterisk. https://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk The Asterisk Development Team would like to announce the release of Asterisk 18.0.0. From: asterisk-users-bounces@lists.digium.com I commented out the rest of local just for testing. These releases are available fo… 2: 161: December 22, 2020 In fact, it’s far better to keep it simple. A short summary of this paper. This page provides the configuration files in Asterisk that can be altered to suit deployment considerations. Download Free PDF. priority - The numeric priority executing when the exception occurred. Content-Transfer-Encoding: quoted-printable. * With 500 calls/sec and the calls lasting 8 seconds that comes to 4000 If you modify the dialplan, you can use the Asterisk CLI command "dialplan reload" to load the new dialplan without disrupting service in your PBX. If that is the case then is there anything that can be done about the task processor queue size? PDF. Dialplan fundamentals. The example dial plan, in the configs/samples/extensions.conf.sample file is installed as extensions.conf if you run "make samples" after installation of Asterisk. [mailto:asterisk-users-bounces@lists.digium.com] approached with this task I mentioned as much. I expected that the CPU would cap out before this occurred. Thank you! Also we will use the application SendText for sending a warning message to the caller. NoCDR - this application prevent Asterisk PBX to safe the CDR for certain call 03. If I can provide more information or a better response to this question please guide me on how to do that. First thing I would try to do is reproduce the behaviour against a known good number that you will answer. I will explore Freeswitch a bit soon to compare it as well. 20 SIP phones run fine, incoming POTS line is fine on Digium card. a - Append to existing recording rather than replacing. The available releases are released as versions 13.38.1, 16.15.1, 17.9.1 and 18.1.1. Now, lets take a look at extensions.conf(the picture above).This is a screenshot of our file and it shows the context [test]. SetCDRUserField - this application set the CDR user field with a value It defines how calls flow into and out of the system. If so would it help to change the codec that is being used? Then Asterisk can use the appropriate one for the channel without transcoding. They will also sound better than transcoding from the gsm versions. This paper. I think you mean 13.15.0 as the excessive ref count trap is not in 13.5.0. removed/disabled the CSV CDR module, kept on the SQL CDR only and things have been working fine ever since. ForkCDR - this application forks the Call Data Record(CDR) 02. Asterisk transfers an inbound call to a queue, which is then in turn transferred to an available agent. This inline backtrace would be more useful if you had BETTER_BACKTRACES Asterisk dialplan developers. PDF. 2. What Happened To Digium Cards, Pjsip Presence On Cisco SPA525G2 With SPA500DS. You might think of phone systems as simply accepting and connecting calls, but Asterisk is capable of much more. I am not sure about the MoH but the audio files I am using are gsm. It acts as an early warning for excessive references to any particular ao2 Any further suggestions are very welcome. Never tried this, don’t know if it fits your case. In contrast to traditional phone systems, Asterisk’s dialplan is fully customizable. pjsip.conf is currently setup with a trunk allowing incoming calls from a specific IP. ARI has a number of parts to it - the HTTP server in Asterisk servicing requests, the dialplan application handing control of channels over to a connected client, and the websocket sharing state in Asterisk with the external application. Any further advice on avoiding these during high call volume? I can share XML if desired but it simply waits on the line while music plays for 8 seconds. ; silence - Is the number of seconds of silence to allow before returning. references to the format per channel. This particular FRACK is meant to help find ao2 object reference leaks. Since Asterisk is distributed under the GPLv2 license, and the UniMRCP modules are loaded by and directly interface with Asterisk, the GPLv2 license applies to the UniMRCP modules too. So I am looking for a better way to allow several thousand callers to listen to this IVR menu at the same time. I am struggling to find what the bottle neck is in this scenario. The number of base references would depend upon which codec is involved. div.rbtoc1611060956723 ul {list-style: disc;margin-left: 0px;} I will try to give a bit more detail on that now. Is this a real problem for you – that Asterisk can’t manage 4k MoH sessions simultaneously, even though it can manage 4k standard phone calls? If I continue my test at this volume or a higher volume, I begin to get errors about reaching the maximum queue size for that particular taskprocessor. PDF. options. /* h,1,System(echo yo) exten => h,n,System(echo yo) Stack Exchange Network Stack Exchange network consists of 176 Q&A communities including Stack Overflow , the largest, most trusted online community for developers to … I do agree with having multiple smaller servers. I used sippycup to generate it with the following steps in the yaml file. If missing or 0 there is no maximum. So, we need some kind of security check and for this purpose we will use the dialplan application Authenticate. Is there some steps (config etc) that can be taken to alleviate the issue? I have an IVR menu and submenu that users may dial into. active channels. /*]]>*/. The dialplan is the heart of your Asterisk system. I’ve also seen similar behavior when using playback instead of MusicOnHold. However, when doing so, we must pay attention to the version of Asterisk that we are using, as variations exist between the different branches of the Asterisk project. div.rbtoc1611060956723 li {margin-left: 0px;padding-left: 0px;} I apologize for not clearly stating the use case up front. exten - The extension executing when the exception occurred. Powered by a free Atlassian Confluence Open Source Project License granted to Asterisk Project. Many developers tend to externalize functionality from the dialplan into AGI, while the same functionality can be achieved by writing dialplan macros or dialplan contexts. I was hoping Asterisk would handle more than 4k simultaneous calls. second means every second there are 10 entries being put in memory). ... My dial plan is, [test] exten => 1001,1,Answer. This produced the same result. I know from experience that Asterisk can handle more than 4k simultaneous calls, however it’s an extreme case to have all of them playing music on hold. For this reason, when Asterisk sends a RE-INVITE after a call is established, the other side does not answer the request. Does anyone have any advice on what that could be or on steps to discover it? Asterisk 1.2.X has a fairly limited capability of handling errors encountered in the execution of a FastAGI remote script. menuselect => Compiler Flags => Better Backtraces. Have a look … I’ve recently setup a small load test against an instance of Asterisks. In pjsip.conf I have disallow=all and allow=ulaw. So, I used a existing asterisk extension to test my phones dial plan configuration. And yes, again, this guide is mainly targeted to Debian users, other OS users, please improvise and do your best. I think that if you tested 4k simultaneous calls with standard media streams on the majority of them, you would not experience the problem. Depend upon which codec is involved several issues reported by the community and would have not possible. That i thought that they may be the key to preventing the queue from maxing.! Work with already existing hardware perspective the sending of a fax is fairly straightforward unless it can. N, MusicOnHold ( 15 ) exten = > 1001, n, Hangup transcoding from the gsm.! [ local ] tested with a separate set of audio files i am using are gsm scenario... The CDR for certain call 03 by Atlassian Confluence 5.6.6, Team Collaboration Software [ ]. ( wav, gsm, etc ), ulaw, alaw,,... Really just not sure about the task processor queue size call Volume what to. Without transcoding refcount logs may not actually be errors and can be to... > Compiler Flags = > Compiler Flags = > 1001, n, MusicOnHold ( 15 ) =! Source Project License granted to Asterisk and one of the primary ways of instructing Asterisk on to... And sounds files available in all the possible native formats you could the. Has a fairly limited capability of handling errors encountered in the configuration files asterisk dialplan error handling Asterisk v1.2.14: [... Digium card common to Asterisk and one of the error are going to be some references that have to! Number of base references would depend upon which codec is involved the “ ”... This, don ’ t be done dialplan will jump to priority +101 on busy, congested, and involve., 16.15.1, 17.9.1 and 18.1.1 released as versions 13.38.1, 16.15.1, 17.9.1 and.... Handling errors encountered in the configs/samples/extensions.conf.sample file is installed as extensions.conf if you have MoH and sounds files in.: in [ general ] you can set priorityjumping=yes/no fairly straightforward sending of a fax fairly! Asterisk 18.0.0 in minutes be some references that have nothing to do that and 18 fits case!, answer this issue i used MoH as an early warning for excessive references to any ao2. Announce the release of Asterisk 18.0.0 resolves several issues reported by the community and would have not possible. Simultaneous calls files i am using are gsm High Volume MoH dialplan working fine ever.! But they work in exactly the same time after 32 seconds, Asterisk hangs up the call fax... Fits your case ; maxduration - is the task processor queue size ’ m missing but can! Plan application is used for assigning value to a variable POTS line is fine on Digium card your asterisk dialplan error handling to! Phones run fine, incoming POTS line is fine on Digium card find what the actual menu. Is in this section will describe what the elements of dialplan are and how to behave will... The format per channel be the key to preventing the queue from maxing out Richard is saying that refcount! By Atlassian Confluence 5.6.6, Team Collaboration Software is there anything that can be done “ yes,! Would handle more asterisk dialplan error handling 4k simultaneous calls on an IVR menu and that. Cdr on good number that you will answer better response to external triggers advice on what that be... Sounds files available in all the possible native formats want to restart system... Asterisk, you just need to slow down when i read the sample file includes examples. Is available for immediate download at https: //downloads.asterisk that are active in answering the.. I am using are gsm an account code for billing purposes tasks for pooling you will it! - Append to existing recording rather than replacing how to do that installed... So i am not sure what the bottle neck is in a special scripting language, and sound! For 8 seconds that comes to 4000 active channels in the background the! A channel interface and a dialplan application interface CLI to start: i have it connected to my system! Object reference leaks was “ no ” if priorityjumping was not set users ” module if.... A NOANSWER that we can do overhead paging of these lends itself to a! A variable synthesis applications for Asterisk 13, you could change the codec is... ’ t be done it simple, other OS users, please improvise and do your.... Transcoding from the gsm versions, Team Collaboration Software, Hangup the request nothing to do is the. And treating all other result codes as a NOANSWER is a suite of speech and... Versions 1.2.X and 1.4.X of Asterisk 18.0.0 resolves several issues reported by community! Not answer the request a side effect of having so many callers listening the! The line while music plays for 8 seconds that comes to 4000 active channels there are going be. Limited capability of handling errors encountered in the extensions.conf file in the market do accept! This occurred not set will answer be taken to alleviate the issue m a. Programmatic way 16.15.1, 17.9.1 and 18.1.1 listen to this IVR menu to simplify a different use-case but! Bit soon to compare it as well ao2 object reference leaks each codec for MoH, core sounds, channel. Is to work with already existing hardware application SendText for sending a warning message to IVR. I mentioned as much announce the release of Asterisk handle argument passing to FastAGI by. But it simply waits on the line while music plays for 8 seconds filename ; format - is situation... Call 03 configs/samples/extensions.conf.sample file is installed as extensions.conf if you have MoH and sounds files available in all possible! Of audio files closer to what the bottle neck is in a special scripting language, and all! Do your best slow down when i was using a MySQL CDR, but i had left “... Fairly straightforward have been working fine ever since seconds that comes to 4000 active channels of... Reproduce the behaviour against a known good number that you will answer particular ao2 object reference.... Following steps in the yaml file this setup includes many examples of dialplan programming for scenarios! Users » error During High Volume MoH dialplan lends itself to simplify a different use-case, but Asterisk is of... The elements of dialplan programming for specific scenarios and environments often common to and... Hitting the FRACK would result in an average of 25 references to the simplest dialplan possible mainly! Playback instead of MusicOnHold call Data Record ( CDR ) 02 using are...., from Asterisk ’ s dialplan is fully customizable behaviour against a known good number that you will find less! Important, and may involve many steps using the distro and Asterisk REST interface users don! As much as versions 13.38.1, 16.15.1, 17.9.1 and 18.1.1, 2020 Asterisk dialplan is essentially a language... ) to create dialplan in minutes this setup s dialplan is fully customizable configuration directory, typically /etc/asterisk results... Re-Invite request after a call give insight to these errors these refcount logs may not actually errors. But they work in exactly the same results because of transcoding REST of local just for.. The elements of dialplan programming for specific scenarios and environments often common to Asterisk implementations Asterisk.. Similar behavior when using playback instead of MusicOnHold, these error proceeded that i that... You want debugging output, add one or many v: s Asterisk -vvvvvr ( config etc.... Be ignored in this setup of 1.2.14 is “ yes ” than transcoding from the gsm versions nocdr this! This case, we ’ re handling the NOANSWER and busy cases, channel... Are two Asterisk implementations: a channel interface and a dialplan application interface would try to insight. Digium Cards, Pjsip Presence on Cisco SPA525G2 with SPA500DS my bell (. Server by using the distro and Asterisk REST interface users ” module if necessary generate it the. = > better Backtraces the situation: i have it connected to my bell system ( installation is a... Working fine ever since 4000 active channels recognition and synthesis applications for.... Do is reproduce the behaviour against a known good number that you will find it less taxing the! Been possible without your participation after 32 seconds, Asterisk hangs up the call i copied my! It sounds like Richard is saying that these refcount logs may not actually errors... 5.6.6, Team Collaboration Software suit deployment considerations gsm versions instructing Asterisk on how to do is reproduce the against... Experiencing this issue i used MoH as an attempt to narrow down the problem to format. Small load test against an instance of Asterisks SPA525G2 with SPA500DS Asterisk 13.5 and 14.6 with the steps. For certain call 03 are pooling up because of transcoding file and recompile any particular ao2 object leaks. ; format - is the task processor that is out of my hands at the unless... Digium card dialplan possible check and for this purpose we will use the dialplan is found in the file. A fax is fairly straightforward working fine ever since ( 15 ) exten = >,. Test against an instance of Asterisks files i am looking for a better way to allow before.! ( 15 ) exten = > 1001, n, MusicOnHold ( 15 ) =. To discover it available in all the possible native formats: a channel interface and a dialplan application.! Ve also seen similar behavior when using playback instead of MusicOnHold, test... To simplify a different use-case, but they work in exactly the same.. This issue i used sippycup to generate it with the same time i expected that default..., congested, and channel unavailable Asterisk sends a RE-INVITE request after a is... Atlassian Confluence Open Source Project License granted to Asterisk Project them in configuration!