similar to: Controlling Asterisk from within the dialplan

Hi all, we have just upgraded from Asterisk 11 to Asterisk 16. After porting all the config to 16 we figured out some major load problems. the majority running of our Asterisk instances is still having Asterisk 11 so we can compare load handling on both versions. On the same hardware configuration we see load differences that Asterisk 16 takes four times the load as Asterisk 11 (on 11 we see

Anybody having issues with ztdummy under the current 2.6 RC7? I get the following errors when trying to modprobe ztdummy: "Unable to register zaptel rtc driver" Doing a Google on the error shows reference to a message from 2004 that said you might not have RTC compiled into the kernel. Checking via: cd /usr/src/linux-2.6.13-rc7 grep -i rtc .config shows: CONFIG_APM_RTC_IS_GMT=y

Can you say a Virtualized Asterisk with a PRI card! http://www.phoronix.com/scan.php?page=news_item&px=OTY0OQ Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

Is there a way to limit outbound calls when feeding files to the outgoing directory in asterisk? I several thousand files i need to feed asterisk, hoping to copy it to the outgoing directory all at 1 time.

Hi, @josh, We are using AGI. It is a very simple perl script. If we need to move to FastAGI we most likely would port it to ARI instead. But I let you know. @john, we using Perl. To see if it is a problem with the perl i had put an "exit 0" just at the first lines so there is no logic done at the AGI. It's only the start up and return from AGI what produces the most of the load.

Jon, We are only using FastAgi. On the second system (running Asterisk 16) there are no agi's running (just some bash scripts on call hangup). I did add some hackey code (netstat -nua | grep -v 'udp 0 0' | grep -v udp6 | grep -v ' 0 0.0.0.0' | grep udp) to my bash script to check out the packet queue (with the help of

I am trying to connect two machines to each other with an T1 crossover cable. The first machine has two TE120P cards - one connecting to the telco on an ISDN PRI. The second to a crossover T1 cable to a second machine which has one TE120P card. Telco <-cA-> Machine1 <-cB-> Machine2 Machine1: Two TE120P cards Machine2: One TE120P card cA: Standard T1 Cable cB: Crossover T1

We have 2 PRI from AT&T And all is well but only few numbers having following issue. We are getting hangup cause 18 do you guys have any idea ? We have just migrate 1.2 to 1.8 and this issue raised [Jun 7 17:57:10] VERBOSE[23717] sig_pri.c: -- Span 2: Channel 0/3 got hangup request, cause 18 [Jun 7 17:57:10] DEBUG[24856] sig_pri.c: Not yet hungup... Calling hangup once with icause,

On Tue, Jun 26, 2018 at 7:28 PM, Doug Lytle <support at drdos.info> wrote: > On 06/26/2018 07:20 PM, Dovid Bender wrote: > > Doug, > > I tried that as well. Even with my dialplan looking like this: > > > > Ordering by includes works for me under Asterisk 11 and 13 > > What does the output of the below show? > > dialplan show from-external > >

> exten => _13XXXXXXX,1,dial(${OPERATOR},20) Hello "SIP/2.0 401 Unauthorized" Unfortunately the negative. An asterisk indicates a 404 error. On Tue, Mar 5, 2019 at 12:51 PM Doug Lytle <support at drdos.info> wrote: > > On 3/5/19 2:46 AM, Gokan Atmaca wrote: > > Asterisk can send calls, but I don't get a call. What could be the problem? > > >

We currently have an Asterisk system that is hooked up to our old paging speakers via sound card, plugged into two amps. Each amp drives up to 8 analog speakers in each warehouse (we have 2). Both warehouses are around 30k square feet. Both have a large number of printing presses. The computer system is that is running Asterisk is around 10 years old and starting to fail. I'm looking to

Hey everybody, I've, within the last 3 weeks, moved over to a PRI from SBC/AT&T. I've received several complaints about dropped calls. Reviewing the archives on PRI and dropped calls shows that I should set the resetinterval=never in the zapata.conf and restart. This hasn't helped. The dropped calls have to date only been on outbound calls. Usually within 2 to 3 minutes

Hi All, I hope someone can give me a hint. We try to reload the asterisk dialplan config using ansible command module. Using this we just trigger asterisk -rx "dialplan reload" Now we want ansibe to fail if there is something wrong in the dialplan. If we put a bad config in extensions.conf dialplan reload prints some warning in Asterisk console but the command "asterisk

Hello, I am using the voicemail module of asterisk. When I did some test calls from my mobile phone, sometimes the beginning of the prompt was missing, e.g. instead of something like "number 12345 not available" I was only hearing "345 not available". Verbose level 5 on the asterisk console didn't give me any hint on this, it only shows that playback of the prompt started

Hey all I have been beating on this all weekend long. Any feed back would be appreciated. We stood up a 11.6 system. We tested everything we could think of. We moved over to it and all seemed to be working good than a customer told us that they were not hearing our vociemail greetings. When we call into the system and it drops to voicemail we just get a beep no greeting played. We checked

> Show us the parts of your dial plan which: > > a) work for recording incoming / outgoing calls > > b) do not work for recording internal calls > > then we might be able to give you a clue what's wrong. Hello For example: My phone number is 1000, the other's number is 1001. These numbers are in the same PBX (asterisk). I want 1000, 1001 can call to record audio

Since putting my Tellabs EC into place around 2 weeks ago, the echo problem has almost been eliminated. Reports of some very faint echo, but everybody is happy. My question is, if I were to also turn on the Asterisk Software EC, would this remove any residual echo that may make it past the Tellabs Hardware EC. Thanks, Doug

On Tuesday 25 June 2019 at 16:49:23, Doug Lytle wrote: > We have Polycom phones (I'm using a VVX601, the destination is a VVX301). > We're also on Asterisk 13. > > I forwarded my call to the VVX301 and then dialed my phones DID. Surely that is "call forwarding", which is quite different from either a blind or attended transfer? A transfer involves a call coming

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I've moved 1 of our facilities over from 1.2 to 1.4 two weeks back. So far, the only issue that I've encounted is. I have a scheduled CRON job that runs at 3am every Sunday, that issues a: asterisk -rx 'restart when convenient' The first Sunday that it ran, Asterisk never restarted. The CRON logs show that it issued the command successfully. This Sunday, it ran but never