similar to: is there some blocking in 11.21.0

>Are you saying that this worked in earlier versions but you started to >get the delay when you updated to 11.21.0? Or just that you happened to >be using 11.21.0 the first time you tried this scenario? I should have said "first time" trying this. Any thoughts? Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am running a heartbeat... Asterisk 11.15.0 - same behaviour is noticed on 1.4.43 also I issue a call through the API that does the below. just UserEvent and Hangup -- Executing [s at heartbeat:1] UserEvent("Local/s at heartbeat-0000000f;2", "HeartBeat, Noop") in new stack -- Executing [s at heartbeat:2] Hangup("Local/s at heartbeat-0000000f;2",

Dear, I'm currently using a Dial command with multiple destinations and channels eg: Dial(SIP/100&SIP/101) I simply would like to know, in real time during the call (from dial plan or AGI), who has picked up the call. Can I find this information in a variable somewhere ? Thank you for your help Patrick

The Asterisk Development Team has announced the release of Asterisk 11.21.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.21.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

I am using asterisk 1.4.41 and the AMI I am trying to execute a command over AMI, specifically "core show channels concise" "sometimes" I get this back: asterisk_command_show_channels() execute failed. 'Response: Follows[CR ][LF ]Privilege: Command[CR ][LF ]OutgoingSpoolFailed!smvoice-dialout!failed!1!Down!AGI!smvoice|-digium_failed!!!3!0!(None)[LF ]' I'm not

I am using AMI to call a phone and play a wave file. That works fine to SIP/401. Now I am trying to "redirect" that call that is ringing to another phone (SIP/404). When I do it the other phone rings but the first phone continues to ring also. Then when I answer on SIP/404, I get a ring not the wave file. Action: Setvar Channel: SIP/401-00000004 Variable: SMVOICE_CALLAT

Hey guys, I've got a part of my dialplan that dials multiple people: exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME}) Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g. SIP/100&SIP/101&SIP/105 etc This works great, however I want to see if I can find a way to work out (and run an AGI script) when the call is picked up by someone. Thanks all!

All: Some of the people who dial into to our system will press the pound key when entering an extension for the directory key. When waitexten gets that, I get an error messages as, for example 123# doesn't match any extension. I was going to use ${EXTEN} to just use the first three numbers, but I'm not sure how to use this with WaitExten. so I have exten =>

Hey, I have a few dids with callcentric. They seem to work fine most of the time but at some points I get "handle_request_invite: Failed to authenticate user <sip:PSTNnumber" This happens intermittently. The way I understand it the insecure=port,invite should tell asterisk not to authenticate users coming from that host. But its not working for some reason. This is my sip.conf

I am writing a dialplan context under asterisk 11.21.0 to handle SIP message routing between registered SIP peers using chan_sip. I am having trouble with double-quotes when the source peer uses a display name, which appears in quotes before the SIP URI. I want to extract the SIP URI from MESSAGE(from) in order to (conditionally) route a failure message back to the source peer. My test dialplan

I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root at elx4 ~]#

Has anyone created like a broadcast dialplan, if so care to share it. What I'd like to do is create an extension so when someone calls that extension they can leave a voicemail. Right after it is recorded, I need that voicemail to played on all phones on that system... E.g.: 1) Administrator --> Dial special number 2) Record emergency message (e.g. Snow day don't come in) 3) Hang up 4)

I am trying to block/hide outgoing caller id on a PRI E1. It seems that it should be fairly simple, but it is defeating me. https://wiki.asterisk.org/wiki/display/AST/Function_CALLERID says: "to hide your caller id, use: Set(CALLERID(num-pres)=prohib)" That doesn't seem to do it. The calls are originated from AMI and I have tried a blank "CallerId:" line and

Dear all, I want to make call through socket i have set code given below: #!/usr/bin/perl -w use IO::Socket::INET; sub asterisk_command () { # my $command=$_[0]; my $ami=IO::Socket::INET->new(PeerAddr=>'127.0.0.1',PeerPort=>5038,Proto=>'tcp') or die "failed to connect to AMI!"; print $ami "Action: Login\r\nUsername:

On Thu, Jan 21, 2016 at 4:04 PM, Craig Topper <craig.topper at gmail.com> wrote: > Isn't it also marked 'final' so it can't be inherited from anyway? What's > your need to inherit from it? > -- > ~Craig > Oops, missed the final part. I need to change the alignment. For my target it is sort of independent of the data layout. I was going to overwrite void

Does anybody know if there is a particular reason why MCAsmStreamer doesn't have its own .h file? https://github.com/llvm-mirror/llvm/blob/0e66a5f53c74056f95d178c86531d7d9cfb23da9/lib/MC/MCAsmStreamer.cpp#L41 It seems like it is a good idea to have this class declared as its own module ( its own .cpp and .h files). That would make it easier to inherit from it if there is a need (like in my

I am getting this: make gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pritest.o -MF .pritest.o.d -MP -c -o pritest.o pritest.c pritest.c:50:28: fatal error: dahdi/tonezone.h: No such file or directory #include <dahdi/tonezone.h> I am using dahdi-linux-complete-2.11.0+2.11.0 I did "find . | grep timezone.h" from the dahdi root and its not

Hi All, What is the correct way to provide a CentOS 7 - WMware image for ESX ? As an amateur to VMware - I thought - great I can get VMplayer and ESX should be able to import my image... Wrong... I even went through the trouble of "converting" to VMWare workstation and thinking ESX could import that - Apparently still Wrong... I cannot for the life of me understand how one product

Hi Can anyone with distinctive ring on their 7960's possibly post how they've got it to work? I understand that the ALERT_INFO variable is involved but using the examples for the variable value from the WiKi I'm just getting an error message from the Asterisk concole. Thanks in advance. P

I am using Xen 4.6.0-8.el7 on CentOS 7 with 3 Linux DomUs. I did have Xen 4.? on CentOS 6.5 on my previous system. I installed using yum install centos-release-xen. On shutting down the Hypervisor it used to wait for the DomUs to shut down gracefully (one can take a minute or more), then on starting up again it would restart the DomUs, all nice and controlled. For some reason the DomUs (CentOS