similar to: dialplan reload context

Hello how can I set the fromuser field of the SIP INVITE when using the Dial()-command in the dialplan ? None of the below Dial() command give the correct result : exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762 at myprovider.biz) exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762 at myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user762 at

Thanks Scott. I was able to get the basic concept to run. However, it seems PJSIP INVITE for the Dial also does not support added headers. The Local channel dial plan did have the channel variable values. I added them as SIP headers, then Dial(PJSIP/Agent). The INVITE for the Dial on PJSIP continues to not include the SIP Headers I added. For chan_sip, I have no problem with this. Even the

Hello, I have this in my dialplan : exten => s,n,Set(vgLabel=vg(${number}+1)) exten => s,n,GoTo(${vgLabel}) But in stead of vgLabel becoming the SUM of 2 numbers, it is just a string : [Nov 3 16:17:27] -- Executing [s at macro-f:43] Set("SIP/test-00000002", "vgLabel=vg(1+1)") in new stack [Nov 3 16:17:27] -- Executing [s at macro-f:44]

Thanks Scott. I?m taking over for someone else?s code, so I must admit I?m still learning the Agent and Queue concepts. Local channels are something I have not used either. Would local channels essentially be an internal bridge? How would I ?Register Local/number at agent in the queue on behalf of the agent (replace number with the agent's extension number)? From: asterisk-users-bounces

Hello list, using Asterisk 1.4.30. I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. This is the dialplan : exten => 10,1,NoOp() exten => 10,n,SIPAddHeader(Remote-Party-ID: "eric" <sip:10 at 192.168.1.150>;party=called ) exten => 10,n,Dial(SIP/test2) This is what the CLI shows : /[Jul 12

Nick- Are you wanting to recreate the dialplan Echo() application in stasis? Why not just send the call to Echo() instead of Stasis()? On Fri, May 22, 2015 at 11:25 AM, Matthew Jordan <mjordan at digium.com> wrote: > On Fri, May 22, 2015 at 4:41 AM, Nick Awesome <jleed at me.com> wrote: > > Can anyone tell me how can I create echo test using ARI stasis > application?

Hello I have 2 strange errors when using the Background()-application and DTMF-input that is received. First of all, my first 2 lines are not being executed. The first line being executed is the Set() application, thus line 3. Secondly, the received digits (911) is not the same as the EXTEN (which is set to 91). exten => ivr,n,Set(TIMEOUT(digit)=2) exten =>

I'm pretty sure there isn't a way to do that currently. ?My best guess would be that a new special type of bridge technology could be created that would implement the per-channel echo (no audio bridged between channels in the bridge). That would require new C code in Asterisk for the bridge, and then the usual methods of moving channels in to bridges with ARI could be used.? On Sat, May

The Cisco/Linksys SPA devices are also able to be provisioned automatically. On Wed, Apr 15, 2015 at 3:20 PM, Bryant Zimmerman <BryantZ at zktech.com> wrote: > Alejandro > > All of the Grandstream devices can be remote provisioned if you know what > you are doing. > > Bryant > > ------------------------------ > *From*: "Alejandro" <cdgraff at

Hello i have the following field (text string) in a MySQL database : "${KNUMMER} ${phone_number_to} ${phone_number_from} ${CHANNEL:4}" I read this string form the database and want to have the dialplan variables to be replaced with the correct content. How can I do this ? Currently this is not working. The variable ${PARAMS} contains the exact string of the database field : my

Okay, Scott, I think we are on the wrong path. Maybe I'm wrong though. I will summarize again briefly the problems together: The peer ip address could be another than the ip address of incoming invites After an re-register the REGISTER is send to the new SIP server, answered with OK. But the peer ip address is still the old one (sip show peers). If now is a INVITE, the request is answered

Scott, I have changed the configuration as said it and will test it. I?m curious. Can you briefly explain what insecure=invite,port does? ;insecure=port ; Allow matching of peer by IP address without ; matching port number ;insecure=invite ; Do not require authentication of incoming INVITEs ;insecure=port,invite ; (both) Do I understand correctly that

On Thu, 28 May 2015 11:15:45 -0500 Scott Griepentrog <sgriepentrog at digium.com> wrote: > The string "5a2600300339934f704528bb14ed05e9 at MyAsterisk:5060" is the unique > identifier for the call in SIP known as the Call-ID. If you have a packet > capture of the port 5060 SIP traffic, that identifier will be in each SIP > message related to the call, which also

Is it possible to make use of queues for incoming calls but to have agents that do not need to log in ? If I create a queue and make certain SIP-users member of the queue, do these SIP-users always need to log in to the queue to be able to receive calls that are in the queue ?? Can't a member be just available when the phone is registered to the Asterisk-server ? In stead of also having to

Howdy, I built an LXC container with an "image" of asterisk 11.18 precompiled and installed. It runs fine on the dev platform, which is a Dell R320 running Ubuntu 14.04LTS. I shutdown the container, tarred it up, and untarred on a Dell PE1850, also running Ubuntu 14.04LTS. The container itself is Ubuntu 14.04LTS. Both platforms as far as I know are amd64. The container boots

Hi All I have a few lines like this at asterisk/messages. [May 25 15:22:42] WARNING[27725] chan_sip.c: Hanging up call 5a2600300339934f704528bb14ed05e9 at MyAsterisk:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). Since we have hundreds of clients with hundreds of simultaneous calls, how is it possible to know to which customer/IP

I have a call coming in. I need to add a SIP Header to the channel. Then, I need to send the call to the Queue so it is sent to the Agent. The SIP header I added, I need to have appear in the INVITE sent to the Agent. It works in chan_sip. I send the call to a macro which does... n,SIPAddHeader(My-Header-Name:${MY-HEADER-VALUE}) n,Queue(${ARG2}) In PJSIP , this doesn't seem to work. Is

Anyone found any good smartphone apps that connect with their asterisk boxes that provides basic mobility features? The main problem we are trying to solve is when our staff forward to their cell phones they cant distinguish if the call was directed at their cell phone or the business DID. We also would like to give user ability to control DND and forwarding of their extension from the

Hello, I'm trunking with an ITSP that, when treating an outbound to an unknown destination, either: - send a SIP error code (I can't be more explicit, at the moment), - or cast a pre-recorded audio message using Early Media. At the same time, I'm also trunking with Contact Center solution which doesn't support Early Media. Beside asking my ITSP to treat calls consistently or

Ok, I have tested dnsmgr. This is not a solution, the situation has not changed. With dnsmgr I can not place outbound calls. I do not know why and what dnsmgr really do. My current solution is as follows: Say allowguest=yes, configure the default context that there can not be placed outbound calls. Use iptables to DROP all at your SIP port and allow only your local phones and the sip trunk ip