similar to: Asterisk not matching longest prefix with include

On Tue, Jun 26, 2018 at 7:06 PM, Doug Lytle <support at drdos.info> wrote: > On 06/26/2018 06:57 PM, Dovid Bender wrote: > >> Hi, >> >> My dialplan looks like this: >> [from-external] >> >> Exten => _X.,1,Noop(CALL IS COMING INTO FROM EXTERNAL CONTEXT) >> Exten => _X.,n,Noop(IP: ${CHANNEL(recvip)}) >> Exten => _X.,n,Noop(CALLED

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 > >

Hello, How can I get peerip, recvip, from, uri, useragent, peername, t38passthrough variables in (within) my custom Asterisk application? I can't use chan_sip.c internal structures (such as sip_pvt) in my custom application, because there's no chan_sip.h and I can't include it into my application (maybe there's other way?). I can do like this: exten =>

I am currently suffering various SIP attacks. I am using the following extension to record the caller's IP address: exten => h,n,set(CDR(srcip)=${CHANNEL(recvip)}) However, in recent attacks, this IP address is not correct, and I believe that they are spoofing it. I am using asterisk 1.6.2.15. Does the CHANNEL(recvip) variable record IP show in the SIP header instead of the real, UDP

Hi there; I didn't see any "G" option in the example above, and the usage for the option parameters is entirely undocumented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial The G options are as below G - If the call is answered, transfer the calling party to the specified priority and the called party to the specified priority plus one. context exten

Hi, I have the problem that WaitExten is not responding to key presses. Here are the sections from my extensions.conf: [globals] incoming_call=0 menu=0 announce=0 [internal] exten => 777,1,Goto(hotline,${EXTEN},1) [hotline] exten => _X.,1,Set(CALLERID(name)=Hotline) exten => _X.,n,Set(original_extension=${EXTEN}) exten => _X.,n,GotoIf($[${announce}=1]?4:10) exten =>

Hello We had this situation: Some bot-net did try to guess SIP logins and finally succeeded. The Asterisk Server was abused to call a large number of expensive destinations. It is clear that the sip logins have been passed to various persons (probably posted on a forum somewhere inviting to do 'free calls'). Right after the affected password was changed, the message log shows which

Hi, I am facing this issue, where I get a delay of aroud five seconds when switching between contexts (through extension.conf) . This is how my extensions looks like. [salesivr] exten => _X.,1,NoOp(Incoming call from user ${EXTEN} and caller id ${CALLERID}) exten => _X.,2,Playback(emptyy) exten => _X.,3,Background(Main_Sales) exten => _X.,4,WaitExten(2) exten => _X.,5,Goto(_X.,3)

I'd like multiple people to be able to dial in and listen to various live radio streams. I was told that the correct resource-friendly way would be to setup a MoH class, and then select that from the dialplan. This works well, but how do I switch between streams? Someone correct me if I'm wrong, but from previous similar questions a few years ago it seems like once you've entered a

Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension

I have a Asterisk PBX System with Redhat Linux Fedora 4, Webmin version 1.400 and I am simply trying to configure into the "Extensions.conf" script an entry that will add to the "Auto-Attendant" a line that will allow a "Caller" to enter a "0" (Zero) will then ring the extension(s) of the "Operator" to speak directly with the "OPERATOR"

Hi everyone! I've set up asterisk ip-pbx to implement IVR menu and encountered such a problem: when users dial the destinaion phone number and end it up with "#" asterisk still waits until timeout in WaitExten() is reached. // Here comes the context where user is prompted for a dest. number: context ivr-dialout { s => { Background(enter-your-dest-number);

You can use the following: switch3*CLI> show function SIPCHANINFO switch3*CLI> -= Info about function 'SIPCHANINFO' =- [Syntax] SIPCHANINFO(item) [Synopsis] Gets the specified SIP parameter from the current channel [Description] Valid items are: - peerip The IP address of the peer. - recvip The source IP address of the peer. - from

Thanks Joshua and everyone, Joshua's solution seems a lot simpler and works well. Only one thing now - The reason I named the classes as I did, was so that I could select the class based on callerID plus extension. Unless I've misread it, I'm limited to 9 switchable classes via the "digit=#" option, is that correct? Or is there a clever hack around this? extensions.conf

I cannot find in Asterisk 12, the channel variable ${CHANNEL(recvip)}, so if I use PJSIP, for scalability, how do I read what the signalling IP where the inbound call is coming from and what is the inbound codec? You would think that the new channel would set those variables up, isn't it? Philip Orleans

A few months ago I started using and had to abandon PJSIP because my dialplan could not read the inbound signalling IP address, which I can read now in Asterisk11 using CHANNEL(recvip). My app relies on this information. The question is, is it possible now access the signalling IP of an incoming SIP call using PJSIP? Philip

Hi, I am upgrading from Asterisk 1.4 to 12.4. I am able to authenticate the user and call AgentLogin. But after that when I call AgentRequest I keep getting Agent '1234' is busy. If I put a delay of 5 second or more before calling AgentRequest then it works most of the times. Here's my dialplan: [login] exten => s,1,Background(thank-you-for-calling) same =>

Is there any known reason that the DISA() routine should behave differently than WaitExten() as far as recognizing DTMF tones? If not, I suspect there's a bug here. Try it yourself--two DID's on our PRI, numbers below let you test each routine: It is my observation that some setups/phones DO and some DO NOT express this variance. --I could not show any variance on a sprint mobile phone

I have the following senerio. Call file calls 1st party. When connected give called party option to connect to second party. Issue Dial to second party. Caller answers and the two are bridged together. My issue is that 4 out of 5 calls fail to bridge the audio. Am I missing something or is there some kind of bug? Here is my test dialplan ;Dialer Base Code Files. ;Variables

hello list i have an IVR menu in asterisk 1.4 like below exten => 600,1,Ringing() exten => 600,n,Wait(2) exten => 600,n,Goto(home,s,1) [home] exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/) exten => s,n,Background(${sounds_path}music1) exten => s,n,Background(${sounds_path}music2) exten => s,n,Background(${sounds_path}music3) exten =>