Re: [asterisk-users] Call does not go to voicemail

Tim, I've tested similar dialplan on my home-server and it works perfectly. (same setting, slightly different extensions) but same idea: exten => 418,1,Dial(SIP/55,15,trw) exten => 418,n,GotoIf($["${DIALSTATUS}"="BUSY"]?vmail:line2) exten => 418,n(line2),Dial(SIP/218,15,rw) exten =>

Re: [asterisk-users] Call does not go voicemail

The way you have the GotoIf is making it so that no matter what the busy condition of the line, it will execute the next line in the dial plan. What you'd need is an "if" or "then" which goes to a tagged line in the dial plan. How it reads now is: "If [busy] then line2, else execute next line".

[asterisk-users] Some questions regarding jitterbuffer in asterisk / pjsip

Hello! I just implemented a jitterbuffer for pjsip in the dialplan in a SBC: [fromtrunk] exten => _[+0-9]!,1,Set(JITTERBUFFER(fixed)=default) This jitterbuffer catches all calls coming from ISP. My understanding is, that the incoming rtp stream in leg1a is now buffered and handed out

Re: [asterisk-users] SwitchVox and Asterisk

On Monday 08 May 2017 at 16:43:24, Luca Pradovera wrote: > Hello, > I need to have an extension on a SwitchVox server dial out to one on an > Asterisk (FreePBX actually) box which will host a voice directory. What's a voice directory? > The Asterisk server will then need to dial one of the

[asterisk-users] SwitchVox and Asterisk

Hello, I need to have an extension on a SwitchVox server dial out to one on an Asterisk (FreePBX actually) box which will host a voice directory. The Asterisk server will then need to dial one of the SwitchVox extensions if it gets a voice match. Anyone has done that, and could let me know how?

Re: [asterisk-users] Dial an extension to modify dialplan

On Monday 08 May 2017, Frank Vanoni wrote: > By dialing 4000 or 4001, the dialplan is modified and reloaded > accordingly. > > Is there a better solution? That's an . interesting . way of doing things! We would be thinking in terms of using a GLOBAL variable, or an ASTDB entry, to

Re: [asterisk-users] Dial an extension to modify dialplan

You could use the DIALGROUP function for this and not need to shell out. https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_DIALGROUP On Mon, May 8, 2017 at 7:35 AM, Frank Vanoni wrote: > Hello > > I have the following scenario: > > [mynicecontext] >

Re: [asterisk-users] SwitchVox and Asterisk

Hello, sorry for not being clear, the application part of this (the voice directory) is already built, mostly working and I have no problem with that. It is based on LumenVox if anyone would like to know, with just a plain XML grammar. I do need to get SwitchVox to send a call to

Re: [asterisk-users] Dial an extension to modify dialplan

https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+ManagerAction_DialplanExtensionAdd Is it enough? Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres

Re: [asterisk-users] Call does not go voicemail

Thank you for the input Tim. Yes, that worked. exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?line2:vmail) exten => 4,n(vmail),Voicemail(4) Though, I'm not sure why are you saying line 2 is FD_L2 needs to be fixed. Do I need to removde "t", the call can not be transferred? Even when I put: exten

Re: [asterisk-users] Dial an extension to modify dialplan

On Monday 08 May 2017 at 15:44:47, Marcelo Terres wrote: > https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+ManagerAction_Dialpl > anExtensionAdd > > Is it enough? Is there a similar call to delete an extension, or to modify an existing one? On the basis that the OP already has extension

Re: [asterisk-users] Dial an extension to modify dialplan

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_DialplanExtensionRemove Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 8 May 2017 at

[asterisk-users] Dial an extension to modify dialplan

Hello I have the following scenario: [mynicecontext] exten => 2000,1,Dial(SIP/deviceA/deviceB/deviceC) As expected, by dialing 2000, all three devices will ring. And that's fine. However, there are situations where I only want "deviceA" and "deviceB" to ring. I would like to have an extension

Re: [asterisk-users] Dial an extension to modify dialplan

> Hello > I have the following scenario: > [mynicecontext] > exten => 2000,1,Dial(SIP/deviceA/deviceB/deviceC) > As expected, by dialing 2000, all three devices will ring. And that's fine. > However, there are situations where I only want "deviceA" and "deviceB" to ring. I would like to have an

Re: [asterisk-users] Call does not go voicemail

The "error" I was talking about was in your log: "...== Spawn extension (extensions, 4, 3) exited non-zero on 'IAX2/home_server-6364'..." The call terminated here in a error which prevented the dialplan from continuing. Something there is broken, my recommendation is to check you registrations

Re: [asterisk-users] Call does not go to voicemail

So, good, we're on the same page so far I think. As I last stated, the original code suggestion would be what you want to do for the serial phone ring-down (hunt), now you just need to figure out why your Line_2 phone is answering and then hanging up immediately (or why Asterisk thinks it is).

Re: [asterisk-users] Call does not go to voicemail

On 05/08/2017 04:37 PM, Tim S wrote: > The "error" I was talking about was in your log: > > "...== Spawn extension (extensions, 4, 3) exited non-zero on > 'IAX2/home_server-6364'..." > > The call terminated here in a error which prevented the dialplan from > continuing. Something there is

[asterisk-users] Cisco 7942G (SIP42.9-4-2) Failover Configuration [SEC=UNCLASSIFIED]

Hi all, It's slightly OT, but hopefully someone can help. I'm struggling with getting Cisco IP Phone 7942G to fail over to our secondary Asterisk server in the event of a primary failure. We recently bought a bunch of new Cisco 7942G phones, which now come with the requirement of FW >