[asterisk-users] PJSIP DNS ISSUE

2019-02-20 Thread John T. Bittner
Anyone know how to disable DNS in asterisk so PJSIP still works when the internet goes down. I tried a few things but nothing is working. I even installed BIND on the asterisk box ...that didn't even work. Once I pull the plug on the internet, I cant dial anything. John Bittner CTO

Re: [asterisk-users] PJSIP DNS ISSUE

2019-02-20 Thread Eric Wieling
Make sure the IP of every interface address is listed in /etc/hosts Use dnsmgr Install local BIND, which you already did. On 02/20/2019 11:29 AM, John T. Bittner wrote: Anyone know how to disable DNS in asterisk so PJSIP still works when the internet goes down. I tried a few things but

[asterisk-users] if function when the true value has a colon in it?

2019-02-20 Thread Brian J. Murrell
Following up on my previously asked question if I rewrite the branching example (not that it negates the more general branching question) I was using as such: exten => s,n,Set(EXT=${IF($[${SIP}=PJSIP]?${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,PJSIP/,)})}:${ARG2})}) exten =>

Re: [asterisk-users] PJSIP DNS ISSUE

2019-02-20 Thread Joshua C. Colp
On Wed, Feb 20, 2019, at 12:30 PM, John T. Bittner wrote: > > Anyone know how to disable DNS in asterisk so PJSIP still works when > the internet goes down. What exactly isn't working? What's the scenario? PJSIP uses asynchronous DNS, so it shouldn't block Asterisk like chan_sip did. There

Re: [asterisk-users] PJSIP DNS ISSUE

2019-02-20 Thread Ryan, Travis
Can't you just reference everything in IPs? If not, then hardcode the IPs in your /etc/hosts file. I think that's a bad idea, but that's one way to ensure you always have the Ip of a domain name. From: asterisk-users On Behalf Of John T. Bittner Sent: Wednesday, February 20, 2019 11:30 AM To:

[asterisk-users] branching in extensions.conf?

2019-02-20 Thread Brian J. Murrell
Is there any less cumbersome way of doing conditionalized/branching in extensions.conf other than something like: exten => s,n,GotoIf($["${SIP}" = "PJSIP" ]?pjsip) exten => s,n,Dial(${ARG2},20,TtWw) exten => s,n,Goto(afterdial) exten =>

Re: [asterisk-users] branching in extensions.conf?

2019-02-20 Thread John Kiniston
Use the IF function to evaluate and change the dial command directly. My braces and parens may be off in this example sorry if it doesn't work out of the box. exten => s,n,Dial(${IF($["${SIP}" = "PJSIP"]? ${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,"PJSIP/","")})}{ARG2})},20,TtWw) On Wed, Feb 20,

Re: [asterisk-users] branching in extensions.conf?

2019-02-20 Thread Brian J. Murrell
On Wed, 2019-02-20 at 11:46 -0700, John Kiniston wrote: > Use the IF function to evaluate and change the dial command directly. Thanks for taking the time, but that doesn't actually answer the question I asked. It in fact answers the caveat I specifically mentioned: > Granted the particular

Re: [asterisk-users] if function when the true value has a colon in it?

2019-02-20 Thread Stefan Tichy
On Wed, Feb 20, 2019 at 12:08:14PM -0500, Brian J. Murrell wrote: > exten => > s,n,Set(EXT=${IF($[${SIP}=PJSIP]?${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,PJSIP/,)})}:${ARG2})}) > > But that ${IF expression?tval:fval} doesn't work because tval has a : > in it which the if function is taking as the

Re: [asterisk-users] branching in extensions.conf?

2019-02-20 Thread John Kiniston
I don't see any other messages in this thread other than your initial one and my response, perhaps the listserv hasn't relayed it to me yet. I switch between ExecIf/GotoIf/GosubIf and the IF function as needed in my dialplan design, if It's just a one liner I'll usually use either the IF function

Re: [asterisk-users] branching in extensions.conf?

2019-02-20 Thread Eric Wieling
If you want your dialplan code to look pretty, use AEL. On 02/20/2019 11:41 AM, Brian J. Murrell wrote: Is there any less cumbersome way of doing conditionalized/branching in extensions.conf other than something like: exten => s,n,GotoIf($["${SIP}" = "PJSIP" ]?pjsip) exten =>

Re: [asterisk-users] branching in extensions.conf?

2019-02-20 Thread Brian J. Murrell
On Wed, 2019-02-20 at 12:38 -0700, John Kiniston wrote: > I don't see any other messages in this thread other than your initial > one > and my response, perhaps the listserv hasn't relayed it to me yet. I started a new thread:

[asterisk-users] ARI delayed 183 due to device_state lookup

2019-02-20 Thread Jöran Vinzens
Hi all, we are searching for shorter post dial delay and we were wondering why the asterisk takes about 190ms from an ARI Command Playback or Answer until the SIP Message is send out. We took the latest code from github and used the python scripts provided by sangoma in order to start a