search for: wiater

In PHP something like: $pid = pcntl_fork(); if ($pid != 0) { // we are the parent // do parent stuff exit; } // we are the child, detatch from terminal $sid = posix_setsid(); if ($sid < 0) { die; } // do child stuff On 04/19/2019 02:00 PM, Mark Wiater wrote: > On 4/19/2019 1:49 PM, Dovid Bender wrote: >> Mark, >> >> I am using PHP agi and when forking the call does not continue util >> the forked process is done. Am I doing it wrong? >> >> >> On Wed, Apr 10, 2019 at 4:27 PM Mark Wiater <mark.wiate...

I have an AGI that can sometimes take time complete. I don't want the dialplan to be held up by the agi. Is there any way to call it and have Asterisk continue with the dialplan? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190410/4c704231/attachment.html>

...oft phone is shut down, on hardware phone only an led is > flashing to show an incoming call but no sound. > > both phones use the same extension. that is the reason why I use pjsip. > > On 06.02.19 14:59, Antony Stone wrote: >> On Wednesday 06 February 2019 at 13:54:44, Mark Wiater wrote: >> >>> These two phones are not using the same extension, are they? >> >> If you shut down the softphone, does the hardware phone then ring? >> >> >> Antony. >> >>> On 2/6/2019 8:49 AM, basti wrote: >>>> both phones are...

On Wednesday 06 February 2019 at 13:54:44, Mark Wiater wrote: > These two phones are not using the same extension, are they? If you shut down the softphone, does the hardware phone then ring? Antony. > On 2/6/2019 8:49 AM, basti wrote: > > both phones are registered. and the hardware phone can also make calls. > > but an incoming...

....168.1.13)... and it worked flawlessly... on the SAME Networkcable in the same plug... so it must have something to do with the polycom phone config... remember... when I use tcp the phone tries to register, but does not even try with udp... thank you, yves Am 21.12.2016 um 13:34 schrieb Mark Wiater: > Yves, > > Didn't you say that > >> AsteriskServer: 192.168.1.211 >> SIP-user: 165 > ? > > On 12/21/2016 4:24 AM, Yves wrote: >> . It is sure for 100% that there is no firewall or something else >> mangeling >> in between... another Hardphon...

...an led is >>> flashing to show an incoming call but no sound. >>> >>> both phones use the same extension. that is the reason why I use pjsip. >>> >>> On 06.02.19 14:59, Antony Stone wrote: >>>> On Wednesday 06 February 2019 at 13:54:44, Mark Wiater wrote: >>>> >>>>> These two phones are not using the same extension, are they? >>>> >>>> If you shut down the softphone, does the hardware phone then ring? >>>> >>>> >>>> Antony. >>>> >>>>...

Hello all, I've got dialout= and callback= set in my voicemail.conf so that I can have users return calls to folks who have left messages. They really like this feature. But when the callback is over, a normal hangup occurs instead of the caller being put back into voicemail at the next message. Is it possible that the users be returned into the voicemail system where they left off? thanks

Hi, I am facing a problem where for legal obligations (LI) I have to copy/mirror/forward the RTP streams for some selected call to an external address/port and I have not found a way to do it with built-in functionality. Do I miss something? The basic requirements are: * Raw RTP (no transcoding, header and payload as is) * Direction (did it arrive at asterisk or was it sent) * End

...o register, but does not even try with >> udp... >> >> thank you, >> yves >> > I am a bit confused: is your problematic phone's IP 192.168.0.13 > (what the error log is reporting below) or 192.168.1.13? > >> Am 21.12.2016 um 13:34 schrieb Mark Wiater: >> >> Yves, >> >> Didn't you say that >> >> AsteriskServer: 192.168.1.211 >> SIP-user: 165 >> >> ? >> >> On 12/21/2016 4:24 AM, Yves wrote: >> >> . It is sure for 100% that there is no firewall or something else mang...

Does anyone on this list use Windstream as a SIP trunk provider? If so, would you mind sharing your peer settings? I'm using asterisk 13.7.2 and can't seem to get the inbound working correctly (using registration). Outbound is fine, but they are seeing an authentication error on their end. Here are my inbound peer settings: username=<accountnumber> secret=<secret>

Hello, One common mean to remotely configure a phone is to send it some XML data using HTTP. Of course, this XML data is vendor specific but thanks to Asterisk multiple tools, it is quite easy to customize your dialplan to both build and send this specific XML data. I have just discovered one interesting capability from one phone vendor: getting XML data from incoming SIP NOTIFY messages instead

Hi, I'm trying to integrate Google cloud speech recognition v2 in it. I can get the audio recorded, have created Service key and API key but whenever I try to access it, I just get 403 access denied. I am at my wits end here. Has anybody tried it ? were you successful ? Could you please guide me how to do it ? I'll be grateful to you if this works ! -- Warm Regds. MathuRahul

Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the

Hello, I have some user that had have a hardwarephone and an softphone. I use pjsip driver and set "Max Contacts = 2" to have register both at the same time. But Only the softphone is ring. the hardware phone is mute. How can i fix this?

You need to ensure that traffic to the SIP box is sent to the correct IP. Also if you use split-tunnel (eg: not redirect-gateway def1) you must make sure NAT and traffic redirection works as is so the Asus router knows it should send the traffic through tunnel and not via WAN. IMPORTANT: Then you must, in the ASUS RT-N66U make a port forward inwards from TUN to the phone client. I would suggest

I have an extension that looks like this: exten => 5555551111,1,Verbose(Door buzzer calling) same => n,Dial(SIP/user1&SIP/user2&SIP/user3) The idea is that any of the three users can answer the phone to let someone in. The problem is that if, say, user2 unplugs his phone then the call immediately goes to his voice mail and the other two do not have the ability to open the door.

Hello I find very little feedback on the following warning/error when trying to connect to Google calendar : [Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:118 auth_credentials: Invalid username or password for CalDAV calendar 'cal1' [Oct 26 12:11:14] WARNING[24926]: res_calendar_caldav.c:157 caldav_request: Unknown response to CalDAV calendar cal1, request REPORT to

I sent this last night but it never showed up in the thread list so I'm trying again. Please pardon me if it duplicates. So I've been banging my head against the rack on this one and am now turning to the group for help. I'm in the process of bringing five Asterisk servers (all originally built from source code by myself) from various versions (1.6.2.x,11.6-cert13, and 13.1-cert2) up

On 6/18/2015 1:27 PM, Greg Woods wrote: > My provider claims that I am somehow sending an old number that > doesn't appear anywhere <snip> > (I just moved from a POTS provider Century Link to a VOIP provider). Set(CALLERID(number)=${var}) works fine for me. Perhaps some debugging on the channel could help? Set sip debug if it's a sip trunk? You'll at least get to