search for: nyigc

...lt;khamlichi.khalil at gmail.com> wrote: >> >> Hi, >> >> Is there any way I can get exact sip status from pjsip after a dial ? >> or all we can >> get is asterisk hangup causes ? >> >> Thanks in advance. >> >> KKh > -- http://help.nyigc.net/

...xIntoPeers}CurrentCallsCount} - 1]) same => n,Set(GLOBAL(${PeerBeingConsidered}EpochAtCallEnd)=${EPOCH}) same => n,Return() I've also tried replacing the Dial above with: same => n,Dial(${DialForPeer},,g) Cheers, David On Tue, Jun 5, 2018 at 7:38 AM, Eric Wieling <ewieling at nyigc.com> wrote: > Use hangup handlers, they work around the issues with the 'h' extension. > > On 06/05/2018 05:33 AM, David P wrote: > >> Thanks, Anthony. >> >> I added both 'g' and 'F' options. Now, when the caller hangs-up, my >> cleanu...

...TACT: line holds the correct ip! Only the FROM: line holds the wrong (private) IP. I'm still learning SIP...but I assume the FROM should also hold the rewritten public IP. Just don't know how to force Asterisk to do that. -----Original Message----- From: Eric Wieling [mailto:ewieling at nyigc.com] Sent: Wednesday, June 21, 2023 2:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>; TTT <lists at telium.io> Subject: Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE type=endpoint rewrite_co...

...t? I searched >> around and could not find any..... >> Any idea is appreciated! >> >> Kind regards >> Patrick Wakano >> > > I think this is what you may be looking for: > > http://php.net/manual/en/function.set-time-limit.php > -- http://help.nyigc.net/

...hing it looks like this command has disappeared in Asterisk 20 (actually everything after ver 13). I thought it was replaced with CALLERPRES(allowed) but this generated an error too in Asterisk 20. Is there a replacement command? -----Original Message----- From: Eric Wieling [mailto:ewieling at nyigc.com] Sent: Saturday, July 1, 2023 1:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>; TTT <lists at telium.io> Subject: Re: [asterisk-users] AGI script commands You have to read stdin to accept the data Asterisk sends when the AGI...

...a stanza for this transport with: > > local_net=172.31.0.0/16 > > Is that all that's needed for Asterisk to replace the from IP with the external IP? I'm not clear on why Asterisk is not substituting the private FROM ip with a public one... > > > -- http://help.nyigc.net/

...> > to that end, I modified cli.conf > > [startup_commands] > > originate local/s extension s at default = yes > > But now I upgraded to Asterisk18 and there is no longer a local channels > > Does anybody have any idea of a workaround? > > -- http://help.nyigc.net/

...erson, but in perl, i check the process id that's > returned from fork() and exit if it's 1 (parent) and keep processing if > it's the child (greater than 1). > > I think php uses pcntl_fork(). > > Is that how you're doing it? > > > -- http://help.nyigc.net/

...s of the language or AGI type, Asterisk > itself should be able to timeout a long running script and return to the > dialplan. However looks like there is nothing of this sort..... > > Kind regards, > Patrick Wakano > > On Sat, 15 Sep 2018 at 03:56, Eric Wieling <ewieling at nyigc.com> wrote: > >> I don't know AGIspeedy, but I have some PHP scripts where I set a >> connect timeout using streams. >> >> Example using https, but should be easily adaptable to non-s http.: >> >> $pbxsh_bin = @file_get_contents("https://blah.blah....

Hello, Currently my asterisk system pickup incoming call after 3 or 4 rings. How can I ask it to answer the call on the first ring? I put immediate=yes on /etc/asterisk/chan_dahdi.conf but result in no different. Thanks in advance :) BR, Anam -- Sent from my mobile device

...> elapsing between them. > The first thing I achieved by changing a parameter in asterisk.conf, > but how do I conquer the second goal? > > > > ------------------------------ > > Message: 2 > Date: Fri, 6 Jun 2014 13:08:36 -0400 > From: Eric Wieling <EWieling at nyigc.com> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Subject: Re: [asterisk-users] Shorten time between DTMF > Message-ID: > <616B4ECE1290D441AD56124FEBB03D082D165E9BEF at mailserver2007.nyigc.globe&gt...

Hi Yves.. This may be silly... but what is the useragent of your sip configuration? In the case that useragent has some special characters like "(.", please remove it and tell us if there is any change!!. Regards. rv 2014-11-22 14:50 GMT-03:00 Eric Wieling <EWieling at nyigc.com>: > Try setting directmedia=no in sip.conf. > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto: > asterisk-users-bounces at lists.digium.com] On Behalf Of Yves A. > Sent: Saturday, November 22, 2014 8:06 AM > To: asterisk-users at...

I am trying to figure out the best way to deal with this. I want all of the calls in the network to be G729 and this is working. I do have hardware that provides me 30 g729 licenses. I am setting each extensions to disallow=all and allow=g729. However when I have this setup, I get no voice mail prompts. I tried setting to disallow=all and allow=g729,alaw and I still have no audio when calling

Thanks, Anthony. I added both 'g' and 'F' options. Now, when the caller hangs-up, my cleanup code is run by both the caller channel and the peer channel, but I only want the caller channel to do that. Also, when the peer hangs-up, there is no execution of the priorities following the Dial. Finally, is there a way to reset all globals, maybe as a variant of "dialplan

Hi, Is there any way I can get exact sip status from pjsip after a dial ? or all we can get is asterisk hangup causes ? Thanks in advance. KKh

It's a great idea but it doesn't work. Maybe this should be the way that works. -----Original Message----- From: Eric Wieling <ewieling at nyigc.com> Sent: Thursday, August 17, 2023 3:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>; Federico <federico at digitalipvoice.com> Subject: Re: [asterisk-users] Alternative to Local channel You can't set the variable in glob...

Hi, I have an Asterisk box which has an IVR that plays random gsm files. The box has SSD's and two CPU E5-2695 v2 cpus with 64GB ram. The Asterisk CPU usage along with the load seems to jump around. With about 500 callers it hovers between 250-400% CPU (so 2.5 to 4 cores) which seems reasonable. Every so often the load average spikes. The idle never drops below 85%. When the load average

...hing it looks like this command has disappeared in Asterisk 20 (actually everything after ver 13). I thought it was replaced with CALLERPRES(allowed) but this generated an error too in Asterisk 20. Is there a replacement command? -----Original Message----- From: Eric Wieling [mailto:ewieling at nyigc.com] Sent: Saturday, July 1, 2023 1:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>; TTT <lists at telium.io> Subject: Re: [asterisk-users] AGI script commands You have to read stdin to accept the data Asterisk sends when the AGI...

Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 <----------------------> g711

...TACT: line holds the correct ip! Only the FROM: line holds the wrong (private) IP. I'm still learning SIP...but I assume the FROM should also hold the rewritten public IP. Just don't know how to force Asterisk to do that. -----Original Message----- From: Eric Wieling [mailto:ewieling at nyigc.com] Sent: Wednesday, June 21, 2023 2:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>; TTT <lists at telium.io> Subject: Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE type=endpoint rewrite_co...