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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

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: asteri...

...${Peer${IndexIntoPeers}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 &gt...

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

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

Hi All, I have installed asteriskNow with Asterisk 1.6.2.11 and FreePBX 2.7.0.10. I have configured x100p fxo card in my asterisk box. But in my cli mode i am not getting the command *"dahdi show status"* Output of CLI : astrisks*CLI> *dahdi show status* No such command 'dahdi show status' (type 'core show help dahdi show' for other possible commands) I

...same => n,Return When the caller hangs-up, handleHangupByCaller is run first, then handleHangupByPeer runs. (And strangely, the value of global CB${IndexIntoPeers}CurrentCallsCount isn't accessible in handleHangupByPeer.) Cheers, David On Tue, Jun 5, 2018 at 12:58 PM, Eric Wieling <ewieling at nyigc.com> wrote: > Don't use the _. pattern. Ever. > > The call has two channels so it needs two hangup handlers, something like > this, though I've not tested it. > > [some_context] > exten => _X.,1,Noop > same => n,Set(CHANNEL(hangup_handler_push...

...BTW, the CONTACT: 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=endpoin...

On Tue, 30 Aug 2016 10:39:14 -0400 Eric Wieling <ewieling at nyigc.com> wrote: > The dialplan below cannot go to voicemail, either something else is Of course not. It's the individual extensions that have voice mail. I have a similar problem when one of those destinations is a cell phone but I know that there is no Asterisk solution for t...

...google searching 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 w...

hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . ?service zaptel restart? or there is any other command Thanks and regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

In article <alpine.DEB.2.20.1705311339370.15080 at ws.sedwards.com>, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Wed, 31 May 2017, Steve Edwards wrote: > > > I want to capture all SIP messages. > > > > I have about 30 hosts in about 6 colos. > > > > My first thought was dumpcap, but the output file name format bugs me. > > >

Hello Everyone, We are currently experiencing some higher load on our servers, and since signaling comes into our servers on G729, we would like to implement G729 pass-through. A few questions arise, do we need to convert all the recording to the codec, and what about voicemail? We are also using A2Billing (hope I am not violating any thread rules), and would like to convert all that recording

I think HANGUPCAUSE is channel agnostic. See: core show function HANGUPCAUSE Some thing like this IIRC: Set(my_cause=${HANGUPCAUSE(${CHANNEL(name)},tech)}) Remember the incoming leg of the call and the outgoing leg of the call are different channels. Make sure you are giving HANGUPCAUSE the correct channel. On 06/09/2018 02:01 PM, Khalil Khamlichi wrote: > It seems very weird to me

...on regardless 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:...

...ever I set in my answer handler does not show up in the hangup handler. In order to do billing I can't rely on the g option where the caller hangs up the call. Looks like I can either use h or a hangup handler along with the shared function. On Tue, Dec 26, 2017 at 4:40 PM, Eric Wieling <ewieling at nyigc.com> wrote: > Don't use an 'h' extension, use a hangup handler. > Use the MASTER_CHANNEL() function to set variables to ensure they are > always set in the "top most" channel. Below is an untested example, but is > inspired by dialplan code I use i...

Hi There, We have single PRI with multiple DID numbers and its working fine in receiving call. And if you make outbound call it will send main-line CallerID (company name). Now we want individual caller id for per extensions on outbound calls. like if i call someone he will get my extension as callerid ( 617-838-XXXX) XXXX is my sip extension something like this so next time i direct get call

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.blah", FALSE, @stream_context_create(array('https' => array('timeout' => 5, "verify_peer"=>false,

Hi Guys! This is strange issue with 1.8 I have restarted my asterisk and it destroy all registered SIP peers now only solution is i manually reboot all phones to get them register back. I have never seen issue like this before. Any idea what would be the issue ? Thanks S -------------- next part -------------- An HTML attachment was scrubbed... URL:

...nge the time > 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.ny...