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...( Extension 300 is hylafax with iaxmodem. On the upper Asterisk gw it's the same, despite the faxdetect set to no we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile phone calling the 0123456789 PSTN number. -- Executing [0123456789 at from-internal:1] Set("SIP/TOOTAi-00008262", "FAXOPT(faxdetect)=no") in new stack -- Executing [0123456789 at from-internal:2] Macro("SIP/TOOTAi-00008262", "Fax") in new stack -- Executing [s at macro-Fax:1] Dial("SIP/TOOTAi-00008262", "IAX2/300,,") in new stack...

I solved the issue by not answering the call as I assume others have done. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Wednesday, February 18, 2015 12:50 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Res_fax - FAXOPT(faxdetect) Hello Le 17/02/2015 17:00, Administrator TOOTAI a ?crit : > Hi, > > as stated in the documentation, it's allowed to set > FAXOPT(faxdetect...

...ed a ticket to their support but in the mean time we want to know if someone is using successfully a PJSIP channel against Kamailio. Another one: despite the fact that they use 5061 port, it's not TLS but UDP. Our asterisk16 has no TLS configured. We use wizard which looks like: [Provider-tootai](!) ; type = wizard sends_auth = yes sends_registrations = yes accepts_auth = no accepts_registrations = no endpoint/call_group = 1 endpoint/pickup_group = 1 endpoint/accountcode = TOOTAi endpoint/language = fr endpoint/allow = !all,ulaw,alaw,g729 endpoint/context = incoming-Provider endpoint/direc...

On Tue, Apr 26, 2016 at 11:07 AM, Administrator TOOTAI <admin at tootai.net> wrote: > Le 26/04/2016 17:23, Mamadou NGOM a ?crit : > >> Hello, >> >> >> Having installed DAHDI to be able to use the meetme() application , when >> I start the dahdi service it generates me the following error: >> >> -ba...

Hi, Le 12/03/2015 17:28, Salaheddine Elharit a ?crit : > hello list, > > i use the code below > > [macro-chanspy] > exten => s,1,Authenticate(${ARG1}) > exten => s,n,ChanSpy(SIP/${EXTEN:3},__dqs) Here you have a problem: ${EXTEN} value is s [...] Daniel

----- Original Message ----- > From: "Administrator TOOTAI" <admin at tootai.net> > To: asterisk-users at lists.digium.com > Sent: Thursday, April 30, 2015 4:43:33 PM > Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In > > > I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and > &gt...

Hi, I want to check the status of a blind transfer (only sip endpoint) between various phones. Transfer is working perfectly, using ## from features.conf or using transfer key from phone, here SNOM320. My problem is that if party to transfer to is busy, the transfer fail and the call is ended. What I want to do is to return the call to the party who originate the transfer. I checked

Hello Le 17/02/2015 17:00, Administrator TOOTAI a ?crit : > Hi, > > as stated in the documentation, it's allowed to set > FAXOPT(faxdetect)=yes/no to allow fax detection. > > It's done (see below) but still fax detection :-( Extension 300 is > hylafax with iaxmodem. > > On the upper Asterisk gw it's the sam...

On Tue, 30 Aug 2016 17:56:35 +0200 Administrator TOOTAI <admin at tootai.net> wrote: > Something like > > exten => 5555551111,1,Verbose(Door buzzer calling) > same => n,Set(toRing=) > same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NOT IN > USE"]?Set(toRing=${toRing}&SIP/user1) Failed. I ch...

----- Original Message ----- > From: "Administrator TOOTAI" <admin at tootai.net> > To: asterisk-users at lists.digium.com > Sent: Friday, May 1, 2015 6:42:38 AM > Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In > > Le 01/05/2015 00:05, Andrew Martin a ?crit : > > ----- Original Message ----- &gt...

On Mon, Apr 6, 2020 at 2:06 PM Administrator <admin at tootai.net> wrote: > Hello, > > We have a provider which is using Kamailio as front end. Our asterisk > 13/chan_sip server has no problem to register and pass/receive calls > form this provider. > > Now we want to move to asterisk 16/pjsip and face problem. Registration > is OK...

Hi list, is there any existing IAX command to add information to a call like SipAddHeader? Another solution is sending text frame (0x07) frame type, but I don know how do it in a dialplan. Thanks for any hint. -- Daniel

Le 27/11/2018 à 12:13, Joshua C. Colp a écrit : > On Tue, Nov 27, 2018, at 5:49 AM, Administrator TOOTAI wrote:[...] >> >> [TOOTAiAudio] >> ; >> ; Call our gateway >> >> exten = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${ARG1}) >>  same = n,Dial(PJSIP/${ARG1}@TOOTAiAudio,,T) >>  same = n,Return >> >> exten = h,1,NoOp() >>  s...

I am attempting installation of spandsp on to an Asterisk installation on Linux RH9 the distribution i am using is that are URL http://ftp2.tootai.net - the README for which i have followed verbatim - my only issue on this was the target for the port.h / tif_dir.h / tiffiop.h files in the 'headers' folder of the distribtion i put these in the /usr/include folder based simply on the fact that there is nothing in the /usr/local/inclu...

Hi list, we wanted to patch our servers with 11.18.0 patch against 11.17.0 actual running version. Patch failed with zone-s:/usr/src/asterisk-11.18.0# patch --dry-run -p0 < ../asterisk-11.18.0-patch can't find file to patch at input line 5 Perhaps you used the wrong -p or --strip option? The text leading up to this was: -------------------------- |diff --git a/.version b/.version |index

On 12/07/2018 03:36 PM, Administrator TOOTAI wrote: > Le 07/12/2018 à 14:32, hw a écrit : > > [...] >> >> Queues seem to be the only way to have several phones ring at once, or >> are there other ways? > > Dial(SIP/Phone1&SIP/Phone2&...&SIP/Phonex,,) > Good to know, thanks! What are the e...

...ERCONFIG to 160. There is no telephony card in the server, we want to use ztdummy for meetme. Does anyone now want can be the problem, knowing that we installed other servers -eg Dell- with similar setup (asterisk 1.4.18) and without facing this problem. Thanks for any information. -- Daniel TOOTAi Network

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 05/05/2015 10:59 AM, Andrew Martin wrote: > > > ----- Original Message ----- >> From: "Administrator TOOTAI" <admin at tootai.net> To: >> asterisk-users at lists.digium.com Sent: Friday, May 1, 2015 6:42:38 >> AM Subject: Re: [asterisk-users] OpenVPN Clients Intermittently >> Cannot Call In >> >> Le 01/05/2015 00:05, Andrew Martin a ?crit : >>> ----- Ori...

On Thu, 1 Sep 2016 11:02:57 +0200 Administrator TOOTAI <admin at tootai.net> wrote: > > [Aug 31 21:52:00] WARNING[-1][C-0001fed5] pbx.c: No application > > 'ExecIf' for extension (unauthenticated, 5555551111, 3) > > > > Is there a module that I need to load? > > > > In case it matters I am running Aster...

Le 08/07/2015 17:36, Richard Mudgett a ?crit : > > > On Wed, Jul 8, 2015 at 8:14 AM, Administrator TOOTAI <admin at tootai.net > <mailto:admin at tootai.net>> wrote: > > Hi list, > > we wanted to patch our servers with 11.18.0 patch against 11.17.0 > actual running version. Patch failed with > > zone-s:/usr/src/asterisk-11.18.0# patch --dry-run -p0 &l...