similar to: Remote Attended Transfer

Hello, Throwing calls from Asterisk to PSTN (via a VoIP gateway) some operators sends an explaining audio, in situations as: The phone number does is not assigned The phone is powered off etc. The audio is sent before the call to be answered. So, in an automatic dialling application I'd like to recognize that audio to know what to do with those calls (queue them to a service, mark as wrong

Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something related to directmedia, [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 answered SIP/MAN-000a4b48 [May 17 00:33:35] DTMF[4238] channel.c:

On Sat, Mar 14, 2020 at 2:02 PM John Roman <john at dev1ce.com> wrote: > greetings asterisk users :) > ive just deployed version 17 and migrated as best I can to pjsip. I can > receive calls, and get to my mailbox prompt, however placing calls seems > impossible with the following error on dial: > > Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel

Hello, I use asterisk to to IAX2 trunking between London POP & Reunion Island pop. I would like to know if it's possible to do a kind of call-limit (i.e. restrict to XX) channels but on a per dialcode and / or destination basis. For example: [trunk] ; reunion proper, i want to send no more than 24 channels exten => _0262XXXXXX,1,Dial(IAX2/mytrunk/${EXTEN}) ; reunion mobile, i want

greetings asterisk users :) ive just deployed version 17 and migrated as best I can to pjsip. I can receive calls, and get to my mailbox prompt, however placing calls seems impossible with the following error on dial: Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel (pid = 517890) dunkel*CLI> dunkel*CLI> == Setting global variable 'SIPDOMAIN' to

ive enabled logging. aside from a realm error i see on my endpoint, im still not sure whats up Asterisk GIT-master-0cde95ec89, Copyright (C) 1999 - 2018, Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General

Hello, I need help to solve a problem that I am having using Asterisk 13, PJSIP and T38. My setup is as follows: SIP Provider --> Asterisk 13 --> Patton --> Physical Fax I need to get the fax directly in T38 to Patton. The provider sends me the fax in T38. If I receive the T38 fax on Asterisk (using an hylafax device), I can properly receive the fax. If I send a T38 fax with Asterisk

Hello, For lab testing, I'm trying to build two differents PJSIP trunks between two Asterisk 13.8.0enabled boxes. I thought I could set up both trunks like this: Box A/port 5060 <------ Trunk1 -----> Box B/port 5060 Box A/port 5062 <------ Trunk2 -----> Box B/port 5062 and declare trunks like this: [foobar1] type=endpoint transport=simpletrans context=from-customer

If understand correctly type=identify is more for sip trunk configuration ? ;[mytrunk] ;type=identify ;endpoint=mytrunk ;match=198.51.100.1 ;match=198.51.100.2 In chan_sip it was just reply 200 OK on keepalive packet without need define trunks. volga629 On Sun, 3 Dec, 2017 at 10:45 AM, Joshua Colp <jcolp at digium.com> wrote: > On Sun, Dec 3, 2017, at 10:42 AM, volga629 at

Using pjsip 2.7.2 on Asterisk 15.5 Really struggling to make sense of translating these old 1.8 SIP instructions into a neat pjsip_wizard conf suitable for 2018 http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18 In pjsip_wizard.conf, I have the following, which seems to get me registered, and it responds to an incoming call, but I always get this: [Jul 28 18:32:29]

Hi, I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS. When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial command breaks and the call control go to hangup block instead of next priority. The error in CLI says "*Dial requires an argument (technology/resource)*". This error seems legit as there are no contacts for an offline endpoint. The dialplan

Hello Joshua, What will be example of endpoint configuration that not require authentication from specific ip ? volga629 On Sun, 3 Dec, 2017 at 11:01 AM, Joshua Colp <jcolp at digium.com> wrote: > On Sun, Dec 3, 2017, at 10:55 AM, volga629 at networklab.ca wrote: >> If understand correctly type=identify is more for sip trunk >> configuration ? >> >>

On 2 February 2015 at 15:12, Joshua Colp <jcolp at digium.com> wrote: > Sunny wrote: > >> Hello, >> >> >> In Asterisk 11 it is possible to set extensions on DB table (sipppers) >> and also in sip.conf. >> >> But in Asterisk 13 apparently this is not possible: as I tried to set in >> ps_endpoints and also in pjsip.conf but only the

INFORMACAO CONFIDENCIAL Prezado(a) Amigo(a): Esta carta/e-mail nada tem de semelhante As muitas "aldrabices" que circulam pela Internet. Ela ? uma mensagem rara que tem um conteUdo que pode modificar a sua vida para melhor. Assim, peco-lhe um pouco de paciencia, e que a leia com atencao, muita atencao, e no final, muito provavelmente, se sentira recompensado(a). Este e um assunto que

INFORMACAO CONFIDENCIAL Prezado(a) Amigo(a): Esta carta/e-mail nada tem de semelhante As muitas "aldrabices" que circulam pela Internet. Ela ? uma mensagem rara que tem um conteUdo que pode modificar a sua vida para melhor. Assim, peCo-lhe um pouco de paciencia, e que a leia com atencao, muita atencao, e no final, muito provavelmente, se sentira recompensado(a). Este e

Hello friends. I have a softswitch in which I cannot create a list of blocked source numbers; So, I have thought to use Asterisk and return a 302 message when the number can make the call, my dialplan is as follows: [from-external]   exten => _AX.,1,Verbose(=======> ${CALLERID(num)} to ${EXTEN})    same =>      n,Set(MYDESTINY=${REPLACE(${EXTEN},A,)})    same =>     

Sunny wrote: <snip> > Yeah, it works now. Thank you Joshua! > > However I'm getting following error: > [2015-02-02 16:55:56] ERROR[32605] res_sorcery_config.c: Could not > create an object of type 'endpoint' with id '192.168.1.3' from > configuration file 'pjsip.conf' > > my endpoint is defined as: > [192.168.1.3] > type = endpoint

On 2 February 2015 at 17:23, Joshua Colp <jcolp at digium.com> wrote: > Sunny wrote: > > <snip> > > Yeah, it works now. Thank you Joshua! >> >> However I'm getting following error: >> [2015-02-02 16:55:56] ERROR[32605] res_sorcery_config.c: Could not >> create an object of type 'endpoint' with id '192.168.1.3' from >>