similar to: call-limit on a per destination basis

Hello List, The list sorted my problem thus I shall contribute back ;-) PROBLEM: ======== I am posting this example, where I have a "Reunion" link of 30 channels. If i send all the traffic (proper + mobile) on the link, the less profitable proper traffic fills the link and leaves no channel for more profitable mobile traffic. Some kind of priority is needed to always leave space for

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:

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

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

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

> -----Message d'origine----- > De : Andrew Bartlett [mailto:abartlet at samba.org] > Envoyé : samedi 26 août 2017 12:40 > À : HB; samba at lists.samba.org > Objet : Re: [Samba] DC Upgrade from 4.1.7 to 4.6.7 > > On Sat, 2017-08-26 at 12:32 +0400, HB via samba wrote: > > > > > Here is the output of samba-tool dbcheck : > > # samba-tool dbcheck >

> -----Message d'origine----- > De : samba [mailto:samba-bounces at lists.samba.org] De la part de Rowland > Penny via samba > Envoyé : samedi 26 août 2017 12:00 > À : samba at lists.samba.org > Objet : Re: [Samba] DC Upgrade from 4.1.7 to 4.6.7 > ... > On Sat, 26 Aug 2017 11:28:00 +0400 > > Hi, > > > > I have begun to add a new 4.6.7 DC (following

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'm trying to find more information about this Remote Attended Transfers, as is explained in https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Remote+Attended+Transfers for Asterisk 12 using pjsip stack Was Remote Attended Transfer implemented in previous versions of Asterisk (versions without PJSIP, Asterisk 11 and previous)? Where can I find configuration examples to do it work

Hello! Looks like faq, but... Could you , please, point me on how to convert this [cisco] type=friend host=192.168.22.253 insecure=port,invite to pjsip? as you can see another side is very old cisco router, so I can't change anything there. I don't see any examples here

Gaurav, James and I are attending the GSoC reunion in San Jose, California, USA this weekend, so if anyone else is around, do come and say hello. We're also going to be holding a documentation sprint on Monday and Tuesday next week in San Francisco. If anyone else wants to take part (remotely would work) let me know. The only plan we have so far is to come up with a plan over the weekend.

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

Dear ALL, I fear my question has already been answered many times before, but I haven't fund that in archives... I am working on spatial datasets and, in most arrays I'm handling, there are two columns dedicated to (x,y)-coords. For different reasons - notably to draw image() plots, I need to have these two columns sorted in increasing order. But sort() and order() seem to apply to

Hello, 1) What is the difference between a "data frame" (J H Maindonald, Using R, p. 12) and a "vector"? In Using R, the author asks the reader to enter the following data in a data frame, which I will call "mydata": year snow.cover 1970 6.5 1971 12.0 1972 14.9 1973 10.0 1974 10.7 1975 7.9 ... mydata=data.frame(year=c(1970,...),snow.cover=c(6.5,...)) 2) How to

I cannot operate lattice plots, notably those in package gstat, within for() loops. (I can neither read such graphics included in a .R (or .txt) file with the source() function). However same commands executed one by one don't fail. Could you please help me find a means to execute lattice plots inside a loop ? Jacques VESLOT CIRAD -----------------------------------------------------

I am new to Asterisk and currently work mainly with Cisco Callmanager. With Callmanager I can setup partitions and call search spaces to determine where a given phone can and can't dial. Does Asterisk offer this type of functionality, and if so how? Blake Parker -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Everybody First I want to said thanks for the great job you've been done with samba4. I'm french and I live in Reunion Island near South Africa. Sorry if i'm english is wrong lol. Here my trouble. I follow the how to on the wiki samba. I successfully join the domain I create. But when I try to log a user on windows xp ( create via swat and configure in 2003 active

I have some simple code so that when you click on a thumbnail, it''s supposed to display the image enlarged in the enlargement area. The page looks right, but when you click on a thumb nothing happens, it just moves up the page. Why is the href=# happening ? The generated html comes after the sample rhtml <%= image_tag @enlargement, :width => "279", :height =>