similar to: What this attacks means?

Hi List, The below error caused the Asterisk to crash, if anyone have idea on this please reply,(Asterisk version :1.8.9) [Sep 2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel SIP/18202-0002d011 setting write format to ilbc from ulaw native formats 0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: Unable to find a codec translation path from 0x4 (ulaw) to

Hello list! I?m working on a fresh Asterisk install over CentOS7 base. I?m using ?Asterisk. The Definite guide? book as a reference. I connect and work using SSH Problem I have - I can?t connect to asterisk from remote. Getting error: $ sudo asterisk -rvvvvvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) Yes, it exist, and service runs: [asteriskpbx at

using asterisk 11.6.0-rc1 i just converted my "nat=yes" to "nat=auto_force_rport,auto_comedia" I have my asterisk box on the same subnet as a cisco 1760 (vgw1). a few times per day, Asterisk thinks vgw1 is dead (by qualify/options). A 'sip reload' always fixes the problem. i left 'sip set debug peer vgw1' on the console. but i dont see what's

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Hi, Am making a simple SIP trunk between two Asterisk server, Server 1 sip.conf [usman02] type=peer username=usman02 secret=usman02 host=10.30.2.58 context=man02-trunk port=5060 qualify=yes disallow=all ;allow=g729 allow=g729 ;allow=alaw nat=force_rport,comedia dtmfmode=rfc2833 relaxdtmf=yes insecure=invite,port extensions.conf [man02-trunk] exten => _1X.,1,Dial(SIP/usman02/${EXTEN}) exten

On 2018-09-09 10:27, Antony Stone wrote: <snip > 1. Try removing one of the two commas. > > 2. Take a copy of your dialplan, and then strip out *everything* except > the > one context and the one number you want to match: > > [0705680837] > exten => 31705680837,1,NooP( Incoming 31705680837 on CC) > same => n,Answer(); > same =>

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes

Asterisk Project Security Advisory - AST-2011-013 Product Asterisk Summary Possible remote enumeration of SIP endpoints with differing NAT settings Nature of Advisory Unauthorized data disclosure Susceptibility Remote

hello folks, this might be a simple question... I just installed asterisk in a debian server. All seems to be running fine, but the audio sent by the server. If I have one of my registered peers call and extension (102) that plays back audio (extension.conf and sip.conf coffee-pasted below), Asterisk answers and prints no errors. Its `sip show channels` prints: Peer User/ANR Call ID

Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are starting to complaint about packets loss, conversations are choppy! I don't even know where to start looking! Choppy conversations happened within users. I am using sip.conf [1091] type=friend context=sip-phone call-limit=2 trustrpid=no callerid="dev1" <1091> disallow=all allow=ulaw

Hi list, I've been googling this issue and found some good resources however I am still running into problems with the following combo ... Here's my story; - Asterisk 13.4 with FreePBX 12. - Migrating from Asterisk 11 / FreePBX 2.11 - Mix of Cisco 79xx handsets, mostly 7940G's. My problems started with (the very common) issue of the 7940 not replying to 401

I'm quite new to Asterisk and using Asterisk 13 on FreeBSD current. Asterisk is behind a NAT router, the physical setup is very much a trivial one. The Asterisk PBX is supposed to act as the telephone gateway for several VoIP/SIP phones. I'm using throughout pjsip as configuration, I have no experience with chan_sip since I started recently using Asterisk for several SoHo and lab's

I?ve gotten to the bottom of this; Seems that the pjsip.endpoint_custom.conf isn?t getting included properly, or my syntax is wrong. If I put force_rport=no into pjsip.endpoint.conf and reload only Asterisk, everything works perfectly. Unfortunately, I?m using FreePBX, so it owns this file and my changes won?t persist a FreePBX reload. Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F

Hi, we had an attack on a server and we don't understand how it was possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL, network 188.161.128.0/18 Hacked account had following setup: [111] type=friend username=111 context=from-111 host=11.22.33.44 dtmfmode=auto qualify=yes nat=yes canreinvite=no defaultip=11.22.33.44 port=35060 disallow=all allow=ulaw,alaw call-limit=2

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

Hi Guys, Just new to Asterisk and am completely stumped. I have created two accounts as instructed. Please see below for the config of the user accounts. [Peter] type=friend host=IP address disallow=all allow=ulaw allow=alaw callerid=Peter <6004> secret=XXXXXXX context=default port=9060 nat=force_rport,comedia deny=0.0.0.0

On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan at amtelco.com> wrote: > > I am not sure if I entered the correct settings for the transport > information. > > For the local_net, I entered my local ip address, but no mask. I will > check with the network admin so he can verify the settings I entered. > > > You need the network and mask. For example if the ip

Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > What settings have you got for directmedia? > > Could you try > > nat=force_rport,comedia > directmedia=no Tried. Peer always unreachable, call not possible... :( Other idea? Thanks Luca Bertoncello (lucabert at lucabert.de)

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia