similar to: allowguest defaults to yes for SIP

Hello, I want that people from other servers like ekiga.net can make calls to my users. When I do an "allowguest=no" then people from other domains cannot call me. So I think I need "allowguest=yes". Maybe something like this? ------------- <default> include => users <dialout> include => users exten=_0.,1,Dial(SIP/trunk/0${EXTEN:1},30,tT) <users>

Hi i have a PRI connected to a TE205P. Actually, can i send and receive FAX through Asterisk using stable solutions? Or shall i connect an ATA to Asterisk and then a modem with Hylafax? -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser

Hi Last month, people reported a crash with Asterisk 1.2.13 and spandsp-0.0.3 when receiving a fax using fax detection. Today I've tried 1.2.14 with the latest spandsp 0.0.3pre27 and with the snapshots for app_rxfax.c and app_txfax.c. The problem still happens. Has anyone found how to resolve this issue? I tried emailing Steve Underwood (with crash backtrace) but he hasn't

I'm using IAXmodem and Hylafax with 'faxdetect=incoming' and things mostly work pretty well. My main lines come in via T1 DID. Today, HR got tired of having someone read and forward their faxes to them and requested we bring their physical machine back on line. I have been able to get the fax forwarded to the appropriate zap channel, but I cannot seem to get it to stop

Hi! from 2 days I'm trying to run hylafax server and iaxmodem with Asterisk 1.8.5. I have 2 computers in the lan, one is the Asterisk PBX and the other is the server with hylafax and iaxmodem installed. In Asterisk I set up an IAX trunk in this way: ___________________________ iax.conf [iaxmodem] type=friend context=outgoing-fax disallow=all allow=ulaw username=iaxmodem secret=password

Hi, I had the following odd behaviour in Asterisk 1.2 - We are migrating to 1.6, and I will re-test ASAP, though it is quite hard to replicate, but I am curious to know whether it is a known IAX issue in 1.2. We had 2 users in iax.conf: [user1] username=user1 secret=secret1 context=context1 host=iax.hostname.com [user2] username=user2 secret= context=context2 host=dynamic deny=0.0.0.0/0.0.0.0

Hi, I'm trying to get a softphone configured. In Sip.conf [general] I found an example that said I need: nat=yes localnet=192.168.xxx.xxx Is localnet supposed to be a LAN IP or a WAN IP? Thank you, Gary

I need some help understanding SIP dialog. Some actor is trying to access my server, but I can't figure out what he's trying to do ,or how. I'm getting a lot of these warnings. [May 17 10:08:08] WARNING[1532]: chan_sip.c:4068 retrans_pkt: Retransmission timeout reached on transmission _zIr9tDtBxeTVTY5F7z8kD7R.. for seqno 101 With SIP DEBUG I tracked the Call-ID to this INVITE :

http://www.asterisk.org/node/48327 I mean, really... you're kidding me, right? Lee.

I do not want set allowguest=yes. The problem is, there is no official list with ip addresses of Telekom Germany. But I think all ip addresses comes from the ip range 217.0.0.0/13. I have now the following addition to sip.conf. I think it is the only safe option. Or what would you say? [telekom](!) context=from-trunk type=peer defaultuser= authuser= remotesecret= fromdomain=tel.t-online.de

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

> what is the sip.conf for 1239 > which I'm going to assume is a extension on the TNT > > Barry > > JR Richardson wrote: > > Hi All, > > > > I have a lab setup with two asterisk servers and a MAX TNT in the > > middle like this: > > > > asterisk sip >< sip TNT pri >< pri asterisk exten 1239 is the CID Number from the

On 05/17/2018 11:38 AM, Frank Vanoni wrote: > On Thu, 2018-05-17 at 11:18 -0400, sean darcy wrote: > >> 3. How do I set up the server to block these ? >> >> 4. Can I stop the retransmitting of the 401 Unauthorized packets ? > > I'm happy with Fail2Ban protecting my Asterisk 13. Here is my > configuration: > > in /etc/asterisk/logger.conf: > >

Hi list, i noticed from the cli my asterisk box is accepting unauthenticated calls how can i prevent this? CLI: -- Accepting UNAUTHENTICATED call from 192.168.0.2: > requested format = gsm, > requested prefs = (), > actual format = ulaw, > host prefs = (g729|ulaw|alaw), > priority = mine -------------- next part -------------- An HTML

Dear OpenSSH developers, a publicly accessible sshd on port 22 generates a lot of log clutter from unauthenticated connections. For an exemplary host on a university network, sshd accumulates 5~20k log lines on a single day (more than 90% of the total amount of syslog lines). That is despite the host having a restricted configuration (no SSH password authentication, firewall rate limit for

Hi i need to implement "Inward" SIP usring dialing in my Asterisk IPpbx, So anybody can recah me by dialing my SIP uri. same time my DNS on same server where currently Asterisk running. how ican implement this. Please help me with config details at DNS & Asterisk point of view. anybody can provide me config exmple? I am using Asterisk 1.4.9. Plz help me Regards Amit

Asterisk Project Security Advisory - AST-2008-003 +------------------------------------------------------------------------+ | Product | Asterisk | |--------------------+---------------------------------------------------| | Summary | Unauthenticated calls allowed from SIP channel | | | driver

Asterisk Project Security Advisory - AST-2008-003 +------------------------------------------------------------------------+ | Product | Asterisk | |--------------------+---------------------------------------------------| | Summary | Unauthenticated calls allowed from SIP channel | | | driver

John Hodrien wrote: > On Thu, 22 Feb 2018, hw wrote: > >> That seems neither useful, nor feasible for customers wanting to use the >> wireless network we would set up for them with their cell phones.? Are cell >> phones even capable of this kind of authentication? > > Yes, entirely capable.? WPA2-Enterprise isn't some freakish and unusual > solution. Ok, so

Not sure if this is a bug yet, so I wanted to ask around to see if anyone else was having this issue. I have pollmailboxes=yes set in voicemail.conf but externnotify is not called. I know it isn't the externnotify script because if the changes are done in asterisk then it is called properly, if the changes are done via our webserver then it is not. Also, we use odbc voicemail storage. Thanks