similar to: HELP: BYE-request not sent to SIP-peer

Hello, How can I get peerip, recvip, from, uri, useragent, peername, t38passthrough variables in (within) my custom Asterisk application? I can't use chan_sip.c internal structures (such as sip_pvt) in my custom application, because there's no chan_sip.h and I can't include it into my application (maybe there's other way?). I can do like this: exten =>

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: > >

because of problems you are facing i decided to go way with second table CREATE TABLE `cdr_extended` ( `id` int(11) unsigned NOT NULL AUTO_INCREMENT, `uniqueid` varchar(32) NOT NULL DEFAULT '', `callid` varchar(256) NOT NULL DEFAULT '' COMMENT 'sip call-id', `hangupcause` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL COMMENT 'info about

core show function SIP<TAB> I use: set(PEERIP=${SIPCHANINFO(peerip)}) in one of my dialplans. For AGI, whatever function in your library that executes 'GET FULL VARIABLE' should do the trick. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline

Hello! I want to use "asterisk -rx "show version"" from a php script called in the browser using the local apache, which runs as user "apache". Asterisk is running as root. I added the following line to /etc/sudoers using visudo: apache ALL = NOPASSWD: /usr/sbin/asterisk When i am on the command line of my linux box it looks like this:

I'm trying to isolate the IP address of inbound calls to my switch over IAX2. Is the proper way to get that information as follows: ${IAXPEER(IP)} If the caller was inbound via SIP, this works: ${SIPCHANINFO(PEERIP)} So I'm looking to return the IP address of the caller via IAX2. Thanks Lee -------------- next part -------------- An HTML attachment was

What is the variable like $peerip to get the registered ip address for a peer Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do

On 12/30/2017 08:18 PM, Dovid Bender wrote: > Script kiddies trying to find vulnerable systems that they can make > calls on. Lock down the box with iptables and use fail2ban to block > them. The via is probably bogus unless a box at the DoD was comprimised. > > > > On Sat, Dec 30, 2017 at 6:49 PM, sean darcy <seandarcy2 at gmail.com > <mailto:seandarcy2 at

You can use the following: switch3*CLI> show function SIPCHANINFO switch3*CLI> -= Info about function 'SIPCHANINFO' =- [Syntax] SIPCHANINFO(item) [Synopsis] Gets the specified SIP parameter from the current channel [Description] Valid items are: - peerip The IP address of the peer. - recvip The source IP address of the peer. - from

Hi List, Which will be the appropriate variable to get called peer IP address? I tried following channel variables peerip, recvip, URI, from and following SIP channel variables: SIPURI,SIPDOMAIN They all return calling peer IP but not the destination/called peer IP. unfortunately set(CDR(calledip)=${CHANNEL(to)}) doesn't work Regards, Zohair Raza -------------- next part --------------

I updated from CVS yesterday and today and still have the problem. IaxComm cannot transfer the call when it's an outgoing call. ('outgoing' is from the dial plan point of view). details : First I call the IaxComm phone and accept the call. Then I'm not able to transfer it from the IaxComm phone. If the call is an incoming call it works fine. details : First I call a phone

Hi, I am having trouble configuring CAPI so that call transfers work. I make a SIP call to asterisk which goes out on ISDn via CAPI. Then I try to do a transfer from the SIP phone which doesn't work and results in the call being disconnected. The error message given by asterisk is that it chan_capi can't find an entry for the outgoing msn for the transfer however the outgoing msn is the

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

Hello when setting a local forward (in this case to extension 23) on a SIP phone, I see the following on the Asterisk CLI : [Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back from 11.22.33.44:40670 [Aug 31 14:59:34] -- Now forwarding Local/myaccount184 at CallFromQueue-000007f4;2 to 'Local/23 at from-internal' (thanks to SIP/myaccount184-00003729)

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 :

Hello. Voice quality when calling - this is one of the most important in the PBX. You need to record the quality parameters for each call to improve. Because the overall quality of a call can only be determined upon completion, I did it in the HangUp handler and wrote in custom fields of CDR. This worked well in asterisk 11. In asterisk 13 I did not find a handler after the call, but before

I am not proxying the media, but never the less I am forced to store the source media IP in my CDR, for regulatory reasons. Asterisk gets that information when the reinvite comes, but how do I store it? If I don't figure this out my next email will be from Federal Prison. Kindly help me stay away from those guys. Eventually we all need to save that information or we shall not be able to stay

using successfully active/active replica. trying to add a proxy node in front. this proxy node should do the auth with the same ldap passdb settings as the replica in addition (later with kerberos). so i add to 10-auth.conf on the proxy: default_fields = proxy=y host=imap.myserver.lan port=993 any idea why on the backend the user is empty in the logs? on the proxy: imap-login: Error:

I have the need to modify the behavior of the virtual network driver's behavior and how it deals with routed networks. I'm running libvirt-0.8.3-2.fc14. According to http://libvirt.org/firewall.html, the following is automatically added to the FORWARD chain of iptables when a network type of "routed" is started up: "Allow inbound, but only to our expected subnet.

Hi List, Is there easy way to get klips ipsec stack into centos 6? As it makes firewalling ipsec traffic much easier.. Eero