search for: peerip

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&...

...ending a BYE request to it's peer, so the peer doen't know to end the session and continues to send RTP packages to me. Does anyone know how to fix this? Here's the complete trace from ngrep(make call, speak for 5 seconds, hangup): UDP port 5060 in all directions U [myIP]:5060 -> [peerIP]:5060 INVITE sip:011423663900828@sip.provider.com SIP/2.0..Via: SIP/2.0/UDP [myIP]:5060;branch=z9hG4bK4246930c..From: "423663098668" <sip: user@sip.provider.com>;tag=as10b2c259..To: <sip:011423663900828@sip.provider.com>..Contact: <sip:user@[myIP]>..Call -ID: 6dccb...

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 :

...one, 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 -- Got SIP response 488 "Not acceptable here" back from (PeerIP) -- SIP/easycall-838e is circuit-busy ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060127/5be0ac94/attachment.htm

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

...> > in /etc/asterisk/sip.conf: > > allowguest=yes > context=unauthenticated > > > in /etc/asterisk/extensions.conf: > > [unauthenticated] > ;; Incomming calls from unauthenticated caller -> Fail2Ban > exten => _X.,1,Log(WARNING,fail2ban='${CHANNEL(peerip)}') > exten => _X.,2,Set(CDR(UserField)=SIP PEER IP: ${CHANNEL(peerip)}) > exten => _X.,3,HangUp() > > exten => _+X.,1,Log(WARNING,fail2ban='${CHANNEL(peerip)}') > exten => _+X.,2,Set(CDR(UserField)=SIP PEER IP: ${CHANNEL(peerip)}) > exten => _+X.,3,Han...

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

...nt(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 hangup', `peerip` varchar(15) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL, `recvip` varchar(15) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL, `from_u` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL, `uri` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL, `useragent...

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. A...

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 scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090603/87cd735f/attachment.htm

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...

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

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 The URI from the From: header. - uri The URI from the Contact: header. - useragent The useragent. - peername The name of the peer....

Hi, Is there a variable that contains the Sip Peer name? I was using ${CALLERID(num)} for outgoing calls, but when a call is being transferred, that variable contains something else. I need a variable that is always set to the SIP Peer's name. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I am trying to set the CDR(userfield) to a certain vaule using the application map of features.conf but I am not able to do it. When I receive a call I would like to tag it with a client code (3 digit numeric) so I can referenci it later from the CDR. I have edited features.conf with something like: code => #111,self,SET(CDR(userfield(111)) or

...e IaxComm phone and accept the call on the other phone. Then I'm able to transfer the call from the IaxComm phone. I saw that the manager api naming convention changed for IAX channels (no more brackets). Any other change ? I changed the dial string for my IAX phones. Instead of using IAX2/peerip , I used IAX2/recordname. (recordname is the name in brackets in the first line of the phone entry in iax.conf) With that change, one phone was fixed and the other was still not able to transfer. I was not able to find the difference between the two. Uh ! Any information ? Thanks

Hi, incoming SIP calls have a channel name in the form of: SIP/<ip-adresss-of-peer>-<handle> This is a way to get fetch the IP address of the remote side of a SIP call - in most cases. However, sometimes, instead of the IP address, there is a host name in the channel name. I assume, this value in the channel name is not the real IP address, but just a field filled in by the remote

I am currently suffering various SIP attacks. I am using the following extension to record the caller's IP address: exten => h,n,set(CDR(srcip)=${CHANNEL(recvip)}) However, in recent attacks, this IP address is not correct, and I believe that they are spoofing it. I am using asterisk 1.6.2.15. Does the CHANNEL(recvip) variable record IP show in the SIP header instead of the real, UDP

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 -------------- An HTML attachment was...