similar to: PJSIP question

Hi all I'd like to log the client IP addr and port used for SIP and RTP *during* in a call. The IPs must be the real source IPs (internet accessible). How are these parameters available from dialplan? For instance, ${SIPURI} holds the internal "IP:port" if the client is behind NAT. I need the external IP:port Regards Ethy

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

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

Hi!I've a problem using ssh without password: I want use rsync for automatic scripts,I'm using this 2 names for my asterisk@home2.5 linux (based on red hat), rsync11 and rsync12. This is the way I use to change the configuration and then using without password , but the password is always asked: [rsync11@asterisk11]$ ssh-keygen -t rsa Generating public/private rsa key pair. Enter file

I cannot find in Asterisk 12, the channel variable ${CHANNEL(recvip)}, so if I use PJSIP, for scalability, how do I read what the signalling IP where the inbound call is coming from and what is the inbound codec? You would think that the new channel would set those variables up, isn't it? Philip Orleans

Hi, My dialplan looks like this: [from-external] Exten => _X.,1,Noop(CALL IS COMING INTO FROM EXTERNAL CONTEXT) Exten => _X.,n,Noop(IP: ${CHANNEL(recvip)}) Exten => _X.,n,Noop(CALLED NUMBER: ${EXTEN}) Exten => _X.,n,Ringing Exten => _X.,n,WaitExten(15) Exten => _X.,n,Congestion() Exten => _X.,n,Hangup() include => test [test] Exten => 8282,1,Noop(--- WE GOT TO

Hi all, For one of my inverstigations it looks like i'm back to "square one" I'm trying to accept an incoming xmpp call and forward it conditionally to a sip, isdn, or voicemail. No google is involved as i use a local xmpp server (ejabberd) I was experimenting on 1.8.15.1 (with jabber.conf, jingle.conf), but some suggested me to have a look at asterisk11,so i did... I

On Tue, Jun 26, 2018 at 7:06 PM, Doug Lytle <support at drdos.info> wrote: > On 06/26/2018 06:57 PM, Dovid Bender wrote: > >> Hi, >> >> My dialplan looks like this: >> [from-external] >> >> Exten => _X.,1,Noop(CALL IS COMING INTO FROM EXTERNAL CONTEXT) >> Exten => _X.,n,Noop(IP: ${CHANNEL(recvip)}) >> Exten => _X.,n,Noop(CALLED

I have been seeing errors saying the Asterisk cannot establish an RTP connection, so I did this: lsof -i -n -P | grep asterisk | wc -l 10483 but I have only Asterisk 11 has 1 open calls Asterisk 12 has 21 open calls Asterisk 14 has 19 open calls Asterisk 15 has 22 open calls Asterisk 16 has 15 open calls Asterisk 17 has 15 open calls Asterisk 30 has 71 open calls Total 164 active calls The

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

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 =>

Hi! I' ve this problem: I've 2 asterisks box, asterisk11 and asterisk12, and one wi_fi phone. I call from wi_fi to a X-lite phone on a windows xp.I've setuped the X-lite to my vrrp IP (vrid IP) and the call is ok, I call from the wi_fi to X-lite and from the X-lite to wi_fi. In asterisk panell is all ok, and I listen the voice to the xp and in the wi_fi phone. asterisk12 is my master.

Hi everyone! I tried to install chan_dongle for Asterisk 11 in a Ubuntu 14.04, but my Huawei E153 is not working in my Asterisk. I fallow this rules http://blog.denisbondar.com/post/asterisk11-chan_dongle_e1550-ubuntu14 But not successes. Thanks in advanced,

?Asterisk 13.11 rc1 ./configure LDFLAGS="-z muldefs" --libdir=/usr/lib64 --with-unixodbc=$(odbc_config --include-prefix)/ --disable-dev-mode --with-pjproject-bundled ?checking for pjsip_dlg_create_uas_and_inc_lock in -lpjsip... no checking for pjsip_tsx_create_uac2 in -lpjsip... no checking if "pjmedia_mod_offer_flag flag = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE" compiles using

Hello We had this situation: Some bot-net did try to guess SIP logins and finally succeeded. The Asterisk Server was abused to call a large number of expensive destinations. It is clear that the sip logins have been passed to various persons (probably posted on a forum somewhere inviting to do 'free calls'). Right after the affected password was changed, the message log shows which

I am getting this message: Failed to authenticate device <sip:2010 at X.X.X.X>;tag=149853321 for INVITE, code = -1 but it does not report the "connecting" address. Who is failing connecting ? I either need to block someone or fix something - I'm thinking block - but I dont know who. How do I found out the connecting IP? Jerry -------------- next part -------------- An HTML

I am stuck in channel PJSIP trying to see the real flow of SIP messages, what in regular sip we used to type "sip set debug on" Also, is there an automated way to convert sip.conf options to pjsip.conf? Philip

I did what we use to dim that is add a line to pjsip.conf like #include /etc/asterisk/pjpeers.conf but the file is not loaded. Am I doing something wrong this functionality is disabled?

I could not find a way to set a max on the calls allowed through a PJSIP endpoint. In case we decide to add it, the we need another reason for the call to fail in the Dial application, something like "limit reached" Am I missing this capability?

I found that PJSIP allows only one asterisk per box. I tried to start several asterisks with the parameter -C and PJSIP only worked on the first process. In the other processes, the command "pjsip reload" was absent. Each pjsip transport in the second and subsequent processes was bound to a different IP in a multihomed box, something I routinely do with regular SIP. Am I wrong?