similar to: Capture Media IP in CDR

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 =>

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

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

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, 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:

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

hi, i have strange problem with call-limit/groupcount limiting. i set up limit of 2 calls. i'm using both methods but a for few times i have problem with successfull fraud with more calls than 2 asterisk is 1.8.22 someone with the same problem? any ideas how to solve or debug this problem? -- --------------------------------------- Marek =======================================

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

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)

A few months ago I started using and had to abandon PJSIP because my dialplan could not read the inbound signalling IP address, which I can read now in Asterisk11 using CHANNEL(recvip). My app relies on this information. The question is, is it possible now access the signalling IP of an incoming SIP call using PJSIP? Philip

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

Dear Tariq; About elastix.org, this can be use with Asterisk or it is coming as a complete IP Telephony, Call Center, IVR and Reporting? Because, I do not need to install another IP Telephony on the server which already has asterisk which is an IP Telephony, this will cause a problem in the service (for example, when listening for SIP port of 5060).

* Thomas Leuxner <tlx at leuxner.net> 2015.08.25 09:45: > > http://hg.dovecot.org/dovecot-2.2/rev/64c73e6bd397 > > ==> /var/log/dovecot/dovecot.log <== > Aug 25 09:42:07 nihlus dovecot: imap(tlx at leuxner.net): Error: net_connect_unix(/var/run/dovecot/imap-hibernate) failed: Permission denied > Aug 25 09:42:07 nihlus dovecot: imap(tlx at leuxner.net): Error:

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://hg.dovecot.org/dovecot-2.2/rev/64c73e6bd397 Today I finally committed the "imap-hibernate" feature that I first started developing about a year ago (and had been thinking about for several years before that). The main purpose here is to reduce the number of imap processes and the amount of memory they use by moving IDLEing connections into imap-hibernate processes where they are

Hello all, I am looking for a way to restrict users from logging in two separate phones with the same authorization name/password at the same time. Meaning, I only want users to be able to place a call from one phone in one location, but have the ability to move from computer to computer. Has anyone found any sort of solution for this type scenario? Thanks, Bryan Mahin Please visit us @

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

Although RAW sockets can be used when specifying the source address of packets (defeating one of the aspects of the jail) some people may find it usefull to use utilities like ping(8) or traceroute(8) from inside jails. Enclosed is a patch I have written which gives you the option of allowing prison-root to create raw sockets inside the prison, so

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