similar to: How to obtain SIPCHANINFO variables within custom application?

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, I am trying to find a way to add the following info in CDRs (with asterisk 1.4.23.1): 1. Codec used 2. RTP QoS statistics 3. RTP IP of remote host 4. For answered calls, the peer that requested to end the conversation I have managed to get 1 and 2 for the caller, like that: exten => h,1,Set(CDR(userfield)=RIP=${SIPCHANINFO(recvip)}

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

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

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 @

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

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

Hi, I'm using Asterisk (1.2.4) as a voicemail system for our Softswitch. When forwarding a call to Voicemail, here is somehow what the softswitch sends to Asterisk : In INVITE : Vm Phone Number ( to route the call ) In To : Person who has been called ! In From : Person who was calling ! Of course, I need to send the call into the "Called User" Mailbox (Thus To SIP header) ! 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

Hello, When i have a "Hangup" in my dialplan (extensions.conf) the RFC says to terminate the session is to send a BYE request to UA. After tracing the traffic on port 5060 UDP i recognized that my asterisk is NOT sending 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

On Fri, 12 Jun 2015 11:49:05 -0700 John Kiniston <johnkiniston at gmail.com> wrote: > Try this for CHAN_SIP: > > same => n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer > same => n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the > mailbox same => n,VoicemailMain(${MailBox}@LocalSets,s) ; If we > have a mailbox defined log into it Perfect.

(This is a shameless copy-paste from the note I posted on http://bugs.digium.com/view.php?id=5090) I have again backported the whole T.38 shebang to the stable branch. The port was based on two versions of the t38passthrough branch: r19125, the latest unconflicted automerge, and r13623, the latest version without the new chan_sip flag structure. Basically, the port contains everything that

I have this in my sip.conf: exten => *98,1,Verbose(0,CALLERID number is "${CALLERID(num)}") same => n,VoicemailMain(${CALLERID(num)}@LocalSets,s) same => n,Hangup However, my extensions are set up so that they always show the external number, not the extension: [foobar2](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxxxxx callerid=Candace <5555551212>

It's not just you. Same thing happens here. I went back to 1.0.7. Stefan Gofferje wrote: > Hi folks, > > I used to have some constructions like > > exten => number/callerid,1,Goto(somewhere) > > After updating to 1.0.8 those does not work any more. > Any hints? > > Regards, > Stefan >

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 :

the gdb of the core taken from the asterisk as the time of crash is as below I run asterisk-1.2.5 on fedora core 3 with chan_ss7 can someone help out? #0 ast_var_name (var=0x1) at chanvars.c:71 71 if (var->name[0] == '_') { (gdb) bt #0 ast_var_name (var=0x1) at chanvars.c:71 #1 0x0808934e in pbx_builtin_getvar_helper (chan=0x0, name=0xf5bc2d46

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

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: > >

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