similar to: revesecharge and asterisk 11

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

>> There are SOME protocol level things accessible using CHANNEL[1] but that's it. >> [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL I am trying to use the CHANNEL function listed above from the AMI. Since it is not an AMI “action”, but rather a dialplan “function”, I’m trying to figure out how to call this from the AMI. Using a telnet session

The uppercase command made a difference. I now get a call-id as show below. However, does the call-id look valid? The @0.0.0.0 seems strange. action: Getvar actionid: act1 channel: PJSIP/Twilio-NA-W-3-In-00000028 Variable: CHANNEL(pjsip,call-id) Response: Success ActionID: act1 Variable: CHANNEL(pjsip,call-id) Value: 4decf884e3ae74595906283a74f7154e at 0.0.0.0 As well,

On Mon, Jun 26, 2023 at 4:04 PM TTT <lists at telium.io> wrote: > It looks like if I call Getvar and pass PJSIP_HEADERS() I can get the > entire SIP header for a channel. I also read (on stackoverflow) that the > PJSIP_HEADER function will only return the headers from the INVITE of the > *inbound* channel. > > > > If that’s correct, how would I get the headers from

I think that’s getting me close. I’m trying to get (or recreate) the FROM and TO lines of the header, from a system running PJSIP. I think if I use CHANNEL to get local_uri and local_tag I can recreate a FROM line like: FROM=<URI>;tag=TAG And if I use CHANNEL to get remote_uri and remote_tag I can recreate a FROM line like: TO=<URI>;tag=TAG Would it be correct to assume

On Sat, Jun 17, 2023 at 2:55 PM TTT <lists at telium.io> wrote: > Based on postings it should be possible to get the SIP Call-ID header > value from the ARI. At what point is this value available ? As well, how > do I retrieve that value – something like > > > > GET /channels/{channelId}/pjsip_header?key=Call-Id > > > > But that doesn’t work. >

Hello, I have a problem with some variables in 1.8.6.0. I set on extension the following lines: exten => h, n, Set (CDR (LLP) = $ {CHANNEL (rtpqos, audio, local_lostpackets)}) ; lost packets by local end ** exten => h, n, Set (CDR (PCR) = $ {CHANNEL (rtpqos, audio, remote_lostpackets)}) ; lost packets by remote end exten => h, n, Set (CDR (ljitt) = $ {CHANNEL (rtpqos,

On Mon, 18 Jan 2016, Ethy H. Brito wrote: >> how to flush user input before READ()? How about a read() to a dummy variable with a 1 second timeout to consume the octothorpe and password? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

Hi I had a bad experience upgrading Ubuntu a few months ago. Today I made a "dd" copy to another harddisk and tried to dist-upgrade. I get "Illegal instruction (core dumped)" running "service asterisk debug" at random places. Any help will be appreciated to spot this problem. I think it is worth to mention that the dadhi hardware is not present at the copied

Hi All We've been reading this in the CLI a lot lately: Received SSL traffic on RTP instance '0x7fe7481faad8' without an SSL session How can we find details about this particular RTP instance? "rtp set debug" needs an IP which is precisely what I want to know (and I don't)! Cheers Ethy

Hi all I am using a pass trhu router and I need to QoS some clients output by its IP address. The problem is that QoS is due after NATing. Is there some clever way of doing this besides MARKing every packet with some IP hashing in POSTROUTING NAT table? Regards Ethy

Hi all These questions were asked back in 2009 at lists.digium.com and got unanswered: - Has someone been successful in using IMPORT on a Local channel ? - Is there a known limitation in doing so ? I run into the same problem. ${IMPORT(Local/1234 at example-abcd;2,CALLERID(dstchannel))} returns nothing. But I can read the dstchannel for it into the CDR. Asterisk is 11.7.0~dfsg-1ubuntu1

I visited yesican.chsoft.biz and the author proposes a way to match packets by less than some size . Here is the thing: match u16 0x0000 0xffb0 at 2 With this match he says that packet with less than 80 bytes will match the rule. Well, 0xffb0 translates to 1111 1111 1011 0000 (which is -80 BTW). So, if I am correct any packet with bits 4 and/or 5 set (together with any of the 4

Hi All What, by definition, goes to the cdr table's "dst" column ?? In our setup, to get outside the user has to dial X before any number. This goes to the dst with the X stripped out. I recently made some changes in a macro and after that the X appeared in this dst column! I run through the script and see no differences between the old and the new one. Then the question: What,

Hi All After a Dial() I get: WARNING[7964][C-000075a8]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) if the subscriber is not registered. Is there a way from dialplan to know, *before* Dial(), if a destination Subscriber is a) not registered or b) busy ? I need to redirect a call to some other Subscriber if (s)he is not there

Dear All Why NAT rules stays valid even if I flush nat anf table chains?? I have: iptables -P FORWARD DROP iptables -A FORWARD -m state --state ESTABLISHED,RELATED -j ACCEPT iptables -A FORWARD -s SOME_IP -d SOME_BCP_5_IP --dport 1234 -j ACCEPT iptables -i nat -A PREROUTING -s SOME_IP -d MY_INTERNET_IP \\ --dport 1234 -j DNAT --to-destination SOME_BCP_5_IP The conection is

Hi how to flush user input before READ()? I wrote a small script to ask for user password before granting access to outside, but some telefones memorize the full user input, including "#". So, when the user press redial, for instance 5556789#123, asterisk accepts the number and the password "123" and gives access to the outside word to whomever redials that terminal. Any

Hi All I have a few lines like this at asterisk/messages. [May 25 15:22:42] WARNING[27725] chan_sip.c: Hanging up call 5a2600300339934f704528bb14ed05e9 at MyAsterisk:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). Since we have hundreds of clients with hundreds of simultaneous calls, how is it possible to know to which customer/IP

On Thu, 28 May 2015 11:15:45 -0500 Scott Griepentrog <sgriepentrog at digium.com> wrote: > The string "5a2600300339934f704528bb14ed05e9 at MyAsterisk:5060" is the unique > identifier for the call in SIP known as the Call-ID. If you have a packet > capture of the port 5060 SIP traffic, that identifier will be in each SIP > message related to the call, which also

Hi all Any of you guys could point me in the right direction? I need to make that a blind transfer to return to the transferrer when the transferee does not answer. Scenario: . Miss Jane Doe, our front desk attendant, picks up an external call to Mr. Smith; . Miss Doe flashes, dial Mr. Smith's extension and then hangup; . Mr Smith's phone rings until timeout; . At this point, how