similar to: SIP and RTP port and IP addresses

Hi All I installing a new S.O. (Ubuntu 14.04) and upgrading my asterisk 1.8. to Asterisk 11 Asterisk 11 is not in production right now but is scheduled to be soon. From its logs (Asterisk 11) I read a lots of: [Jul 11 12:07:45] WARNING[24074][C-00000000] func_channel.c: Unknown or unavailable item requested: 'reversecharge' One of the lines triggering the warning at extensions.conf

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

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

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, 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 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

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 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

Well, things seem to be going really well with tbf, prio and sfq. But I''m a nosey bugger. :-) I''d love to be able to audit what''s going through each of the prio bands. The super ideal solution would be to be able to attach tcpdump to each band and see what''s going through it with the benefit of tcpdump''s filtering so that I can examine and filter

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 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

Hi all I trying to sync a samba password with Linux password. I did: encrypt passwords = yes ; passwd program = /usr/bin/passwd %u passwd program = /tmp/teste %u update encrypted = no unix password sync = yes passwd chat debug = yes passwd chat = *New*password* %n\n*Re-enter*new*password* %n\n*changed* The problem is 'passwd program' is not even called. I

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

Hi All Can someone tell me what am I doing wrong? These was suposed to catch icmp type 8. Why ''match u8'' does not work? # tc filter add dev eth0 protocol ip parent 1: prio 2 \ u32 \ match u16 0x0800 0xffff at -2 \ match u8 8 0xff at 20 \ flowid 1:5 Illegal "match" # # tc filter add dev eth0 protocol ip parent 1: prio 2 \ u32 \ match

On Thu, 22 Oct 2015 15:06:59 +0300 Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: > On Wed, Oct 21, 2015 at 01:33:27PM -0200, Ethy H. Brito wrote: > > > > Hi > > > > Who should insert dahdi_echocan_oslec.ko module in Ubuntu 14.04? > > dahdi_echocan_oslec should be built by default. What you may miss is the > 'echo.ko' (OSLEC) kernel module.