similar to: Nube question: where is chan_sip.so?

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

The Asterisk Development Team has announced the release of Asterisk 13.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

The Asterisk Development Team has announced the first release candidate of Asterisk 14.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.0.0-rc1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues

Dne 13.8.2015 v 17:20 Rusty Newton napsal(a): > On Thu, Aug 13, 2015 at 3:54 AM, Marek ?ervenka <cervajs at fpf.slu.cz > <mailto:cervajs at fpf.slu.cz>> wrote: > > hello, > > is it possible simultaneously use chan_sip and chan_pjsip? > > if yes, can you recommend settings > > i'm thinking about > - chan_sip - for sip

Buenas, me gustaría saber su experiencia sobre servicios en la nube. Se que hay varios que usan los servicios de MS Azure. Tengo una duda respecto a eso, funciona como una maquina virtual, no? Lo que significa que para correr procesos debo estar conectado a la maquina, o puedo dejar corriendo procesos en segundo plano y volver luego para ver el resultado. Saludos y muchas gracias! --

You gotta start somewhere I figure. I am enjoying wine 1.1.0 and have used it on and off mostly off for several years. Now I am getting serious and I haven't figured out the documentaiton. I got some bad advice from old docs which discouraged me. Anyway, where does one put the "native" dll's from one's windows installation? I know how to use winecfg to activate them and

just to show i''m not a complete dork: i receive emails from the list through gmail and i''ve already searched for the answer to this problem and since i''ve been a member i''ve not received an email about it. Google search results are mixed at best. so this is my problem: i have a form: <div id="container"

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'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

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