similar to: MWI - message waiting indication

Hi, A brief 6-step guide on how to hardcode a change in the Asterisk source that will allow it to work from behind a nat device. I know it?s messy, but it may prove useful to some people. 1. First punch a whole in your nat device. I just forwarded the port 5060 (for sip) and all ports between 10000 to 10020 (for rtp) to my asterisk gateway. 2. Now make sure your /etc/asterisk/rtp.conf correctly

Hello, If I understand it correctly, when sending INVITE, o= and c= sdp fields are built using p->ourip IP address. At this point RTP packets will be coming to the default asterisk IP address. For the machine with multiple interfaces this could be not the right one (not what we want). Could it be configured (in rtp.conf or in sip.conf per context) ? Thank you. Alex Zarubin --------------

Revision 77616 Modified Sat Jul 28 07:44:16 2007 UTC (3 months ago) by rizzo File length: 681368 byte(s) Diff to previous 77538 make use of received= and rport= fields in sip replies. In a nutshell, these fields are used to tell a sip entity the address and port its request came from, and are extremely useful in the presence of NATs, especially with symmetric NATs where STUN is totally

Hi, The Polycom 600 phones do not natively bridge with Asterisk. I've solved the problem, but I'm not sure how general it is, so I thought I'd ask this list for advice. It's necessary to use a recent Asterisk CVS for this, since there was a problem with session versions in earlier CVS builds. The problem now is the Via field. When the reinvite goes out, the branch number

Can someone help me with upgrading to the lastest version. I am using the same sip.conf file, but the headers have changed and registration fails. Has something change in the conf file that would cause this? Notice 1.2.5 has no Authoization at all... Regards, Jason Version 1.0.9 --------------------------- REGISTER sip:voip-ca35323.ocn.ne.jp SIP/2.0 Via: SIP/2.0/UDP

Hi list, My need is to append a site specific parameter to the Contact: header on all INVITEs exiting * via a SIP trunk. I'd like it to look something like this: Contact: <bob:3125551212 at 10.10.10.10;SITE-ID=us.here> where SITE-ID=us.here is set in a config file that * parses on startup. Or in a Dial() command option? Or I don't care exactly how. :-) It is possible to

Hi, I know this has already been discussed here, but I still have the problem even with 1.2.6: When I call a mailbox in a context "company" is doesn't play my busy message... It goes directly to the temp message... Am I doing something wrong? == Everyone is busy/congested at this time (1:0/1/0) -- Executing NoOp("SIP/200.234.208.250-0840f548", "Voicemail de

Hi, I input "0203#" after "mailbox?" voice prompt from Voicemail cmd on extensions.conf such as exten => 0021,1,Ringing exten => 0021,2,Wait(1) exten => 0021,3,Voicemail exten => 0021,4,Hangup *CLI> -- Executing [0021 at sip:1] Ringing("SIP/0103-09a308b0", "") in new stack -- Executing [0021 at sip:2]

Hello everyone, i’m using LLVM (updated to 3.1 after seeing that bug, but it’s the same with 3.0) for running a bitcode on a C++ program, and Clang for compiling it. My code work perfectly, as expected on x64, but crash on x86. I’m on Windows 7 x64 and LLVM + Clang was compiled using Visual Studio 2010 (tested in both Release and Debug build). Project was make using CMake. Here is my code:

Hi guys, When I try to get ring tones when dialing out with the command Dial(SIP/sipout/${PHONE},15,r), I get the error message indicated in the subject. I've checked my indications.conf file using the sample file provided with asterisk 1.4.10 (the version I'm using) and it's not better. Any idea ? Regards. -- Cyril SCETBON

When you press *1 in Asterisk (1.6.2.7) to start/stop call recording, the console CLI> shows: > User hit '*1' to record call. filename: wav,auto-1273791789-103-5551212,m Is it possible to play a sound to back to the person who pressed *1 to indicate to them that recording has actually started or stopped? Something like "Recording" / "Record Off", or else sounds

hi all, I have Asterisk 1.6.0.5 Installed and kamailio 1.0.5 version installed when i leave voicemail On Asterisk i need MWI Indication on kamailio extension there are some methods i tried but still cant get success All other feature are working fine also try voip-info.org methods can anybody suggest me for different method and have some different setting on SIP . any help appreciated

When I do a "reload" in the Asterisk CLI I get a long list "Unregistered indication country" lines during the parsing of the features.conf file. Then, when parsing the indications.conf file, they seem to all get re-registered (lines saying "Registered indication country" are displayed). What do these lines mean and why are they unregistered and then registered?

Remarque : message transf?r? en pi?ce jointe. ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version sur http://fr.messenger.yahoo.com -------------- next part -------------- An embedded message was scrubbed... From: harry gaillac <gaillacharry@yahoo.fr>

I'm definitely missing this feature; Best, Patrick

So I just realised that things were changed and the presence of the 'quiet' keyword on the command line suppresses some status messages about loading of the kernel/initrd. I had thought this keyword was strictly for the kernel.. I only realise this now from the note in the change log for 3.74 - so I suppose my suggestion is not quite so important if you remove the 'quiet' keyword

On Thu, Mar 12, 2015 at 9:02 AM, Paul Civati via Syslinux <syslinux at zytor.com> wrote: > Hi all, > > back in previous versions (3.x I think) when you loaded the kernel or > initrd (at least through PXELINUX) you would get some loading progress > on screen with a series of periods. Correct, this was changed in 5.00 as the loading code moved to pure C rather than ASM and

On 18 Mar 2015, at 1:39 am, Gene Cumm <gene.cumm at gmail.com> wrote: >> Would code be accepted if someone were to contribute it? > > Likely but bear in mind the printing would have to be quite conditional. In what respect? I would opt for some kind of progress bar or percentage, but I have realised that whatever is coded would have to play nicely over serial as well as VGA

Hi all, i want lock Zap channels via global var FREE1 if FREE1 = 1 then call should go on with nothing and waiting for digits to go in _X. Otherwise hangup the channel But if the GotoIf goes to s|4 (NoOp) then comes a ringing indication.... !? The "immediate" property in the zapat.conf is "yes" [tel1] exten => s,1,GotoIf($[${FREE1} = 1]?s|4:s|2) exten =>

Does anybody know of any software that can show the status of voicemail messages? Or at least provide a visual indication that I have new voicemail? Right now I am using Gnophone and I'm checking manually. Thanks in advance. -- Christopher Lewis