search for: behing

I have a client requirement that multiple phones can be dialed, however they don't want the pstn phone to pick up automatically because of voicemail etc, nothing can be changed on the phones, how can I handle this requirement, by the way no zap channels are involved, all the pstn phones are behing another sip gateway.

Hi all, I use tinc since 2 years between my MPLS sites in router mode, works like a charm. Today i am trying to make an ethernet bridge beetween 2 sites for video conference needs, the raison is that 1 side is behind nat, and i dont have access on the gate, other side is on public range. Because off H323 that doesn't handle unconfigured NAT, i would like to give a public IP to the video

What dbPowerAmp does is encapsulate the wav file into a FLAC container without actually compressing it. I guess it can be a bit arguable if you can technically call it a FLAC file :) It's like using "Store" mode in RAR, for instance. The idea behing this "uncompressed FLAC" is to give the FLAC tagging abilities to people who don't want to compress their files. But as WAV is lacking in the tags department (some workarounds, but most players are not compatible with those), dbPowerAmp uses the FLAC structure and inputs the uncom...

Greetings, I was wondering if somebody is working on a solution to the NAT/SIP-issues? It seems to me that the problem has been identified, is that correct? Just hoping that someone with more skills will provide us with a solution sooner or later... Regards, Stig -------------- next part -------------- An HTML attachment was scrubbed... URL:

...+Clang double-patch is not worth the trouble for the noise. > > For both of them, we need an external server, as there's no way to > update a repository's property from another. That makes it fragile, and that’s why I disagree with your “90% done” assessment. What if the service behing the hook is down for a few days? Who will maintain it? — Mehdi > > Multiple commits eventually getting into a single umbrella revision > can be innocuous for development, but they can make controlling the > version for releases a bit more complicated. Though, it would also > ha...

Hi, We have a 256 kbits/s (kilobits per second) link to the internet. it is a router running Linux that belongs to our ISP. They have given us 8 internet ips. (i.e- subnet is 255.255.255.248). one has been given to this router. I have given another internet ip to the firewall running CentOS 4.5. iptables is running on it. And also, I have installed iproute2 pkg as well. pls see below for

Hi all, Anyone got a link or the file to Mr. Guy's notes from his speech at Astricon.. I looked for them or relevant info he showed ad FWD's webpage but have not been able to find them. He had a working FWD registered thru Asterisk behing a NAT and a couple of samples of their new scripts, talking weblog (from Mr. Pulver) and other cool new features. If anyone got these please send em over or post.. maybe they should be added to the the wiki. Thanks! Lethol

...ing dropped all over the place. The one thing I do not understand is that at least 30 phones on the internal network went to "No Service". Since they are on the same network segment and on the same subnet I do not see why the Internet connection sould affect them. The asterisk server is behing NAT and we use externip=201.161.XXX.XXX for outside sip phones and providers. Any recommendations? -- ?Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrub...

Hello everybody. I have a problem with an integration between an Asterisk (1.4.24.1) on FreeBSD 7.0 and a Shoretel 7.5 server. To make a very long story short, when someone behind asterisk call an extension behing shoretel everything work as expected. When someone behing the shoretel server call someone behind asterisk the first 10 seconds of the call seems ok but then the line is dropped with asterisk reporting something like this: [Apr 20 18:01:46] WARNING[17362] chan_sip.c: Maximum retries exceeded on...

...ve about 3 E1 worth of g729 traffic going thru. So I'm planning to use IAX2 trunking to reduce bandwith requirement and squeeze out each and every bit of this (expensive) bandwith. I've set up two boxes (debian etch), one in a remote data center (which has plenty of bandwith) and one behing the SDSL link. To make things consistent I've installed the same kernel, latest stable zapata + asterisk on both ends. I've done some tests with about 1/2 E1 (15 channels) worth of calls and so far it's been working good - and the call statistics (ASR, ACD, PDD) are roughtly th...

> That makes it fragile, and that’s why I disagree with your “90% done” assessment. > What if the service behing the hook is down for a few days? In the long-term view, a pretty trivial catch-up script ought to be able to synthesize a sane history after any amount of down-time. People could even run it locally for their bisecting needs if it was that important to them. In the short term, I don't think i...

...C headers because that would pollute the global namespace and users might prefer to import the missing symbols (`size_t` and `FILE`) selectively. Also that wouldn't help static analysis within the header files since the analysers use the C path. So I have guarded inclusion of standard C headers behing a `__cplusplus` check. If that makes sense, would R core consider applying the attached patch to the R sources? Best, Lionel

...y a text-based sniffer (no X available in the Asterisk live CD I'm using), I'm interested :-) Thank you. PS: FWIW, extension 203 (softphone) and 204 (IP phone) are both located on the same network and behind a NAT router, and both connect out to an Asterisk server somewhere on the Net behing its own NAT router: slast*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 204/204 82.237.x.y D 5060 UNREACHABLE 203/203 82.237.x.y D N 46838 OK (925 ms)

Hi guys, I need your help ... I have a couple of DIDs that reach my Asterisk box .... But I'd like to set my DIDs automatically via Macro or other routine based on the number called by my agent ... Ex: My agent called 954-111-1111 ... So I'd like to set the callerid as 954-222-2222 (That is my DID) Thanks in advance, Marcelo

...tart''", "/usr/lib/ruby/gems/1.8/gems/rails-0.10.0/lib/webrick_server.rb:20:in `dispatch''", "/usr/local/rss_feed/script/server:51"] [2005-03-02 15:17:14] ERROR `/rss_feed/index'' not found. The rails server is listening on port 3082. It''s behing an apache proxy with this config: ProxyPass / http://127.0.0.1:3082/ ProxyPassReverse / http://127.0.0.1:3082/ I access the url http://servername/rss_feed/index Running on debian, with all ruby1.8 packages installed. Any idea? Thanks

...em with Perl either, or Python. Tcl is much less well-known. > > Note that I don't particularly like any of these languages but I'm trying > not to let personal preference get in the way. :) Neither I do. This sub-thread started with a discussion about the feasibility of leaving behing DejaGNU and using `bash' instead or: > > If a language conceals the differences between POSIX and Microsoftian C > > for interprocess control, hides badly non-POSIX filepaths from me, and > > has a reasonable track record of forward compatibility, and has a > > reasonab...

...at would pollute the global namespace > and users might prefer to import the missing symbols (`size_t` and > `FILE`) selectively. Also that wouldn't help static analysis within > the header files since the analysers use the C path. So I have guarded > inclusion of standard C headers behing a `__cplusplus` check. > > If that makes sense, would R core consider applying the attached patch > to the R sources? > > Best, > Lionel >

On 30 June 2016 at 05:14, Tim Northover <t.p.northover at gmail.com> wrote: >> That makes it fragile, and that’s why I disagree with your “90% done” assessment. >> What if the service behing the hook is down for a few days? > > In the long-term view, a pretty trivial catch-up script ought to be > able to synthesize a sane history after any amount of down-time. > People could even run it locally for their bisecting needs if it was > that important to them. Yup. If the sc...

...now this subject have been The most written subject about VoIP Right... but I just want to make clear, just one time ! If Asterisk is on a Public IP Address and a softphone behind the nat, sip.conf must contains for this phone: nat=yes .... Now if I want to configure my sipphone (X-Lite) placing behing the NAT, it must have in "Domain/Realm" the external IP address? If Asterisk is behind the NAT, sip.conf must have in [globals] externip = <External IP address> localnet = < Internal NETWORK address> localmask = <mask of localnet> Is that right? If someone have clea...

...es are no longer ringing (busy) after 1 call rings & is unanswered, prior to sending a 2nd call. In other words, I think that what we are currently experiencing is this: Incoming call gets routed to our queue. It rings all phones. In the meantime, a 2nd caller gets routed to our queue (in line behing the first caller that is currently ringing our phones). One queue agent answers the first phone call in the queue. Asterisk immediately starts ringing all queue agent extensions again with the 2nd caller. However, most of the agents extensions are reporting busy, and so their phones don't ring....