similar to: sip-user status

Hello For customers who need a small IP PBX to handle up to four ISDN lines (in France, so I guess that means EuroISDN) instead of a PC + Asterisk and an ISDN gateway box, has someone already played with the Atcom IP-4B? www.atcom.cn/IP-BRIM.html Any feedback appreciated.

Retrying, falling of the list some how :-( -------- Original Message -------- Subject: feeling n00b again Date: 2018-08-20 09:51 From: asterisk at a-domani.nl To: asterisk-users at lists.digium.com Hi all, Long time ago, I followed a Asterisk training, and both at work and at home, was able to deploy Asterisk, make all sorts of internal call (hard/soft voip-phones, incoming/outgoing,

Hello, I'm wondering if it is possible to ring X number of extensions simultaneously, and each answered call can be handled with some code. I can do a huntgroup-esque way of dialling, but I want all the dialled numbers to be picked up. I hope this makes sense.. If not please say.. Many thanks! Andy -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've been looking at the list at: http://www.voip-info.org/wiki/view/chan_mobile But when googling of any of the "known working" devices, there ain't any for sale anymore, probably replaced by more recent types. So, anyone around here who bought recently an BT-dongle that is working with asterisk? hw

Good afternoon/morning readers. This is the first time I am trying to run some Bayesian computation in R, and am experiencing a few problems. I am working on a Poisson model for cancer rates which has a conjugate Gamma prior. 1) The first question is precisely how I work out the parameters. #Suppose I assign values to theta with *seq()* *theta<-seq(0,1,len=500)* #Then I try out the

Hi, I''m new in developint rails (but I love rails). I would like to react on changing a field in the view (no model field) from a selection box to show the number of rows (which I used in the controller). In the controller I read the params-hash: ... @rowsperpage = params[''rowsperpage''].to_i ... In the view I have the field, where I like to react directly. ...

Ok, here's the challenge: I would like to be able to find, match - and then react - upon prompts that are presented by the outbound/remote side of a call. Think mobile phone and "This user is temporarily unavailable". Collecting a limited number of known prompt snippets should not be a problem, but how would you then detect their presence in a longer recording (or live audio

Hi all, Been reading about chan_motif / chan_xmpp in the wiki's for 1.8, 10 and 11 version of asterisk. In each example i got the impression that the asterisk server is registering on a XMPP server as a single user with the credentials as specified in jabber.conf. Instead of a single xmpp-user, could that also be multiple users? For instance, for each sip-user an xmpp-user? When i skim

Greetings- I'm about to dive into the process of virtualizing some of my Asterisk (primarily 1.4.x) infrastructure. In the past, when looking at virt solutions, the primary issue preventing me from moving was the lack of proper timing. We do not need it for MeetMe but rather for IAX2 trunking. I'd like to use either OpenVZ or KVM, but each seem to have independent "issues" that

Hi all, Perhaps someone has dealt with it before. I want to activate a bunch of my own scripts after someone has registred om my asterisk, or when his cient has de-registerded. have been skimming through AGI and AMI, and seen a lot of nice features, but not the (de-)registering events. Kind regards, Hans

Just curious, I noticed that with SetLanguage() you can change it into a lot of other languages. Yes one can record them easily enough with "record", but don't like to re-invent the wheel.. Browsed through a lot of google-pages but failed to find any other languages (except for FR and ES on the digium site) Any pointers for german, dutch, greek, italian, .... prompts? Hans --

Hi all, Perhaps someone can give me a hint i the right direction... Sometimes dtmf is recognized, sometimes not. I'm using 1.2.19 asterisk with misdn for my hfc card. When i got in incoming sip-call, dtmf is recognized, When i phone my self (isdn-phone or gsm-phone) no problem with dtmf When SOME (not all) people phone me (isdn-incoming) DTMF is not recognized. How come? Either it works

Morning! This may be the "day of the dead" in some regions, but we expect the usual lively discussion today at 9AM PDT, 11 Central, 12 Noon EDT, 4PM UK and Portugal, 5PM Paris, $deity-forsaken hour down under. This Sunday, I believe the USA falls back to Standard time. Future VUC are still at 12 Noon EST. Info site: http://voipusersconference.org PSTN (724) 444-7444 enter 22622# 1#

To all you great C programmers out there who have been looking for a chance to take on a really cool open source project in video: liboggplay is a library that vastly simplifies the decoding and playback of Ogg encapsulated audio-visual content for programmers. It does everything apart from the actual display of audio and video and has thus been selected as the thinnest library to provide

To all you great C programmers out there who have been looking for a chance to take on a really cool open source project in video: liboggplay is a library that vastly simplifies the decoding and playback of Ogg encapsulated audio-visual content for programmers. It does everything apart from the actual display of audio and video and has thus been selected as the thinnest library to provide

Dear All, Thank you for your comments and suggestions about my term 'VoIP Farm'. It was done in jest, and I'm glad to see how the community reacted, with off list emails with some with advice, some with warnings, some with funny comments, and some sending me a promise of $30 for usage of the word for three years! To lay it to rest, I know I can't trademark VoIP Farm, but thanks

The test relies on the timeout program. Also, since the nbdkit_nanosleep code relies on the Linux extension POLLRDHUP to detect early client closure, we may have to relax that part of the test when porting to platforms that lack ppoll/POLLRDHUP. (That is, while we should still be able to let a signal to the server shut down nbdkit quickly, it's harder to let a client close()ing its end cause

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040728/3306e606/attachment.htm -------------- next part -------------- ? Hi, I'm dialling 1234 in the softphone or Grandstream Phone and without disconnecting the phone i want to dial "10" after dialling "1234". Is it possible to do? Regards shan

I have read and read the dovecot docs on maildir folders and think that this is what I want. However, the MUA programs that I use, SylpheedClaws, Evolution, Mozilla seem to react differently to how I setup the folders. Is there a good FAQ on this subject somewhere? I think that I want to setup a virtual domain and am looking for documents. Alex

anyone else finding it difficult to connect to vorbis.com? seems the machine is down. connecting times out. does not react to pings. -- Alexander Wörndle NoAlexanderSpam@Woerndle.Net GnuPG-Key unter http://www.woerndle.net verfügbar. --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to