similar to: Asterisk supports SIP-T?

Hi List. It is possible to travel over the X.25 protocol on Asterisk SIP? -- Atenciosamente Daviramos Roussenq Fortunato -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100928/7d6ca5fd/attachment.htm

Hi List. I have a small problem in using the transfer key transfer of IP Phone in Asterisk 1.6, I think I spend some detail in the configuration but can not find. What happens is, when I do a transfer using the Transfer button, the phone, does not play the music on hold, which is waiting on the phone is silent, and I have the same settings on a 1.4 server, and the music plays correctly when

Boa Tarde Lista. I'm having problems in tramiss?o a fax using T.38. My scenario is: Asterisk 1.6.0.5 2 ATA of Intelbras 2210. ReceiveFAX in the asterisk. Unable to fax when it is a ATA to another user on the Asterisk means, if I directly between the ATA works perfectly, is a step to the ATA ReceiveFAX of Asterisk works perfect, but if I try to pass between two Branches

Hello, Is there like the kruskal wallis test in relation to ANOVA (no restrictions on normallity and variance homogenity) something (in R) for MANOVA? thanks -- Dr.Nicolaas Busscher Universit?t GH Kassel Nordbahnhofstrasse: 1a, D-37213 Witzenhausen Phone: 0049-(0)5542-98-1715, Fax: 0049-(0)5542-98-1713

Hello, I am working with zero inflated models for a current project and I am getting wildly different results from R's zeroinfl(y ~ x, dist="negbin") command and Stata's zinb command. Does anyone know why this may be? I find it odd considering that zeroinfl(y ~ x, dist="poisson") gives identical to output to Stata's zip function. Thanks, --david [[alternative

I am aware there is a SIP over TCP patch. Will this ever become part of a release, if so are there any timelines? Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080213/1f05009a/attachment.htm

Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows RTP to use different IP as SIP ip. Is there any way to configure it? GUI or CLI? or , will we support it in future? Thanks. -- Rgds, -- Rgds, Hans Yin Web: homeofhans.homeip.net Email: hansyin at gmail.com MSN: hansyin at hotmail.com Skype: hans_yin_vancouver

I am working on a project that requires shared extension. Where shared line looks at the status of a line/trunk, shared extension would look at a series of channels as the same "extension". The users would like to add destination channels on the fly, to provide roaming extensions, but maintaining fixed channels as well. If a call comes in on an extension, the system needs to honor the

Dear Andrea. Your request seems vague. And also: be careful, even if some of the people reading the list understand italian you are expected to post your requests in english. In short: read the FAQ and at least "An introduction to R". But you seem to be lucky, on the CRAN you will find a document by Vito Ricci, in italian, about time series analysis:

Hi, I'm trying to implement a Voice Drop service within Asterisk dial-plan. The service is supposed to work as following: 1. A initiates a call to B 2. The call is answered by B's answering machine 3. A hears the answering machine's greeting and the recording beep 4. A speaks a few words into the recording to personalize the message 5. A presses some DTMF keys (say, '##') to

What I would like to do is have two identical * servers which accept registrations of sip extensions 4000-4999. If I define a rrDNS or LinuxHA then I should have load-balanced registrations. However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's

I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead? Doug. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML

Hi all, I just upgraded to Asterisk 1.4.18 a few days ago and I don't use Broadvoice TOO often, however I have a Vermont number with them and so my mother in law calls it to talk to my wife once in a while, so that's why it took me so long to notice it wasn't working. Anyway, when she calls she gets a busy signal (as I've tested when calling it from my cell). When I enable

Vuoi Davvero Guadagnare con Internet? Bene, salva su disco questa pagina per averla a portata di mano anche se il tuo PC non ? connesso a Internet, poi copia quanto segue in Word o in Blocco Note e stampalo, cos? lo potrai leggere con pi? attenzione. Questo Sistema ? diverso da tutti gli altri, quindi non essere precipitoso nel valutarlo senza averlo compreso a fondo, ma ti assicuro che

Hi all, I am a system administrator new to Ruby and Rails. I have developed a simple web-based tallysheet program with the following tools: view: phpFormGenerator controller: Bash, Gawk, and other Linux utilities model: Flat file (Just playing with what i''ve learned about Rails) :-) Here''s a sample snapshot of the application:

Hi List. I'm having problems with Asterisk 1.6 + DAHDI 2.1.0.3 PlayBack does not ring, is still in command, and not later in the following context. Disabling the dahdi operates normally. I'm using dahdi_dummy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090218/0029d92d/attachment.htm

Hi List. I have an IP Phone when I'm on a call and tightness in the Transfer button, it opens a new channel for me to make a new connection. But the extension is on hold is that mute the music without the wait. How should I proceed to solve my problem. I'm using asterisk 1.6.0.5 -------------- next part -------------- An HTML attachment was scrubbed... URL: