similar to: Asterisk and Cisco AS53xx/54xx Access Server Platform

Dear all, Did anyone successfully test T38 fax pass thru to Cisco as53xx? We've tried 1.2.4 with latest patch and latest svn trunk and T38 patch but still not work. Reinvites from Cisco are correctly passed back to the originating gateway, but fax never able to connect. Cisco IOS 12.3.x configuration voice service voip fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback

Do somebody using CISCO AS5350 with Asterisk? Which protocol do you using: H323, MGCP, SIP? This direction: [12sp->Asterisk->h323->as5350->isdnPSTN] is ok But reverse: [isdnPSTN->as5350->h323->Asterisk->12sp] cannot hear 12sp, but 12sp hear PSTN (codec g711u) -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I am relatively new to R and I am working through the code that is provided in the book Numerical Ecology with R and I have run across an error message that I can't seem to figure out. I am using the vegan, ade4, gclus and cluster packages. The code is as follows: # Ordered community table # Species are ordered by their weighted averages on site scores or <- vegemite(spe,

Has anyone succesfully integrated * with a cisco voice gateway ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031214/3b1ba7b3/attachment.htm

Hi all, >From a list of strings, I desire to filter out the followings: 1. Digits at the beginning of the strings 2. Character "SPE" following the digits (if it exists) 3. Any characters followed by hyphen The following produces the desired result, but would like to know whether this can be done more efficiently. Any suggestions would be much appreciated. dat <- c("2148

Dear list, I want to calculate SPE (squared prediction error) in x-space, can someone help? Here are my codes: fit.pls<- plsr(Y~X,data=DAT,ncomp=3,scale=T,method='oscorespls',validation="CV",x= T) actual<-fit.pls$model$X pred<-fit.pls$scores %*% t(fit.pls$loadings) SPE.x<-rowSums((actual-pred)^2) Am I missing something here? Thanks in advance. Stella Sim

Hi all. I have a strange problem, I've got a AS5350 hooked up to a telco using two trunked E1's The 5350 should only act as a GW to a sipproxyserver. THe thing is it seems to be only oneway audio? There are no firewall at all, and the audio still only get one-way When I call from pstn --> as5350 --> sip-sip-phone I can here the sip-phone ,, but the sipphone cannot her the

Here is my Configuration PSTN -> Cisco AS5350 -> SIP -> ASTERISK -> SIP -> ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong?

Hi, I am a new R user and am trying to construct a palaeoenvironmental transfer function (weighted averaging method) using the package rioja. I've managed to insert the two matrices (the species abundance and the environmental data) and have assigned them to the y and x values respectively. When I try and enter the 'WA' function though, I get an 'Error in FUN' message (see

I'm stuck running a old copy of asterisk because something strange is going on in later versions of the CVS.. When I call in from a PSTN into my cisco 2610XM gateway which then routes the call to my asterisk box via sip.. Asterisk can no longer process DTMF tones generated by the calling party. This affects DISA, prompts and menus.. Has anyone else had this problem?? and use.. I DO have

Hello that might would be an easy question for someone, but im in doubt Is there any possibility to pass a call from one asterisk to another and then to ZAP channel. For instance I have "A" asterisk with numbering 45670 "B" asterisk with numbering 45680 second asterisk has TE110P card with single PRI port connected to Siemens EWSD. When I originate call from asterisk

Dear R-gurus, Just supose I have a dara.frame that looks like myDF<-read.table(stdin(),head=T,sep=",") codID,namesp,k1,k2,k3,k4 1,spA,2,5,6,3 2,spB,4,5,4,6 3,spC,2,1,5,6 4,spC,5,4,3,2 5,spD,1,2,3,4 6,spE,2,4,3,1 I need to update the columns k1-k4 with the namesp, but considering the math between Kx and codID. My desired output must looks like: codID,namesp,k1,k2,k3,k4

Hi, I am currently interconnecting to a PRI using a Cisco AS5350. I'd like to be able to dial specific numbers out by a specific isdn channel, so for e.g. if I dial 999 01 12341234 it should send 12341234 out via isdn channel one from the Cisco AS5350. If somebody would be able to guide on this, it would be appreciated. Regards, Sahil Gupta VoiceValley

On Thu, 30 Oct 1997, Nathan Neulinger wrote: > > you'll need to do encrypted passwords for your users. what version of > > unix are you using? have you looked into PAMs? (plug-in authentication > > modules) > > The problem is, we'll never have cleartext passwords for the user. > Maintaining a separate password database is unnaceptable. If we wanted to

Dear all, Is there any function for calculating confidence limits for coefficients in an lm() object? I know of the confint() function in the MASS library working very well on my binomial GLMs and I have tried it (using glm () , family=gaussian) but it gives NAs according to below. Does the confint() function not accept gaussian GLMs? Could there be convergence problems in the GLM? Note the

Hello, I am using vegan to do an NMDS plot and I would like to suppress the labels for the loading vectors. Is this possible? Alternatively, how can I avoid overlap? Many thanks for the help. Example code: #perform NMDS using metaMDS() function spe.nmds<-metaMDS(data, distance='bray',k=2 , engine = "isoMDS", autotransform=F, trymax=1000) #calculate the loading (i.e.,

Hi Jean-Marc, We needed to use a release version of the SDK because this part of production code for our Sony implementations. I'm not sure how stable that branch is. Is the encoding and decoding already stable? In other words, what is not stable about that branch? Thanks, --------------------------- Saad Nader Middleware Engineer Powered By Gamespy, IGN -----Original Message----- From:

Guys. I need some advice on some h323 issues. I need to test connectivity from Asterisk to a Cisco AS5300 that has PSTN lines and to cisco 3600 voip routers. H323 needs to be used here but I was wondering if anybody has linked Asterisk to these Cisco routers before? Thank you for any pointers.

Hi, I have a custom lowering operation on ISD::BITCAST for the PowerPC/SPE target, to convert 'f64 bitcast (i64 build_pair i32, i32)' into a 'f64 BUILD_SPE64 i32, i32' node, which can be seen at https://reviews.llvm.org/D54583. However, when building compiler-rt's lib/builtins/divdc3.c an assertion is triggered that BUILD_PAIR is not legal on line 24. There should be no

Could someone give me a 10,000 foot view of what the differences are between Ser and Asterisk? I'd like to implement one or the other handle a small number of local ip phones, tie a couple of asterisk (or ser) machines together across the Internet, implement a couple of FX gateways (to handle incoming pstn calls, and for outgoing pstn calls), and use features mostly common to pbx's. No