similar to: Polycom - missed calls dial back

This is not an Asterisk specific question but doesn't anyone know if you can automatically prepend a 9 on the call lists so clients can return dial without having to repunch in the number? If you go to directories now it just shows the number without a 9 (obviously). Maybe on the Asterisk side?? Bill -------------- next part -------------- An HTML attachment was scrubbed... URL:

192.168.1.107 199 6bd3fb49505 00102/00000 ulaw No Tx: ACK 192.168.0.100 110 5c5a4953-65 00101/00005 ulaw Yes Rx: ACK Those channels are stuck talking to each other. The phones are disconnected yet that connection remains. I can clear w/ a restart obviously, but is there any way to tear down a call like that from the CLI? Bill -------------- next

I have a situation that has repeated itself a few times. Someone calls into Asterisk and is connected with a voice extension. At some point during the call, the log shows "chan_zap.c: DTMF digit: f on Zap/2-1". At this point, the call is redirected to receive a fax and the Asterisk voice extension is hung up. The users report that there were no noticable tones heard just before the

I have read that 1.4 has shared line appearances, which I assume will work with Polycom phones. Has anyone configured this and verified it working? I was going to start playing around with it but wanted to see if anyone else has tackled it yet. Bill -------------- next part -------------- An HTML attachment was scrubbed... URL:

CVS Head from 07/07/2005 I'm trying to make an IP-501 auto answer a call. exten => 301,1,SetVar(_ALERT_INFO="Ring_Ans") exten => 301,2,SetVar(ALERT_INFO="Ring_Ans") # Tried both combinations exten => 301,3,Dial(SIP/5001,15) exten => 301,4,Hangup Sip.cfg for Polycom phone <alertInfo voIpProt.SIP.alertInfo.2.value="Ring_Ans"

2 Asterisk servers 1.2.12.1 Connected via IAX2, same switch, GigE, no packet loss, etc 1 with a Sangoma A101 for a PRI to the PSTN Ulaw QoS enabled NAT for the registered ATA boxes, no nat between the * servers Faxing inbound: Call from PRI hits the first Asterisk server Then talks to the 2nd via IAX2 NVFaxDetect receives the fax, converts to PDF and emails it out Works great!

I've been playing with Asterisk for a few weeks and it's working great. I have a question about getting multi-line receptionist phones working. I was thinking about getting one of these expansion ports: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0 9186a008008883d.html What are people using for receptionist phones that show all the extensions in

A follow up (late better than never) Finally had time to sit down and look at this sip.cfg <keys key.scrolling.timeout="1" key.IP_500.31.function.prim="BuddyStatus"/> This turns the Services key which I never use on a 501 into the Buddy Status. It even works while on a call. One touch! Bill ________________________________ From: Bill Gibbs

I know this is not asterisk specific but we all know this group is used for Polycom issues as well... I have hints working ok on Asterisk. However the Polycom phone will only show the buddies key if there is not a call. This defeats the purpose of using the buddies to see if you can transfer a call to another extension (using the Buddy key to see if they are on the phone). Polycom sip

I am testing out hylafax and iaxmodem. Submitting jobs to the hylafax server and having it then dial using iaxmodem is working fine. Hylafax server is talking to my Asterisk box that has a Sangoma A101 in it via iaxmodem via an IAX channel using ulaw. A call coming into a certain test DID comes into the Sangoma A101 then it goes to another box via IAX ulaw that uses the rxfax app to

Ok so I added to my Freepbx config running Asterisk 1.2.4 in extensions_custom.conf ; intercom exten => _7XXX,1,SIPAddHeader(Alert-Info: Auto Answer) exten => _7XXX,2,Dial(SIP/${EXTEN:1},12,Tt) and configured my Polycoms via this page http://www.voip-info.org/wiki/view/Polycom+auto-answer+config for auto answer and that works fine if I dial 7 then the 3 digit extension. No

I know the Polycoms work with NAT, but you have to specify the public IP. Is there anyway for it to discover the external IP automatically? I like the phones (been playing with a 301) but for some of our clients who have a dynamic IP (and no hope of getting a static ie cable or residential DSL) I'd be afraid to use them since you have to specify the IP. What about the Cisco phones?

Hello, thank you very much for your previous answers about the C++ code. I am interested in the application of the Gibbs Sampler in the IRT models, so in the function MCMCirt1d and MCMCirtkd. I've found the C++ source codes, as you suggested, but I cannot find anything about the Gibbs Sampler. All the files are for the Metropolis algorithm. Maybe I am not able to read them very well, by the

via google, I found the reference regarding digit maps for the Polycom phones: http://lists.digium.com/pipermail/asterisk-users/2005-January/082884.html But I don't see any instruction for prepending digits to the number dialed. Does anyone know how to prepend a digit to the number dialed (from the Polycom side, not Asterisk)? I can do this pretty easily on a Sipura. i.e. Say I want to

I suspect it''s probably not a good idea but I was wondering if someone could clarify the details. I have 4 250G SATA(150) disks and 1 250G PATA(133) disk. Would it cause problems if I created a raidz1 pool across all 5 drives? I know the PATA drive is slower so would it slow the access across the whole pool or just when accessing that disk? Thanks for your input. - Chris

I have been using the rtmvt function in the {tmvtnorm} package i'm getting the warning: "Acceptance rate is very low and rejection sampling becomes inefficient. Consider using Gibbs sampling." but i AM specifying the gibbs algorithm!!: rtmvt(M, mean=q[,,i,j], sigma=((u[i,j] + nu[i])/(p+nu[i]))*delta[,,i], df=ceiling(nu[i]+p), lower=c(0,0), algorithm="gibbs") Any

An embedded and charset-unspecified text was scrubbed... Name: n?o dispon?vel URL: <https://stat.ethz.ch/pipermail/r-help/attachments/20110405/06486a8a/attachment.pl>

Hello R community, following is my code and it shows error, can some one fix this error and explain why this occurs? gibbs <-function(m,n, theta = 0, lambda = 1){ alpha <- 1.5 beta <- 1.5 gamma <- 1.5 x<- array(0,c(m+1, 3)) x[1,1] <- theta x[1,2] <- lambda x[1,3]<- n for(t in 2:m+1){ x[t,1] <- rbinom(x[t-1,3], 1, x[t-1,1])

I have downloaded the Umacs (Universal Markov chain sampler) and submitted the following sample code from Kerman and Gelman.   s <-Sampler( J=8, sigma.y  =c(15,10,16,11,9,11,10,18),           y  =c(28, 8,-3,7,-1,1,18,12),      theta =Gibbs(theta.update,theta.init),           V =Gibbs(V.update,mu.init),         mu =Gibbs(mu.update,mu.init),         tau =Gibbs(tau.update,tau.init),       

Hi, I want to use R_qsort_int_I() in my C function, but getting the following error. It looks like there is a conflict between Rmath.h, which I use to generate random numbers, and R_ext/Boolean.h I would appreciate any help to fix this problem. gcc -ansi -g -o Gibbs gibbs.c subroutines.o rand.o vector.o -lm -lRmath -llapack -lblas -lfrtbegin -lg2c -lm -shared-libgcc In file included from